manual_src~dfsg/0000755000000000000000000000000012270515241012741 5ustar rootrootmanual_src~dfsg/command/0000755000000000000000000000000012262561504014363 5ustar rootrootmanual_src~dfsg/command/flags.xml0000644000000000000000000016576412262561504016224 0ustar rootroot Command Line Flags Csound command line csound Csound command. Description The csound command executes Csound. Syntax csound orchname scorename csound csdfilename Csound command line flags Listed below are the command line flags available in Csound6 in alphabetical order. Various platform implementations may not react the same way to different flags! You can view the command line flags organized by category in Command-line Flags (by Category). The command line arguments are of 2 types: flags arguments (beginning with a -,-- or -+), and name arguments (such as filenames). Certain flag arguments take a following name or numeric argument. Flags that start with -- and -+ usually take an argument themselves using =. Command-line Flags -@ FILE-@ Provide an extended command-line in file FILE -3-3, --format=24bit--format=24bit Use 24-bit audio samples. -8-8, --format=uchar--format=uchar Use 8-bit unsigned character audio samples. --format=type--format Set the audio file output format to one of the formats available in libsndfile. At present the list is aiff, au, avr, caf, flac, htk, ircam, mat4, mat5, MPC, nist, ogg, paf, pvf, raw, sd2, sds, svx, voc, w64, W64, wav, wavex, WVE, xi. Can also be used as --format=type:format or --format=format:type to set both the file type (wav, aiff, etc.) and sample format (short, long, float, etc.) at the same time. -A-A, --aiff--aiff, --format=aiff--format=aiff Write an AIFF format soundfile. Use with the -c, -s, -l, or -f flags. -a-a, --format=alaw--format=alaw Use a-law audio samples. -B NUM-B, --hardwarebufsamps=NUM--hardwarebufsamps Number of audio sample-frames held in the DAC hardware buffer. This is a threshold on which software audio I/O (above) will wait before returning. A small number reduces audio I/O delay; but the value is often hardware limited, and small values will risk data lates. In the case of portaudio output (the default real-time output), the -B parameter (more precisely, -B / sr) is passed as the "suggested latency" value. Other than that, Csound has no control over how PortAudio interprets the parameter. The default is 1024 on Linux, 4096 on Mac OS X and 16384 on Windows. -b NUM-b, --iobufsamps=NUM--iobufsamps Number of audio sample-frames per sound i/o software buffer. Large is efficient, but small will reduce audio I/O delay and improve the accuracy of the timing of real time events. The default is 256 on Linux, 1024 on MacOS X, and 4096 on Windows. In real-time performance, Csound waits on audio I/O on NUM boundaries. It also processes audio (and polls for other input like MIDI) on orchestra ksmps boundaries. The two can be made synchronous. For convenience, if NUM is negative, the effective value is ksmps * -NUM (audio synchronous with k-period boundaries). With NUM small (e.g. 1) polling is then frequent and also locked to fixed DAC sample boundaries. Note: if both -iadc and -odac are used at the same time (full duplex real time audio), the -b option should be set to an integer multiple of ksmps. -C-C, --cscore--cscore Use Cscore processing of the scorefile. -c-c, --format=schar--format=schar Use 8-bit signed character audio samples. --csd-line-nums=NUM--csd-line-nums Determines how line numbers are counted and displayed for error messages when processing a Csound Unified Document file (.csd). This flag has no effect if separate orchestra and score files are used. (Csound 5.08 and later). 0 = line numbers are relative to the beginning of the orchestra or score sections of the CSD 1 = line numbers are relative to the beginning of the CSD file. This is the default as of Csound 5.08. -D-D, --defer-gen1--defer-gen1 Defer GEN01 soundfile loads until performance time. -d-d, --nodisplays--nodisplays Suppress all displays. See -O if you want to save the log to a file. -d--daemon Run in daemon mode: do not exit if CSD/orchestra is not given, is empty or does not compile. --displays--displays Enables displays, reverting the effect of any previous -d flag. --default-paths--no-default-paths Reenables adding of directory of CSD/ORC/SCO to search paths, if it has been disabled by a previous --no-default-paths (e.g. in .csoundrc). --env:NAME=VALUE--env:NAME=VALUE Set environment variable NAME to VALUE. Note: not all environment variables can be set this way, because some are read before parsing the command line. INCDIR, SADIR, SFDIR, and SSDIR are known to work. --env:NAME+=VALUE--env:NAME+=VALUE Append VALUE to ';' separated list of search paths in environment variable NAME (should be INCDIR, SADIR, SFDIR, or SSDIR). If a file is found in multiple directories, the last will be used. --expression-opt--expression-opt Since Csound 5. Turns on some optimizations in expressions: Redundant assignment operations are eliminated whenever possible. This means that for example this line a1 = a2 + a3 will compile as a1 Add a2, a3 instead of #a0 Add a2, a3 a1 = #a0 saving a temporary variable and an opcode call. Less opcode calls result in reduced CPU usage (an average orchestra may compile about 10% faster with --expression-opt, but it depends largely on how many expressions are used, what the control rate is (see also below), etc.; thus, the difference may be less, but also much more). number of a- and k-rate temporary variables is significantly reduced. This expression (a1 + a2 + a3 + a4) will compile as #a0 Add a1, a2 #a0 Add #a0, a3 #a0 Add #a0, a4 ; (the result is in #a0) instead of #a0 Add a1, a2 #a1 Add #a0, a3 #a2 Add #a1, a4 ; (the result is in #a2) The advantages of less temporary variables are: less cache memory is used, which may improve performance of orchestras with many a-rate expressions and a low control rate (e.g. ksmps = 100) large orchestras may load faster due to less different identifier names index overflow errors (i.e. when messages like this Case2: indx=-56004 (ffff253c); (short)indx = 9532 (253c) are printed and odd behavior or a Csound crash occurs) may be fixed, because such errors are triggered by too many different (especially a-rate) variable names in a single instrument. Note that this optimization (due to technical reasons) is not performed on i-rate temporary variables. Warning When --expression-opt is turned on, it is not allowed to use the i() function with an expression argument, and relying on the value of k-rate expressions at i-time is unsafe. -F FILE-F, --midifile=FILE--midifile Read MIDI events from MIDI file FILE. The file should have only one track in Csound versions 4.xx and earlier; this limitation is removed in Csound 5.00. -f-f, --format=float--format=float Use single-format float audio samples (not playable on some systems, but can be read by -i, soundin and GEN01 -G-G, --postscriptdisplay--postscriptdisplay Suppress graphics, use PostScript displays instead. -g-g, --asciidisplay--asciidisplay Suppress graphics, use ASCII displays instead. -H#-H, --heartbeat=NUM--heartbeat Print a heartbeat after each soundfile buffer write: no NUM, a rotating bar. NUM = 1, a rotating bar. NUM = 2, a dot (.) NUM = 3, filesize in seconds. NUM = 4, sound a bell. -h-h, --noheader--noheader No header on output soundfile. Don't write a file header, just binary samples. --help--help Display on-line help message. -I-I, --i-only--i-only i-time only. Allocate and initialize all instruments as per the score, but skip all p-time processing (no k-signals or a-signals, and thus no amplitudes and no sound). Provides a fast validity check of the score pfields and orchestra i-variables. This option is exclusive of the --syntax-check-only flag. -i FILE-i, --input=FILE--input Input soundfile name. If not a full pathname, the file will be sought first in the current directory, then in that given by the environment variable SSDIR (if defined), then by SFDIR. The name stdin will cause audio to be read from standard input. The name devaudio or adc will request sound from the host audio input device. It is possible to select a device number by appending an integer value in the range 0 to 1023, or a device name separated by a : character (e.g. -iadc3, -iadc:hw:1,1). It depends on the host audio interface whether a device number or a name should be used. In the first case, an out of range number usually results in an error and listing the valid device numbers. The audio coming in using -i can be received using opcodes like inch. -+id_artist=string-+id_artist (max. length = 200 characters) Artist tag in output soundfile (no spaces) -+id_comment=string-+id_comment (max. length = 200 characters) Comment tag in output soundfile (no spaces) -+id_copyright=string-+id_copyright (max. length = 200 characters) Copyright tag in output soundfile (no spaces) -+id_date=string-+id_date (max. length = 200 characters) Date tag in output soundfile (no spaces) -+id_software=string-+id_software (max. length = 200 characters) Software tag in output soundfile (no spaces) -+id_title=string-+id_title (max. length = 200 characters) Title tag in output soundfile (no spaces) -+ignore_csopts=integer-+ignore_csopts= If set to 1, Csound will ignore all options specified in the csd file's CsOptions section. See Unified File Format for Orchestras and Scores. -+input_stream=string-+input_stream Pulseaudio input stream name. -J-J, --ircam--ircam, --format=ircam--format=ircam Write an IRCAM format soundfile. -+jack_client=[client_name]-+jack_client The client name used by Csound, defaults to 'csound5'. If multiple instances of Csound connect to the JACK server, different client names need to be used to avoid name conflicts. (Linux and Mac OS X only) -+jack_inportname=[input port name prefix], -+jack_outportname=[output port name prefix] -+jack_inportname -+jack_outportname Name prefix of Csound JACK input/output ports; the default is 'input' and 'output'. The actual port name is the channel number appended to the name prefix. (Linux and Mac OS X only) Example: with the above default settings, a stereo orchestra will create these ports in full duplex operation: csound5:input1 (record left) csound5:input2 (record right) csound5:output1 (playback left) csound5:output2 (playback right) -K-K, --nopeaks--nopeaks Do not generate any PEAK chunks. -k NUM-k, --control-rate=NUM--control-rate Override the control rate (KR) supplied by the orchestra. -L DEVICE-L, --score-in=DEVICE--score-in Read line-oriented real-time score events from device DEVICE. The name stdin will permit score events to be typed at your terminal, or piped from another process. Each line-event is terminated by a carriage-return. Events are coded just like those in a standard numeric score, except that an event with p2=0 will be performed immediately, and an event with p2=T will be performed T seconds after arrival. Events can arrive at any time, and in any order. The score carry feature is legal here, as are held notes (p3 negative) and string arguments, but ramps and pp or np references are not. The -L flag is only valid on *NIX systems which have pipes. It doesn't work on Windows. -l-l, --format=long--format=long Use long integer audio samples. -M DEVICE-M, --midi-device=DEVICE--midi-device Read MIDI events from device DEVICE. If using ALSA MIDI (-+rtmidi=alsa), devices are selected by name and not number. So, you need to use an option like -M hw:CARD,DEVICE where CARD and DEVICE are the card and device numbers (e.g. -M hw:1,0). In the case of PortMidi and MME, DEVICE should be a number, and if it is out of range, an error occurs and the valid device numbers are printed.When using PortMidi, you can use '-Ma' to enable all devices. This is also convenient when you don't have devices as it will not generate an error. -m NUM-m, --messagelevel=NUM--messagelevel Message level for standard (terminal) output. Takes the sum of any of the following values: 1 = note amplitude messages 2 = samples out of range message 4 = warning messages 128 = print benchmark information And exactly one of these to select note amplitude format: 0 = raw amplitudes, no colours 32 = dB, no colors 64 = dB, out of range highlighted with red 96 = dB, all colors 256 = raw, out of range highlighted with red 512 = raw, all colours The default is 135 (128+4+2+1), which means all messages, raw amplitude values, and printing elapsed time at the end of performance. The coloring of raw amplitudes was introduced in version 5.04. --m-amps=NUM Message level for amplitudes on standard (terminal) output. 0 = no note amplitude messages 1 = note amplitude messages --m-range=NUM Message level for out of range messages on standard (terminal) output. 0 = no samples out of range message 1 = samples out of range message --m-warnings=NUM Message level for warnings on standard (terminal) output. 0 = no warning messages 1 = warning messages --m-dB=NUM Message level for amplitude format on standard (terminal) output. 0 = absolute amplitude messages 1 = dB amplitude messages --m-colours=NUM Message level for amplitude format on standard (terminal) output. 0 = no colouring of amplitude messages 1 = colouring of amplitude messages --m-benchmarks=NUM Message level for benchmark information on standard (terminal) output. 0 = no benchnark numbers 1 = print benchnark numbers -+max_str_len=integer-+max_str_len (min: 10, max: 10000) Maximum length of string variables + 1; defaults to 256 allowing a length of 255 characters. The length of string constants is not limited by this parameter. --midi-key=N--midi-key=N Route MIDI note on message key number to pfield N as MIDI value [0-127]. --midi-key-cps=N--midi-key-cps=N Route MIDI note on message key number to pfield N as cycles per second. --midi-key-oct=N--midi-key-oct=N Route MIDI note on message key number to pfield N as linear octave. --midi-key-pch=N--midi-key-pch=N Route MIDI note on message key number to pfield N as oct.pch (pitch class). --midi-velocity=N--midi-velocity=N Route MIDI note on message velocity number to pfield N as MIDI value [0-127]. --midi-velocity-amp=N--midi-velocity-amp=N Route MIDI note on message velocity number to pfield N as amplitude [0-0dbFS]. --midioutfile=FILENAME--midioutfile Save MIDI output to a file (Csound 5.00 and later only). -+msg_color=boolean-+msg_color Enable message attributes (colors etc.); might need to be disabled on some terminals which print strange characters instead of modifying text attributes. default: true. -+mute_tracks=string-+mute_tracks (max. length = 255 characters) Ignore events (other than tempo changes) in MIDI file tracks defined by pattern (for example, -+mute_tracks=00101 will mute the third and fifth tracks). -N-N, --notify--notify Notify (ring the bell) when score or MIDI track is done. -n-n, --nosound--nosound No sound. Do all processing, but bypass writing of sound to disk. This flag does not change the execution in any other way. --no-default-paths--no-default-paths Disables adding of directory of CSD/ORC/SCO to search paths. --no-expression-opt--no-expression-opt Disables expression optimization. -O FILE-O, --logfile=FILE--logfile Log output to file FILE. If FILE is null (i.e. -O null or --logfile=null) all printing of messages to the console is disabled. -o FILE-o, --output=FILE--output Output soundfile name. If not a full pathname, the soundfile will be placed in the directory given by the environment variable SFDIR (if defined), else in the current directory. The name stdout will cause audio to be written to standard output, while null results in no sound output similarly to the -n flag. If no name is given, the default name will be test. The name devaudio or dac (you can use -odac or -o dac) will request writing sound to the host audio output device. It is possible to select a device number by appending an integer value in the range 0 to 1023, or a device name separated by a : character (e.g. -odac3, -odac:hw:1,1). It depends on the host audio interface whether a device number or a name should be used. In the first case, an out of range number usually results in an error and listing the valid device numbers. --ogg--ogg Set output file format to ogg. (csound 5.18 and later) --omacro:XXX=YYY--omacro Set orchestra macro XXX to value YYY --opcode-lib=LIBNAME--opcode-lib Load plugin library LIBNAME. --orc orcname--orc Set the argument as the orchestrra file. Used when not score is required>. (Csound 5.18 and later). -+output_stream=string-+output_stream Pulseaudio output stream name. --port=N--port Set UDP port on which to listen instruments/orchestra code (implies --daemon) -Q DEVICE-Q Enables MIDI OUT operations to device id DEVICE. This flag allows parallel MIDI OUT and DAC performance. Unfortunately the real-time timing implemented in Csound is completely managed by DAC buffer sample flow. So MIDI OUT operations can present some time irregularities. These irregularities can be reduced by using a lower value for the -b flag. If using ALSA MIDI (-+rtmidi=alsa), devices are selected by name and not number. So, you need to use an option like -Q hw:CARD,DEVICE where CARD and DEVICE are the card and device numbers (e.g. -Q hw:1,0). In the case of PortMidi and MME, DEVICE should be a number, and if it is out of range, an error occurs and the valid device numbers are printed. -R-R, --rewrite--rewrite Continually rewrite the header while writing the soundfile (WAV/AIFF). -r NUM-r, --sample-rate=NUM--sample-rate Override the sampling rate (SR) supplied by the orchestra. -+raw_controller_mode=boolean-+raw_controller_mode Disable special handling of MIDI controllers like sustain pedal, all notes off etc., allowing the use of all the 128 controllers for any purpose. This will also set the initial value of all controllers to zero. Default: no. -+rtaudio=string-+rtaudio (max. length = 20 characters) Real time audio module name. The default is PortAudio. Also available, depending on platform and build options: Linux: alsa, jack; Windows: mme; Mac OS X: CoreAudio. In addition, null can be used on all platforms, to disable the use of any real time audio plugin. -+rtmidi=string-+rtmidi (max. length = 20 characters) Real time MIDI module name. Defaults to PortMidi, other options (depending on build options): Linux: alsa; Windows: mme, winmm. In addition, null can be used on all platforms, to disable the use of any real time MIDI plugin. ALSA MIDI devices are selected by name and not number. So, you need to use an option like -M hw:CARD,DEVICE where CARD and DEVICE are the card and device numbers (e.g. -M hw:1,0). -s-s, --format=short--format=short Use short integer audio samples. --sample-accurate--sample-accurate Start and stop instances of instruments at the nearest sample to the requested time. This is in contrast to traditional Csound which rounds the times to the nearest k-cycle. Note that this does not work with tied notes. --sched--sched Linux only. Use real-time scheduling and lock memory. (Also requires -d and either -o dac or -o devaudio). See also --sched=N below. --sched=N--sched=N Linux only. Same as --sched, but allows specifying a priority value: if N is positive (in the range 1 to 99) the scheduling policy SCHED_RR will be used with a priority of N; otherwise, SCHED_OTHER is used with the nice level set to N. Can also be used in the format --sched=N,MAXCPU,TIME to enable the use of a "watchdog" thread that terminates Csound if the average CPU usage exceeds MAXCPU percents over a peroid of TIME seconds (new in Csound 5.00). -+server=string-+server Pulseaudio server name. -+skip_seconds=float-+skip_seconds (min: 0) Start playback at the specified time (in seconds), skipping earlier events in the score and MIDI file. --smacro:XXX=YYY--smacro Set score macro XXX to value YYY --strset--strset Csound 5. The --strset option allows setting strset string values from the command line, in the format '--strsetN=VALUE'. It is useful for passing parameters to the orchestra (e.g. file names). --syntax-check-only--syntax-check-only Causes Csound to exit immediately after the orchestra and score parsers finish checking the syntax of the input files and before the orchestra performs the score. This option is exclusive of the --i-only flag. (Csound 5.08 and later). -T-T, --terminate-on-midi--terminate-on-midi Terminate the performance when the end of MIDI file is reached. -t0-t0, --keep-sorted-score--keep-sorted-score Prevents Csound from deleting the sorted score file, score.srt, upon exit. -t NUM-t, --tempo=NUM--tempo=NUM Use the uninterpreted beats of score.srt for this performance, and set the initial tempo at NUM beats per minute. When this flag is set, the tempo of score performance is also controllable from within the orchestra. WARNING: this mode of operation is experimental and may be unreliable. -U UTILITY-U, --utility=UTILITY--utility Invoke the utility program UTILITY. Use any invalid name to list the available utilities. -u-u, --format=ulaw--format=ulaw Use u-law audio samples. -v-v, --verbose--verbose Verbose translate and run. Prints details of orch translation and performance, enabling errors to be more clearly located. -W-W, --wave--wave, --format=wave--format=wave Write a WAV format soundfile. -x FILE-x, --extract-score=FILE--extract-score Extract a portion of the sorted score, score.srt, using the extract file FILE (see Extract). -Z-Z, --dither--dither Switch on dithering of audio conversion from internal floating point to 32, 16 and 8-bit formats. The default form of the dither is triangular. -Z-Zd, --dither--triangular--dither--triangular, --dither--uniform--dither--uniform Switch on dithering of audio conversion from internal floating point to 32, 16 and 8-bit formats. In the case of -Z the next digit should be a 1 (for trangular) or a 2 (for uniform). The exact interpretation depends on the output system. -z NUM-z, --list-opcodesNUM--list-opcodesNUM List opcodes in this version: no NUM, just show names NUM = 0, just show names NUM = 1, show arguments to each opcode using the format <opname> <outargs> <inargs> manual_src~dfsg/command/descXO.xml0000644000000000000000000000624312262561504016277 0ustar rootroot
Description of the command syntax The csound command is followed by a set of Command Line Flags and the name of the orchestra (.orc) and score (.sco) files or the Unified csd file (containing both orchestra and score) to process. Command Line Flags to control input and output configuration may appear anywhere in the command line, either separately or bundled together. A flag taking a Name or Number will find it in that argument, or in the immediately subsequent one. The following are thus equivalent commands: csound -nm3 orchname -Sxxfilename scorename csound -n -m 3 orchname -x xfilename -S scorename All flags and names are optional. The default values are: csound -s -otest -b1024 -B1024 -m7 -P128 orchname scorename where orchname is a file containing Csound orchestra code, and scorename is a file of score data in standard numeric score format, optionally presorted and time-warped. If scorename is omitted, there are two default options: if real-time input is expected (e.g. -L, -M, -iadc or -F), a dummy score file is substituted consisting of the single statement 'f 0 3600' (i.e. listen for RT input for one hour) else Csound uses the previously processed score.srt in the current directory. Csound reports on the various stages of score and orchestra processing as it goes, doing various syntax and error checks along the way. Once the actual performance has begun, any error messages will derive from either the instrument loader or the unit generators themselves. A CSound command may include any rational combination of flag arguments. Running the examples in this manual from the command line Most of the manual's examples come ready to run without the need of adding any command line flags since they specify options within the csd file's <CsOptions> tag. So you only need to type something like: csound oscil.csd within the examples folder, and realtime audio output should be generated.
manual_src~dfsg/command/preproc.xml0000644000000000000000000000570512262561504016566 0ustar rootroot
Score File Preprocessing
The Extract Feature extract This feature will extract a segment of a sorted numeric score file according to instructions taken from a control file. The control file contains an instrument list and two time points, from and to, in the form: instruments 1 2 from 1:27.5 to 2:2 The component labels may be abbreviated as i, f and t. The time points denote the beginning and end of the extract in terms of: [section no.] : [beat no.]. Each of the three parts of the argument is optional. The default values for missing i, f or t are: all instruments, beginning of score, end of score.
Independent Pre-Processing with Scsort Scsort Although the result of all score preprocessing is retained in the file score.srt after orchestra performance (it exists as soon as score preprocessing has completed), the user may sometimes want to run these phases independently. The command scot filename will process the Scot formatted filename, and leave a standard numeric score result in a file named score for perusal or later processing. The command scscort < infile > outfile will put a numeric score infile through Carry, Tempo, and Sort preprocessing, leaving the result in outfile. Likewise extract, also normally invoked as part of the Csound command, can be invoked as a standalone program: extract xfile < score.sort > score.extract This command expects an already sorted score. An unsorted score should first be sent through Scsort then piped to the extract program: scsort < scorefile | extract xfile > score.extract
manual_src~dfsg/command/unifile.xml0000644000000000000000000002502612262561504016545 0ustar rootroot
Unified File Format for Orchestras and Scores
Description The Unified File FormatUnified File Format, introduced in Csound version 3.50, enables the orchestra and score files, as well as command line flags, to be combined in one file. The file has the extension .csd. This format was originally introduced by Michael Gogins in AXCsound. .csd .csoundrc <CsFileB> <CsInstruments> <CsMidifileB> <CsOptions> <CsSampleB> <CsScore> <CsoundSynthesizer> <CsVersion> <CsLicense> The file is a structured data file which uses markup language, similar to any SGML such as HTML. Start tags (<tag>) and end tags (</tag>) are used to delimit the various elements. The file is saved as a text file. Structured Data File Format Mandatory Elements The first tag in the file must be the start tag <CsoundSynthesizer>. The last tag in the file must be the end tag </CsoundSynthesizer>. This element is used to alert the csound compiler to the .csd format. All text before the start tag and after the end tag is ignored by Csound. The tag may also be spelled <CsoundSynthesiser>. Options (<CsOptions>) Csound command line flags are put in the Options Element. This section is delimited by the start tag <CsOptions> and the end tag </CsOptions> Lines beginning with # or ; are treated as comments. Orchestra (<CsInstruments>) The instrument definitions (orchestra) are put into the Instruments Element. The statements and syntax in this section are identical to the Csound orchestra file, and have the same requirements, including the header statements (sr, kr, etc.) This Instruments Element is delimited with the start tag <CsInstruments> and the end tag </CsInstruments>. Score (<CsScore>) Csound score statements are put in the Score Element. The statements and syntax in this section are identical to the Csound score file, and have the same requirements. The Score Element is delimited by the start tag <CsScore> and the end tag </CsScore>. As an alternative Csound score statements can also be generated by an external program using the CsScore scheme with an attribute bin. The lines upto the closing tag </CsScore> are copied to a file and the external program named is called with that file name and the destination score file. The external program should create a standard Csound score. Optional Elements Included Base64 Files (<CsFileB>) Base64-encoded files may be included with the tag <CsFileB filename=filename>, where filename is the name of the file to be included. The Base64-encoded data should be terminated with a </CsFileB> tag. For encoding files, the csb64enc and makecsd utilities (included with Csound 5.00 and newer) can be used. The file will be extracted to the current directory, and deleted at end of performance. If there is an already existing file with the same name, it is not overwritten, but an error will occur instead. Base64-encoded MIDI files may be included with the tag <CsMidifileB filename=filename>, where filename is the name of the file containing the MIDI information. There is no matching end tag. This was added in Csound version 4.07. Note: using this tag is not recommended; use <CsFileB> instead. Base64-encoded sample files may be included with the tag <CsSampleB filename=filename>, where filename is the name of the file containing the sample. There is no matching end tag. This was added in Csound version 4.07. Note: using this tag is not recommended; use <CsFileB> instead. Version Blocking (<CsVersion>) Versions of Csound may blocked by placing one of the following statements between the start tag <CsVersion> and the end tag </CsVersion>: Before #.# or After #.# where #.# is the requested Csound version number. The second statement may be written simply as: #.# This was added in Csound version 4.09. Licence Information (<CsLicence> or <CsLicense>) Licencing details can be included in between the start tag <CsLicence> and the end tag </CsLicence>. There is no format for this information, any text is acceptable. This text will be printed by Csound to the console when the CSD is run.
Example Below is a sample file, test.csd, which renders a .wav file at 44.1 kHz sample rate containing one second of a 1 kHz sine wave. Displays are suppressed. test.csd was created from two files, tone.orc and tone.sco, with the addition of command line flags. <CsoundSynthesizer>; ; test.csd - a Csound structured data file <CsOptions> -W -d -o tone.wav </CsOptions> <CsVersion> ; optional section Before 4.10 ; these two statements check for After 4.08 ; Csound version 4.09 </CsVersion> <CsInstruments> ; originally tone.orc sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 a1 oscil p4, p5, 1 ; simple oscillator out a1 endin </CsInstruments> <CsScore> ; originally tone.sco f1 0 8192 10 1 i1 0 1 20000 1000 ; play one second of one kHz tone e </CsScore> </CsoundSynthesizer>
Command Line Parameter File (.csoundrc) If the file .csoundrc exists, it will be used to set the command line parameters. These can be overridden. Csound 5.00 and newer versions read this file from the HOME directory first (or the full path file name defined by the CSOUNDRC environment variable), and then the current directory. If both exist, options in the .csoundrc in the current directory will have higher precedence. It uses the same form as a .csd file, but no tags are needed. Lines beginning with # or ; are treated as comments. A .csoundrc file can contain something like this: -+rtaudio=portaudio -odac2 -iadc2 -+rtmidi=winmme -M1 -Q1 -m0 In this case, csound will generate real-time output and take realtime input from device 2, using the portaudio driver interface. It will input and output realtime MIDI on interface 1. It will print very few messages (-m0). These options will be used by default when other options are not given inside the <CsOptions> of the .csd file or the command line (See Order of precendence).
manual_src~dfsg/command/environmentXO.xml0000644000000000000000000002262512262561504017727 0ustar rootroot
Csound Environment Variables The following environment variables can be used by Csound: SFDIR: Default directory for sound files. Used if no full path is given for sound files. SSDIR: Default directory for input (source) audio and MIDI files. Used if no full path is given for sound files. May be used in conjunction with SFDIR to set separate input and output directories. Please note that MIDI files as well as audio files are also sought inside SSDIR. SADIR: Default directory for analysis files. Used if no full path is given for analysis files. SFOUTYP: Sets the default output file type. Currently only 'WAV', 'AIFF' and 'IRCAM' are valid. This flag is checked by the csound executable and the utilities and is used if no file output type is specified. INCDIR: Include directory. Specifies the location of files used by #include statements. OPCODEDIR: Defines the location of csound opcode plugins for the single precision float (32-bit) version. OPCODEDIR64: Defines the location of csound opcode plugins for the double precision float (64-bit) version. SNAPDIR: Is used by the FLTK widget opcodes when loading and saving snapshots. CSOUNDRC: Defines the csound resource (or configuration) file. A full path and filename containing csound flags must be specified. This variable defaults to .csoundrc if not present. CSSTRNGS: In Csound 5.00 and later versions, the localisation of messages is controlled by two environment variables CSSTRNGS and CS_LANG, both of which are optional. CSSTRNGS points to a directory containing .xmg files. CS_LANG: Selects a language for csound messages. RAWWAVE_PATH: Is used by the STK opcodes to find the raw wave files. Only relevant if you are using STK wrapper opcodes like STKBowed or STKBrass. CSNOSTOP: If this environment variable is set to "yes", then any graph displays are closed automatically at the end of performance (meaning that you possibly will not see much of them in the case of a short non-realtime render). Otherwise, you need to click "Quit" in the FLTK display window to exit, allowing for viewing the graphs even after the end of score is reached. MFDIR: Default directory for MIDI files. Used if no full path is given for MIDI files. Please note that MIDI files are sought in SSDIR and SFDIR as well. For more information about SFDIR, SSDIR, SADIR, MFDIR and INCDIR see Directories and files. The only mandatory environment variables are OPCODEDIR and OPCODEDIR64. It is very important to set them correctly, otherwise most of the opcodes will not be available. Make sure you set the path correctly depending on the precision of your binary. if you run csound on a command line without any arguments you should see some text like : Csound version 5.01.0 beta (float samples) Mar 23 2006. This text refers to the single precision version. CSSTRNGS and CS_LANG currently have very limited use since Csound has not yet been completely translated into other languages. Other environment variables which are not exclusive to Csound but which might be of importance are: PATH: The directory containing csound executables should be listed in this variable. PYTHONPATH: If you intend to use CsoundVST and python, the directory containing the _CsoundVST shared library and the CsoundVST.py file must be in your PYTHONPATH environment variable (or the default path python searches in), so that the Python runtime knows how to load these files. LADSPA_PATH and DSSI_PATH: These environment variables are required if you are using the dssi4cs (LADSPA and DSSI host) plug-in opcodes. CSDOCDIR: Specifies the directory where the html help files are located. Though not used by Csound directly, this environment variable can help front-ends and editors (which implement it) to find the csound manual. Setting environment variables On the command line You can set environment variables on the command line or the configuration file .csoundrc by using the command line flag --env:NAME=VALUE or --env:NAME+=VALUE, where NAME is the environment variable name, and VALUE is its value. See Command-line Flags Please note that this method of setting environment variables will not work for variables which are parsed before the command line arguments. SADIR, SSDIR, SFDIR, INCDIR, SNAPDIR, RAWWAVE_PATH, CSNOSTOP, SFOUTYP should work, but the following environment variables must be set on the system prior to running csound: OPCODEDIR, OPCODEDIR64, CSSTRINGS, and CS_LANG. CSOUNDRC can currently (v. 5.02) be set using --env, but this behavior is not guaranteed for future versions. Windows To set a csound environment on Windows XP and 2000 go to Control Panel->System->Advanced and click on the button 'Environment Variables'. On other Windows earlier than XP you set environment variables in the autoexec.bat file. Go to 'My Computer', select C: drive, right click on autoexec.bat, and select 'Edit'. The statement format is: SET NAME=VALUE . Linux You can set environment variables on Linux in many ways. You can set them using the export shell command, by setting them on .bashrc or similar files or by adding them to the /etc/profile file. Mac If the user has a Mac that shipped with an OS X version prior to 10.3 (includes 10.2 and 10.1) then it is possible that the default shell is the Tenex C-shell (tcsh). If this is the case, then you either have to type: ~% setenv OPCODEDIR "/Users/you/your/Csound5/build" or change your /etc/profile and or edit your .tcshrc file. If the user has a Mac that shipped with OS X 10.3 or 10.4 then it likely has the "Bourne-again" C-shell (bash) as the default shell. If this is the case, then the user must type something like: ~$ export OPCODEDIR=/Users/you/your/Csound5/build in addition if the bash shell is the default, then it is usually easier to edit your .bashrc or /etc/profile. Note that if users choose one of the above methods, ie editing the .bashrc file then the environment variables are executed when a new shell is created. This can be problematic if your application implements a Quartz or Aqua interface and does not use the commandline. If this is the case, then the standard solution (up to OS 10.3.9 and unless the application uses the csoundAPI and sets the environ variables directly) is to create an XML property list file (called a .plist file by the OS). This file should nominally be located at ~/.MacOSX/Environment.plist. This has been a solution specifically for the [csoundapi~] object for Pd on OS X. Since Pd uses an OS X native .app style packaging, and runs off of the Aqua interface, the standard means of supplying environment variables to Csound do not work. The solution is to set Csound's environment variables for the Aqua environment. Likely, most users will not have the hidden folder .MacOSX located in their $HOME directory (aka ~/) This folder must first be created and the Environment.plist added to this folder. The contents of the Environment.plist file should be something like: <?xml version="1.0" encoding='UTF-8"?> <!DOCTYPE plist PUBLIC "-//Apple Computer//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd"> <plist version="1.0"> <dict> <key>OPCODEDIR</key> <string>/Library/Frameworks/CsoundLib.framework/Versions/5.1/Resources/Opcodes</string> <key>OPCODEDIR64</key> <string>/Volumes/ExternalHD/devel/csound5/lib64</string> <key>INCDIR</key> <string>/Volumes/ExternalHD/CSOUND/include</string> <key>SFDIR</key> <string>/Volumes/ExternalHD/iTunes/csoundaudio</string> </dict> </plist> and so on, using the XML <key> tag for each environment variable required by the API and the <string> tag for it's corresponding path on the system. Please note that you must login out and login in for these changes to take effect.
manual_src~dfsg/command/topXO.xml0000644000000000000000000000466512262561504016171 0ustar rootroot The Csound Command Csound is a command to generate a sound output from an orchestra and score files (or a unified csd file). It is designed to be called from a terminal or DOS window, but can be called from an easier to use front-end. The score file can be in one of many different formats, according to user preference. Translation, sorting, and formatting into orchestra-readable numeric text is handled by various preprocessors; all or part of the score is then sent on to the orchestra. Orchestra performance is influenced by command flags, which set the level of displays and console reports, specify I/0 filenames and sample formats, and declare the nature of real-time sensing and control.
Order of Precedence There are five places where options for Csound performance may be set. They are processed in the following order: Csound's own defaults File defined by CSOUNDRC environment variable, or .csoundrc file in the HOME directory .csoundrc file in the current directory <CsOptions> tag in a .csd file Csound command line The lower options in the list will override any earlier ones. As of version 5.01, sample and control rate override flags (-r and -k) specified anywhere override sr, kr, and ksmps in the orchestra header.
&commanddesc; &commandflags; &commandflagscategory; &commandenvironment; &commandunifile; manual_src~dfsg/command/desc.xml0000644000000000000000000000625412262561504016032 0ustar rootroot
Description of the command syntax The csound command is followed by a set of Command Line Flags and the name of the orchestra (.orc) and score (.sco) files or the Unified csd file (containing both orchestra and score) to process. Command Line Flags to control input and output configuration may appear anywhere in the command line, either separately or bundled together. A flag taking a Name or Number will find it in that argument, or in the immediately subsequent one. The following are thus equivalent commands: csound -nm3 orchname -Sxxfilename scorename csound -n -m 3 orchname -x xfilename -S scorename All flags and names are optional. The default values are: csound -s -otest -b1024 -B1024 -m7 -P128 orchname scorename where orchname is a file containing Csound orchestra code, and scorename is a file of score data in standard numeric score format, optionally presorted and time-warped. If scorename is omitted, there are two default options: if real-time input is expected (e.g. -L, -M, -iadc or -F), a dummy score file is substituted consisting of the single statement 'f 0 3600' (i.e. listen for RT input for one hour) else Csound uses the previously processed score.srt in the current directory. Csound reports on the various stages of score and orchestra processing as it executes, performing various syntax and error checks along the way. Once the actual performance has begun, any error messages will derive from either the instrument loader or the unit generators themselves. A CSound command may include any rational combination of flag arguments. Running the examples in this manual from the command line Most of the manual's examples come ready to run without the need of adding any command line flags since they specify options within the csd file's <CsOptions> tag, so you only need to type something like: csound oscil.csd within the examples folder, and realtime audio output should be generated.
manual_src~dfsg/command/top.xml0000644000000000000000000000502512262561504015711 0ustar rootroot The Csound Command Csound is a command to generate a sound output from an orchestra file and a score file (or a unified csd file). It is designed to be called from a terminal or DOS window, but can be called from an easier-to-use front-end. The score file can be in one of many different formats, according to user preference. Translation, sorting, and formatting into orchestra-readable numeric text is handled by various preprocessors; all or part of the score is then sent on to the orchestra. Orchestra performance is influenced by command flags, which set the level of displays and console reports, specify I/0 filenames and sample formats, and declare the nature of real-time sensing and control.
Order of Precedence There are five places where options for Csound performance may be set. They are processed in the following order: Csound's own defaults File defined by the CSOUNDRC environment variable, or .csoundrc file in the HOME directory A .csoundrc file in the current directory <CsOptions> tag in a .csd file Passed on the Csound command line The later options in the list will override any earlier ones. As of version 5.01 of Csound, sample and control rate override flags (-r and -k) specified anywhere override sr, kr, and ksmps defined in the orchestra header.
&commanddesc; &commandflags; &commandflagscategory; &commandenvironment; &commandunifile; &commandpreproc; manual_src~dfsg/command/flagscategory.xml0000644000000000000000000016402412262561504017746 0ustar rootroot
Command-line Flags (by Category) Listed below are the command line available in Csound5 organized by categories. Various platform implementations may not react the same way to different flags! You can view the command line flags organized alphabetically in Command-line Flags (Alphabetically). The format of a command is either: csound orchname scorename or csound csdfilename where the arguments are of 2 types: flags arguments (beginning with a -,-- or -+), and name arguments (such as filenames). Certain flag arguments take a following name or numeric argument. Flags that start with -- and -+ usually take an argument themselves using =. Audio File Ouput -3-3, --format=24bit--format=24bit Use 24-bit audio samples. -8-8, --format=uchar--format=uchar Use 8-bit unsigned character audio samples. -A-A, --aiff--aiff, --format=aiff--format=aiff Write an AIFF format soundfile. Use with the -c, -s, -l, or -f flags. -a-a, --format=alaw--format=alaw Use a-law audio samples. -c-c, --format=schar--format=schar Use 8-bit signed character audio samples. -f-f, --format=float--format=float Use single-format float audio samples (not playable on some systems, but can be read by -i, soundin and GEN01 --format=type--format Set the audio file output format to one of the formats available in libsndfile. At present the list is aiff, au, avr, caf, flac, htk, ircam, mat4, mat5, nis, paf, pvf, raw, sd2, sds, svx, voc, w64, wav, wavex and xi. Can also be used as --format=type:format or --format=format:type to set both the file type (wav, aiff, etc.) and sample format (short, long, float, etc.) at the same time. -h-h, --noheader--noheader No header on output soundfile. Don't write a file header, just binary samples. -i FILE-i, --input=FILE--input Input soundfile name. If not a full pathname, the file will be sought first in the current directory, then in that given by the environment variable SSDIR (if defined), then by SFDIR. The name stdin will cause audio to be read from standard input. The name devaudio or adc will request sound from the host audio input device. It is possible to select a device number by appending an integer value in the range 0 to 1023, or a device name separated by a : character. It depends on the host audio interface whether a device number or a name should be used. In the first case, an out of range number usually results in an error and listing the valid device numbers. The audio coming in using -i can be received using opcodes like inch. -J-J, --ircam--ircam, --format=ircam--format=ircam Write an IRCAM format soundfile. -K-K, --nopeaks--nopeaks Do not generate any PEAK chunks. -l-l, --format=long--format=long Use long integer audio samples. -n-n, --nosound--nosound No sound. Do all processing, but bypass writing of sound to disk. This flag does not change the execution in any other way. -o FILE-o, --output=FILE--output Output soundfile name. If not a full pathname, the soundfile will be placed in the directory given by the environment variable SFDIR (if defined), else in the current directory. The name stdout will cause audio to be written to standard output, while null results in no sound output similarly to the -n flag. If no name is given, the default name will be test. The name dac or devaudio (you can use -odac or -o dac) will request writing sound to the host audio output device. It is possible to select a device number by appending an integer value in the range 0 to 1023, or a device name separated by a : character. It depends on the host audio interface whether a device number or a name should be used. In the first case, an out of range number usually results in an error and listing the valid device numbers. --ogg--ogg Set output file format to ogg. (Csound 5.18 and later). -R-R, --rewrite--rewrite Continually rewrite the header while writing the soundfile (WAV/AIFF). -s-s, --format=short--format=short Use short integer audio samples. -u-u, --format=ulaw--format=ulaw Use u-law audio samples. -W-W, --wave--wave, --format=wave--format=wave Write a WAV format soundfile. -Z-Z, --dither--dither Switch on dithering of audio conversion from internal floating point to 32, 16 and 8-bit formats. The default form of the dither is triangular. -Z-Zd, --dither--triangular--dither--triangular, --dither--uniform--dither--uniform Switch on dithering of audio conversion from internal floating point to 32, 16 and 8-bit formats. In the case of -Z the next digit should be a 1 (for trangular) or a 2 (for uniform). The exact interpretation depends on the output system. Output File Id tags -+id_artist=string-+id_artist (max. length = 200 characters) Artist tag in output soundfile (no spaces) -+id_comment=string-+id_comment (max. length = 200 characters) Comment tag in output soundfile (no spaces) -+id_copyright=string-+id_copyright (max. length = 200 characters) Copyright tag in output soundfile (no spaces) -+id_date=string-+id_date (max. length = 200 characters) Date tag in output soundfile (no spaces) -+id_software=string-+id_software (max. length = 200 characters) Software tag in output soundfile (no spaces) -+id_title=string-+id_title (max. length = 200 characters) Title tag in output soundfile (no spaces) Realtime Audio Input/Output -i adc[DEVICE]-i, --input=adc[DEVICE]--input The name devaudio or adc will request sound from the host audio input device. It is possible to select a device number by appending an integer value in the range 0 to 1023, or a device name separated by a : character (e.g. -iadc3, -iadc:hw:1,1). It depends on the host audio interface whether a device number or a name should be used. In the first case, an out of range number usually results in an error and listing the valid device numbers. -o dac[DEVICE]-o, --output=dac[DEVICE]--output The name dac or devaudio (you can use -odac or -o dac) will request writing sound to the host audio output device. It is possible to select a device number by appending an integer value in the range 0 to 1023, or a device name separated by a : character (e.g. -odac3, -odac:hw:1,1). It depends on the host audio interface whether a device number or a name should be used. In the first case, an out of range number usually results in an error and listing the valid device numbers. -+rtaudio=string-+rtaudio (max. length = 20 characters) Real time audio module name. The default is PortAudio (all platforms). Also available, depending on platform and build options: Linux: alsa, jack; Windows: mme; Mac OS X: CoreAudio. In addition, null can be used on all platforms, to disable the use of any real time audio plugin. -+server=string-+server Pulseaudio server name. -+output_stream=string-+output_stream Pulseaudio output stream name. -+input_stream=string-+input_stream Pulseaudio input stream name. -+jack_client=[client_name]-+jack_client The client name used by Csound, defaults to 'csound5'. If multiple instances of Csound connect to the JACK server, different client names need to be used to avoid name conflicts. (Linux and Mac OS X only) -+jack_inportname=[input port name prefix], -+jack_outportname=[output port name prefix] -+jack_inportname -+jack_outportname Name prefix of Csound JACK input/output ports; the default is 'input' and 'output'. The actual port name is the channel number appended to the name prefix. (Linux and Mac OS X only) Example: with the above default settings, a stereo orchestra will create these ports in full duplex operation: csound5:input1 (record left) csound5:input2 (record right) csound5:output1 (playback left) csound5:output2 (playback right) MIDI File Input/Ouput -F FILE-F, --midifile=FILE--midifile Read MIDI events from MIDI file FILE. The file should have only one track in Csound versions 4.xx and earlier; this limitation is removed in Csound 5.00. --midioutfile=FILENAME--midioutfile Save MIDI output to a file (Csound 5.00 and later only). -+mute_tracks=string-+mute_tracks (max. length = 255 characters) Ignore events (other than tempo changes) in MIDI file tracks defined by pattern (for example, -+mute_tracks=00101 will mute the third and fifth tracks). -+raw_controller_mode=boolean-+raw_controller_mode Disable special handling of MIDI controllers like sustain pedal, all notes off etc., allowing the use of all the 128 controllers for any purpose. This will also set the initial value of all controllers to zero. Default: no. -+skip_seconds=float-+skip_seconds (min: 0) Start playback at the specified time (in seconds), skipping earlier events in the score and MIDI file. -T-T, --terminate-on-midi--terminate-on-midi Terminate the performance when the end of MIDI file is reached. MIDI Realtime Input/Ouput -M DEVICE-M, --midi-device=DEVICE--midi-device Read MIDI events from device DEVICE. If using ALSA MIDI (-+rtmidi=alsa), devices are selected by name and not number. So, you need to use an option like -M hw:CARD,DEVICE where CARD and DEVICE are the card and device numbers (e.g. -M hw:1,0). In the case of PortMidi and MME, DEVICE should be a number, and if it is out of range, an error occurs and the valid device numbers are printed. When using PortMidi, you can use '-Ma' to enable all devices. This is also convenient when you don't have devices as it will not generate an error. --midi-key=N--midi-key=N Route MIDI note on message key number to pfield N as MIDI value [0-127]. --midi-key-cps=N--midi-key-cps=N Route MIDI note on message key number to pfield N as cycles per second. --midi-key-oct=N--midi-key-oct=N Route MIDI note on message key number to pfield N as linear octave. --midi-key-pch=N--midi-key-pch=N Route MIDI note on message key number to pfield N as oct.pch (pitch class). --midi-velocity=N--midi-velocity=N Route MIDI note on message velocity number to pfield N as MIDI value [0-127]. --midi-velocity-amp=N--midi-velocity-amp=N Route MIDI note on message velocity number to pfield N as amplitude [0-0dbFS]. --midioutfile=FILENAME--midioutfile Save MIDI output to a file (Csound 5.00 and later only). -+rtmidi=string-+rtmidi (max. length = 20 characters) Real time MIDI module name. Defaults to PortMidi, other options (depending on build options): Linux: alsa; Windows: mme, winmm. In addition, null can be used on all platforms, to disable the use of any real time MIDI plugin. ALSA MIDI devices are selected by name and not number. So, you need to use an option like -M hw:CARD,DEVICE where CARD and DEVICE are the card and device numbers (e.g. -M hw:1,0). -Q DEVICE-Q Enables MIDI OUT operations to device id DEVICE. This flag allows parallel MIDI OUT and DAC performance. Unfortunately the real-time timing implemented in Csound is completely managed by DAC buffer sample flow. So MIDI OUT operations can present some time irregularities. These irregularities can be reduced by using a lower value for the -b flag. If using ALSA MIDI (-+rtmidi=alsa), devices are selected by name and not number. So, you need to use an option like -Q hw:CARD,DEVICE where CARD and DEVICE are the card and device numbers (e.g. -Q hw:1,0). In the case of PortMidi and MME, DEVICE should be a number, and if it is out of range, an error occurs and the valid device numbers are printed. Display --csd-line-nums=NUM--csd-line-nums Determines how line numbers are counted and displayed for error messages when processing a Csound Unified Document file (.csd). This flag has no effect if separate orchestra and score files are used. (Csound 5.08 and later). 0 = line numbers are relative to the beginning of the orchestra or score sections of the CSD 1 = line numbers are relative to the beginning of the CSD file. This is the default as of Csound 5.08. -d-d, --nodisplays--nodisplays Suppress all displays. See -O if you want to save the log to a file. --displays--displays Enables displays, reverting the effect of any previous -d flag. -G-G, --postscriptdisplay--postscriptdisplay Suppress graphics, use PostScript displays instead. -g-g, --asciidisplay--asciidisplay Suppress graphics, use ASCII displays instead. -H#-H, --heartbeat=NUM--heartbeat Print a heartbeat after each soundfile buffer write: no NUM, a rotating bar. NUM = 1, a rotating bar. NUM = 2, a dot (.) NUM = 3, filesize in seconds. NUM = 4, sound a bell. -m NUM-m, --messagelevel=NUM--messagelevel Message level for standard (terminal) output. Takes the sum of any of the following values: 1 = note amplitude messages 2 = samples out of range message 4 = warning messages 128 = print benchmark information And exactly one of these to select note amplitude format: 0 = raw amplitudes, no colours 32 = dB, no colors 64 = dB, out of range highlighted with red 96 = dB, all colors 256 = raw, out of range highlighted with red 512 = raw, all colours The default is 135 (128+4+2+1), which means all messages, raw amplitude values, and printing elapsed time at the end of performance. The coloring of raw amplitudes was introduced in version 5.04 --m-amps=NUM Message level for amplitudes on standard (terminal) output. 0 = no note amplitude messages 1 = note amplitude messages --m-range=NUM Message level for out of range messages on standard (terminal) output. 0 = no samples out of range message 1 = samples out of range message --m-warnings=NUM Message level for warnings on standard (terminal) output. 0 = no warning messages 1 = warning messages --m-dB=NUM Message level for amplitude format on standard (terminal) output. 0 = absolute amplitude messages 1 = dB amplitude messages --m-colours=NUM Message level for amplitude format on standard (terminal) output. 0 = no colouring of amplitude messages 1 = colouring of amplitude messages --m-benchmarks=NUM Message level for benchmark information on standard (terminal) output. 0 = no benchnark numbers 1 = print benchnark numbers -+msg_color=boolean-+msg_color Enable message attributes (colors etc.); might need to be disabled on some terminals which print strange characters instead of modifying text attributes. default: true. -v-v, --verbose--verbose Verbose translate and run. Prints details of orch translation and performance, enabling errors to be more clearly located. -z NUM-z, --list-opcodesNUM--list-opcodesNUM List opcodes in this version: no NUM, just show names NUM = 0, just show names NUM = 1, show arguments to each opcode using the format <opname> <outargs> <inargs> Performance Configuration and Control -B NUM-B, --hardwarebufsamps=NUM--hardwarebufsamps Number of audio sample-frames held in the DAC hardware buffer. This is a threshold on which software audio I/O (above) will wait before returning. A small number reduces audio I/O delay; but the value is often hardware limited, and small values will risk data lates. In the case of portaudio output (the default real-time output), the -B parameter (more precisely, -B / sr) is passed as the "suggested latency" value. Other than that, Csound has no control over how PortAudio interprets the parameter. The default is 1024 on Linux, 4096 on Mac OS X and 16384 on Windows. -b NUM-b, --iobufsamps=NUM--iobufsamps Number of audio sample-frames per sound i/o software buffer. Large is efficient, but small will reduce audio I/O delay and improve the accuracy of the timing of real time events. The default is 256 on Linux, 1024 on MacOS X, and 4096 on Windows. In real-time performance, Csound waits on audio I/O on NUM boundaries. It also processes audio (and polls for other input like MIDI) on orchestra ksmps boundaries. The two can be made synchronous. For convenience, if NUM is negative, the effective value is ksmps * -NUM (audio synchronous with k-period boundaries). With NUM small (e.g. 1) polling is then frequent and also locked to fixed DAC sample boundaries. Note: if both -iadc and -odac are used at the same time (full duplex real time audio), the -b option should be set to an integer multiple of ksmps. -d--daemon Run in daemon mode: do not exit if CSD/orchestra is not given, is empty or does not compile. -k NUM-k, --control-rate=NUM--control-rate Override the control rate (KR) supplied by the orchestra. -L DEVICE-L, --score-in=DEVICE--score-in Read line-oriented real-time score events from device DEVICE. The name stdin will permit score events to be typed at your terminal, or piped from another process. Each line-event is terminated by a carriage-return. Events are coded just like those in a standard numeric score, except that an event with p2=0 will be performed immediately, and an event with p2=T will be performed T seconds after arrival. Events can arrive at any time, and in any order. The score carry feature is legal here, as are held notes (p3 negative) and string arguments, but ramps and pp or np references are not. The -L flag is only valid on *NIX systems which have pipes. It doesn't work on Windows. --omacro:XXX=YYY--omacro Set orchestra macro XXX to value YYY --port=N--port Set UDP port on which to listen instruments/orchestra code (implies --daemon) -r NUM-r, --sample-rate=NUM--sample-rate Override the sampling rate (SR) supplied by the orchestra. --sample-accurate--sample-accurate Start and stop instances of instruments at the nearest sample to the requested time. This is in contrast to traditional Csound which rounds the times to the nearest k-cycle. Note that this does not work with tied notes. --sched--sched Linux only. Use real-time scheduling and lock memory. (Also requires -d and either -o dac or -o devaudio). See also --sched=N below. --sched=N--sched=N Linux only. Same as --sched, but allows specifying a priority value: if N is positive (in the range 1 to 99) the scheduling policy SCHED_RR will be used with a priority of N; otherwise, SCHED_OTHER is used with the nice level set to N. Can also be used in the format --sched=N,MAXCPU,TIME to enable the use of a "watchdog" thread that terminates Csound if the average CPU usage exceeds MAXCPU percents over a peroid of TIME seconds (new in Csound 5.00). --smacro:XXX=YYY--smacro Set score macro XXX to value YYY --strset--strset Csound 5. The --strset option allows setting strset string values from the command line, in the format '--strsetN=VALUE'. It is useful for passing parameters to the orchestra (e.g. file names). -+skip_seconds=float-+skip_seconds (min: 0) Start playback at the specified time (in seconds), skipping earlier events in the score and MIDI file. -t NUM-t, --tempo=NUM--tempo=NUM Use the uninterpreted beats of score.srt for this performance, and set the initial tempo at NUM beats per minute. When this flag is set, the tempo of score performance is also controllable from within the orchestra. WARNING: this mode of operation is experimental and may be unreliable. Miscellaneous -@ FILE-@ Provide an extended command-line in file FILE -C-C, --cscore--cscore Use Cscore processing of the scorefile. --default-paths--no-default-paths Reenables adding of directory of CSD/ORC/SCO to search paths, if it has been disabled by a previous --no-default-paths (e.g. in .csoundrc). -D-D, --defer-gen1--defer-gen1 Defer GEN01 soundfile loads until performance time. --env:NAME=VALUE--env:NAME=VALUE Set environment variable NAME to VALUE. Note: not all environment variables can be set this way, because some are read before parsing the command line. INCDIR, SADIR, SFDIR, and SSDIR are known to work. --env:NAME+=VALUE--env:NAME+=VALUE Append VALUE to ';' separated list of search paths in environment variable NAME (should be INCDIR, SADIR, SFDIR, or SSDIR). If a file is found in multiple directories, the last will be used. --expression-opt--expression-opt Since Csound 5. Turns on some optimizations in expressions: Redundant assignment operations are eliminated whenever possible. This means that for example this line a1 = a2 + a3 will compile as a1 Add a2, a3 instead of #a0 Add a2, a3 a1 = #a0 saving a temporary variable and an opcode call. Less opcode calls result in reduced CPU usage (an average orchestra may compile about 10% faster with --expression-opt, but it depends largely on how many expressions are used, what the control rate is (see also below), etc.; thus, the difference may be less, but also much more). number of a- and k-rate temporary variables is significantly reduced. This expression (a1 + a2 + a3 + a4) will compile as #a0 Add a1, a2 #a0 Add #a0, a3 #a0 Add #a0, a4 ; (the result is in #a0) instead of #a0 Add a1, a2 #a1 Add #a0, a3 #a2 Add #a1, a4 ; (the result is in #a2) The advantages of less temporary variables are: less cache memory is used, which may improve performance of orchestras with many a-rate expressions and a low control rate (e.g. ksmps = 100) large orchestras may load faster due to less different identifier names index overflow errors (i.e. when messages like this Case2: indx=-56004 (ffff253c); (short)indx = 9532 (253c) are printed and odd behavior or a Csound crash occurs) may be fixed, because such errors are triggered by too many different (especially a-rate) variable names in a single instrument. Note that this optimization (due to technical reasons) is not performed on i-rate temporary variables. Warning When --expression-opt is turned on, it is not allowed to use the i() function with an expression argument, and relying on the value of k-rate expressions at i-time is unsafe. --help--help Display on-line help message. -I-I, --i-only--i-only i-time only. Allocate and initialize all instruments as per the score, but skip all p-time processing (no k-signals or a-signals, and thus no amplitudes and no sound). Provides a fast validity check of the score pfields and orchestra i-variables. This option is exclusive of the --syntax-check-only flag. -+ignore_csopts=integer-+ignore_csopts= If set to 1, Csound will ignore all options specified in the csd file's CsOptions section. See Unified File Format for Orchestras and Scores. -+max_str_len=integer-+max_str_len (min: 10, max: 10000) Maximum length of string variables + 1; defaults to 256 allowing a length of 255 characters. The length of string constants is not limited by this parameter. -N-N, --notify--notify Notify (ring the bell) when score or MIDI track is done. --no-default-paths--no-default-paths Disables adding of directory of CSD/ORC/SCO to search paths. --no-expression-opt--no-expression-opt Disables expression optimization. -O FILE-O, --logfile=FILE--logfile Log output to file FILE. If FILE is null (i.e. -O null or --logfile=null) all printing of messages to the console is disabled. --opcode-lib=LIBNAME--opcode-lib Load plugin library LIBNAME. --orc orcname--orc Set the argument as the orchestra file. Used when not score is required>. (Csound 5.18 and later). --syntax-check-only--syntax-check-only Causes Csound to exit immediately after the orchestra and score parsers finish checking the syntax of the input files and before the orchestra performs the score. This option is exclusive of the --i-only flag. (Csound 5.08 and later). -t0-t0, --keep-sorted-score--keep-sorted-score Prevents Csound from deleting the sorted score file, score.srt, upon exit. -U UTILITY-U, --utility=UTILITY--utility Invoke the utility program UTILITY. Use any invalid name to list the available utilities. -x FILE-x, --extract-score=FILE--extract-score Extract a portion of the sorted score, score.srt, using the extract file FILE (see Extract).
manual_src~dfsg/command/environment.xml0000644000000000000000000002320312262561504017451 0ustar rootroot
Csound Environment Variables The following environment variables can be used by Csound: SFDIR: Default directory for sound files. Used if no full path is given for sound files. SSDIR: Default directory for input (source) audio and MIDI files. Used if no full path is given for sound files. May be used in conjunction with SFDIR to set separate input and output directories. Please note that MIDI files as well as audio files are also sought inside SSDIR. SADIR: Default directory for analysis files. Used if no full path is given for analysis files. SFOUTYP: Sets the default output file type. Currently only 'WAV', 'AIFF' and 'IRCAM' are valid. This flag is checked by the csound executable and the utilities and is used if no file output type is specified. INCDIR: Include directory. Specifies the location of files used by #include statements. OPCODEDIR: Defines the location of csound opcode plugins for the single precision float (32-bit) version. OPCODEDIR64: Defines the location of csound opcode plugins for the double precision float (64-bit) version. SNAPDIR: Is used by the FLTK widget opcodes when loading and saving snapshots. CSOUNDRC: Defines the csound resource (or configuration) file. A full path and filename containing csound flags must be specified. This variable defaults to .csoundrc if not present. CSSTRNGS: In Csound 5.00 and later versions, the localisation of messages is controlled by two environment variables CSSTRNGS and CS_LANG, both of which are optional. CSSTRNGS points to a directory containing .xmg files. CS_LANG: Selects a language for csound messages. RAWWAVE_PATH: Is used by the STK opcodes to find the raw wave files. Only relevant if you are using STK wrapper opcodes like STKBowed or STKBrass. CSNOSTOP: If this environment variable is set to "yes", then any graph displays are closed automatically at the end of performance (meaning that you possibly will not see much of them in the case of a short non-realtime render). Otherwise, you need to click "Quit" in the FLTK display window to exit, allowing for viewing the graphs even after the end of score is reached. MFDIR: Default directory for MIDI files. Used if no full path is given for MIDI files. Please note that MIDI files are sought in SSDIR and SFDIR as well. CS_OMIT_LIBS: Allows defining a list of plugin libraries that should be skipped. Libraries can be separated with commas, and don't require the "lib" prefix. For more information about SFDIR, SSDIR, SADIR, MFDIR and INCDIR see Directories and files. The only mandatory environment variables are OPCODEDIR and OPCODEDIR64. It is very important to set them correctly, otherwise most of the opcodes will not be available. Make sure you set the path correctly depending on the precision of your binary. if you run csound on a command line without any arguments you should see some text like : Csound version 5.01.0 beta (float samples) Mar 23 2006. This text refers to the single precision version. CSSTRNGS and CS_LANG currently have very limited use since Csound has not yet been completely translated into other languages. Other environment variables which are not exclusive to Csound but which might be of importance are: PATH: The directory containing csound executables should be listed in this variable. PYTHONPATH: If you intend to use CsoundVST and python, the directory containing the _CsoundVST shared library and the CsoundVST.py file must be in your PYTHONPATH environment variable (or the default path python searches in), so that the Python runtime knows how to load these files. LADSPA_PATH and DSSI_PATH: These environment variables are required if you are using the dssi4cs (LADSPA and DSSI host) plug-in opcodes. CSDOCDIR: Specifies the directory where the html help files are located. Though not used by Csound directly, this environment variable can help front-ends and editors (which implement it) to find the csound manual. Setting environment variables On the command line You can set environment variables on the command line or the configuration file .csoundrc by using the command line flag --env:NAME=VALUE or --env:NAME+=VALUE, where NAME is the environment variable name, and VALUE is its value. See Command-line Flags Please note that this method of setting environment variables will not work for variables which are parsed before the command line arguments. SADIR, SSDIR, SFDIR, INCDIR, SNAPDIR, RAWWAVE_PATH, CSNOSTOP, SFOUTYP should work, but the following environment variables must be set on the system prior to running csound: OPCODEDIR, OPCODEDIR64, CSSTRINGS, and CS_LANG. CSOUNDRC can currently (v. 5.02) be set using --env, but this behavior is not guaranteed for future versions. Windows To set a csound environment on Windows XP and 2000 go to Control Panel->System->Advanced and click on the button 'Environment Variables'. On other versions of Windows earlier than Windows XP and Windows 2000 you set environment variables in the autoexec.bat file. Go to 'My Computer', select C: drive, right click on autoexec.bat, and select 'Edit'. The statement format is: SET NAME=VALUE . Linux You can set environment variables on Linux in many ways. You can set them using the export shell command, by setting them on .bashrc or similar files or by adding them to the /etc/profile file. Mac If the user has a Mac that shipped with an OS X version prior to 10.3 (includes 10.2 and 10.1) then it is possible that the default shell is the Tenex C-shell (tcsh). If this is the case, then you either have to type: ~% setenv OPCODEDIR "/Users/you/your/Csound5/build" or change your /etc/profile and or edit your .tcshrc file. If the user has a Mac that shipped with OS X 10.3 or 10.4 then it likely has the "Bourne-again" C-shell (bash) as the default shell. If this is the case, then the user must type something like: ~$ export OPCODEDIR=/Users/you/your/Csound5/build in addition if the bash shell is the default, then it is usually easier to edit your .bashrc or /etc/profile. Note that if users choose one of the above methods, ie editing the .bashrc file then the environment variables are executed when a new shell is created. This can be problematic if your application implements a Quartz or Aqua interface and does not use the commandline. If this is the case, then the standard solution (up to OS 10.3.9 and unless the application uses the csoundAPI and sets the environ variables directly) is to create an XML property list file (called a .plist file by the OS). This file should nominally be located at ~/.MacOSX/Environment.plist. This has been a solution specifically for the [csoundapi~] object for Pd on OS X. Since Pd uses an OS X native .app style packaging, and runs off of the Aqua interface, the standard means of supplying environment variables to Csound do not work. The solution is to set Csound's environment variables for the Aqua environment. Likely, most users will not have the hidden folder .MacOSX located in their $HOME directory (aka ~/) This folder must first be created and the Environment.plist added to this folder. The contents of the Environment.plist file should be something like: <?xml version="1.0" encoding='UTF-8"?> <!DOCTYPE plist PUBLIC "-//Apple Computer//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd"> <plist version="1.0"> <dict> <key>OPCODEDIR</key> <string>/Library/Frameworks/CsoundLib.framework/Versions/5.1/Resources/Opcodes</string> <key>OPCODEDIR64</key> <string>/Volumes/ExternalHD/devel/csound5/lib64</string> <key>INCDIR</key> <string>/Volumes/ExternalHD/CSOUND/include</string> <key>SFDIR</key> <string>/Volumes/ExternalHD/iTunes/csoundaudio</string> </dict> </plist> and so on, using the XML <key> tag for each environment variable required by the API and the <string> tag for it's corresponding path on the system. Please note that you must login out and login in for these changes to take effect.
manual_src~dfsg/command/flagscategoryXO.xml0000644000000000000000000016010712262561504020213 0ustar rootroot
Command-line Flags Listed below are the command line available in OLPCsound. The format of a command is either: csound orchname scorename or csound csdfilename where the arguments are of 2 types: flags arguments (beginning with a -,-- or -+), and name arguments (such as filenames). Certain flag arguments take a following name or numeric argument. Flags that start with -- and -+ usually take an argument themselves using =. Audio File Ouput -a-a, --format=alaw--format=alaw Use a-law audio samples. -f-f, --format=float--format=float Use single-format float audio samples (not playable on some systems, but can be read by -i, soundin and GEN01 --format=type--format Set the audio file output format to one of the formats available. At present the options are: aiff, au, avr, caf, flac, htk, ircam, mat4, mat5, nis, paf, pvf, raw, sd2, sds, svx, voc, w64, wav, wavex and xi. Can also be used as --format=type:format or --format=format:type to set both the file type (wav, aiff, etc.) and sample format (short, long, float, etc.) at the same time. -i FILE-i, --input=FILE--input Input soundfile name. If not a full pathname, the file will be sought first in the current directory, then in that given by the environment variable SSDIR (if defined), then by SFDIR. The name stdin will cause audio to be read from standard input. The name devaudio or adc will request sound from the host audio input device. It is possible to select a device number by appending an integer value in the range 0 to 1023, or a device name separated by a : character. It depends on the host audio interface whether a device number or a name should be used. In the first case, an out of range number usually results in an error and listing the valid device numbers. The audio coming in using -i can be received using opcodes like inch. -n-n, --nosound--nosound No sound. Do all processing, but bypass writing of sound to disk. This flag does not change the execution in any other way. -o FILE-o, --output=FILE--output Output soundfile name. If not a full pathname, the soundfile will be placed in the directory given by the environment variable SFDIR (if defined), else in the current directory. The name stdout will cause audio to be written to standard output, while null results in no sound output similarly to the -n flag. If no name is given, the default name will be test. The name dac or devaudio (you can use -odac or -o dac) will request writing sound to the host audio output device. It is possible to select a device number by appending an integer value in the range 0 to 1023, or a device name separated by a : character. It depends on the host audio interface whether a device number or a name should be used. In the first case, an out of range number usually results in an error and listing the valid device numbers. -s-s, --format=short--format=short Use short integer audio samples. -u-u, --format=ulaw--format=ulaw Use u-law audio samples. -W-W, --wave--wave, --format=wave--format=wave Write a WAV format soundfile. -Z-Z, --dither--dither Switch on dithering of audio conversion from internal floating point to 32, 16 and 8-bit formats. The default form of the dither is triangular. -Z-Zd, --dither--triangular--dither--triangular, --dither--uniform--dither--uniform Switch on dithering of audio conversion from internal floating point to 32, 16 and 8-bit formats. In the case of -Z the next digit should be a 1 (for trangular) or a 2 (for uniform). Output File Id tags -+id_artist=string-+id_artist (max. length = 200 characters) Artist tag in output soundfile (no spaces) -+id_comment=string-+id_comment (max. length = 200 characters) Comment tag in output soundfile (no spaces) -+id_copyright=string-+id_copyright (max. length = 200 characters) Copyright tag in output soundfile (no spaces) -+id_date=string-+id_date (max. length = 200 characters) Date tag in output soundfile (no spaces) -+id_software=string-+id_software (max. length = 200 characters) Software tag in output soundfile (no spaces) -+id_title=string-+id_title (max. length = 200 characters) Title tag in output soundfile (no spaces) Realtime Audio Input/Output -i adc[DEVICE]-i, --input=adc[DEVICE]--input The name devaudio or adc will request sound from the host audio input device. It is possible to select a device number by appending an integer value in the range 0 to 1023, or a device name separated by a : character (e.g. -iadc3, -iadc:hw:1,1). It depends on the host audio interface whether a device number or a name should be used. In the first case, an out of range number usually results in an error and listing the valid device numbers. -o dac[DEVICE]-o, --output=dac[DEVICE]--output The name dac or devaudio (you can use -odac or -o dac) will request writing sound to the host audio output device. It is possible to select a device number by appending an integer value in the range 0 to 1023, or a device name separated by a : character (e.g. -odac3, -odac:hw:1,1). It depends on the host audio interface whether a device number or a name should be used. In the first case, an out of range number usually results in an error and listing the valid device numbers. -+rtaudio=string-+rtaudio (max. length = 20 characters) Real time audio module name. The default is PortAudio (all platforms). Also available, depending on platform and build options: Linux: alsa, jack, pulse; Windows: mme; Mac OS X: CoreAudio. In addition, null can be used on all platforms, to disable the use of any real time audio plugin. -+server=string-+server Pulseaudio server name. -+output_stream=string-+output_stream Pulseaudio output stream name. -+input_stream=string-+input_stream Pulseaudio input stream name. -+jack_client=[client_name]-+jack_client The client name used by Csound, defaults to 'csound5'. If multiple instances of Csound connect to the JACK server, different client names need to be used to avoid name conflicts. (Linux and Mac OS X only) -+jack_inportname=[input port name prefix], -+jack_outportname=[output port name prefix] -+jack_inportname -+jack_outportname Name prefix of Csound JACK input/output ports; the default is 'input' and 'output'. The actual port name is the channel number appended to the name prefix. (Linux and Mac OS X only) Example: with the above default settings, a stereo orchestra will create these ports in full duplex operation: csound5:input1 (record left) csound5:input2 (record right) csound5:output1 (playback left) csound5:output2 (playback right) MIDI File Input/Ouput -F FILE-F, --midifile=FILE--midifile Read MIDI events from MIDI file FILE. The file should have only one track in Csound versions 4.xx and earlier; this limitation is removed in Csound 5.00. --midioutfile=FILENAME--midioutfile Save MIDI output to a file (Csound 5.00 and later only). -+mute_tracks=string-+mute_tracks (max. length = 255 characters) Ignore events (other than tempo changes) in MIDI file tracks defined by pattern (for example, -+mute_tracks=00101 will mute the third and fifth tracks). -+raw_controller_mode=boolean-+raw_controller_mode Disable special handling of MIDI controllers like sustain pedal, all notes off etc., allowing the use of all the 128 controllers for any purpose. This will also set the initial value of all controllers to zero. Default: no. -+skip_seconds=float-+skip_seconds (min: 0) Start playback at the specified time (in seconds), skipping earlier events in the score and MIDI file. -T-T, --terminate-on-midi--terminate-on-midi Terminate the performance when the end of MIDI file is reached. MIDI Realtime Input/Ouput -M DEVICE-M, --midi-device=DEVICE--midi-device Read MIDI events from device DEVICE. If using ALSA MIDI (-+rtmidi=alsa), devices are selected by name and not number. So, you need to use an option like -M hw:CARD,DEVICE where CARD and DEVICE are the card and device numbers (e.g. -M hw:1,0). In the case of PortMidi and MME, DEVICE should be a number, and if it is out of range, an error occurs and the valid device numbers are printed. --midi-key=N--midi-key=N Route MIDI note on message key number to pfield N as MIDI value [0-127]. --midi-key-cps=N--midi-key-cps=N Route MIDI note on message key number to pfield N as cycles per second. --midi-key-oct=N--midi-key-oct=N Route MIDI note on message key number to pfield N as linear octave. --midi-key-pch=N--midi-key-pch=N Route MIDI note on message key number to pfield N as oct.pch (pitch class). --midi-velocity=N--midi-velocity=N Route MIDI note on message velocity number to pfield N as MIDI value [0-127]. --midi-velocity-amp=N--midi-velocity-amp=N Route MIDI note on message velocity number to pfield N as amplitude [0-0dbFS]. --midioutfile=FILENAME--midioutfile Save MIDI output to a file (Csound 5.00 and later only). -+rtmidi=string-+rtmidi (max. length = 20 characters) Real time MIDI module name. Defaults to PortMidi, other options (depending on build options): Linux: alsa; Windows: mme, winmm. In addition, null can be used on all platforms, to disable the use of any real time MIDI plugin. ALSA MIDI devices are selected by name and not number. So, you need to use an option like -M hw:CARD,DEVICE where CARD and DEVICE are the card and device numbers (e.g. -M hw:1,0). -Q DEVICE-Q Enables MIDI OUT operations to device id DEVICE. This flag allows parallel MIDI OUT and DAC performance. Unfortunately the real-time timing implemented in Csound is completely managed by DAC buffer sample flow. So MIDI OUT operations can present some time irregularities. These irregularities can be reduced by using a lower value for the -b flag. If using ALSA MIDI (-+rtmidi=alsa), devices are selected by name and not number. So, you need to use an option like -Q hw:CARD,DEVICE where CARD and DEVICE are the card and device numbers (e.g. -Q hw:1,0). In the case of PortMidi and MME, DEVICE should be a number, and if it is out of range, an error occurs and the valid device numbers are printed. Display --csd-line-nums=NUM--csd-line-nums Determines how line numbers are counted and displayed for error messages when processing a Csound Unified Document file (.csd). This flag has no effect if separate orchestra and score files are used. (Csound 5.08 and later). 0 = line numbers are relative to the beginning of the orchestra or score sections of the CSD 1 = line numbers are relative to the beginning of the CSD file. This is the default as of Csound 5.08. --displays--displays Enables displays, reverting the effect of any previous -d flag. -g-g, --asciidisplay--asciidisplay Suppress graphics, use ASCII displays instead. -m NUM-m, --messagelevel=NUM--messagelevel Message level for standard (terminal) output. Takes the sum of any of the following values: 1 = note amplitude messages 2 = samples out of range message 4 = warning messages 128 = print benchmark information And exactly one of these to select note amplitude format: 0 = raw amplitudes, no colours 32 = dB, no colors 64 = dB, out of range highlighted with red 96 = dB, all colors 256 = raw, out of range highlighted with red 512 = raw, all colours The default is 135 (128+4+2+1), which means all messages, raw amplitude values, and printing elapsed time at the end of performance. The coloring of raw amplitudes was introduced in version 5.04 --m-amps=NUM Message level for amplitudes on standard (terminal) output. 0 = no note amplitude messages 1 = note amplitude messages --m-range=NUM Message level for out of range messages on standard (terminal) output. 0 = no samples out of range message 1 = samples out of range message --m-warnings=NUM Message level for warnings on standard (terminal) output. 0 = no warning messages 1 = warning messages --m-dB=NUM Message level for amplitude format on standard (terminal) output. 0 = absolute amplitude messages 1 = dB amplitude messages --m-colours=NUM Message level for amplitude format on standard (terminal) output. 0 = no colouring of amplitude messages 1 = colouring of amplitude messages --m-benchmarks=NUM Message level for benchmark information on standard (terminal) output. 0 = no benchnark numbers 1 = print benchnark numbers -+msg_color=boolean-+msg_color Enable message attributes (colors etc.); might need to be disabled on some terminals which print strange characters instead of modifying text attributes. default: true. -v-v, --verbose--verbose Verbose translate and run. Prints details of orch translation and performance, enabling errors to be more clearly located. Performance Configuration and Control -B NUM-B, --hardwarebufsamps=NUM--hardwarebufsamps Number of audio sample-frames held in the DAC hardware buffer. This is a threshold on which software audio I/O (above) will wait before returning. A small number reduces audio I/O delay; but the value is often hardware limited, and small values will risk data lates. In the case of portaudio output (the default real-time output), the -B parameter (more precisely, -B / sr) is passed as the "suggested latency" value. Other than that, Csound has no control over how PortAudio interprets the parameter. The default is 1024 on Linux, 4096 on Mac OS X and 16384 on Windows. -b NUM-b, --iobufsamps=NUM--iobufsamps Number of audio sample-frames per sound i/o software buffer. Large is efficient, but small will reduce audio I/O delay and improve the accuracy of the timing of real time events. The default is 256 on Linux, 1024 on MacOS X, and 4096 on Windows. In real-time performance, Csound waits on audio I/O on NUM boundaries. It also processes audio (and polls for other input like MIDI) on orchestra ksmps boundaries. The two can be made synchronous. For convenience, if NUM is negative, the effective value is ksmps * -NUM (audio synchronous with k-period boundaries). With NUM small (e.g. 1) polling is then frequent and also locked to fixed DAC sample boundaries. Note: if both -iadc and -odac are used at the same time (full duplex real time audio), the -b option should be set to an integer multiple of ksmps. -k NUM-k, --control-rate=NUM--control-rate Override the control rate (KR) supplied by the orchestra. -L DEVICE-L, --score-in=DEVICE--score-in Read line-oriented real-time score events from device DEVICE. The name stdin will permit score events to be typed at your terminal, or piped from another process. Each line-event is terminated by a carriage-return. Events are coded just like those in a standard numeric score, except that an event with p2=0 will be performed immediately, and an event with p2=T will be performed T seconds after arrival. Events can arrive at any time, and in any order. The score carry feature is legal here, as are held notes (p3 negative) and string arguments, but ramps and pp or np references are not. --omacro:XXX=YYY--omacro Set orchestra macro XXX to value YYY -r NUM-r, --sample-rate=NUM--sample-rate Override the sampling rate (SR) supplied by the orchestra. --sched--sched Linux only. Use real-time scheduling and lock memory. (Also requires -d and either -o dac or -o devaudio). See also --sched=N below. --sched=N--sched=N Linux only. Same as --sched, but allows specifying a priority value: if N is positive (in the range 1 to 99) the scheduling policy SCHED_RR will be used with a priority of N; otherwise, SCHED_OTHER is used with the nice level set to N. Can also be used in the format --sched=N,MAXCPU,TIME to enable the use of a "watchdog" thread that terminates Csound if the average CPU usage exceeds MAXCPU percents over a peroid of TIME seconds (new in Csound 5.00). --smacro:XXX=YYY--smacro Set score macro XXX to value YYY --strset--strset Csound 5. The --strset option allows setting strset string values from the command line, in the format '--strsetN=VALUE'. It is useful for passing parameters to the orchestra (e.g. file names). -+skip_seconds=float-+skip_seconds (min: 0) Start playback at the specified time (in seconds), skipping earlier events in the score and MIDI file. -t NUM-t, --tempo=NUM--tempo=NUM Use the uninterpreted beats of score.srt for this performance, and set the initial tempo at NUM beats per minute. When this flag is set, the tempo of score performance is also controllable from within the orchestra. WARNING: this mode of operation is experimental and may be unreliable. Miscellaneous -@ FILE-@ Provide an extended command-line in file FILE --default-paths--no-default-paths Reenables adding of directory of CSD/ORC/SCO to search paths, if it has been disabled by a previous --no-default-paths (e.g. in .csoundrc). -D-D, --defer-gen1--defer-gen1 Defer GEN01 soundfile loads until performance time. --env:NAME=VALUE--env:NAME=VALUE Set environment variable NAME to VALUE. Note: not all environment variables can be set this way, because some are read before parsing the command line. INCDIR, SADIR, SFDIR, and SSDIR are known to work. --env:NAME+=VALUE--env:NAME+=VALUE Append VALUE to ';' separated list of search paths in environment variable NAME (should be INCDIR, SADIR, SFDIR, or SSDIR). If a file is found in multiple directories, the last will be used. --expression-opt--expression-opt Since Csound 5. Turns on some optimizations in expressions: Redundant assignment operations are eliminated whenever possible. This means that for example this line a1 = a2 + a3 will compile as a1 Add a2, a3 instead of #a0 Add a2, a3 a1 = #a0 saving a temporary variable and an opcode call. Less opcode calls result in reduced CPU usage (an average orchestra may compile about 10% faster with --expression-opt, but it depends largely on how many expressions are used, what the control rate is (see also below), etc.; thus, the difference may be less, but also much more). number of a- and k-rate temporary variables is significantly reduced. This expression (a1 + a2 + a3 + a4) will compile as #a0 Add a1, a2 #a0 Add #a0, a3 #a0 Add #a0, a4 ; (the result is in #a0) instead of #a0 Add a1, a2 #a1 Add #a0, a3 #a2 Add #a1, a4 ; (the result is in #a2) The advantages of less temporary variables are: less cache memory is used, which may improve performance of orchestras with many a-rate expressions and a low control rate (e.g. ksmps = 100) large orchestras may load faster due to less different identifier names index overflow errors (i.e. when messages like this Case2: indx=-56004 (ffff253c); (short)indx = 9532 (253c) are printed and odd behavior or a Csound crash occurs) may be fixed, because such errors are triggered by too many different (especially a-rate) variable names in a single instrument. Note that this optimization (due to technical reasons) is not performed on i-rate temporary variables. Warning When --expression-opt is turned on, it is not allowed to use the i() function with an expression argument, and relying on the value of k-rate expressions at i-time is unsafe. --help--help Display on-line help message. -I-I, --i-only--i-only i-time only. Allocate and initialize all instruments as per the score, but skip all p-time processing (no k-signals or a-signals, and thus no amplitudes and no sound). Provides a fast validity check of the score pfields and orchestra i-variables. This option is exclusive of the --syntax-check-only flag. -+ignore_csopts=integer-+ignore_csopts= If set to 1, Csound will ignore all options specified in the csd file's CsOptions section. See Unified File Format for Orchestras and Scores. -+max_str_len=integer-+max_str_len (min: 10, max: 10000) Maximum length of string variables + 1; defaults to 256 allowing a length of 255 characters. The length of string constants is not limited by this parameter. --no-default-paths--no-default-paths Disables adding of directory of CSD/ORC/SCO to search paths. --no-expression-opt--no-expression-opt Disables expression optimization. -O FILE-O, --logfile=FILE--logfile Log output to file FILE. If FILE is null (i.e. -O null or --logfile=null) all printing of messages to the console is disabled. --syntax-check-only--syntax-check-only Causes Csound to exit immediately after the orchestra and score parsers finish checking the syntax of the input files and before the orchestra performs the score. This option is exclusive of the --i-only flag. (Csound 5.08 and later). -t0-t0, --keep-sorted-score--keep-sorted-score Prevents Csound from deleting the sorted score file, score.srt, upon exit. -U UTILITY-U, --utility=UTILITY--utility Invoke the utility program UTILITY. Use any invalid name to list the available utilities.
manual_src~dfsg/quickref.py0000644000000000000000000002322312262561504015132 0ustar rootroot# -*- coding: utf-8 -*- # This script generates the misc/quickref.html file # by Andres Cabrera June 2006 # Licensed under the GPL licence version 3 or later # modification for empty arg in command and links on opcodes by Francois Pinot February 2007 from xml.dom import minidom import os, glob, sys # categories holds the list of valid categories for opcodes from categories import categories XO = False opcodelist = [] if sys.argv.count("-x"): XO = True file = open("opcode_listXO.txt", 'r') removedopcodes = file.read().split() file.close() entries = [] example_cats = [] for i in categories: entries.append([]) example_cats.append([]) # Generate a file listing all examples outfilename = 'misc/examples.xml' examples = open(outfilename,'w') examples.write("\n\n") examples.write(''' List of examples''') files = glob.glob('opcodes/*.xml') try: files.remove('opcodes/template.xml') except: pass files.sort() for i,filename in enumerate(files): entry = '' #print filename source = open(filename, 'r') text = source.read() source.close() cat_text = text[text.find("") + 21: text.find("")] if categories.count(cat_text) > 0: cat = categories.index(cat_text); #print cat_text, cat while text.find("examples-xml/") != -1: file_to_add = text[text.find("examples-xml/"):text.find(".csd",text.find("examples-xml/") )+ 4] if len(file_to_add) < 45: #yes, very flaky, I know... entries[cat].append(file_to_add) else: print file_to_add text = text[text.find("examples-xml/") + 10:] for i,catname in enumerate(categories): print catname if len(entries[i]) > 0: title ="" title += catname #title += "\n" title += "\n" examples.write(title) for j,ex in enumerate(entries[i]): if ex != "": line = '' + ex.replace("examples-xml/",'') print line #line += "\n" line += "\n" examples.write(line) examples.write("\n") examples.write('\n') examples.close() #Now generate the quickref file #Reset entries entries = [] for i in categories: entries.append([]) outfilename = 'misc/quickref.xml' if not XO else 'misc/quickrefXO.xml' quickref = open(outfilename,'w') quickref.write("\n\n") quickref.write(''' Opcode Quick Reference Opcode Quick Reference''') manualfilename = 'manual.xml' if not XO else 'manualXO.xml' manual = open(manualfilename, 'r') text = manual.read() manual.close() files = glob.glob('opcodes/*.xml') try: files.remove('opcodes/template.xml') except: pass files[len(files):]=glob.glob('opcodes/*/*.xml') files[len(files):]=glob.glob('vectorial/*.xml') files[len(files):]=glob.glob('utility/*.xml') files.sort() headerText = text[0:text.find(' 0): # print 'Trimming file: ', filename, ' ', refStart newfile = newfile[refStart:] # Necessary to define entities newfile = headerText + newfile #print text try: xmldoc = minidom.parseString(newfile) except: print '>>> Failed to parse:',filename continue xmldocId = xmldoc.documentElement.getAttribute('id') # Some files need special treatment (adds, dollar, divides, modulus, multiplies, # opbitor, opor, raises, subtracts, assign, ifdef, ifndef, define, include, undef) # There must be a better way to avoid loosing the entities when parsing the XML # file. Anyone??? if (filename == 'opcodes' + os.sep + 'adds.xml'): entry = 'a '+'+ b (no rate restriction)\n' elif (filename == 'opcodes' + os.sep + 'dollar.xml'): entry = ''+'$NAME \n' elif (filename == 'opcodes' + os.sep + 'divides.xml'): entry = 'a '+'/ b (no rate restriction)\n' elif (filename == 'opcodes' + os.sep + 'modulus.xml'): entry = 'a '+'% b (no rate restriction)\n' elif (filename == 'opcodes' + os.sep + 'multiplies.xml'): entry = 'a '+'* b (no rate restriction)\n' elif (filename == 'opcodes' + os.sep + 'opbitor.xml'): entry = 'a '+'| b (bitwise OR)\n' elif (filename == 'opcodes' + os.sep + 'opor.xml'): entry = 'a '+'|| b (logical OR; not audio-rate)\n' elif (filename == 'opcodes' + os.sep + 'raises.xml'): entry = 'a '+'ˆ b (b not audio-rate)\n' elif (filename == 'opcodes' + os.sep + 'subtracts.xml'): entry = 'a '+'− b (no rate restriction)\n' # elif (filename == 'opcodes' + os.sep + 'assign.xml'): # entry = ''' '+'−a (no rate restriction) # '+'+a (no rate restriction)\n''' elif (filename == 'opcodes' + os.sep + 'ifdef.xml'): entry = ''+'#ifdef NAME ....' + \ ''+'#else  ....' + \ ''+'#end \n' elif (filename == 'opcodes' + os.sep + 'ifndef.xml'): entry=''+'#ifndef NAME ....' + \ ''+'#else  ....' + \ ''+'#end \n' elif (filename == 'opcodes' + os.sep + 'define.xml'): entry=''+'#define NAME # replacement text #\n' + \ ''+'#define NAME(a' b' c') # replacement text #\n' elif (filename == 'opcodes' + os.sep + 'include.xml'): entry=''+'#include "filename"\n' elif (filename == 'opcodes' + os.sep + 'undef.xml'): entry=''+'#undef NAME\n' else: synopsis = xmldoc.getElementsByTagName('synopsis') if (len(synopsis) != 0): # There can be more than 1 synopsis per file for num in range(len(synopsis)): tmp = synopsis[num].toxml() if XO: opcodename = tmp[tmp.find('') + 9:tmp.find('')] else: opcodename = "" if XO and removedopcodes.count(opcodename) == 0: print "Removed ----------------------", opcodename else: if tmp[-21:] == "": # no arg, insert nbsp tmp = tmp[:-11] + " " tmp = tmp.replace('', '') entry += tmp.replace('', '') if entry != '': entry += '' else: #print "no synopsis tag for file: " + file entry = '' #print "Entry ------ ", entry info = xmldoc.getElementsByTagName('refentryinfo') if (len(info)!=0 and entry != ''): category = info[0].toxml() category = category[21:-23] print category else: print "no refentryinfo tag for file " + filename category = "Miscellaneous" if (entry!=''): print filename + " sent to Miscellaneous" #print category match = False for j, thiscategory in enumerate(categories): if (category == thiscategory): entries[j].append(entry+ '\n') match = True if match == False: print filename + "WARNING! No Category Match!" for i in range(len(categories)): if (len(entries[i])==0): print "No entries for category: "+categories[i]+"...Skipping" continue quickref.write("\n") quickref.write(""+ categories[i] + "\n\n\n") count = 0 for j in range(len(entries[i])): quickref.write(entries[i].pop(0)) # + '\n') count += 1 quickref.write("\n") print str(count) + " entries in category: " + categories[i] quickref.write('\n') quickref.close() print entries manual_src~dfsg/sigmod/0000755000000000000000000000000012262561504014227 5ustar rootrootmanual_src~dfsg/sigmod/panspatlXO.xml0000644000000000000000000000623412262561504017047 0ustar rootroot
Panning and Spatialization Amplitude spatialization locsend locsig pan pan2 space spdist spsend Binaural spatialization hrtfer hrtfmove hrtfmove2 hrtfstat Ambisonics bformdec bformenc
manual_src~dfsg/sigmod/siglimit.xml0000644000000000000000000000074012262561504016573 0ustar rootroot
Signal Limiters Opcodes that can be used to limit signals are: limit mirror wrap
manual_src~dfsg/sigmod/reverbtn.xml0000644000000000000000000000236112262561504016602 0ustar rootroot
Reverberation The opcodes one can use for reverberation are: alpass babo comb freeverb nestedap nreverb (also called reverb2) reverb reverbsc valpass vcomb
manual_src~dfsg/sigmod/panspatl.xml0000644000000000000000000000621612262561504016600 0ustar rootroot
Panning and Spatialization Amplitude spatialization locsend locsig pan pan2 space spdist spsend 3D spatialization with simulation of room acoustics spat3d spat3di spat3dt Vector Base Amplitude Panning vbap16 vbap16move vbap4 vbap4move vbap8 vbap8move vbaplsinit vbapz vbapzmove Binaural spatialization hrtfer hrtfmove hrtfmove2 hrtfstat Ambisonics bformdec bformenc
manual_src~dfsg/sigmod/delayops.xml0000644000000000000000000000472512262561504016601 0ustar rootroot
Delay Fixed delays delay delay1 delayk Delay Lines delayr delayw deltap deltap3 deltapi deltapn deltapx deltapxw Variable delays vdelay vdelay3 vdelayx vdelayxs vdelayxq vdelayxw vdelayxwq vdelayxws Multitap delays multitap
manual_src~dfsg/sigmod/speciali.xml0000644000000000000000000000263112262561504016544 0ustar rootroot
Specialized Filters High pass filters dcblock dcblock2 Parametric EQ pareq rbjeq eqfil Other filters nlfilt filter2 fofilter hilbert zfilter2
manual_src~dfsg/sigmod/wavguide.xml0000644000000000000000000000075712262561504016575 0ustar rootroot
Waveguides The opcodes that use waveguides to modify a signal are: streson wguide1 wguide2
manual_src~dfsg/sigmod/compaccum.xml0000644000000000000000000000275112262561504016725 0ustar rootroot
Comparators and Accumulators The following opcodes perform comparisons between signals at a-rate or k-rate, find maxima or minima, or accumulate the results of several computations or comparisons: max max_k maxabs maxabsaccum maxaccum min minabs minabsaccum minaccum vincr clear
manual_src~dfsg/sigmod/top.xml0000644000000000000000000000045212262561504015554 0ustar rootroot Signal Modifiers &sigmodampmod; &sigmodconmorph; &sigmoddelayops; &sigmodpanspatl; &sigmodreverbtn; &sigmodsample; &sigmodsiglimit; &sigmodspeciale; &sigmodstandard; &sigmodspeciali; &sigmodwavguide; &sigmodwavshape; manual_src~dfsg/sigmod/sample.xml0000644000000000000000000000304212262561504016231 0ustar rootroot
Sample Level Operators The opcodes one may use to modify signals are: a(k) denorm diff downsamp fold i(k) integ interp k(i) ntrpol samphold upsamp vaget vaset
manual_src~dfsg/sigmod/speciale.xml0000644000000000000000000000143112262561504016535 0ustar rootroot
Special Effects Opcodes that generate special effects are: distort distort1 flanger harmon phaser1 phaser2
manual_src~dfsg/sigmod/ampmod.xml0000644000000000000000000000154712262561504016235 0ustar rootroot
Amplitude Modifiers and Dynamic processing The opcodes that modify amplitude are: balance compress clip dam gain The opcode 0dbfs facilitates the use of amplitude by removing the need to use of explicit sample values.
manual_src~dfsg/sigmod/wavshape.xml0000644000000000000000000000326612262561504016576 0ustar rootroot
Waveshaping and Phase Distortion These opcodes can perform dynamic waveshaping or phaseshaping (a.k.a. phase distortion). They differ from traditional table-based methods of waveshaping by directly calculating the transfer function with one or more variable parameters for affecting the amount or results of the shaping. Most of these opcodes could be used on either an audio signal (for waveshaping) or a phasor (for phaseshaping) but tend to work best for one of these applications. These opcodes are good for waveshaping: chebyshevpoly clip distort distort1 polynomial powershape These opcodes are good for phaseshaping: pdclip pdhalf pdhalfy
manual_src~dfsg/sigmod/standard.xml0000644000000000000000000001006512262561504016553 0ustar rootroot
Standard Filters Resonant Low-pass filters areson lowpass2 lowres lowresx lpf18 moogvcf moogladder reson resonr resonx resony resonz rezzy statevar svfilter tbvcf vlowres bqrez Standard filters Hi-pass filters: atone, atonex Low-pass filters: tone, tonex Biquad filters: biquad and biquada. Butterworth filters: butterbp, butterbr, butterhp, butterlp (which are also called butbp, butbr, buthp, butlp) General filters: clfilt Control signal filters aresonk atonek lineto port portk resonk resonxk tlineto tonek
manual_src~dfsg/sigmod/conmorph.xml0000644000000000000000000000154312262561504016601 0ustar rootroot
Convolution and Morphing The opcodes that convolve and morph signals are: convolve also called convle cross2 dconv ftconv ftmorf pconvolve
manual_src~dfsg/cscore/0000755000000000000000000000000012262561504014223 5ustar rootrootmanual_src~dfsg/cscore/compile.xml0000644000000000000000000003173512262561504016406 0ustar rootroot
Compiling a Cscore Program A Cscore program can be invoked either as a standalone program or as part of Csound in between sorting the score and performing the score with the orchestra: cscore scorefilein > scorefileout or csound -C orchname scorename Before trying to compile your own Cscore program, you will most likely want to obtain a copy of the Csound source code. Either download the latest source distribution for your platform or check out a copy of the csound5 module from Sourceforge CVS. There are several files in the sources that will help you. Within the examples/cscore/ directory are a number of examples of Cscore control programs, including all of the examples contained in this manual. And in the frontends/cscore/ directory are the two files cscoremain.c and cscore.c. cscoremain.c contains a simple main function that performs all of the initialization that a standalone Cscore program needs to do before it calls your control function. This main stub initializes Csound, reads the commandline arguments, opens the input and output score files, and then calls a function cscore(). As described above, it is expected that you will write the cscore() function and provide it in another file. The file frontends/cscore/cscore.c shows the simplest example of a cscore() function that reads in a score of any length and writes it to the output unchanged. So, to create a standalone program, write a control program as shown in the previous section. Let's assume that you saved this program in a file named mycscore.c. Next, you need to compile and link this program with the Csound library and cscoremain.c in order to create an exectuable by following the set of directions below that apply to your operating system. It will be helpful to already have some familiarity with the C compiler on your computer since the information below cannot be complete for all possible systems. Linux and Unix The following commands assume that you have copied your file mycscore.c into the same directory as cscoremain.c, that you have opened a terminal to that same directory, and that you have previously installed a binary distribution of Csound that placed a library libcsound.a or libcsound.so into /usr/local/lib and the header files for the Csound API into /usr/local/include/csound. To compile and link: gcc mycscore.c cscoremain.c -o cscore -lcsound -L/usr/local/lib -I/usr/local/include/csound To run (sending the results to standard output): ./cscore test.sco It is possible that on some Unix systems, the C compiler will be named cc or something else other than gcc. Windows Csound is usually compiled on Windows using the MinGW environment that makes GCC -- the same compiler used on Linux -- available using a Unix-like command shell (MSYS). Since pre-compiled libraries for Csound on Windows are built in this way, you may need to use MinGW as well to link to them. If you have built Csound using another compiler, then you should be able to build Cscore with that compiler as well. Compiling standalone Cscore programs using MinGW should be similar to the procedure for Linux above with library and header paths changed appropriately for where Csound is installed on the Windows system. (Please feel free to contribute more detailed instructions here as the editor has been unable to test Cscore on a Windows machine). OS X The following commands assume that you have copied your file mycscore.c into the same directory as cscoremain.c and that you have opened a terminal to that same directory. In addition, the Apple-supplied developer tools (including the GCC compiler) should be installed on your system and you should have previously installed a binary distribution of Csound that placed the CsoundLib framework into /Library/Frameworks. Use this command compile and link. (You may get a warning about "multiple definitions of symbol _cscore"). gcc cscore.c cscoremain.c -o cscore -framework CsoundLib -I/Library/Frameworks/CsoundLib.framework/Headers To run (sending the results to standard output): ./cscore test.sco MacOS 9 You will need CodeWarrior or some other development environment installed on your computer (MPW may work). Download the source code distribution for OS 9 (it will have a name like Csound5.05_OS9_src.smi.bin). If using CodeWarrior, find and open the project file "Cscore5.cw8.mcp" in the folder "Csound5.04-OS9-source:macintosh:Csound5Library:". This project file is configured to use the source files cscore.c and cscoremain_MacOS9.c from the csound5 source tree and the Csound5Lib shared library produced by compiling Csound with the "Csound5.cw8.mcp" project file. You should substitute your own Cscore program file for cscore.c and either compile Csound5Lib first or substitute a copy of the library in the project from the binary distribution of Csound for OS 9. The file cscoremain_MacOS9.c contains specialized code for configuring CodeWarrior's SIOUX console library and allows commandline arguments to be entered before the program is run. Once you have the proper files included in the project window, click the "Make" button and CodeWarrior should produce an application named Cscore. When you run this application, it first displays a window allowing you to type in the arguments to the main function. You only need to type in the filename or pathname to the input score -- do not type in "cscore". The input file should be in the same folder as the application or else you will need to type a full or relative pathname to the file. Output will be displayed in the console window. You can use the Save command from the File menu before quitting if you wish. Alternatively, in the commandline dialog, you can choose to redirect the output to a file by clicking on the File button on the right side of the dialog. (Note that the console window can only display about 32,000 characters, so writing to a file is necessary for long scores). Making Cscore usable from within Csound To operate from Csound, first follow the instructions for compiling Csound (see Building Csound) according to the operating system that you are using. Once you have successfully built an unmodified Csound system, then substitute your own cscore() function for the one in the file Top/cscore_internal.c, and rebuild Csound. The resulting executable is your own special Csound, usable as above. The -C flag will invoke your Cscore program after the input score is sorted into score.srt. The details of what happens when you run Csound with the -C flag are given in the next section. Csound 5 also provides an additional way to run your own Cscore program from within Csound. Using the API, a host application can set a Cscore callback function, which is a function that Csound will call instead of using the built-in cscore() function. One advantage of this approach is that it is not necessary to recompile the entirety of Csound. Another benefit is that the host application can select at runtime from more than one Cscore function to designate as the callback. The disadvantage is that you need to write a host application. A simple approach to using a Cscore callback via the API would be to modify the standard Csound main program -- which is a simple Csound host -- contained in the file frontends/csound/csound_main.c. Adding a call to csoundSetCscoreCallback() after the call to csoundCreate() but before the call to csoundCompile() should do the job. Recompiling this file and linking to an existing Csound library will make a commandline version of Csound that works similarly to the one described above. Don't forget to use the -C flag. Notes about score formats and run-time behavior As stated previously, the input files to Cscore may be in original or time-warped and pre-sorted form; this modality will be preserved (section by section) in reading, processing, and writing scores. Standalone processing will most often use unwarped sources and create unwarped new files. When running from within Csound, the input score will arrive already warped and sorted, and can thus be sent directly (normally section by section) to the orchestra. One advantage of this method of using Cscore is that all of the syntactical conveniences of the full Csound score language may be used -- macros, arithmetic expressions, carry, ramp, etc. -- since the score will go through the "Carry, Tempo, Sort" phases of score processing before being passed to the user-supplied Cscore program. When running within Csound, a list of events can be conveyed to a Csound orchestra using cscoreListPlay(). There may be any number of cscoreListPlay() calls in a Cscore program. Each list so conveyed can be either time-warped or not, but each list must be in strict p2-chronological order (either from presorting or using cscoreListSort()). If there is no cscoreListPlay() in a Cscore module run from within Csound, all events written out (via cscorePutEvent(), cscorePutString(), or cscoreListPut()) are written to a new score in the current directory with the name cscore.out. Csound then invokes the score sorter again before sending this new score to the orchestra for performance. The final, sorted, output score is written to a file named cscore.srt. A standalone Cscore program will normally use the put commands to write into its output file. If a standalone Cscore program calls cscoreListPlay(), the events thus intended for performance will be sent to the output in the same way as if cscoreListPut() had been called instead. A note list sent by cscoreListPlay() for performance should be temporally distinct from subsequent note lists. No note-end should extend past the next list's start time, since cscoreListPlay() will complete each list before starting the next (i.e. like a Section marker that doesn't reset local time to zero). This is important when using cscoreListGetNext() or cscoreListGetUntil() to fetch and process score segments prior to performance, because these functions may only read part of an unsorted section.
manual_src~dfsg/cscore/main.xml0000644000000000000000000004230112262561504015671 0ustar rootroot
Writing a Cscore Control Program The general format for a Cscore control program is: #include "cscore.h" void cscore(CSOUND *cs) { /* VARIABLE DECLARATIONS */ /* PROGRAM BODY */ } The include statement will define the event and list structures and all of the Cscore API functions for the program. The name of the user function needs to be cscore if it will be linked with the standard main program in cscormai.c or linked as the internal Cscore routine for a personal Csound executable. This cscore() function receives one argument from cscormai.c or Csound -- CSOUND *cs -- which is a pointer to a Csound object. The pointer cs must be passed as the first parameter to every Cscore API function that the program calls. The following C program will read from a standard numeric score, up to (but not including) the first s or e statement, then write that data (unaltered) as output. #include "cscore.h" void cscore(CSOUND *cs) { EVLIST *a; /* a is allowed to point to an event list */ a = cscoreListGetSection(cs); /* read events in, return the list pointer */ cscoreListPut(cs, a); /* write these events out (unchanged) */ cscorePutString(cs, "e"); /* write the string e to output */ } After execution of cscoreListGetSection(), the variable a points to a list of event addresses, each of which points to a stored event. We have used that same pointer to enable another list function -- cscoreListPut() -- to access and write out all of the events that were read. If we now define another symbol e to be an event pointer, then the statement e = a->e[4]; will set it to the contents of the 4th slot in the EVLIST structure, a. The contents is a pointer to an event, which is itself comprised of an array of parameter field values. Thus the term e->p[5] will mean the value of parameter field 5 of the 4th event in the EVLIST denoted by a. The program below will multiply the value of that pfield by 2 before writing it out. #include "cscore.h" void cscore(CSOUND *cs) { EVENT *e; /* a pointer to an event */ EVLIST *a; a = cscoreListGetSection(cs); /* read a score as a list of events */ e = a->e[4]; /* point to event 4 in event list a */ e->p[5] *= 2; /* find pfield 5, multiply its value by 2 */ cscoreListPut(cs, a); /* write out the list of events */ cscorePutString(cs, "e"); /* add a "score end" statement */ } Now consider the following score, in which p[5] contains frequency in Hz. f 1 0 257 10 1 f 2 0 257 7 0 300 1 212 .8 i 1 1 3 0 440 10000 i 1 4 3 0 256 10000 i 1 7 3 0 880 10000 e If this score were given to the preceding main program, the resulting output would look like this: f 1 0 257 10 1 f 2 0 257 7 0 300 1 212 .8 i 1 1 3 0 440 10000 i 1 4 3 0 512 10000 ; p[5] has become 512 instead of 256. i 1 7 3 0 880 10000 e Note that the 4th event is in fact the second note of the score. So far we have not distinguished between notes and function table setup in a numeric score. Both can be classed as events. Also note that our 4th event has been stored in e[4] of the structure. For compatibility with Csound pfield notation, we will ignore p[0] and e[0] of the event and list structures, storing p1 in p[1], event 1 in e[1], etc. The Cscore functions all adopt this convention. As an extension to the above, we could decide to use the same pointers a and e to examine each of the events in the list. Note that e was not set to the numeral 4, but to the location of the 4th slot in the list. To inspect p5 of the previous event in the list, we need only redefine e with the assignment e = a->e[3]; and reference the 5th slot of the pfield array using the expression e->p[5] More generally, we can use an integer variable as an index to the array e[], and access each event in sequence by using a loop and incrementing the index. The number of events stored in an EVLIST is contained in the nevents member of the struct. int index; /* start with e[1] because e[0] is not used */ for (index = 1; index <= a->nevents; index++) { e = a->e[index]; /* do something with e */ } The above example starts with e[1] and increases the index each time through the loop (index++) until it is greater than a->nevents, the index of the last event in the list. The statements inside the for loop do execute a final time when index equals a->nevents. In the following program we will use the same input score. This time we will separate the ftable statements from the note statements. We will next write the three note-events stored in the list a to the output, then create a second score section consisting of the original pitch set and a transposed version of itself. This will bring about an octave doubling. Here, our index to the array is n and we increment n as part of a for block which iterates nevents times, allowing one statement to act upon the same pfield of each successive event. #include "cscore.h" void cscore(CSOUND *cs) { EVENT *e, *f; EVLIST *a, *b; int n; a = cscoreListGetSection(cs); /* read score into event list "a" */ b = cscoreListSeparateF(cs, a); /* separate f statements */ cscoreListPut(cs, b); /* write f statements out to score */ cscoreListFreeEvents(cs, b); /* and release the spaces used */ e = cscoreDefineEvent(cs, "t 0 120"); /* define event for tempo statement */ cscorePutEvent(cs, e); /* write tempo statement to score */ cscoreListPut(cs, a); /* write the notes */ cscorePutString(cs, "s"); /* section end */ cscorePutEvent(cs, e); /* write tempo statement again */ b = cscoreListCopyEvents(cs, a); /* make a copy of the notes in "a" */ for (n = 1; n <= b->nevents; n++) /* iterate the following lines nevents times: */ { f = b->e[n]; f->p[5] *= 0.5; /* transpose pitch down one octave */ } a = cscoreListAppendList(cs, a, b); /* now add these notes to original pitches */ cscoreListPut(cs, a); cscorePutString(cs, "e"); } The output of this program is: f 1 0 257 10 1 f 2 0 257 7 0 300 1 212 .8 t 0 120 i 1 1 3 0 440 10000 i 1 4 3 0 256 10000 i 1 7 3 0 880 10000 s t 0 120 i 1 1 3 0 440 10000 i 1 4 3 0 256 10000 i 1 7 3 0 880 10000 i 1 1 3 0 220 10000 i 1 4 3 0 128 10000 i 1 7 3 0 440 10000 e If the output is only being written to a file, then the unsorted order of the events is not a problem. The output is written to a file (or standard output) whenever the function cscoreListPut() is used. However, if this program were to be called during a Csound performance and the function cscoreListPlay() replaced cscoreListPut(), then the events would be sent to the orchestra instead of to a file and they should then be sorted beforehand by calling the function cscoreListSort(). The details of score output and playing when using Cscore from within Csound are described in the next section. Next we extend the above program by using the for loop to look at p[5] and p[6]. In the original score p[6] denotes amplitude. To create a diminuendo in the added lower octave, which is independent from the original set of notes, a variable called dim will be used. #include "cscore.h" void cscore(CSOUND *cs) { EVENT *e, *f; EVLIST *a, *b; int n, dim; /* declare two integer variables */ a = cscoreListGetSection(cs); b = cscoreListSeparateF(cs, a); cscoreListPut(cs, b); cscoreListFreeEvents(cs, b); e = cscoreDefineEvent(cs, "t 0 120"); cscorePutEvent(cs, e); cscoreListPut(cs, a); cscorePutString(cs, "s"); cscorePutEvent(cs, e); /* write out another tempo statement */ b = cscoreListCopyEvents(cs, a); dim = 0; /* initialize dim to 0 */ for (n = 1; n <= b->nevents; n++) { f = b->e[n]; f->p[6] -= dim; /* subtract current value of dim */ f->p[5] *= 0.5; /* transpose pitch down one octave */ dim += 2000; /* increase dim for each note */ } a = cscoreListAppendList(cs, a, b); /* now add these notes to original pitches */ cscoreListPut(cs, a); cscorePutString(cs, "e"); } Using the same input score again, the output from this program is: f 1 0 257 10 1 f 2 0 257 7 0 300 1 212 .8 t 0 120 i 1 1 3 0 440 10000 i 1 4 3 0 256 10000 i 1 7 3 0 880 10000 s t 0 120 i 1 1 3 0 440 10000 ; Three original notes at i 1 4 3 0 256 10000 ; beats 1,4 and 7 with no dim. i 1 7 3 0 880 10000 i 1 1 3 0 220 10000 ; three notes transposed down one octave i 1 4 3 0 128 8000 ; also at beats 1,4 and 7 with dim. i 1 7 3 0 440 6000 e In the following program the same three-note sequence will be repeated at various time intervals. The starting time of each group is determined by the values of the array cue. This time the dim will occur for each group of notes rather than each note. Note the position of the statement which increments the variable dim outside the inner for loop. #include "cscore.h" int cue[3] = {0,10,17}; /* declare an array of 3 integers */ void cscore(CSOUND *cs) { EVENT *e, *f; EVLIST *a, *b; int n, dim, cuecount; /* declare new variable cuecount */ a = cscoreListGetSection(cs); b = cscoreListSeparateF(cs, a); cscoreListPut(cs, b); cscoreListFreeEvents(cs, b); e = cscoreDefineEvent(cs, "t 0 120"); cscorePutEvent(cs, e); dim = 0; for (cuecount = 0; cuecount <= 2; cuecount++) /* elements of cue are numbered 0, 1, 2 */ { for (n = 1; n <= a->nevents; n++) { f = a->e[n]; f->p[6] -= dim; f->p[2] += cue[cuecount]; /* add values of cue */ } printf("; diagnostic: cue = %d\n", cue[cuecount]); dim += 2000; cscoreListPut(cs, a); } cscorePutString(cs, "e"); } Here the inner for loop looks at the events of list a (the notes) and the outer for loop looks at each repetition of the events of list a (the pitch group "cues"). This program also demonstrates a useful trouble-shooting device with the printf function. The semi-colon is first in the character string to produce a comment statement in the resulting score file. In this case the value of cue is being printed in the output to insure that the program is taking the proper array member at the proper time. When output data is wrong or error messages are encountered, the printf function can help to pinpoint the problem. Using the same input file, the C program above will generate the following score. Can you determine why the last set of notes starts at the wrong time and how to correct the problem? f 1 0 257 10 1 f 2 0 257 7 0 300 1 212 .8 t 0 120 ; diagnostic: cue = 0 i 1 1 3 0 440 10000 i 1 4 3 0 256 10000 i 1 7 3 0 880 10000 ; diagnostic: cue = 10 i 1 11 3 0 440 8000 i 1 14 3 0 256 8000 i 1 17 3 0 880 8000 ; diagnostic: cue = 17 i 1 28 3 0 440 4000 i 1 31 3 0 256 4000 i 1 34 3 0 880 4000 e
manual_src~dfsg/cscore/top.xml0000644000000000000000000000577112262561504015561 0ustar rootroot Cscore Cscore Cscore is an API (application programming interface) for generating and manipulating numeric score files. It is a part of the larger Csound API and includes a number of functions that can be called by a user-designed program written in the C language. Cscore can be invoked either as a standalone score preprocessor, or as part of a Csound performance by including the -C flag in its arguments: cscore scorefilein > scorefileout (where cscore is the name of your user-written program), or csound -C orchname scorename The available API functions augment the C language library functions; they can read either standard numeric scores or pre-sorted score files, can massage and expand the data in various ways, then make it available for performance by a Csound orchestra. The user-written control program is written in C, and is compiled and linked to the Csound library (or the csound commandline program) by the user. It is not essential to know the C language well to write this program, since the function calls have a simple syntax, and are powerful enough to do most of the complicated work. Additional power can come from C later as the need arises. The following sections explain all of the steps needed to make use of Cscore: Events, Lists, and Operations - Explains the syntax of Cscore functions and data structures. Writing a Cscore Control Program - Illustrates by example how to write your own control program. Compiling a Cscore Program - Outlines the steps for compiling and linking with the Csound library. More Advanced Examples - Addresses advanced issues such as multiple input scores and the details of running Cscore inside of a Csound performance. &cscoreevents; &cscoremain; &cscorecompile; &cscoreadvancd; manual_src~dfsg/cscore/advancd.xml0000644000000000000000000001140512262561504016346 0ustar rootroot
More Advanced Examples The following program demonstrates reading from two different input files. The idea is to switch between two 2-section scores, and write out the interleaved sections to a single output file. #include "cscore.h" /* CSCORE_SWITCH.C */ cscore(CSOUND* cs) /* callable from either Csound or standalone cscore */ { EVLIST *a, *b; FILE *fp1, *fp2; /* declare two scorefile stream pointers */ fp1 = cscoreFileGetCurrent(cs); /* this is the command-line score */ fp2 = cscoreFileOpen(cs, "score2.srt"); /* this is an additional score file */ a = cscoreListGetSection(cs); /* read section from score 1 */ cscoreListPut(cs, a); /* write it out as is */ cscorePutString(cs, "s"); cscoreFileSetCurrent(cs, fp2); b = cscoreListGetSection(cs); /* read section from score 2 */ cscoreListPut(cs, b); /* write it out as is */ cscorePutString(cs, "s"); cscoreListFreeEvents(cs, a); /* optional to reclaim space */ cscoreListFreeEvents(cs, b); cscoreFileSetCurrent(cs, fp1); a = cscoreListGetSection(cs); /* read next section from score 1 */ cscoreListPut(cs, a); /* write it out */ cscorePutString(cs, "s"); cscoreFileSetCurrent(cs, fp2); b = cscoreListGetSection(cs); /* read next sect from score 2 */ cscoreListPut(cs, b); /* write it out */ cscorePutString(cs, "e"); } Finally, we show how to take a literal, uninterpreted score file and imbue it with some expressive timing changes. The theory of composer-related metric pulses has been investigated at length by Manfred Clynes, and the following is in the spirit of his work. The strategy here is to first create an array of new onset times for every possible sixteenth-note onset, then to index into it so as to adjust the start and duration of each note of the input score to the interpreted time-points. This also shows how a Csound orchestra can be invoked repeatedly from a run-time score generator. #include "cscore.h" /* CSCORE_PULSE.C */ /* program to apply interpretive durational pulse to */ /* an existing score in 3/4 time, first beats on 0, 3, 6 ... */ static float four[4] = { 1.05, 0.97, 1.03, 0.95 }; /* pulse width for 4's */ static float three[3] = { 1.03, 1.05, .92 }; /* pulse width for 3's */ cscore(CSOUND* cs) /* This example should be called from Csound */ { EVLIST *a, *b; EVENT *e, **ep; float pulse16[4*4*4*4*3*4]; /* 16th-note array, 3/4 time, 256 measures */ float acc16, acc1,inc1, acc3,inc3, acc12,inc12, acc48,inc48, acc192,inc192; float *p = pulse16; int n16, n1, n3, n12, n48, n192; /* fill the array with interpreted ontimes */ for (acc192=0.,n192=0; n192<4; acc192+=192.*inc192,n192++) for (acc48=acc192,inc192=four[n192],n48=0; n48<4; acc48+=48.*inc48,n48++) for (acc12=acc48,inc48=inc192*four[n48],n12=0;n12<4; acc12+=12.*inc12,n12++) for (acc3=acc12,inc12=inc48*four[n12],n3=0; n3<4; acc3+=3.*inc3,n3++) for (acc1=acc3,inc3=inc12*four[n3],n1=0; n1<3; acc1+=inc1,n1++) for (acc16=acc1,inc1=inc3*three[n1],n16=0; n16<4; acc16+=.25*inc1*four[n16],n16++) *p++ = acc16; /* for (p = pulse16, n1 = 48; n1--; p += 4) /* show vals & diffs */ /* printf("%g %g %g %g %g %g %g %g\n", *p, *(p+1), *(p+2), *(p+3), /* *(p+1)-*p, *(p+2)-*(p+1), *(p+3)-*(p+2), *(p+4)-*(p+3)); */ a = cscoreListGetSection(cs); /* read sect from tempo-warped score */ b = cscoreListSeparateTWF(cs, a); /* separate warp & fn statements */ cscoreListPlay(cs, b); /* and send these to performance */ a = cscoreListAppendStringEvent(cs, a, "s"); /* append a sect statement to note list */ cscoreListPlay(cs, a); /* play the note-list without interpretation */ for (ep = &a->e[1], n1 = a->nevents; n1--; ) { /* now pulse-modifiy it */ e = *ep++; if (e->op == 'i') { e->p[2] = pulse16[(int)(4. * e->p2orig)]; e->p[3] = pulse16[(int)(4. * (e->p2orig + e->p3orig))] - e->p[2]; } } cscoreListPlay(cs, a); /* now play modified list */ }
manual_src~dfsg/cscore/events.xml0000644000000000000000000002540312262561504016255 0ustar rootroot
Events, Lists, and Operations An event in Cscore is equivalent to one statement of a standard numeric score or a time-warped score (the format in which Csound writes a sorted score -- see any score.srt), and is stored internally in time-warped format. It is important to note that when Cscore is used in standalone-mode, it cannot understand any of the non-numeric "conveniences" that Csound allows in the input score format. Therefore, scores making use of features such as carry, ramp, expressions, and others will have to either be sorted first with the scsort utility or used with a modified Csound executable that contains the user's Cscore program. Score opcodes with macro arguments (r, m, n, and {}) are not understood. Score events are each read in from an existing score file and stored in a C structure. The structures main components are an opcode and an array of pfield values. Cscore handles reading the events and storing them in memory for you. The format of the structure starts as follows: typedef struct { CSHDR h; /* space-managing header */ char *strarg; /* address of optional string argument */ char op; /* opcode-t, w, f, i, a, s or e */ short pcnt; MYFLT p2orig; /* unwarped p2, p3 */ MYFLT p3orig; MYFLT p[1]; /* array of pfields p0, p1, p2 ... */ } EVENT; MYFLT is either the C type float or double depending on how your copy of the Csound library was compiled. You should just declare any floating-point variables as MYFLT in your user program for compatibility. Any Cscore function that creates, reads, or copies an event will return a pointer to the storage structure holding the event data. The event pointer can be used to access any component of the structure, in the form of e->op or e->p[n]. Each newly stored event will give rise to a new pointer, and a sequence of new events will generate a sequence of distinct pointers that must themselves be stored. Groups of event pointers are stored in an event list, which has its own structure: typedef struct { CSHDR h; int nslots; /* max events in this event list */ int nevents; /* number of events present */ EVENT *e[1]; /* array of event pointers e0, e1, e2.. */ } EVLIST; Any Cscore function that creates or modifies a list will return a pointer to the new list. The list pointer can be used to access any of its component event pointers, in the form of a->e[n]. Event pointers and list pointers are thus primary tools for manipulating the data of a score file. Pointers and lists of pointers can be copied and reordered without modifying the data values they refer to. This means that notes and phrases can be copied and manipulated from a high level of control. Alternatively, the data within an event or group of events can be modified without changing the event or list pointers. The Cscore API functions enable scores to be created and manipulated in this way. With Csound 5, the names of all of the Cscore API functions have changed to be more explicit. In addition, each function now requires a pointer to a CSOUND object as its first argument. The structure of the CSOUND object is unimportant (and indeed cannot be modified in a user program). How to obtain this CSOUND pointer will be shown in the next section. The Cscore functions and data structures are available in the cscore.h header file, which you must include in your program code before you can you use them. The names of the Cscore functions specify whether they operate on single events or event lists. In the following summary of available function calls, some simple naming conventions are used: The symbol cs is a pointer to a CSOUND object (CSOUND *); The symbols e, f are pointers to events (notes); The symbols a, b are pointers to lists (arrays) of such events; The symbol n is an integer parameter of type int; "..." indicates a string parameter (either a constant or variable of type char *); The symbol fp is a score input stream file pointer (FILE *); calling syntax description -------------- ----------- /* Functions for working with single events */ e = cscoreCreateEvent(cs, n); create a blank event with n pfields e = cscoreDefineEvent(cs, "..."); defines an event as per the character string ... e = cscoreCopyEvent(cs, f); make a new copy of event f e = cscoreGetEvent(cs); read the next event in the score input file cscorePutEvent(cs, e); write event e to the score output file cscorePutString(cs, "..."); write the string-defined event to score output /* Functions for working with event lists */ a = cscoreListCreate(cs, n); create an empty event list with n slots a = cscoreListAppendEvent(cs, a, e); append event e to list a a = cscoreListAppendStringEvent(cs, a, "..."); append a string-defined event to list a; a = cscoreListCopy(cs, b); copy the list b (but not the events) a = cscoreListCopyEvents(cs, b); copy the events of b, making a new list a = cscoreListGetSection(cs); read all events from score input, up to next s or e a = cscoreListGetNext(cs, nbeats); read next nbeats beats from score input (nbeats is MYFLT) a = cscoreListGetUntil(cs, beatno); read all events from score input up to beat beatno (MYFLT) a = cscoreListSeparateF(cs, b); separate the f statements from list b into list a a = cscoreListSeparateTWF(cs, b); separate the t,w & f statements from list b into list a a = cscoreListAppendList(cs, a, b); append the list b onto the list a a = cscoreListConcatenate(cs, a, b); concatenate (append) the list b onto the list a (same as previous) cscoreListSort(cs, a); sort the list a into chronological order by p[2] n = cscoreListCount(cs, a); returns the number of events in list a a = cscoreListExtractInstruments(cs, b, "..."); extract notes of instruments ... (no new events) a = cscoreListExtractTime(cs, b, from, to); extract notes of time-span, creating new events (from and to are MYFLT) cscoreListPut(cs, a); write the events of list a to the score output file cscoreListPlay(cs, a); send events of list a to the Csound orchestra for immediate performance (or print events if no orchestra) /* Functions for reclaiming memory */ cscoreFreeEvent(cs, e); release the space of event e cscoreListFree(cs, a); release the space of list a (but not the events) cscoreListFreeEvents(cs, a); release the events of list a, and the list space /* Functions for working with multiple input score files */ fp = cscoreFileGetCurrent(cs); get the currently active input scorefile pointer (initially finds the command-line input scorefile pointer) fp = cscoreFileOpen(cs, "filename"); open another input scorefile (maximum of 5) cscoreFileSetCurrent(cs, fp); make fp the currently active scorefile pointer cscoreFileClose(cs, fp); close the scorefile relating to FILE *fp Under Csound 4, the function names and parameters were as follows: calling syntax description -------------- ----------- e = createv(n); create a blank event with n pfields e = defev("..."); defines an event as per the character string ... e = copyev(f); make a new copy of event f e = getev(); read the next event in the score input file putev(e); write event e to the score output file putstr("..."); write the string-defined event to score output a = lcreat(n); create an empty event list with n slots int n; a = lappev(a,e); append event e to list a a = lappstrev(a,"..."); append a string-defined event to list a; a = lcopy(b); copy the list b (but not the events) a = lcopyev(b); copy the events of b, making a new list a = lget(); read all events from score input, up to next s or e a = lgetnext(nbeats); read next nbeats beats from score input float nbeats; a = lgetuntil(beatno); read all events from score input up to beat beatno float beatno; a = lsepf(b); separate the f statements from list b into list a a = lseptwf(b); separate the t,w & f statements from list b into list a a = lcat(a,b); concatenate (append) the list b onto the list a lsort(a); sort the list a into chronological order by p[2] a = lxins(b,"..."); extract notes of instruments ... (no new events) a = lxtimev(b,from,to); extract notes of time-span, creating new events float from, to; lput(a); write the events of list a to the score output file lplay(a); send events of list a to the Csound orchestra for immediate performance (or print events if no orchestra) relev(e); release the space of event e lrel(a); release the space of list a (but not the events) lrelev(a); release the events of list a, and the list space fp = getcurfp(); get the currently active input scorefile pointer (initially finds the command-line input scorefile pointer) fp = filopen("filename"); open another input scorefile (maximum of 5) setcurfp(fp); make fp the currently active scorefile pointer filclose(fp); close the scorefile relating to FILE *fp
manual_src~dfsg/opcodes.xml0000644000000000000000000101602012262561504015123 0ustar rootroot Sets the value of 0 decibels using full scale amplitude.0dbfs = iarg Sets the control rate.kr = iarg Sets the number of samples in a control period.ksmps = iarg Sets the number of channels of audio output.nchnls = iarg Sets the number of channels of audio input.nchnls_i = iarg Sets the audio sampling rate.sr = iarg Ends the current instrument block.endin Marks the end of an user-defined opcode block.endop Starts an instrument block.instr i, j, ... Defines the start of user-defined opcode block.opcode name, outtypes, intypes Defines a macro.#define NAME # replacement text # #define NAME(a' b' c') # replacement text # Calls a defined macro.$NAME Conditional reading of code.#ifdef NAME ....#else ....#end Conditional reading of code.#ifndef NAME ....#else ....#end Includes an external file for processing.#include "filename" Un-defines a macro.#undef NAME Output is an additive set of individually controlled sinusoids, using an oscillator bank.ares adsyn kamod, kfmod, ksmod, ifilcod Performs additive synthesis with an arbitrary number of partials, not necessarily harmonic.ares adsynt kamp, kcps, iwfn, ifreqfn, iampfn, icnt [, iphs] Performs additive synthesis with an arbitrary number of partials -not necessarily harmonic- with interpolation.ar adsynt2 kamp, kcps, iwfn, ifreqfn, iampfn, icnt [, iphs] An oscillator which takes tonality and brightness as arguments.ares hsboscil kamp, ktone, kbrite, ibasfreq, iwfn, ioctfn \ [, ioctcnt] [, iphs] A low frequency oscillator of various shapes.kres lfo kamp, kcps [, itype]ares lfo kamp, kcps [, itype] Mixes the output of any number of oscillators.ares oscbnk kcps, kamd, kfmd, kpmd, iovrlap, iseed, kl1minf, kl1maxf, \ kl2minf, kl2maxf, ilfomode, keqminf, keqmaxf, keqminl, keqmaxl, \ keqminq, keqmaxq, ieqmode, kfn [, il1fn] [, il2fn] [, ieqffn] \ [, ieqlfn] [, ieqqfn] [, itabl] [, ioutfn] A simple oscillator.ares oscil xamp, xcps [, ifn, iphs]kres oscil kamp, kcps [, ifn, iphs] A simple oscillator with cubic interpolation.ares oscil3 xamp, xcps [, ifn, iphs]kres oscil3 kamp, kcps [, ifn, iphs] A simple oscillator with linear interpolation.ares oscili xamp, xcps, ifn [, iphs]kres oscili kamp, kcps, ifn [, iphs] A linearly interpolated oscillator that allows changing the table number at k-rate.ares oscilikt xamp, xcps, kfn [, iphs] [, istor]kres oscilikt kamp, kcps, kfn [, iphs] [, istor] A linearly interpolated oscillator that allows allows phase modulation.ares osciliktp kcps, kfn, kphs [, istor] A linearly interpolated oscillator with sync status that allows changing the table number at k-rate.ares oscilikts xamp, xcps, kfn, async, kphs [, istor] Accesses table values at a user-defined frequency.ares osciln kamp, ifrq, ifn, itimes A simple, fast sine oscillatorares oscils iamp, icps, iphs [, iflg] High precision oscillator.ares poscil aamp, acps [, ifn, iphs]ares poscil aamp, kcps [, ifn, iphs]ares poscil kamp, acps [, ifn, iphs]ares poscil kamp, kcps [, ifn, iphs]ires poscil kamp, kcps [, ifn, iphs]kres poscil kamp, kcps [, ifn, iphs] High precision oscillator with cubic interpolation.ares poscil3 aamp, acps [, ifn, iphs]ares poscil3 aamp, kcps [, ifn, iphs]ares poscil3 kamp, acps [, ifn, iphs]ares poscil3 kamp, kcps [, ifn, iphs]ires poscil3 kamp, kcps [, ifn, iphs]kres poscil3 kamp, kcps [, ifn, iphs] Easier-to-use user-controllable vibrato.kout vibr kAverageAmp, kAverageFreq, ifn Generates a natural-sounding user-controllable vibrato.kout vibrato kAverageAmp, kAverageFreq, kRandAmountAmp, kRandAmountFreq, kAmpMinRate, kAmpMaxRate, kcpsMinRate, kcpsMaxRate, ifn [, iphs Output is a set of harmonically related sine partials.ares buzz xamp, xcps, knh, ifn [, iphs] Output is a set of harmonically related cosine partials.ares gbuzz xamp, xcps, knh, klh, kmul, ifn [, iphs] Generates a set of impulses.ares mpulse kamp, kintvl [, ioffset] Implementation of a band limited, analog modeled oscillator.ares vco xamp, xcps, iwave, kpw [, ifn] [, imaxd] [, ileak] [, inyx] \ [, iphs] [, iskip] Implementation of a band-limited oscillator using pre-calculated tables.ares vco2 kamp, kcps [, imode] [, kpw] [, kphs] [, inyx] Returns a table number at k-time for a given oscillator frequency and wavform.kfn vco2ft kcps, iwave [, inyx] Returns a table number at i-time for a given oscillator frequency and wavform.ifn vco2ift icps, iwave [, inyx] Calculates tables for use by vco2 opcode.ifn vco2init iwave [, ibasfn] [, ipmul] [, iminsiz] [, imaxsiz] [, isrcft] Two mutually frequency and/or phase modulated oscillators.a1, a2 crossfm xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2]a1, a2 crossfmi xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2]a1, a2 crosspm xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2]a1, a2 crosspmi xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2]a1, a2 crossfmpm xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2]a1, a2 crossfmpmi xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] Uses FM synthesis to create a Hammond B3 organ sound.ares fmb3 kamp, kfreq, kc1, kc2, kvdepth, kvrate[, ifn1, ifn2, ifn3, \ ifn4, ivfn] Uses FM synthesis to create a tublar bell sound.ares fmbell kamp, kfreq, kc1, kc2, kvdepth, kvrate[, ifn1, ifn2, ifn3, \ ifn4, ivfn, isus] Uses FM synthesis to create aares fmmetal kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, \ ifn4, ivfn Uses FM synthesis to create a percussive flute sound.ares fmpercfl kamp, kfreq, kc1, kc2, kvdepth, kvrate[, ifn1, ifn2, \ ifn3, ifn4, ivfn] Uses FM synthesis to create a Fender Rhodes electric piano sound.ares fmrhode kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, \ ifn3, ifn4, ivfn FM Singing Voice Synthesisares fmvoice kamp, kfreq, kvowel, ktilt, kvibamt, kvibrate[, ifn1, \ ifn2, ifn3, ifn4, ivibfn] Uses FM synthesis to create a Wurlitzer electric piano sound.ares fmwurlie kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, \ ifn4, ivfn A basic frequency modulated oscillator.ares foscil xamp, kcps, xcar, xmod, kndx, ifn [, iphs] Basic frequency modulated oscillator with linear interpolation.ares foscili xamp, kcps, xcar, xmod, kndx, ifn [, iphs] Synchronous granular synthesis, using a soundfile as source.asig diskgrain Sfname, kamp, kfreq, kpitch, kgrsize, kprate, \ ifun, iolaps [,imaxgrsize , ioffset] Produces sinusoid bursts useful for formant and granular synthesis.ares fof xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, \ ifna, ifnb, itotdur [, iphs] [, ifmode] [, iskip] Produces sinusoid bursts including k-rate incremental indexing with each successive burst.ares fof2 xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, \ ifna, ifnb, itotdur, kphs, kgliss [, iskip] Audio output is a succession of grains derived from data in a stored function tableares fog xamp, xdens, xtrans, aspd, koct, kband, kris, kdur, kdec, \ iolaps, ifna, ifnb, itotdur [, iphs] [, itmode] [, iskip] Generates granular synthesis textures.ares grain xamp, xpitch, xdens, kampoff, kpitchoff, kgdur, igfn, \ iwfn, imgdur [, igrnd] Easy-to-use granular synthesis texture generator.ares grain2 kcps, kfmd, kgdur, iovrlp, kfn, iwfn [, irpow] \ [, iseed] [, imode] Generate granular synthesis textures with more user control.ares grain3 kcps, kphs, kfmd, kpmd, kgdur, kdens, imaxovr, kfn, iwfn, \ kfrpow, kprpow [, iseed] [, imode] A more complex granular synthesis texture generator.ares granule xamp, ivoice, iratio, imode, ithd, ifn, ipshift, igskip, \ igskip_os, ilength, kgap, igap_os, kgsize, igsize_os, iatt, idec \ [, iseed] [, ipitch1] [, ipitch2] [, ipitch3] [, ipitch4] [, ifnenv] Granular synthesizer with "per grain" control over many of its parameters. Has a sync input to sychronize its internal grain scheduler clock to an external clock source.a1 [, a2, a3, a4, a5, a6, a7, a8] partikkel agrainfreq, \ kdistribution, idisttab, async, kenv2amt, ienv2tab, ienv_attack, \ ienv_decay, ksustain_amount, ka_d_ratio, kduration, kamp, igainmasks, \ kwavfreq, ksweepshape, iwavfreqstarttab, iwavfreqendtab, awavfm, \ ifmamptab, kfmenv, icosine, ktraincps, knumpartials, kchroma, \ ichannelmasks, krandommask, kwaveform1, kwaveform2, kwaveform3, \ kwaveform4, iwaveamptab, asamplepos1, asamplepos2, asamplepos3, \ asamplepos4, kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains \ [, iopcode_id] Outputsasync [,aphase] partikkelsync iopcode_id Reads a mono sound sample from a table and applies time-stretching and/or pitch modification.ares [, ac] sndwarp xamp, xtimewarp, xresample, ifn1, ibeg, iwsize, \ irandw, ioverlap, ifn2, itimemode Reads a stereo sound sample from a table and applies time-stretching and/or pitch modification.ar1, ar2 [,ac1] [, ac2] sndwarpst xamp, xtimewarp, xresample, ifn1, \ ibeg, iwsize, irandw, ioverlap, ifn2, itimemode Synchronous granular synthesis.asig syncgrain kamp, kfreq, kpitch, kgrsize, kprate, ifun1, \ ifun2, iolaps Synchronous granular synthesis.asig syncloop kamp, kfreq, kpitch, kgrsize, kprate, klstart, \ klend, ifun1, ifun2, iolaps[,istart, iskip] Simple vocal simulation based on glottal pulses with formant characteristics.ar vosim kamp, kFund, kForm, kDecay, kPulseCount, kPulseFactor, ifn [, iskip] Allows one-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables.hvs1 kx, inumParms, inumPointsX, iOutTab, iPositionsTab, iSnapTab [, iConfigTab] Allows two-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables.hvs2 kx, ky, inumParms, inumPointsX, inumPointsY, iOutTab, iPositionsTab, iSnapTab [, iConfigTab] Allows three-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables.hvs3 kx, ky, kz, inumParms, inumPointsX, inumPointsY, inumPointsZ, iOutTab, iPositionsTab, iSnapTab [, iConfigTab] Trace a series of line segments between specified points with cosine interpolation.ares cosseg ia, idur1, ib [, idur2] [, ic] [...]kres cosseg ia, idur1, ib [, idur2] [, ic] [...] Trace a series of line segments between specified absolute points with cosine interpolation.ares cossegb ia, itim1, ib [, itim2] [, ic] [...]kres cossegb ia, itim1, ib [, itim2] [, ic] [...] Trace a series of line segments between specified points with cosine interpolation, including a release segment.ares cossegr ia, idur1, ib [, idur2] [, ic] [...], irel, izkres cossegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz This opcode implements a formula for generating a normalised exponential curve in range 0 - 1. It is based on the Max / MSP work of Eric Singer (c) 1994.kout expcurve kindex, ksteepness Trace an exponential curve between specified points.ares expon ia, idur, ibkres expon ia, idur, ib Trace a series of exponential segments between specified points.ares expseg ia, idur1, ib [, idur2] [, ic] [...]kres expseg ia, idur1, ib [, idur2] [, ic] [...] An exponential segment generator operating at a-rate.ares expsega ia, idur1, ib [, idur2] [, ic] [...] Trace a series of exponential segments between specified absolute points.ares expsegb ia, itim1, ib [, itim2] [, ic] [...]kres expsegb ia, itim1, ib [, itim2] [, ic] [...] An exponential segment generator operating at a-rate with absolute times.ares expsegba ia, itim1, ib [, itim2] [, ic] [...] Trace a series of exponential segments between specified points including a release segment.ares expsegr ia, idur1, ib [, idur2] [, ic] [...], irel, izkres expsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz An implementation of a logarithmic gain curve which is similar to the gainslider~ object from Cycling 74 Max / MSP.kout gainslider kindex A jitter-spline generator.ares jspline xamp, kcpsMin, kcpsMaxkres jspline kamp, kcpsMin, kcpsMax Trace a straight line between specified points.ares line ia, idur, ibkres line ia, idur, ib Trace a series of line segments between specified points.ares linseg ia, idur1, ib [, idur2] [, ic] [...]kres linseg ia, idur1, ib [, idur2] [, ic] [...] Trace a series of line segments between specified absolute points.ares linsegb ia, itim1, ib [, itim2] [, ic] [...]kres linsegb ia, itim1, ib [, itim2] [, ic] [...] Trace a series of line segments between specified points including a release segment.ares linsegr ia, idur1, ib [, idur2] [, ic] [...], irel, izkres linsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz This opcode implements a formula for generating a normalised logarithmic curve in range 0 - 1. It is based on the Max / MSP work of Eric Singer (c) 1994.kout logcurve kindex, ksteepness Generate control signal consisting of linear segments delimited by two or more specified points.ksig loopseg kfreq, ktrig, iphase, kvalue0, ktime0 [, kvalue1] [, ktime1] \ [, kvalue2] [, ktime2][...] Control signals based on linear segments.ksig loopsegp kphase, kvalue0, kdur0, kvalue1 \ [, kdur1, ... , kdurN-1, kvalueN] Generate control signal consisting of exponential or linear segments delimited by two or more specified points.ksig looptseg kfreq, ktrig, ktime0, kvalue0, ktype0, [, ktime1] [, kvalue1] [,ktype1] \ [, ktime2] [, kvalue2] [,ktype2] [...][, ktimeN] [, kvalueN] Generate control signal consisting of exponential segments delimited by two or more specified points.ksig loopxseg kfreq, ktrig, iphase, ktime0, kvalue0 [, ktime1] [, kvalue1] \ [, ktime2] [, kvalue2] [...] Generate control signal consisting of held segments.ksig lpshold kfreq, ktrig, iphase, ktime0, kvalue0 [, ktime1] [, kvalue1] [, ktime2] [, kvalue2] [...] Control signals based on held segments.ksig lpsholdp kphase, ktrig, ktime0, kvalue0 [, ktime1] [, kvalue1] \ [, ktime2] [, kvalue2] [...] Generate random spline curves.ares rspline xrangeMin, xrangeMax, kcpsMin, kcpsMaxkres rspline krangeMin, krangeMax, kcpsMin, kcpsMax Arbitrary signal scaling.kscl scale kinput, kmax, kmin Constructs a user-definable envelope.ares transeg ia, idur, itype, ib [, idur2] [, itype] [, ic] ...kres transeg ia, idur, itype, ib [, idur2] [, itype] [, ic] ... Constructs a user-definable envelope in absolute time.ares transegb ia, itim, itype, ib [, itim2] [, itype] [, ic] ...kres transegb ia, itim, itype, ib [, itim2] [, itype] [, ic] ... Constructs a user-definable envelope with extended release segment.ares transegr ia, idur, itype, ib [, idur2] [, itype] [, ic] ...kres transegr ia, idur, itype, ib [, idur2] [, itype] [, ic] ... Calculates the classical ADSR envelope using linear segments.ares adsr iatt, idec, islev, irel [, idel]kres adsr iatt, idec, islev, irel [, idel] Applies an envelope consisting of 3 segments.ares envlpx xamp, irise, idur, idec, ifn, iatss, iatdec [, ixmod]kres envlpx kamp, irise, idur, idec, ifn, iatss, iatdec [, ixmod] Theares envlpxr xamp, irise, idec, ifn, iatss, iatdec [, ixmod] [,irind]kres envlpxr kamp, irise, idec, ifn, iatss, iatdec [, ixmod] [,irind] Applies a straight line rise and decay pattern to an input amp signal.ares linen xamp, irise, idur, ideckres linen kamp, irise, idur, idec Theares linenr xamp, irise, idec, iatdeckres linenr kamp, irise, idec, iatdec Calculates the classical ADSR envelope using theares madsr iatt, idec, islev, irel [, idel] [, ireltim]kres madsr iatt, idec, islev, irel [, idel] [, ireltim] Calculates the classical ADSR envelope using theares mxadsr iatt, idec, islev, irel [, idel] [, ireltim]kres mxadsr iatt, idec, islev, irel [, idel] [, ireltim] Calculates the classical ADSR envelope.ares xadsr iatt, idec, islev, irel [, idel]kres xadsr iatt, idec, islev, irel [, idel] Semi-physical model of a bamboo sound.ares bamboo kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ [, ifreq1] [, ifreq2] Creates a tone similar to a struck metal bar.ares barmodel kbcL, kbcR, iK, ib, kscan, iT30, ipos, ivel, iwid Semi-physical model of a cabasa sound.ares cabasa iamp, idettack [, inum] [, idamp] [, imaxshake] Simulates Chua's oscillator, an LRC oscillator with an active resistor, proved capable of bifurcation and chaotic attractors, with k-rate control of circuit elements.aI3, aV2, aV1 chuap kL, kR0, kC1, kG, kGa, kGb, kE, kC2, iI3, iV2, iV1, ktime_step Semi-physical model of a crunch sound.ares crunch iamp, idettack [, inum] [, idamp] [, imaxshake] Semi-physical model of a water drop.ares dripwater kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ [, ifreq1] [, ifreq2] Dynamic stochastic approach to waveform synthesis conceived by Iannis Xenakis.ares gendy kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl [, initcps] [, knum]kres gendy kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl [, initcps] [, knum] Dynamic stochastic approach to waveform synthesis using cubic interpolation.ares gendyc kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl [, initcps] [, knum]kres gendyc kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl [, initcps] [, knum] Variation of the dynamic stochastic approach to waveform synthesis conceived by Iannis Xenakis.ares gendyx kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl, kcurveup, kcurvedown [, initcps] [, knum]kres gendyx kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl, kcurveup, kcurvedown [, initcps] [, knum] Audio output is a tone related to the striking of a cow bell or similar.ares gogobel kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivfn Semi-physical model of a guiro sound.ares guiro kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] [, ifreq1] Implements the Lorenz system of equations.ax, ay, az lorenz ksv, krv, kbv, kh, ix, iy, iz, iskip [, iskipinit] Mandelbrot setkiter, koutrig mandel ktrig, kx, ky, kmaxIter An emulation of a mandolin.ares mandol kamp, kfreq, kpluck, kdetune, kgain, ksize, ifn [, iminfreq] Physical model related to the striking of a wooden block.ares marimba kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec \ [, idoubles] [, itriples] An emulation of a mini-Moog synthesizer.ares moog kamp, kfreq, kfiltq, kfiltrate, kvibf, kvamp, iafn, iwfn, ivfn Simulates a planet orbiting in a binary star system.ax, ay, az planet kmass1, kmass2, ksep, ix, iy, iz, ivx, ivy, ivz, idelta \ [, ifriction] [, iskip] Creates a tone similar to a piano string prepared in a Cageian fashion.ares prepiano ifreq, iNS, iD, iK, \ iT30,iB, kbcl, kbcr, imass, ifreq, iinit, ipos, ivel, isfreq, \ isspread[, irattles, irubbers]al,ar prepiano ifreq, iNS, iD, iK, \ iT30,iB, kbcl, kbcr, imass, ifreq, iinit, ipos, ivel, isfreq, \ isspread[, irattles, irubbers] Semi-physical model of a sandpaper sound.ares sandpaper iamp, idettack [, inum] [, idamp] [, imaxshake] Semi-physical model of a sekere sound.ares sekere iamp, idettack [, inum] [, idamp] [, imaxshake] Sounds like the shaking of a maraca or similar gourd instrument.ares shaker kamp, kfreq, kbeans, kdamp, ktimes [, idecay] Semi-physical model of a sleighbell sound.ares sleighbells kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ [, ifreq1] [, ifreq2] Semi-physical model of a stick sound.ares stix iamp, idettack [, inum] [, idamp] [, imaxshake] Semi-physical model of a tambourine sound.ares tambourine kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ [, ifreq1] [, ifreq2] Physical model related to the striking of a metal block.ares vibes kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec An emulation of a human voice.ares voice kamp, kfreq, kphoneme, kform, kvibf, kvamp, ifn, ivfn Produce a normalized moving phase value.ares phasor xcps [, iphs]kres phasor kcps [, iphs] Produce an arbitrary number of normalized moving phase values.ares phasorbnk xcps, kndx, icnt [, iphs]kres phasorbnk kcps, kndx, icnt [, iphs] Produces a normalized moving phase value with sync input and output.aphase, asyncout syncphasor xcps, asyncin, [, iphs] Beta distribution random number generator (positive values only).ares betarand krange, kalpha, kbetaires betarand krange, kalpha, kbetakres betarand krange, kalpha, kbeta Exponential distribution random number generator.ares bexprnd krangeires bexprnd krangekres bexprnd krange Cauchy distribution random number generator.ares cauchy kalphaires cauchy kalphakres cauchy kalpha Cauchy distribution random number generator with interpolation.ares cauchyi klambda, xamp, xcpsires cauchyi klambda, xamp, xcpskres cauchyi klambda, xamp, xcps Continuous USER-defined-distribution RaNDom generator.aout cuserrnd kmin, kmax, ktableNumiout cuserrnd imin, imax, itableNumkout cuserrnd kmin, kmax, ktableNum Discrete USER-defined-distribution RaNDom generator.aout duserrnd ktableNumiout duserrnd itableNumkout duserrnd ktableNum Random impulses.ares dust kamp, kdensitykres dust kamp, kdensity Random impulses.ares dust2 kamp, kdensitykres dust2 kamp, kdensity Exponential distribution random number generator (positive values only).ares exprand klambdaires exprand klambdakres exprand klambda Exponential distribution random number generator with interpolation (positive values only).ares exprandi klambda, xamp, xcpsires exprandi klambda, xamp, xcpskres exprandi klambda, xamp, xcps A fractal noise generator.ares fractalnoise kamp, kbeta Gaussian distribution random number generator.ares gauss krangeires gauss krangekres gauss krange Gaussian distribution random number generator with interpolation.ares gaussi krange, xamp, xcpsires gaussi krange, xamp, xcpskres gaussi krange, xamp, xcps Random impulses around a certain frequency.ares gausstrig kamp, kcps, kdev [, imode]kres gausstrig kamp, kcps, kdev [, imode] Generates a segmented line whose segments are randomly generated.kout jitter kamp, kcpsMin, kcpsMax Generates a segmented line with user-controllable random segments.kout jitter2 ktotamp, kamp1, kcps1, kamp2, kcps2, kamp3, kcps3 Linear distribution random number generator (positive values only).ares linrand krangeires linrand krangekres linrand krange A white noise generator with an IIR lowpass filter.ares noise xamp, kbeta Cauchy distribution random number generator (positive values only).ares pcauchy kalphaires pcauchy kalphakres pcauchy kalpha Generates approximate pink noise.ares pinkish xin [, imethod] [, inumbands] [, iseed] [, iskip] Poisson distribution random number generator (positive values only).ares poisson klambdaires poisson klambdakres poisson klambda Generates a controlled random number series.ares rand xamp [, iseed] [, isel] [, ioffset]kres rand xamp [, iseed] [, isel] [, ioffset] Generates random numbers and holds them for a period of time.ares randh xamp, xcps [, iseed] [, isize] [, ioffset]kres randh kamp, kcps [, iseed] [, isize] [, ioffset] Generates a controlled random number series with interpolation between each new number.ares randi xamp, xcps [, iseed] [, isize] [, ioffset]kres randi kamp, kcps [, iseed] [, isize] [, ioffset] Generates a controlled pseudo-random number series between min and max values.ares random kmin, kmaxires random imin, imaxkres random kmin, kmax Generates random numbers with a user-defined limit and holds them for a period of time.ares randomh kmin, kmax, xcps [,imode] [,ifirstval]kres randomh kmin, kmax, kcps [,imode] [,ifirstval] Generates a user-controlled random number series with interpolation between each new number.ares randomi kmin, kmax, xcps [,imode] [,ifirstval]kres randomi kmin, kmax, kcps [,imode] [,ifirstval] 31-bit bipolar random opcodes with controllable distribution.ax rnd31 kscl, krpow [, iseed]ix rnd31 iscl, irpow [, iseed]kx rnd31 kscl, krpow [, iseed] Sets the global seed value.seed ival Generates a controlled pseudo-random number series between min and max values according to a trigger.kout trandom ktrig, kmin, kmax Triangular distribution random number generatorares trirand krangeires trirand krangekres trirand krange Uniform distribution random number generator (positive values only).ares unirand krangeires unirand krangekres unirand krange truly random opcodes with controllable range.ax urandom [imin, imax]ix urandom [imin, imax]kx urandom [imin, imax] A discrete user-defined-distribution random generator that can be used as a function.aout = urd(ktableNum)iout = urd(itableNum)kout = urd(ktableNum) Weibull distribution random number generator (positive values only).ares weibull ksigma, ktauires weibull ksigma, ktaukres weibull ksigma, ktau Generates breakbeat-style cut-ups of a mono audio stream.a1 bbcutm asource, ibps, isubdiv, ibarlength, iphrasebars, inumrepeats \ [, istutterspeed] [, istutterchance] [, ienvchoice ] Generates breakbeat-style cut-ups of a stereo audio stream.a1,a2 bbcuts asource1, asource2, ibps, isubdiv, ibarlength, iphrasebars, \ inumrepeats [, istutterspeed] [, istutterchance] [, ienvchoice] Function-table-based crossfading looper.asig flooper kamp, kpitch, istart, idur, ifad, ifn Function-table-based crossfading looper.asig flooper2 kamp, kpitch, kloopstart, kloopend, kcrossfade, ifn \ [, istart, imode, ifenv, iskip] Collects all audio from all Fluidsynth engines in a performancealeft, aright fluidAllOut Sends a MIDI controller data message to fluid.fluidCCi iEngineNumber, iChannelNumber, iControllerNumber, iValue Sends a MIDI controller data message to fluid.fluidCCk iEngineNumber, iChannelNumber, iControllerNumber, kValue Sends MIDI note on, note off, and other messages to a SoundFont preset.fluidControl ienginenum, kstatus, kchannel, kdata1, kdata2 Instantiates a fluidsynth engine.ienginenum fluidEngine [iReverbEnabled] [, iChorusEnabled] [,iNumChannels] [, iPolyphony] Loads a SoundFont into a fluidEngine, optionally listing SoundFont contents.isfnum fluidLoad soundfont, ienginenum[, ilistpresets] Plays a note on a channel in a fluidSynth engine.fluidNote ienginenum, ichannelnum, imidikey, imidivel Outputs sound from a given fluidEnginealeft, aright fluidOut ienginenum Assigns a preset from a SoundFont to a channel on a fluidEngine.fluidProgramSelect ienginenum, ichannelnum, isfnum, ibanknum, ipresetnum Set interpolation method for channel in Fluid EnginefluidSetInterpMethod ienginenum, ichannelnum, iInterpMethod Read sampled sound from a table.ar1 [,ar2] loscil xamp, kcps, ifn [, ibas] [, imod1] [, ibeg1] [, iend1] \ [, imod2] [, ibeg2] [, iend2] Read sampled sound from a table using cubic interpolation.ar1 [,ar2] loscil3 xamp, kcps, ifn [, ibas] [, imod1] [, ibeg1] [, iend1] \ [, imod2] [, ibeg2] [, iend2] Loop oscillator.ar1 [, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, ar13, ar14, \ ar15, ar16] loscilx xamp, kcps, ifn \ [, iwsize, ibas, istrt, imod1, ibeg1, iend1] Generates a table index for sample playbackares lphasor xtrns [, ilps] [, ilpe] [, imode] [, istrt] [, istor] Read sampled sound from a table with looping and high precision.ares lposcil kamp, kfreqratio, kloop, kend, ifn [, iphs] Read sampled sound from a table with high precision and cubic interpolation.ares lposcil3 kamp, kfreqratio, kloop, kend, ifn [, iphs] Read sampled sound from a table with looping and high precision.ar lposcila aamp, kfreqratio, kloop, kend, ift [,iphs] Read stereo sampled sound from a table with looping and high precision.ar1, ar2 lposcilsa aamp, kfreqratio, kloop, kend, ift [,iphs] Read stereo sampled sound from a table with looping and high precision.ar1, ar2 lposcilsa2 aamp, kfreqratio, kloop, kend, ift [,iphs] Prints a list of all instruments of a previously loaded SoundFont2 (SF2) file.sfilist ifilhandle Plays a SoundFont2 (SF2) sample instrument, generating a stereo sound.ar1, ar2 sfinstr ivel, inotenum, xamp, xfreq, instrnum, ifilhandle \ [, iflag] [, ioffset] Plays a SoundFont2 (SF2) sample instrument, generating a stereo sound with cubic interpolation.ar1, ar2 sfinstr3 ivel, inotenum, xamp, xfreq, instrnum, ifilhandle \ [, iflag] [, ioffset] Plays a SoundFont2 (SF2) sample instrument, generating a mono sound with cubic interpolation.ares sfinstr3m ivel, inotenum, xamp, xfreq, instrnum, ifilhandle \ [, iflag] [, ioffset] Plays a SoundFont2 (SF2) sample instrument, generating a mono sound.ares sfinstrm ivel, inotenum, xamp, xfreq, instrnum, ifilhandle \ [, iflag] [, ioffset] Loads an entire SoundFont2 (SF2) sample file into memory.ir sfload "filename" Plays a SoundFont2 (SF2) sample preset, generating a stereo sound, with user-defined time-varying crossfade looping.ar1, ar2 sflooper ivel, inotenum, kamp, kpitch, ipreindex, kloopstart, kloopend, kcrossfade \ [, istart, imode, ifenv, iskip] Assigns all presets of a SoundFont2 (SF2) sample file to a sequence of progressive index numbers.sfpassign istartindex, ifilhandle[, imsgs] Plays a SoundFont2 (SF2) sample preset, generating a stereo sound.ar1, ar2 sfplay ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset] [, ienv] Plays a SoundFont2 (SF2) sample preset, generating a stereo sound with cubic interpolation.ar1, ar2 sfplay3 ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset] [, ienv] Plays a SoundFont2 (SF2) sample preset, generating a mono sound with cubic interpolation.ares sfplay3m ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset] [, ienv] Plays a SoundFont2 (SF2) sample preset, generating a mono sound.ares sfplaym ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset] [, ienv] Prints a list of all presets of a SoundFont2 (SF2) sample file.sfplist ifilhandle Assigns an existing preset of a SoundFont2 (SF2) sample file to an index number.ir sfpreset iprog, ibank, ifilhandle, ipreindex A sound looper with pitch control.asig, krec sndloop ain, kpitch, ktrig, idur, ifad A simple time stretch by repeating cycles.ares waveset ain, krep [, ilen] Copies from one table to another with a gain control.scanhammer isrc, idst, ipos, imult Generate audio output using scanned synthesis.ares scans kamp, kfreq, ifn, id [, iorder] A simpler scanned synthesis implementation.aout scantable kamp, kpch, ipos, imass, istiff, idamp, ivel Compute the waveform and the wavetable for use in scanned synthesis.scanu init, irate, ifnvel, ifnmass, ifnstif, ifncentr, ifndamp, kmass, \ kstif, kcentr, kdamp, ileft, iright, kpos, kstrngth, ain, idisp, id Allows the position and velocity of a node in a scanned process to be read.kpos, kvel xscanmap iscan, kamp, kvamp [, iwhich] Fast scanned synthesis waveform and the wavetable generator.ares xscans kamp, kfreq, ifntraj, id [, iorder] Allows the position and velocity of a node in a scanned process to be read.xscansmap kpos, kvel, iscan, kamp, kvamp [, iwhich] Compute the waveform and the wavetable for use in scanned synthesis.xscanu init, irate, ifnvel, ifnmass, ifnstif, ifncentr, ifndamp, kmass, \ kstif, kcentr, kdamp, ileft, iright, kpos, kstrngth, ain, idisp, id STKBandedWG uses banded waveguide techniques to model a variety of sounds.asignal STKBandedWG ifrequency, iamplitude, [kpress, kv1[, kmot, kv2[, klfo, kv3[, klfodepth, kv4[, kvel, kv5[, kstrk, kv6[, kinstr, kv7]]]]]]] STK Hammond-oid organ-like FM synthesis instrument.asignal STKBeeThree ifrequency, iamplitude, [kop4, kv1[, kop3, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] STKBlowBotl uses a helmholtz resonator (biquad filter) with a polynomial jet excitation.asignal STKBlowBotl ifrequency, iamplitude, [knoise, kv1[, klfo, kv2[, klfodepth, kv3[, kvol, kv4]]]] STK clarinet physical model with one register hole and one tonehole.asignal STKBlowHole ifrequency, iamplitude, [kreed, kv1[, knoise, kv2[, khole, kv3[, kreg, kv4[, kbreath, kv5]]]]] STKBowed is a bowed string instrument.asignal STKBowed ifrequency, iamplitude, [kpress, kv1[, kpos, kv2[, klfo, kv3[, klfodepth, kv4[, kvol, kv5]]]]] STKBrass is a simple brass instrument.asignal STKBrass ifrequency, iamplitude, [klip, kv1[, kslide, kv2[, klfo, kv3[, klfodepth, kv4[, kvol, kv5]]]]] STKClarinet uses a simple clarinet physical model.asignal STKClarinet ifrequency, iamplitude, [kstiff, kv1[, knoise, kv2[, klfo, kv3[, klfodepth, kv4[, kbreath, kv5]]]]] STKDrummer is a drum sampling synthesizer.asignal STKDrummer ifrequency, iamplitude STKFMVoices is a singing FM synthesis instrument.asignal STKFMVoices ifrequency, iamplitude, [kvowel, kv1[, kspec, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] STKFlute uses a simple flute physical model.asignal STKFlute ifrequency, iamplitude, [kjet, kv1[, knoise, kv2[, klfo, kv3[, klfodepth, kv4[, kbreath, kv5]]]]] STKHevyMetl produces metal sounds.asignal STKHevyMetl ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] STKMandolin produces mamdolin-like sounds.asignal STKMandolin ifrequency, iamplitude, [kbody, kv1[, kpos, kv2[, ksus, kv3[, kdetune, kv4[, kmic, kv5]]]]] STKModalBar is a resonant bar instrument.asignal STKModalBar ifrequency, iamplitude, [khard, kv1[, kpos, kv2[, klfo, kv3[, klfodepth, kv4[, kmix, kv5[, kvol, kv6[, kinstr, kv7]]]]]]] STKMoog produces moog-like swept filter sounds.asignal STKMoog ifrequency, iamplitude, [kq, kv1[, krate, kv2[, klfo, kv3[, klfodepth, kv4[, kvol, kv5]]]]] STKPercFlut is a percussive flute FM synthesis instrument.asignal STKPercFlut ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] STKPlucked uses a plucked string physical model.asignal STKPlucked ifrequency, iamplitude STKResonate is a noise driven formant filter.asignal STKResonate ifrequency, iamplitude, [kfreq, kv1[, kpole, kv2[, knotch, kv3[, kzero, kv4[, kenv, kv5]]]]] STK Fender Rhodes-like electric piano FM synthesis instrument.asignal STKRhodey ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] STKSaxofony is a faux conical bore reed instrument.asignal STKSaxofony ifrequency, iamplitude, [kstiff, kv1[, kapert, kv2[, kblow, kv3[, knoise, kv4[, klfo, kv5[, klfodepth, kv6[, kbreath, kv7]]]]]]] STKShakers is an instrument that simulates environmental sounds or collisions of multiple independent sound producing objects.asignal STKShakers ifrequency, iamplitude, [kenerg, kv1[, kdecay, kv2[, kshake, kv3[, knum, kv4[, kres, kv5[, kinstr, kv6]]]]]] STKSimple is a wavetable/noise instrument.asignal STKSimple ifrequency, iamplitude, [kpos, kv1[, kcross, kv2[, kenv, kv3[, kgain, kv4]]]] STKSitar uses a plucked string physical model.asignal STKSitar ifrequency, iamplitude STKStifKarp is a plucked stiff string instrument.asignal STKStifKarp ifrequency, iamplitude, [kpos, kv1[, ksus, kv2[, kstretch, kv3]]] STKTubeBell is a tubular bell (orchestral chime) FM synthesis instrument.asignal STKTubeBell ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] STKVoicForm is a four formant synthesis instrument.asignal STKVoicForm ifrequency, iamplitude, [kmix, kv1[, ksel, kv2[, klfo, kv3[, klfodepth, kv4[, kloud, kv5]]]]] STKWhistle produces whistle sounds.asignal STKWhistle ifrequency, iamplitude, [kmod, kv1[, knoise, kv2[, kfipfreq, kv3[, kfipgain, kv4[, kvol, kv5]]]]] STKWurley simulates a Wurlitzer electric piano FM synthesis instrument.asignal STKWurley ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] Accesses table values by incremental sampling.kres oscil1 idel, kamp, idur [, ifn] Accesses table values by incremental sampling with linear interpolation.kres oscil1i idel, kamp, idur [, ifn] Accesses table values by direct indexing.ares ptable andx, ifn [, ixmode] [, ixoff] [, iwrap]ires ptable indx, ifn [, ixmode] [, ixoff] [, iwrap]kres ptable kndx, ifn [, ixmode] [, ixoff] [, iwrap] Accesses table values by direct indexing with cubic interpolation.ares ptable3 andx, ifn [, ixmode] [, ixoff] [, iwrap]ires ptable3 indx, ifn [, ixmode] [, ixoff] [, iwrap]kres ptable3 kndx, ifn [, ixmode] [, ixoff] [, iwrap] Accesses table values by direct indexing with linear interpolation.ares ptablei andx, ifn [, ixmode] [, ixoff] [, iwrap]ires ptablei indx, ifn [, ixmode] [, ixoff] [, iwrap]kres ptablei kndx, ifn [, ixmode] [, ixoff] [, iwrap] Fast table opcodes.ir tab_i indx, ifn[, ixmode]kr tab kndx, ifn[, ixmode]ar tab xndx, ifn[, ixmode]tabw_i isig, indx, ifn [,ixmode]tabw ksig, kndx, ifn [,ixmode]tabw asig, andx, ifn [,ixmode] Accesses table values by direct indexing.ares table andx, ifn [, ixmode] [, ixoff] [, iwrap]ires table indx, ifn [, ixmode] [, ixoff] [, iwrap]kres table kndx, ifn [, ixmode] [, ixoff] [, iwrap] Accesses table values by direct indexing with cubic interpolation.ares table3 andx, ifn [, ixmode] [, ixoff] [, iwrap]ires table3 indx, ifn [, ixmode] [, ixoff] [, iwrap]kres table3 kndx, ifn [, ixmode] [, ixoff] [, iwrap] Accesses table values by direct indexing with linear interpolation.ares tablei andx, ifn [, ixmode] [, ixoff] [, iwrap]ires tablei indx, ifn [, ixmode] [, ixoff] [, iwrap]kres tablei kndx, ifn [, ixmode] [, ixoff] [, iwrap] A simple wave-terrain synthesis opcode.aout wterrain kamp, kpch, k_xcenter, k_ycenter, k_xradius, k_yradius, \ itabx, itaby Produces a naturally decaying plucked string or drum sound.ares pluck kamp, kcps, icps, ifn, imeth [, iparm1] [, iparm2] Physical model of the plucked string.ares repluck iplk, kamp, icps, kpick, krefl, axcite A string resonator with variable fundamental frequency.ares streson asig, kfr, ifdbgain Creates a tone similar to a bowed string.ares wgbow kamp, kfreq, kpres, krat, kvibf, kvamp, ifn [, iminfreq] A physical model of a bowed bar.ares wgbowedbar kamp, kfreq, kpos, kbowpres, kgain [, iconst] [, itvel] \ [, ibowpos] [, ilow] Creates a tone related to a brass instrument.ares wgbrass kamp, kfreq, ktens, iatt, kvibf, kvamp, ifn [, iminfreq] Creates a tone similar to a clarinet.ares wgclar kamp, kfreq, kstiff, iatt, idetk, kngain, kvibf, kvamp, ifn \ [, iminfreq] Creates a tone similar to a flute.ares wgflute kamp, kfreq, kjet, iatt, idetk, kngain, kvibf, kvamp, ifn \ [, iminfreq] [, ijetrf] [, iendrf] A high fidelity simulation of a plucked string.ares wgpluck icps, iamp, kpick, iplk, idamp, ifilt, axcite Physical model of the plucked string.ares wgpluck2 iplk, kamp, icps, kpick, krefl Periodically writes an orchestra control-signal value to an external file.dumpk ksig, ifilname, iformat, iprd Periodically writes two orchestra control-signal values to an external file.dumpk2 ksig1, ksig2, ifilname, iformat, iprd Periodically writes three orchestra control-signal values to an external file.dumpk3 ksig1, ksig2, ksig3, ifilname, iformat, iprd Periodically writes four orchestra control-signal values to an external file.dumpk4 ksig1, ksig2, ksig3, ksig4, ifilname, iformat, iprd Closes a previously opened file.ficlose ihandleficlose Sfilename Read signals from a file at a-rate.fin ifilename, iskipframes, iformat, ain1 [, ain2] [, ain3] [,...] Read signals from a file at i-rate.fini ifilename, iskipframes, iformat, in1 [, in2] [, in3] [, ...] Read signals from a file at k-rate.fink ifilename, iskipframes, iformat, kin1 [, kin2] [, kin3] [,...] Opens a file in a specific mode.ihandle fiopen ifilename, imode Outputs a-rate signals to an arbitrary number of channels.fout ifilename, iformat, aout1 [, aout2, aout3,...,aoutN] Outputs i-rate signals of an arbitrary number of channels to a specified file.fouti ihandle, iformat, iflag, iout1 [, iout2, iout3,....,ioutN] Outputs i-rate signals from an arbitrary number of channels to a specified file.foutir ihandle, iformat, iflag, iout1 [, iout2, iout3,....,ioutN] Outputs k-rate signals of an arbitrary number of channels to a specified file, in raw (headerless) format.foutk ifilename, iformat, kout1 [, kout2, kout3,....,koutN] Similar to printks but prints to a file.fprintks "filename", "string", [, kval1] [, kval2] [...] Similar to prints but prints to a file.fprints "filename", "string" [, ival1] [, ival2] [...] Read a line of text from an external file.Sres, kline readf ifilname Read a line of text from an external file.Sres, iline readfi ifilname Periodically reads an orchestra control-signal value from an external file.kres readk ifilname, iformat, iprd Periodically reads two orchestra control-signal values from an external file.kr1, kr2 readk2 ifilname, iformat, iprd Periodically reads three orchestra control-signal values from an external file.kr1, kr2, kr3 readk3 ifilname, iformat, iprd Periodically reads four orchestra control-signal values from an external file.kr1, kr2, kr3, kr4 readk4 ifilname, iformat, iprd Deprecated. Reads audio data from an external device or stream and can alter its pitch.ar1 [, ar2 [, ar3 [, ... arN]]] diskin ifilcod, kpitch [, iskiptim] \ [, iwraparound] [, iformat] [, iskipinit] Reads audio data from a file, and can alter its pitch using one of several available interpolation types, as well as convert the sample rate to match the orchestra sr setting.a1[, a2[, ... aN]] diskin2 ifilcod, kpitch[, iskiptim \ [, iwrap[, iformat [, iwsize[, ibufsize[, iskipinit]]]]]] Reads mono audio data from an external device or stream.ar1 in Reads a 32-channel audio signal from an external device or stream.ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, ar13, ar14, \ ar15, ar16, ar17, ar18, ar19, ar20, ar21, ar22, ar23, ar24, ar25, ar26, \ ar27, ar28, ar29, ar30, ar31, ar32 in32 Reads from numbered channels in an external audio signal or stream.ain1[, ...] inch kchan1[,...] Reads six-channel audio data from an external device or stream.ar1, ar2, ar3, ar4, ar5, ar6 inh Reads eight-channel audio data from an external device or stream.ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8 ino Reads quad audio data from an external device or stream.ar1, ar2, ar3, a4 inq Allow input from a range of adjacent audio channels from the audio input deviceinrg kstart, ain1 [,ain2, ain3, ..., ainN] Reads stereo audio data from an external device or stream.ar1, ar2 ins Reads a k-rate signal from a user-defined channel.kvalue invalue "channel name"Sname invalue "channel name" Reads a 16-channel audio signal from an external device or stream.ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, \ ar13, ar14, ar15, ar16 inx Reads multi-channel audio samples into a ZAK array from an external device or stream.inz ksig1 Reads stereo audio data from an external MP3 file.ar1, ar2 mp3in ifilcod[, iskptim, iformat, iskipinit, ibufsize] Reads audio data from an external device or stream.ar1[, ar2[, ar3[, ... a24]]] soundin ifilcod [, iskptim] [, iformat] \ [, iskipinit] [, ibufsize] A MIDI delay opcode.mdelay kstatus, kchan, kd1, kd2, kdelay Returns the audio spout frame.aout1 [,aout2 ... aoutX] monitor Writes mono audio data to an external device or stream.out asig Writes 32-channel audio data to an external device or stream.out32 asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8, asig10, \ asig11, asig12, asig13, asig14, asig15, asig16, asig17, asig18, \ asig19, asig20, asig21, asig22, asig23, asig24, asig25, asig26, \ asig27, asig28, asig29, asig30, asig31, asig32 Writes audio data with an arbitrary number of channels to an external device or stream.outc asig1 [, asig2] [...] Writes multi-channel audio data, with user-controllable channels, to an external device or stream.outch kchan1, asig1 [, kchan2] [, asig2] [...] Writes 6-channel audio data to an external device or stream.outh asig1, asig2, asig3, asig4, asig5, asig6 Writes 8-channel audio data to an external device or stream.outo asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8 Writes 4-channel audio data to an external device or stream.outq asig1, asig2, asig3, asig4 Writes samples to quad channel 1 of an external device or stream.outq1 asig Writes samples to quad channel 2 of an external device or stream.outq2 asig Writes samples to quad channel 3 of an external device or stream.outq3 asig Writes samples to quad channel 4 of an external device or stream.outq4 asig Allow output to a range of adjacent audio channels on the audio output deviceoutrg kstart, aout1 [,aout2, aout3, ..., aoutN] Writes stereo audio data to an external device or stream.outs asig1, asig2 Writes samples to stereo channel 1 of an external device or stream.outs1 asig Writes samples to stereo channel 2 of an external device or stream.outs2 asig Sends a k-rate signal or string to a user-defined channel.outvalue "channel name", kvalueoutvalue "channel name", "string" Writes 16-channel audio data to an external device or stream.outx asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8, \ asig9, asig10, asig11, asig12, asig13, asig14, asig15, asig16 Writes multi-channel audio data from a ZAK array to an external device or stream.outz ksig1 Deprecated. Writes audio output to a disk file.soundout asig1, ifilcod [, iformat] Deprecated. Writes audio output to a disk file.soundouts asigl, asigr, ifilcod [, iformat] Reads data from the software buskval chani kchanaval chani kchan Send data to the outwards software buschano kval, kchanchano aval, kchan Declare a channel of the named software bus. chn_k Sname, imode[, itype, idflt, imin, ima, ix, iy, iwidth, iheight, Sattributes] chn_a Sname, imode chn_S Sname, imode Clears an audio output channel of the named software bus.chnclear Sname Export a global variable as a channel of the bus.gival chnexport Sname, imode[, itype, idflt, imin, imax]gkval chnexport Sname, imode[, itype, idflt, imin, imax]gaval chnexport Sname, imodegSval chnexport Sname, imode Reads data from the software bus.ival chnget Snamekval chnget Snameaval chnget SnameSval chnget Sname Writes audio data to the named software bus, mixing to the previous output.chnmix aval, Sname Query parameters of a channel.itype, imode, ictltype, idflt, imin, imax chnparams Recieves data from the software bus.ival chnrecv Snamekval chnrecv Snameaval chnrecv SnameSval chnrecv Sname Sends data via the named software bus.chnsend ival, Snamechnsend kval, Snamechnsend aval, Snamechnsend Sval, Sname Writes data to the named software bus.chnset ival, Snamechnset kval, Snamechnset aval, Snamechnset Sval, Sname Sets the local ksmps value in an instrument or user-defined opcode blocksetksmps iksmps Passes variables to a user-defined opcode block,xinarg1 [, xinarg2] ... [xinargN] xin Retrieves variables from a user-defined opcode block,xout xoutarg1 [, xoutarg2] ... [, xoutargN] Displays the Fourier Transform of an audio or control signal.dispfft xsig, iprd, iwsiz [, iwtyp] [, idbout] [, iwtflg] Displays the audio or control signals as an amplitude vs. time graph.display xsig, iprd [, inprds] [, iwtflg] Allows text to be displayed from instruments like slidersflashtxt iwhich, String Displays the values init (i-rate) variables.print iarg [, iarg1] [, iarg2] [...] printf-style formatted outputprintf_i Sfmt, itrig, [iarg1[, iarg2[, ... ]]]printf Sfmt, ktrig, [xarg1[, xarg2[, ... ]]] Prints one k-rate value at specified intervals.printk itime, kval [, ispace] Prints a new value every time a control variable changes.printk2 kvar [, inumspaces] Prints at k-rate using a printf() style syntax.printks "string", itime [, kval1] [, kval2] [...] Prints at init-time using a printf() style syntax.prints "string" [, kval1] [, kval2] [...] Returns the number of bits in each sample in a sound file.ir filebit ifilcod [, iallowraw] Returns the length of a sound file.ir filelen ifilcod, [iallowraw] Returns the number of channels in a sound file.ir filenchnls ifilcod [, iallowraw] Returns the peak absolute value of a sound file.ir filepeak ifilcod [, ichnl] Returns the sample rate of a sound file.ir filesr ifilcod [, iallowraw] Checks that a file can be used.ir filevalid ifilcod Returns the length of an MP3 sound file.ir mp3len ifilcod Adjust one audio signal according to the values of another.ares balance asig, acomp [, ihp] [, iskip] Clips a signal to a predefined limit.ares clip asig, imeth, ilimit [, iarg] Compress, limit, expand, duck or gate an audio signal.ar compress aasig, acsig, kthresh, kloknee, khiknee, kratio, katt, krel, ilook A dynamic compressor/expander.ares dam asig, kthreshold, icomp1, icomp2, irtime, iftime Adjusts the amplitude audio signal according to a root-mean-square value.ares gain asig, krms [, ihp] [, iskip] Convolves a signal and an impulse response.ar1 [, ar2] [, ar3] [, ar4] convolve ain, ifilcod [, ichannel] Cross synthesis using FFT's.ares cross2 ain1, ain2, isize, ioverlap, iwin, kbias A direct convolution opcode.ares dconv asig, isize, ifn Low latency multichannel convolution, using a function table as impulse response source.a1[, a2[, a3[, ... a8]]] ftconv ain, ift, iplen[, iskipsamples \ [, iirlen[, iskipinit]]] Morphs between multiple ftables as specified in a list.ftmorf kftndx, iftfn, iresfn Convolution based on a uniformly partitioned overlap-save algorithmar1 [, ar2] [, ar3] [, ar4] pconvolve ain, ifilcod [, ipartitionsize, ichannel] Delays an input signal by some time interval.ares delay asig, idlt [, iskip] Delays an input signal by one sample.ares delay1 asig [, iskip] Delays an input signal by some time interval.kr delayk ksig, idel[, imode]kr vdel_k ksig, kdel, imdel[, imode] Reads from an automatically established digital delay line.ares delayr idlt [, iskip] Writes the audio signal to a digital delay line.delayw asig Taps a delay line at variable offset times.ares deltap kdlt Taps a delay line at variable offset times, uses cubic interpolation.ares deltap3 xdlt Taps a delay line at variable offset times, uses interpolation.ares deltapi xdlt Taps a delay line at variable offset times.ares deltapn xnumsamps Read from or write to a delay line with interpolation.aout deltapx adel, iwsize Mixes the input signal to a delay line.deltapxw ain, adel, iwsize Multitap delay line implementation.ares multitap asig [, itime1, igain1] [, itime2, igain2] [...] Short description. Single line for opcode listing.outarg1, outarg2 newopcodename inarg1, inarg2 An interpolating variable time delay.ares vdelay asig, adel, imaxdel [, iskip] A variable time delay with cubic interpolation.ares vdelay3 asig, adel, imaxdel [, iskip] A variable delay opcode with high quality interpolation.aout vdelayx ain, adl, imd, iws [, ist] A 4-channel variable delay opcode with high quality interpolation.aout1, aout2, aout3, aout4 vdelayxq ain1, ain2, ain3, ain4, adl, imd, iws [, ist] A stereo variable delay opcode with high quality interpolation.aout1, aout2 vdelayxs ain1, ain2, adl, imd, iws [, ist] Variable delay opcodes with high quality interpolation.aout vdelayxw ain, adl, imd, iws [, ist] Variable delay opcodes with high quality interpolation.aout1, aout2, aout3, aout4 vdelayxwq ain1, ain2, ain3, ain4, adl, \ imd, iws [, ist] Variable delay opcodes with high quality interpolation.aout1, aout2 vdelayxws ain1, ain2, adl, imd, iws [, ist] Deprecated. Decodes an ambisonic B format signal.ao1, ao2 bformdec isetup, aw, ax, ay, az [, ar, as, at, au, av \ [, abk, al, am, an, ao, ap, aq]]ao1, ao2, ao3, ao4 bformdec isetup, aw, ax, ay, az [, ar, as, at, \ au, av [, abk, al, am, an, ao, ap, aq]]ao1, ao2, ao3, ao4, ao5 bformdec isetup, aw, ax, ay, az [, ar, as, \ at, au, av [, abk, al, am, an, ao, ap, aq]]ao1, ao2, ao3, ao4, ao5, ao6, ao7, ao8 bformdec isetup, aw, ax, ay, az \ [, ar, as, at, au, av [, abk, al, am, an, ao, ap, aq]]] Decodes an ambisonic B format signalao1, ao2 bformdec1 isetup, aw, ax, ay, az [, ar, as, at, au, av \ [, abk, al, am, an, ao, ap, aq]]ao1, ao2, ao3, ao4 bformdec1 isetup, aw, ax, ay, az [, ar, as, at, \ au, av [, abk, al, am, an, ao, ap, aq]]ao1, ao2, ao3, ao4, ao5 bformdec1 isetup, aw, ax, ay, az [, ar, as, \ at, au, av [, abk, al, am, an, ao, ap, aq]]ao1, ao2, ao3, ao4, ao5, ao6, ao7, ao8 bformdec1 isetup, aw, ax, ay, az \ [, ar, as, at, au, av [, abk, al, am, an, ao, ap, aq]]] Deprecated. Codes a signal into the ambisonic B format.aw, ax, ay, az bformenc asig, kalpha, kbeta, kord0, kord1aw, ax, ay, az, ar, as, at, au, av bformenc asig, kalpha, kbeta, \ kord0, kord1 , kord2aw, ax, ay, az, ar, as, at, au, av, ak, al, am, an, ao, ap, aq bformenc \ asig, kalpha, kbeta, kord0, kord1, kord2, kord3 Codes a signal into the ambisonic B format.aw, ax, ay, az bformenc1 asig, kalpha, kbetaaw, ax, ay, az, ar, as, at, au, av bformenc1 asig, kalpha, kbetaaw, ax, ay, az, ar, as, at, au, av, ak, al, am, an, ao, ap, aq bformenc1 \ asig, kalpha, kbeta Generates 3D binaural audio with high-fidelity early reflections in a parametric room using a Phase Truncation algorithm.aleft, aright, irt60low, irt60high, imfp hrtfearly asrc, ksrcx, ksrcy, ksrcz, klstnrx, klstnry, klstnrz, \ ifilel, ifiler, idefroom [,ifade, isr, iorder, ithreed, kheadrot, iroomx, iroomy, iroomz, iwallhigh, \ iwalllow, iwallgain1, iwallgain2, iwallgain3, ifloorhigh, ifloorlow, ifloorgain1, ifloorgain2, \ ifloorgain3, iceilinghigh, iceilinglow, iceilinggain1, iceilinggain2, iceilinggain3] Creates 3D audio for two speakers.aleft, aright hrtfer asig, kaz, kelev, HRTFcompact Generates dynamic 3d binaural audio for headphones using magnitude interpolation and phase truncation.aleft, aright hrtfmove asrc, kAz, kElev, ifilel, ifiler [, imode, ifade, isr] Generates dynamic 3d binaural audio for headphones using a Woodworth based spherical head model with improved low frequency phase accuracy.aleft, aright hrtfmove2 asrc, kAz, kElev, ifilel, ifiler [,ioverlap, iradius, isr] A binaural, dynamic FDN based diffuse-field reverberator. The opcode works independently as an efficient, flexible reverberator.aleft, aright, idel hrtfreverb asrc, ilowrt60, ihighrt60, ifilel, ifiler [,isr, imfp, iorder] Generates static 3d binaural audio for headphones using a Woodworth based spherical head model with improved low frequency phase accuracy. aleft, aright hrtfstat asrc, iAz, iElev, ifilel, ifiler [,iradius, isr] Distributes the audio signals of a previousa1, a2 locsenda1, a2, a3, a4 locsend Takes an input signal and distributes between 2 or 4 channels.a1, a2 locsig asig, kdegree, kdistance, kreverbsenda1, a2, a3, a4 locsig asig, kdegree, kdistance, kreverbsend Distribute an audio signal amongst four channels.a1, a2, a3, a4 pan asig, kx, ky, ifn [, imode] [, ioffset] Distribute an audio signal across two channels.a1, a2 pan2 asig, xp [, imode] Distributes an input signal among 4 channels using cartesian coordinates.a1, a2, a3, a4 space asig, ifn, ktime, kreverbsend, kx, ky Positions the input sound in a 3D space and allows moving the sound at k-rate.aW, aX, aY, aZ spat3d ain, kX, kY, kZ, idist, ift, imode, imdel, iovr [, istor] Positions the input sound in a 3D space with the sound source position set at i-time.aW, aX, aY, aZ spat3di ain, iX, iY, iZ, idist, ift, imode [, istor] Can be used to render an impulse response for a 3D space at i-time.spat3dt ioutft, iX, iY, iZ, idist, ift, imode, irlen [, iftnocl] Calculates distance values from xy coordinates.k1 spdist ifn, ktime, kx, ky Generates output signals based on a previously defineda1, a2, a3, a4 spsend Distributes an audio signal among many channels.ar1[, ar2...] vbap asig, kazim [, kelev] [, kspread] [, ilayout] Distributes an audio signal among 16 channels.ar1, ..., ar16 vbap16 asig, kazim [, kelev] [, kspread] Distribute an audio signal among 16 channels with moving virtual sources.ar1, ..., ar16 vbap16move asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] Distributes an audio signal among 4 channels.ar1, ar2, ar3, ar4 vbap4 asig, kazim [, kelev] [, kspread] Distributes an audio signal among 4 channels with moving virtual sources.ar1, ar2, ar3, ar4 vbap4move asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] Distributes an audio signal among 8 channels.ar1, ..., ar8 vbap8 asig, kazim [, kelev] [, kspread] Distributes an audio signal among 8 channels with moving virtual sources.ar1, ..., ar8 vbap8move asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] Calculates the gains for a sound location between multiple channels.k1[, k2...] vbapg kazim [,kelev] [, kspread] [, ilayout] Configures VBAP output according to loudspeaker parameters.vbaplsinit idim, ilsnum [, idir1] [, idir2] [...] [, idir32] Distributes an audio signal among many channels with moving virtual sources.ar1[, ar2...] vbapmove asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] Writes a multi-channel audio signal to a ZAK array.vbapz inumchnls, istartndx, asig, kazim [, kelev] [, kspread] Writes a multi-channel audio signal to a ZAK array with moving virtual sources.vbapzmove inumchnls, istartndx, asig, idur, ispread, ifldnum, ifld1, \ ifld2, [...] Reverberates an input signal with a flat frequency response.ares alpass asig, krvt, ilpt [, iskip] [, insmps] A physical model reverberator.a1, a2 babo asig, ksrcx, ksrcy, ksrcz, irx, iry, irz [, idiff] [, ifno] Reverberates an input signal with aares comb asig, krvt, ilpt [, iskip] [, insmps] Reverberates an input signal with aares combinv asig, krvt, ilpt [, iskip] [, insmps] Opcode version of Jezar's FreeverbaoutL, aoutR freeverb ainL, ainR, kRoomSize, kHFDamp[, iSRate[, iSkip]] Three different nested all-pass filters.ares nestedap asig, imode, imaxdel, idel1, igain1 [, idel2] [, igain2] \ [, idel3] [, igain3] [, istor] A reverberator consisting of 6 parallel comb-lowpass filters.ares nreverb asig, ktime, khdif [, iskip] [,inumCombs] [, ifnCombs] \ [, inumAlpas] [, ifnAlpas] Models the reverberation of a metal plate.a1[, a2, ...] platerev itabexcite. itabouts, kbndry, iaspect, istiff, idecay, iloss, aexcite1[, aexcite2, ...] Reverberates an input signal with aares reverb asig, krvt [, iskip] Same as the nreverb opcode.ares reverb2 asig, ktime, khdif [, iskip] [,inumCombs] \ [, ifnCombs] [, inumAlpas] [, ifnAlpas] 8 delay line stereo FDN reverb, based on work by Sean CostelloaoutL, aoutR reverbsc ainL, ainR, kfblvl, kfco[, israte[, ipitchm[, iskip]]] Variably reverberates an input signal with a flat frequency response.ares valpass asig, krvt, xlpt, imaxlpt [, iskip] [, insmps] Variably reverberates an input signal with aares vcomb asig, krvt, xlpt, imaxlpt [, iskip] [, insmps] Mixes low level noise to a list of a-rate signalsdenorm a1[, a2[, a3[, ... ]]] Modify a signal by differentiation.ares diff asig [, iskip]kres diff ksig [, iskip] Modify a signal by down-sampling.kres downsamp asig [, iwlen] Adds artificial foldover to an audio signal.ares fold asig, kincr Modify a signal by integration.ares integ asig [, iskip]kres integ ksig [, iskip] Converts a control signal to an audio signal using linear interpolation.ares interp ksig [, iskip] [, imode] Calculates the weighted mean value of two input signals.ares ntrpol asig1, asig2, kpoint [, imin] [, imax]ires ntrpol isig1, isig2, ipoint [, imin] [, imax]kres ntrpol ksig1, ksig2, kpoint [, imin] [, imax] Converts a k-rate parameter to an a-rate value with interpolation.a(x) (control-rate args only) Returns an init-type equivalent of a k-rate argument, or directly returns an i-rate argument.i(x) (control-rate or init-rate arg) Converts a i-rate parameter to an k-rate value. Or converts an a-rate value to a k-rate value by down-sampling.k(x) (i-rate args only)k(x) (a-rate args only) Performs a sample-and-hold operation on its input.ares samphold asig, agate [, ival] [, ivstor]kres samphold ksig, kgate [, ival] [, ivstor] Modify a signal by up-sampling.ares upsamp ksig Access values of the current buffer of an a-rate variable by indexing.kval vaget kndx, avar Write value of into the current buffer of an a-rate variable by index.vaset kval, kndx, avar Sets the lower and upper limits of the value it processes.ares limit asig, klow, khighires limit isig, ilow, ihighkres limit ksig, klow, khigh Reflects the signal that exceeds the low and high thresholds.ares mirror asig, klow, khighires mirror isig, ilow, ihighkres mirror ksig, klow, khigh Wraps-around the signal that exceeds the low and high thresholds.ares wrap asig, klow, khighires wrap isig, ilow, ihighkres wrap ksig, klow, khigh Distort an audio signal via waveshaping and optional clipping.ar distort asig, kdist, ifn[, ihp, istor] Modified hyperbolic tangent distortion.ares distort1 asig, kpregain, kpostgain, kshape1, kshape2[, imode] A user controlled flanger.ares flanger asig, adel, kfeedback [, imaxd] Analyze an audio input and generate harmonizing voices in synchrony.ares harmon asig, kestfrq, kmaxvar, kgenfreq1, kgenfreq2, imode, \ iminfrq, iprd Analyze an audio input and generate harmonizing voices in synchrony with formants preserved.ares harmon2 asig, koct, kfrq1, kfrq2, icpsmode, ilowest[, ipolarity]ares harmon3 asig, koct, kfrq1, \ kfrq2, kfrq3, icpsmode, ilowest[, ipolarity]ares harmon4 asig, koct, kfrq1, \ kfrq2, kfrq3, kfrq4, icpsmode, ilowest[, ipolarity] First-order allpass filters arranged in a series.ares phaser1 asig, kfreq, kord, kfeedback [, iskip] Second-order allpass filters arranged in a series.ares phaser2 asig, kfreq, kq, kord, kmode, ksep, kfeedback A hi-pass filter whose transfer functions are the complements of theares atone asig, khp [, iskip] Emulates a stack of filters using the atone opcode.ares atonex asig, khp [, inumlayer] [, iskip] A sweepable general purpose biquadratic digital filter.ares biquad asig, kb0, kb1, kb2, ka0, ka1, ka2 [, iskip] A sweepable general purpose biquadratic digital filter with a-rate parameters.ares biquada asig, ab0, ab1, ab2, aa0, aa1, aa2 [, iskip] Same as the butterbp opcode.ares butbp asig, kfreq, kband [, iskip] Same as the butterbr opcode.ares butbr asig, kfreq, kband [, iskip] Same as the butterhp opcode.ares buthp asig, kfreq [, iskip] Same as the butterlp opcode.ares butlp asig, kfreq [, iskip] A band-pass Butterworth filter.ares butterbp asig, kfreq, kband [, iskip] A band-reject Butterworth filter.ares butterbr asig, kfreq, kband [, iskip] A high-pass Butterworth filter.ares butterhp asig, kfreq [, iskip] A low-pass Butterworth filter.ares butterlp asig, kfreq [, iskip] Implements low-pass and high-pass filters of different styles.ares clfilt asig, kfreq, itype, inpol [, ikind] [, ipbr] [, isba] [, iskip] A fast and robust method for approximating sound propagation, achieving convincing Doppler shifts without having to solve equations.ashifted doppler asource, ksourceposition, kmicposition [, isoundspeed, ifiltercutoff] A median filter, a variant FIR lowpass filter.ares median asig, ksize, imaxsize [, iskip] A median filter, a variant FIR lowpass filter.kres mediank kin, ksize, imaxsize [, iskip] A filter that simulates a mass-spring-damper systemaout mode ain, kfreq, kQ [, iskip] A first-order recursive low-pass filter with variable frequency response.ares tone asig, khp [, iskip] Emulates a stack of filters using the tone opcode.ares tonex asig, khp [, inumlayer] [, iskip] A notch filter whose transfer functions are the complements of the reson opcode.ares areson asig, kcf, kbw [, iscl] [, iskip] A second-order multi-mode filter.ares bqrez asig, xfco, xres [, imode] [, iskip] A resonant lowpass filter.ares lowpass2 asig, kcf, kq [, iskip] Another resonant lowpass filter.ares lowres asig, kcutoff, kresonance [, iskip] Simulates layers of serially connected resonant lowpass filters.ares lowresx asig, kcutoff, kresonance [, inumlayer] [, iskip] A 3-pole sweepable resonant lowpass filter.ares lpf18 asig, kfco, kres, kdist [, iskip] Moog ladder lowpass filter.asig moogladder ain, kcf, kres[, istor] A digital emulation of the Moog diode ladder filter configuration.ares moogvcf asig, xfco, xres [,iscale, iskip] A digital emulation of the Moog diode ladder filter configuration.ares moogvcf2 asig, xfco, xres [,iscale, iskip] A second-order resonant filter.ares reson asig, kcf, kbw [, iscl] [, iskip] A bandpass filter with variable frequency response.ares resonr asig, kcf, kbw [, iscl] [, iskip] Emulates a stack of filters using the reson opcode.ares resonx asig, kcf, kbw [, inumlayer] [, iscl] [, iskip] A bank of second-order bandpass filters, connected in parallel.ares resony asig, kbf, kbw, inum, ksep [, isepmode] [, iscl] [, iskip] A bandpass filter with variable frequency response.ares resonz asig, kcf, kbw [, iscl] [, iskip] A resonant low-pass filter.ares rezzy asig, xfco, xres [, imode, iskip] State-variable filter.ahp,alp,abp,abr statevar ain, kcf, kq [, iosamps, istor] A resonant second order filter, with simultaneous lowpass, highpass and bandpass outputs.alow, ahigh, aband svfilter asig, kcf, kq [, iscl] Models some of the filter characteristics of a Roland TB303 voltage-controlled filter.ares tbvcf asig, xfco, xres, kdist, kasym [, iskip] A bank of filters in which the cutoff frequency can be separated under user control.ares vlowres asig, kfco, kres, iord, ksep A notch filter whose transfer functions are the complements of the reson opcode.kres aresonk ksig, kcf, kbw [, iscl] [, iskip] A hi-pass filter whose transfer functions are the complements of thekres atonek ksig, khp [, iskip] Generate glissandos starting from a control signal.kres lineto ksig, ktime Applies portamento to a step-valued control signal.kres port ksig, ihtim [, isig] Applies portamento to a step-valued control signal.kres portk ksig, khtim [, isig] A second-order resonant filter.kres resonk ksig, kcf, kbw [, iscl] [, iskip] Control signal resonant filter stack.kres resonxk ksig, kcf, kbw[, inumlayer, iscl, istor] Generate glissandos starting from a control signal.kres tlineto ksig, ktime, ktrig A first-order recursive low-pass filter with variable frequency response.kres tonek ksig, khp [, iskip] A DC blocking filter.ares dcblock ain [, igain] A DC blocking filter.ares dcblock2 ain [, iorder] [, iskip] Equalizer filterasig eqfil ain, kcf, kbw, kgain[, istor] Performs filtering using a transposed form-II digital filter lattice with no time-varying control.ares filter2 asig, iM, iN, ib0, ib1, ..., ibM, ia1, ia2, ..., iaNkres filter2 ksig, iM, iN, ib0, ib1, ..., ibM, ia1, ia2, ..., iaN Formant filter.asig fofilter ain, kcf, kris, kdec[, istor] A Hilbert transformer.ar1, ar2 hilbert asig A filter with a non-linear effect.ares nlfilt ain, ka, kb, kd, kC, kL A filter with a non-linear effect and blowup protection.ares nlfilt2 ain, ka, kb, kd, kC, kL Implementation of Zoelzer's parametric equalizer filters.ares pareq asig, kc, kv, kq [, imode] [, iskip] Parametric equalizer and filter opcode with 7 filter types, based on algorithm by Robert Bristow-Johnson.ar rbjeq asig, kfco, klvl, kQ, kS[, imode] Performs filtering using a transposed form-II digital filter lattice with radial pole-shearing and angular pole-warping.ares zfilter2 asig, kdamp, kfreq, iM, iN, ib0, ib1, ..., ibM, \ ia1,ia2, ..., iaN A simple waveguide model consisting of one delay-line and one first-order lowpass filter.ares wguide1 asig, xfreq, kcutoff, kfeedback A model of beaten plate consisting of two parallel delay-lines and two first-order lowpass filters.ares wguide2 asig, xfreq1, xfreq2, kcutoff1, kcutoff2, \ kfeedback1, kfeedback2 Efficiently evaluates the sum of Chebyshev polynomials of arbitrary order.aout chebyshevpoly ain, k0 [, k1 [, k2 [...]]] Performs linear clipping on an audio signal or a phasor.aout pdclip ain, kWidth, kCenter [, ibipolar [, ifullscale]] Distorts a phasor for reading the two halves of a table at different rates.aout pdhalf ain, kShapeAmount [, ibipolar [, ifullscale]] Distorts a phasor for reading two unequal portions of a table in equal periods.aout pdhalfy ain, kShapeAmount [, ibipolar [, ifullscale]] Waveshapes a signal by raising it to a variable exponent.aout powershape ain, kShapeAmount [, ifullscale] Produces a signal that is the maximum of any number of input signals.amax max ain1, ain2 [, ain3] [, ain4] [...]kmax max kin1, kin2 [, kin3] [, kin4] [...] Local maximum (or minimum) value of an incoming asig signalknumkout max_k asig, ktrig, itype Produces a signal that is the maximum of the absolute values of any number of input signals.amax maxabs ain1, ain2 [, ain3] [, ain4] [...]kmax maxabs kin1, kin2 [, kin3] [, kin4] [...] Accumulates the maximum of the absolute values of audio signals.maxabsaccum aAccumulator, aInput Accumulates the maximum value of audio signals.maxaccum aAccumulator, aInput Produces a signal that is the minimum of any number of input signals.amin min ain1, ain2 [, ain3] [, ain4] [...]kmin min kin1, kin2 [, kin3] [, kin4] [...] Produces a signal that is the minimum of the absolute values of any number of input signals.amin minabs ain1, ain2 [, ain3] [, ain4] [...]kmin minabs kin1, kin2 [, kin3] [, kin4] [...] Accumulates the minimum of the absolute values of audio signals.minabsaccum aAccumulator, aInput Accumulates the minimum value of audio signals.minaccum aAccumulator, aInput Stops one of a number of internal clocks.clockoff inum Starts one of a number of internal clocks.clockon inum Compares two values for equality.(a == b ? v1 : v2) Determines if one value is greater than or equal to another.(a >= b ? v1 : v2) Determines if one value is greater than another.(a > b ? v1 : v2) Determines if one value is less than or equal to another.(a <= b ? v1 : v2) Determines if one value is less than another.(a < b ? v1 : v2) Determines if one value is not equal to another.(a != b ? v1 : v2) compiles a new orchestra from an ASCII fileires compileorc Sfilename compiles a new orchestra passed in as an ASCII stringires compilestr Sorch Evalstrs evaluates a string containing Csound code, returning a value.ires evalstr Scode Returns a value from an instrument.return ival Creates a held note.ihold Enables an instrument to turn itself off.turnoff Turn off instance(s) of other instruments at performance time.turnoff2 kinsno, kmode, krelease Activate an instrument for an indefinite time.turnon insnum [, itime] Generates a score event from an instrument.event "scorechar", kinsnum, kdelay, kdur, [, kp4] [, kp5] [, ...]event "scorechar", "insname", kdelay, kdur, [, kp4] [, kp5] [, ...] Generates a score event from an instrument.event_i "scorechar", iinsnum, idelay, idur, [, ip4] [, ip5] [, ...]event_i "scorechar", "insname", idelay, idur, [, ip4] [, ip5] [, ...] Mutes/unmutes new instances of a given instrument.mute insnum [, iswitch]mute "insname" [, iswitch] Read, preprocess and schedule a score from an input string.readscore Sin Removes the definition of an instrument.remove insnum Adds a new score event generated by a k-rate trigger.schedkwhen ktrigger, kmintim, kmaxnum, kinsnum, kwhen, kdur \ [, ip4] [, ip5] [...]schedkwhen ktrigger, kmintim, kmaxnum, "insname", kwhen, kdur \ [, ip4] [, ip5] [...] Similar to schedkwhen but uses a named instrument at init-time.schedkwhennamed ktrigger, kmintim, kmaxnum, "name", kwhen, kdur \ [, ip4] [, ip5] [...] Adds a new score event.schedule insnum, iwhen, idur [, ip4] [, ip5] [...]schedule "insname", iwhen, idur [, ip4] [, ip5] [...] Adds a new score event.schedwhen ktrigger, kinsnum, kwhen, kdur [, ip4] [, ip5] [...]schedwhen ktrigger, "insname", kwhen, kdur [, ip4] [, ip5] [...] Issues one or more score line events from an instrument.scoreline Sin, ktrig Issues one or more score line events from an instrument at i-time.scoreline_i Sin Conditionally transfer control on every pass.cggoto condition, label Conditionally transfer control during the i-time pass.cigoto condition, label Conditionally transfer control during the p-time passes.ckgoto condition, label Transfers control on every pass when a condition is not true.cngoto condition, label Executes a block of code when an "if...then" condition is false.else Defines another "if...then" condition when a "if...then" condition is false.elseif xa R xb then Closes a block of code that begins with an "if...then" statement.endif Transfer control on every pass.goto label Branches conditionally at initialization or during performance time.if ia R ib igoto labelif ka R kb kgoto labelif xa R xb goto labelif xa R xb then Transfer control during the i-time pass.igoto label Transfer control during the performance-time passes.kgoto label Looping constructions.loop_ge indx, idecr, imin, labelloop_ge kndx, kdecr, kmin, label Looping constructions.loop_gt indx, idecr, imin, labelloop_gt kndx, kdecr, kmin, label Looping constructions.loop_le indx, incr, imax, labelloop_le kndx, kncr, kmax, label Looping constructions.loop_lt indx, incr, imax, labelloop_lt kndx, kncr, kmax, label Transfer control at i-time when a new note is being tied onto a previously held notetigoto label Conditional branch during p-time depending on elapsed note time.timout istrt, idur, label A syntactic looping construction.until condition do ... od Returns the number of active instances of an instrument.ir active insnum [,iopt]ir active Sinsname [,iopt]kres active kinsnum [,iopt] Reports the usage of cpu either total or per core.ktot[,kcpu1, kcpu2,...]cpumeter ifreq Control allocation of cpu resources on a per-instrument basis, to optimize realtime output.cpuprc insnum, ipercentcpuprc Sinsname, ipercent Exit Csound as fast as possible, with no cleaning up.exitnow Start/stop jack_transport and can optionally relocate the playback head.jacktransport icommand [, ilocation] Limits the number of allocations of an instrument.maxalloc insnum, icountmaxalloc Sinsname, icount Creates space for instruments but does not run them.prealloc insnum, icountprealloc "insname", icount Performs a simple assignment.ares = xargires = iargkres = kargires, ... = iarg, ...kres, ... = karg, ...table [ kval] = karg Puts the value of the i-time expression into a k-, a-rate or t- variable.ares init iargires init iargkres init iargares, ... init iarg, ...ires, ... init iarg, ...kres, ... init iarg, ...tab init isize[, ival] Returns the number of a named instrument.insno nstrnum "name" Show the value in a given p-field.p(x) Assigns a range of p-fields to ivariables. ivar1, ... passign [istart] Defines and initializes numeric arrays at orchestra load time.pset icon1 [, icon2] [...] Suspends a performance while a special initialization pass is executed.reinit label Transfers control during a reinit pass.rigoto label Terminates a reinit pass.rireturn Puts the value of the instrument's internalir tival Sense on-screen controls.kres button knum k-rate signal change detector.ktrig changed kvar1 [, kvar2,..., kvarN] Sense on-screen controls.kres checkbox knum Configurable slider controls for realtime user input.kres control knum Envelope follower unit generator.ares follow asig, idt Another controllable envelope extractor.ares follow2 asig, katt, krel Return Csound settings.Svalue getcfg iopt Reads data from a joystick controller.kres joystick kdevice ktab Trigger Metronomektrig metro kfreq [, initphase] Returns the current tempo at k-rate, of either the MIDI file (if available) or the scoreksig miditempo Reads data from a P5 Glove controller.p5gconnect Reads data fields from an external P5 Glove.kres p5gdata kcontrol Returns the number of pfields belonging to a note event.icount pcount Maintains the output equal to the highest absolute value received.kres peak asigkres peak ksig Returns the value of a specified pfield.ivalue pindex ipfieldIndex Tracks the pitch of a signal.koct, kamp pitch asig, iupdte, ilo, ihi, idbthresh [, ifrqs] [, iconf] \ [, istrt] [, iocts] [, iq] [, inptls] [, irolloff] [, iskip] Follows the pitch of a signal based on the AMDF method.kcps, krms pitchamdf asig, imincps, imaxcps [, icps] [, imedi] \ [, idowns] [, iexcps] [, irmsmedi] Tracks the pitch of a signal.acps, alock plltrack asig, kd [, kloopf, kloopq, klf, khf, kthresh] Tracks the pitch of a signal.kcps, kamp ptrack asig, ihopsize[,ipeaks] returns a value stored in the instance of an instrument.ival readscratch[index] Rewinds the playback position of the current score performance. rewindscore Determines the root-mean-square amplitude of an audio signal.kres rms asig [, ihp] [, iskip] Returns the ASCII code of a key that has been pressed.kres[, kkeydown] sensekey Generates a trigger signal according to the values stored in a table.ktrig_out seqtime ktime_unit, kstart, kloop, kinitndx, kfn_times Generates a trigger signal according to the values stored in a table.ktrig_out seqtime2 ktrig_in, ktime_unit, kstart, kloop, kinitndx, kfn_times Configurable slider controls for realtime user input.setctrl inum, ival, itype Sets the playback position of the current score performance to a given position. setscorepos ipos Split a trigger signalsplitrig ktrig, kndx, imaxtics, ifn, kout1 [,kout2,...,koutN] Estimate the tempo of beat patterns in a control signal.ktemp tempest kin, iprd, imindur, imemdur, ihp, ithresh, ihtim, ixfdbak, \ istartempo, ifn [, idisprd] [, itweek] Apply tempo control to an uninterpreted score.tempo ktempo, istartempo Reads the current value of the tempo.kres tempoval Time Variant Sequencerktrig timedseq ktimpnt, ifn, kp1 [,kp2, kp3, ...,kpN] Informs when a krate signal crosses a threshold.kout trigger ksig, kthreshold, kmode Accepts a trigger signal as input and outputs a group of values.trigseq ktrig_in, kstart, kloop, kinitndx, kfn_values, kout1 [, kout2] [...] Reads data from a number of external Nintendo Wiimote controllers.ires wiiconnect [itimeout, imaxnum] Reads data fields from a number of external Nintendo Wiimote controllers.kres wiidata kcontrol[, knum] Sets scaling and range limits for certain Wiimote fields. wiirange icontrol, iminimum, imaximum[, inum] Sends data to one of a number of external Nintendo Wiimote controllers.kres wiisend kcontrol, kvalue[, knum] writes a value into the scratchpad of the instance of an instrument.writescratchival[, index] Sense the cursor position in an output windowkx, ky xyin iprd, ixmin, ixmax, iymin, iymax [, ixinit] [, iyinit] Pops values from the global stack.xval1, [xval2, ... , xval31] popival1, [ival2, ... , ival31] pop Pops an f-sig frame from the global stack.fsig pop_f Pushes a value into the global stack.push xval1, [xval2, ... , xval31]push ival1, [ival2, ... , ival31] Pushes an f-sig frame into the global stack.push_f fsig Initializes the stack.stack iStackSize Creates and runs a numbered instrument instance.a1, [...] [, a8] subinstr instrnum [, p4] [, p5] [...]a1, [...] [, a8] subinstr "insname" [, p4] [, p5] [...] Creates and runs a numbered instrument instance at init-time.subinstrinit instrnum [, p4] [, p5] [...]subinstrinit "insname" [, p4] [, p5] [...] Returns the number seconds since a base date.ir date Returns as a string the date and time specified.Sir dates [ itime] Reads the value of an internal clock.ir readclock inum Read the real time clock from the operating system.ires rtclockkres rtclock Read absolute time in k-rate cycles.kres timeinstk Read absolute time in seconds.kres timeinsts Read absolute time in k-rate cycles.ires timekkres timek Read absolute time in seconds.ires timeskres times Receives an audio signal from a Jack port.asignal JackoAudioIn ScsoundPortName Creates an audio connection from a Jack port to Csound.JackoAudioInConnect SexternalPortName, ScsoundPortName Sends an audio signal to a Jack port.JackoAudioOut ScsoundPortName, asignal Creates an audio connection from Csound to a Jack port.JackoAudioOutConnect ScsoundPortName, SexternalPortName Turns Jack's freewheeling mode on or off.JackoFreewheel [ienabled] Prints information about the Jack system.JackoInfo Initializes Csound as a Jack client.JackoInit SclientName, ServerName Creates a MIDI connection from a Jack port to Csound.JackoMidiInConnect SexternalPortName, ScsoundPortName Sends a MIDI channel message to a Jack port.JackoMidiOut ScsoundPortName, kstatus, kchannel, kdata1[, kdata2] Creates a MIDI connection from Csound to a Jack port.JackoMidiOutConnect ScsoundPortName, SexternalPortName Sends a MIDI channel message to a Jack port.JackoNoteOut ScsoundPortName, kstatus, kchannel, kdata1[, kdata2] Enables or disables all Jack ports.JackoOn [iactive] Control the Jack transport.JackoTransport kcommand, [kposition] Executes an arbitrary block of Lua code from the Csound orchestra.lua_exec Sluacode Calls a Lua opcode at i-rate only. Any number of output and/or input arguments may be passed. All arguments must be passed on the right-hand side. Outputs are returned in the argument.lua_iopcall Sname, ...lua_ikopcall Sname, ...lua_iaopcall Sname, ...lua_iopcall_off Sname, ...lua_ikopcall_off Sname, ...lua_iaopcall_off Sname, ... Define an opcode in Lua at i-rate. The opcode can take any number of output and/or input arguments of any type.lua_opdef Sname, Sluacode Open a serial port.iPort serialBegin SPortName [, ibaudRate] Close a serial port. serialEnd iPort Flush data from a serial port. serialFlush iPort Print data from a serial port. serialPrint iPort Read data from a serial port.kByte serialRead iPort Write data to a serial port. serialWrite iPort, iByte serialWrite iPort, kByte serialWrite iPort, SBytes Write data to a serial port. serialWrite_i iPort, iByte serialWrite_i iPort, SBytes Deletes function table.ftfree ifno, iwhen Generate a score function table from within the orchestra.gir ftgen ifn, itime, isize, igen, iarga [, iargb ] [...] Generate a score function table from within the orchestra, which is deleted at the end of the note.ifno ftgentmp ip1, ip2dummy, isize, igen, iarga, iargb, ... Loads a sound file into memory for use bysndload Sfname[, ifmt[, ichns[, isr[, ibas[, iamp[, istrt \ [, ilpmod[, ilps[, ilpe]]]]]]]]] Generate a vector with initial values.karray[] array ival1, ival2,.....ivaln Generate a vector with initial values.karray[] fillarray ival1, ival2,.....ivaln Returns the number of channels in a stored function table.ftchnls(x) (init-rate args only) Returns the base frequency of a stored function table in Hz.ftcps(x) (init-rate args only) Returns the size of a stored function table.ftlen(x) (init-rate args only) Returns the loop segment start-time of a stored function table number.ftlptim(x) (init-rate args only) Returns the sampling-rate of a stored function table.ftsr(x) (init-rate args only) Generate a vector with an arithmetic sequence.karray genarray kstart, kens[, inc]iarray genarray istart, iens[, inc] Generate a vector with an arithmetic sequence.karray genarray_i istart, iend [,inc] Calculate the length of a vector.ir lenarray karraykr lenarray karray Apply a function to every element of a vector.karray maparray kinarray, Stringkarray maparray_i kinarray, String Returns the number of samples loaded into a stored function table number.nsamp(x) (init-rate args only) Take a slice of a vector.karray slicearray kinarray, istart, iend Interrogates a function table for length.ires tableng ifnkres tableng kfn Adding values in a range of a table.kr tabsum ifn[[, kmin] [, kmax]] Table Read Access inside expressions.tb0_init ifntb1_init ifntb2_init ifntb3_init ifntb4_init ifntb5_init ifntb6_init ifntb7_init ifntb8_init ifntb9_init ifntb10_init ifntb11_init ifntb12_init ifntb13_init ifntb14_init ifntb15_init ifniout = tb0(iIndex)kout = tb0(kIndex)iout = tb1(iIndex)kout = tb1(kIndex)iout = tb2(iIndex)kout = tb2(kIndex)iout = tb3(iIndex)kout = tb3(kIndex)iout = tb4(iIndex)kout = tb4(kIndex)iout = tb5(iIndex)kout = tb5(kIndex)iout = tb6(iIndex)kout = tb6(kIndex)iout = tb7(iIndex)kout = tb7(kIndex)iout = tb8(iIndex)kout = tb8(kIndex)iout = tb9(iIndex)kout = tb9(kIndex)iout = tb10(iIndex)kout = tb10(kIndex)iout = tb11(iIndex)kout = tb11(kIndex)iout = tb12(iIndex)kout = tb12(kIndex)iout = tb13(iIndex)kout = tb13(kIndex)iout = tb14(iIndex)kout = tb14(kIndex)iout = tb15(iIndex)kout = tb15(kIndex) Provides k-rate control over table numbers.ares tableikt xndx, kfn [, ixmode] [, ixoff] [, iwrap]kres tableikt kndx, kfn [, ixmode] [, ixoff] [, iwrap] Provides k-rate control over table numbers.ares tablekt xndx, kfn [, ixmode] [, ixoff] [, iwrap]kres tablekt kndx, kfn [, ixmode] [, ixoff] [, iwrap] Reads function tables with linear, cubic, or sinc interpolation.ares tablexkt xndx, kfn, kwarp, iwsize [, ixmode] [, ixoff] [, iwrap] Load a set of previously-allocated tables from a file.ftload "filename", iflag, ifn1 [, ifn2] [...] Load a set of previously-allocated tables from a file.ftloadk "filename", ktrig, iflag, ifn1 [, ifn2] [...] Save a set of previously-allocated tables to a file.ftsave "filename", iflag, ifn1 [, ifn2] [...] Save a set of previously-allocated tables to a file.ftsavek "filename", ktrig, iflag, ifn1 [, ifn2] [...] Change the contents of existing function tables of any length.ptablew asig, andx, ifn [, ixmode] [, ixoff] [, iwgmode]ptablew isig, indx, ifn [, ixmode] [, ixoff] [, iwgmode]ptablew ksig, kndx, ifn [, ixmode] [, ixoff] [, iwgmode] Simple, fast table copy opcode.tablecopy kdft, ksft Filters a source table and writes result into a destination table.knumpassed tablefilter kouttable, kintatble, kmode, kparam Filters a source table and writes result into a destination table.inumpassed tablefilteri iouttable, iintatble, imode, iparam Writes a table's guard point.tablegpw kfn Simple, fast table copy opcode.tableicopy idft, isft Writes a table's guard point.tableigpw ifn Mixes two tables.tableimix idft, idoff, ilen, is1ft, is1off, is1g, is2ft, is2off, is2g Change the contents of existing function tables.tableiw isig, indx, ifn [, ixmode] [, ixoff] [, iwgmode] Mixes two tables.tablemix kdft, kdoff, klen, ks1ft, ks1off, ks1g, ks2ft, ks2off, ks2g Reads tables in sequential locations.ares tablera kfn, kstart, koff Change the contents of existing function tables.tablew asig, andx, ifn [, ixmode] [, ixoff] [, iwgmode]tablew isig, indx, ifn [, ixmode] [, ixoff] [, iwgmode]tablew ksig, kndx, ifn [, ixmode] [, ixoff] [, iwgmode] Writes tables in sequential locations.kstart tablewa kfn, asig, koff Change the contents of existing function tables.tablewkt asig, andx, kfn [, ixmode] [, ixoff] [, iwgmode]tablewkt ksig, kndx, kfn [, ixmode] [, ixoff] [, iwgmode] Allow morphing between a set of tables.kout tabmorph kindex, kweightpoint, ktabnum1, ktabnum2, \ ifn1, ifn2 [, ifn3, ifn4, ...,ifnN] Allow morphing between a set of tables at audio rate with interpolation.aout tabmorpha aindex, aweightpoint, atabnum1, atabnum2, \ ifn1, ifn2 [, ifn3, ifn4, ... ifnN] Allow morphing between a set of tables at audio rate with interpolation.aout tabmorphak aindex, kweightpoint, ktabnum1, ktabnum2, \ ifn1, ifn2 [, ifn3, ifn4, ... ifnN] Allow morphing between a set of tables with interpolation.kout tabmorphi kindex, kweightpoint, ktabnum1, ktabnum2, \ ifn1, ifn2 [, ifn3, ifn4, ..., ifnN] Playing-back control signals.tabplay ktrig, knumtics, kfn, kout1 [,kout2,..., koutN] Recording of control signals.tabrec ktrig_start, ktrig_stop, knumtics, kfn, kin1 [,kin2,...,kinN] A FLTK container opcode that groups child widgets.FLgroup "label", iwidth, iheight, ix, iy [, iborder] [, image] Marks the end of a group of FLTK child widgets.FLgroupEnd Provides the functionality of compressing and aligning FLTK widgets.FLpack iwidth, iheight, ix, iy, itype, ispace, iborder Marks the end of a group of compressed or aligned FLTK widgets.FLpackEnd Creates a window that contains FLTK widgets.FLpanel "label", iwidth, iheight [, ix] [, iy] [, iborder] [, ikbdcapture] [, iclose] Marks the end of a group of FLTK widgets contained inside of a window (panel).FLpanelEnd A FLTK opcode that adds scroll bars to an area.FLscroll iwidth, iheight [, ix] [, iy] A FLTK opcode that marks the end of an area with scrollbars.FLscrollEnd Creates a tabbed FLTK interface.FLtabs iwidth, iheight, ix, iy Marks the end of a tabbed FLTK interface.FLtabsEnd A FLTK widget opcode that creates a counter.kout, ihandle FLcount "label", imin, imax, istep1, istep2, itype, \ iwidth, iheight, ix, iy, iopcode [, kp1] [, kp2] [, kp3] [...] [, kpN] A FLTK opcode that acts like a joystick.koutx, kouty, ihandlex, ihandley FLjoy "label", iminx, imaxx, iminy, \ imaxy, iexpx, iexpy, idispx, idispy, iwidth, iheight, ix, iy A FLTK widget opcode that creates a knob.kout, ihandle FLknob "label", imin, imax, iexp, itype, idisp, iwidth, \ ix, iy [, icursorsize] A FLTK widget that creates a transversal knob.kout, ihandle FLroller "label", imin, imax, istep, iexp, itype, idisp, \ iwidth, iheight, ix, iy Puts a slider into the corresponding FLTK container.kout, ihandle FLslider "label", imin, imax, iexp, itype, idisp, iwidth, \ iheight, ix, iy A FLTK widget opcode that creates a textbox.kout, ihandle FLtext "label", imin, imax, istep, itype, iwidth, \ iheight, ix, iy A FLTK widget that displays text inside of a box.ihandle FLbox "label", itype, ifont, isize, iwidth, iheight, ix, iy [, image] A FLTK widget opcode that creates a bank of buttons.kout, ihandle FLbutBank itype, inumx, inumy, iwidth, iheight, ix, iy, \ iopcode [, kp1] [, kp2] [, kp3] [, kp4] [, kp5] [....] [, kpN] A FLTK widget opcode that creates a button.kout, ihandle FLbutton "label", ion, ioff, itype, iwidth, iheight, ix, \ iy, iopcode [, kp1] [, kp2] [, kp3] [, kp4] [, kp5] [....] [, kpN] A FLTK widget opcode that creates a button that will close the panel window it is a part of.ihandle FLcloseButton "label", iwidth, iheight, ix, iy A FLTK widget opcode that creates a button that executes a command.ihandle FLexecButton "command", iwidth, iheight, ix, iy Retrieves a previously stored FLTK snapshot.inumsnap FLgetsnap index [, igroup] Displays a box with a grid useful for visualizing two-dimensional Hyper Vectorial Synthesis.ihandle FLhvsBox inumlinesX, inumlinesY, iwidth, iheight, ix, iy [, image] Sets the cursor position of a previously-declared FLhvsBox widget.FLhvsBox kx, ky, ihandle Reports keys pressed (on alphanumeric keyboard) when an FLTK panel has focus.kascii FLkeyIn [ifn] Loads all snapshots into the memory bank of the current orchestra.FLloadsnap "filename" [, igroup] Returns the mouse position and the state of the three mouse buttons.kx, ky, kb1, kb2, kb3 FLmouse [imode] A FLTK opcode that prints a k-rate value at specified intervals.FLprintk itime, kval, idisp A FLTK opcode that prints a new value every time a control-rate variable changes.FLprintk2 kval, idisp Starts the FLTK widget thread.FLrun Saves all snapshots currently created into a file.FLsavesnap "filename" [, igroup] Stores the current status of all FLTK valuators into a snapshot location.inumsnap, inumval FLsetsnap index [, ifn, igroup] Determines the snapshot group for FL valuators.FLsetSnapGroup igroup Sets the value of a FLTK valuator at control-rate.FLsetVal ktrig, kvalue, ihandle Sets the value of a FLTK valuator to a number provided by the user.FLsetVal_i ivalue, ihandle A FLTK widget containing a bank of horizontal sliders.FLslidBnk "names", inumsliders [, ioutable] [, iwidth] [, iheight] [, ix] \ [, iy] [, itypetable] [, iexptable] [, istart_index] [, iminmaxtable] A FLTK widget containing a bank of horizontal sliders.FLslidBnk2 "names", inumsliders, ioutable, iconfigtable [,iwidth, iheight, ix, iy, istart_index] modify the values of a slider bank.FLslidBnk2Set ihandle, ifn [, istartIndex, istartSlid, inumSlid] modify the values of a slider bank.FLslidBnk2Setk ktrig, ihandle, ifn [, istartIndex, istartSlid, inumSlid] gets the handle of last slider bank created.ihandle FLslidBnkGetHandle modify the values of a slider bank.FLslidBnkSet ihandle, ifn [, istartIndex, istartSlid, inumSlid] modify the values of a slider bank.FLslidBnkSetk ktrig, ihandle, ifn [, istartIndex, istartSlid, inumSlid] Same as the FLrun opcode.FLupdate Shows the current value of a FLTK valuator.ihandle FLvalue "label", iwidth, iheight, ix, iy An FLTK widget opcode that creates a virtual keyboard widget.FLvkeybd "keyboard.map", iwidth, iheight, ix, iy A FLTK widget containing a bank of vertical sliders.FLvslidBnk "names", inumsliders [, ioutable] [, iwidth] [, iheight] [, ix] \ [, iy] [, itypetable] [, iexptable] [, istart_index] [, iminmaxtable] A FLTK widget containing a bank of vertical sliders.FLvslidBnk2 "names", inumsliders, ioutable, iconfigtable [,iwidth, iheight, ix, iy, istart_index] Senses the mouse cursor position in a user-defined area inside an FLpanel.koutx, kouty, kinside FLxyin ioutx_min, ioutx_max, iouty_min, iouty_max, \ iwindx_min, iwindx_max, iwindy_min, iwindy_max [, iexpx, iexpy, ioutx, iouty] Allows one-dimensional HVS (Hyper-Vectorial Synthesis).vphaseseg kphase, ioutab, ielems, itab1,idist1,itab2 \ [,idist2,itab3, ... ,idistN-1,itabN] A FLTK opcode that sets the primary colors.FLcolor ired, igreen, iblue [, ired2, igreen2, iblue2] A FLTK opcode that sets the secondary (selection) color.FLcolor2 ired, igreen, iblue Hides the target FLTK widget.FLhide ihandle A FLTK opcode that modifies the appearance of a text label.FLlabel isize, ifont, ialign, ired, igreen, iblue Sets the text alignment of a label of a FLTK widget.FLsetAlign ialign, ihandle Sets the appearance of a box surrounding a FLTK widget.FLsetBox itype, ihandle Sets the primary color of a FLTK widget.FLsetColor ired, igreen, iblue, ihandle Sets the secondary (or selection) color of a FLTK widget.FLsetColor2 ired, igreen, iblue, ihandle Sets the font type of a FLTK widget.FLsetFont ifont, ihandle Sets the position of a FLTK widget.FLsetPosition ix, iy, ihandle Resizes a FLTK widget.FLsetSize iwidth, iheight, ihandle Sets the label of a FLTK widget.FLsetText "itext", ihandle Sets the color of the text label of a FLTK widget.FLsetTextColor ired, iblue, igreen, ihandle Sets the size of the text label of a FLTK widget.FLsetTextSize isize, ihandle Sets some font attributes of the text label of a FLTK widget.FLsetTextType itype, ihandle Restores the visibility of a previously hidden FLTK widget.FLshow ihandle Addition operatora + b (no rate restriction) Division operator.a / b (no rate restriction) Modulus operator.a % b (no rate restriction) Multiplication operator.a * b (no rate restriction) Logical AND operator.a && b (logical AND; not audio-rate) Bitwise AND operator.a & b (bitwise AND) Bitwise NOT operator.~ a (bitwise NOT) Bitwise OR operator.a | b (bitwise OR) Bitshift left operator.a << b (bitshift left) Bitshift right operator.a >> b (bitshift left) Bitwise NON EQUIVALENCE operator.a # b (bitwise NON EQUIVALENCE) Logical OR operator.a || b (logical OR; not audio-rate) a ^ b (b not audio-rate) Subtraction operator.a - b (no rate restriction) Zeroes a list of audio signals.clear avar1 [, avar2] [, avar3] [...] Accumulates audio signals.vincr accum, aincr Returns an absolute value.abs(x) (no rate restriction) Returns the smallest integer not less thanceil(x) (init-, control-, or audio-rate arg allowed) Returns e raised to the x-th power.exp(x) (no rate restriction) Returns the largest integer not greater thanfloor(x) (init-, control-, or audio-rate arg allowed) Returns the fractional part of a decimal number.frac(x) (init-rate or control-rate args; also works at audio rate in Csound5) Extracts an integer from a decimal number.int(x) (init-rate or control-rate; also works at audio rate in Csound5) Returns a natural log.log(x) (no rate restriction) Returns a base 10 log.log10(x) (no rate restriction) Returns a base 2 log.log2(x) (no rate restriction) Performs a logarithmic base two calculation.logbtwo(x) (init-rate or control-rate args only) Performs a power-of-two calculation.powoftwo(x) (init-rate or control-rate args only) Questions whether the argument is a infinite numberqinf(x) (no rate restriction) Questions whether the argument is a infinite numberqnan(x) (no rate restriction) Returns the integer value nearest toround(x) (init-, control-, or audio-rate arg allowed) Returns a square root value.sqrt(x) (no rate restriction) Performs a cosine function.cos(x) (no rate restriction) Performs a hyperbolic cosine function.cosh(x) (no rate restriction) Performs a arccosine function.cosinv(x) (no rate restriction) Performs a signum function.signum(x) (no rate restriction) Performs a sine function.sin(x) (no rate restriction) Performs a hyperbolic sine function.sinh(x) (no rate restriction) Performs an arcsine function.sininv(x) (no rate restriction) Performs a tangent function.tan(x) (no rate restriction) Performs a hyperbolic tangent function.tanh(x) (no rate restriction) Performs an arctangent function.taninv(x) (no rate restriction) Returns the amplitude equivalent of the decibel value x.ampdb(x) (no rate restriction) Returns the amplitude equivalent (in 16-bit signed integer scale) of the full scale decibel (dB FS) valueampdbfs(x) (no rate restriction) Returns the amplitude equivalent for a given decibel amount.db(x) Returns the decibel equivalent of the raw amplitudedbamp(x) (init-rate or control-rate args only) Returns the decibel equivalent of the raw amplitudedbfsamp(x) (init-rate or control-rate args only) Returns a random number in a bi-polar range.birnd(x) (init- or control-rate only) Returns a random number in a unipolar range at the rate given by the input argument.rnd(x) (init- or control-rate only) Safely divides two numbers.ares divz xa, xb, ksubstires divz ia, ib, isubstkres divz ka, kb, ksubst...divz(ka, kb, ksubst)... (no rate restriction) Multiplies and accumulates a- and k-rate signals.ares mac ksig1, asig1 [, ksig2] [, asig2] [, ksig3] [, asig3] [...] Multiply and accumulate a-rate signals only.ares maca asig1 , asig2 [, asig3] [, asig4] [, asig5] [...] Efficiently evaluates a polynomial of arbitrary order.aout polynomial ain, k0 [, k1 [, k2 [...]]] Computes one argument to the power of another argument.ares pow aarg, kpow [, inorm]ires pow iarg, ipow [, inorm]kres pow karg, kpow [, inorm] Multiplies any number of a-rate signals.ares product asig1, asig2 [, asig3] [...] Sums any number of a-rate signals.ares sum asig1 [, asig2] [, asig3] [...] Returns an arctangent.ares taninv2 ay, axires taninv2 iy, ixkres taninv2 ky, kx...taninv2(ky, kx)... (no rate restriction) Calculates a factor to raise/lower a frequency by a given amount of cents.cent(x) Converts a Midi note number value to cycles-per-second.cpsmidinn (MidiNoteNumber) (init- or control-rate args only) Converts an octave-point-decimal value to cycles-per-second.cpsoct (oct) (no rate restriction) Converts a pitch-class value to cycles-per-second.cpspch (pch) (init- or control-rate args only) Calculates a factor to raise/lower a frequency by a given amount of octaves.octave(x) Converts a cycles-per-second value to octave-point-decimal.octcps (cps) (init- or control-rate args only) Converts a Midi note number value to octave-point-decimal.octmidinn (MidiNoteNumber) (init- or control-rate args only) Converts a pitch-class value to octave-point-decimal.octpch (pch) (init- or control-rate args only) Converts a Midi note number value to octave point pitch-class units.pchmidinn (MidiNoteNumber) (init- or control-rate args only) Converts an octave-point-decimal value to pitch-class.pchoct (oct) (init- or control-rate args only) Calculates a factor to raise/lower a frequency by a given amount of semitones.semitone(x) Converts a pitch-class value into cycles-per-second (Hz) for equal divisions of the octave.icps cps2pch ipch, iequal Returns micro-tuning values at k-rate.kcps cpstun ktrig, kindex, kfn Returns micro-tuning values at init-rate.icps cpstuni index, ifn Converts a pitch-class value into cycles-per-second (Hz) for equal divisions of any interval.icps cpsxpch ipch, iequal, irepeat, ibase Get the current after-touch value for this channel.kaft aftouch [imin] [, imax] Get the current value of a MIDI channel controller.ival chanctrl ichnl, ictlno [, ilow] [, ihigh]kval chanctrl ichnl, ictlno [, ilow] [, ihigh] Allows a floating-point 14-bit MIDI signal scaled with a minimum and a maximum range.idest ctrl14 ichan, ictlno1, ictlno2, imin, imax [, ifn]kdest ctrl14 ichan, ictlno1, ictlno2, kmin, kmax [, ifn] Allows a floating-point 21-bit MIDI signal scaled with a minimum and a maximum range.idest ctrl21 ichan, ictlno1, ictlno2, ictlno3, imin, imax [, ifn]kdest ctrl21 ichan, ictlno1, ictlno2, ictlno3, kmin, kmax [, ifn] Allows a floating-point 7-bit MIDI signal scaled with a minimum and a maximum range.idest ctrl7 ichan, ictlno, imin, imax [, ifn]kdest ctrl7 ichan, ictlno, kmin, kmax [, ifn]adest ctrl7 ichan, ictlno, kmin, kmax [, ifn] [, icutoff] Sets the initial values for a set of MIDI controllers.ctrlinit ichnl, ictlno1, ival1 [, ictlno2] [, ival2] [, ictlno3] \ [, ival3] [,...ival32] Initializes the controllers used to create a 14-bit MIDI value.initc14 ichan, ictlno1, ictlno2, ivalue Initializes the controllers used to create a 21-bit MIDI value.initc21 ichan, ictlno1, ictlno2, ictlno3, ivalue Initializes the controller used to create a 7-bit MIDI value.initc7 ichan, ictlno, ivalue Assigns a MIDI channel number to a Csound instrument.massign ichnl, insnum[, ireset]massign ichnl, "insname"[, ireset] Allows a floating-point 14-bit MIDI signal scaled with a minimum and a maximum range.idest midic14 ictlno1, ictlno2, imin, imax [, ifn]kdest midic14 ictlno1, ictlno2, kmin, kmax [, ifn] Allows a floating-point 21-bit MIDI signal scaled with a minimum and a maximum range.idest midic21 ictlno1, ictlno2, ictlno3, imin, imax [, ifn]kdest midic21 ictlno1, ictlno2, ictlno3, kmin, kmax [, ifn] Allows a floating-point 7-bit MIDI signal scaled with a minimum and a maximum range.idest midic7 ictlno, imin, imax [, ifn]kdest midic7 ictlno, kmin, kmax [, ifn] Get the current value (0-127) of a specified MIDI controller.ival midictrl inum [, imin] [, imax]kval midictrl inum [, imin] [, imax] Get a note number from a MIDI event.ival notnum Get the current pitch-bend value for this channel.ibend pchbend [imin] [, imax]kbend pchbend [imin] [, imax] Assigns an instrument number to a specified MIDI program.pgmassign ipgm, inst[, ichn]pgmassign ipgm, "insname"[, ichn] Returns the polyphonic after-touch pressure of the selected note number.ires polyaft inote [, ilow] [, ihigh]kres polyaft inote [, ilow] [, ihigh] Get the velocity from a MIDI event.ival veloc [ilow] [, ihigh] Sends a Non-Registered Parameter Number to the MIDI OUT port.nrpn kchan, kparmnum, kparmvalue Sends MIDI aftertouch messages at i-rate.outiat ichn, ivalue, imin, imax Sends MIDI controller output at i-rate.outic ichn, inum, ivalue, imin, imax Sends 14-bit MIDI controller output at i-rate.outic14 ichn, imsb, ilsb, ivalue, imin, imax Sends polyphonic MIDI aftertouch messages at i-rate.outipat ichn, inotenum, ivalue, imin, imax Sends MIDI pitch-bend messages at i-rate.outipb ichn, ivalue, imin, imax Sends MIDI program change messages at i-rateoutipc ichn, iprog, imin, imax Sends MIDI aftertouch messages at k-rate.outkat kchn, kvalue, kmin, kmax Sends MIDI controller messages at k-rate.outkc kchn, knum, kvalue, kmin, kmax Sends 14-bit MIDI controller output at k-rate.outkc14 kchn, kmsb, klsb, kvalue, kmin, kmax Sends polyphonic MIDI aftertouch messages at k-rate.outkpat kchn, knotenum, kvalue, kmin, kmax Sends MIDI pitch-bend messages at k-rate.outkpb kchn, kvalue, kmin, kmax Sends MIDI program change messages at k-rate.outkpc kchn, kprog, kmin, kmax Get the velocity of the current MIDI event.iamp ampmidi iscal [, ifn] Musically map MIDI velocity to peak amplitude within a specified dynamic range in decibels.iamplitude ampmidid ivelocity, idecibelskamplitude ampmidid kvelocity, idecibels Get the note number of the current MIDI event, expressed in cycles-per-second.icps cpsmidi Get the note number of the current MIDI event and modify it by the current pitch-bend value, express it in cycles-per-second.icps cpsmidib [irange]kcps cpsmidib [irange] Get a MIDI note number (allows customized micro-tuning scales).icps cpstmid ifn Get the note number, in octave-point-decimal units, of the current MIDI event.ioct octmidi Get the note number of the current MIDI event and modify it by the current pitch-bend value, express it in octave-point-decimal.ioct octmidib [irange]koct octmidib [irange] Get the note number of the current MIDI event, expressed in pitch-class units.ipch pchmidi Get the note number of the current MIDI event and modify it by the current pitch-bend value, express it in pitch-class units.ipch pchmidib [irange]kpch pchmidib [irange] Returns a generic MIDI message received by the MIDI IN port.kstatus, kchan, kdata1, kdata2 midiin Sends a generic MIDI message to the MIDI OUT port.midiout kstatus, kchan, kdata1, kdata2 Indicates whether a note is in itskflag release Extend the duration of real-time generated events.xtratim iextradur Generates MIDI note messages at k-rate.midion kchn, knum, kvel Sends noteon and noteoff messages to the MIDI OUT port.midion2 kchn, knum, kvel, ktrig Sends a stream of the MIDI notes.moscil kchn, knum, kvel, kdur, kpause Send a noteoff message to the MIDI OUT port.noteoff ichn, inum, ivel Send a noteon message to the MIDI OUT port.noteon ichn, inum, ivel Sends a noteon and a noteoff MIDI message both with the same channel, number and velocity.noteondur ichn, inum, ivel, idur Sends a noteon and a noteoff MIDI message both with the same channel, number and velocity.noteondur2 ichn, inum, ivel, idur Gets a MIDI channel's aftertouch value.midichannelaftertouch xchannelaftertouch [, ilow] [, ihigh] Returns the MIDI channel number from which the note was activated.ichn midichn Gets a MIDI control change value.midicontrolchange xcontroller, xcontrollervalue [, ilow] [, ihigh] Changes values, depending on MIDI activation.mididefault xdefault, xvalue Gets a MIDI noteoff value.midinoteoff xkey, xvelocity Gets a MIDI note number as a cycles-per-second frequency.midinoteoncps xcps, xvelocity Gets a MIDI note number value.midinoteonkey xkey, xvelocity Gets a MIDI note number value as octave-point-decimal value.midinoteonoct xoct, xvelocity Gets a MIDI note number as a pitch-class value.midinoteonpch xpch, xvelocity Gets a MIDI pitchbend value.midipitchbend xpitchbend [, ilow] [, ihigh] Gets a MIDI polyphonic aftertouch value.midipolyaftertouch xpolyaftertouch, xcontrollervalue [, ilow] [, ihigh] Gets a MIDI program change value.midiprogramchange xprogram Sends a MIDI CLOCK message.mclock ifreq Send system real-time messages to the MIDI OUT port.mrtmsg imsgtype Creates a bank of 16 different 14-bit MIDI control message numbers.i1,...,i16 s16b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, \ initvalue1, ifn1,..., ictlno_msb16, ictlno_lsb16, imin16, imax16, initvalue16, ifn16k1,...,k16 s16b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, \ initvalue1, ifn1,..., ictlno_msb16, ictlno_lsb16, imin16, imax16, initvalue16, ifn16 Creates a bank of 32 different 14-bit MIDI control message numbers.i1,...,i32 s32b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, \ initvalue1, ifn1,..., ictlno_msb32, ictlno_lsb32, imin32, imax32, initvalue32, ifn32k1,...,k32 s32b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, \ initvalue1, ifn1,..., ictlno_msb32, ictlno_lsb32, imin32, imax32, initvalue32, ifn32 Creates a bank of 16 different MIDI control message numbers.i1,...,i16 slider16 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum16, imin16, imax16, init16, ifn16k1,...,k16 slider16 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum16, imin16, imax16, init16, ifn16 Creates a bank of 16 different MIDI control message numbers, filtered before output.k1,...,k16 slider16f ichan, ictlnum1, imin1, imax1, init1, ifn1, \ icutoff1,..., ictlnum16, imin16, imax16, init16, ifn16, icutoff16 Stores a bank of 16 different MIDI control messages to a table.kflag slider16table ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, .... , ictlnum16, imin16, imax16, init16, ifn16 Stores a bank of 16 different MIDI control messages to a table, filtered before output.kflag slider16tablef ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, icutoff1, .... , ictlnum16, imin16, imax16, init16, ifn16, icutoff16 Creates a bank of 32 different MIDI control message numbers.i1,...,i32 slider32 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum32, imin32, imax32, init32, ifn32k1,...,k32 slider32 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum32, imin32, imax32, init32, ifn32 Creates a bank of 32 different MIDI control message numbers, filtered before output.k1,...,k32 slider32f ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1, \ ..., ictlnum32, imin32, imax32, init32, ifn32, icutoff32 Stores a bank of 32 different MIDI control messages to a table.kflag slider32table ichan, ioutTable, ioffset, ictlnum1, imin1, \ imax1, init1, ifn1, .... , ictlnum32, imin32, imax32, init32, ifn32 Stores a bank of 32 different MIDI control messages to a table, filtered before output.kflag slider32tablef ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, icutoff1, .... , ictlnum32, imin32, imax32, init32, ifn32, icutoff32 Creates a bank of 64 different MIDI control message numbers.i1,...,i64 slider64 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum64, imin64, imax64, init64, ifn64k1,...,k64 slider64 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum64, imin64, imax64, init64, ifn64 Creates a bank of 64 different MIDI control message numbers, filtered before output.k1,...,k64 slider64f ichan, ictlnum1, imin1, imax1, init1, ifn1, \ icutoff1,..., ictlnum64, imin64, imax64, init64, ifn64, icutoff64 Stores a bank of 64 different MIDI control messages to a table.kflag slider64table ichan, ioutTable, ioffset, ictlnum1, imin1, \ imax1, init1, ifn1, .... , ictlnum64, imin64, imax64, init64, ifn64 Stores a bank of 64 different MIDI control messages to a table, filtered before output.kflag slider64tablef ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, icutoff1, .... , ictlnum64, imin64, imax64, init64, ifn64, icutoff64 Creates a bank of 8 different MIDI control message numbers.i1,...,i8 slider8 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum8, imin8, imax8, init8, ifn8k1,...,k8 slider8 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum8, imin8, imax8, init8, ifn8 Creates a bank of 8 different MIDI control message numbers, filtered before output.k1,...,k8 slider8f ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1, \ ..., ictlnum8, imin8, imax8, init8, ifn8, icutoff8 Stores a bank of 8 different MIDI control messages to a table.kflag slider8table ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1,..., ictlnum8, imin8, imax8, init8, ifn8 Stores a bank of 8 different MIDI control messages to a table, filtered before output.kflag slider8tablef ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, icutoff1, .... , ictlnum8, imin8, imax8, init8, ifn8, icutoff8 Creates a bank of 16 different MIDI control message numbers from a KAWAI MM-16 midi mixer.k1, k2, ...., k16 sliderKawai imin1, imax1, init1, ifn1, \ imin2, imax2, init2, ifn2, ..., imin16, imax16, init16, ifn16 Activates the indicated instrument in the orchestra header, without need for an i statement. alwayson Tinstrument [p4, ..., pn] Connects a source outlet to a sink inlet.connect Tsource1, Soutlet1, Tsink1, Sinlet1 Generate a function table from within an instrument definition, without duplication of data.ifno ftgenonce ip1, ip2dummy, isize, igen, iarga, iargb, ... Receives an arate signal into an instrument through a named port.asignal inleta Sname Receives an frate signal (fsig) into an instrument from a named port.fsignal inletf Sname Receives a krate signal into an instrument from a named port.ksignal inletk Sname Receives a krate signal into an instrument from a named port.ksignal inletkid Sname, SinstanceID Sends an arate signal out from an instrument to a named port.outleta Sname, asignal Sends a frate signal (fsig) out from an instrument to a named port.outletf Sname, fsignal Sends a krate signal out from an instrument to a named port.outletk Sname, ksignal Sends a krate signal out from an instrument to a named port.outletkid Sname, SinstanceID, ksignal Deprecated.ktableseg ifn1, idur1, ifn2 [, idur2] [, ifn3] [...] Reads from aares pvadd ktimpnt, kfmod, ifilcod, ifn, ibins [, ibinoffset] \ [, ibinincr] [, iextractmode] [, ifreqlim] [, igatefn] Reads from a phase vocoder analysis file and makes the retrieved data available.pvbufread ktimpnt, ifile Applies the amplitudes from one phase vocoder analysis file to the data from a second file.ares pvcross ktimpnt, kfmod, ifile, kampscale1, kampscale2 [, ispecwp] Interpolates between the amplitudes and frequencies of two phase vocoder analysis files.ares pvinterp ktimpnt, kfmod, ifile, kfreqscale1, kfreqscale2, \ kampscale1, kampscale2, kfreqinterp, kampinterp Implements signal reconstruction using an fft-based phase vocoder.ares pvoc ktimpnt, kfmod, ifilcod [, ispecwp] [, iextractmode] \ [, ifreqlim] [, igatefn] Reads from a phase vocoder analysis file and returns the frequency and amplitude from a single analysis channel or bin.kfreq, kamp pvread ktimpnt, ifile, ibin Creates a new function table by making linear segments between values in stored function tables.tableseg ifn1, idur1, ifn2 [, idur2] [, ifn3] [...] Creates a new function table by making exponential segments between values in stored function tables.tablexseg ifn1, idur1, ifn2 [, idur2] [, ifn3] [...] Implements signal reconstruction using an fft-based phase vocoder and an extra envelope.ares vpvoc ktimpnt, kfmod, ifile [, ispecwp] [, ifn] Resynthesises a signal from the data passed internally by a previous lpread, applying formant shifting.ares lpfreson asig, kfrqratio Computes a new set of poles from the interpolation between two analysis.lpinterp islot1, islot2, kmix Reads a control file of time-ordered information frames.krmsr, krmso, kerr, kcps lpread ktimpnt, ifilcod [, inpoles] [, ifrmrate] Resynthesises a signal from the data passed internally by a previous lpread.ares lpreson asig Selects the slot to be use by further lp opcodes.lpslot islot Perform a weighted add of two input spectra.wsig specaddm wsig1, wsig2 [, imul2] Finds the positive difference values between consecutive spectral frames.wsig specdiff wsigin Displays the magnitude values of the spectrum.specdisp wsig, iprd [, iwtflg] Filters each channel of an input spectrum.wsig specfilt wsigin, ifhtim Accumulates the values of successive spectral frames.wsig spechist wsigin Estimates the pitch of the most prominent complex tone in the spectrum.koct, kamp specptrk wsig, kvar, ilo, ihi, istr, idbthresh, inptls, \ irolloff [, iodd] [, iconfs] [, interp] [, ifprd] [, iwtflg] Scales an input spectral datablock with spectral envelopes.wsig specscal wsigin, ifscale, ifthresh Sums the magnitudes across all channels of the spectrum.ksum specsum wsig [, interp] Generate a constant-Q, exponentially-spaced DFT.wsig spectrum xsig, iprd, iocts, ifrqa [, iq] [, ihann] [, idbout] \ [, idsprd] [, idsinrs] PVS tracks to amplitude+frequency conversion.fsig binit fin, isize Partial track spectral analysis.ftrks partials ffr, fphs, kthresh, kminpts, kmaxgap, imaxtracks Copies spectral data to t-variables.kframe pvs2tab tvar, fsig Resynthesize using a fast oscillator-bank.ares pvsadsyn fsrc, inoscs, kfmod [, ibinoffset] [, ibinincr] [, iinit] Generate an fsig from a mono audio source ain, using phase vocoder overlap-add analysis.fsig pvsanal ain, ifftsize, ioverlap, iwinsize, iwintype [, iformat] [, iinit] Arpeggiate the spectral components of a streaming pv signal.fsig pvsarp fsigin, kbin, kdepth, kgain A band pass filter working in the spectral domain.fsig pvsbandp fsigin, xlowcut, xlowfull, \ xhighfull, xhighcut[, ktype] A band reject filter working in the spectral domain.fsig pvsbandr fsigin, xlowcut, xlowfull, \ xhighfull, xhighcut[, ktype] Obtain the amp and freq values off a PVS signal bin.kamp, kfr pvsbin fsig, kbin Average the amp/freq time functions of each analysis channel for a specified time.fsig pvsblur fsigin, kblurtime, imaxdel This opcode creates and writes to a circular buffer for f-signals (streaming PV signals).ihandle, ktime pvsbuffer fsig, ilen This opcode reads a circular buffer of f-signals (streaming PV signals).fsig pvsbufread ktime, khandle[, ilo, ihi, iclear] This opcode reads a circular buffer of f-signals (streaming PV signals), with binwise additional delays.fsig pvsbufread2 ktime, khandle, ift1, ift2 Scale the frequency components of a pv stream.fsig pvscale fsigin, kscal[, kkeepform, kgain, kcoefs] Calculate the spectral centroid of a signal.kcent pvscent fsig Performs cross-synthesis between two source fsigs.fsig pvscross fsrc, fdest, kamp1, kamp2 Spectral azimuth-based de-mixing of stereo sources.fsig pvsdemix fleft, fright, kpos, kwidth, ipoints Read a selected channel from a PVOC-EX analysis file.fsig pvsdiskin SFname,ktscal,kgain[,ioffset, ichan] Displays a PVS signal as an amplitude vs. freq graph.pvsdisp fsig[, ibins, iwtflg] Multiply amplitudes of a pvoc stream by those of a second pvoc stream, with dynamic scaling.fsig pvsfilter fsigin, fsigfil, kdepth[, igain] Read a selected channel from a PVOC-EX analysis file.fsig pvsfread ktimpt, ifn [, ichan] Freeze the amplitude and frequency time functions of a pv stream according to a control-rate trigger.fsig pvsfreeze fsigin, kfreeza, kfreezf Reads amplitude and/or frequency data from function tables.pvsftr fsrc, ifna [, ifnf] Writes amplitude and/or frequency data to function tables.kflag pvsftw fsrc, ifna [, ifnf] Write a fsig to a PVOCEX file.pvsfwrite fsig, ifile Scale the amplitude of a pv stream.fsig pvsgain fsigin, kgain Shift the frequency components of a pv stream, stretching/compressing its spectrum.fsig pvshift fsigin, kshift, klowest[, kkeepform, igain, kcoefs] Instantaneous Frequency Distribution, magnitude and phase analysis.ffr,fphs pvsifd ain, ifftsize, ihopsize, iwintype[,iscal] Retrieve an fsig from the input software bus; a pvs equivalent to chani.fsig pvsin kchan[, isize, iolap, iwinsize, iwintype, iformat] Get information from a PVOC-EX formatted source.ioverlap, inumbins, iwinsize, iformat pvsinfo fsrc Initialise a spectral (f) variable to zero.fsig pvsinit isize[, iolap, iwinsize, iwintype, iformat] Frequency lock an input fsigfsig pvslock fsigin, klock Modify amplitudes using a function table, with dynamic scaling.fsig pvsmaska fsrc, ifn, kdepth Mix 'seamlessly' two pv signals.fsig pvsmix fsigin1, fsigin2 Smooth the amplitude and frequency time functions of a pv stream using parallel 1st order lowpass IIR filters with time-varying cutoff frequency.fsig pvsmooth fsigin, kacf, kfcf Performs morphing (or interpolation) between two source fsigs.fsig pvsmorph fsig1, fsig2, kampint, kfrqint PVS-based oscillator simulator.fsig pvsosc kamp, kfreq, ktype, isize [,ioverlap] [, iwinsize] [, iwintype] [, iformat] Write a fsig to the pvs output bus.pvsout fsig, kchan Track the pitch and amplitude of a PVS signal.kfr, kamp pvspitch fsig, kthresh Phase vocoder analysis processing with onset detection/processing.fsig pvstanal ktimescal, kamp, kpitch, ktab, [kdetect, kwrap, ioffset,ifftsize, ihop, idbthresh] Transforms a pvoc stream according to a masking function table.fsig pvstencil fsigin, kgain, klevel, iftable Combine the spectral envelope of one fsig with the excitation (frequencies) of another.fsig pvsvoc famp, fexc, kdepth, kgain [,kcoefs] Warp the spectral envelope of a PVS signalfsig pvswarp fsigin, kscal, kshift[, klowest, kmeth, kgain, kcoefs] Resynthesise using a FFT overlap-add.ares pvsynth fsrc, [iinit] Streaming partial track additive synthesis with cubic phase interpolation with pitch control and support for timescale-modified inputasig resyn fin, kscal, kpitch, kmaxtracks, ifn Streaming partial track additive synthesis with cubic phase interpolationasig sinsyn fin, kscal, kmaxtracks, ifn Copies spectral data from t-variables.fsig tab2pvs tvar[,ihopsize, iwinsize, iwintype] Streaming partial track additive synthesisasig tradsyn fin, kscal, kpitch, kmaxtracks, ifn Streaming partial track cross-synthesis.fsig trcross fin1, fin2, ksearch, kdepth [, kmode] Streaming partial track filtering.fsig trfilter fin, kamnt, ifn Extracts the highest-frequency track from a streaming track input signal.fsig, kfr, kamp trhighest fin1, kscal Extracts the lowest-frequency track from a streaming track input signal.fsig, kfr, kamp trlowest fin1, kscal Streaming partial track mixing.fsig trmix fin1, fin2 Streaming partial track frequency scaling.fsig trscale fin, kpitch[, kgain] Streaming partial track frequency scaling.fsig trshift fin, kpshift[, kgain] Streaming partial track frequency splitting.fsiglow, fsighi trsplit fin, ksplit[, kgainlow, kgainhigh] uses the data from an ATS analysis file to perform additive synthesis.ar ATSadd ktimepnt, kfmod, iatsfile, ifn, ipartials[, ipartialoffset, \ ipartialincr, igatefn] uses the data from an ATS analysis file to perform noise resynthesis.ar ATSaddnz ktimepnt, iatsfile, ibands[, ibandoffset, ibandincr] reads data from and ATS data file and stores it in an internal data table of frequency, amplitude pairs.ATSbufread ktimepnt, kfmod, iatsfile, ipartials[, ipartialoffset, \ ipartialincr] perform cross synthesis from ATS analysis files.ar ATScross ktimepnt, kfmod, iatsfile, ifn, kmylev, kbuflev, ipartials \ [, ipartialoffset, ipartialincr] reads data out of the header of an ATS file.idata ATSinfo iatsfile, ilocation allows a user to determine the frequency envelope of anykamp ATSinterpread kfreq returns a frequency, amplitude pair from ankfrq, kamp ATSpartialtap ipartialnum reads data from an ATS file.kfreq, kamp ATSread ktimepnt, iatsfile, ipartial reads data from an ATS file.kenergy ATSreadnz ktimepnt, iatsfile, iband uses the data from an ATS analysis file to perform resynthesis.ar ATSsinnoi ktimepnt, ksinlev, knzlev, kfmod, iatsfile, ipartials \ [, ipartialoffset, ipartialincr] Morphs two stored sets of bandwidth-enhanced partials and stores a new set of partials representing the morphed sound. The morph is performed by linearly interpolating the parameter envelopes (frequency, amplitude, and bandwidth, or noisiness) of the bandwidth-enhanced partials according to control-rate frequency, amplitude, and bandwidth morphing functions.lorismorph isrcidx, itgtidx, istoreidx, kfreqmorphenv, kampmorphenv, kbwmorphenv renders a stored set of bandwidth-enhanced partials using the method of Bandwidth-Enhanced Additive Synthesis implemented in the Loris software, applying control-rate frequency, amplitude, and bandwidth scaling envelopes.ar lorisplay ireadidx, kfreqenv, kampenv, kbwenv Imports a set of bandwidth-enhanced partials from a SDIF-format data file, applying control-rate frequency, amplitude, and bandwidth scaling envelopes, and stores the modified partials in memory.lorisread ktimpnt, ifilcod, istoreidx, kfreqenv, kampenv, kbwenv[, ifadetime] Calculate the spectral centroid of a signal.kcent centroid asig, ktrig, ifftsize Phase-locked vocoder processing.asig mincer atimpt, kamp, kpitch, ktab, klock[,ifftsize,idecim] Phase-locked vocoder processing with onset detection/processing, 'tempo-scaling'.asig temposcal ktimescal, kamp, kpitch, ktab, klock [,ifftsize, idecim, ithresh] Set string variable to value from strset table or string p-fieldSdst strget indx Allows a string to be linked with a numeric value.strset iarg, istring Print a string constant or variableputs Sstr, ktrig[, inonl] printf-style formatted output to a string variable.Sdst sprintf Sfmt, xarg1[, xarg2[, ... ]] printf-style formatted output to a string variable at k-rate.Sdst sprintfk Sfmt, xarg1[, xarg2[, ... ]] Concatenate stringsSdst strcat Ssrc1, Ssrc2 Concatenate strings (k-rate)Sdst strcatk Ssrc1, Ssrc2 Compare stringsires strcmp S1, S2 Compare stringskres strcmpk S1, S2 Assign value to a string variableSdst strcpy SsrcSdst = Ssrc Assign value to a string variable (k-rate)Sdst strcpyk Ssrc Return the position of the first occurence of a string in another stringipos strindex S1, S2 Return the position of the first occurence of a string in another stringkpos strindexk S1, S2 Return the length of a stringilen strlen Sstr Return the length of a stringklen strlenk Sstr Return the position of the last occurence of a string in another stringipos strrindex S1, S2 Return the position of the last occurence of a string in another stringkpos strrindexk S1, S2 Extract a substringSdst strsub Ssrc[, istart[, iend]] Extract a substringSdst strsubk Ssrc, kstart, kend Return the ASCII code of a character in a stringichr strchar Sstr[, ipos] Return the ASCII code of a character in a stringkchr strchark Sstr[, kpos] Convert a string to lower caseSdst strlower Ssrc Convert a string to lower caseSdst strlowerk Ssrc Converts a string to a float (i-rate).ir strtod Sstrir strtod indx Converts a string to a float (k-rate).kr strtodk Sstrkr strtodk kndx Converts a string to a signed integer (i-rate).ir strtol Sstrir strtol indx Converts a string to a signed integer (k-rate).kr strtolk Sstrkr strtolk kndx Convert a string to upper caseSdst strupper Ssrc Convert a string to upper caseSdst strupperk Ssrc Read vectors (from tables -or arrays of vectors).vtaba andx, ifn, aout1 [, aout2, aout3, .... , aoutN ] Read vectors (from tables -or arrays of vectors).vtabi indx, ifn, iout1 [, iout2, iout3, .... , ioutN ] Read vectors (from tables -or arrays of vectors).vtabk kndx, ifn, kout1 [, kout2, kout3, .... , koutN ] Read a vector (several scalars simultaneously) from a table.vtable1k kfn,kout1 [, kout2, kout3, .... , koutN ] Read vectors (from tables -or arrays of vectors).vtablea andx, kfn, kinterp, ixmode, aout1 [, aout2, aout3, .... , aoutN ] Read vectors (from tables -or arrays of vectors).vtablei indx, ifn, interp, ixmode, iout1 [, iout2, iout3, .... , ioutN ] Read vectors (from tables -or arrays of vectors).vtablek kndx, kfn, kinterp, ixmode, kout1 [, kout2, kout3, .... , koutN ] Write vectors (to tables -or arrays of vectors).vtablewa andx, kfn, ixmode, ainarg1 [, ainarg2, ainarg3 , .... , ainargN ] Write vectors (to tables -or arrays of vectors).vtablewi indx, ifn, ixmode, inarg1 [, inarg2, inarg3 , .... , inargN ] Write vectors (to tables -or arrays of vectors).vtablewk kndx, kfn, ixmode, kinarg1 [, kinarg2, kinarg3 , .... , kinargN ] Write vectors (to tables -or arrays of vectors).vtabwa andx, ifn, ainarg1 [, ainarg2, ainarg3 , .... , ainargN ] Write vectors (to tables -or arrays of vectors).vtabwi indx, ifn, inarg1 [, inarg2, inarg3 , .... , inargN ] Write vectors (to tables -or arrays of vectors).vtabwk kndx, ifn, kinarg1 [, kinarg2, kinarg3 , .... , kinargN ] Adds a scalar value to a vector in a table.vadd ifn, kval, kelements [, kdstoffset] [, kverbose] Adds a scalar value to a vector in a table.vadd_i ifn, ival, ielements [, idstoffset] Performs power-of operations between a vector and a scalarvexp ifn, kval, kelements [, kdstoffset] [, kverbose] Performs power-of operations between a vector and a scalarvexp_i ifn, ival, ielements[, idstoffset] Multiplies a vector in a table by a scalar value.vmult ifn, kval, kelements [, kdstoffset] [, kverbose] Multiplies a vector in a table by a scalar value.vmult_i ifn, ival, ielements [, idstoffset] Raises each element of a vector to a scalar power.vpow ifn, kval, kelements [, kdstoffset] [, kverbose] Raises each element of a vector to a scalar powervpow_i ifn, ival, ielements [, idstoffset] Performs addition between two vectorial control signals.vaddv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] Performs addition between two vectorial control signals at init time.vaddv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Copies between two vectorial control signalsvcopy ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [, kverbose] Copies a vector from one table to another.vcopy_i ifn1, ifn2, ielements [,idstoffset, isrcoffset] Performs division between two vectorial control signalsvdivv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] Performs division between two vectorial control signals at init time.vdivv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Performs exponential operations between two vectorial control signalsvexpv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] Performs exponential operations between two vectorial control signals at init time.vexpv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Maps elements from a vector according to indexes contained in another vector.vmap ifn1, ifn2, ielements [,idstoffset, isrcoffset] Performs mutiplication between two vectorial control signalsvmultv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] Performs mutiplication between two vectorial control signals at init time.vmultv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Performs power-of operations between two vectorial control signalsvpowv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] Performs power-of operations between two vectorial control signals at init time.vpowv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Performs subtraction between two vectorial control signalsvsubv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] Performs subtraction between two vectorial control signals at init time.vsubv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Vectorial envelope generatorvexpseg ifnout, ielements, ifn1, idur1, ifn2 [, idur2, ifn3 [...]] Vectorial envelope generatorvlinseg ifnout, ielements, ifn1, idur1, ifn2 [, idur2, ifn3 [...]] Limiting and Wrapping Vectorial Signalsvlimit ifn, kmin, kmax, ielements Limiting and Wrapping Vectorial Signalsvmirror ifn, kmin, kmax, ielements Limiting and Wrapping Vectorial Signalsvwrap ifn, kmin, kmax, ielements k-rate variable time delay.kout vdelayk ksig, kdel, imaxdel [, iskip, imode] Vectorial Control-rate Delay Pathsvecdelay ifn, ifnIn, ifnDel, ielements, imaxdel [, iskip] Vectorial Control-rate Delay Pathsvport ifn, khtime, ielements [, ifnInit] Generates a vector of random numbers stored into a table, holding the values for a period of time.vrandh ifn, krange, kcps, ielements [, idstoffset] [, iseed] \ [, isize] [, ioffset] Generate a sort of 'vectorial band-limited noise'vrandi ifn, krange, kcps, ielements [, idstoffset] [, iseed] \ [, isize] [, ioffset] Cellular Automatoncell ktrig, kreinit, ioutFunc, initStateFunc, iRuleFunc, ielements Cellular Automatavcella ktrig, kreinit, ioutFunc, initStateFunc, \ iRuleFunc, ielements, irulelen [, iradius] Clears one or more variables in the za space.zacl kfirst, klast Establishes zak space.zakinit isizea, isizek Modulates one a-rate signal by a second one.ares zamod asig, kzamod Reads from a location in za space at a-rate.ares zar kndx Reads from a location in za space at a-rate, adds some gain.ares zarg kndx, kgain Writes to a za variable at a-rate without mixing.zaw asig, kndx Writes to a za variable at a-rate with mixing.zawm asig, kndx [, imix] Reads from a location in zk space at i-rate.ir zir indx Writes to a zk variable at i-rate without mixing.ziw isig, indx Writes to a zk variable to an i-rate variable with mixing.ziwm isig, indx [, imix] Clears one or more variables in the zk space.zkcl kfirst, klast Facilitates the modulation of one signal by another.kres zkmod ksig, kzkmod Reads from a location in zk space at k-rate.kres zkr kndx Writes to a zk variable at k-rate without mixing.zkw ksig, kndx Writes to a zk variable at k-rate with mixing.zkwm ksig, kndx [, imix] Activates or deactivates a DSSI or LADSPA plugin.dssiactivate ihandle, ktoggle Processes audio using a LADSPA or DSSI plugin.[aout1, aout2, ..., aout9] dssiaudio ihandle, [ain1, ain2, ..., ain9] Send control information to a LADSPA or DSSI plugin.dssictls ihandle, iport, kvalue, ktrigger Loads a DSSI or LADSPA plugin.ihandle dssiinit ilibraryname, iplugindex [, iverbose] Lists all available DSSI and LADSPA plugins.dssilist VST audio output.aout1,aout2 vstaudio instance, [ain1, ain2]aout1,aout2 vstaudiog instance, [ain1, ain2] Loads parameter banks to a VST plugin.vstbankload instance, ipath Opens the GUI editor window for a VST plugin.vstedit instance Displays the parameters and the programs of a VST plugin.vstinfo instance Load a VST plugin into memory for use with the other vst4cs opcodes.instance vstinit ilibrarypath [,iverbose] Sends MIDI information to a VST plugin.vstmidiout instance, kstatus, kchan, kdata1, kdata2 Sends a MIDI note with definite duration to a VST plugin.vstnote instance, kchan, knote, kveloc, kdur Used for parameter comunication to and from a VST plugin.vstparamset instance, kparam, kvaluekvalue vstparamget instance, kparam Loads parameter banks to a VST plugin.vstprogset instance, kprogram Start a listening process for OSC messages to a particular port.ihandle OSCinit iport Listen for OSC messages to a particular path.kans OSClisten ihandle, idest, itype [, xdata1, xdata2, ...] Sends data to other processes using the OSC protocolOSCsend kwhen, ihost, iport, idestination, itype [, kdata1, kdata2, ...] Instantiates and runs a compiled Faust program.ihandle,a1[,a2,...] faustaudio ifac[,ain1,...] Invokes the just-in-time compiler to produce a instantiable DSP process from a Faust program.ihandle faustcompile Scode, Sargs Adjusts a given control in a Faust DSP instance.faustctl idsp,Scontrol,kval Compiles, Instantiates and runs a compiled Faust program.ihandle,a1[,a2,...] faustgen SCode[,ain1,...] Defines the port for use with the remote system.remoteport iportnum Receives data from other processes using the low-level UDP or TCP protocolsasig sockrecv iport, ilengthasigl, asigr sockrecvs iport, ilengthasig strecv Sipaddr, iport Sends data to other processes using the low-level UDP or TCP protocolssocksend asig, Sipaddr, iport, ilengthsocksends asigl, asigr, Sipaddr, iport, ilengthstsend asig, Sipaddr, iport An opcode which can be used to implement a remote orchestra. This opcode will send note events from a source machine to many destinations.insglobal isource, instrnum [,instrnum...] An opcode which can be used to implement a remote orchestra. This opcode will send note events from a source machine to one destination.insremot idestination, isource, instrnum [,instrnum...] An opcode which can be used to implement a remote midi orchestra. This opcode will broadcast the midi events to all the machines involved in the remote concert.midglobal isource, instrnum [,instrnum...] An opcode which can be used to implement a remote midi orchestra. This opcode will send midi events from a source machine to one destination.midremot idestination, isource, instrnum [,instrnum...] Resets all channels of a buss to 0.MixerClear Gets the level of a send to a buss.kgain MixerGetLevel isend, ibuss Receives an arate signal from a channel of a buss.asignal MixerReceive ibuss, ichannel Mixes an arate signal into a channel of a buss.MixerSend asignal, isend, ibuss, ichannel Sets the level of a send to a buss.MixerSetLevel isend, ibuss, kgain Sets the level of a send to a buss.MixerSetLevel_i isend, ibuss, igain Assign the value of the given Csound variable to a Python variable possibly destroying its previous content.pyassign "variable", kvaluepyassigni "variable", ivaluepylassign "variable", kvaluepylassigni "variable", ivaluepyassignt ktrigger, "variable", kvaluepylassignt ktrigger, "variable", kvalue Invoke the specified Python callable at k-time and i-time (i suffix), passing the given arguments. The call is perfomed in the global environment, and the result (the returning value) is copied into the Csound output variables specified. pycall "callable", karg1, ...kresult pycall1 "callable", karg1, ...kresult1, kresult2 pycall2 "callable", karg1, ...kr1, kr2, kr3 pycall3 "callable", karg1, ...kr1, kr2, kr3, kr4 pycall4 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5 pycall5 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6 pycall6 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7 pycall7 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7, kr8 pycall8 "callable", karg1, ... pycallt ktrigger, "callable", karg1, ...kresult pycall1t ktrigger, "callable", karg1, ...kresult1, kresult2 pycall2t ktrigger, "callable", karg1, ...kr1, kr2, kr3 pycall3t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4 pycall4t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5 pycall5t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6 pycall6t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7 pycall7t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7, kr8 pycall8t ktrigger, "callable", karg1, ... pycalli "callable", karg1, ...iresult pycall1i "callable", iarg1, ...iresult1, iresult2 pycall2i "callable", iarg1, ...ir1, ir2, ir3 pycall3i "callable", iarg1, ...ir1, ir2, ir3, ir4 pycall4i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5 pycall5i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5, ir6 pycall6i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5, ir6, ir7 pycall7i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5, ir6, ir7, ir8 pycall8i "callable", iarg1, ...pycalln "callable", nresults, kresult1, ..., kresultn, karg1, ...pycallni "callable", nresults, iresult1, ..., iresultn, iarg1, ... pylcall "callable", karg1, ...kresult pylcall1 "callable", karg1, ...kresult1, kresult2 pylcall2 "callable", karg1, ...kr1, kr2, kr3 pylcall3 "callable", karg1, ...kr1, kr2, kr3, kr4 pylcall4 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5 pylcall5 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6 pylcall6 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7 pylcall7 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7, kr8 pylcall8 "callable", karg1, ... pylcallt ktrigger, "callable", karg1, ...kresult pylcall1t ktrigger, "callable", karg1, ...kresult1, kresult2 pylcall2t ktrigger, "callable", karg1, ...kr1, kr2, kr3 pylcall3t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4 pylcall4t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5 pylcall5t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6 pylcall6t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7 pylcall7t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7, kr8 pylcall8t ktrigger, "callable", karg1, ... pylcalli "callable", karg1, ...iresult pylcall1i "callable", iarg1, ...iresult1, iresult2 pylcall2i "callable", iarg1, ...ir1, ir2, ir3 pylcall3i "callable", iarg1, ...ir1, ir2, ir3, ir4 pylcall4i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5 pylcall5i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5, ir6 pylcall6i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5, ir6, ir7 pylcall7i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5, ir6, ir7, ir8 pylcall8i "callable", iarg1, ...pylcalln "callable", nresults, kresult1, ..., kresultn, karg1, ...pylcallni "callable", nresults, iresult1, ..., iresultn, iarg1, ... Evaluate a generic Python expression and store the result in a Csound variable at k-time or i-time (i suffix).kresult pyeval "expression"iresult pyevali "expression"kresult pyleval "expression"iresult pylevali "expression"kresult pyevalt ktrigger, "expression"kresult pylevalt ktrigger, "expression" Execute a script from a file at k-time or i-time (i suffix).pyexec "filename"pyexeci "filename"pylexec "filename"pylexeci "filename"pyexect ktrigger, "filename"plyexect ktrigger, "filename" Initialize the Python interpreter.pyinit Run a Python statement or block of statements.pyrun "statement"pyruni "statement"pylrun "statement"pylruni "statement"pyrunt ktrigger, "statement"pylrunt ktrigger, "statement" Create an empty image of a given size.iimagenum imagecreate iwidth, iheight Frees memory allocated for a previously loaded or created image.imagefree iimagenum Return the RGB pixel values of a previously opened or created image.ared, agreen, ablue imagegetpixel iimagenum, ax, aykred, kgreen, kblue imagegetpixel iimagenum, kx, ky Load an image.iimagenum imageload filename Save a previously created image.imagesave iimagenum, filename Set the RGB value of a pixel inside a previously opened or created image.imagesetpixel iimagenum, ax, ay, ared, agreen, ablueimagesetpixel iimagenum, kx, ky, kred, kgreen, kblue Return the width and height of a previously opened or created image.iwidth, iheight imagesize iimagenum Copy data from a vector to an f-table.copya2ftab tab, kftbl Copy data from an f-table to a vector.copyf2array tab, kftbl returns the maximum value in an array.kmax maxarray karray returns the minimum value in an array.kmin minarray karray Scales the values in a range of a vector (one dimensional array).scalearray tarray, kmin, kmax[, kleft, kright] returns the sum of the elements in an array.ksum sumarray karray returns the length of a Farey Sequence.kfl fareylen kfn returns the length of a Farey Sequence.ifl fareyleni ifn Modulation matrix opcode with optimizations for sparse matrices.modmatrix iresfn, isrcmodfn, isrcparmfn, imodscale, inum_mod, \\ inum_parm, kupdate Asks the underlying operating system for the current directory name as a string.Sres pwd Call an external program via the system callires system_i itrig, Scmd, [inowait]kres system ktrig, Scmd, [knowait] shuffles the content of a function table so that each element of the source table is put into a different random position.tableshuffle ktablenumtableshufflei itablenum Performs ATS analysis on a soundfile.csound -U atsa [flags] infilename outfilename Starts Csound with a set of options that can be controlled by environment variables, and input and output files determined by the specified filename stem.cs [-OPTIONS] <name> [CSOUND OPTIONS ... ] Converts a binary file to a Base64 encoded text file.csb64enc [OPTIONS ... ] infile1 [ infile2 [ ... ]] Converts a soundfile into a single Fourier transform frame.csound -U cvanal [flags] infilename outfilenamecvanal [flags] infilename outfilename Reduces noise in a file.dnoise [flags] -i noise_ref_file -o output_soundfile input_soundfile Extracts the envelope of a file to a text file.envext [-flags] soundfilecsound -U envext [-flags] soundfile Extract a section of audio from an audio file.extractor [OPTIONS ... ] infile Converts a .het file to a comma separated text file.het_export het_file cstext_filecsound -U het_export het_file cstext_file Converts a comma-separated text file to a .het file.het_import cstext_file het_filecsound -U het_import cstext_file het_file Decomposes an input soundfile into component sinusoids.csound -U hetro [flags] infilename outfilenamehetro [flags] infilename outfilename Performs both linear predictive and pitch-tracking analysis on a soundfile.csound -U lpanal [flags] infilename outfilenamelpanal [flags] infilename outfilename Creates a CSD file from the specified input files.makecsd [OPTIONS ... ] infile1 [ infile2 [ ... ]] Mixes together a number of soundfiles.mixer [OPTIONS ... ] infile [[OPTIONS... ] infile] ... Outputs a catalog of opcodes and plugin modules.mkdb [-m] [base_directory] Converts a .pvx file to a comma separated text file.pv_export pv_file cstext_filecsound -U pv_export pv_file cstext_file Converts a comma-separated text file to a .pvx file.pv_import cstext_file pv_filecsound -U pv_import cstext_file pv_file Converts a soundfile into a series of short-time Fourier transform frames.csound -U pvanal [flags] infilename outfilenamepvanal [flags] infilename outfilename View formatted text output of STFT analysis files.csound -U pvlook [flags] infilenamepvlook [flags] infilename Scale the amplitude of a sound file.scale [OPTIONS ... ] infile Converts SDIF files to files usable by adsyn.sdif2ad [flags] infilename outfilename Displays information about a soundfile.csound -U sndinfo [options] soundfilenames ...sndinfo [options] soundfilenames ... Converts the sample rate of an audio file.srconv [flags] infile manual_src~dfsg/csound.css0000644000000000000000000001172612262561504014761 0ustar rootroot/* csound.css - for Csound5 html manual */ /* (c) Francois Pinot, 2007 */ /* (c) Andres Cabrera, 2008 */ /* Free software, licensed by GNU Free Documentation License */ pre.programlisting { background-color: #e5e5e5; padding: 10px 2em; border-style: dotted; border-width: 1px; border-color: #663333; } pre.screen { background-color: #f0f0f0; padding: 10px 2em; } div.literallayout { font-family: monospace; padding-left: 1.5cm; } span.strong { font-weight: 700; } div.table table, div.informaltable table { border-collapse: collapse; } div.table thead { border-bottom: solid; } div.table td, div.table th, div.informaltable table td { /* font-size: 150% */ padding: 2px 6px; } div.chapter pre { margin-bottom: 0; margin-top: 0; } div.chapter pre.synopsis { background-color: #e0e0e0; padding: 0px 2em; } div.chapter a { color: blue; font-weight: bolder; 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*/ color: #666633; font-size: large; } h2 { font-family: sans-serif; /* padding-left: 1cm; */ /* font-size: x-small; */ color: #996633; } h2.subtitle { font-family: sans-serif; padding-left: 1.5cm; color: #336699; font-size: x-large; } h3.author { font-family: sans-serif; padding-left: 2cm; font-size: medium; color: #663333; } h3 { font-family: sans-serif; /* padding-left: 1cm; */ /* font-size: x-small; */ color: #996633; } h3.editor { font-family: sans-serif; padding-left: 2cm; font-size: x-small; color: #663333; } h4 { font-family: sans-serif; /* padding-left: 1cm; */ /* font-size: x-small; */ color: #996633; } span.orgname { font-family: serif; padding-left: 2cm; font-size: medium; } div.mediaobject div.caption { font-style: italic; } p { text-align: justify; /* Most text */ /* font-size: 11px; */ /* padding-left: 1cm; */ } h2 span.refentrytitle { font-family: sans-serif; /* padding-left: 0.5cm; */ color: #666633; font-size: xx-large; } /*dt span.refentrytitle { font-family: sans-serif; 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padding: 10px 2em; } /* Syntax highlighting rules */ pre.programlisting span.csdtag { color: brown; } pre.programlisting span.comment { color: green; font-style: italic; } pre.programlisting span.Ccomment { color: peru; font-style: italic; } pre.programlisting span.ohdr { color: darkred; font-weight: bold; } pre.programlisting span.oblock { color: purple; font-weight: bold; } pre.programlisting span.opc { color: blue; font-weight: bold; } pre.programlisting span.op { font-weight: bolder; } pre.programlisting span.octrl { color: green; font-weight: bold; } pre.programlisting span.olabel { color: indianred; font-weight: bold; } pre.programlisting span.omacro { color: mediumorchid; font-weight: bold; } pre.programlisting span.stamnt { color: blue; font-weight: bold; } manual_src~dfsg/overview/0000755000000000000000000000000012262561504014613 5ustar rootrootmanual_src~dfsg/overview/functional.xml0000644000000000000000000000415412262561504017503 0ustar rootroot
Function Syntax in Csound6 Csound 6 has introduced a new alternative syntax for orchestra code. This is initially an experimental feature, which has some limitations, as explained below. It will also allow the introduction of some non-backwards-compatible language features. Overview The main aspect of the new syntax is that some opcodes can be called as functions, and inlined in orchestra code. The general form of this is the expression: var* = op(exprlist*) where * indicates optional, var is a single variable in one of Csound6 types, and exprlist is a comma separated list of expressions (or a single expression or variable). These expressions can be placed anywhere inside instrument or opcode blocks. I-time operations can also be placed outside instrument blocks. The functional syntax can be intermixed with standard Csound code. Here are some examples of these expressions: a1 = oscil(p4,p5) out(vco2(p4*linen(1,0.1,p3,0.1),p5) outs(oscili(in(),p5), in()) Limitations The main limitation is that only opcodes with single outputs (or no outputs) are allowed. In addition, opcodes with multiple optional outputs will not be parsed successfully in this form. An alternative to allow for these is to wrap them in user-defined opcodes, or just to intermix standard Csound syntax with this new style. To resolve opcode ambiguities, we have introduced type annotations, in the form of op:type(exprlist). For instance the code: a1 = oscili(oscili:k(p4,p5), 440) will choose a control-rate opcode to modulate the amplitude of the audio carrier, rather than an audio rate one. There will be cases where the type annotation will be required, when the input arguments cannot be used to determine the correct type of opcode to be applied.
manual_src~dfsg/overview/frontends.xml0000644000000000000000000003135512262561504017346 0ustar rootroot Front Ends Front ends are programs that provide some form of user interface for Csound. Within these programs, Csound is used to generate sound, and familiarity with Csound code is required in order to use them. Front ends typically add helpful features, such as syntax coloring, graphic widgets, or tools for algorithmic score generation, that are not part of Csound itself. Most of these programs were created by a single person, so some of them are not being maintained. Below is a list (certainly not complete, and perhaps not up to date) of front ends available for Csound. Most often, you'll want to download and install Csound itself before downloading and installing a front end. Some front ends require particular versions of Csound, so if you plan to use a front end, it's recommended that you verify its compatibility before installing Csound. CsoundQt CsoundQt is a versatile, cross-platform GUI (graphical user interface) which is bundled with the standard Csound distribution. Created and maintained by Andres Cabrera, QuteCsound provides a multi-tabbed editor, graphic widgets for real-time sound control, and an opcode help system that links to this manual. At this writing (2013) CsoundQt is in active development, so the version installed in your system when you install Csound may not be the most current. The most recent version can be found at http://qutecsound.sourceforge.net/. Blue A cross-platform composition-oriented front end written by Steven Yi in Java. The user interface provides a timeline structured somewhat like a digital multitrack, but differs in that timelines can be embedded within timelines (polyObjects). This allows for a compositional organization in time that many users will find intuitive, informative, and flexible. Each instrument and score section in a blue project has its own editing window, which makes organizing large projects easier. Blue can be downloaded at Blue Home Page. WinXound A convenient front-end for Windows with syntax highlighting. You can get it at the WinXsound Front Page. Winsound Winsound was formerly part of the main Csound tree. It is now available only as source code. Winsound is a cross-platform FLTK port of Barry Vercoe's original front-end for csound. Some partially sighted or unsighted users report success using Winsound with text-to-speech software.
CsoundAC Python Scripting You can use CsoundAC as a Python extension module. You can do this in a standard Python interpreter, such as Python command line or the Idle Python GUI. To use CsoundAC in a standard Python interpreter, import CsoundAC. import CsoundAC The CsoundAC module automatically creates an instance of CppSound named csound, which provides an object-oriented interface to the Csound API. In a standard Python interpreter, you can load a Csound .csd file and perform it like this: C:\Documents and Settings\mkg>python Python 2.3.3 (#51, Dec 18 2003, 20:22:39) [MSC v.1200 32 bit (Intel)] on win32 Type "help", "copyright", "credits" or "license" for more information. >>> import CsoundAC >>> csound.load("c:/projects/csound5/examples/trapped.csd") 1 >>> csound.exportForPerformance() 1 >>> csound.perform() BEGAN CppSound::perform(5, 988ee0)... BEGAN CppSound::compile(5, 988ee0)... Using default language 0dBFS level = 32767.0 Csound version 5.00 beta (float samples) Jun 7 2004 libsndfile-1.0.10pre6 orchname: temp.orc scorename: temp.sco orch compiler: 398 lines read instr 1 instr 2 instr 3 instr 4 instr 5 instr 6 instr 7 instr 8 instr 9 instr 10 instr 11 instr 12 instr 13 instr 98 instr 99 sorting score ... ... done Csound version 5.00 beta (float samples) Jun 6 2004 displays suppressed 0dBFS level = 32767.0 orch now loaded audio buffered in 16384 sample-frame blocks SFDIR undefined. using current directory writing 131072-byte blks of shorts to test.wav WAV SECTION 1: ENDED CppSound::compile. ftable 1: ftable 2: ftable 3: ftable 4: ftable 5: ftable 6: ftable 7: ftable 8: ftable 9: ftable 10: ftable 11: ftable 12: ftable 13: ftable 14: ftable 15: ftable 16: ftable 17: ftable 18: ftable 19: ftable 20: ftable 21: ftable 22: new alloc for instr 1: B 0.000 .. 1.000 T 1.000 TT 1.000 M: 32.7 0.0 new alloc for instr 1: B 1.000 .. 3.600 T 3.600 TT 3.600 M: 207.6 0.1 ... B 93.940 .. 94.418 T 98.799 TT281.799 M: 477.6 85.0 B 94.418 ..100.000 T107.172 TT290.172 M: 118.9 11.5 end of section 4 sect peak amps: 25950.8 26877.4 inactive allocs returned to freespace end of score. overall amps: 32204.8 31469.6 overall samples out of range: 0 0 0 errors in performance 782 131072-byte soundblks of shorts written to test.wav WAV Elapsed time = 13.469000 seconds. ENDED CppSound::perform. 1 >>> The koch.py script shows how to use Python to do algorithmic composition for Csound. You can use Python triple-quoted string literals to hold your Csound files right in your script, and assign them to Csound: csound.setOrchestra('''sr = 44100 kr = 441 ksmps = 100 nchnls = 2 0dbfs = .1 instr 1,2,3,4,5 ; FluidSynth General MID I; INITIALIZATION ; Channel, bank, and program determine the preset, that is, the actual sound. ichannel = p1 iprogram = p6 ikey = p4 ivelocity = p5 + 12 ijunk6 = p6 ijunk7 = p7 ; AUDIO istatus = 144; print iprogram, istatus, ichannel, ikey, ivelocityaleft, aright fluid "c:/projects/csound5/samples/VintageDreamsWaves-v2.sf2", \\ iprogram, istatus, ichannel, ikey, ivelocity, 1 outs aleft, arightendin''') csound.setCommand("csound --opcode-lib=c:/projects/csound5/fluid.dll \\ -RWdfo ./koch.wav ./temp.orc ./temp.sco") csound.exportForPerformance() csound.perform()
CsoundVST CsoundVST is a multi-function front end for Csound, based on the Csound API. CsoundVST runs as a stand-alone graphical user interface to Csound, and it also runs as a VST instrument or effect plugin in VST hosts such as Cubase with the same user interface. CsoundVST is part of the main csound source tree, but is not included in standard distributions, due to licensing limitations of Steinberg's VST SDK. Standalone To run CsoundVST as a stand-alone front end to Csound, execute CsoundVST. When the program has loaded, you will see a graphical user interface with a row of buttons along the top. Click on the Open... button to load a .csd file. You can also click on the Open... button and load a .orc file, then click on the Import... button to add a .sco file. You can edit the Csound command, the orchestra file, or the score file in the respective tabs of the user interface. When all is satisfactory, click on the Perform button to run Csound. You can stop a performance at any time by clicking on the Stop button. VST Plugin The following instructions are for Cubase 4.0. You would follow roughly similar procedures in other hosts. Use the Devices menu, Plug-In Information dialog, VST Plug-Ins tab, VST 2.x Plug-in Paths dialog, Add button to add your csound/bin directory to Cubase's plugin path. You can have multiple directories separated by semicolons. Then select the CsoundVST path and click on the Set as Shared Folder button. Quit Cubase, and start it again. Use the File menu, New Project dialog to create a new song. Use the Project menu, Add Track submenu, to add a new MIDI track. Use the pencil tool to draw a Part on the track a few measures long. Write some music in the Part using the Event editor or the Score editor. Use the Devices menu (or the F11 key) to open the VST Instruments dialog. Click on one of the No VST Instrument labels, and select CsoundVST from the list that pops up. Click on the e (for edit) button to open the CsoundVST dialog. On the Settings page, check the Instrument box in the VST Plugin group, and the Classic box in the Csound performance mode group. Then click on the Apply button. Click on the Open button to bring up the file selector dialog. Navigate to a directory containing a Csound csd file suitable for MIDI performance, such as csound/examples/CsoundVST.csd. Click on the OK button to load the file. You can also open and import a suitable .orc and .sco file as described above. In any event, the command line in the Classic Csound command line text box must specify -+rtmidi=null -M0, and should read something like this: csound -f -h -+rtmidi=null -M0 -d -n -m7 --midi-key-oct=4 --midi-velocity=5 temp.orc temp.sco Click on the VST Instruments dialog's on/off button to turn it on. This should compile the Csound orchestra. In the Cubase Track Inspector, click on the out: Not Assigned label and select CsoundVST from the list that pops up. On the ruler at the top of the Arrangement window, select the loop end point and drag it to the end of your part, then click on the loop button to enable looping. Click on the play button on the Transport bar. You should hear your music played by CsoundVST. Try assigning your track to different channels; a different Csound instrument will perform each channel. When you save your song, your Csound orchestra will be saved as part of the song and re-loaded when you re-load the song. You can click on the Orchestra tab and edit your Csound instruments while CsoundVST is playing. To hear your changes, just click on the CsoundVST Perform button to recompile the orchestra. You can assign up to 16 channels to a single CsoundVST plugin.
manual_src~dfsg/overview/design.xml0000644000000000000000000002272312262561504016614 0ustar rootroot
How Csound5 works Csound processes and generates output using "unit generators" (ugens) called opcodes. These opcodes are used to define instruments in the orchestra. When you run Csound, the engine loads the base Opcodes, and the opcodes contained in separate loadable "opcode libraries" . It then interprets the orchestra (through the orchestra reader). The engine sets up an instrument processing chain, which then receives events from the score or in real-time. The processing chain uses the input/output modules to generate output. There are modules that can write to file, or generate real-time audio output. [The Csound5 Modular structure.] The Csound5 Modular structure. Csound's processing buffers Csound processes audio in sample blocks called buffers. There are three separate buffer layers: spout = Csound's innermost software buffer, contains ksmps sample frames. Csound processes real-time control events once every ksmps sample frames. -b = Csound's intermediate software buffer (the "software" buffer), in sample frames. Should be (but does not need to be) an integral multiple of ksmps (can equal ksmps too). Once per ksmps sample frames, Csound copies spout to the -b buffer. Once per -b sample frames, Csound copies the -b buffer to the -B "hardware" buffer. -B = The sound card's internal buffer (the "hardware" buffer), in sample frames. Should be (and may need to be) an integral multiple of -b. If Csound misses delivering a -b one time, the extra -b sample frames in -b are still there for the sound card to keep playing while Csound catches up. But they can be the same size if you're willing to bet Csound can always keep up with the sound card.
Amplitude values in Csound Amplitude values in Csound are always relative to a "0dbfs" value representing the peak available amplitude before clipping, in either an AD/DA codec, or in a soundfile with a defined range (which both WAVE and AIFF are). In the original Csound, this value was always 32767, corresponding to the bipolar range of a 16bit soundfile or 16bit AD/DA codec, Csound's only possible output back then. This remains the default peak amplitude for Csound, for backward compatibility and you will find some of this manual's examples still use this value (hence you find large amplitude values like 10000). The 0dbfs value enables Csound to produce appropriately scaled values to whatever output format is being used, whether 24bit integer, 32bit floats, or even 32bit integers. Put another way, the literal amplitude values you write in a Csound instrument only match those written literally to the file if the 0dbfs value in Csound corresponds exactly to that of the output sample format. The consequence of this approach is that you can write a piece with a certain amplitude and have it render correctly and identically (setting aside of course the better dynamic range of the high-res formats) whether written to an integer or floats file, or rendered in real-time. The one exception to this is if you choose to write to a "raw" (headerless) file format. In such cases the internal 0dbfs value is meaningless, and whatever values you use are written unmodified. This does enable arbitrary data to be generated or processed by Csound. It is a relatively exotic thing to do, but some users need it. You can choose to redefine the 0dbfs value in the orchestra header, purely for your own convenience or preference. Many people will choose 1.0 (the standard for SAOL, other software like Pure Data, and for many plugin standards such as VST, LADSPA, CoreAudio AudioUnits, etc), but any value is possible. The common factor in defining amplitudes is the decibel (dB) scale, with 0dBFS always understood as digital peak; hence "0dbfs" means "0dB Full-Scale value". This measure is different to actual amplitude values, since amplitude values are a linear scale which show the actual oscillation around 0, so they can be positive or negative. Decibel values are an absolute logarithmic scale, but can be useful for most opcodes as well. You can convert amplitude to and from decibels using the ampdb,ampdbfs, dbamp and dbfsamp functions. This way, Csound enables the programmer to express all amplitudes in dB - lower amplitudes will then be represented by negative dB values. This reflects industry practice (e.g. in level meters in mixers, etc). For example the same dB level of -6dB (half the amplitude) or -20dB are actually a different linear amplitude according to 0dbfs like this: dB<subscript>FS</subscript> in relation to amplitude dBFS 0dbfs = 32767 (default) 0dbfs = 1 0dbfs = 1000 (unusual) 0 dB 32767 1 1000 -6 dB 16384 0.5 500 -20 dB 3276.7 0.1 100
Some Csound users might therefore be minded to express all levels in dBFS, and obviate any confusion or ambiguity of level that may otherwise arise when using explicit amplitude values. The decibel scale reflects the response of the ear pretty closely, and that when you want to express a really quiet level, it might be easier and more expressive to write "-46dB" than "0.005" or "163.8". The reason for using 0dbfs is very simple: digital peak equates to maximum level regardless of sample resolution. If you then define a signal at -110dB you will lose it if rendering to a 16bit file, but retain it (audibly or not) if rendering to 24bit or better. In other words, there is a fixed ceiling, but a moveable floor - you can define sounds as quietly as you like (e.g. envelope tails), in a predictable way,and preserve them or not (without changing the orch code at all), depending on the resolution (file or audio i/o) you render to. A note on digital amplitude, decibels and dynamic range A convenient aproximation of dynamic range for a certain digital precision is to calculate the decibel interval between the minimum value and the maximum value for a sample. As a rule of thumb, 1 bit (doubling of level) is 6dB, so 16bits = 96dB. This is not entirely accurate since audio sample values are represented on a bipolar scale with positive and negative values, and 1 bit is used for the sign. Therefore, for 16bit integer samples actually use 1 bit for the sign and 15 bits for the values, so the actual dynamic range is 90dB.
manual_src~dfsg/overview/configuring.xml0000644000000000000000000000601412262561504017650 0ustar rootroot Configuring Once you have either unpacked a binary distribution, or built Csound from sources, you will need to configure Csound so that it will run properly on your system. Installers usually perform these steps for you automatically. On all platforms, make sure the directory or directories containing Csound's plugin libraries are in an OPCODEDIR or OPCODEDIR64 environment variable depending on the precision of the compiled binary. The Python opcodes currently require at least Python 2.4, which can be downloaded from www.python.org if it is not already on your system. You can check if it is available by typing 'python' on a command prompt or DOS window. Windows On Windows, make sure the directory or directories (normally the C:\Program Files\Csound directory) containing the Csound executables directory are in your PATH variable, or else copy all the executable files to your Windows system32 directory. Depending on your installation method, you might also need to set the OPCODEDIR and OPCODEDIR64 environment variables. Assuming that Csound is installed to the default location of C:\Program Files\Csound you can use (otherwise set the paths accordingly): set OPCODEDIR=C:\Program Files\Csound\plugins set OPCODEDIR64=C:\Program Files\Csound\plugins64 set PATH=%PATH%;C:\Program Files\Csound\bin Missing python24.dll or python25.dll If you get a pop-up about the missing Python library (python24.dll or python25.dll) and don't need the python opcodes, just delete C:\Program Files\Csound\plugins\py.dll and C:\Program Files\Csound\plugins64\py.dll, and the pop-up about the missing Python library should be gone. Unix and Linux On Unix and Linux, either install the Csound program in one of the system bin directories, typically /usr/local/bin, and the Csound and plugin shared libraries in places like /usr/local/lib/csound/plugins or /usr/local/lib/csound/plugins64 and make sure that OPCODEDIR and OPCODEDIR64 environment variable are set correctly. CsoundAC CsoundAC requires some additional configuration. On all platforms, CsoundAC requires that you have Python installed on your computer. The directory containing the _csoundAC shared library and the CsoundAC.py file must be in your PYTHONPATH environment variable, so that the Python runtime knows how to load these files. manual_src~dfsg/overview/usingXO.xml0000644000000000000000000004332112262561504016734 0ustar rootroot Using Csound Csound can be operated in a variety of modes and configurations. The original method for running Csound was as a console program (DOS prompt for Windows, Terminal for Mac OS X). This, of course, still works. Running csound without any arguments prints out a list of command-line options, which are more fully explained in the Command Line Flags (by Category) section. Normally, the user executes something like: csound myfile.csd or separate orchestra (orc) and score (sco) files can be used: csound myorchestra.orc myscore.sco You can find many .csd files in the examples folder. Most opcode entries in this manual also include simple .csd files showing the usage of the opcode. Csound also has several ways of producing output. It can: Read and write soundfiles (off-line rendering) - Using the -o and -i flags specifying an output filename. Read and write digital audio using a sound card (real-time rendering) - Using the -odac and -iadc flags Read and write MIDI files (non-realtime) - Using the -F and --midioutfile flags. Read and write MIDI using a MIDI interface and controller (real-time control) - Using the -M and -Q flags.
Csound's Console Output When Csound runs, it prints a text output to the console, which shows data about the Csound run. A Console output looks something like this: time resolution is 0.455 ns PortMIDI real time MIDI plugin for Csound virtual_keyboard real time MIDI plugin for Csound PortAudio real-time audio module for Csound 0dBFS level = 32768.0 Csound version 5.10 beta (float samples) Apr 19 2009 libsndfile-1.0.17 Reading options from $HOME/.csoundrc UnifiedCSD: oscil.csd STARTING FILE Creating options Creating orchestra Creating score orchname: /tmp/csound-XYACV6.orc scorename: /tmp/csound-IYtLAJ.sco rtaudio: ALSA module enabled rtmidi: PortMIDI module enabled orch compiler: 17 lines read instr 1 Elapsed time at end of orchestra compile: real: 0.129s, CPU: 0.020s sorting score ... ... done Elapsed time at end of score sort: real: 0.130s, CPU: 0.020s Csound version 5.10 beta (float samples) Apr 19 2009 displays suppressed 0dBFS level = 32768.0 orch now loaded audio buffered in 256 sample-frame blocks ALSA input: total buffer size: 1024, period size: 256 reading 1024-byte blks of shorts from adc (RAW) ALSA output: total buffer size: 1024, period size: 256 writing 1024-byte blks of shorts to dac SECTION 1: ftable 1: new alloc for instr 1: B 0.000 .. 2.000 T 2.000 TT 2.000 M: 10000.0 10000.0 Score finished in csoundPerform(). inactive allocs returned to freespace end of score. overall amps: 10000.0 10000.0 overall samples out of range: 0 0 0 errors in performance Elapsed time at end of performance: real: 2.341s, CPU: 0.050s 345 1024-byte soundblks of shorts written to dac Removing temporary file /tmp/csound-CoVcrm.srt ... Removing temporary file /tmp/csound-IYtLAJ.sco ... Removing temporary file /tmp/csound-XYACV6.orc ... The console output of Csound is quite verbose, particularly before the actual performance (like version, plugins loaded, etc.). Performance actually started when the console printed: SECTION 1: In this particular run, the lines: new alloc for instr 1: B 0.000 .. 2.000 T 2.000 TT 2.000 M: 10000.0 10000.0 Show that a single note for instrument 1, that lasted 2 seconds starting at time 0.000, was produced with an amplitude of 10000 for both channel 1 and 2. An important section of the console output is: end of score. overall amps: 10000.0 10000.0 overall samples out of range: 0 0 Which shows the overall amplitude and the number of samples which were clipped because they were out of range. The line: Elapsed time at end of performance: real: 2.341s, CPU: 0.050s Shows the clock time and the CPU time it takes for the processor to complete the task. If CPU time is lower than clock time it means the csd can run in realtime (unless it has some sections which are extremely CPU intensive). The "real time" figure is the total running time and it is larger because it acounts for disk access. module loading, etc. (CPU time is strictly number-crunching time). If you have a sound that lasts for 100s and it takes 5s to generate it offline, it means that you are taking around 5% of CPU, and that it runs on 0.05 of realtime.
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Real-Time Audio The -i and -o flags can are used to specify realtime output instead of the ordinary non-realtime file output. You should use -o dac for realtime output and -i adc for realtime input. Naturally, you can use either one or both if your hardware supports it. You can also specify the hardware you want to use by appending a device number or name to the flag (See -i and -o). Period & Buffer Sizes Period and buffer sizes will vary greatly from one machine to another. Lower buffer sizes will result in lower latency, but might cause breakups or clicks in the audio. The Csound flags which control period and buffer sizes are -b and -B, respectively. Buffer size is hardware dependant, and some experimentation may be necessary to find the optimal balance between low latency performance and uninterrupted audio output. The values given to -b and -B should be powers of two, and the value of -B should be at least one power of two higher than that of -b. Currently, with -B set to 512, audio output latency is about 12 milliseconds, fast enough for reasonably responsive keyboad playing. Even shorter latencies, are feasible on some systems. Control Rate Low values for ksmps will in general give a higher quality of synthesis, but will consume more system resources. There is no hard and fast rule for setting ksmps - different orchestras will require different control rates. A waveguide instrument will need a ksmps of 1 (and may not be suitable for realtime use), whereas a simple FM synth may be run with a higher ksmps without noticeable degradation of sound. If the FM synth were to be used to play a monophonic bassline, a very low ksmps may be used, however more complex note clusters will require a higher ksmps. A well-tuned Linux system should be capable of running even complex polyphonic synths with ksmps values as low as 4 or 8. If full duplex audio is required, -b must be an integer multiple of ksmps. Bearing this in mind, a rule of thumb might be to only use powers of two for ksmps.
Optimizing Audio I/O Latency To achieve the lowest latency possible without audio break ups, a combination of variables needs to be tweaked. The final values will be platform and system dependent, and will also depend on the complexity of the audio calculations performed. You need to adjust ksmps in the orchestra, as well as the software (-b) and harware buffer (-B) sizes. Usually the simplest solution is the following: Set ksmps to a value with a good tradeoff between quality and performance, without adjusting -B at all. Set -b to a negative power of two of this value. To get the optimal values, start with something you think is going to be too low, ie -1, and then continue "upwards", -2, -4 and so on, until you stop getting x-runs (glitches). The real value of -b will be the absolute value of -b * ksmps. Reduce the hardware buffer (-B). Bring it down from the default (1024 on Linux, 4096 on Mac OS X, 16384 on Windows), halving it each time, until you start to get x-runs (glitches) again. Then take it back up again until performance is continuous. This process assumes you have a 16-bit soundcard. If you have a 24-bit soundcard, then -B should be 3/2, or 3 times -b, rather than 2 or 4 times. Csound works with 32-bit floats, or 64-bit doubles whereas most soundcards are 16 or 24-bit integer. -b is the internal buffer, so it's dealing with the 32 or 64-bit side of things, whereas -B is the hardware buffer, so it's dealing with the 16 or 24-bit side. The csound default for floats is -B = 4 * -b. This is a sane value for a 16 bit card. You can usually get away with -B = 2 * -b, but this is the absolute minimum. For example, if you set -b1024 -B2048, csound will tell you that: audio buffered in 1024 sample-frame blocks writing 4096-byte blocks to dac 4096 bytes is 32768 bits. 32768/32 = 1024, our sample-frame size, 1024 * 32/16 = 2048, our buffer size. Were we to reduce the value of -B, we would need to reduce the value of -b by a corresponding amount in order to continue to write 16-bit integers to dac. The minimum size of -b is (-B * bitrate)/32. That is to say that the minimum ratio of -b to -B should be: 16-bit: 1:2 24-bit: 2:3 32-bit: 1:1 While there is no theoretical maximum ratio, it makes no sense to have a very high ratio here, as the software buffer has to fill the hardware buffer before returning. If the ratio is high, it will take a long time, defeating the purpose of setting a small value for -b. The value of -b is something that will need to be varied depending on the complexity of the instrument you're working with, but because it's intimately related to the value of ksmps, it's better to synchronise it with ksmps and go from there. One way to do it is to decide how long the release on your envelopes might need to be at maximum (for desired effect), set the release on all envelopes to maximum, give yourself a generous value for -b, and then play. If it breaks up, double ksmps, repeat until smooth, then bring the value of -b down as far as possible. The value of -B is primarily determined by operating system and soundcard. Figure out (using above method) how low you can go, and use that value (or one higher for safety). If you have problems you'll know that it's probably because of an inappropriate value for ksmps, too low a value for -b, or denormals (see denorm).
manual_src~dfsg/overview/realtimewindows.xml0000644000000000000000000000624712262561504020563 0ustar rootroot
Windows Real-time Audio Windows users can use either the default PortAudio Realtime module, or the winmm Realtime Audio Module. The winmm module is a native windows module which provides great stability, but latency will usually be too high for realtime interaction. To activate a realtime module, you can use the -+rtaudio flag with value of portaudio or winmme. The default value is portaudio, which is activated by default without specifying it. You also need to specify the sound device you want to use, and specify that you want to generate real-time audio ouput instead of soundfile to disk output. To do this, you must use the -odac or -o dac flag, which tells csound to output to the Digital-to-Analog converters instead of a file. By adding a number after the flag (e.g. -odac2), you can choose the device number you want. To find out available devices in your system, you can use a large out of range number (e.g. -odac99), and csound will report an error, and list available devices. When choosing the device number under Portaudio, you are also choosing the driver interface, since Portaudio supports WinMME, DirectX and ASIO. If you have an ASIO capable interface or an ASIO driver emulator like ASIO4ALL, the device will show multiple times, once for each driver interface. ASIO will give you the best latency on your system, so if available it should be your choice for realtime audio output. Enabling realtime audio input is done using -iadc, which makes csound listen to the realtime audio outputs. You can again select the device by its number, and check for available devices using an out of range number. Note that for input you use 'adc' instead of 'dac'. Make sure you have the appropriate input selected in your soundcard's control panel. Real-time MIDI To enable Real-time MIDI on Windows, you can use the -M flag for MIDI input and the -Q flag for MIDI output. You might need to specify the device number after the flag (e.g. -M2), and again, you can find the available devices by giving an out of range number. Csound will use PortMidi as the default MIDI module, but there's also a native winmme module, which can be activated with the flag: -+rtmidi=winmme A typical set of flags to enable Real-time Audio and MIDI I/O can look like: -+rtmidi=winmme -M1 -Q1 -+rtaudio=portaudio -odac3 -iadc3
manual_src~dfsg/overview/building.xml0000644000000000000000000005665612262561504017154 0ustar rootroot Building Csound Csound has become a complex project and can involve many dependencies. Unless you are a Csound developer or need to develop Csound plugins, you should try to use one of the precompiled distributions from http://www.sourceforge.net/projects/csound. However, building from source is probably the best option on GNU/Linux. This section focuses on the main Csound 6 build system, which uses cmake, a cross-platform, open-source build system designed to build, test and package software. When building Csound from source instead of using a precompiled package, you first need to obtain the sources for a release of Csound at http://www.sourceforge.net/projects/csound. The source packages have either a zip or tar.gz extension. The latest (possibly unstable) Csound source code is also available through GIT. The Csound GIT front page is located at: http://csound.git.sourceforge.net/git/gitweb-index.cgi. Using git for just checking out and compiling is pretty easy. Install GIT for your platform and use this command to clone the Csound6 git repository: git clone git://git.code.sf.net/p/csound/csound6-git csound-csound6-git This will checkout a read-only (meaning, you can not commit back to the central git repository) version of the Csound6 repo. To update with the latest from the master repo, use: git pull The same process is used for the Csound manual. Use this command to clone: git clone git://git.code.sf.net/p/csound/manual6-git csound-manual6-git Basic requirements to build Csound 6 on any platform Install libdsndfile version 1.0.13 or later from www.mega-nerd.com/libsndfile. Install the Cmake build system from www.scons.org or from your system packages. These are the minimum requirements for a build, but csound has many optional components which enhance functionality and add opcodes which may require additional libraries. Optional configurations (ALL platforms) In most cases it is best to install the most recent stable versions of the optional libraries. Real-time audio can use the cross-platform PortAudio library (trunk version or devel-19 branch) from www.portaudio.com/usingcvs.html. Please note that stable version 18 will not work. Csound can also use several platform specific audio APIs like ALSA, JACK, CoreAudio and the Windows multimedia library, see each platform notes for details. On Windows, you can use the pre-built PortAudio from https://github.com/adfernandes/precompiled-portaudio-windows Real-time MIDI can use the cross-platform PortMidi library from www.cs.cmu.edu/~music/portmusic For GUI widgets, install FLTK 1.1 or 1.3 from www.fltk.org. You must configure and build FLTK with --enable-shared --enable-threads. For generating Python and Java interfaces, install the Software Interface and Wrapper Generator (SWIG) from http://www.swig.org. Install Python from www.python.org. In most cases it is best to install the most recent 2.7 version. CsoundAC requires FLTK and the boost C++ template libraries for random numbers and linear algebra, from http://www.boost.org. CsoundAC requires at least version 1.32.1. The fluid opcodes require the Fluidsynth library from http://savannah.nongnu.org/download/fluid. The OSC opcodes require the latest version of the liblo library from http://plugin.org.uk/liblo. On Windows, liblo requires a Windows version of the POSIX thread library (pthreads) which is available from http://sourceware.org/pthreads-win32; copy libpthreadGC2.a to libpthread.a. You may also need the latest version of autoconf from MinGW. The STK opcodes require STK source code from http://ccrma.stanford.edu/software/stk, copied into csound6/Opcodes/stk. The Loris opcodes requires Loris 1.8 to be installed: http://sourceforge.net/projects/loris/files. It will create the Csound Loris opcodes (along with Loris itself). When you run Csound use an additional command line flag: --opcode-lib. For example in Linux: --opcode-lib=/usr/local/lib/libloris.so Windows The following is needed to build on Windows (more complete build instructions for Windows may be found in How_to_Build_Csound_on_Windows.doc): Install a compiler like gcc or Microsoft Visual Studio. If using MinGW (gcc), download the MinGW-build installer for 32-bit architecture, Posix threads, dwarf2 exception handling from http://sourceforge.net/projects/mingwbuilds/files/latest/download?source=files and run it. Install to a directory such as D:/mingw32-4.8.1. In addition to the compiler, building Csound and its dependencies requires some other tools. Install MSys from http://sourceforge.net/projects/mingwbuilds/files/external-binary-packages/msys%2B7za%2Bwget%2Bsvn%2Bgit%2Bmercurial%2Bcvs-rev13.7z/download. This includes bison, flex, autotools and some other goodies. Edit the MSys etc/fstab file to map the MinGW directory to /mingw. Options for the MinGW C compiler should be -g -O2 -fopenmp -march=nocona -mstackrealign -DNDEBUG -I/usr/local/include. Options for the MinGW C++ compiler should be -g -O2 -std=gnu++11 -fopenmp -march=nocona -mstackrealign -DNDEBUG -I/usr/local/include. Optional configurations for Windows include the following: Real-time audio and MIDI can use the Windows multimedia library. This module will be built automatically if the headers are found. The VST Host opcodes require both the Steinberg VST headers. Linux Optional configurations for Linux include the following: Real-time audio on Linux can use ALSA (www.alsa-project.org) and JACK (www.jackaudio.org/) in addition to PortAudio. Distributions usually provide the appropriate dev packages for these systems through their repositories. The DSSI Host opcodes require both the LADSPA and DSSI headers. Mac OS X Optional configurations for Mac OS X include the following: Real-time audio can use CoreAudio (OSX builtin native audio system) and Jack, apart from PortAudio. The DSSI Host opcodes require both the LADSPA and DSSI headers. Building Csound 6 with CMake When you have all the necessary packages and their sources (or -dev packages) to support your particular requirements on your hardware platform it is best to review the file Custom.cmake.ex to discover the current configuration options. make any changes and place your customisations in Custom.cmake Building is made considerably easier if, when installing, the downloaded headers and libraries are installed in their default locations. To modify the default build, in particular to handle non-standard options for third-party dependencies, such as where headers and libraries are to be found: Avoid modifying the CMakeList.txt files. With CMake it is usual to build in a difference directory to the sources. Create a directory for your build, change to that directory and run cmake path-to-source-directory. All being well this will create Makefiles as necessary. You can then run "make" and "make install". If things go wrong it is sometime necessary to delete the file CMakeCache.txt. It is important that you set the environment variable OPCODE6DIR to the directory where plugin libraries are installed; in the case of a double precision build, OPCODE6DIR64 should be set instead. Installers usually take care of this, but it is necessary when building from source so Csound can find its plugin libraries. manual_src~dfsg/overview/realtimemac.xml0000644000000000000000000000015012262561504017614 0ustar rootroot
Mac Coming Soon...
manual_src~dfsg/overview/links.xml0000644000000000000000000000371312262561504016461 0ustar rootroot Csound Links Csound's "home page" is maintained by Richard Boulanger at http://csounds.com. The Csound source code is maintained by John ffitch and others at http://www.sourceforge.net/projects/csound. The most recent versions and precompiled packages for most platforms also can be downloaded here. A Csound mailing list exists to discuss Csound. It is run by John ffitch of Bath University, UK. To have your name put on the mailing list send an empty message to: csound-subscribe@lists.bath.ac.uk. You can also subscribe to the digest (1 message per day) by sending an empty email to: csound-digest-subscribe@lists.bath.ac.uk. Posts sent to csound@lists.bath.ac.uk go to all subscribed members of the list. You can browse the csound mailing list archives here Similarly, the Csound-devel mailing list exists to discuss Csound development. For more information on this list, go to http://lists.sourceforge.net/lists/listinfo/csound-devel. Posts sent to csound-devel@lists.sourceforge.net go to all subscribed members of the list. manual_src~dfsg/overview/introduction.xml0000644000000000000000000000471212262561504020062 0ustar rootroot Introduction Csound is a unit generator-based, user-programmable computer music system. It was originally written by Barry Vercoe at the Massachusetts Institute of Technology in 1984 as the first C language version of this type of software. Since then Csound has received numerous contributions from researchers, programmers, and musicians from around the world. Around 1991, John ffitch ported Csound to Microsoft DOS. Csound currently runs on many varieties of UNIX and Linux, Microsoft DOS and Windows, all versions of the Macintosh operating system including Mac OS X, and others. There are newer computer music systems that have graphical patch editors (e.g. Max/MSP, PD, jMax, or Open Sound World), or that use more advanced techniques of software engineering (e.g. Nyquist or SuperCollider). Yet Csound still has the largest and most varied set of unit generators, is the best documented, runs on the most platforms, and is the easiest to extend. It is possible to compile Csound using double-precision arithmetic throughout for superior sound quality. In short, Csound must be considered one of the most powerful musical instruments ever created. In addition to this "canonical" version of Csound and CsoundAC, there are other versions of Csound and other front ends for Csound, many of which can be found at http://csounds.com. manual_src~dfsg/overview/using.xml0000644000000000000000000004326412262561504016473 0ustar rootroot Using Csound Csound can be operated in a variety of modes and configurations. The original method for running Csound was as a console program (DOS prompt for Windows, Terminal for Mac OS X). This, of course, still works. Running csound without any arguments prints out a list of command-line options, which are more fully explained in the Command Line Flags (by Category) section. Normally, the user executes something like: csound myfile.csd or separate orchestra (orc) and score (sco) files can be used: csound myorchestra.orc myscore.sco You can find many .csd files in the examples folder. Most opcode entries in this manual also include simple .csd files showing the usage of the opcode. There are also many Front-Ends which can be used to run csound. A Front-End is a graphical program that eases the process of running csound, and sometimes provides editing and composing functions. Csound also has several ways of producing output. It can: Read and write soundfiles (off-line rendering) - Using the -o and -i flags specifying an output filename. Read and write digital audio using a sound card (real-time rendering) - Using the -odac and -iadc flags Read and write MIDI files (non-realtime) - Using the -F and --midioutfile flags. Read and write MIDI using a MIDI interface and controller (real-time control) - Using the -M and -Q flags.
Csound's Console Output When Csound runs, it prints a text output to the console, which shows data about the Csound run. A Console output looks something like this: time resolution is 0.455 ns PortMIDI real time MIDI plugin for Csound virtual_keyboard real time MIDI plugin for Csound PortAudio real-time audio module for Csound 0dBFS level = 32768.0 Csound version 5.10 beta (float samples) Apr 19 2009 libsndfile-1.0.17 Reading options from $HOME/.csoundrc UnifiedCSD: oscil.csd STARTING FILE Creating options Creating orchestra Creating score orchname: /tmp/csound-XYACV6.orc scorename: /tmp/csound-IYtLAJ.sco rtaudio: ALSA module enabled rtmidi: PortMIDI module enabled orch compiler: 17 lines read instr 1 Elapsed time at end of orchestra compile: real: 0.129s, CPU: 0.020s sorting score ... ... done Elapsed time at end of score sort: real: 0.130s, CPU: 0.020s Csound version 5.10 beta (float samples) Apr 19 2009 displays suppressed 0dBFS level = 32768.0 orch now loaded audio buffered in 256 sample-frame blocks ALSA input: total buffer size: 1024, period size: 256 reading 1024-byte blks of shorts from adc (RAW) ALSA output: total buffer size: 1024, period size: 256 writing 1024-byte blks of shorts to dac SECTION 1: ftable 1: new alloc for instr 1: B 0.000 .. 2.000 T 2.000 TT 2.000 M: 10000.0 10000.0 Score finished in csoundPerform(). inactive allocs returned to freespace end of score. overall amps: 10000.0 10000.0 overall samples out of range: 0 0 0 errors in performance Elapsed time at end of performance: real: 2.341s, CPU: 0.050s 345 1024-byte soundblks of shorts written to dac Removing temporary file /tmp/csound-CoVcrm.srt ... Removing temporary file /tmp/csound-IYtLAJ.sco ... Removing temporary file /tmp/csound-XYACV6.orc ... The console output of Csound is quite verbose, particularly before the actual performance (like version, plugins loaded, etc.). Performance actually started when the console printed: SECTION 1: In this particular run, the lines: new alloc for instr 1: B 0.000 .. 2.000 T 2.000 TT 2.000 M: 10000.0 10000.0 Show that a single note for instrument 1, that lasted 2 seconds starting at time 0.000, was produced with an amplitude of 10000 for both channel 1 and 2. An important section of the console output is: end of score. overall amps: 10000.0 10000.0 overall samples out of range: 0 0 Which shows the overall amplitude and the number of samples which were clipped because they were out of range. The line: Elapsed time at end of performance: real: 2.341s, CPU: 0.050s Shows the clock time and the CPU time it takes for the processor to complete the task. If CPU time is lower than clock time it means the csd can run in realtime (unless it has some sections which are extremely CPU intensive). The "real time" figure is the total running time and it is larger because it acounts for disk access. module loading, etc. (CPU time is strictly number-crunching time). If you have a sound that lasts for 100s and it takes 5s to generate it offline, it means that you are taking around 5% of CPU, and that it runs on 0.05 of realtime.
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Real-Time Audio The following information applies mostly to csound being run directly from the command line. Front-ends implement these features in different ways, but knowledge of them is necessary in some of them. The -i and -o flags can are used to specify realtime output instead of the ordinary non-realtime file output. You should use -o dac for realtime output and -i adc for realtime input. Naturally, you can use either one or both if your hardware supports it. You can also specify the hardware you want to use by appending a device number or name to the flag (See -i and -o). You might also need to use the -+rtaudio flag to specify the driver interface to be used. Csound defaults to using Portaudio, which is cross-plaform and reliable, but for better performance, you might need to use ALSA or JACK on linux, and CoreAudio on Mac. You can use ASIO on Windows if your version of Portaudio has been compiled with ASIO support. You can see a list of available devices by giving a device number which is out of range, for instance -o dac99. This will also reveal if you have ASIO available if you are using PortAudio. Period & Buffer Sizes Period and buffer sizes will vary greatly from one machine to another. Lower buffer sizes will result in lower latency, but might cause breakups or clicks in the audio. The Csound flags which control period and buffer sizes are -b and -B, respectively. Buffer size is hardware dependant, and some experimentation may be necessary to find the optimal balance between low latency performance and uninterrupted audio output. The values given to -b and -B should be powers of two, and the value of -B should be at least one power of two higher than that of -b. Currently, with -B set to 512, audio output latency is about 12 milliseconds, fast enough for reasonably responsive keyboad playing. Even shorter latencies, are feasible on some systems. Control Rate Low values for ksmps will in general give a higher quality of synthesis, but will consume more system resources. There is no hard and fast rule for setting ksmps - different orchestras will require different control rates. A waveguide instrument will need a ksmps of 1 (and may not be suitable for realtime use), whereas a simple FM synth may be run with a higher ksmps without noticeable degradation of sound. If the FM synth were to be used to play a monophonic bassline, a very low ksmps may be used, however more complex note clusters will require a higher ksmps. A well-tuned Linux system should be capable of running even complex polyphonic synths with ksmps values as low as 4 or 8. If full duplex audio is required, -b must be an integer multiple of ksmps. Bearing this in mind, a rule of thumb might be to only use powers of two for ksmps. Some settings differ according to platform. See further below for information for each platform. &oviewrealtimelinux; &oviewrealtimewindows; &oviewrealtimemac;
Optimizing Audio I/O Latency To achieve the lowest latency possible without audio break ups, a combination of variables needs to be tweaked. The final values will be platform and system dependent, and will also depend on the complexity of the audio calculations performed. You need to adjust ksmps in the orchestra, as well as the software (-b) and harware buffer (-B) sizes. Usually the simplest solution is the following: Set ksmps to a value with a good tradeoff between quality and performance, without adjusting -B at all. Set -b to a negative power of two of this value. To get the optimal values, start with something you think is going to be too low, ie -1, and then continue "upwards", -2, -4 and so on, until you stop getting x-runs (glitches). The real value of -b will be the absolute value of -b * ksmps. Reduce the hardware buffer (-B). Bring it down from the default (1024 on Linux, 4096 on Mac OS X, 16384 on Windows), halving it each time, until you start to get x-runs (glitches) again. Then take it back up again until performance is continuous. This process assumes you have a 16-bit soundcard. If you have a 24-bit soundcard, then -B should be 3/2, or 3 times -b, rather than 2 or 4 times. Csound works with 32-bit floats, or 64-bit doubles whereas most soundcards are 16 or 24-bit integer. -b is the internal buffer, so it's dealing with the 32 or 64-bit side of things, whereas -B is the hardware buffer, so it's dealing with the 16 or 24-bit side. The csound default for floats is -B = 4 * -b. This is a sane value for a 16 bit card. You can usually get away with -B = 2 * -b, but this is the absolute minimum. For example, if you set -b1024 -B2048, csound will tell you that: audio buffered in 1024 sample-frame blocks writing 4096-byte blocks to dac 4096 bytes is 32768 bits. 32768/32 = 1024, our sample-frame size, 1024 * 32/16 = 2048, our buffer size. Were we to reduce the value of -B, we would need to reduce the value of -b by a corresponding amount in order to continue to write 16-bit integers to dac. The minimum size of -b is (-B * bitrate)/32. That is to say that the minimum ratio of -b to -B should be: 16-bit: 1:2 24-bit: 2:3 32-bit: 1:1 While there is no theoretical maximum ratio, it makes no sense to have a very high ratio here, as the software buffer has to fill the hardware buffer before returning. If the ratio is high, it will take a long time, defeating the purpose of setting a small value for -b. The value of -b is something that will need to be varied depending on the complexity of the instrument you're working with, but because it's intimately related to the value of ksmps, it's better to synchronise it with ksmps and go from there. One way to do it is to decide how long the release on your envelopes might need to be at maximum (for desired effect), set the release on all envelopes to maximum, give yourself a generous value for -b, and then play. If it breaks up, double ksmps, repeat until smooth, then bring the value of -b down as far as possible. The value of -B is primarily determined by operating system and soundcard. Figure out (using above method) how low you can go, and use that value (or one higher for safety). If you have problems you'll know that it's probably because of an inappropriate value for ksmps, too low a value for -b, or denormals (see denorm).
manual_src~dfsg/overview/recentdev.xml0000644000000000000000000002367012262561504017324 0ustar rootroot Recent Developments In the time since Barry Vercoe wrote the original Preface to this manual, printed above, many further contributions have been made to Csound. CsoundAC is an extended version of Csound 5.
Features of Csound 5 Csound 5 begins a new major version of Csound that includes the following new features: Now licensed under the GNU Lesser General Public License, an open source license. A new, easier to manage build system using SCons. The use of widely--accepted open source libraries: libsndfile for soundfile input and output. PortAudio with ASIO drivers for low-latency, real-time audio input and output. FLTK for graphical widgets that can be programmed in orchestra code. PortMidi for real-time MIDI input and output. In addition, Istvan Varga has contributed native MIDI and audio drivers for Windows and Linux. Simplified audio buffering system. Status returns from all internal functions, including opcode functions. MIDI interop opcodes, that enable the same instrument definitions to be used interchangeably for either live MIDI performance or off-line, score-driven performance. Plugin opcodes are working and becoming more widely accepted. Many opcodes have been moved to plugins. Most new opcodes are plugins, including: The FluidSynth-based SoundFont opcodes. Python opcodes allowing Python code to execute in the orchestra header or in instrument code, at i-rate or k-rate. Loris opcodes for time/frequency analysis and resynthesis. Control bus opcodes. Audio mixer opcodes. String conversion opcodes. Improved Open Sound Control (OSC) opcodes. Vectorial opcodes. The pvs opcodes for real-time spectral processing, a port of Mark Dolson's phase vocoder code. The ATS opcodes for spectral Analysis, Transformation, and Synthesis of sound based on a sinusoidal plus critical-band noise model. A sound in ATS is a symbolic object representing a spectral model that can be sculpted using a variety of transformation functions. These opcodes can read, transform and resynthesize ATS analysis files. Please note that you need the ATS application to produce analysis files. The STK opcodes, consisting of Perry Cook's original Synthesis Toolkit in C++ instruments, in C++, adapted as opcodes. DSSI and LADSPA adapter opcodes for hosting DSSI and LADSPA plugins in Csound. vst4cs VST adapter opcodes for hosting VST plugins in Csound. (Distributed in source form only due to the VST SDK licence restrictions.) The OpcodeBase.hpp header file for writing plugin opcodes in C++. This is based on the technique of static polymorphism via template inheritance. Istvan Varga's csound5gui frontend for Csound, simplifying the editing of Csound, the use of Csound especially for live performance, and the monitoring of performances. Victor Lazzarini's Tcl/Tk frontends for Csound, cstclsh and cswish. The Csound API is becoming more standardized and more widely used. There are interfaces or wrappers to the API in the following languages: C (include csound.h). C++ (include csound.hpp)). This API includes Csound score and orchestra file container functions. Python (import csnd). Java (import csnd.*;). Lua (require "csnd";). Lisp (use the CFFI file csound5.lisp). Csound is now truly re-entrant, meaning that multiple instances of Csound can run at the same time, in the same process. John ffitch plans to replace the handwritten parser with one written using a parser generator, which should make it more bug-free and perhaps more efficient.
Features of CsoundAC CsoundAC is a Python extension module for writing music by programming in Python. CsoundAC is based on Michael Gogins' concept of music graphs, in which a score is represented as hierarchical tree of nodes, which can contain notes, score generators, score transforms, and other nodes. CsoundAC also provides a Python interface to the Csound API. This makes it very easy to use Csound to render CsoundAC compositions. Using Python's triple quotes, it is even possible to embed the formatted Csound orchestra code for a piece directly into the Python code for that piece, so that all programming for a composition can be maintained in a single file. The coordinate system in CsoundAC is based on a Euclidean music space with dimensions {time, duration, event type, instrument number, pitch as MIDI key, loudness as MIDI velocity, phase, spatial X coordinate, spatial Y coordinate, spatial Z coordinate, pitch-class set, 1}. A point in music space can be a note, an inflection of a note, or even a grain of sound. A music graph is a directed acyclical graph, or tree, of nodes in music space. These nodes are associated with local transformations of coordinate system. There are nodes for containing scores or fragments of scores, for generating scores, and for transforming scores. In addition, any node may contain child nodes that inherit the parent's coordinate system. Thus, it is possible to compose a musical score by containing or generating notes in lower level nodes, assembling them into a score using higher level nodes, and finally rendering the score with Csound. The process is strictly analogous to the construction of a 3-dimensional scene in computer graphics by generating primitive objects such as spheres, cones, and cubes and moving them around in space to assemble a scene. Some of the node classes included in CsoundAC are: ScoreNode: Simply contains a sequence of notes or other points in music space, perhaps imported from a MIDI file. Rescale: Rescales child points to fit a desired range in time, duration, pitch, and/or other dimensions. Cell: Repeats child points in a sequence at regular intervals; the interval can be shorter or longer than the actual duration of the child points. Hocket: Hockets points produced by child nodes. Lindenmayer: Generates scores using O-L Lindenmayer systems. StrangeAttractor: Generates scores from a variety of tunable chaotic dynamical systems. MCRM: Generates scores using the Multiple Copy Reducing Machine algorithm. ImageToScore: Generates scores by translating image files into points in music space. Random: Randomizes child points on any dimension or dimensions of music space, using a variety of random variables. VoiceleadingNode: Generates chord progressions and voice-leadings for child notes, using operations based on the mathematical music theory of Dmitri Tymoczko. Finally, is is possible to derive a new Node class in Python from any existing Node, in order to create new score generators and transforms as part of the composing process.
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Realtime I/O on Linux Under Linux, the default PortAudio/PortMidi settings will result in higher latency than that which can be achieved using ALSA and/or JACK. The PortMusic plugins are audio and MIDI servers, which provide an interface to the ALSA drivers, in a manner which is in some respects similar but fundamentally different from that provided by JACK. For a more detailed comparison please refer to: http://jackaudio.org/faq Using ALSA The highest level of control and the lowest possible level of latency are to be achieved using the ALSA plugins in combination with the --sched flag. Using --sched requires that Csound be run as the root user, which may be impossible or undesirable in some circumstances. The ALSA plugins require the "name" of a "card" and a "device". Unless you have named your "cards" in ~/.asoundrc (or /etc/asound.conf), the "names" will actually be numbers. In order to obtain a list of the possible configurations, use the command line utilities "aplay", "arecord" and "amidi". These utilities are included with most Linux distros, or can be downloaded and built from source here: ftp://ftp.alsa-project.org/pub/utils/ Audio Output Running the following command: aplay -l will give you a list of the audio playback devices available on your system. Typically this list will look something like: [....] **** List of PLAYBACK Hardware Devices **** card 0: A5451 [ALI 5451], device 0: ALI 5451 [ALI 5451] [....] If you have more than one card on your system, or if there is more than one device on your card, the list will of course be more complicated, however in all cases the information that is pertinent is the number/name of the card/device. In order to use the above soundcard for audio output, the following flag would be added to the Csound command line, ~/.csoundrc, or the <CsOptions>section of a CSD: -+rtaudio=alsa -o dac Output with dmix If you would like to use Csound with dmix and your soundcard does not support hardware mixing of audio streams, special care is needed in setting up of software (-b) and hardware (-B) buffers. If you get a message from Csound's ALSA driver that looks like the following: ALSA: -B 8192 not allowed on this device; use 7526 instead there is a good chance that you may be using dmix. If you are using dmix, the -b and -B settings of Csound must be synced the period_size and buffer_size of dmix respectively, using a ratio of the sr for the Csound project to the sample rate that dmix is set up to. The following formula will determine what settings to use for Csound given the settings of dmix: -b = (csound_sr/dmix_sample_rate) * dmix_period_size -B = (csound_sr/dmix_sample_rate) * dmix_buffer_size For example, if dmix is set to 48000 sample rate, a period_size of 1024, and a buffer_size of 8192, when running a Csound project with sr=48000, the settings for buffers should be "-b 1024 -B8192". If the sr=24000, the settings for buffers should be "-b 512 -B4096". Because of this relationship, if a Csound project's sr does not evenly divide into the sample_rate used by dmix, then it may be difficult if not imposible to set the correct setting for -b and -B due to rounding errors. It is suggested then that if you are using sample rates different than what your setting is for dmix, then you may want to configure dmix to have a period_size and buffer_size that can be evenly divided by the ratio between the csound sr and dmix sample_rate. For example, to run a project with sr=16000, the following dmix setting: pcm.amix { type dmix ipc_key 50557 slave { pcm "hw:0,0" period_time 0 #period_size 1024 #buffer_size 8192 period_size 1536 buffer_size 12288 } bindings { 0 0 1 1 } } # route ALSA software through pcm.amix pcm.!default { type plug slave.pcm "amix" } with period_size 1536 and buffer_size 12288 will divide nicely by 3 (the ratio of the csound sr to the dmix sample_rate) to get "-b 512 -B4096" ( (16000/48000) * 1536 = 512, (16000/48000) * 12288 = 4096 ). Note For most soundcards that this affects, the default sample rate for the card will be 48000 and the defaults for dmix will be 1024 and 8192. Audio Input Typically the same card will be used for both input and output, so to continue using the foregoing example, the flag: -i adc:hw:0,0 would be added for audio input from Card 0 Device 0. To use the default card employ one of the following flags, with the forementioned warning that this will not necessarily work: -i adc If you wish to use a different card or device for input, running the following utility from the command line will provide a list of input devices: arecord -l If, by way of an example, you wanted to use a USB audio interface, which is the second "card" in your system, for output, but wanted to use your internal soundcard, the first card in your setup, for input, you would put the following flags somewhere useful: -+rtaudio=alsa -i adc:hw:0,0 -o dac:hw:1,0 If you wanted to use the second device on your USB interface, to send audio to a specific channel, for instance, you would use the following flags: -+rtaudio=alsa -i adc:hw:0,0 -o dac:hw:1,1 MIDI Input Csound does not automatically create its own ALSA sequencer port. It creates an ALSA raw midi port each time it runs. In order to enable your orchestra to receive MIDI input you can use VirMIDI or MIDIThru, whichever you prefer. Setting up these virtual MIDI ports is a topic that has been covered extensively elsewhere, see The Linux MIDI how-to or browse your distro's documentation or the ALSA documentation for instructions on how to install and configure either VirMIDI or MIDIThru (seqdummy). Once you have done so run: amidi -l for a list of available devices. Typically this will look something like the following: [....] Device Name hw:1,0 Virtual Raw MIDI (16 subdevices) hw:1,1 Virtual Raw MIDI (16 subdevices) hw:1,2 Virtual Raw MIDI (16 subdevices) hw:1,3 Virtual Raw MIDI (16 subdevices) hw:2,0,0 PCR MIDI hw:2,0,1 PCR 1 In this example, Csound can connect to any of the four available Virtual Raw MIDI ports, where it will listen for MIDI input. The following flag instructs Csound to listen on the first of these ports: -+rtmidi=alsa -Mhw:1,0 You will then need to connect your hardware or software controller to the port which is hosting your Csound synthesizer. The simplest way to do this is using the "aconnect" utility. Run: aconnect -li for a list of available input devices, and: aconnect -lo for a list of available output devices (including the port to which Csound has been connected). These should give something like the following: #aconnect -li client 0: 'System' [type=kernel] 0 'Timer ' 1 'Announce ' Connecting To: 15:0 client 20: 'Virtual Raw MIDI 1-0' [type=kernel] 0 'VirMIDI 1-0 ' client 21: 'Virtual Raw MIDI 1-1' [type=kernel] 0 'VirMIDI 1-1 ' client 22: 'Virtual Raw MIDI 1-2' [type=kernel] 0 'VirMIDI 1-2 ' client 23: 'Virtual Raw MIDI 1-3' [type=kernel] 0 'VirMIDI 1-3 ' client 24: 'PCR' [type=kernel] 0 'PCR MIDI ' 1 'PCR 1 ' 2 'PCR 2 ' #aconnect -lo client 20: 'Virtual Raw MIDI 1-0' [type=kernel] 0 'VirMIDI 1-0 ' client 21: 'Virtual Raw MIDI 1-1' [type=kernel] 0 'VirMIDI 1-1 ' client 22: 'Virtual Raw MIDI 1-2' [type=kernel] 0 'VirMIDI 1-2 ' client 23: 'Virtual Raw MIDI 1-3' [type=kernel] 0 'VirMIDI 1-3 ' client 24: 'PCR' [type=kernel] 0 'PCR MIDI ' 1 'PCR 1 ' In the following example, the USB keyboard which is listed above as client 24 will be connected to a Csound synthesizer which is listening on the first VirMIDI port. The keyboard has three output ports. The first (24:0) transmits messages received on the MIDI in port, the second (24:1) transmits keyboard and controller messages, and the third (24:2) transmits system exclusive messages. The following command connects the second port of the keyboard to the Csound synthesizer: aconnect 24:1 20:0 Remember that Csound acts as a raw MIDI device and is not an ALSA sequencer client. This means that Csound will not appear in MIDI device listings and will not be available for use directly with aconnect, so you must connect to a virtual device (like 'virtual raw MIDI' or 'MIDI through') for persistent connections, or conect directly to the destination using command line flags. MIDI Output Csound can be connected to any device which shows up on the ALSA sequencer list of output ports, obtained by "amidi -l" as above. In order to connect a Csound synthesizer to the MIDI out port of the keyboard listed above, the following flag would be used: -Qhw:2,0,0 Scheduling If you are able to run Csound as the root user, using the "--sched" flag will dramatically improve realtime performance, when using ALSA, however you may hang your system if you do something stupid. DO NOT use "--sched" if you are using JACK for audio output. JACK controls scheduling for the audio applications connected to it, and also tries to run at the highest possible priority. If the "--sched" flag is used, Csound and JACK will be competing rather than cooperating, resulting in extremely poor performance. Using JACK The simplest way to use the JACK plugin enabling input and output is as follows: -+rtaudio=jack -i adc -o dac Additionally, there are some command line options specific to JACK: JACK Command-line Flags -+jack_client=[client_name] The client name used by Csound, defaults to 'csound5'. If multiple instances of Csound connect to the JACK server, different client names need to be used to avoid name conflicts. -+jack_inportname=[input port name prefix], -+jack_outportname=[output port name prefix] Name prefix of Csound JACK input/output ports; the default is 'input' and 'output'. The actual port name is the channel number appended to the name prefix. Example: with the above default settings, a stereo orchestra will create these ports in full duplex operation: csound5:input1 (record left) csound5:input2 (record right) csound5:output1 (playback left) csound5:output2 (playback right) -+jack_sleep_time=[sleep time in microseconds] As of Csound version 5.01, this option is deprecated and ignored. Connecting Csound to other JACK clients By default, no connections are made (you need to use jack_connect, qjackctl, or a similar utility); however, the plugin can connect to ports specified as '-iadc:portname_prefix' or '-odac:portname_prefix', where portname_prefix is the full name of a port without a channel number, such as 'alsa_pcm:capture_' (for -i adc), or 'alsa_pcm:playback_' (for -o dac). Notes on buffer sizes Audio data is received from and sent to the JACK server by Csound using a ring buffer that is controlled by the -b and -B flags. -B is the total size of the buffer, while -b is the size of a single period. These values are rounded so that the total size is an integer multiple of, and greater than the period size. The difference of the Csound buffer and period size must be greater than or equal to the JACK period size. If both -iadc and -odac are used at the same time, the -b option should be set to an integer multiple of ksmps. An example of buffer settings for low latency on a fast system: jackd -d alsa -P -r 48000 -p 64 -n 4 -zt & csound -+rtaudio=jack -b 64 -B 256 [...] with real time scheduling (as root): jackd -R -P 90 -d alsa -P -r 48000 -p 64 -n 2 -zt & csound --sched=80,90,10 -d -+rtaudio=jack -b 64 -B 192 [...] To improve performance, use ksmps values like 32 and 64. The sample rate of the orchestra must be the same as that of the JACK server. Using Pulseaudio Support for Pulseaudio was added in Csound 5.09. You can specify the following settings: Sink names: it's possible to use a number instead of the full name, so -odac:1 would select your second device (count starts at 0). Server name: it's possible to connect to a specific server by using -+server=<server_string> where server_string is a name of a server or a more complex server selection string (see pulseaudio.org on server strings). This should be network transparent and should allow connections to remote machines. Stream names: it is possible to label the streams generated by csound, by using -+output_stream=<stream-name> and -+input_stream=<stream-name> Here's an example command line: csound -odac:1 examples/trapped.csd -+rtaudio=pulse -+server=unix:/tmp/pulse-victor/native -+output_stream=trapped
manual_src~dfsg/deprecated/0000755000000000000000000000000012262561504015045 5ustar rootrootmanual_src~dfsg/deprecated/top.xml0000644000000000000000000000477612262561504016407 0ustar rootroot Deprecated Orchestra Opcodes and GEN Routines
Deprecated Orchestra Opcodes The following opcodes are deprecated. They are still distributed with Csound for backward compatibility. The new opcodes which are to be used in place of the deprecated ones are indicated within each of the following entries.
&opcodesabetarand; &opcodesabexprnd; &opcodesacauchy; &opcodesaexprand; &opcodesagauss; &opcodesagogobel; &opcodesalinrand; &opcodesapcauchy; &opcodesapoisson; &opcodesapow; &opcodesatrirand; &opcodesaunirand; &opcodesaweibull; &opcodesbformenc; &opcodesbformdec; &opcodesclock; &opcodesdiskin; &opcodesibetarand; &opcodesibexprnd; &opcodesicauchy; &opcodesictrl14; &opcodesictrl21; &opcodesictrl7; &opcodesiexprand; &opcodesigauss; &opcodesilinrand; &opcodesimidic14; &opcodesimidic21; &opcodesimidic7; &opcodesinstimek; &opcodesinstimes; &opcodesioff; &opcodesion; &opcodesiondur; &opcodesiondur2; &opcodesioutat; &opcodesioutc; &opcodesioutc14; &opcodesioutpat; &opcodesioutpb; &opcodesioutpc; &opcodesipcauchy; &opcodesipoisson; &opcodesipow; &opcodesis16b14; &opcodesis32b14; &opcodesislider16; &opcodesislider32; &opcodesislider64; &opcodesislider8; &opcodesitablecopy; &opcodesitablegpw; &opcodesitablemix; &opcodesitablew; &opcodesitrirand; &opcodesiunirand; &opcodesiweibull; &opcodeskbetarand; &opcodeskbexprnd; &opcodeskcauchy; &opcodeskdump; &opcodeskdump2; &opcodeskdump3; &opcodeskdump4; &opcodeskexprand; &opcodeskfilter2; &opcodeskgauss; &opcodesklinrand; &opcodeskon; &opcodeskoutat; &opcodeskoutc; &opcodeskoutc14; &opcodeskoutpat; &opcodeskoutpb; &opcodeskoutpc; &opcodeskpcauchy; &opcodeskpoisson; &opcodeskpow; &opcodeskread; &opcodeskread2; &opcodeskread3; &opcodeskread4; &opcodesktableseg; &opcodesktrirand; &opcodeskunirand; &opcodeskweibull; &opcodespeakk; &opcodessoundout; &opcodessoundouts;
Deprecated Gen Routines The following GEN routine is deprecated. It is still distributed with Csound for backward compatibility. The new GEN routine which is to be used in place of the deprecated one is indicated within the following entry.
&scoregensgen22; manual_src~dfsg/plugin/0000755000000000000000000000000012262561504014243 5ustar rootrootmanual_src~dfsg/plugin/dssi4cs.xml0000644000000000000000000000253112262561504016342 0ustar rootroot
DSSI and LADSPA for Csound dssi4cs enables the use of DSSI and LADSPA plugin effects and synthesizers within Csound on Linux. The following opcodes are available: dssiinit - Loads a plugin. dssiactivate - Activates or deactivates a plugin if it has this facility dssilist - Lists all available plugins found in the LADSPA_PATH and DSSI_PATH global variables. dssiaudio - Process audio using a Plugin. dssictls - Send control information to a plugin's control port. See the entry for dssiinit for a usage example. Currently only LADSPA plugins are supported, but DSSI support is planned.
manual_src~dfsg/plugin/vst4cs.xml0000644000000000000000000000376412262561504016225 0ustar rootroot
VST for Csound vst4cs enables the use of VST plugin effects and synthesizers within Csound. The following opcodes are available: vstinit - Loads a plugin. vstaudio, vstaudiog - Returns a plugin's output. vstmidiout - Sends MIDI data to a plugin. vstparamset, vstparamget - Sends and receives automation data to and from the plugin. vstnote - Sends a MIDI note with definite duration. vstinfo - Outputs the Parameter and Program names for a plugin. vstbankload - Loads an .fxb Bank. vstprogset - Sets a Program in an .fxb Bank. vstedit - Opens the GUI editor for the plugin, when available. Credits By: Andres Cabrera and Michael Gogins Uses code from Hermann Seib's VSTHost and Thomas Grill's vst~ object. VST is a trademark of Steinberg Media Technologies GmbH. VST Plug-In Technology by Steinberg.
manual_src~dfsg/plugin/top.xml0000644000000000000000000000052612262561504015572 0ustar rootroot Plugin Hosting Csound currently hosts external plugins using dssi4cs (for LADSPA plugins) on Linux and vst4cs (for VST plugins) on Windows and Mac OS X. &plugindssi4cs; &pluginvst4cs; manual_src~dfsg/csbeats/0000755000000000000000000000000012262561504014371 5ustar rootrootmanual_src~dfsg/csbeats/syntax.xml0000644000000000000000000001115312262561504016442 0ustar rootroot The csbeats language is very simple, having only 5 kinds of statement, and only one of them has any complexity. In general the introductory word for each statement type is case insensitive, so "QUIT", "quit", "QuIt"... are all the same. Comments can be introduced in either ANSI C89 format or C++ (that is either /* ... */ or // to the end of line) or Csound's semicolon. QUIT Causes csbeats to exit. For flexibility the command END is also accepted for the identical action. BEATS=integer Sets the number of beats per minute for the following score until the end or until it is reset. The default value is 60bpm. The token BPS is also acceptable instead of BEATS. PERMEASURE=integer Sets the number of beats in a bar. The default value is 4. BAR Start a new bar. BAR integer Start the bar whose number is given. i integer attributes Specified a note event for the numbered instrument. The attributes may be any of a pitch, duration or dynamic, or a positioning of the note to a beat or measure, and can be in any order. Pitches are specified with a conventional note name (English form) in upper case optionally followed by a #, x (for double sharp), b (for flat) or bb (for double flat). A note of Z is a rest (think zzzz). All notes except rests must be followed by an octave number, with A4 being international A (440Hz). Pitches are passed to Csound in Hertz in the parameter p4, and are twelve tone equal temprament. Durations are coded in lower case with the initial letter of the name for whole, half, quarter, eigth and sixteeth notes, or th for thirtysecondth notes. Except for w these can be moderated by appending as suffix: d or . Dotted note (half as long again) dd or .. Dotted note (three quarters as long again) t Triplet note (third notes in two) t. Dotted triplet note q Quintuplet (Five notes in four) s Septuplet (Seven notes in eight) Durations can be added together by giving more that one duration. To make this more intuitive a + sign can be used instead of white space. The dynamics are written in conventional notation, that is fff, ff, f, mf, mp, p, pp, ppp, pppp. These are passed to the instrument as p5 as 0 for fff, and one less dB for each step below. The default dynamic is fortissimo. Additional parameters (p-fields) can be specified in an instrument statement by giving statements like p6=42 which would set the p6 field to 42 until it is changed. The number must be 6 or greater as the first 5 fields are taken with fixed information. Each instrument number has its own additional parameters. If any of these attributes is missing it carried forward from the previous note, with beat position being incremented to the end of the previous note. In addition an event can be directed to a particular measure with an m attribute or a particular beat with a b. manual_src~dfsg/csbeats/top.xml0000644000000000000000000000124212262561504015714 0ustar rootroot Csbeats Csbeats Csbeats is an alternative score language that is aimed at specifying simple scores in standard western tunings and rhythms. Csbeats can be invoked via the CsScore component of a standard .csd score with bin="csbeats" or as stand-alone program which generates a standard numeric score. As a stand-alone the program reads standard input and writes to standard output. &csbeatssyntax; &csbeatsexample; manual_src~dfsg/csbeats/example.xml0000644000000000000000000001213412262561504016547 0ustar rootroot The opening of Bach's Goldberg variation number 3 can be coded as: ; Bach - Goldberg Variations - Variato 3 ; by Brian Baughn 3-14-05 ; bbaughn@berklee.net beats = 120 permeasure = 6 i101 m1 b1 B4 mp qd+s i101 C5 s i101 D5 i101 C5 i101 D5 i101 E5 i101 A4 qd+s i101 B4 s i101 C5 i101 B4 i101 C5 i101 D5 i101 m2 b1 G4 qd i101 G5 qd+e i101 A5 s i101 G5 i101 F#5 i101 G5 i101 A5 e i101 m3 b1.5 D5 s i101 C5 i101 B4 i101 A4 i101 B4 e i101 C5 s i101 B4 i101 A4 i101 B4 i101 G4 e i101 E5 i101 D5 i101 C5 i101 F#5 i101 A5 i101 m4 b1 B4 q i101 G5 e i101 G5 q i101 F#5 e i101 Z e // Z is a rest (zzzzz..) i101 e i101 B5 e i101 A5 q i101 D5 e quit This produces output ;;;setting bpm=120.000000 ;;;setting permeasure=6 i101 0.000000 0.875000 493.883621 -4 i101 0.875000 0.125000 523.251131 -4 i101 1.000000 0.125000 587.329536 -4 i101 1.125000 0.125000 523.251131 -4 i101 1.250000 0.125000 587.329536 -4 i101 1.375000 0.125000 659.255114 -4 i101 1.500000 0.875000 440.000000 -4 i101 2.375000 0.125000 493.883621 -4 i101 2.500000 0.125000 523.251131 -4 i101 2.625000 0.125000 493.883621 -4 i101 2.750000 0.125000 523.251131 -4 i101 2.875000 0.125000 587.329536 -4 i101 3.000000 0.750000 391.995436 -4 i101 3.750000 1.000000 783.990872 -4 i101 4.750000 0.125000 880.000000 -4 i101 4.875000 0.125000 783.990872 -4 i101 5.000000 0.125000 739.988845 -4 i101 5.125000 0.125000 783.990872 -4 i101 5.250000 0.250000 880.000000 -4 i101 6.250000 0.125000 587.329536 -4 i101 6.375000 0.125000 523.251131 -4 i101 6.500000 0.125000 493.883621 -4 i101 6.625000 0.125000 440.000000 -4 i101 6.750000 0.250000 493.883621 -4 i101 7.000000 0.125000 523.251131 -4 i101 7.125000 0.125000 493.883621 -4 i101 7.250000 0.125000 440.000000 -4 i101 7.375000 0.125000 493.883621 -4 i101 7.500000 0.250000 391.995436 -4 i101 7.750000 0.250000 659.255114 -4 i101 8.000000 0.250000 587.329536 -4 i101 8.250000 0.250000 523.251131 -4 i101 8.500000 0.250000 739.988845 -4 i101 8.750000 0.250000 880.000000 -4 i101 9.000000 0.500000 493.883621 -4 i101 9.500000 0.250000 783.990872 -4 i101 9.750000 0.500000 783.990872 -4 i101 10.250000 0.250000 739.988845 -4 ;;rest at 10.500000 for 0.250000 ;;rest at 10.750000 for 0.250000 i101 11.000000 0.250000 987.767243 -4 i101 11.250000 0.500000 880.000000 -4 i101 11.750000 0.250000 587.329536 -4 e A Complete Example Here is a simple example of the csbeats score generator. It uses the file csbeats.csd. A simple example of csbeats. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. manual_src~dfsg/miscopcodes/0000755000000000000000000000000012262561504015255 5ustar rootrootmanual_src~dfsg/miscopcodes/top.xml0000644000000000000000000000076212262561504016606 0ustar rootroot Miscellaneous opcodes Here is a list of opcodes that don't fall in any category: system - Call an external program via the system call. modmatrix - modulation matrix opcode with optimizations for sparse matrices. manual_src~dfsg/STKopcodes/0000755000000000000000000000000012262561504014763 5ustar rootrootmanual_src~dfsg/STKopcodes/top.xml0000644000000000000000000001104512262561504016310 0ustar rootroot STK Opcodes These opcodes are an optional component of Csound5. You can check if they are installed by using the command 'csound -z' which lists all available opcodes. The STK opcodes can be built in several ways. If you are already building Csound, simply follow the directions at the top of the Opcodes/stk/stkOpcodes.cpp file. If you do not wish to build all of Csound, you can still build the STK opcodes as an opcode plugin (shared library). You can adapt the example from the Csound GIT repository in the pluginSDK directory. Replace the pluginSDK/examplePlugin.c file with the opcodes/stk/stkOpcodes.cpp file, and change the name of this file in the SConstruct build file. Then use scons to build the opcodes. Be sure to use the scons command-line option useDouble=0 if you have single-precision (32 bit) Csound, and useDouble=1 if you have double-precision (64 bit) Csound. You must also either add all the STK source code files and include path to the SConstruct file, or you must add a prebuilt STK library and include path to the SConstruct file. The STK source files would go near line 296 in the SConstruct file: makePlugin(pluginEnvironment, 'stkOpcodes', Split(''' stkOpcodes.cpp ... STK source files ... ''')) The STK family of opcodes wraps: STKBandedWG STKBeeThree STKBlowBotl STKBlowHole STKBowed STKBrass STKClarinet STKDrummer STKFlute STKFMVoices STKHevyMetl STKMandolin STKModalBar STKMoog STKPercFlut STKPlucked STKResonate STKRhodey STKSaxofony STKShakers STKSimple STKSitar STKStifKarp STKTubeBell STKVoicForm STKWhistle STKWurley For more information about the STK opcodes, visit The Synthesis ToolKit in C++ (STK) web site at https://ccrma.stanford.edu/software/stk. Credits This implementation of the STK unit generators was written by Perry R. Cook and Gary P. Scavone. The opcodes were further adapted as a plugin for Csound 5 by Michael Gogins. manual_src~dfsg/copying.txt0000644000000000000000000005044112262561504015162 0ustar rootrootCopyright (c) 1986, 1992 by the Massachusetts Institute of Technology. All rights reserved. Developed by Barry L. Vercoe at the Experimental Music Studio, Media Laboratory, M.I.T., Cambridge, Massachusetts, with partial support from the System Development Foundation and from National Science Foundation Grant # IRI-8704665. Copyright (c) 2003 by Kevin Conder for modifications made to the Public Csound Reference Manual. Permission is granted to copy, distribute and/or modify this document under the terms of the GNU Free Documentation License, Version 1.2 or any later version published by the Free Software Foundation; with no Invariant Sections, no Front-Cover Texts, and no Back-Cover Texts. A copy of this license is available in the examples sub-directory. A legal notice from the Public Csound Reference Manual... "The original Hypertext Edition of the MIT Csound Manual was prepared for the World Wide Web by Peter J. Nix of the Department of Music at the University of Leeds and Jean Piche of the Faculte de musique de l'Universite de Montreal. A Print Edition, in Adobe Acrobat format, was then maintained by David Boothe. The editors fully acknowledge the rights of the authors of the original documentation and programs, as set out above, and further request that this notice appear wherever this material is held." The Public Csound Reference Manual's last known network location was http://www.lakewoodsound.com/csound/hypertext/manual.htm. The Alternative Csound Reference Manual's network location, for both the Transparent and Opaque copies, is http://kevindumpscore.com/download.html#csound-manual. GNU Free Documentation License Version 1.2, November 2002 Copyright (C) 2000,2001,2002 Free Software Foundation, Inc. 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA Everyone is permitted to copy and distribute verbatim copies of this license document, but changing it is not allowed. 0. PREAMBLE The purpose of this License is to make a manual, textbook, or other functional and useful document "free" in the sense of freedom: to assure everyone the effective freedom to copy and redistribute it, with or without modifying it, either commercially or noncommercially. Secondarily, this License preserves for the author and publisher a way to get credit for their work, while not being considered responsible for modifications made by others. This License is a kind of "copyleft", which means that derivative works of the document must themselves be free in the same sense. It complements the GNU General Public License, which is a copyleft license designed for free software. We have designed this License in order to use it for manuals for free software, because free software needs free documentation: a free program should come with manuals providing the same freedoms that the software does. But this License is not limited to software manuals; it can be used for any textual work, regardless of subject matter or whether it is published as a printed book. We recommend this License principally for works whose purpose is instruction or reference. 1. APPLICABILITY AND DEFINITIONS This License applies to any manual or other work, in any medium, that contains a notice placed by the copyright holder saying it can be distributed under the terms of this License. Such a notice grants a world-wide, royalty-free license, unlimited in duration, to use that work under the conditions stated herein. The "Document", below, refers to any such manual or work. Any member of the public is a licensee, and is addressed as "you". You accept the license if you copy, modify or distribute the work in a way requiring permission under copyright law. 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If the Document does not specify a version number of this License, you may choose any version ever published (not as a draft) by the Free Software Foundation. manual_src~dfsg/csd2docbook.py0000644000000000000000000002607112262561504015521 0ustar rootroot# -*- coding: utf-8 -*- # # Copyright (C) 2007 Francois Pinot # # This program is free software; you can redistribute it and/or # modify it under the terms of the GNU Lesser General Public License # as published by the Free Software Foundation; either version 2.1 # of the License, or (at your option) any later version. # # This program is distributed in the hope that it will be useful, # but WITHOUT ANY WARRANTY; without even the implied warranty of # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. # See the GNU Lesser General Public License for more details. # # You should have received a copy of the GNU Lesser General Public # License along with this program; if not, write to the # Free Software Foundation, Inc., # 59 Temple Place - Suite 330, Boston, MA 02111, USA # or connect to: # http://www.fsf.org/licensing/licenses/lgpl.html # import csnd6 import getopt, sys, os, glob, re from xml.dom import minidom class OrchestraDict(dict): def __init__(self): self.initOpcodes() self.initHeaderSymbols() self.initBlockDelim() self.initOperators() self.initSeqCtrl() self.initMacros() self.labels = [] def initOpcodes(self): '''Read the list of csound opcodes from csound 5 API. Thus we're sure that the opcode list is up to date, however it will be complete only if csound has been built with all the plugins. It is needed to compile an orchestra and score before calling CsoundOpcodeList() in order to get the opcode list. So we use a dummy csd file for that purpose (tricky isn't it?) ''' csound = csnd6.CppSound() dummy = ''' -dodac instr 1 print p3 endin i1 0 1 e ''' f = open("dummy.csd", 'w') f.write(dummy) f.close() csound.Compile("dummy.csd") lo = csnd6.CsoundOpcodeList(csound.GetCsound()) n = lo.Count() print n, "Opcodes registered" for i in xrange(n): self[lo.Name(i)] = 'opc' lo.Clear() csound.Reset() os.remove("dummy.csd") def initHeaderSymbols(self): self.update({ 'sr':'ohdr', 'kr':'ohdr', 'ksmps':'ohdr', 'nchnls':'ohdr', '0dbfs':'ohdr', 'ctrlinit':'ohdr', 'ftgen':'ohdr', 'massign':'ohdr', 'pgmassign':'ohdr', 'seed':'ohdr', 'strset':'ohdr'}) def initBlockDelim(self): self.update({ 'instr':'oblock', 'endin':'oblock', 'opcode':'oblock', 'endop':'oblock'}) def initOperators(self): self.update({ '==':'op', '!=':'op', '?':'op', '>':'op', '>=':'op', 'LeSsThAn':'op', 'LeSsThAn=':'op', '=':'op', '+':'op', '-':'op', '*':'op', '/':'op', '%':'op', '^':'op', 'AmPeRsAnDAmPeRsAnD':'op', '||':'op', 'AmPeRsAnD':'op', '|':'op', '~':'op', ':':'op'}) def initSeqCtrl(self): self.update({ 'cggoto':'octrl', 'cigoto':'octrl', 'ckgoto':'octrl', 'cngoto':'octrl', 'elseif':'octrl', 'else':'octrl', 'endif':'octrl', 'goto':'octrl', 'if':'octrl', 'igoto':'octrl', 'kgoto':'octrl', 'tigoto':'octrl', 'timout':'octrl', 'loop_ge':'octrl', 'loop_gt':'octrl', 'loop_le':'octrl', 'loop_lt':'octrl'}) def initMacros(self): self.update({ '#define':'omacro', '#ifdef':'omacro', '#ifndef':'omacro', '#end':'omacro', '#else':'omacro', '#include':'omacro', '#undef':'omacro'}) def addLabel(self, key): self[key] = 'olabel' self.labels.append(key) def clearLabels(self): for key in self.labels: if self.has_key(key): self.pop(key) self.labels = [] def get(self, key, x=None): if self.has_key(key): return '' + key + '' return key class OrchestraTransform(object): def __init__(self): motif = '''( \s+ | \( | == | != | LeSsThAn= | LeSsThAn | >= | > | \? | = | \+ | - | \* | / | % | ^ | AmPeRsAnDAmPeRsAnD | \|\| | AmPeRsAnD | \| | ~ | : )''' self.pattern = re.compile(motif, re.VERBOSE) self.labelPattern = re.compile('[a-zA-Z]\w*:') self.orcDict = OrchestraDict() def detectLabels(self, lines): for s in lines: c = self.labelPattern.match(s) if c: self.orcDict.addLabel(s[:c.end()-1]) def comment(self, s): return '' + s + '' def orcLine(self, line): inTokens = self.pattern.split(line) outTokens = [] for t in inTokens: if len(t) > 0: outTokens.append(self.orcDict.get(t, line)) return ''.join(outTokens) def transformLines(self, lines): self.orcDict.clearLabels() self.detectLabels(lines) outLines = [] for s in lines: i = s.find(';') if i == 0: outLines.append(self.comment(s)) elif i > 0: outLines.append(self.orcLine(s[:i]) + self.comment(s[i:])) else: outLines.append(self.orcLine(s)) return outLines def transform(self, filename): f = open(filename, 'r') s = f.read() f.close() s = s.replace('&', 'AmPeRsAnD') s = s.replace('<', 'LeSsThAn') s = s.replace('LeSsThAnCs', '\n') fout.write(s) fout.write('') fout.close() print filename class ScoreTransform(object): def __init__(self): self.pattern = re.compile('^\s*[abefimnqrstvx]') def comment(self, s): return '' + s + '' def scoLine(self, line): r = self.pattern.search(line) if r: s = line[r.end()-1] return line.replace(s, '' + s + '') return line def transformLines(self, lines): outLines = [] for s in lines: i = s.find(';') if i == 0: outLines.append(self.comment(s)) elif i > 0: outLines.append(self.scoLine(s[:i]) + self.comment(s[i:])) else: outLines.append(self.scoLine(s)) return outLines class CsdTransform(object): def __init__(self): self.orc_t = OrchestraTransform() self.sco_t = ScoreTransform() def text(self, node): s = node.childNodes[0].nodeValue if s[0] == '\n': start = 1 else: start = 0 if s[-1] == '\n': end = -1 else: end = None return s[start:end].split('\n') def optTransform(self, lines): comment = lambda s: '' + s + '' outLines = [] for s in lines: i = s.find(';') if i == 0: outLines.append(comment(s)) elif i > 0: outLines.append(s[:i] + comment(s[i:])) else: outLines.append(s) return outLines def transform(self, filename): tag = lambda s: '<' + s + '>' f = open(filename, 'r') print filename s = f.read() f.close() # We convert & and < in "pseudo" entities because we don't want that # minidom resolves those entities s = s.replace('&', 'AmPeRsAnD') s = s.replace('<', 'LeSsThAn') s = s.replace('LeSsThAnCs', '') outLines.append(tag("CsoundSynthesizer")) if optionsPresent: outLines.append(tag("CsOptions")) outLines.extend(self.optTransform(self.text(options))) outLines.append(tag("/CsOptions")) outLines.append(tag("CsInstruments")) outLines.extend(self.orc_t.transformLines(self.text(orchestra))) outLines.append(tag("/CsInstruments")) outLines.append(tag("CsScore")) outLines.extend(self.sco_t.transformLines(self.text(score))) outLines.append(tag("/CsScore")) outLines.append(tag("/CsoundSynthesizer")) outLines.append('') csddoc.unlink() # We convert back our "pseudo" entities in actual entities s = '\n'.join(outLines).replace('AmPeRsAnD', '&') s = s.replace('LeSsThAn', '<') outfn = filename.replace('examples', 'examples-xml') fout = open(outfn + '.xml', 'w') fout.write(s) fout.close() def main(): '''Usage: python csd2xml option [filename] where option is one of the following: -f filename or --file=filename transform examples/filename to examples-xml/filename.xml where filename is the name of a csd file or orc file -a or --all transform all the csd files of the example directory to xml files in the examples-xml directory.''' try: opts, args = getopt.getopt(sys.argv[1:], "f:a", ["file=", "all"]) except getopt.GetoptError: print main.__doc__ sys.exit(2) for o, a in opts: if o in ("-f", "--file") and a: if a.find('examples') < 0: a = 'examples' + os.sep + a if os.path.exists(a): suf = os.path.splitext(a)[1] if suf == '.csd': CsdTransform().transform(a) elif suf == '.orc': OrchestraTransform().transform(a) else: print a + ' is not a csd or orc filename' sys.exit(0) else: print a + " doesn't exists!" sys.exit(2) if o in ("-a", "--all"): ct = CsdTransform() infiles = glob.glob('examples/*.csd') for f in infiles: ct.transform(f) sys.exit(0) print main.__doc__ sys.exit(2) if __name__ == '__main__': main() manual_src~dfsg/strings/0000755000000000000000000000000012262561504014436 5ustar rootrootmanual_src~dfsg/strings/top.xml0000644000000000000000000000701012262561504015760 0ustar rootroot Strings String variables are variables with a name starting with S or gS (for a local or global string variable, respectively), and can store any string with a maximum length defined by the -+max_str_len command line flag (255 characters by default). These variables can be used as input argument to any opcode that exepcts a quoted string constant, and can be manipulated at initialization or performance time with the opcodes listed below. It is also possible to use string p-fields. The string p-field can be used by many orchestra opcodes directly, or it can be copied to a string variable first: a1 diskin2 p5, 1 Sname strget p5 a1 diskin2 Sname, 1 Strings within Csound can be expressed using traditional double quotes (" "), an also using {{ }}. The second method is useful to allow ';' and '$' characters within the string without having to used ASCII codes. Note String variables and related opcodes are not available in Csound versions older than 5.00. Strings can also be linked to a number using strset and strget. Csound 5 also has improvements in parsing string constants. It is possible to specify a multi-line string by enclosing it within {{ and }} instead of the usual double quote characters (note that the length of string constants is not limited, and is not affected by the -+max_str_len option), and the following escape sequences are automatically converted: \a alert bell \b backspace \n new line \r carriage return \t tab \\ a single '\' character \nnn the character of which the ASCII code (in octal) is nnn It can be useful together with the system opcode: instr 1 ; csound5 lets you make a string with line returns inside double brackets system {{ ps date cd ~/Desktop pwd ls -l whois csounds.com }} endin And the python opcodes, among others: pyruni {{ import random pool = [(1 + i/10.0) ** 1.2 for i in range(100)] def get_number_from_pool(n, p): if random.random() < p: i = int(random.random() * len(pool)) pool[i] = n return random.choice(pool) }} &stringsmanipulate; &stringsconvert; manual_src~dfsg/strings/convert.xml0000644000000000000000000000273612262561504016650 0ustar rootroot
String Conversion Opcodes These opcodes convert string variables (note: most of the opcodes run at init time only, and have a version with a "k" suffix that runs at both init and performance time; exceptions to this rule include puts and strget): strtod and strtodk - Converts string value to a floating point value at i-rate. strtol and strtolk - Converts string value to signed integer at i-rate. strchar and strchark - Returns the ASCII code of a character in a string. strlower and strlowerk - Converts a string to lower case. strupper and strupperk - Converts a string to upper case.
manual_src~dfsg/strings/manipulate.xml0000644000000000000000000000523112262561504017320 0ustar rootroot
String Manipulation Opcodes These opcodes perform operations on string variables (note: most of the opcodes run at init time only, and have a version with a "k" suffix that runs at both init and performance time; exceptions to this rule include puts and strget): strcpy and strcpyk - Assigns to a string variable. strcat and strcatk - Concatenates strings, and stores the result in a variable. strcmp and strcmpk - Compares strings. strget - Assigns to a string variable, from strset table at the specified index, or string score p-field. strlen and strlenk - Returns the length of a string. sprintf - printf-style formatted output conversion, storing the result in a string variable. sprintfk - printf-style formatted output conversion, storing the result in a string variable at k-rate. puts - Prints a string constant or variable. strindex and strindexk - Returns the first occurence of a string in another string. strrindex and strrindexk - Returns the last occurence of a string in another string. strsub and strsubk - Returns a substring of the input string.
manual_src~dfsg/orch/0000755000000000000000000000000012262561504013700 5ustar rootrootmanual_src~dfsg/orch/tvars.xml0000644000000000000000000000104012262561504015554 0ustar rootroot
K-Rate Vectors Csound allows the declaration and deployment of one-dimensional vectors or tables. They are local to an instrument, and need to be declared for size (with the init opcode. Individual elements are read as part of any expression with square brackets to give an index at k-rate. Individual elements can be assigned, and there are a number of opcodes to query and modify tables.
manual_src~dfsg/orch/dirfiles.xml0000644000000000000000000001006512262561504016225 0ustar rootroot
Directories and Files Many generators and the Csound command itself specify filenames to be read from or written to. These are optionally full pathnames, whose target directory is fully specified. When not a full path, filenames are sought in several directories in order, depending on their type and on the setting of certain environment variables. The latter are optional, but they can serve to partition and organize the directories so that source files can be shared rather than duplicated in several user directories. The environment variables can define directories for soundfiles SFDIR, sound samples SSDIR, sound analysis SADIR, and include files for orchestra and score files INCDIR. SFDIR SSDIR SADIR INCDIR MFDIR In Csound version 5.00 and later, these environment variables can specify multiple directories as a ; separated list. If a file is found in more than one location, the first one has the highest precedence. The search order is: Soundfiles being written are placed in SFDIR (if it exists), else the current directory. Soundfiles for reading are sought in the current directory. If default paths are not disabled, files will next be sought for relative to the CSD/ORC/SCO file. Finally they will be sought in SSDIR and then SFDIR. Analysis control files for reading are sought in the current directory. If default paths are not disabled, files will next be sought for relative to the CSD/ORC/SCO file. Finally they will be sought in SADIR. MIDI files for reading are sought in the current directory. If default paths are not disabled, files will next be sought for relative to the CSD/ORC/SCO file. Finally they will be sought in MFDIR, SSDIR and SFDIR. Files of code to be included in orchestra and score files (with #include) are sought first in the current directory, then in the same directory as the orchestra or score file (as appropriate), then finally INCDIR.
manual_src~dfsg/orch/statemnt.xml0000644000000000000000000000177112262561504016267 0ustar rootroot
Ordinary Statements An ordinary statement runs at either init time or performance time or both. Operations which produce a result formally run at the rate of that result (that is, at init time for i-rate results; at performance time for k- and a-rate results), with the sole exception of the init opcode. Most generators and modifiers, however, produce signals that depend not only on the instantaneous value of their arguments but also on some preserved internal state. These performance-time units therefore have an implicit init-time component to set up that state. The run time of an operation which produces no result is apparent in the opcode. Arguments are values that are sent to an operation. Most arguments will accept arithmetic expressions composed of constants, variables, reserved symbols, value converters, arithmetic operations, and conditional values.
manual_src~dfsg/orch/nomen.xml0000644000000000000000000000502112262561504015534 0ustar rootroot
Nomenclature Throughout this document, opcodes are indicated in boldface and their argument and result mnemonics, when mentioned in the text, are given in italics. Argument names are generally mnemonic (amp, phs), and the result is usually denoted by the letter r. Both are preceded by a type qualifier i, k, a, or x (e.g. kamp, iphs, ar). The prefix i denotes scalar values valid at note init time; prefixes k or a denote control (scalar) and audio (vector) values, modified and referenced continuously throughout performance (i.e. at every control period while the instrument is active). Arguments are used at the prefix-listed times; results are created at their listed times, then remain available for use as inputs elsewhere. With few exceptions, argument rates may not exceed the rate of the result. The validity of inputs is defined by the following: arguments with prefix i must be valid at init time; arguments with prefix k can be either control or init values (which remain valid); arguments with prefix a must be vector inputs; arguments with prefix x may be either vector or scalar (the compiler will distinguish). All arguments, unless otherwise stated, can be expressions whose results conform to the above. Most opcodes (such as linen and oscil) can be used in more than one mode, which one being determined by the prefix of the result symbol. Thoughout this manual, the term "opcode" is used to indicate a command that usually produces an a-, k-, or i-rate output, and always forms the basis of a complete Csound orchestra statement. Items such as "+" or "sin(x)" or, "( a >= b ? c : d)" are called "operators."
manual_src~dfsg/orch/iblock.xml0000644000000000000000000000262612262561504015673 0ustar rootroot
Instrument and Opcode Block Statements An instrument block is comprised of ordinary statements that set values, control the logical flow, or invoke the various signal processing subroutines that lead to audio output. Statements that define an instrument block are: instr endin An instrument block looks like this: instr 1 ;A simple sine wave oscillator aout oscils 10000, 440, 0 out aout endin Statements that define a user defined opcode (UDO) block are opcode endop See the UDO section for more information.
manual_src~dfsg/orch/macros.xml0000644000000000000000000000320712262561504015710 0ustar rootroot
Macros Orchestra macros work like C preprocessor macros, and replace the content of the macro in the orchestra before it is compiled. The opcodes one can use to create, call, or undefine orchestra macros are: #define $NAME #ifdef #ifndef #end #else #include #undef Orchestra macros can also be defined using the command line flag --omacro:. More information and examples on the usage of orchestra macros can be found in the entry for #define. These opcodes refer to orchestra macros; for score macros, refer to Score Macros.
manual_src~dfsg/orch/kvar.xml0000644000000000000000000001603112262561504015366 0ustar rootroot
Types, Constants and Variables Constants are floating point numbers, such as 1, 3.14159, or -73.45. They are available continuously and do not change in value. Variables are named cells containing numbers. They are available continuously and may be updated at one of the four update rates (setup only, i-rate, k-rate, or a-rate). i- and k-rate variables are scalars (i.e. they take on only one value at any given time) and are primarily used to store and recall controlling data, that is, data that changes at the note rate (for i-rate variables) or at the control rate (for k-rate variables). i- and k-variables are therefore useful for storing note parameter values, pitches, durations, slow-moving frequencies, vibratos, etc. a-rate variables, on the other hand, are arrays or vectors of information. Though renewed on the same perf-time control pass as k-rate variables, these array cells represent a finer resolution of time by dividing the control period into sample periods (see ksmps). a-rate variables are used to store and recall data changing at the audio sampling rate (e.g. output signals of oscillators, filters, etc.). Some types of variables can be thought of as signals. For example a-rate and k-rate variables are signals that have a constant update frequency (see kr and sr). This abstraction is generally quite useful, but be aware that a-rate signals are actually vectors which are processed at k-rate, i.e. Csound works at k-rate internally but processes ksmps number samples for each a-rate variable on every control pass. There are other types of signals that require rates that don't match kr or sr. f-rate and w-rate signals are used for spectral processing and their rate is determined by the window size and overlap factor. A further distinction is that between local and global variables. local variables are private to a particular instrument, and cannot be read from or written into by any other instrument. Their values are preserved, and they may carry information from pass to pass (e.g. from initialization time to performance time) within a single instrument. Local variable names begin with the letter p, i, k, or a. The same local variable name may appear in two or more different instrument blocks without conflict. Global variables are cells that are accessible by all instruments. The names are either like local names preceded by the letter g, or are special reserved symbols. Global variables are used for broadcasting general values, for communicating between instruments (semaphores), or for sending sound from one instrument to another (e.g. mixing prior to reverberation). Given these distinctions, there are nine forms of local and global variables: Types of Variables Type When Renewable Local Global reserved symbols permanent -- rsymbol score pfields i-time p number -- init variables i-time i name gi name control signals p-time, k-rate k name gk name audio signals p-time, k-rate (all audio samples in a k-pass) a name ga name spectral data types k-rate w name -- streaming spectral data types k-rate f name gf name string variables i-time and optionally k-rate S name gS name vector variables k-rate t name
Where rsymbol is a special reserved symbol (e.g. sr, kr), number is a positive integer referring to a score pfield or sequence number, and name is a string of letters, the underscore character, and/or digits with local or global meaning. As might be apparent, score parameters are local i-rate variables whose values are copied from the invoking score statement just prior to the init pass through an instrument, while MIDI controllers are variables which can be updated asynchronously from a MIDI file or MIDI device.
Variable Initialization Opcodes that let one initialize variables are: assign divz init tival Predefined Math Constant Macros Csound defines several important math constants as Macros. You can see the full list here.
manual_src~dfsg/orch/top.xml0000644000000000000000000000750512262561504015233 0ustar rootroot Syntax of the Orchestra The Csound orchestra (.orc) or the <CsInstruments> section of a csd file, contains: A header section, which specifies global options for instrument performance A list of User defined opcodes and instrument blocks containing UDO and instrument definitions. The orchestra header, instrument blocks, and UDOs contain Orchestra statements. An orchestra statement in Csound has the format: label: result opcode argument1, argument2, ... ;comments The label is optional and identifies the basic statement that follows as the potential target of a go-to operation (see Program Flow Control). A label has no effect on the statement per se. Depending on their function, some opcodes produce no output, so they have no result value. Others take no arguments and only produce a result. Every orchestra statement must be on a single line, however long lines can be wrapped to a new line using the '\' character. This character indicates that the next line is part of the current one, this way you can split a line for easier reading, like this: a2 oscbnk kcps, 1.0, kfmd1, 0.0, 40, 203, 0.1, 0.2, kamfr, kamfr2, 148, \ 0, 0, 0, 0, 0, 0, -1, \ kfnum, 3, 4 Comments are optional and are for the purpose of letting the user document his orchestra code. Comments begin with a semicolon (;) and extend to the end of the line. Comments can optionally be in C-style, spanning multiple lines like this: /* Anything in here -------- is a comment which can span several lines --------- */ The remainder (result, opcode, and arguments) form the basic statement. This also is optional, i.e. a line may have only a label or comment or be entirely blank. If present, the basic statement must be complete on one line, and is terminated by a carriage return and line feed. The opcode determines the operation to be performed; it usually takes some number of input values (or arguments, with a maximum value of about 800); and it usually has a result field variable to which it sends output values at some fixed rate. There are four possible rates: once only, at orchestra setup time (effectively a permanent assignment) once at the beginning of each note (at initialization (init) time: i-rate) once every performance-time control loop (perf-time control rate, or k-rate) once each sound sample of every control loop (perf-time audio rate, or a-rate) &orchheader; &orchiblock; &orchstatemnt; &orchkvar; &orchexpress; &orchdirfiles; &orchnomen; &orchmacros; &orchnamedinstruments; &orchUDO; &orchtab; &oviewfunctional; manual_src~dfsg/orch/express.xml0000644000000000000000000000210312262561504016107 0ustar rootroot
Expressions Expressions may be composed to any depth. Each part of an expression is evaluated at its own proper rate. For instance, if the terms within a sub-expression all change at the control rate or slower, the sub-expression will be evaluated only at the control rate; that result might then be used in an audio-rate evaluation. For example, in k1 + abs(int(p5) + frac(p5) * 100/12 + sqrt(k1)) the 100/12 would be evaluated at orch init, the p5 expressions evaluated at note i-time, and the remainder of the expression evaluated every k-period. The whole might occur in a unit generator argument position, or be part of an assignment statement.
manual_src~dfsg/orch/namedInstruments.xml0000644000000000000000000001646512262561504017776 0ustar rootroot
Named Instruments As a recent addition to the orchestra syntax, instruments can be defined with string names. Such named instruments are callable from the score, and are supported by a number of opcodes. Syntax A named instrument is declared as shown below: instr Name[, Name2[, Name3[, ...]]] [...] endin A single instrument can have any number of names, and any of these names can be used to call the instrument. Additionally, it is possible to use numbers as name, denoting a standard numbered instrument, so the following declaration is also valid: instr 100, Name1, 99, Name2, 1, 2, 3 An instrument name may consist of any number of letters, digits, and the underscore (_) character, however, the first character must not be a digit. Optionally, the instrument name may be prefixed with the '+' character (see below), for example: instr +Reverb For all instrument names, a number is automatically assigned (note: if the message level (-m) is not zero, these numbers are printed to the console during orchestra compilation), following these rules: any unused instrument numbers are taken up in ascending order, starting from 1 the numbers are assigned in the order of instrument name definition, so named instruments that are defined later will always have a higher number (except if the '+' modifier is used) if the instrument name was prefixed with '+', the assigned number will be higher than that of any of the (both numbered and named) other instruments without '+'. If there are multiple '+' instruments, the numbering of these will follow the order of definition, according to the above rule. Using '+' is mainly useful for global output or effect instruments, that must be performed after the other instruments. An example for instrument numbers: instr 1, 2 endin instr Instr1 endin instr +Effect1, Instr2 endin instr 100, Instr3, +Effect2, Instr4, 5 endin In this example, the instrument numbers are assigned as follows: Instr1: 3 Effect1: 101 Instr2: 4 Instr3: 6 Effect2: 102 Instr4: 7 Using Named Instruments Named instruments can be called by using the name in double quotes as the instrument number (note: the '+' character should be omitted). Currently (as of Csound 4.22.4), named instruments are supported by: 'i' and 'q' score events Notes in score files, unmatched quotes, and spaces or other invalid characters in the strings should be avoided, otherwise (at least with current version) unpredictable behavior may occur (this problem does not exist for -L line events). However, there is checking for undefined instruments, and in such cases, the event is simply ignored with a warning. Stand-alone utilities (score sort and extract) do not support named instruments. It is still possible to sort such scores by using the -t0 option of the main Csound executable) real-time line events (-L) event, schedkwhen, subinstr, and subinstrinit opcodes massign, pgmassign, prealloc, and mute opcodes Additionaly, there is a new opcode (nstrnum) that returns the number of a named instrument: insno nstrnum "name" With the above example, nstrnum "Effect1" would return 101. If an instrument with the specified name does not exist, an init error occurs, and -1 is returned. Example ; ---- orchestra ---- sr = 44100 ksmps = 10 nchnls = 1 prealloc "SineWave", 20 prealloc "MIDISineWave", 20 massign 1, "MIDISineWave" gaOutSend init 0 instr +OutputInstr out gaOutSend clear gaOutSend endin instr SineWave a1 oscils p4, p5, 0 vincr gaOutSend, a1 endin instr MIDISineWave iamp veloc inote notnum icps = cpsoct(inote / 12 + 3) a1 oscils iamp * 100, icps, 0 vincr gaOutSend, a1 endin ; ---- score ---- i "SineWave" 0 2 12000 440 i "OutputInstr" 0 3 e Author Istvan Varga 2002
manual_src~dfsg/orch/header.xml0000644000000000000000000000543412262561504015660 0ustar rootroot
Orchestra Header Statements The Orchestra Header contains global information that applies to all instruments and defines aspects of Csound output. It is sometimes referred to as instr 0, because it behaves as an instrument, but without k- or a-rate processing (i.e. only opcodes and instructions that work at i-rate are allowed). An orchestra header statement operates once only, at orchestra setup time. It is most commonly an assignment of some value to a global reserved symbol , e.g. sr = 20000. All orchestra header statements belong to a pseudo instrument 0, an init pass of which is run prior to all other instruments at score time 0. Any ordinary statement can serve as an orchestra header statement, eg. gifreq = cpspch(8.09) provided it is an init-time only operation. Statements that are normally placed in an orchestra header are: 0dbfs ctrlinit ftgen kr ksmps massign nchnls pgmassign pset seed sr strset For example, a Csound header may look like: sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 1 massign 1, 10
manual_src~dfsg/orch/UDO.xml0000644000000000000000000000510612262561504015053 0ustar rootroot
User Defined Opcodes (UDO) Csound allows the definition of opcodes inside the orchestra header using the opcodes opcode and endop. The defined opcode may run with a different number of control samples (ksmps) using setksmps. To connect inputs and outputs for the UDO, use xin and xout. An UDO looks like this: opcode Lowpass, a, akk setksmps 1 ; need sr=kr ain, ka1, ka2 xin ; read input parameters aout init 0 ; initialize output aout = ain*ka1 + aout*ka2 ; simple tone-like filter xout aout ; write output endop This UDO called Lowpass takes 3 inputs (the first is a-rate, and the next two are k-rate), and delivers 1 a-rate output. Notice the use of xin to receive inputs and xout to deliver outputs. Also note the use of setksmps, which is needed for the filter to work properly. To use this UDO within an instrument, you would do something like: afiltered Lowpass asource, kvalue1, kvalue2 See the entry for opcode for detailed information on UDO definition. You can find many ready made UDO's (or contribute your own) at Csounds.com's User Defined Opcode Database.
manual_src~dfsg/examples/0000755000000000000000000000000012270515241014557 5ustar rootrootmanual_src~dfsg/examples/checkbox.csd0000644000000000000000000000116412262561504017046 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o checkbox.wav -W ;;; for file output any platform sr = 44100 ksmps = 10 nchnls = 2 instr 1 ; Get the value from the checkbox. k1 checkbox 1 ; If the checkbox is selected then k2=440, otherwise k2=880. k2 = (k1 == 0 ? 440 : 880) a1 oscil 10000, k2, 1 outs a1, a1 endin ; sine wave. f 1 0 32768 10 1 i 1 0 10 e manual_src~dfsg/examples/opand.csd0000644000000000000000000000136512262561504016364 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o opand.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kval randomh 0, 1.2, 20 ;choose between 0 and 1.2 if kval >0 && kval<=.5 then ;3 possible outcomes kval = 1 elseif kval >.5 && kval<=1 then kval =2 elseif kval >1 then kval =3 endif printk2 kval ;print value when it changes asig poscil .7, 440*kval, 1 outs asig, asig endin f1 0 16384 10 1 i 1 0 5 e manual_src~dfsg/examples/sqrt.csd0000644000000000000000000000131412262561504016246 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sqrt.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 asig pluck 0.7, 55, 55, 0, 1 kpan line 0,p3,1 kleft = sqrt(1-kpan) kright = sqrt(kpan) printks "square root of left channel = %f\\n", 1, kleft ;show coarse of sqaure root values outs asig*kleft, asig*kright ;where 0.707126 is between 2 speakers endin i 1 0 10 e manual_src~dfsg/examples/vdelay3.csd0000644000000000000000000000133212262561504016624 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelay3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ims = 100 ;maximum delay time in msec aout poscil .8, 220, 1 ;make a signal a2 poscil ims/2, 1/p3, 1 ;make an LFO a2 = a2 + ims/2 ;offset the LFO so that it is positive asig vdelay3 aout, a2, ims ;use the LFO to control delay time outs asig, asig endin f1 0 8192 10 1 i 1 0 5 e manual_src~dfsg/examples/FLcount.csd0000644000000000000000000000326312262561504016634 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLcount.wav -W ;;; for file output any platform ; Demonstration of the flcount opcode ; clicking on the single arrow buttons ; increments the oscillator in semitone steps ; clicking on the double arrow buttons ; increments the oscillator in octave steps sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Counter", 900, 400, 50, 50 ; Minimum value output by counter imin = 6 ; Maximum value output by counter imax = 12 ; Single arrow step size (semitones) istep1 = 1/12 ; Double arrow step size (octave) istep2 = 1 ; Counter type (1=double arrow counter) itype = 1 ; Width of the counter in pixels iwidth = 200 ; Height of the counter in pixels iheight = 30 ; Distance of the left edge of the counter ; from the left edge of the panel ix = 50 ; Distance of the top edge of the counter ; from the top edge of the panel iy = 50 ; Score event type (-1=ignored) iopcode = -1 gkoct, ihandle FLcount "pitch in oct format", imin, imax, istep1, istep2, itype, iwidth, iheight, ix, iy, iopcode, 1, 0, 1 ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 iamp = 15000 ifn = 1 asig oscili iamp, cpsoct(gkoct), ifn out asig endin ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e manual_src~dfsg/examples/hvs2.csd0000644000000000000000000000401512262561504016140 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O sr=44100 ksmps=100 nchnls=2 0dbfs = 1 ginumLinesX init 4 ginumLinesY init 4 ginumParms init 3 giOutTab ftgen 5,0,8, -2, 0 giPosTab ftgen 6,0,32, -2, 3,2,1,0,4,5,6,7,8,9,10, 11, 15, 14, 13, 12 giSnapTab ftgen 8,0,64, -2, 1,1,1, 2,0,0, 3,2,0, 2,2,2, 5,2,1, 2,3,4, 6,1,7, 0,0,0, \ 1,3,5, 3,4,4, 1,5,8, 1,1,5, 4,3,2, 3,4,5, 7,6,5, 7,8,9 tb0_init giOutTab FLpanel "Prova HVS2",600,400,10,100,0 gk1, gk2, ih1, ih2 FLjoy "HVS controller XY", 0, 1, 1, 0, 0, 0, -1, -1, 300, 300, 0, 50 ; *ihandle, *numlinesX, *numlinesY, *iwidth, *iheight, *ix, *iy,*image; gihandle FLhvsBox ginumLinesX, ginumLinesY, 300, 300, 300, 50, 1 FLpanel_end FLrun instr 1 ; Smooth control signals to avoid clicks kx portk gk1, 0.02 ky portk gk2, 0.02 ; kx, ky, inumParms, inumlinesX, inumlinesY, iOutTab, iPosTab, iSnapTab [, iConfigTab] hvs2 kx, ky, ginumParms, ginumLinesX, ginumLinesY, giOutTab, giPosTab, giSnapTab ;, iConfigTab ; *kx, *ky, *ihandle; FLhvsBoxSetValue gk1, gk2, gihandle k0 init 0 k1 init 1 k2 init 2 printk2 tb0(k0) printk2 tb0(k1), 10 printk2 tb0(k2), 20 kris init 0.003 kdur init 0.02 kdec init 0.007 ; Make parameters of synthesis depend on the table values produced by hvs ares1 fof 0.2, tb0(k0)*100 + 50, tb0(k1)*100 + 200, 0, tb0(k2) * 10 + 50, 0.003, 0.02, 0.007, 20, \ 1, 2, p3 ares2 fof 0.2, tb0(k1)*100 + 50, tb0(k2)*100 + 200, 0, tb0(k0) * 10 + 50, 0.003, 0.02, 0.007, 20, \ 1, 2, p3 outs ares1, ares2 endin f 1 0 1024 10 1 ;Sine wave f 2 0 1024 19 0.5 0.5 270 0.5 ;Grain envelope table f0 3600 i1 0 3600 manual_src~dfsg/examples/copya2ftab.csd0000644000000000000000000000226112262561504017311 0ustar rootroot ; Select audio/midi flags here according to platform -n sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;an 'empty' function table with 10 points giTable ftgen 0, 0, -10, 2, 0 instr 1 ;print inital values of giTable puts "\nInitial table content:", 1 indx = 0 until indx == ftlen(giTable) do iVal table indx, giTable printf_i "Table index %d = %f\n", 1, indx, iVal indx += 1 od ;create array kArr[] init 10 ;fill in values kArr genarray 1, 10 ;print array values printf "%s", 1, "\nArray content:\n" kndx = 0 until kndx == lenarray(kArr) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr[kndx] kndx += 1 od ;copy array values to table copya2ftab kArr, giTable ;print modified values of giTable printf "%s", 1, "\nModified table content after copya2ftab:\n" kndx = 0 until kndx == ftlen(giTable) do kVal table kndx, giTable printf "Table index %d = %f\n", kndx+1, kndx, kVal kndx += 1 od ;turn instrument off turnoff endin i 1 0 0.1 manual_src~dfsg/examples/kickroll.het0000644000000000000000000004733212262561504017110 0ustar rootroot  6 o6RfSPZ>?!M$C(B+5.(2#5P"8k<?;Bv8FQI[LKSPB_SZAWYZ<]TIa8d/g$k'nfquBAxf;{W9Z?M"aAU<M92b'( l"WB R 1H7QX`bTZ\;kOC"J1)( @(?q# &) -! 03 7:2 > A Da H K-N5RPU]XU\ \_@bNfn@i Jlq1p,s*vzP}Ms(k   3 BV   l  )6hM%ZAYZBM@K1/V,u@ 6    <|  s  !$$7(&J+V/%]2,Y5E9K<yA?rKC1F1I,MjPASWZ]ad g7 k? nqu x| ?     8 FXT`,WITI$CK33,j1:   )M  } @ vt9ARdT MFbE(J55 +#&3*-047{:K> Ad DhHK NR U !$t(z+.r25a8o<?$BFILPSWZh]aldxgkkqnbqux{ej|gm]A!> <3}l~z QhCX$#&*):-(0337[:>ADyH%KNRUX\_bfqil{p}svMzp}5o"=35ulTDww~Lv0vP+?  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BMFTI}LPStW6Z]'aldgkn q u xU { Krrv YI/ a  - ) S E w PB"+>  =  # &- )6 -$ 0 3u 7z: > A D H K N RUXe\N_bfilppsV vz }  sC ) 9 -  Z  &     yWS  / r{ ]& < 3  C #y W >  ! $ (+/259< ?CFI M&PX S WZ]H agdg# k> n6 q u5 x|       s gQXu   "a u @ 8  5 O       5&7Z  #&l * -04Y 7:> A@ D8 H KG NR;U;!  z!$(.s2o58i<?nBFILPSWZ]adgyknqY ux{R_`KH{iRSL  r{ ~VWA#W&_)-0D3N7ADUH KTNRUX\_bfilpvsvzm}Z`VdO |{s[{gWg P CN" !$(+/259<?CFIMtPzS\WsZ]oadWgikquRxE|~n\mvXiSGV  #j&*]-g04|7:W>gAHKSNHR   t   !$(g+\.258 < ? 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CaF4IMG P SW Z ] a g k n qu x |\  T .4      gs  @    M  d     h P 2 I     [> # &k *d- 0 4 76 A% D HlK[ NL R U  v !$(+s.a2B5`8 < ?BFeI>LPSWZ]adlgWknq ux{8$z"a|jfiO^ICFYOFMO BmvZM TrPK\#-0B37:F>ADcH_ KYNRUX\_bJfilphsMvMzk}[EKZCI B)jRIl\DLX{N L!H$ (+/259 <?CFlIUMDPlS\WDZM]WavdZgx|MtJ*nXAk\FOUqfL|haU o Z#&i*]-I0O4S7l:p>NRJU`  9  !$U( +Z.H2 58 <?BFIKLP S WZ ]Q adgk(n7qux{F o  r 'VH   G G   ;+ :" 9p#W- 0W37 :>A1DHKNRUFX\ _ bfDi l pCsvz } 5  0j @A<]LF Y  eE      !d3 R R @!$@(+S/2 5 9C <~? C FwIMP) S1 W Z ] a^dg x |A\ > %] 3  O  Mv  *  Y0 k   89i' }    R X#& * -x 0 47U:P> NR U manual_src~dfsg/examples/ftmorf.csd0000644000000000000000000000143112262561504016552 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftmorf.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 instr 1 kndx line 0, p3, 7 ftmorf kndx, 1, 2 asig oscili .8, 440, 2 outs asig, asig endin f1 0 8 -2 3 4 5 6 7 8 9 10 f2 0 1024 10 1 /*contents of f2 dont matter */ f3 0 1024 10 1 f4 0 1024 10 0 1 f5 0 1024 10 0 0 1 f6 0 1024 10 0 0 0 1 f7 0 1024 10 0 0 0 0 1 f8 0 1024 10 0 0 0 0 0 1 f9 0 1024 10 0 0 0 0 0 0 1 f10 0 1024 10 1 1 1 1 1 1 1 i1 0 15 e manual_src~dfsg/examples/moscil.csd0000644000000000000000000000132712262561504016547 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1;;;RT audio I/O with MIDI in sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 instr 1 ;Triggered by MIDI notes on channel 1 inote notnum ivel veloc kpitch = 40 kfreq = 2 kdur = .04 kpause = .1 k1 lfo kpitch, kfreq,5 ;plays a stream of notes of kdur duration on MIDI channel 1 moscil 1, inote + k1, ivel, kdur, kpause endin ; Dummy ftable f0 60 manual_src~dfsg/examples/STKPlucked.csd0000644000000000000000000000107512262561504017232 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKPlucked.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;STKPlucked - has no controllers ifrq = p4 asig STKPlucked cpspch(ifrq), 1 outs asig, asig endin i 1 0 2 6.00 i 1 + 8 5.00 i 1 + .5 8.00 e manual_src~dfsg/examples/outleta.csd0000644000000000000000000000156612262561504016743 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outleta.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 connect "1", "Outl", "reverby", "InL" connect "1", "Outr", "reverby", "InR" alwayson "reverby", 1 instr 1 aIn diskin2 "fox.wav", 1 outleta "Outl", aIn outleta "Outr", aIn endin instr reverby aInL inleta "InL" aInR inleta "InR" al, ar reverbsc aInL, aInR, 0.7, 21000 ifxlev = 0.5 al = (aInL*ifxlev)+(al*(1-ifxlev)) ar = (aInR*ifxlev)+(ar*(1-ifxlev)) outs al, ar endin i 1 0 3 e4 manual_src~dfsg/examples/assign.csd0000644000000000000000000000106712262561504016546 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o assign.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 ; Assign a value to the variable i1. i1 = 1234 ; Print the value of the i1 variable. print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/loop_lt.csd0000644000000000000000000000162612262561504016733 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loop_lt.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 gisine ftgen 0, 0, 2^10, 10, 1 instr 1 ;master instrument ininstr = 5 ;number of called instances indx = 0 loop: prints "play instance %d\\n", indx ipan random 0, 1 ifreq random 100, 1000 iamp = 1/ininstr event_i "i", 10, 0, p3, iamp, ifreq, ipan loop_lt indx, 1, ininstr, loop endin instr 10 ipeak random 0, 1 ;where is the envelope peak asig poscil3 p4, p5, gisine aenv transeg 0, p3*ipeak, 6, 1, p3-p3*ipeak, -6, 0 aL,aR pan2 asig*aenv, p6 outs aL, aR endin i1 0 10 e manual_src~dfsg/examples/octave.csd0000644000000000000000000000141112262561504016534 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o octave.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iroot = 440 ; root note is A above middle-C (440 Hz) koct lfo 5, 1, 5 ; generate sawtooth, go from 5 octaves higher to root koc = int(koct) ; produce only whole numbers kfactor = octave(koc) ; for octave knew = iroot * kfactor printk2 knew asig pluck 1, knew, 1000, 0, 1 asig dcblock asig ;remove DC outs asig, asig endin i 1 0 5 e manual_src~dfsg/examples/pvsfwrite.csd0000644000000000000000000000132012262561504017303 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsfwrite.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ;By Victor Lazzarini 2008 instr 1 asig oscili 10000, 440, 1 fss pvsanal asig, 1024,256,1024,0 pvsfwrite fss, "mypvs.pvx" ase pvsynth fss out ase endin instr 2 ; must be called after instr 1 finishes ktim timeinsts fss pvsfread ktim, "mypvs.pvx" asig pvsynth fss out asig endin f1 0 16384 10 1 i1 0 1 i2 1 1 e manual_src~dfsg/examples/hvs2-2.csd0000644000000000000000000000621312262561504016301 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o hvs2.wav -W ;;; for file output any platform sr=48000 ksmps=100 nchnls=2 ; Example by James Hearon 2008 ; Edited by Andres Cabrera ginumPointsX init 16 ginumPointsY init 16 ginumParms init 3 ;Generate 9 tables with arbitrary points gitmp ftgen 100, 0, 16, -2, 70, 260, 390, 180, 200, 300, 980, 126, \ 330, 860, 580, 467, 220, 399, 1026, 1500 gitmp ftgen 200, 0, 16, -2, 100, 200, 300, 140, 600, 700, 880, 126, \ 330, 560, 780, 167, 220, 999, 1026, 1500 gitmp ftgen 300, 0, 16, -2, 400, 200, 300, 540, 600, 700, 880, 126, \ 330, 160, 780, 167, 820, 999, 1026, 1500 gitmp ftgen 400, 0, 16, -2, 100, 200, 800, 640, 600, 300, 880, 126, \ 330, 660, 780, 167, 220, 999, 1026, 1500 gitmp ftgen 500, 0, 16, -2, 200, 200, 360, 440, 600, 700, 880, 126, \ 330, 560, 380, 167, 220, 499, 1026, 1500 gitmp ftgen 600, 0, 16, -2, 100, 600, 300, 840, 600, 700, 880, 126, \ 330, 260, 980, 367, 120, 399, 1026, 1500 gitmp ftgen 700, 0, 16, -2, 100, 200, 300, 340, 200, 500, 380, 126, \ 330, 860, 780, 867, 120, 999, 1026, 1500 gitmp ftgen 800, 0, 16, -2, 100, 600, 300, 240, 200, 700, 880, 126, \ 130, 560, 980, 167, 220, 499, 1026, 1500 gitmp ftgen 900, 0, 16, -2, 100, 800, 200, 140, 600, 700, 680, 126, \ 330, 560, 780, 167, 120, 299, 1026, 1500 giOutTab ftgen 5,0,8, -2, 0 giPosTab ftgen 6,0,32, -2, 0,1,2,3,4,5,6,7,8,9,10, 11, 15, 14, 13, 12 giSnapTab ftgen 8,0,64, -2, 1,1,1, 2,0,0, 3,2,0, 2,2,2, \ 5,2,1, 2,3,4, 6,1,7, 0,0,0, 1,3,5, 3,4,4, 1,5,8, 1,1,5, \ 4,3,2, 3,4,5, 7,6,5, 7,8,9 tb0_init giOutTab FLpanel "hsv2",440,100,10,10,0 gk1,ih1 FLslider "X", 0,1, 0, 5, -1, 400,20, 20,10 gk2, ih2 FLslider "Y", 0, 1, 0, 5, -1, 400, 20, 20, 50 FLpanel_end FLpanel "hvsBox",280,280,500,1000,0 ;ihandle FLhvsBox inumlinesX, inumlinesY, iwidth, iheight, ix, iy [, image] gih1 FLhvsBox 16, 16, 250, 250, 10, 1 FLpanel_end FLrun instr 1 FLhvsBoxSetValue gk1, gk2, gih1 hvs2 gk1,gk2, ginumParms, ginumPointsX, ginumPointsY, giOutTab, giPosTab, giSnapTab ;, iConfigTab k0 init 0 k1 init 1 k2 init 2 kspeed init 0 kspeed = int((tb0(k2)) + 1)*.10 kenv oscil 25000, kspeed*16, 10 k1 phasor kspeed ;slow phasor: 200 sec. kpch tableikt k1 * 16, int((tb0(k1)) +1)*100 ;scale phasor * length a1 oscilikt kenv, kpch, int(tb0(k0)) +1000;scale pitch slightly ahp butterlp a1, 2500 outs ahp, ahp endin f 10 0 1024 20 5 ;use of windowing function f1000 0 1024 10 .33 .25 .5 f1001 0 1024 10 1 f1002 0 1024 10 .5 .25 .05 f1003 0 1024 10 .05 .10 .3 .5 1 f1004 0 1024 10 1 .5 .25 .125 .625 f1005 0 1024 10 .33 .44 .55 .66 f1006 0 1024 10 1 1 1 1 1 f1007 0 1024 10 .05 .25 .05 .25 .05 1 f0 3600 i1 0 3600 manual_src~dfsg/examples/scoreline.csd0000644000000000000000000000137312262561504017245 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o scoreline.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktrig metro 3 ;trigger 3 times a second scoreline {{ ;so it sounds like an echo i 2 0 3 "flute.aiff" i 2 1 3 "beats.wav" }}, ktrig ktrig = 0 endin instr 2 asig soundin p4 outs asig*.3, asig*.3 endin i1 0 2 ;play for 2 seconds, so the samples are played 6 times e manual_src~dfsg/examples/readf.csd0000644000000000000000000000074412262561504016344 0ustar rootroot -n instr 1 Swd pwd printf_i "Working directory is '%s'\n", 1, Swd prints "Reading myself =):\n" read: Sline, kLinNum readf "readf.csd" printf "Line %d: %s", kLinNum, kLinNum, Sline if kLinNum != -1 then kgoto read else turnoff endif endin i1 0 1 e manual_src~dfsg/examples/octmidi.csd0000644000000000000000000000122012262561504016701 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o octmidi.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; This example expects MIDI note inputs on channel 1 i1 octmidi print i1 endin ;Dummy f-table to give time for real-time MIDI events f 0 8000 e manual_src~dfsg/examples/pvspitch.csd0000644000000000000000000000146712262561504017126 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvspitch.wav -W ;;; for file output any platform sr = 44100 ksmps = 128 nchnls = 1 giwave ftgen 0, 0, 4096, 10, 1, 0.5, 0.333, 0.25, 0.2, 0.1666 instr 1 ifftsize = 1024 iwtype = 1 /* cleaner with hanning window */ a1 inch 1 ;Realtime audio input ;a1 soundin "input.wav" ;Use this line for file input fsig pvsanal a1, ifftsize, ifftsize/4, ifftsize, iwtype kfr, kamp pvspitch fsig, 0.01 adm oscil kamp, kfr * 1.5, giwave ;Generate note a fifth above detected pitch out adm endin i 1 0 30 e manual_src~dfsg/examples/midinoteoff.csd0000644000000000000000000000144612262561504017566 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M1 -Q1 ;;;realtime audio out and midi in and midi out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midinoteoff.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;uses external midi device kkey init 0 ;initialize key number kvel init 0 ;initialize velocity midinoteoff kkey,kvel ;MIDI noteoff value printk2 kvel ;display noteoff value midion 1, kkey, kvel ;sent note to external device endin f 0 30 ;runs for 30 seconds i 1 0 2 62 ; play these notes from score as well i 1 + 2 65 e manual_src~dfsg/examples/cpsmidi.csd0000644000000000000000000000104512262561504016706 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M0 ;;;RT audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpsmidi.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi asig oscil 0.6, icps, 1 print icps outs asig, asig endin f0 20 ;sine wave. f 1 0 16384 10 1 e manual_src~dfsg/examples/butterbp.csd0000644000000000000000000000116612262561504017111 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o butterbp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; White noise signal asig rand 0.6 outs asig, asig endin instr 2 ;filtered noise asig rand 1 abp butterbp asig, 2000, 100 ;passing only 1950 to 2050 Hz outs abp, abp endin i 1 0 2 i 2 2.5 2 e manual_src~dfsg/examples/exprand.csd0000644000000000000000000000142012262561504016714 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o exprand.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values klamda exprand 20 printk .2, klamda ; look aout oscili 0.8, 440+klamda, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 klamda exprand 20 printk .2, klamda ; look aout oscili 0.8, 440+klamda, 1 ; & listen outs aout, aout endin ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/syncphasor-CZresonance.csd0000644000000000000000000000755512262561504021673 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o syncphasor-CZresonance.wav -W ;;; for file output any platform ; by Anthony Kozar. February 2008 ; http://www.anthonykozar.net/ ; Imitation of the Casio CZ-series synthesizer's "Resonance" waveforms ; using a synced phasor to read a sinusoid table. The jumps at the sync ; points are smoothed by multiplying with a windowing function controlled ; by the master phasor. ; Based on information from the Wikipedia article on phase distortion: ; http://en.wikipedia.org/wiki/Phase_distortion_synthesis ; Sawtooth Resonance waveform. Smoothing function is just the inverted ; master phasor. ; The Wikipedia article shows an inverted cosine as the stored waveform, ; which implies that it must be unipolar for the smoothing to work. ; I have substituted a sine wave in the first phrase to keep the output ; bipolar. The second phrase demonstrates the much "rezzier" sound of the ; bipolar cosine due to discontinuities. instr 1 ifreq = cpspch(p4) initReson = p5 itable = p6 imaxamp = 10000 anosync init 0.0 kslavecps line ifreq * initReson, p3, ifreq amaster, async syncphasor ifreq, anosync ; pair of phasors aslave, async2 syncphasor kslavecps, async ; slave synced to master aosc tablei aslave, itable, 1 ; use slave phasor to read a (co)sine table aout = aosc * (1.0 - amaster) ; inverted master smoothes jumps adeclick linseg 0.0, 0.05, 1.0, p3 - 0.1, 1.0, 0.05, 0.0 out aout * adeclick * imaxamp endin ; Triangle or Trapezoidal Resonance waveform. Uses a second table to change ; the shape of the smoothing function. (This is my best guess so far as to ; how these worked). The cosine table works fine with the triangular smoothing ; but we once again need to use a sine table with the trapezoidal smoothing. ; (It might be interesting to be able to vary the "width" of the trapezoid. ; This could be done with the pdhalf opcode). instr 2 ifreq = cpspch(p4) initReson = p5 itable = p6 ismoothtbl = p7 imaxamp = 10000 anosync init 0.0 kslavecps line ifreq * initReson, p3, ifreq amaster, async syncphasor ifreq, anosync ; pair of phasors aslave, async2 syncphasor kslavecps, async ; slave synced to master aosc tablei aslave, itable, 1 ; use slave phasor to read a (co)sine table asmooth tablei amaster, ismoothtbl, 1 ; use master phasor to read smoothing table aout = aosc * asmooth adeclick linseg 0.0, 0.05, 1.0, p3 - 0.1, 1.0, 0.05, 0.0 out aout * adeclick * imaxamp endin f1 0 16385 10 1 f3 0 16385 9 1 1 270 ; inverted cosine f5 0 4097 7 0.0 2048 1.0 2049 0.0 ; unipolar triangle f6 0 4097 7 1.0 2048 1.0 2049 0.0 ; "trapezoid" ; Sawtooth resonance with a sine table i1 0 1 7.00 5.0 1 i. + 0.5 7.02 4.0 i. + . 7.05 3.0 i. + . 7.07 2.0 i. + . 7.09 1.0 i. + 2 7.06 12.0 f0 6 s ; Sawtooth resonance with a cosine table i1 0 1 7.00 5.0 3 i. + 0.5 7.02 4.0 i. + . 7.05 3.0 i. + . 7.07 2.0 i. + . 7.09 1.0 i. + 2 7.06 12.0 f0 6 s ; Triangle resonance with a cosine table i2 0 1 7.00 5.0 3 5 i. + 0.5 7.02 4.0 i. + . 7.05 3.0 i. + . 7.07 2.0 i. + . 7.09 1.0 i. + 2 7.06 12.0 f0 6 s ; Trapezoidal resonance with a sine table i2 0 1 7.00 5.0 1 6 i. + 0.5 7.02 4.0 i. + . 7.05 3.0 i. + . 7.07 2.0 i. + . 7.09 1.0 i. + 2 7.06 12.0 e manual_src~dfsg/examples/vosim.csd0000644000000000000000000000730312262561504016416 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vosim.wav -W ;;; for file output any platform sr = 44100 ksmps = 100 nchnls = 1 ;################################################# ; By Rasmus Ekman 2008 ; Square each point in table #p4. This should only be run once in the performance. instr 10 index tableng p4 index = index - 1 ; start from last point loop: ival table index, p4 ival = ival * ival tableiw ival, index, p4 index = index - 1 if index < 0 igoto endloop igoto loop endloop: endin ;################################################# ; Main vosim instrument. Sweeps from a fund1/form1 to fund2/form2, ; trying for narrowest formant bandwidth (still quite wide by the looks of it) ; p4: amp ; p5, p6: fund beg-end ; p7, p8: form beg-end ; p9: amp decay (ignored) ; p10: pulse count (ignored - calc internally) ; p11: pulse length mod ; p12: skip (for tied events) ; p13: don't fade out (if followed by tied note) instr 1 kamp init p4 ; freq start, end kfund line p5, p3, p6 ; formant start, end kform line p7, p3, p8 ; Try for constant ratio burst/silence, and narrowest formant bandwidth kPulseCount = (kform / kfund) ;init p10 ; Attempt to smooth steps between format bandwidths, ; increasing decay before we are forced to a lower pulse count kDecay = kPulseCount/(kform % kfund) ; init p9 if (kDecay * kPulseCount) > kamp then kDecay = kamp / kPulseCount endif kDecay = 0.3 * kDecay kPulseFactor init p11 ; ar vosim kamp, kFund, kForm, kDecay, kPulseCount, kPulseFactor, ifn [, iskip] ar1 vosim kamp, kfund, kform, kDecay, kPulseCount, kPulseFactor, 17, p12 ; scale amplitude for 16-bit files, with quick fade out amp init 20000 if (p13 != 0) goto nofade amp linseg 20000, p3-.02, 20000, .02, 0 nofade: out ar1 * amp endin f1 0 32768 9 1 1 0 ; sine wave f17 0 32768 9 0.5 1 0 ; half sine wave i10 0 0 17 ; init run only, square table 17 ; Vosim score ; Picking some formants from the table in Csound manual ; p4=amp fund form decay pulses pulsemod [skip] nofade ; tenor a -> e i1 0 .5 .5 280 240 650 400 .03 5 1 i1 . . .3 . . 1080 1700 .03 5 . i1 . . .2 . . 2650 2600 .03 5 . i1 . . .15 . . 2900 3200 .03 5 . ; tenor a -> o i1 0.6 .2 .5 300 210 650 400 .03 5 1 0 1 i1 . . .3 . . 1080 800 .03 5 . . . i1 . . .2 . . 2650 2600 .03 5 . . . i1 . . .15 . . 2900 2800 .03 5 . . . ; tenor o -> aah i1 .8 .3 .5 210 180 400 650 .03 5 1 1 1 i1 . . .3 . . 800 1080 .03 5 . . . i1 . . .2 . . 2600 2650 .03 5 . . . i1 . . .15 . . 2800 2900 .03 5 . . . ; tenor aa -> i i1 1.1 .2 .5 180 250 650 290 .03 5 1 1 1 i1 . . .3 . . 1080 1870 .03 5 . . . i1 . . .2 . . 2650 2800 .03 5 . . . i1 . . .15 . . 2900 3250 .03 5 . . . ; tenor i -> u i1 1.3 .3 .5 250 270 290 350 .03 5 1 1 0 i1 . . .3 . . 1870 600 .03 5 . . . i1 . . .2 . . 2800 2700 .03 5 . . . i1 . . .15 . . 3250 2900 .03 5 . . . e manual_src~dfsg/examples/tableikt.csd0000644000000000000000000000270012262561504017054 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tableikt.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ilowfn = p4 ;lowest ftable wave ihighfn = p5 ;highest ftable wave kswpenv line 1, p3, 0 ;sweep envelope, calculate current table pair and interpolation amount inumtables = ihighfn - ilowfn ;1 less than number of tables kfn1 = int(kswpenv*inumtables) + ilowfn printks "play table no: %d\n", 1, kfn1 kfn2 = kfn1 + 1 kinterp = frac(kswpenv*inumtables) ixmode = 1 ;read tables with phasor aphase phasor 40 asig tableikt aphase, kfn1, ixmode ;normalized index if kswpenv == 1.0 kgoto skipfn2 ;if kfn1 is last table, there is no kfn2 asig2 tableikt aphase, kfn2, ixmode skipfn2: amix ntrpol asig, asig2, kinterp ;interpolate between tables and output outs amix*.5, amix*.5 endin f 1 0 16384 10 1 f 2 0 16384 10 1 .5 f 3 0 16384 1 "fox.wav" 0 0 0 ;a sample f 4 0 16384 10 1 .5 .3 .25 .2 .16 .14 .125 .111 ;sawtooth f 5 0 16384 10 1 .4 .3 .25 .2 f 6 0 16384 10 1 .3 .3 .25 .2 .16 f 7 0 16384 10 1 1 1 1 .7 .5 .3 .1 ;pulse f 8 0 16384 1 "beats.wav" 0 0 0 ;a sample i 1 0 10 1 8 e manual_src~dfsg/examples/opcode_example.csd0000644000000000000000000000407512262561504020250 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o opcode_example.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 /* example opcode 1: simple oscillator */ opcode Oscillator, a, kk kamp, kcps xin ; read input parameters a1 vco2 kamp, kcps ; sawtooth oscillator xout a1 ; write output endop /* example opcode 2: lowpass filter with local ksmps */ opcode Lowpass, a, akk setksmps 1 ; need sr=kr ain, ka1, ka2 xin ; read input parameters aout init 0 ; initialize output aout = ain*ka1 + aout*ka2 ; simple tone-like filter xout aout ; write output endop /* example opcode 3: recursive call */ opcode RecursiveLowpass, a, akkpp ain, ka1, ka2, idep, icnt xin ; read input parameters if (icnt >= idep) goto skip1 ; check if max depth reached ain RecursiveLowpass ain, ka1, ka2, idep, icnt + 1 skip1: aout Lowpass ain, ka1, ka2 ; call filter xout aout ; write output endop /* example opcode 4: de-click envelope */ opcode DeClick, a, a ain xin aenv linseg 0, 0.02, 1, p3 - 0.05, 1, 0.02, 0, 0.01, 0 xout ain * aenv ; apply envelope and write output endop /* instr 1 uses the example opcodes */ instr 1 kamp = .7 ; amplitude kcps expon 50, p3, 500 ; pitch a1 Oscillator kamp, kcps ; call oscillator kflt linseg 0.4, 1.5, 0.4, 1, 0.8, 1.5, 0.8 ; filter envelope a1 RecursiveLowpass a1, kflt, 1 - kflt, 10 ; 10th order lowpass a1 DeClick a1 outs a1, a1 endin i 1 0 4 e5 ;extra second before quitting manual_src~dfsg/examples/monitor.csd0000644000000000000000000000201212262561504016740 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o monitor.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 asig poscil3 .5, 880, giSine ;write a raw file: 32 bits with header fout "fout_880.wav", 15, asig outs asig, asig endin instr 2 klfo lfo 1, 2, 0 asig poscil3 .5*klfo, 220, giSine ;write an aiff file: 32 bits with header fout "fout_aif.aiff", 25, asig ; fout "fout_all3.wav", 14, asig outs asig, asig endin instr 99 ;read the stereo csound output buffer allL, allR monitor ;write the output of csound to an audio file ;to a wav file: 16 bits with header fout "fout_all.wav", 14, allL, allR endin i 1 0 2 i 2 0 3 i 99 0 3 e manual_src~dfsg/examples/ATSinfo-2.csd0000644000000000000000000000154012262561504016720 0ustar rootroot ; Select audio/midi flags here according to platform -n ;;;no audio out ;example by joachim heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "fox.ats" has been created by ATSanal Sfile = "fox.ats" isr ATSinfo Sfile, 0 ifs ATSinfo Sfile, 1 iws ATSinfo Sfile, 2 inp ATSinfo Sfile, 3 inf ATSinfo Sfile, 4 ima ATSinfo Sfile, 5 imf ATSinfo Sfile, 6 id ATSinfo Sfile, 7 ift ATSinfo Sfile, 8 prints {{ Sample rate = %d Hz Frame Size = %d samples Window Size = %d samples Number of Partials = %d Number of Frames = %d Maximum Amplitude = %f Maximum Frequency = %f Hz Duration = %f seconds ATS file Type = %d }}, isr, ifs, iws, inp, inf, ima, imf, id, ift endin i 1 0 0 manual_src~dfsg/examples/ficlose.csd0000644000000000000000000000107312262561504016703 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ficlose.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gihand fiopen "test1.txt", 0 instr 1 ires random 0, 100 fouti gihand, 0, 1, ires ficlose gihand endin i 1 0 1 e manual_src~dfsg/examples/linsegb.csd0000644000000000000000000000154312262561504016704 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o linseg.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kcps = cpspch(p4) kenv linsegb 0, 0.25, 1, 1, 0 asig poscil kenv, kcps, giSine outs asig, asig endin instr 2 ; scaling to duration kcps = cpspch(p4) kenv linseg 0, p3*0.25, 1, p3, 0 asig poscil kenv, kcps, giSine outs asig, asig endin i 1 0 1 7.00 ; = 1 sec, p3 fits exactly i 1 2 2 7.00 ; = 2 sec, p3 truncated at 1 sec i 2 4 1 7.00 ; scales to duration i 2 6 2 7.00 ; of p3 e manual_src~dfsg/examples/inletk.csd0000644000000000000000000000137612262561504016553 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o inletk.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 connect "bend", "bendout", "guitar", "bendin" instr bend kbend line p4, p3, p5 outletk "bendout", kbend endin instr guitar kbend inletk "bendin" kpch pow 2, kbend/12 printk2 kpch asig oscili .4, 440*kpch, 1 outs asig, asig endin f1 0 1024 10 1 i"guitar" 0 5 8.00 i"bend" 3 .2 -12 12 i"bend" 4 .1 -17 40 e manual_src~dfsg/examples/m.csd0000644000000000000000000000130212262561504015506 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o m.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .8*aenv, p4, 1 outs asig, asig endin f 1 0 16384 10 1 ;sine wave m foo ;mark section i 1 0 1 110 i 1 1.5 1 220 i 1 3 1 440 i 1 4.5 1 880 s ;second section i 1 0 2 110 i 1 2 2 220 s n foo ;repeat marked section e manual_src~dfsg/examples/copyf2array.csd0000644000000000000000000000114012262561504017513 0ustar rootroot ; Select audio/midi flags here according to platform -n sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;8 points sine wave function table giSine ftgen 0, 0, 8, 10, 1 instr 1 ;create array kArr[] init 8 ;copy table values in it copyf2array kArr, giSine ;print values kndx = 0 until kndx == lenarray(kArr) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr[kndx] kndx += 1 od ;turn instrument off turnoff endin i 1 0 0.1 manual_src~dfsg/examples/metro-2.csd0000644000000000000000000000133112262561504016541 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o metro-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kpch random 1,20 ;produce values at k-rate ktrig metro 10 ;trigger 10 times per second kval samphold kpch, ktrig ;change value whenever ktrig = 1 asig buzz 1, 220, kval, 1;harmonics outs asig, asig endin f 1 0 4096 10 1 ; sine i 1 0 10 e manual_src~dfsg/examples/spat3d_quad.csd0000644000000000000000000000535512262561504017476 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o spat3d_quad.wav -W ;;; for file output any platform /* Written by Istvan Varga */ sr = 48000 kr = 1000 ksmps = 48 nchnls = 4 /* room parameters */ idep = 3 /* early reflection depth */ itmp ftgen 1, 0, 64, -2, \ /* depth1, depth2, max delay, IR length, idist, seed */ \ idep, 48, -1, 0.01, 0.25, 123, \ 1, 21.982, 0.05, 0.87, 4000.0, 0.6, 0.7, 2, /* ceil */ \ 1, 1.753, 0.05, 0.87, 3500.0, 0.5, 0.7, 2, /* floor */ \ 1, 15.220, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* front */ \ 1, 9.317, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* back */ \ 1, 17.545, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* right */ \ 1, 12.156, 0.05, 0.87, 5000.0, 0.8, 0.7, 2 /* left */ instr 1 /* some source signal */ a1 phasor 150 ; oscillator a1 butterbp a1, 500, 200 ; filter a1 = taninv(a1 * 100) a2 phasor 3 ; envelope a2 mirror 40*a2, -100, 5 a2 limit a2, 0, 1 a1 = a1 * a2 * 9000 kazim line 0, 2.5, 360 ; move sound source around kdist line 1, 10, 4 ; distance ; convert polar coordinates kX = sin(kazim * 3.14159 / 180) * kdist kY = cos(kazim * 3.14159 / 180) * kdist kZ = 0 a1 = a1 + 0.000001 * 0.000001 ; avoid underflows imode = 2 ; change this to 3 for 8 spk in a cube, ; or 1 for simple stereo aW, aX, aY, aZ spat3d a1, kX, kY, kZ, 1.0, 1, imode, 2, 2 aW = aW * 1.4142 ; stereo ; ;aL = aW + aY /* left */ ;aR = aW - aY /* right */ ; quad (square) ; aFL = aW + aX + aY /* front left */ aFR = aW + aX - aY /* front right */ aRL = aW - aX + aY /* rear left */ aRR = aW - aX - aY /* rear right */ ; eight channels (cube) ; ;aUFL = aW + aX + aY + aZ /* upper front left */ ;aUFR = aW + aX - aY + aZ /* upper front right */ ;aURL = aW - aX + aY + aZ /* upper rear left */ ;aURR = aW - aX - aY + aZ /* upper rear right */ ;aLFL = aW + aX + aY - aZ /* lower front left */ ;aLFR = aW + aX - aY - aZ /* lower front right */ ;aLRL = aW - aX + aY - aZ /* lower rear left */ ;aLRR = aW - aX - aY - aZ /* lower rear right */ outq aFL, aFR, aRL, aRR endin /* Written by Istvan Varga */ t 0 60 i 1 0 10 e manual_src~dfsg/examples/dumpk.csd0000644000000000000000000000152212262561504016376 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o dumpk.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 20 nchnls = 1 ; By Andres Cabrera 2008 instr 1 ; Write fibonacci numbers to file "fibonacci.txt" ; as ascii long integers (mode 7), using the orchestra's ; control rate (iprd = 0) knumber init 0 koldnumber init 1 ktrans init 1 ktrans = knumber knumber = knumber + koldnumber koldnumber = ktrans dumpk knumber, "fibonacci.txt", 7, 0 printk2 knumber endin ;Write to the file for 1 second. Since control rate is 20, 20 values will be written i 1 0 1 manual_src~dfsg/examples/STKRhodey.csd0000644000000000000000000000132212262561504017070 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKRhodey.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;(FM) Modulator Index One kv5 = p7 ;ADSR 2 and 4 target asig STKRhodey cpspch(p4), 1, 2, kv1, 4, 10, 11, 100, 1, 3, 128, kv5 outs asig, asig endin i 1 0 .5 7.00 75 0 0 i 1 + . 8.00 120 0 120 i 1 + 1 6.00 50 120 50 i 1 + 4 8.00 10 120 100 e manual_src~dfsg/examples/semitone.csd0000644000000000000000000000144112262561504017101 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o semitone.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iroot = 440 ; root note is A above middle-C (440 Hz) ksem lfo 12, .5, 5 ; generate sawtooth, go from 5 octaves higher to root ksm = int(ksem) ; produce only whole numbers kfactor = semitone(ksm) ; for semitones knew = iroot * kfactor printk2 knew printk2 kfactor asig pluck 1, knew, 1000, 0, 1 asig dcblock asig ;remove DC outs asig, asig endin i 1 0 5 e manual_src~dfsg/examples/FLxyin-2.csd0000644000000000000000000000151212262561504016625 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O sr=44100 kr=441 ksmps=100 nchnls=2 ; Example by Gabriel Maldonado FLpanel "Move the mouse inside this panel to hear the effect",400,400 FLpanel_end FLrun instr 1 k1, k2, kinside FLxyin 50, 1000, 50, 1000, 100, 300, 50, 250, -2,-3 ;if k1 <= 50 || k1 >=5000 || k2 <=100 || k2 >= 8000 kgoto end ; if cursor is outside bounds, then don't play!!! a1 oscili 3000, k1, 1 a2 oscili 3000, k2, 1 outs a1,a2 printk2 k1 printk2 k2, 10 printk2 kinside, 20 end: endin f1 0 1024 10 1 f2 0 17 19 1 1 90 1 f3 0 17 19 2 1 90 1 i1 0 3600 manual_src~dfsg/examples/vmultv.csd0000644000000000000000000000170512262561504016616 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ifn2 = p5 ielements = p6 idstoffset = p7 isrcoffset = p8 kval init 25 vmultv ifn1, ifn2, ielements, idstoffset, isrcoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 16 17 f 2 0 16 -7 1 16 2 i2 0.0 0.2 1 i2 0.2 0.2 2 i1 0.4 0.01 1 2 5 3 8 i2 0.8 0.2 1 i1 1.0 0.01 1 2 5 10 -2 i2 1.2 0.2 1 i1 1.4 0.01 1 2 8 14 0 i2 1.6 0.2 1 i1 1.8 0.01 1 2 8 0 14 i2 2.0 0.2 1 i1 2.2 0.002 1 1 8 5 2 i2 2.4 0.2 1 e manual_src~dfsg/examples/filebit.csd0000644000000000000000000000115012262561504016671 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o filebit.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the number of channels in the ; audio file "mary.wav". ibits filebit "mary.wav" print ibits endin ; Play Instrument #1 for 1 second. i 1 0 1 e manual_src~dfsg/examples/bamboo.csd0000644000000000000000000000073412262561504016521 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o bamboo.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 asig bamboo p4, 0.01 outs asig, asig endin i1 0 1 20000 e manual_src~dfsg/examples/outkpc_fltk.csd0000644000000000000000000000114512262561504017604 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1;;;RT audio I/O with MIDI in sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 FLpanel "outkpc",200,100,90,90;start of container gkpg, gihandle FLcount "Midi-Program change",0,127,1,5,1,152,40,16,23,-1 FLpanelEnd FLrun instr 1 ktrig changed gkpg outkpc ktrig,gkpg,0,127 endin ; Run instrument 1 for 60 seconds i 1 0 60 manual_src~dfsg/examples/filepeak.csd0000644000000000000000000000207412262561504017041 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o filepeak.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iscaldb = p4 ;set peak amplitude in dB ipeak filepeak "Church.wav" iscal = ampdb(iscaldb)/ipeak ;calculate amp multiplier printf_i "\nPeak value in file '%s' is %f (%.3f dB).\n\n", 1, "Church.wav", ipeak, dbamp(ipeak) asnd soundin "Church.wav" outs asnd, asnd ; scale & write file to disk asig = asnd*iscal ;scale to p4 fout "Church_norm.wav", 14, asig endin instr 2 ; play scaled file aout soundin "Church_norm.wav" ipknew filepeak "Church_norm.wav" printf_i "\nPeak value in file '%s' is %f (%.3f dB).\n\n", 1, "Church_norm.wav", ipknew, dbamp(ipknew) outs aout, aout endin i 1 0 2 -6 ; normalize audio to -6 dB i 2 2 2 e manual_src~dfsg/examples/move20000644000000000000000000000007012262561504015533 0ustar rootroot 0 -100 1 3 0 0 5.6 1 20manual_src~dfsg/examples/doppler.csd0000644000000000000000000000212312262561504016721 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o doppler.wav -W ;;; for file output any platform sr = 48000 ksmps = 128 nchnls = 2 0dbfs = 1 instr 1 iattack init 0.05 irelease init 0.05 isustain init p3 p3 init iattack + isustain + irelease kdamping linseg 0.0, iattack, 1.0, isustain, 1.0, irelease, 0.0 kmic init 4 ; Position envelope, with a changing rate of change of position. ; transeg a dur ty b dur ty c dur ty d kposition transeg 4, p3*.4, 0, 120, p3*.3, -3, 50, p3*.3, 2, 4 ismoothinghz init 6 ispeedofsound init 340.29 asignal vco2 0.5, 110 aoutput doppler asignal, kposition, kmic, ispeedofsound, ismoothinghz outs aoutput*kdamping, aoutput * kdamping endin i1 0.0 20 e1 manual_src~dfsg/examples/128,8-cylinderX0000755000000000000000000000573312262561504017136 0ustar rootroot 0 1 0 8 0 120 1 0 1 2 1 9 1 121 2 1 2 3 2 10 2 122 3 2 3 4 3 11 3 123 4 3 4 5 4 12 4 124 5 4 5 6 5 13 5 125 6 5 6 7 6 14 6 126 7 6 7 15 7 127 8 0 8 9 8 16 9 1 9 8 9 10 9 17 10 2 10 9 10 11 10 18 11 3 11 10 11 12 11 19 12 4 12 11 12 13 12 20 13 5 13 12 13 14 13 21 14 6 14 13 14 15 14 22 15 7 15 14 15 23 16 8 16 17 16 24 17 9 17 16 17 18 17 25 18 10 18 17 18 19 18 26 19 11 19 18 19 20 19 27 20 12 20 19 20 21 20 28 21 13 21 20 21 22 21 29 22 14 22 21 22 23 22 30 23 15 23 22 23 31 24 16 24 25 24 32 25 17 25 24 25 26 25 33 26 18 26 25 26 27 26 34 27 19 27 26 27 28 27 35 28 20 28 27 28 29 28 36 29 21 29 28 29 30 29 37 30 22 30 29 30 31 30 38 31 23 31 30 31 39 32 24 32 33 32 40 33 25 33 32 33 34 33 41 34 26 34 33 34 35 34 42 35 27 35 34 35 36 35 43 36 28 36 35 36 37 36 44 37 29 37 36 37 38 37 45 38 30 38 37 38 39 38 46 39 31 39 38 39 47 40 32 40 41 40 48 41 33 41 40 41 42 41 49 42 34 42 41 42 43 42 50 43 35 43 42 43 44 43 51 44 36 44 43 44 45 44 52 45 37 45 44 45 46 45 53 46 38 46 45 46 47 46 54 47 39 47 46 47 55 48 40 48 49 48 56 49 41 49 48 49 50 49 57 50 42 50 49 50 51 50 58 51 43 51 50 51 52 51 59 52 44 52 51 52 53 52 60 53 45 53 52 53 54 53 61 54 46 54 53 54 55 54 62 55 47 55 54 55 63 56 48 56 57 56 64 57 49 57 56 57 58 57 65 58 50 58 57 58 59 58 66 59 51 59 58 59 60 59 67 60 52 60 59 60 61 60 68 61 53 61 60 61 62 61 69 62 54 62 61 62 63 62 70 63 55 63 62 63 71 64 56 64 65 64 72 65 57 65 64 65 66 65 73 66 58 66 65 66 67 66 74 67 59 67 66 67 68 67 75 68 60 68 67 68 69 68 76 69 61 69 68 69 70 69 77 70 62 70 69 70 71 70 78 71 63 71 70 71 79 72 64 72 73 72 80 73 65 73 72 73 74 73 81 74 66 74 73 74 75 74 82 75 67 75 74 75 76 75 83 76 68 76 75 76 77 76 84 77 69 77 76 77 78 77 85 78 70 78 77 78 79 78 86 79 71 79 78 79 87 80 72 80 81 80 88 81 73 81 80 81 82 81 89 82 74 82 81 82 83 82 90 83 75 83 82 83 84 83 91 84 76 84 83 84 85 84 92 85 77 85 84 85 86 85 93 86 78 86 85 86 87 86 94 87 79 87 86 87 95 88 80 88 89 88 96 89 81 89 88 89 90 89 97 90 82 90 89 90 91 90 98 91 83 91 90 91 92 91 99 92 84 92 91 92 93 92 100 93 85 93 92 93 94 93 101 94 86 94 93 94 95 94 102 95 87 95 94 95 103 96 88 96 97 96 104 97 89 97 96 97 98 97 105 98 90 98 97 98 99 98 106 99 91 99 98 99 100 99 107 100 92 100 99 100 101 100 108 101 93 101 100 101 102 101 109 102 94 102 101 102 103 102 110 103 95 103 102 103 111 104 96 104 105 104 112 105 97 105 104 105 106 105 113 106 98 106 105 106 107 106 114 107 99 107 106 107 108 107 115 108 100 108 107 108 109 108 116 109 101 109 108 109 110 109 117 110 102 110 109 110 111 110 118 111 103 111 110 111 119 112 104 112 113 112 120 113 105 113 112 113 114 113 121 114 106 114 113 114 115 114 122 115 107 115 114 115 116 115 123 116 108 116 115 116 117 116 124 117 109 117 116 117 118 117 125 118 110 118 117 118 119 118 126 119 111 119 118 119 127 120 0 120 112 120 121 121 1 121 113 121 120 121 122 122 2 122 114 122 121 122 123 123 3 123 115 123 122 123 124 124 4 124 116 124 123 124 125 125 5 125 117 125 124 125 126 126 6 126 118 126 125 126 127 127 7 127 119 127 126 manual_src~dfsg/examples/strget.csd0000644000000000000000000000106612262561504016571 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o strget.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 strset 1, "fox.wav" strset 2, "beats.wav" instr 1 Sfile strget p4 asig soundin Sfile outs asig, asig endin i 1 0 2.7 1 i 1 + 2 2 e manual_src~dfsg/examples/expsega.csd0000644000000000000000000000155412262561504016717 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o expsega.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Define a short percussive amplitude envelope that ; goes from 0.01 to 20,000 and back. aenv expsega 0.01, 0.1, 20000, 0.1, 0.01 a1 oscil aenv, 440, 1 out a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #1 for one second. i 1 1 1 ; Play Instrument #1 for one second. i 1 2 1 ; Play Instrument #1 for one second. i 1 3 1 e manual_src~dfsg/examples/readk2.csd0000644000000000000000000000226612262561504016434 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o readk2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes two control signals to a file kfreq randh 100, 1, 2, 1, 500 ;generates one random number between 400 and 600 per second kdb randh 12, 1, 2, 1, -12 ;amplitudes in dB between -24 and 0 dumpk2 kfreq, kdb, "dumpk2.txt", 8, 1 ;writes the control signals prints "WRITING:\n" printks "kfreq = %f, kdb = %f\n", 1, kfreq, kdb ;prints them endin instr 2 ;reads the file written by instr 1 kf,kdb readk2 "dumpk2.txt", 8, 1 prints "READING:\n" printks "kfreq = %f, kdb = %f\n", 1, kf, kdb ;prints again kdb lineto kdb, .1 ;smoothing amp transition aout poscil ampdb(kdb), kf, giSine outs aout, aout endin i 1 0 5 i 2 5 5 e manual_src~dfsg/examples/fmwurlie.csd0000644000000000000000000000136412262561504017114 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmwurlie.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 440 kc1 = p4 kc2 = 1 kvdepth = 0.05 kvrate = 6 ifn1 = 1 ifn2 = 1 ifn3 = 1 ifn4 = 2 ivfn = 1 asig fmwurlie .5, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn outs asig, asig endin ; sine wave f 1 0 32768 10 1 ; audio file f 2 0 256 1 "fwavblnk.aiff" 0 0 0 i 1 0 3 6 i 1 + 3 30 i 1 + 2 60 e manual_src~dfsg/examples/pitch.csd0000644000000000000000000000176712262561504016400 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pitch.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;clean audio asig soundin p4 outs asig, asig endin instr 2 ;use pitch iupdte = 0.001 ;high definition ilo = 6 ihi = 10 idbthresh = 10 ifrqs = 12 iconf = 10 istrt = 8 asig soundin p4 koct, kamp pitch asig, iupdte, ilo, ihi, idbthresh, ifrqs, iconf, istrt kamp = kamp*.00005 ;lower volume kcps = cpsoct(koct) asig poscil kamp, kcps, 1 ;re-synthesize with sawtooth outs asig, asig endin f1 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ;sawtooth i 1 0 3 "fox.wav" i 2 3 3 "fox.wav" i 1 6 3 "mary.wav" i 2 9 3 "mary.wav" i 1 12 3 "beats.wav" i 2 15 3 "beats.wav" e manual_src~dfsg/examples/pvscale.csd0000644000000000000000000000131312262561504016711 0ustar rootroot -odac sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of the audio-signal fftblur pvscale fftin, p4, p5, p6; scale aout pvsynth fftblur; resynthesis out aout endin i 1 0 3 1 0 1; original sound i 1 3 3 1.5 0 2; fifth higher without ... i 1 6 3 1.5 1 2; ... and with different ... i 1 9 3 1.5 2 5; ... kinds of formant preservation e manual_src~dfsg/examples/oscbnk.csd0000644000000000000000000000670412262561504016544 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o oscbnk.wav -W ;;; for file output any platform /* Written by Istvan Varga */ sr = 48000 kr = 750 ksmps = 64 nchnls = 2 ga01 init 0 ga02 init 0 /* sawtooth wave */ i_ ftgen 1, 0, 16384, 7, 1, 16384, -1 /* FM waveform */ i_ ftgen 3, 0, 4096, 7, 0, 512, 0.25, 512, 1, 512, 0.25, 512, \ 0, 512, -0.25, 512, -1, 512, -0.25, 512, 0 /* AM waveform */ i_ ftgen 4, 0, 4096, 5, 1, 4096, 0.01 /* FM to EQ */ i_ ftgen 5, 0, 1024, 5, 1, 512, 32, 512, 1 /* sine wave */ i_ ftgen 6, 0, 1024, 10, 1 /* room parameters */ i_ ftgen 7, 0, 64, -2, 4, 50, -1, -1, -1, 11, \ 1, 26.833, 0.05, 0.85, 10000, 0.8, 0.5, 2, \ 1, 1.753, 0.05, 0.85, 5000, 0.8, 0.5, 2, \ 1, 39.451, 0.05, 0.85, 7000, 0.8, 0.5, 2, \ 1, 33.503, 0.05, 0.85, 7000, 0.8, 0.5, 2, \ 1, 36.151, 0.05, 0.85, 7000, 0.8, 0.5, 2, \ 1, 29.633, 0.05, 0.85, 7000, 0.8, 0.5, 2 /* generate bandlimited sawtooth waves */ i0 = 0 loop1: imaxh = sr / (2 * 440.0 * exp(log(2.0) * (i0 - 69) / 12)) i_ ftgen i0 + 256, 0, 4096, -30, 1, 1, imaxh i0 = i0 + 1 if (i0 < 127.5) igoto loop1 instr 1 p3 = p3 + 0.4 ; note frequency kcps = 440.0 * exp(log(2.0) * (p4 - 69) / 12) ; lowpass max. frequency klpmaxf limit 64 * kcps, 1000.0, 12000.0 ; FM depth in Hz kfmd1 = 0.02 * kcps ; AM frequency kamfr = kcps * 0.02 kamfr2 = kcps * 0.1 ; table number kfnum = (256 + 69 + 0.5 + 12 * log(kcps / 440.0) / log(2.0)) ; amp. envelope aenv linseg 0, 0.1, 1.0, p3 - 0.5, 1.0, 0.1, 0.5, 0.2, 0, 1.0, 0 /* oscillator / left */ a1 oscbnk kcps, 0.0, kfmd1, 0.0, 40, 200, 0.1, 0.2, 0, 0, 144, \ 0.0, klpmaxf, 0.0, 0.0, 1.5, 1.5, 2, \ kfnum, 3, 0, 5, 5, 5 a2 oscbnk kcps, 1.0, kfmd1, 0.0, 40, 201, 0.1, 0.2, kamfr, kamfr2, 148, \ 0, 0, 0, 0, 0, 0, -1, \ kfnum, 3, 4 a2 pareq a2, kcps * 8, 0.0, 0.7071, 2 a0 = a1 + a2 * 0.12 /* delay */ adel = 0.001 a01 vdelayx a0, adel, 0.01, 16 a_ oscili 1.0, 0.25, 6, 0.0 adel = adel + 1.0 / (exp(log(2.0) * a_) * 8000) a02 vdelayx a0, adel, 0.01, 16 a0 = a01 + a02 ga01 = ga01 + a0 * aenv * 2500 /* oscillator / right */ ; lowpass max. frequency a1 oscbnk kcps, 0.0, kfmd1, 0.0, 40, 202, 0.1, 0.2, 0, 0, 144, \ 0.0, klpmaxf, 0.0, 0.0, 1.0, 1.0, 2, \ kfnum, 3, 0, 5, 5, 5 a2 oscbnk kcps, 1.0, kfmd1, 0.0, 40, 203, 0.1, 0.2, kamfr, kamfr2, 148, \ 0, 0, 0, 0, 0, 0, -1, \ kfnum, 3, 4 a2 pareq a2, kcps * 8, 0.0, 0.7071, 2 a0 = a1 + a2 * 0.12 /* delay */ adel = 0.001 a01 vdelayx a0, adel, 0.01, 16 a_ oscili 1.0, 0.25, 6, 0.25 adel = adel + 1.0 / (exp(log(2.0) * a_) * 8000) a02 vdelayx a0, adel, 0.01, 16 a0 = a01 + a02 ga02 = ga02 + a0 * aenv * 2500 endin /* output / left */ instr 81 i1 = 0.000001 aLl, aLh, aRl, aRh spat3di ga01 + i1*i1*i1*i1, -8.0, 4.0, 0.0, 0.3, 7, 4 ga01 = 0 aLl butterlp aLl, 800.0 aRl butterlp aRl, 800.0 outs aLl + aLh, aRl + aRh endin /* output / right */ instr 82 i1 = 0.000001 aLl, aLh, aRl, aRh spat3di ga02 + i1*i1*i1*i1, 8.0, 4.0, 0.0, 0.3, 7, 4 ga02 = 0 aLl butterlp aLl, 800.0 aRl butterlp aRl, 800.0 outs aLl + aLh, aRl + aRh endin /* Written by Istvan Varga */ t 0 60 i 1 0 4 41 i 1 0 4 60 i 1 0 4 65 i 1 0 4 69 i 81 0 5.5 i 82 0 5.5 e manual_src~dfsg/examples/loscil.csd0000644000000000000000000000145512262561504016550 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ichnls = ftchnls(p4) print ichnls if (ichnls == 1) then asigL loscil .8, 1, p4, 1 asigR = asigL elseif (ichnls == 2) then asigL, asigR loscil .8, 1, p4, 1 ;safety precaution if not mono or stereo else asigL = 0 asigR = 0 endif outs asigL, asigR endin f 1 0 0 1 "mary.wav" 0 0 0 f 2 0 0 1 "kickroll.wav" 0 0 0 i 1 0 3 1 ;mono file i 1 + 2 2 ;stereo file e manual_src~dfsg/examples/schedule.csd0000644000000000000000000000154612262561504017060 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o schedule.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - oscillator with a high note. instr 1 ; Play Instrument #2 at the same time. schedule 2, 0, p3 ; Play a high note. a1 oscils 10000, 880, 1 out a1 endin ; Instrument #2 - oscillator with a low note. instr 2 ; Play a low note. a1 oscils 10000, 220, 1 out a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for half a second. i 1 0 0.5 ; Play Instrument #1 for half a second. i 1 1 0.5 e manual_src~dfsg/examples/signalflowgraph.csd0000644000000000000000000000602412262561504020447 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o madsr.wav -W ;;; for file output any platform /* Written by Michael Gogins */ ; Initialize the global variables. sr = 44100 ksmps = 100 nchnls = 2 ; Connect up the instruments to create a signal flow graph. connect "SimpleSine", "leftout", "Reverberator", "leftin" connect "SimpleSine", "rightout", "Reverberator", "rightin" connect "Moogy", "leftout", "Reverberator", "leftin" connect "Moogy", "rightout", "Reverberator", "rightin" connect "Reverberator", "leftout", "Compressor", "leftin" connect "Reverberator", "rightout", "Compressor", "rightin" connect "Compressor", "leftout", "Soundfile", "leftin" connect "Compressor", "rightout", "Soundfile", "rightin" ; Turn on the "effect" units in the signal flow graph. alwayson "Reverberator", 0.91, 12000 alwayson "Compressor" alwayson "Soundfile" instr SimpleSine ihz = cpsmidinn(p4) iamplitude = ampdb(p5) print ihz, iamplitude ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. isine ftgenonce 0, 0, 4096, 10, 1 a1 oscili iamplitude, ihz, isine aenv madsr 0.05, 0.1, 0.5, 0.2 asignal = a1 * aenv ; Stereo audio outlet to be routed in the orchestra header. outleta "leftout", asignal * 0.25 outleta "rightout", asignal * 0.75 endin instr Moogy ihz = cpsmidinn(p4) iamplitude = ampdb(p5) ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. isine ftgenonce 0, 0, 4096, 10, 1 asignal vco iamplitude, ihz, 1, 0.5, isine kfco line 200, p3, 2000 krez init 0.9 asignal moogvcf asignal, kfco, krez, 100000 ; Stereo audio outlet to be routed in the orchestra header. outleta "leftout", asignal * 0.75 outleta "rightout", asignal * 0.25 endin instr Reverberator ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" idelay = p4 icutoff = p5 aleftout, arightout reverbsc aleftin, arightin, idelay, icutoff ; Stereo output. outleta "leftout", aleftout outleta "rightout", arightout endin instr Compressor ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" kthreshold = 25000 icomp1 = 0.5 icomp2 = 0.763 irtime = 0.1 iftime = 0.1 aleftout dam aleftin, kthreshold, icomp1, icomp2, irtime, iftime arightout dam arightin, kthreshold, icomp1, icomp2, irtime, iftime ; Stereo output. outleta "leftout", aleftout outleta "rightout", arightout endin instr Soundfile ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" outs aleftin, arightin endin ; Not necessary to activate "effects" or create f-tables in the score! ; Overlapping notes to create new instances of instruments. i "SimpleSine" 1 5 60 85 i "SimpleSine" 2 5 64 80 i "Moogy" 3 5 67 75 i "Moogy" 4 5 71 70 e 1 manual_src~dfsg/examples/getcfg.csd0000644000000000000000000000161512262561504016520 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out -iadc ;;;uncomment -iadc if realtime audio input is needed too sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 S1 getcfg 1 ; -+max_str_len S2 getcfg 2 ; -i S3 getcfg 3 ; -o S4 getcfg 4 ; RTaudio S5 getcfg 5 ; -t S6 getcfg 6 ; os system host S7 getcfg 7 ; callback prints "------------------------------" prints "\nMax string len : " prints S1 prints "\nInput file name (-i) : " prints S2 prints "\nOutput file name (-o) : " prints S3 prints "\nRTaudio (-odac) : " prints S4 prints "\nBeat mode (-t)? : " prints S5 prints "\nHost Op. Sys. : " prints S6 prints "\nCallback ? : " prints S7 prints "\n" prints "------------------------------\n" endin i 1 0 0 e manual_src~dfsg/examples/gen12.csd0000644000000000000000000000440312262561504016173 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen12.wav -W ;;; for file output any platform ;example from the Csound Book, page 87 sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = p3 iamp = p4 icarfrq = p5 imodfrq = p6 aenv expseg .01, idur*.1, iamp, idur*.8, iamp*.75, idur*.1, .01 i1 = p7*imodfrq ;p7=modulation index start i2 = p8*imodfrq ;p8=modulation index end adev line i1, idur, i2 ;modulation frequency aindex line p7, idur, p8 ;modulation index ar linseg 1, .1, p9, p3-.2, p10, .1, 1 ; r value envelope: p9-p10 =exp. partial strength parameter start and end amp1 = (aindex*(ar+(1/ar)))/2 afmod oscili amp1, imodfrq, 1 ;FM modulator (sine) atab = (aindex*(ar-(1/ar)))/2 ;index to table alook tablei atab, 37 ;table lookup to GEN12 aamod oscili atab, adev, 2 ;am modulator (cosine) aamod = (exp(alook+aamod))*aenv acar oscili aamod, afmod+icarfrq, 1 ;AFM (carrier) asig balance acar, aenv outs asig, asig endin f 1 0 8192 10 1 f 2 0 8192 9 1 1 90 f37 0 1024 -12 40 ;Bessel function-defined from 0 to 40 i 1 0 2 .2 800 800 1 6 .1 2 i 1 + . . 1900 147 8 1 4 .2 i 1 . . . 1100 380 2 9 .5 2 i 1 . 10 . 100 100 11 3 .2 5 s i 1 0 1 .1 200 100 1 6 .1 2 i 1 + . < < < < < < < i 1 + . . < < < < < < i 1 + . . < < < < < < i 1 + . . < < < < < < i 1 + . . < < < < < < i 1 + . . < < < < < < i 1 + 10 .2 800 800 9 1 .9 6 s i 1 0 11 .25 50 51 1 6 .1 2 i 1 1 9 .05 700 401 1 6 .1 2 i 1 2 8 . 900 147 8 1 4 .2 i 1 3 7 . 1100 381 2 9 .5 2 i 1 4 6 . 200 102 11 3 .2 5 i 1 5 6 . 800 803 9 1 .9 6 e manual_src~dfsg/examples/partikkel-2.csd0000644000000000000000000001141212262561504017402 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out -odac ;;;RT audio ; For Non-realtime ouput leave only the line below: ; -o partikkel.wav -W ;;; for file output any platform sr = 44100 ksmps = 20 nchnls = 2 ; Example by Joachim Heintz and Oeyvind Brandtsegg 2008 giCosine ftgen 0, 0, 8193, 9, 1, 1, 90 ; cosine giDisttab ftgen 0, 0, 32768, 7, 0, 32768, 1 ; for kdistribution giFile ftgen 0, 0, 0, 1, "fox.wav", 0, 0, 0 ; soundfile for source waveform giWin ftgen 0, 0, 4096, 20, 9, 1 ; grain envelope giPan ftgen 0, 0, 32768, -21, 1 ; for panning (random values between 0 and 1) ; ************************************************* ; partikkel example, processing of soundfile ; uses the file "fox.wav" ; ************************************************* instr 1 /*score parameters*/ ispeed = p4 ; 1 = original speed igrainrate = p5 ; grain rate igrainsize = p6 ; grain size in ms icent = p7 ; transposition in cent iposrand = p8 ; time position randomness (offset) of the pointer in ms icentrand = p9 ; transposition randomness in cents ipan = p10 ; panning narrow (0) to wide (1) idist = p11 ; grain distribution (0=periodic, 1=scattered) /*get length of source wave file, needed for both transposition and time pointer*/ ifilen tableng giFile ifildur = ifilen / sr /*sync input (disabled)*/ async = 0 /*grain envelope*/ kenv2amt = 1 ; use only secondary envelope ienv2tab = giWin ; grain (secondary) envelope ienv_attack = -1 ; default attack envelope (flat) ienv_decay = -1 ; default decay envelope (flat) ksustain_amount = 0.5 ; no meaning in this case (use only secondary envelope, ienv2tab) ka_d_ratio = 0.5 ; no meaning in this case (use only secondary envelope, ienv2tab) /*amplitude*/ kamp = 0.4*0dbfs ; grain amplitude igainmasks = -1 ; (default) no gain masking /*transposition*/ kcentrand rand icentrand ; random transposition iorig = 1 / ifildur ; original pitch kwavfreq = iorig * cent(icent + kcentrand) /*other pitch related (disabled)*/ ksweepshape = 0 ; no frequency sweep iwavfreqstarttab = -1 ; default frequency sweep start iwavfreqendtab = -1 ; default frequency sweep end awavfm = 0 ; no FM input ifmamptab = -1 ; default FM scaling (=1) kfmenv = -1 ; default FM envelope (flat) /*trainlet related (disabled)*/ icosine = giCosine ; cosine ftable kTrainCps = igrainrate ; set trainlet cps equal to grain rate for single-cycle trainlet in each grain knumpartials = 1 ; number of partials in trainlet kchroma = 1 ; balance of partials in trainlet /*panning, using channel masks*/ imid = .5; center ileftmost = imid - ipan/2 irightmost = imid + ipan/2 giPanthis ftgen 0, 0, 32768, -24, giPan, ileftmost, irightmost ; rescales giPan according to ipan tableiw 0, 0, giPanthis ; change index 0 ... tableiw 32766, 1, giPanthis ; ... and 1 for ichannelmasks ichannelmasks = giPanthis ; ftable for panning /*random gain masking (disabled)*/ krandommask = 0 /*source waveforms*/ kwaveform1 = giFile ; source waveform kwaveform2 = giFile ; all 4 sources are the same kwaveform3 = giFile kwaveform4 = giFile iwaveamptab = -1 ; (default) equal mix of source waveforms and no amplitude for trainlets /*time pointer*/ afilposphas phasor ispeed / ifildur /*generate random deviation of the time pointer*/ iposrandsec = iposrand / 1000 ; ms -> sec iposrand = iposrandsec / ifildur ; phase values (0-1) krndpos linrand iposrand ; random offset in phase values /*add random deviation to the time pointer*/ asamplepos1 = afilposphas + krndpos; resulting phase values (0-1) asamplepos2 = asamplepos1 asamplepos3 = asamplepos1 asamplepos4 = asamplepos1 /*original key for each source waveform*/ kwavekey1 = 1 kwavekey2 = kwavekey1 kwavekey3 = kwavekey1 kwavekey4 = kwavekey1 /* maximum number of grains per k-period*/ imax_grains = 100 aL, aR partikkel igrainrate, idist, giDisttab, async, kenv2amt, ienv2tab, \ ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, igrainsize, kamp, igainmasks, \ kwavfreq, ksweepshape, iwavfreqstarttab, iwavfreqendtab, awavfm, \ ifmamptab, kfmenv, icosine, kTrainCps, knumpartials, \ kchroma, ichannelmasks, krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, asamplepos4, \ kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains outs aL, aR endin ;i1 st dur speed grate gsize cent posrnd cntrnd pan dist i1 0 2.757 1 200 15 0 0 0 0 0 s i1 0 2.757 1 200 15 400 0 0 0 0 s i1 0 2.757 1 15 450 400 0 0 0 0 s i1 0 2.757 1 15 450 400 0 0 0 0.4 s i1 0 2.757 1 200 15 0 400 0 0 1 s i1 0 5.514 .5 200 20 0 0 600 .5 1 s i1 0 11.028 .25 200 15 0 1000 400 1 1 manual_src~dfsg/examples/maparray.csd0000644000000000000000000000360512262561504017076 0ustar rootroot ; Select audio/midi flags here according to platform -n sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;create an array and fill with numbers kArrSrc[] fillarray 1.01, 2.02, 3.03, 4.05, 5.08, 6.13, 7.21 ;print source array printf "%s", 1, "\nSource array:\n" kndx = 0 until kndx == lenarray(kArrSrc) do printf "kArrSrc[%d] = %f\n", kndx+1, kndx, kArrSrc[kndx] kndx += 1 od ;create an empty array for the results kArrRes[] init 7 ;apply the sqrt() function to each element kArrRes maparray kArrSrc, "sqrt" ;print the result printf "%s", 1, "\nResult after applying sqrt() to source array\n" kndx = 0 until kndx == lenarray(kArrRes) do printf "kArrRes[%d] = %f\n", kndx+1, kndx, kArrRes[kndx] kndx += 1 od ;apply the log() function to each element kArrRes maparray kArrSrc, "log" ;print the result printf "%s", 1, "\nResult after applying log() to source array\n" kndx = 0 until kndx == lenarray(kArrRes) do printf "kArrRes[%d] = %f\n", kndx+1, kndx, kArrRes[kndx] kndx += 1 od ;apply the int() function to each element kArrRes maparray kArrSrc, "int" ;print the result printf "%s", 1, "\nResult after applying int() to source array\n" kndx = 0 until kndx == lenarray(kArrRes) do printf "kArrRes[%d] = %f\n", kndx+1, kndx, kArrRes[kndx] kndx += 1 od ;apply the frac() function to each element kArrRes maparray kArrSrc, "frac" ;print the result printf "%s", 1, "\nResult after applying frac() to source array\n" kndx = 0 until kndx == lenarray(kArrRes) do printf "kArrRes[%d] = %f\n", kndx+1, kndx, kArrRes[kndx] kndx += 1 od ;turn instrument instance off turnoff endin i 1 0 0.1 e manual_src~dfsg/examples/outc.csd0000644000000000000000000000126412262561504016233 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outc.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 5 0dbfs = 1 instr 1 asig vco2 .05, 30 ; sawtooth waveform at low volume kcut line 100, p3, 30 ; Vary cutoff frequency kresonance = 7 inumlayer = 2 asig lowresx asig, kcut, kresonance, inumlayer ; output same sound to 5 channels outc asig,asig,asig,asig,asig endin i 1 0 30 e manual_src~dfsg/examples/FLslider-2.csd0000644000000000000000000001143012262561504017120 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLslider-2.wav -W ;;; for file output any platform sr = 44100 kr = 441 ksmps = 100 nchnls = 1 ;By Andres Cabrera 2007 FLpanel "Slider Types", 410, 260, 50, 50 ; Distance of the left edge of the slider ; from the left edge of the panel ix = 10 ; Distance of the top edge of the slider ; from the top edge of the panel iy = 10 ; Create boxes to display widget values givalue1 FLvalue "1", 60, 20, ix + 330, iy givalue3 FLvalue "3", 60, 20, ix + 330, iy + 40 givalue5 FLvalue "5", 60, 20, ix + 330, iy + 80 givalue2 FLvalue "2", 60, 20, ix + 60, iy + 140 givalue4 FLvalue "4", 60, 20, ix + 195, iy + 140 givalue6 FLvalue "6", 60, 20, ix + 320, iy + 140 ;Horizontal sliders gkdummy1, gihandle1 FLslider "Type 1", 200, 5000, -1, 1, givalue1, 320, 20, ix, iy gkdummy3, gihandle3 FLslider "Type 3", 0, 15000, 0, 3, givalue3, 320, 20, ix, iy + 40 ; Reversed slider gkdummy5, gihandle5 FLslider "Type 5", 1, 0, 0, 5, givalue5, 320, 20, ix, iy + 80 ;Vertical sliders gkdummy2, gihandle2 FLslider "Type 2", 0, 1, 0, 2, givalue2, 20, 100, ix+ 30 , iy + 120 ; Reversed slider gkdummy4, gihandle4 FLslider "Type 4", 1, 0, 0, 4, givalue4, 20, 100, ix + 165 , iy + 120 gkdummy6, gihandle6 FLslider "Type 6", 0, 1, 0, 6, givalue6, 20, 100, ix + 290 , iy + 120 FLpanelEnd FLpanel "Plastic Slider Types", 410, 300, 150, 150 ; Distance of the left edge of the slider ; from the left edge of the panel ix = 10 ; Distance of the top edge of the slider ; from the top edge of the panel iy = 10 ; Create boxes to display widget values givalue21 FLvalue "21", 60, 20, ix + 330, iy givalue23 FLvalue "23", 60, 20, ix + 330, iy + 40 givalue25 FLvalue "25", 60, 20, ix + 330, iy + 80 givalue22 FLvalue "22", 60, 20, ix + 60, iy + 140 givalue24 FLvalue "24", 60, 20, ix + 195, iy + 140 givalue26 FLvalue "26", 60, 20, ix + 320, iy + 140 ;Horizontal sliders gkdummy21, gihandle21 FLslider "Type 21", 200, 5000, -1, 21, givalue21, 320, 20, ix, iy gkdummy23, gihandle23 FLslider "Type 23", 0, 15000, 0, 23, givalue23, 320, 20, ix, iy + 40 ; Reversed slider gkdummy25, gihandle25 FLslider "Type 25", 1, 0, 0, 25, givalue25, 320, 20, ix, iy + 80 ;Vertical sliders gkdummy22, gihandle22 FLslider "Type 22", 0, 1, 0, 22, givalue22, 20, 100, ix+ 30 , iy + 120 ; Reversed slider gkdummy24, gihandle24 FLslider "Type 24", 1, 0, 0, 24, givalue24, 20, 100, ix + 165 , iy + 120 gkdummy26, gihandle26 FLslider "Type 26", 0, 1, 0, 26, givalue26, 20, 100, ix + 290 , iy + 120 ;Button to add color to the sliders gkcolors, ihdummy FLbutton "Color", 1, 0, 21, 150, 30, 30, 260, 0, 10, 0, 1 FLpanelEnd FLrun ;Set some widget's initial value FLsetVal_i 500, gihandle1 FLsetVal_i 1000, gihandle3 instr 10 ; Set the color of widgets FLsetColor 200, 230, 0, gihandle1 FLsetColor 0, 123, 100, gihandle2 FLsetColor 180, 23, 12, gihandle3 FLsetColor 10, 230, 0, gihandle4 FLsetColor 0, 0, 0, gihandle5 FLsetColor 0, 0, 0, gihandle6 FLsetColor 200, 230, 0, givalue1 FLsetColor 0, 123, 100, givalue2 FLsetColor 180, 23, 12, givalue3 FLsetColor 10, 230, 0, givalue4 FLsetColor 255, 255, 255, givalue5 FLsetColor 255, 255, 255, givalue6 FLsetColor2 20, 23, 100, gihandle1 FLsetColor2 200,0 ,123 , gihandle2 FLsetColor2 180, 180, 100, gihandle3 FLsetColor2 180, 23, 12, gihandle4 FLsetColor2 180, 180, 100, gihandle5 FLsetColor2 180, 23, 12, gihandle6 FLsetColor 200, 230, 0, gihandle21 FLsetColor 0, 123, 100, gihandle22 FLsetColor 180, 23, 12, gihandle23 FLsetColor 10, 230, 0, gihandle24 FLsetColor 0, 0, 0, gihandle25 FLsetColor 0, 0, 0, gihandle26 FLsetColor 200, 230, 0, givalue21 FLsetColor 0, 123, 100, givalue22 FLsetColor 180, 23, 12, givalue23 FLsetColor 10, 230, 0, givalue24 FLsetColor 255, 255, 255, givalue25 FLsetColor 255, 255, 255, givalue26 FLsetColor2 20, 23, 100, gihandle21 FLsetColor2 200,0 ,123 , gihandle22 FLsetColor2 180, 180, 100, gihandle23 FLsetColor2 180, 23, 12, gihandle24 FLsetColor2 180, 180, 100, gihandle25 FLsetColor2 180, 23, 12, gihandle26 ; Slider values must be updated for colors to change FLsetVal_i 250, gihandle1 FLsetVal_i 0.5, gihandle2 FLsetVal_i 0, gihandle3 FLsetVal_i 0, gihandle4 FLsetVal_i 0, gihandle5 FLsetVal_i 0.5, gihandle6 FLsetVal_i 250, gihandle21 FLsetVal_i 0.5, gihandle22 FLsetVal_i 500, gihandle23 FLsetVal_i 0, gihandle24 FLsetVal_i 0, gihandle25 FLsetVal_i 0.5, gihandle26 endin f 0 3600 ;Dumy table to make csound wait for realtime events e manual_src~dfsg/examples/scoreline_i.csd0000644000000000000000000000116712262561504017556 0ustar rootroot ; Select audio/midi flags here according to platform -odac --old-parser ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o scoreline.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 scoreline_i {{ i 2 0 3 "flute.aiff" i 2 1 3 "beats.wav" }} endin instr 2 asig soundin p4 outs asig*.8, asig*.8 endin i1 0 1 e manual_src~dfsg/examples/gen01.csd0000644000000000000000000000177212262561504016177 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen01.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;plays deferred and non-deferred sounds with loscil ifn = p4 ibas = 1 asig loscil 1, 1, ifn, ibas outs asig, asig endin instr 2 ;plays only non-deffered sound isnd = p4 aread line sr*p3, p3, 0 ;play this backward asig tablei aread, isnd ;use table 1 outs asig, asig endin f 1 0 131072 1 "beats.wav" 0 0 0 ;non-deferred sound f 2 0 0 1 "flute.aiff" 0 0 0 ;& deferred sounds in f 3 0 0 1 "beats.ogg" 0 0 0 ;different formats i 1 0 1 1 i 1 + 1 2 i 1 + 1 3 i 2 4 2 1 ;non-deffered sound for instr. 2 e manual_src~dfsg/examples/pvsbandr.csd0000644000000000000000000000133612262561504017100 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsbandr.wav -W ;;; for file output any platform sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 Sfile = "fox.wav" klowcut = 100 klowfull = 200 khighfull = 1900 khighcut = 2000 ain soundin Sfile fftin pvsanal ain, 1024, 256, 1024, 1; fft-analysis of the audio-signal fftbp pvsbandr fftin, klowcut, klowfull, khighfull, khighcut ; band reject abp pvsynth fftbp; resynthesis out abp endin i 1 0 3 e manual_src~dfsg/examples/gen25.csd0000644000000000000000000000145512262561504016203 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen25.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 1, 0, 32768, 10, 1 gienv ftgen 2, 0, 1025, 25, 0, 0.01, 200, 1, 400, 1, 513, 0.01 ; y value must be >= 0 instr 1 kcps init 3/p3 ;play 3x over duration of note kndx phasor kcps ixmode = 1 ;normalize to 0-1 kval table kndx, gienv, ixmode kval =kval*100 ;scale up to 0-100 asig poscil 1, 220+kval, gisin ;use table for amplitude outs asig, asig endin i 1 0 4 e manual_src~dfsg/examples/centroid.csd0000644000000000000000000000136612262561504017073 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o centroid.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 16384, 10, 1 instr 1 ktrig init 1 ;calculate centroid a1 oscil3 0.5, p4, giSine ;of the sine wave k1 centroid a1, ktrig, 16384 asig oscil3 0.5, k1, giSine printk2 k1 ;print & compare: outs a1, asig ;left = original, right = centroid signal endin i1 0 2 20 i1 + 2 200 i1 + 2 2000 e manual_src~dfsg/examples/vcopy.csd0000644000000000000000000000153612262561504016423 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vcopy.wav -W ;;; for file output any platform sr=44100 kr=4410 ksmps=10 nchnls=2 instr 1 ;table playback ar lposcil 1, 1, 0, 262144, 1 outs ar,ar endin instr 2 vcopy 2, 1, 20000 ;copy vector from sample to empty table vmult 5, 20000, 262144 ;scale noise to make it audible vcopy 1, 5, 20000 ;put noise into sample turnoff endin instr 3 vcopy 1, 2, 20000 ;put original information back in turnoff endin f1 0 262144 -1 "beats.wav" 0 4 0 f2 0 262144 2 0 f5 0 262144 21 3 30000 i1 0 4 i2 3 1 s i1 0 4 i3 3 1 s i1 0 4 manual_src~dfsg/examples/hrtfstat-2.csd0000644000000000000000000000161512262561504017257 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o hrtfstat-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iAz = p4 iElev = p5 itim = ftlptim(1) ; transeg a dur ty b dur ty c dur ty d kamp transeg 0, p3*.1, 0, .9, p3*.3, -3, .5, p3*.3, -2, 0 ain loscil3 kamp, 50, 1 aleft,aright hrtfstat ain, iAz, iElev, "hrtf-44100-left.dat","hrtf-44100-right.dat" outs aleft, aright endin f 1 0 0 1 "Church.wav" 0 0 0 ;Csound computes tablesize ; Azim Elev i1 0 7 90 0 ;to the right i1 3 7 -90 -40 ;to the left and below i1 6 7 180 90 ;behind and up e manual_src~dfsg/examples/ceil.csd0000644000000000000000000000105212262561504016170 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too sr = 44100 ksmps = 32 nchnls = 2 instr 1 inum = p4 iceil = ceil(inum) print iceil endin i 1 0 0 1 i . . . 0.999999 i . . . 0.000001 i . . . 0 i . . . -0.0000001 i . . . -0.9999999 i . . . -1 e manual_src~dfsg/examples/fin.csd0000644000000000000000000000124612262561504016035 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fin.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asnd init 0 ;input of fin must be initialized fin "fox.wav", 0, 0, asnd ;read audiofile aenv follow asnd, 0.01 ;envelope follower kenv downsamp aenv asig rand kenv ;gate the noise with audiofile outs asig, asig endin i 1 0 3 e manual_src~dfsg/examples/pinkish.csd0000644000000000000000000000122212262561504016720 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pinkish.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 awhite unirand 2.0 ; Normalize to +/-1.0 awhite = awhite - 1.0 apink pinkish awhite, 1, 0, 0, 1 out apink * 30000 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/ATScross.csd0000644000000000000000000000136712262561504016766 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc for RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATScross.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" and "fox.ats" are created by atsa ktime line 0, p3, 4 ktime2 line 0, p3, 4 kline expseg 0.001, .3, 1, p3-.3, 1 kline2 expseg 0.001, p3, 3 ATSbufread ktime2, 1, "fox.ats", 20 aout ATScross ktime, 2, "beats.ats", 1, kline, 0.001 * (4 - kline2), 180 outs aout*2, aout*2 endin ; sine wave. f 1 0 16384 10 1 i 1 0 4 e manual_src~dfsg/examples/07AcousticGuitar.sf20000644000000000000000000043156012262561504020305 0ustar rootrootRIFFh3sfbkLISTINFOifilisng E-mu 10K1INAMGuitar Acoustic SeagullICRDFeb 28, 1999IENGIPRDSBAWE32ICOPPublic DomainICMTISFTSFEDT v1.28:SWAMI v0.9.2LIST0sdtasmpl0{tmiWH:>'<^ncVm{qrjTGWVJ\gb`J%wL[]Z_K=1{twbT\xya]caQ:n`M ef(f}E."- !!!""#$%w&&q'''E(()u*V+D,, -,d,3,Q,N,A,h,j,+*k)q('''&}&5&%{$^#"!! j |:3]b?7g) .lB`Wq܍aڬؒpԲIl8ͱ`0ƒŝ mBoνƽqV+Ǽ1JRȹc8A.ĺ  "Q0LǓʿ/Ρ:Oԏձoܐ`_gX1GlZo5X $. 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LISTpdtaphdrLSeagul Acoustic GitEOPpbagpmod pgen)inst,Seagull Acoustic GiEOIibagimod igenD+ ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mac.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;4 band equalizer klow = p4 ;low gain1 klmid = p5 ;low gain2 kmidh = p6 ;high gain1 khigh = p7 ;high gain2 ifn = p8 ;table ilc1 table 0, ifn ;low freqency range ilc2 table 1, ifn ;low-mid ihc1 table 2, ifn ;mid-high ihc2 table 3, ifn ;high asig diskin2 "fox.wav", 1 alow1 butterlp asig, ilc1 ;lowpass 1 almid butterlp asig, ilc2 ;lowpass 2 amidh butterhp asig, ihc1 ;highpass 1 ahigh butterhp asig, ihc2 ;highpass 2 aout mac klow, alow1, klmid, almid, kmidh, amidh, khigh, ahigh outs aout, aout endin f1 0 4 -2 150 300 600 5000 f2 0 4 -2 75 500 1000 10000 f3 0 4 -2 200 700 1500 3000 ; low lowmid midhigh high table i 1 0 2.8 2 1 1 1 1 i 1 3 2.8 2 3 1 1 2 i 1 6 2.8 2 1 2 3 3 e manual_src~dfsg/examples/maxabsaccum.csd0000644000000000000000000000170412262561504017544 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o maxabsaccum.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;saw asig vco2 .2, p4 outs asig, asig gasaw = asig endin instr 2 ;sine aout poscil .3, p4, 1 outs aout, aout gasin = aout endin instr 10 accum init 0 maxabsaccum accum, gasaw + gasin ;saw and sine accumulated accum dcblock2 accum ;get rid of DC outs accum, accum clear accum endin f 1 0 4096 10 1 ;sine wave i 1 0 7 330 i 2 3 3 440 i 1 10 7 330 ;same notes but without maxabsaccum, for comparison i 2 13 3 440 i 10 0 6 ;accumulation note stops after 6 seconds e manual_src~dfsg/examples/lorisplay.csd0000644000000000000000000000343112262561504017275 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lorisplay.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; clarinet.sdif can be found in /manual/examples ; Play the partials in clarinet.sdif from 0 to 3 sec with 1 ms fadetime ; and no frequency , amplitude, or bandwidth modification. instr 1 ktime linseg 0, p3, 3.0 ; linear time function from 0 to 3 seconds lorisread ktime, "clarinet.sdif", 1, 1, 1, 1, .001 kfrq = p4 ; pitch shifting asig lorisplay 1, kfrq, 1, 1 outs asig, asig endin ; Play the partials in clarinet.sdif from 0 to 3 sec with 1 ms fadetime ; adding tuning and vibrato, increasing the "breathiness" (noisiness) and overall ; amplitude, and adding a highpass filter. instr 2 ktime linseg 0, p3, 3.0 ; linear time function from 0 to 3 seconds ; compute frequency scale for tuning ifscale = cpspch(p4)/cpspch(8.08) ; (original pitch was G#4) ; make a vibrato envelope kvenv linseg 0, p3/6, 0, p3/6, .02, p3/3, .02, p3/6, 0, p3/6, 0 kvib oscil kvenv, 4, 1 ; table 1, sinusoid kbwenv linseg 1, p3/6, 1, p3/6, 100, 2*p3/3, 100 ;lots of noise lorisread ktime, "clarinet.sdif", 1, 1, 1, 1, .001 a1 lorisplay 1, ifscale+kvib, 2, kbwenv asig atone a1, 1000 ; highpass filter, cutoff 1000 Hz outs asig, asig endin ; a sinusoid f 1 0 4096 10 1 i 1 0 3 1.2 ; shifted up i 1 + 1 1.5 i 1 + 6 .5 ; shifted down s ; strt dur ptch i 2 1 3 8.08 i 2 3.5 1 8.04 i 2 4 6 8.00 i 2 4 6 8.07 e manual_src~dfsg/examples/gen21.csd0000644000000000000000000000261012262561504016171 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen21.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 isize = ftlen(ifn) prints "TABLE NUMBER: %d\n", ifn prints "Index\tValue\n" iindex = 0 ;start loop begin_loop: ivalue tab_i iindex, ifn prints "%d:\t%f\n", iindex, ivalue iindex = iindex + 1 if (iindex < isize) igoto begin_loop ;produce sound - and repeat it 10 times so you can hear the patterns: aphase phasor 10/10 ;play all 32 values 10x over 10 seconds aphase = aphase*isize ;step through table afrq table aphase, p4 ;read table number asig poscil .5, (afrq*500)+1000,10 ;scale values of table 500 times, add 1000 Hz outs asig , asig ;so we can distinguish the different tables endin f1 0 32 21 1 ;Uniform (white noise) f2 0 32 21 6 ;Gaussian (mu=0.0, sigma=1/3.83=0.261) f3 0 32 21 6 5.745 ;Gaussian (mu=0.0, sigma=5.745/3.83=1.5) f4 0 32 21 9 1 1 2 ;Beta (note that level precedes arguments) f5 0 32 21 10 1 2 ;Weibull f10 0 8192 10 1 ;Sine wave i 1 0 10 1 i 1 11 10 2 i 1 22 10 3 i 1 33 10 4 i 1 44 10 5 e manual_src~dfsg/examples/move0000644000000000000000000000017512262561504015457 0ustar rootroot 0 -1 1 1 1 1 2 4 4 2.1 -4 -4 3 10 -10 5 -40 0manual_src~dfsg/examples/ihold.csd0000644000000000000000000000164512262561504016363 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ihold.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; A simple oscillator with its note held indefinitely. a1 oscil 10000, 440, 1 ihold ; If p4 equals 0, turn the note off. if (p4 == 0) kgoto offnow kgoto playit offnow: ; Turn the note off now. turnoff playit: ; Play the note. out a1 endin ; Table #1: an ordinary sine wave. f 1 0 32768 10 1 ; p4 = turn the note off (if it is equal to 0). ; Start playing Instrument #1. i 1 0 1 1 ; Turn Instrument #1 off after 3 seconds. i 1 3 1 0 e manual_src~dfsg/examples/STKModalBar.csd0000644000000000000000000000117212262561504017322 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKModalBar.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 = p5 ;stick hardness ; asig STKModalBar cpspch(ifrq), 1, 2, kv1, 4, 120, 11, 0, 1, 0, 8, 10, 16, 1 asig = asig * 3 ;amplify outs asig, asig endin i 1 0 2 8.00 0 i 1 + 2 8.05 120 e manual_src~dfsg/examples/granule.csd0000644000000000000000000000162212262561504016714 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o granule.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 ; k1 linseg 0,0.5,1,(p3-p2-1),1,0.5,0 a1 granule p4*k1,p5,p6,p7,p8,p9,p10,p11,p12,p13,p14,p15,\ p16,p17,p18,p19,p20,p21,p22,p23,p24 a2 granule p4*k1,p5,p6,p7,p8,p9,p10,p11,p12,p13,p14,p15,\ p16,p17,p18,p19, p20+0.17,p21,p22,p23,p24 outs a1,a2 endin ; f statement read sound file mary.wav in the SFDIR ; directory into f-table 1 f1 0 262144 1 "mary.wav" 0 0 0 i1 0 10 2000 64 0.5 0 0 1 4 0 0.005 5 0.01 50 0.02 50 30 30 0.39 \ 1 1.42 0.29 2 e manual_src~dfsg/examples/soundin.csd0000644000000000000000000000135612262561504016742 0ustar rootroot ; Select audio/midi flags here according to platform ;-odac ;;;realtime audio out -iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o soundin.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; choose between mono or stereo file ichn filenchnls p4 ;check number of channels print ichn if ichn == 1 then asig soundin p4 ;mono signal outs asig, asig else ;stereo signal aL, aR soundin p4 outs aL, aR endif endin i 1 0 3 "fox.wav" ;mono signal i 1 5 2 "kickroll.wav" ;stereo signal e manual_src~dfsg/examples/midipitchbend.csd0000644000000000000000000000214312262561504020061 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midipitchbend.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 midinoteonoct p4, p5 ;gets a MIDI note number value as octave-point-decimal value into p4, and MIDI velocity into p5 kvel = p5/127 ;scale midi velocity to 0-1 kpb init 0 midipitchbend kpb printk2 kpb ;display the pitch-bend value when it changes koct = p4+kpb ;add pitchbend values to octave-point-decimal value kcps = cpsoct(koct) ;convert octave-point-decimal value into Hz kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by asig vco2 kenv*kvel, kcps ;velocity value outs asig, asig endin f 0 30 ;runs for 30 seconds f 2 0 4096 10 1 i 1 0 2 8.000 100 ; play these notes from score as well i 1 + 2 8.917 100 e manual_src~dfsg/examples/notnum_complex.csd0000644000000000000000000000176512262561504020336 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in sr = 44100 ksmps = 10 nchnls = 2 ; Set MIDI channel 1 to play instr 1. massign 1, 1 instr 1 ; Returns MIDI note number - an integer in range (0-127) iNum notnum ; Convert MIDI note number to Hz iHz = (440.0*exp(log(2.0)*((iNum)-69.0)/12.0)) ; Generate audio by indexing a table; fixed amplitude. aosc oscil 10000, iHz, 1 ; Since there is no enveloping, there will be clicks. outs aosc, aosc endin ; Generate a Sine-wave to be indexed at audio rate ; by the oscil opcode. f1 0 16384 10 1 ; Keep the score "open" for 1 hour so that MIDI ; notes can allocate new note events, arbitrarily. f0 3600 e manual_src~dfsg/examples/cos.csd0000644000000000000000000000242012262561504016040 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cos.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icos1 = cos(0) ;cosine of 0 is 1 icos2 = cos($M_PI_2) ;cosine of pi/2 (1.5707...) is 0 icos3 = cos($M_PI) ;cosine of pi (3.1415...) is -1 icos4 = cos($M_PI_2 * 3) ;cosine of 3/2pi (4.7123...) is 0 icos5 = cos($M_PI * 2) ;cosine of 2pi (6.2831...) is 1 icos6 = cos($M_PI * 4) ;cosine of 4pi is also 1 as it is periodically to 2pi print icos1, icos2, icos3, icos4, icos5, icos6 endin instr 2 ;cos used in panning, after an example from Hans Mikelson aout vco2 0.8, 220 ; sawtooth kpan linseg p4, p3, p5 ;0 = left, 1 = right kpan = kpan*$M_PI_2 ;range 0-1 becomes 0-pi/2 kpanl = cos(kpan) kpanr = sin(kpan) outs aout*kpanl, aout*kpanr endin i 1 0 0 i 2 0 5 0 1 ;move left to right i 2 6 5 1 0 ;move right to left e manual_src~dfsg/examples/JackoInit.csd0000644000000000000000000000036112262561504017131 0ustar rootroot -n sr = 48000 ksmps = 128 nchnls = 2 0dbfs = 1 instr 1 JackoInit "default", "csound" JackoInfo endin i 1 0 0 e manual_src~dfsg/examples/statevar.csd0000644000000000000000000000133412262561504017110 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o statevar.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 kenv linseg 0,0.1,1, p3-0.2,1, 0.1, 0 ;declick envelope asig buzz .6*kenv, 100, 100, 1 kf expseg 100, p3/2, 5000, p3/2, 1000 ;envelope for filter cutoff ahp,alp,abp,abr statevar asig, kf, 4 outs alp,ahp ; lowpass left, highpass right endin f 1 0 16384 10 1 ;sine wave i1 0 5 e manual_src~dfsg/examples/phasorbnk.csd0000644000000000000000000000225512262561504017251 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o phasorbnk.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Generate a sinewave table. giwave ftgen 1, 0, 1024, 10, 1 ; Instrument #1 instr 1 ; Generate 10 voices. icnt = 10 ; Empty the output buffer. asum = 0 ; Reset the loop index. kindex = 0 ; This loop is executed every k-cycle. loop: ; Generate non-harmonic partials. kcps = (kindex+1)*100+30 ; Get the phase for each voice. aphas phasorbnk kcps, kindex, icnt ; Read the wave from the table. asig table aphas, giwave, 1 ; Accumulate the audio output. asum = asum + asig ; Increment the index. kindex = kindex + 1 ; Perform the loop until the index (kindex) reaches ; the counter value (icnt). if (kindex < icnt) kgoto loop out asum*3000 endin ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/FLhvsBox.csd0000644000000000000000000000110712262561504016750 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O sr=44100 ksmps=128 nchnls=2 ;Example by Andres Cabrera 2007 inumlinesX = 3 inumlinesY = 3 iwidth = 200 iheight = 200 ix = 20 iy = 20 FLpanel "FLhvsBox", 400, 240, -1, -1, 5, 1, 1 ihandle FLhvsBox inumlinesX, inumlinesY, iwidth, iheight, ix, iy ; [, image] FLpanelEnd FLrun 0dbfs = 1 instr 1 endin i 1 0 120 e manual_src~dfsg/examples/logicOR.csd0000644000000000000000000000133312262561504016614 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and virtual midi keyboard ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ;after a UDO from Rory Walsh sr = 44100 ksmps = 32 nchnls = 2 instr 1 ;displays notes, midi channel and control number information kstatus, kchan, kdata1, kdata2 midiin k1 changed kstatus k2 changed kchan k3 changed kdata1 k4 changed kdata2 if((k1==1)||(k2==1)||(k3==1)||(k4==1)) then printks "Value:%d ChanNo:%d CtrlNo:%d\n" , 0, kdata2, kchan, kdata1 endif endin i1 0 60 ;print values for 60 seconds e manual_src~dfsg/examples/FLslidBnk.csd0000644000000000000000000000256212262561504017073 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLslidBnk.wav -W ;;; for file output any platform sr = 44100 kr = 441 ksmps = 100 nchnls = 1 gitypetable ftgen 0, 0, 8, -2, 1, 1, 3, 3, 5, 5, 7, 7 giouttable ftgen 0, 0, 8, -2, 0, 0.2, 0.3, 0.4, 0.5, 0.6, 0.8, 1 FLpanel "Slider Bank", 400, 380, 50, 50 ;Number of sliders inum = 8 ; Table to store output iouttable = giouttable ; Width of the slider bank in pixels iwidth = 350 ; Height of the slider in pixels iheight = 160 ; Distance of the left edge of the slider ; from the left edge of the panel ix = 30 ; Distance of the top edge of the slider ; from the top edge of the panel iy = 10 ; Table containing fader types itypetable = gitypetable FLslidBnk "1@2@3@4@5@6@7@8", inum , iouttable , iwidth , iheight , ix \ , iy , itypetable FLslidBnk "1@2@3@4@5@6@7@8", inum , iouttable , iwidth , iheight , ix \ , iy + 200 , -23 ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 ;Dummy instrument endin ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e manual_src~dfsg/examples/cpsmidinn.csd0000644000000000000000000000167312262561504017251 0ustar rootroot ; Prints a table showing the equivalents of all Midi ; note numbers from 0-127 in cycles-per-second, ; octave.decimal, and octave.pitchclass units. ; Select audio/midi flags here according to platform. ; This example produces no audio, so we render in ; non-realtime and turn off sound to disk: -n instr 1 ; i-time loop to print conversion table imidiNN = 0 loop1: icps = cpsmidinn(imidiNN) ioct = octmidinn(imidiNN) ipch = pchmidinn(imidiNN) print imidiNN, icps, ioct, ipch imidiNN = imidiNN + 1 if (imidiNN < 128) igoto loop1 endin instr 2 ; test k-rate converters kMiddleC = 60 kcps = cpsmidinn(kMiddleC) koct = octmidinn(kMiddleC) kpch = pchmidinn(kMiddleC) printks "%d %f %f %f\n", 1.0, kMiddleC, kcps, koct, kpch endin i1 0 0 i2 0 0.1 e manual_src~dfsg/examples/vlowres.csd0000644000000000000000000000167112262561504016764 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vlowres.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp init p4 asig vco2 kamp, 110 ;saw wave kfco line 30, p3, 300 ;vary the cutoff frequency from 30 to 300 Hz. kres = 20 ksep = p5 ;different resonance values iord = p6 ;and different number of filters aout vlowres asig, kfco, kres, iord, ksep aclp clip aout, 1, 1 ;avoid distortion outs aclp, aclp endin f 1 0 16384 10 1 ;sine s i 1 0 10 .1 5 2 ;compensate volume and i 1 + 10 .1 25 2 ;number of filters = 2 s i 1 0 10 .01 5 6 ;compensate volume and i 1 + 10 .04 15 6 ;number of filters = 6 e manual_src~dfsg/examples/gen49.csd0000644000000000000000000000111012262561504016175 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen49.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = sr/ftlen(1) asig oscil .8, kcps, 1 outs asig, asig endin f 1 0 131072 49 "beats.mp3" 0 1 ;read an audio file (using GEN49). i 1 0 2 e manual_src~dfsg/examples/trigger.csd0000644000000000000000000000144112262561504016721 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trigger.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kmtr lfo 1, 1, 1 ;triangle wave kmode = p4 ktr trigger kmtr, .5, kmode printk2 ktr schedkwhen ktr, 0, 3, 2, 0, .3 endin instr 2 aenv linseg 0,p3*.1,1,p3*.3,1,p3*.6,0 ;envelope a1 poscil .3*aenv, 1000, 1 outs a1, a1 endin f 1 0 16384 10 1 ;sine i 1 0 3 0 ;down-up i 1 4 3 2 ;down-up & up=down e manual_src~dfsg/examples/fluidCCi.csd0000644000000000000000000000201212262561504016733 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidCCi.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "19Trumpet.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 56 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidCCi giengine, 1, 93, 127 ;full chorus & fluidCCi giengine, 1, 91, 127 ;full reverb! fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 7 asigl, asigr fluidOut giengine outs asigl*imvol, asigr*imvol endin i 1 0 5 60 100 ;play one note from score and... i 99 0 60 ;play virtual keyboard for 60 sec. e manual_src~dfsg/examples/readclock.csd0000644000000000000000000000166212262561504017212 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o readclock.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Start clock #1. clockon 1 ; Do something that keeps Csound busy. a1 oscili 10000, 440, 1 out a1 ; Stop clock #1. clockoff 1 ; Print the time accumulated in clock #1. i1 readclock 1 print i1 endin ; Initialize the function tables. ; Table 1: an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for one second starting at 0:00. i 1 0 1 ; Play Instrument #1 for one second starting at 0:01. i 1 1 1 ; Play Instrument #1 for one second starting at 0:02. i 1 2 1 e manual_src~dfsg/examples/compress.csd0000644000000000000000000000157012262561504017114 0ustar rootroot ; Select audio/midi flags here according to platform -odac -iadc ;;;RT audio out and in ; For Non-realtime ouput leave only the line below: ; -o compress.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 ; uncompressed signal asig diskin2 "beats.wav", 1, 0, 1 outs asig, asig endin instr 2 ; compressed signal. ; Use the "beats.wav" audio file and a mic avoice in asig diskin2 "beats.wav", 1, 0, 1 ; duck the audio signal "beats.wav" with your voice. kthresh = 0 kloknee = 40 khiknee = 60 kratio = 3 katt = 0.1 krel = .5 ilook = .02 asig compress asig, avoice, kthresh, kloknee, khiknee, kratio, katt, krel, ilook ; voice-activated compressor outs asig, asig endin i 1 0 5 i 2 6 21 e manual_src~dfsg/examples/hilbert.csd0000644000000000000000000000333012262561504016706 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o hilbert.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = p3 ; Initial amount of frequency shift. ; It can be positive or negative. ibegshift = p4 ; Final amount of frequency shift. ; It can be positive or negative. iendshift = p5 ; A simple envelope for determining the ; amount of frequency shift. kfreq linseg ibegshift, idur, iendshift ; Use the sound of your choice. ain diskin2 "beats.wav", 1, 0, 1 ; Phase quadrature output derived from input signal. areal, aimag hilbert ain ; Quadrature oscillator. asin oscili 1, kfreq, 1 acos oscili 1, kfreq, 1, .25 ; Use a trigonometric identity. ; See the references for further details. amod1 = areal * acos amod2 = aimag * asin ; Both sum and difference frequencies can be ; output at once. ; aupshift corresponds to the sum frequencies. aupshift = (amod1 - amod2) * 0.7 ; adownshift corresponds to the difference frequencies. adownshift = (amod1 + amod2) * 0.7 ; Notice that the adding of the two together is ; identical to the output of ring modulation. outs aupshift, aupshift endin ; Sine table for quadrature oscillator. f 1 0 16384 10 1 ; Starting with no shift, ending with all ; frequencies shifted up by 2000 Hz. i 1 0 6 0 2000 ; Starting with no shift, ending with all ; frequencies shifted down by 250 Hz. i 1 7 6 0 -250 e manual_src~dfsg/examples/pvsftw.csd0000644000000000000000000000161312262561504016610 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsftw.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 inbins = 512 ifn ftgen 0,0,inbins,10,1 ; make ftable fsrc pvsdiskin "fox.pvx", 1, 1 ; read PVOCEX file kflag pvsftw fsrc,ifn ; export amps to table, kamp init 0 if kflag==0 kgoto contin ; only proc when frame is ready tablew kamp,1,ifn ; kill lowest bins, for obvious effect tablew kamp,2,ifn tablew kamp,3,ifn tablew kamp,4,ifn ; read modified data back to fsrc pvsftr fsrc,ifn contin: ; and resynth aout pvsynth fsrc outs aout, aout endin i 1 0 4 e manual_src~dfsg/examples/butterlp.csd0000644000000000000000000000117312262561504017121 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o butterlp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; White noise signal asig rand 0.5 outs asig, asig endin instr 2 ; filtered noise asig rand 0.7 alp butterlp asig, 1000 ;cutting frequencies above 1 KHz outs alp, alp endin i 1 0 2 i 2 2.5 2 e manual_src~dfsg/examples/gen03.csd0000644000000000000000000000406212262561504016174 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen03.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;example by Russell Pinkston - Univ. of Texas (but slightly modified) gisine ftgen 1, 0, 16384, 10, 1 ;sine wave instr 1 ihertz = cpspch(p4) ipkamp = p5 iwsfn = p6 ;waveshaping function inmfn = p7 ;normalization function aenv linen 1, .01, p3, .1 ;overall amp envelope actrl linen 1, 2, p3, 2 ;waveshaping index control aindex poscil actrl/2, ihertz, gisine ;sine wave to be distorted asignal tablei .5+aindex, iwsfn, 1 ;waveshaping anormal tablei actrl, inmfn,1 ;amplitude normalization asig = asignal*anormal*ipkamp*aenv asig dcblock2 asig ;get rid of possible DC outs asig, asig endin ; first four notes are specific Chebyshev polynomials using gen03. The values were obtained from Dodge page 147 f4 0 513 3 -1 1 0 1 ; First-order Chebyshev: x f5 0 257 4 4 1 ; Normalizing function for fn4 f6 0 513 3 -1 1 -1 0 2 ; Second-order Chebyshev: 2x2 - 1 f7 0 257 4 6 1 ; Normalizing function for fn6 f8 0 513 3 -1 1 0 -3 0 4 ; Third-order Chebyshev: 4x3 - 3x f9 0 257 4 8 1 ; Normalizing function for fn8 f10 0 513 3 -1 10 0 -7 0 56 0 -112 0 64 ; Seventh-order Chebyshev: 64x7 - 112x5 + 56x3 - 7x f11 0 257 4 10 1 ; Normalizing function for fn10 f12 0 513 3 -1 1 5 4 3 2 2 1 ; a 4th order polynomial function over the x-interval -1 to 1 f13 0 257 4 12 1 ; Normalizing function for fn12 ; five notes with same fundamental, different waveshape & normalizing functions ; pch amp wsfn nmfn i1 0 3 8.00 .7 4 5 i1 + . . . 6 7 i1 + . . . 8 9 i1 + . . . 10 11 i1 + . . . 12 13 e manual_src~dfsg/examples/filevalid.csd0000644000000000000000000000133712262561504017221 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o filevalid.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 Sfile strget p4 ivld filevalid Sfile if ivld>0 then asig diskin2 Sfile, 1 outs asig, asig else printf_i "Audiofile '%s' does not exist!\n", 1, Sfile endif endin i 1 0 3 "frox.wav";file does not exist!!! i 1 + 3 "fox.wav" ;but this one certainly does... e manual_src~dfsg/examples/lposcil.csd0000644000000000000000000000144512262561504016727 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lposcil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 1.5 ; a fifth up kloop = 0 ; loop start time in samples kend = 45000 ; loop end time in samples asig lposcil 1, kcps, kloop, kend, 1 outs asig, asig endin ; Its table size is deferred, ; and format taken from the soundfile header. f 1 0 0 1 "beats.wav" 0 0 0 ; Play Instrument #1 for 6 seconds. ; This will loop the drum pattern several times. i 1 0 6 e manual_src~dfsg/examples/gendy.csd0000644000000000000000000000077612262561504016376 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aout gendy 0.7, 1, 1, 1, 1, 20, 1000, 0.5, 0.5 outs aout, aout endin i1 0 10 e manual_src~dfsg/examples/vdelayxq.csd0000644000000000000000000000203712262561504017115 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelayxq.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 ims = .5 ;maximum delay time in seconds iws = 1024 ;window size adl = .5 aout1 diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav aout2 diskin2 "fox.wav", 1, 0, 1 ;loop fox.wav aout3 diskin2 "Church.wav", 1, 0, 1 ;loop Church.wav aout4 diskin2 "flute.aiff", 1, 0, 1 ;loop flute.aiff a2 poscil .1, .5, 1 ;make an LFO, 1 cycle per 2 seconds adl = a2 + ims/2 ;offset the LFO so that it is positive aout1, aout2, aout3, aout4 vdelayxq aout1, aout2, aout3, aout4, adl, ims, iws; Use the LFO to control delay time outq aout1, aout2, aout3, aout4 endin f1 0 8192 10 1 i 1 0 10 e manual_src~dfsg/examples/syncloop.csd0000644000000000000000000000135312262561504017126 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o syncloop.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 iolaps = 2 igrsize = 0.01 ifreq = iolaps/igrsize ips = 1/iolaps istr = p4 /* timescale */ ipitch = 1 /* pitchscale */ asig syncloop 1, ifreq, ipitch, igrsize, ips*istr, .3, .75, 1, 2, iolaps outs asig, asig endin f1 0 0 1 "beats.wav" 0 0 0 f2 0 8192 20 2 1 i1 0 6 .5 i1 7 6 .15 e manual_src~dfsg/examples/fmmetal.csd0000644000000000000000000000142012262561504016700 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmmetal.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 440 kvdepth = 0 kvrate = 0 ifn1 = 1 ifn2 = 2 ifn3 = 2 ifn4 = 1 ivfn = 1 kc2 = p5 kc1 line p4, p3, 1 asig fmmetal .5, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn outs asig, asig endin ; sine wave. f 1 0 32768 10 1 ; the "twopeaks.aiff" audio file. f 2 0 256 1 "twopeaks.aiff" 0 0 0 i 1 0 4 6 5 i 1 5 4 .2 10 e manual_src~dfsg/examples/vdelay.csd0000644000000000000000000000135312262561504016544 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelay.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ims = 100 ;maximum delay time in msec aout poscil .8, 220, 1 ;make a signal a2 poscil3 ims/2, 1/p3, 1 ;make an LFO a2 = a2 + ims/2 ;offset the LFO so that it is positive asig vdelay aout, a2, ims ;use the LFO to control delay time outs asig, asig endin f1 0 8192 10 1 ;sine wave i 1 0 5 e manual_src~dfsg/examples/xscans.csd0000644000000000000000000000571512262561504016565 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o xscans.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;the matrices can be found in /manual/examples instr 1 ; Plain scanned syntnesis a0 = 0 xscanu 1, .01, 6, 2, "128,8-cylinderX", 4, 5, 2, .1, .1, -.01, .1, .5, 0, 0, a0, 0, 0 a1 xscans .7, cpspch(p4), 7, 0, 1 outs a1, a1 endin instr 2 ; Scan synthesis with audio injection and dual scan paths ; tap the mic or sing to inject audio into the resonators a0,aa ins a0 = a0/.8 xscanu 1, .01, 6, 2, "128,8-gridX", 14, 5, 2, .01, .05, -.05, .1, .5, 0, 0, a0, 0, 0 a1 xscans .5, cpspch(7.00), 7, 0, 1 a2 xscans .5, cpspch(7.001), 77, 0, 1 outs a1+a2,a1+a2 endin instr 3 ; Vibrating structure with audio injection ; Tap the MIC - to inject audio into the resonators a0,aa ins a0 = a0/.8 xscanu 1, .01, 6, 2, "128-stringcircularX", 14, 5, 2, .01, .05, -.05, .25, .75, 0, 0, a0, 0, 1 endin instr 4 ; Modulated scanners i1 bexprnd 5 i2 bexprnd 1 ko oscil i1, i2, 9 ka1 oscili .5, .15*8, p7 ka2 oscili .5, .15*8, p8 kf oscili 1, .15, p4 kf = 2^(kf/12)*p6*440+ko a1 xscans p9*ka1, kf+i1, 777, 1, 1 a2 xscans p9*ka2, (kf+i1)*2.1, 77, 1, 1 outs a1+a2, a1+a2 endin ; Initial condition f1 0 128 7 0 64 1 64 0 ; Masses f2 0 128 -7 1 128 1 ; Centering force f4 0 128 -7 0 128 2 f14 0 128 -7 2 64 0 64 2 ; Damping f5 0 128 -7 1 128 1 ; Initial velocity f6 0 128 -7 -.0 128 .0 ; Trajectories f7 0 128 -5 .001 128 128 f777 0 128 -23 "128-stringcircular" f77 0 128 -23 "128-spiral-8,16,128,2,1over2" ; Sine f9 0 16384 10 1 ; Pitch tables f100 0 1024 -7 +3 128 +3 128 -2 128 -2 128 +0 128 +0 128 -4 128 -4 128 +3 f101 0 1024 -7 -2 128 -2 128 -2 128 -2 128 -5 128 -5 128 -4 128 -4 128 -2 f102 0 1024 -7 +3 128 +3 128 +2 128 +2 128 +0 128 +0 128 +0 128 +0 128 +3 f103 0 1024 -7 +7 128 +7 128 +5 128 +5 128 +3 128 +3 128 +3 128 +3 128 +7 ; Amplitude tables f200 0 1024 7 1 128 0 128 0 127 0 1 1 128 0 128 0 127 0 1 1 128 0 127 0 1 1 f201 0 1024 7 0 127 0 1 1 127 0 1 1 128 0 127 0 1 1 127 0 1 1 128 0 127 0 1 1 127 0 1 1 f202 0 1024 7 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 f203 0 1024 7 1 1024 0 ;---------------------------------- ; Note list i1 0 10 6.00 1 s i2 1 10 s i3 1 23 i4 1 23 101 1 .5 200 202 1.5 i4 . . 102 0 .5 200 201 1 i4 . . 103 0 .5 200 201 1 i4 . . 100 0 .25 200 200 2 e manual_src~dfsg/examples/nrpn.csd0000644000000000000000000000115312262561504016233 0ustar rootroot ; Select audio/midi flags here according to platform -odac -Q1 ;;;realtime audio out with MIDI out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; change attack time of external synth initc7 1, 6, 0 ; set controller 6 to 0 nrpn 1, 99, 1 ; set MSB nrpn 1, 98, 99 ; set LSB katt ctrl7 1, 6, 1, 127 ; DataEntMSB idur = 2 noteondur2 1, 60, 100, idur ; play note on synth endin i 1 0 3 e manual_src~dfsg/examples/transeg.csd0000644000000000000000000000127312262561504016724 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o transeg.wav -W ;;; for file output any platform sr = 44100 ksmps = 128 nchnls = 2 0dbfs = 1 instr 1 ;p4 and p5 determine the type of curve for each ;section of the envelope kenv transeg 0.01, p3*0.25, p4, 1, p3*0.75, p5, 0.01 a1 oscil kenv, 440, 1 outs a1, a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 i 1 0 2 2 2 i 1 + . 5 5 i 1 + . 1 1 i 1 + . 0 0 i 1 + . -2 -2 i 1 + . -2 2 i 1 + . 2 -2 e manual_src~dfsg/examples/vcomb.csd0000644000000000000000000000376412262561504016376 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc -M0 ;;;RT audio I/O with MIDI in ; Example by Jonathan Murphy and Charles Gran 2007 sr = 44100 ksmps = 10 nchnls = 2 ; new, and important. Make sure that midi note events are only ; received by instruments that actually need them. ; turn default midi routing off massign 0, 0 ; route note events on channel 1 to instr 1 massign 1, 1 ; Define your midi controllers #define C1 #21# #define C2 #22# #define C3 #23# ; Initialize MIDI controllers initc7 1, $C1, 0.5 ;delay send initc7 1, $C2, 0.5 ;delay: time to zero initc7 1, $C3, 0.5 ;delay: rate gaosc init 0 ; Define an opcode to "smooth" the MIDI controller signal opcode smooth, k, k kin xin kport linseg 0, 0.0001, 0.01, 1, 0.01 kin portk kin, kport xout kin endop instr 1 ; Generate a sine wave at the frequency of the MIDI note that triggered the intrument ifqc cpsmidi iamp ampmidi 10000 aenv linenr iamp, .01, .1, .01 ;envelope a1 oscil aenv, ifqc, 1 ; All sound goes to the global variable gaosc gaosc = gaosc + a1 endin instr 198 ; ECHO kcmbsnd ctrl7 1, $C1, 0, 1 ;delay send ktime ctrl7 1, $C2, 0.01, 6 ;time loop fades out kloop ctrl7 1, $C3, 0.01, 1 ;loop speed ; Receive MIDI controller values and then smooth them kcmbsnd smooth kcmbsnd ktime smooth ktime kloop smooth kloop imaxlpt = 1 ;max loop time ; Create a variable reverberation (delay) of the gaosc signal acomb vcomb gaosc, ktime, kloop, imaxlpt, 1 aout = (acomb * kcmbsnd) + gaosc * (1 - kcmbsnd) outs aout, aout gaosc = 0 endin f1 0 16384 10 1 i198 0 10000 e manual_src~dfsg/examples/distort.csd0000644000000000000000000000136612262561504016754 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o distort.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifn ftgen 0,0, 257, 9, .5,1,270 ; define a sigmoid, or better ;gifn ftgen 0,0, 257, 9, .5,1,270,1.5,.33,90,2.5,.2,270,3.5,.143,90,4.5,.111,270 instr 1 kdist line 0, p3, 2 ; and over 10 seconds asig poscil 0.3, 440, 1 aout distort asig, kdist, gifn ; gradually increase the distortion outs aout, aout endin f 1 0 16384 10 1 i 1 0 10 e manual_src~dfsg/examples/luamoog.csd0000644000000000000000000002004212262561504016717 0ustar rootroot sr = 48000 ksmps = 100 nchnls = 1 gibegan rtclock lua_opdef "moogladder", {{ local ffi = require("ffi") local math = require("math") local string = require("string") local csoundApi = ffi.load('csound64.dll.5.2') ffi.cdef[[ int csoundGetKsmps(void *); double csoundGetSr(void *); struct moogladder_t { double *out; double *inp; double *freq; double *res; double *istor; double sr; double ksmps; double thermal; double f; double fc; double fc2; double fc3; double fcr; double acr; double tune; double res4; double input; double i; double j; double k; double kk; double stg[6]; double delay[6]; double tanhstg[6]; }; ]] local moogladder_ct = ffi.typeof('struct moogladder_t *') function moogladder_init(csound, opcode, carguments) local p = ffi.cast(moogladder_ct, carguments) p.sr = csoundApi.csoundGetSr(csound) p.ksmps = csoundApi.csoundGetKsmps(csound) if p.istor[0] == 0 then for i = 0, 5 do p.delay[i] = 0.0 end for i = 0, 3 do p.tanhstg[i] = 0.0 end end return 0 end function moogladder_kontrol(csound, opcode, carguments) local p = ffi.cast(moogladder_ct, carguments) -- transistor thermal voltage p.thermal = 1.0 / 40000.0 if p.res[0] < 0.0 then p.res[0] = 0.0 end -- sr is half the actual filter sampling rate p.fc = p.freq[0] / p.sr p.f = p.fc / 2.0 p.fc2 = p.fc * p.fc p.fc3 = p.fc2 * p.fc -- frequency & amplitude correction p.fcr = 1.873 * p.fc3 + 0.4955 * p.fc2 - 0.6490 * p.fc + 0.9988 p.acr = -3.9364 * p.fc2 + 1.8409 * p.fc + 0.9968 -- filter tuning p.tune = (1.0 - math.exp(-(2.0 * math.pi * p.f * p.fcr))) / p.thermal p.res4 = 4.0 * p.res[0] * p.acr -- Nested 'for' loops crash, not sure why. -- Local loop variables also are problematic. -- Lower-level loop constructs don't crash. p.i = 0 while p.i < p.ksmps do p.j = 0 while p.j < 2 do p.k = 0 while p.k < 4 do if p.k == 0 then p.input = p.inp[p.i] - p.res4 * p.delay[5] p.stg[p.k] = p.delay[p.k] + p.tune * (math.tanh(p.input * p.thermal) - p.tanhstg[p.k]) else p.input = p.stg[p.k - 1] p.tanhstg[p.k - 1] = math.tanh(p.input * p.thermal) if p.k < 3 then p.kk = p.tanhstg[p.k] else p.kk = math.tanh(p.delay[p.k] * p.thermal) end p.stg[p.k] = p.delay[p.k] + p.tune * (p.tanhstg[p.k - 1] - p.kk) end p.delay[p.k] = p.stg[p.k] p.k = p.k + 1 end -- 1/2-sample delay for phase compensation p.delay[5] = (p.stg[3] + p.delay[4]) * 0.5 p.delay[4] = p.stg[3] p.j = p.j + 1 end p.out[p.i] = p.delay[5] p.i = p.i + 1 end return 0 end }} /* Moogladder - An improved implementation of the Moog ladder filter DESCRIPTION This is an new digital implementation of the Moog ladder filter based on the work of Antti Huovilainen, described in the paper \"Non-Linear Digital Implementation of the Moog Ladder Filter\" (Proceedings of DaFX04, Univ of Napoli). This implementation is probably a more accurate digital representation of the original analogue filter. This is version 2 (revised 14/DEC/04), with improved amplitude/resonance scaling and frequency correction using a couple of polynomials,as suggested by Antti. SYNTAX ar Moogladder asig, kcf, kres PERFORMANCE asig - input signal kcf - cutoff frequency (Hz) kres - resonance (0 - 1). CREDITS Victor Lazzarini */ opcode moogladderu, a, akk asig, kcf, kres xin setksmps 1 ipi = 4 * taninv(1) /* filter delays */ az1 init 0 az2 init 0 az3 init 0 az4 init 0 az5 init 0 ay4 init 0 amf init 0 if kres > 1 then kres = 1 elseif kres < 0 then kres = 0 endif /* twice the \'thermal voltage of a transistor\' */ i2v = 40000 /* sr is half the actual filter sampling rate */ kfc = kcf/sr kf = kcf/(sr*2) /* frequency & amplitude correction */ kfcr = 1.8730 * (kfc^3) + 0.4955 * (kfc^2) - 0.6490 * kfc + 0.9988 kacr = -3.9364 * (kfc^2) + 1.8409 * kfc + 0.9968; /* filter tuning */ k2vg = i2v * (1 - exp(-2 * ipi * kfcr * kf)) /* cascade of 4 1st order sections */ ay1 = az1 + k2vg * (tanh((asig - 4 * kres * amf * kacr) / i2v) - tanh(az1 / i2v)) az1 = ay1 ay2 = az2 + k2vg * (tanh(ay1 / i2v) - tanh(az2 / i2v )) az2 = ay2 ay3 = az3 + k2vg * (tanh(ay2 / i2v) - tanh(az3 / i2v)) az3 = ay3 ay4 = az4 + k2vg * (tanh(ay3 / i2v) - tanh(az4 / i2v)) az4 = ay4 /* 1/2-sample delay for phase compensation */ amf = (ay4 + az5) *0.5 az5 = ay4 /* oversampling */ ay1 = az1 + k2vg * (tanh((asig - 4 * kres * amf * kacr) / i2v) - tanh(az1 / i2v)) az1 = ay1 ay2 = az2 + k2vg * (tanh(ay1 / i2v) - tanh(az2 / i2v )) az2 = ay2 ay3 = az3 + k2vg * (tanh(ay2 / i2v) - tanh(az3 / i2v)) az3 = ay3 ay4 = az4 + k2vg * (tanh(ay3 / i2v) - tanh(az4 / i2v)) az4 = ay4 amf = (ay4 + az5) * 0.5 az5 = ay4 xout amf endop instr 1 prints "No filter.\n" kfe expseg 500, p3*0.9, 1800, p3*0.1, 3000 kenv linen 10000, 0.05, p3, 0.05 asig buzz kenv, 100, sr/(200), 1 ; afil moogladder asig, kfe, 1 out asig endin instr 2 prints "Native moogladder.\n" kfe expseg 500, p3*0.9, 1800, p3*0.1, 3000 kenv linen 10000, 0.05, p3, 0.05 asig buzz kenv, 100, sr/(200), 1 afil moogladder asig, kfe, 1 out afil endin instr 3 prints "UDO moogladder.\n" kfe expseg 500, p3*0.9, 1800, p3*0.1, 3000 kenv linen 10000, 0.05, p3, 0.05 asig buzz kenv, 100, sr/(200), 1 afil moogladderu asig, kfe, 1 out afil endin instr 4 prints "Lua moogladder.\n" kres init 1 istor init 0 kfe expseg 500, p3*0.9, 1800, p3*0.1, 3000 kenv linen 10000, 0.05, p3, 0.05 asig buzz kenv, 100, sr/(200), 1 afil init 0 lua_ikopcall "moogladder", afil, asig, kfe, kres, istor out afil endin instr 5 giended rtclock ielapsed = giended - gibegan print ielapsed gibegan rtclock endin f 1 0 65536 10 1 i 5.1 0 1 i 4 1 20 i 5.2 21 1 i 4 22 20 i 5.3 42 1 i 2 43 20 i 5.4 63 1 i 2 64 20 i 5.5 84 1 i 3 85 20 i 5.6 105 1 i 3 106 20 i 5.7 126 1 i 1 127 20 i 5.8 147 1 i 1 148 20 i 5.9 168 1 i 4 169 20 i 4 170 20 i 4 171 20 e manual_src~dfsg/examples/rnd31_time.csd0000644000000000000000000000135112262561504017223 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd31_time.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; i-rate random numbers with linear distribution, ; seeding from the current time. (Note that the seed ; was used only in the first call.) i1 rnd31 1, 0.5, 0 i2 rnd31 1, 0.5 i3 rnd31 1, 0.5 print i1 print i2 print i3 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/opa.csd0000644000000000000000000000130412262561504016033 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o a.wav -W ;;; for file output any platform ; hear the difference between instr.1 and 2 sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;sine wave at k-rate ksig oscil 0.8, 440, 1 ; k-rate to the audio-rate conversion asig = a(ksig) outs asig, asig endin instr 2 ;sine wave at a-rate asig oscil 0.8, 440, 1 outs asig, asig endin ; sine wave. f 1 0 16384 10 1 i 1 0 2 i 2 2 2 e manual_src~dfsg/examples/hrtfstat.csd0000644000000000000000000000157712262561504017127 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o hrtfstat.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasrc init 0 instr 1 ;a plucked string kamp = p4 kcps = cpspch(p5) icps = cpspch(p5) a1 pluck kamp, kcps, icps, 0, 1 gasrc = a1 endin instr 10;uses output from instr1 as source aleft,aright hrtfstat gasrc, 90,0, "hrtf-44100-left.dat","hrtf-44100-right.dat" outs aleft, aright clear gasrc endin i1 0 2 .7 8.00 ; Play Instrument 1: a plucked string i1 .5 2 .7 8.00 i1 1 2 .7 8.00 i1 2 2 .7 7.00 i10 0 12 ; Play Instrument 10 for 2 seconds. manual_src~dfsg/examples/vtablei.csd0000644000000000000000000000107112262561504016703 0ustar rootroot -odac -B441 -b441 sr = 44100 kr = 100 ksmps = 441 nchnls = 2 gindx init 0 instr 1 kindex init 0 ktrig metro 0.5 if ktrig = 0 goto noevent event "i", 2, 0, 0.5, kindex kindex = kindex + 1 noevent: endin instr 2 iout1 init 0 iout2 init 0 iout3 init 0 iout4 init 0 indx = p4 vtablei indx, 1, 1, 0, iout1,iout2, iout3, iout4 print iout1, iout2, iout3, iout4 turnoff endin f 1 0 32 10 1 i 1 0 20 manual_src~dfsg/examples/grain.csd0000644000000000000000000000201012262561504016347 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o grain.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 insnd = 10 ibasfrq = 44100 / ftlen(insnd) ; Use original sample rate of insnd file kamp expseg .001, p3/2, .1, p3/2, .01 ;a swell in amplitude kpitch line ibasfrq, p3, ibasfrq * .8 kdens line 600, p3, 100 kaoff line 0, p3, .1 kpoff line 0, p3, ibasfrq * .5 kgdur line .4, p3, .01 imaxgdur = .5 asigL grain kamp, kpitch, kdens, kaoff, kpoff, kgdur, insnd, 5, imaxgdur, 0.0 asigR grain kamp, kpitch, kdens, kaoff, kpoff, kgdur, insnd, 5, imaxgdur, 0.0 outs asigL, asigR endin f5 0 512 20 2 ; Hanning window f10 0 16384 1 "beats.wav" 0 0 0 i1 0 15 e manual_src~dfsg/examples/aftouch.csd0000644000000000000000000000077512262561504016720 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out No messages MIDI in -odac -d -M0 ;;;RT audio out with MIDI in sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kaft aftouch 0, 1 printk2 kaft ;aftertouch from music keyboard used for volume control asig oscil 0.7 * kaft, 220, 1 outs asig, asig endin ;sine wave. f 1 0 16384 10 1 i 1 0 30 e manual_src~dfsg/examples/STKBlowBotl.csd0000644000000000000000000000124212262561504017363 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKBlowBotl.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = p4 kv1 line p5, p3, p6 ;gain of noise kv4 line 100, p3, 70 ;volume asig STKBlowBotl cpspch(ipch), 1, 4, kv1, 11, 10, 1, 50, 128, kv4 asig = asig * .5 ;too loud outs asig, asig endin i 1 0 2 9.00 20 100 i 1 + 5 8.03 120 0 e manual_src~dfsg/examples/lowpass2.csd0000644000000000000000000000205212262561504017027 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o lowpass2.wav -W ;;; for file output any platform /* Written by Sean Costello */ ; Orchestra file for resonant filter sweep of a sawtooth-like waveform. sr = 44100 kr = 2205 ksmps = 20 nchnls = 1 instr 1 idur = p3 ifreq = p4 iamp = p5 * .5 iharms = (sr*.4) / ifreq ; Sawtooth-like waveform asig gbuzz 1, ifreq, iharms, 1, .9, 1 ; Envelope to control filter cutoff kfreq linseg 1, idur * 0.5, 5000, idur * 0.5, 1 afilt lowpass2 asig,kfreq, 30 ; Simple amplitude envelope kenv linseg 0, .1, iamp, idur -.2, iamp, .1, 0 out afilt * kenv endin /* Written by Sean Costello */ f1 0 8192 9 1 1 .25 i1 0 5 100 1000 i1 5 5 200 1000 e manual_src~dfsg/examples/STKClarinet.csd0000644000000000000000000000122412262561504017400 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKclarinet.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv5 = p5 ;breath pressure kv1 line p6, p3, p7 ;reed stiffness asig STKClarinet cpspch(p4), 1, 2, kv1, 4, 100, 11, 60, 1, 10, 128, kv5 outs asig, asig endin i 1 0 3 8.00 100 127 10 i 1 + 10 8.08 80 60 100 e manual_src~dfsg/examples/pset.csd0000644000000000000000000000116112262561504016230 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pset.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ;this shows an example with non-midi use pset 1, 0, 1, 220, 0.5 asig poscil p5, p4, 1 outs asig, asig endin f 1 0 1024 10 1 ;sine wave i 1 0 1 i 1 1 1 440 i 1 2 1 440 0.1 e manual_src~dfsg/examples/sym10.txt0000644000000000000000000000066412262561504016303 0ustar rootroot0.00077015980911449011 9.5632670722894754e-05 -0.0086412992770224222 -0.0014653825813050513 0.045927239231092203 0.011609893903711381 -0.15949427888491757 -0.070880535783243853 0.47169066693843925 0.7695100370211071 0.38382676106708546 -0.035536740473817552 -0.0319900568824278 0.049994972077376687 0.0057649120335819086 -0.02035493981231129 -0.00080435893201654491 0.0045931735853118284 5.7036083618494284e-05 -0.00045932942100465878 manual_src~dfsg/examples/tabmorpha.csd0000644000000000000000000000203112262561504017227 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tabmorpha.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 giSine ftgen 0, 0, 8193, 10, 1 ; sine wave giSquare ftgen 0, 0, 8193, 7, 1, 4096, 1, 0, -1, 4096, -1 ; square wave giTri ftgen 0, 0, 8193, 7, 0, 2048, 1, 4096, -1, 2048, 0 ; triangle wave giSaw ftgen 0, 0, 8193, 7, 1, 8192, -1 ; sawtooth wave, downward slope instr 1 iamp = .7 aindex phasor 110 ; read table value at this index aweightpoint = 0 ; set weightpoint atabnum1 line 0, p3, 3 ; morph through all tables atabnum2 = 2 ; set to triangle wave asig tabmorpha aindex, aweightpoint, atabnum1,atabnum2, giSine, giSquare, giTri, giSaw asig = asig*iamp outs asig, asig endin i1 0 10 e manual_src~dfsg/examples/jacko.csd0000644000000000000000000000577012262561504016356 0ustar rootroot csound -m255 -M0 -+rtmidi=null -RWf --midi-key=4 --midi-velocity=5 -o jacko_test.wav ;;;; ;;;; NOTE: this csd must be run after starting "aeolus -t". ;;;; sr = 48000 ; The control rate must be BOTH a power of 2 (for Jack) ; AND go evenly into sr. This is about the only one that works! ksmps = 128 nchnls = 2 0dbfs = 1 JackoInit "default", "csound" ; To use ALSA midi ports, use "jackd -Xseq" ; and use "jack_lsp -A -c" or aliases from JackInfo, ; probably together with information from the sequencer, ; to figure out the damn port names. ; JackoMidiInConnect "alsa_pcm:in-131-0-Master", "midiin" JackoAudioInConnect "aeolus:out.L", "leftin" JackoAudioInConnect "aeolus:out.R", "rightin" JackoMidiOutConnect "midiout", "aeolus:Midi/in" ; Note that Jack enables audio to be output to a regular ; Csound soundfile and, at the same time, to a sound ; card in real time to the system client via Jack. JackoAudioOutConnect "leftout", "system:playback_1" JackoAudioOutConnect "rightout", "system:playback_2" JackoInfo ; Turning freewheeling on seems automatically ; to turn system playback off. This is good! JackoFreewheel 1 JackoOn alwayson "jackin" instr 1 ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ichannel = p1 - 1 itime = p2 iduration = p3 ikey = p4 ivelocity = p5 JackoNoteOut "midiout", ichannel, ikey, ivelocity print itime, iduration, ichannel, ikey, ivelocity endin instr jackin ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; JackoTransport 3, 1.0 aleft JackoAudioIn "leftin" aright JackoAudioIn "rightin" ; Aeolus uses MIDI controller 98 to control stops. ; Only 1 data value byte is used, not the 2 data ; bytes often used with NRPNs. ; The format for control mode is 01mm0ggg: ; mm 10 to set stops, 0, ggg group (or Division, 0 based). ; The format for stop selection is 000bbbbb: ; bbbbb for button number (0 based). ; Mode to enable stops for Divison I: b1100010 (98 ; [this controller VALUE is a pure coincidence]). JackoMidiOut "midiout", 176, 0, 98, 98 ; Stops: Principal 8 (0), Principal 4 (1) , Flote 8 (8) , Flote 2 (10) JackoMidiOut "midiout", 176, 0, 98, 0 JackoMidiOut "midiout", 176, 0, 98, 1 JackoMidiOut "midiout", 176, 0, 98, 8 JackoMidiOut "midiout", 176, 0, 98, 10 ; Sends audio coming in from Aeolus out ; not only to the Jack system out (sound card), ; but also to the output soundfile. ; Note that in freewheeling mode, "leftout" ; and "rightout" simply go silent. JackoAudioOut "leftout", aleft JackoAudioOut "rightout", aright outs aright, aleft endin f 0 30 i 1 1 30 60 60 i 1 2 30 64 60 i 1 3 30 71 60 e 2 manual_src~dfsg/examples/ampmidi.csd0000644000000000000000000000117512262561504016702 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o ampmidi.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;expects MIDI note inputs on channel 1 iamp ampmidi 1 ; scale amplitude between 0 and 1 asig oscil iamp, 220, 1 print iamp outs asig, asig endin ;Dummy f-table for 1 minute f 0 60 ;sine wave. f 1 0 16384 10 1 e manual_src~dfsg/examples/midicontrolchange.csd0000644000000000000000000000162412262561504020752 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midicontrolchange.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; use slider of contr. 7 of virtual keyboard kcont init 1 ; max. volume midicontrolchange 7, kcont, 0, 1; use controller 7, scaled between 0 and 1 printk2 kcont ; Display the key value when it changes and key is pressed kenv madsr 0.5, 0.8, 0.8, 0.5 ; envelope multiplied by asig pluck kenv*kcont, 220, 220, 2, 1 ; value of controller 7 outs asig, asig endin f 0 30 f 2 0 4096 10 1 i 1 10 2 ; play a note from score as well e manual_src~dfsg/examples/pgmassign_advanced.mid0000644000000000000000000000016512262561504021075 0ustar rootrootMThdxMTrk_ AB@AJAZ2Q < ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o linseg.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kcps = cpspch(p4) kenv cossegb 0, 0.25, 1, 1, 0 asig poscil kenv, kcps, giSine outs asig, asig endin instr 2 ; scaling to duration kcps = cpspch(p4) kenv cossegb 0, p3*0.25, 1, p3, 0 asig poscil kenv, kcps, giSine outs asig, asig endin i 1 0 1 7.00 ; = 1 sec, p3 fits exactly i 1 2 2 7.00 ; = 2 sec, p3 truncated at 1 sec i 2 4 1 7.00 ; scales to duration i 2 6 2 7.00 ; of p3 e manual_src~dfsg/examples/wgclar.csd0000644000000000000000000000130512262561504016534 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o wgclar.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 330 kstiff = -0.3 iatt = 0.1 idetk = 0.1 kngain init p4 ;vary breath kvibf = 5.735 kvamp = 0.1 asig wgclar .9, kfreq, kstiff, iatt, idetk, kngain, kvibf, kvamp, 1 outs asig, asig endin f 1 0 16384 10 1 ;sine wave i 1 0 2 0.2 i 1 + 2 0.5 ;more breath e manual_src~dfsg/examples/locsig_stereo.csd0000644000000000000000000000165512262561504020126 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o locsig_stereo.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ga1 init 0 ga2 init 0 ; Instrument #1 instr 1 ; Use the "beats.wav" audio file. aout diskin2 "beats.wav", 1, 0, 1 kdegree line 0, p3, 180 kdistance line 1, p3 , 30 a1, a2 locsig aout, kdegree, kdistance, .1 ar1, ar2 locsend ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a2 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 outs a1, a2 ga1=0 ga2=0 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for five seconds. i 1 0 2 i99 0 5 e manual_src~dfsg/examples/tableiw.csd0000644000000000000000000000227612262561504016714 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tableiw.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 ;generate new values every time the instr is played instr 1 ifn = p4 isize = p5 ithresh = 0.5 itemp ftgen ifn, 0, isize, 21, 2 iwrite_value = 0 i_index = 0 loop_start: iread_value tablei i_index, ifn if iread_value > ithresh then iwrite_value = 1 else iwrite_value = -1 endif tableiw iwrite_value, i_index, ifn loop_lt i_index, 1, isize, loop_start turnoff endin instr 2 ifn = p4 isize = ftlen(ifn) prints "Index\tValue\n" i_index = 0 loop_start: ivalue tablei i_index, ifn prints "%d:\t%f\n", i_index, ivalue loop_lt i_index, 1, isize, loop_start ;read table 1 with our index aout oscili .5, 100, ifn ;use table to play the polypulse outs aout, aout endin i 1 0 1 100 16 i 2 0 2 100 e manual_src~dfsg/examples/fwavblnk.aiff0000644000000000000000000000111612262561504017223 0ustar rootrootFORMNAIFF(c) |0h|0COMM@ DSSNDG %'+06 ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trhighest.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "fox.wav", 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .1, 1, 3, 500 ; partial tracking fhi,kfr,kamp trhighest fst, 1 ; highest freq-track aout tradsyn fhi, 1, 1, 1, 1 ; resynthesis of highest frequency outs aout*40, aout*40 ; compensate energy loss endin f1 0 8192 10 1 ;sine wave i 1 0 3 e manual_src~dfsg/examples/mdelay.csd0000644000000000000000000000155112262561504016533 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1;;;RT audio I/O with MIDI in sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 instr 1 ;Triggered by MIDI notes on channel 1 kstatus init 0 ifund notnum ivel veloc noteondur 1, ifund, ivel, 1 kstatus = kstatus + 1 idel1 = .2 idel2 = .4 idel3 = .6 idel4 = .8 ;make four delay lines mdelay kstatus,1,ifund+2, ivel,idel1 mdelay kstatus,1,ifund+4, ivel,idel2 mdelay kstatus,1,ifund+6, ivel,idel3 mdelay kstatus,1,ifund+8, ivel,idel4 endin ; Dummy ftable f 0 60 manual_src~dfsg/examples/locsend_quad.csd0000644000000000000000000000203612262561504017720 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o locsend_quad.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 4 ga1 init 0 ga2 init 0 ga3 init 0 ga4 init 0 ; Instrument #1 instr 1 ; Use the "beats.wav" audio file. aout diskin2 "beats.wav", 1, 0, 1 kdegree line 0, p3, 360 kdistance line 1, p3 , 10 a1, a2, a3, a4 locsig aout, kdegree, kdistance, .1 ar1, ar2, ar3, ar4 locsend ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 3.5, .5 a2 reverb2 ga2, 3.5, .5 a3 reverb2 ga3, 3.5, .5 a4 reverb2 ga4, 3.5, .5 outq a1, a2, a3, a4 ga1 = 0 ga2 = 0 ga3 = 0 ga4 = 0 endin ; Table #1, a sine wave. f 1 0 16384 10 1 i 1 0 2 i99 0 5 e manual_src~dfsg/examples/lposcilsa2.csd0000644000000000000000000000166112262561504017335 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lposcilsa2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 2 ;only integers are allowed kloop = 0 ;loop start time in samples kend = 45000 ;loop end time in samples aenv expsega 0.01, 0.1, 1, 0.1, 0.5, 0.5, 0.01 ;envelope with fast and short segment aL, aR lposcilsa2 aenv, kcps, kloop, kend, 1 ;use it for amplitude outs aL, aR endin ; Its table size is deferred, ; and format taken from the soundfile header. f 1 0 0 1 "kickroll.wav" 0 0 0 ; Play Instrument #1 for 6 seconds. ; This will loop the drum pattern several times. i 1 0 2 e manual_src~dfsg/examples/scalearray.csd0000644000000000000000000000161112262561504017403 0ustar rootroot ; Select audio/midi flags here according to platform -n ;example by joachim heintz seed 0 instr 1 ;create an array with 10 elements kArr[] init 10 ;fill in random numbers and print them out printks "kArr in maximum range 0..100:\n", 0 kIndx = 0 until kIndx == 10 do kNum random 0, 100 kArr[kIndx] = kNum printf "kArr[%d] = %10f\n", kIndx+1, kIndx, kNum kIndx += 1 od ;scale numbers 0...1 and print them out again scalearray kArr, 0, 1 kIndx = 0 printks "kArr in range 0..1\n", 0 until kIndx == 10 do printf "kArr[%d] = %10f\n", kIndx+1, kIndx, kArr[kIndx] kIndx += 1 od turnoff endin i1 0 0.1 e manual_src~dfsg/examples/toot10.csd0000644000000000000000000000314212262561504016404 0ustar rootroot -odac nchnls = 2 garvbsig init 0 instr 10 iattack = .01 irelease = .2 iwhite = 22050 idur = p3 iamp = p4 iswpstart = p5 isweepend = p6 ibndwidth = p7 ibalance = p8 ; 1 = left, .5 = center, 0 = right irvbgain = p9 kamp linen iamp, iattack, idur, irelease ksweep line iswpstart, idur, isweepend asig rand iwhite afilt reson asig, ksweep, ibndwidth arampsig = kamp * afilt outs arampsig * ibalance, arampsig * (1 - ibalance) garvbsig = garvbsig + arampsig * p9 endin instr 100 irvbtime = p4 asig reverb garvbsig, irvbtime outs asig, asig garvbsig = 0 endin ;ins strt dur rvbtime i100 0 15 1.1 i100 15 10 5 ;ins strt dur amp stsw ndsw bdw bal(0-1) rvsnd i10 0 2 .01 5000 500 20 .15 .1 i10 3 1 .01 1500 5000 30 .95 .1 i10 5 2 .01 850 1100 40 .45 .1 i10 8 2 .01 1100 8000 50 .05 .1 i10 8 .5 .01 5000 1000 30 .35 .2 i10 9 .5 .01 1000 8000 40 .75 .1 i10 11 .5 .01 500 2100 50 .14 .2 i10 12 .5 .01 2100 1220 75 .96 .1 i10 13 .5 .01 1700 3500 100 .45 .2 i10 15 5 .005 8000 800 60 .85 .1 e manual_src~dfsg/examples/tival.csd0000644000000000000000000000213612262561504016377 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tival.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = abs(p3) ;make p3 positive even if p3 is negative in score itiv tival i1 = -1 ;assume this is tied note, so keep fase of oscili tigoto slur ;no reinitialisation on tied notes i1 = 0 ;first note, so reset phase aatt line p4, idur, 0 ;primary envelope slur: if itiv==0 kgoto note ;no expression on first and second note aslur linseg 0, idur*.3, p4, idur*.7, 0 ;envelope for slurred note aatt = aatt + aslur note: asig oscili aatt, p5, 1, i1 outs asig, asig endin f1 0 4096 10 1 ;sine wave i1 0 -5 .8 451 ;p3 = 5 seconds i1 1.5 -1.5 .1 512 i1 3 2 .7 440 ;3 notes together--> duration = 5 seconds e manual_src~dfsg/examples/gen01computed.csd0000644000000000000000000000141112262561504017726 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o gen01computed.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 kcps = 1 ifn = 1 ibas = 1 ; Play the audio sample stored in Table #1. a1 loscil kamp, kcps, ifn, ibas out a1 endin ; Table #1: an audio file (using GEN01). ; Since our table size is 0, Csound will compute it. f 1 0 0 1 "beats.wav" 0 0 0 ; Play Instrument #1 for 2 seconds. i 1 0 2 e manual_src~dfsg/examples/sndwarpst.csd0000644000000000000000000000175212262561504017310 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sndwarpst.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktimewarp line 0, p3, 1 ;length of stereo file "kickroll.wav" kresample init 1 ;playback at the normal speed ibeg = 0 iwsize = 4410 irandw = 441 ioverlap = p4 itimemode = 1 ; Use the ktimewarp parameter as a "time" pointer aL, aR sndwarpst .3, ktimewarp, kresample, 1, ibeg, iwsize, irandw, ioverlap, 2, itimemode aL dcblock aL ;get rid of DC offsets for left channel & aR dcblock aR ;right channel outs aL, aR endin f 1 0 65536 1 "kickroll.wav" 0 0 0 f 2 0 16384 9 0.5 1 0 ;half of a sine wave i 1 0 7 2 ;different overlaps i 1 + 7 5 i 1 + 7 15 e manual_src~dfsg/examples/harmon.csd0000644000000000000000000000140412262561504016541 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o harmon.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aout diskin2 "fox.wav", 1 kestfrq = p4 ;different estimated frequency kmaxvar = 0.1 imode = 1 iminfrq = 100 iprd = 0.02 asig harmon aout, kestfrq, kmaxvar, kestfrq*.5, kestfrq*4, \ imode, iminfrq, iprd outs (asig + aout)*.6, (asig + aout)*.6 ;mix dry&wet signal endin i 1 0 2.7 100 i 1 + . 200 i 1 + . 500 e manual_src~dfsg/examples/maca.csd0000644000000000000000000000264012262561504016161 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mac.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;4 band equalizer klow = p4 ;low gain1 aenv1 oscil 0.2, 1, 4 aenv1 = aenv1 + klow klmid = p5 ;low gain2 aenv2 oscil 0.21, 1.1, 4 aenv2 = aenv2 + klmid kmidh = p6 ;high gain1 aenv3 oscil 0.19, 1.2, 4 aenv3 = aenv3 + kmidh khigh = p7 ;high gain2 aenv4 oscil 0.18, 1.3, 4 aenv4 = aenv4 + khigh ifn = p8 ;table ilc1 table 0, ifn ;low freqency range ilc2 table 1, ifn ;low-mid ihc1 table 2, ifn ;mid-high ihc2 table 3, ifn ;high asig diskin2 "fox.wav", 1 alow1 butterlp asig, ilc1 ;lowpass 1 almid butterlp asig, ilc2 ;lowpass 2 amidh butterhp asig, ihc1 ;highpass 1 ahigh butterhp asig, ihc2 ;highpass 2 aout maca aenv1, alow1, aenv2, almid, aenv3, amidh, aenv4, ahigh outs aout, aout endin f1 0 4 -2 150 300 600 5000 f2 0 4 -2 75 500 1000 10000 f3 0 4 -2 200 700 1500 3000 f4 0 4096 10 1 ; low lowmid midhigh high table i 1 0 2.8 2 1 1 1 1 i 1 3 2.8 2 3 1 1 2 i 1 6 2.8 2 1 2 3 3 e manual_src~dfsg/examples/dssi4cs.csd0000644000000000000000000000225212262561504016633 0ustar rootroot ;use appropriate realtime options ksmps = 256 nchnls = 2 dssilist gihandle dssiinit "amp.so", 0, 1 ;gihandle dssiinit "cmt.so", 30 , 2 ;gihandle2 dssiinit "cmt.so", 8 , 1 ;gihandle dssiinit "delayorama_1402", 0 gihandle2 dssiinit "cmt.so", 49 , 1 ;gihandle dssiinit "freq_tracker_1418.so", 0 , 1, 1 ;gihandle dssiinit "g2reverb.so", 0, 1 ;gihandle2 dssiinit "declip_1195.so", 0, 1 ;gihandle2 dssiinit "revdelay_1605.so", 0, 1 ;gihandle2 dssiinit "tap_chorusflanger.so", 0, 1 ;gihandle2 dssiinit "plate_1423.so", 0, 1 gihandle3 dssiinit "gate_1410.so", 0, 1 ;gihandle3 dssiinit "hexter.so", 0, 1 instr 1 print p4 dssiactivate gihandle, p4 dssiactivate gihandle2, p4 dssiactivate gihandle3, p4 endin instr 2 ain1 inch 1 ain2 inch 2 ;aout1,aout2 dssiaudio gihandle, ain1, ain2 aout1 dssiaudio gihandle, ain1 outs aout1,aout1 endin instr 3 kval linen 1, p3 /3, p3, p3/ 3 dssictls gihandle, p4, kval, 1 endin instr 4 ain1 inch 1 aout1 dssiaudio gihandle2, ain1 outs aout1,aout1 endin i 1 1 1 1 i 2 2 15 ;plugin 1 i 3 3 12 0 ;Control port 0 i 4 8 2 ;plugin 2 e manual_src~dfsg/examples/STKDrummer.csd0000644000000000000000000000114112262561504017250 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M0 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKDrummer.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;STK Drummer - has no controllers but plays samples (11) icps cpsmidi iamp ampmidi 1 asig STKDrummer icps, iamp outs asig, asig endin ; play 5 minutes f0 300 e manual_src~dfsg/examples/ATSinfo.csd0000644000000000000000000000054012262561504016560 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out sr = 44100 ksmps = 32 nchnls = 2 instr 1 ; "fox.ats" is created by atsa inum_partials ATSinfo "fox.ats", 3 print inum_partials endin i 1 0 0 e manual_src~dfsg/examples/STKBeeThree.csd0000644000000000000000000000125612262561504017327 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKBeeThree.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kv1 = p6 ;feedback of operator 4 kv2 line p4, p3, p5 ;gain of operator 3 kv5 line 0, p3, 100 ipch = p7 asig STKBeeThree cpspch(ipch), 1, 2, kv1, 4, kv2, 11, 50, 1, 0, 128, kv5 outs asig, asig endin i 1 0 2 20 100 127 8.00 i 1 + 8 120 0 0 6.09 e manual_src~dfsg/examples/pvsvoc.csd0000644000000000000000000000151412262561504016577 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsvoc.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisaw ftgen 1, 0, 2048, 10, 1, 0.5, 0.3, 0.25, 0.2 ;sawtooth-like instr 1 asig in ;get the signal in asyn poscil .6, 150, gisaw ;excitation signal of 150 Hz famp pvsanal asig, 1024, 256, 1024, 1 ;analyse in signal fexc pvsanal asyn, 1024, 256, 1024, 1 ;analyse excitation signal ftps pvsvoc famp, fexc, 1, 1 ;cross it atps pvsynth ftps ;synthesise it outs atps, atps endin i 1 0 10 e manual_src~dfsg/examples/upsamp.csd0000644000000000000000000000356412262561504016573 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o upsamp.wav -W ;;; for file output any platform ;;with code from Steven Cook / David Akbari, Menno Knevel and Joachim Heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 opcode Decimator, a, akk ;UDO Sample rate / Bit depth reducer ;see http://www.csounds.com/udo/displayOpcode.php?opcode_id=73 setksmps 1 ain, kbit, ksrate xin kbits = 2^kbit ;bit depth (1 to 16) kfold = (sr/ksrate) ;sample rate kin downsamp ain ;convert to kr kin = (kin+0dbfs) ;add DC to avoid (-) kin = kin*(kbits/(0dbfs*2)) ;scale signal level kin = int(kin) ;quantise aout upsamp kin ;convert to sr aout = aout*(2/kbits)-0dbfs ;rescale and remove DC a0ut fold aout, kfold ;resample xout a0ut endop instr 1 ;avoid playing this too loud kbit = p4 ksr = 44100 asig diskin "fox.wav", 1 aout Decimator asig, kbit, ksr printks "bitrate = %d, ", 3, kbit printks "with samplerate = %d\\n", 3, ksr outs aout*.7, aout*.7 endin instr 2 ;moving randomly between different bit values (1 - 6) kbit randomi 1, 6, .5, 1 asig diskin "fox.wav", 1, 0, 1 ;loop play aout Decimator asig, kbit, 44100 printks "bitrate = %f\n", .3, kbit outs aout*.7, aout*.7 endin i 1 0 3 16 ;sounds allright but i 1 + 3 5 ;it's getting worse i 1 + 3 2 ;and worse... i 2 9 22 ;or quality moves randomly e manual_src~dfsg/examples/dssiinit.csd0000644000000000000000000000213512262561504017105 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dssiinit.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 gihandle dssiinit "g2reverb.so", 0, 1 gaout init 0 instr 1 ; activate DSSI dssiactivate gihandle, 1 endin instr 2 ain1 diskin2 "beats.wav", 1 gaout = gaout+(ain1*.3) endin instr 3 dssictls gihandle, 4, 100, 1 ; room 10 to 150 dssictls gihandle, 5, 10, 1 ; reverb time 1 to 20 dssictls gihandle, 6, .5, 1 ; input bandwith 0 to 1 dssictls gihandle, 7, .25, 1 ; damping 0 to 1 dssictls gihandle, 8, 0, 1 ; dry -80 to 0 dssictls gihandle, 9, -10, 1 ; reflections -80 to 0 dssictls gihandle, 10, -15, 1 ; rev. tail -80 to 0 endin instr 4 aout1, aout2 dssiaudio gihandle, gaout, gaout ;get beats.wav and outs aout1,aout2 ; stereo DSSI plugin gaout = 0 endin i 1 0 2 i 2 1 10 i 3 1 10 i 4 0 10 e manual_src~dfsg/examples/spsend.csd0000644000000000000000000000247712262561504016564 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o spsend.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 ;stereo output ga1 init 0 ga2 init 0 instr 1 ;sends different amounts to reverb irev = p6 asig diskin2 "fox.wav", 1 a1, a2, a3, a4 space asig, 0, 0, irev, p4, p5 ;take position values from p4, p5 ar1, ar2, ar3, ar4 spsend ;send to reverb ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a2 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 outs a1, a2 ga1=0 ga2=0 endin ;WITH REVERB ;place the sound in the left speaker and near i1 0 1 -1 1 .1 ;place the sound in the right speaker and far i1 1 1 45 45 .1 ;place the sound equally between left and right and in the middle ground distance i1 2 1 0 12 .1 ;NO REVERB ;place the sound in the left speaker and near i1 6 1 -1 1 0 ;place the sound in the right speaker and far i1 7 1 45 45 0 ;place the sound equally between left and right and in the middle ground distance i1 8 1 0 12 0 i 99 0 12 ;keep reverb active all the time e manual_src~dfsg/examples/STKMandolin.csd0000644000000000000000000000126312262561504017403 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKMandolin.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;body size kv3 = p7 ;sustain asig STKMandolin cpspch(ifrq), 1, 2, kv1, 4, 10, 11, kv3, 1, 100, 128, 100 outs asig, asig endin i 1 0 .3 7.00 100 0 20 i 1 + . 8.00 10 100 20 i 1 + . 8.00 100 0 120 i 1 + 4 8.00 10 10 127 e manual_src~dfsg/examples/fout_poly.csd0000644000000000000000000000245112262561504017300 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o fout_poly.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1 - Play an audio file. instr 1 ; Generate an audio signal using ; the audio file "beats.wav". asig soundin "beats.wav" out asig endin ; Instrument #2 - Create a basic tone. instr 2 iamp = 5000 icps = 440 iphs = 0 ; Create an audio signal. asig oscils iamp, icps, iphs out asig endin ; Instrument #99 - Save the global signal to a file. instr 99 ; Read the csound output buffer aoutput monitor ; Write the output of csound to a headerless ; audio file called "fout_poly.raw". fout "fout_poly.raw", 1, aoutput endin ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 every quarter-second. i 2 0.00 0.1 i 2 0.25 0.1 i 2 0.50 0.1 i 2 0.75 0.1 i 2 1.00 0.1 i 2 1.25 0.1 i 2 1.50 0.1 i 2 1.75 0.1 ; Make sure the global instrument, #99, is running ; during the entire performance (2 seconds). i 99 0 2 e manual_src~dfsg/examples/insremotM.csd0000644000000000000000000000112612262561504017233 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o insremotM.wav -W ;;; for file output any platform nchnls = 1 insremot "192.168.1.100", "192.168.1.101", 1 instr 1 aq barmodel 1, 1, p4, 0.001, 0.23, 5, p5, p6, p7 out aq endin i1 0 0.5 3 0.2 500 0.05 i1 0.5 0.5 -3 0.3 1000 0.05 i1 1.0 0.5 -3 0.4 1000 0.1 i1 1.5 4.0 -3 0.5 800 0.05 e manual_src~dfsg/examples/filesr.csd0000644000000000000000000000165712262561504016553 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o filesr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;after an example from Jonathan Murphy instr 1 ;load sound into an ftable Sfile strcpy "beats.wav" ilen filelen Sfile isr filesr Sfile isamps = ilen * isr ;adjust the length of the table to be a power of two closest ;to the actual size of the sound isize init 1 loop: isize = isize * 2 if (isize < isamps) igoto loop itab ftgen 0, 0, isize, 1, Sfile, 0, 0, 0 prints "sample rate = %f, size = %f\n", isr, isize ;prints them endin i1 0 2 e manual_src~dfsg/examples/zarg.csd0000644000000000000000000000216312262561504016223 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zarg.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Generate a simple sine waveform, with an amplitude ; between 0 and 1. asin oscil 1, 440, 1 ; Send the sine waveform to za variable #1. zaw asin, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read za variable #1, multiply its amplitude by 20,000. a1 zarg 1, 20000 ; Generate audio output. out a1 ; Clear the za variables, get them ready for ; another pass. zacl 0, 1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e manual_src~dfsg/examples/a.csd0000644000000000000000000000162212262561504015477 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o a.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .8*aenv, p4, 1 outs asig, asig endin f1 0 16384 10 1 ;sine wave ;two sections s a 0 0 6 ;advance score 6 seconds i 1 0 2 110 ;these first 2 notes i 1 3 2 220 ;will not sound i 1 6 2 440 i 1 9 2 880 s a 0 3 6 ;advance score 6 seconds, but do this after 3 seconds i 1 0 2 110 ;this will sound, because action time (p2) from a statement = 3 i 1 3 2 220 ;so these 2 notes i 1 6 2 440 ;will not sound i 1 9 2 880 ;and this one will e manual_src~dfsg/examples/pvsanal.csd0000644000000000000000000000221612262561504016723 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsanal.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;pvsanal has no influence when there is no transformation of original sound ifftsize = p4 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1 ;von-Hann window Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape ;fft-analysis of the audio-signal fftblur pvscale fftin, p5 ;scale aout pvsynth fftblur ;resynthesis outs aout, aout endin s i 1 0 3 512 1 ;original sound - ifftsize of pvsanal does not have any influence i 1 3 3 1024 1 ;even with different i 1 6 3 2048 1 ;settings s i 1 0 3 512 1.5 ;but transformation - here a fifth higher i 1 3 3 1024 1.5 ;but with different settings i 1 6 3 2048 1.5 ;for ifftsize of pvsanal e manual_src~dfsg/examples/ptrack.csd0000644000000000000000000000145412262561504016546 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ptrack.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ihop = p4 aout diskin2 "fox.wav",1, 0, 1 kf,ka ptrack aout, ihop ; pitch track with different hopsizes kcps port kf, 0.01 ; smooth freq kamp port ka, 0.01 ; smooth amp ; drive an oscillator asig poscil ampdb(kamp)*0dbfs, kcps, 1 outs asig, asig endin ; simple sine wave f 1 0 4096 10 1 i 1 0 5 128 i 1 6 5 512 i 1 12 5 1024 e manual_src~dfsg/examples/active.csd0000644000000000000000000000223712262561504016535 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o active.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a noisy waveform. instr 1 ; Generate a really noisy waveform. anoisy rand 44100 ; Turn down its amplitude. aoutput gain anoisy, 2500 ; Send it to the output. out aoutput endin ; Instrument #2 - counts active instruments. instr 2 ; Count the active instances of Instrument #1. icount active 1 ; Print the number of active instances. print icount endin ; Start the first instance of Instrument #1 at 0:00 seconds. i 1 0.0 3.0 ; Start the second instance of Instrument #1 at 0:015 seconds. i 1 1.5 1.5 ; Play Instrument #2 at 0:01 seconds, when we have only ; one active instance of Instrument #1. i 2 1.0 0.1 ; Play Instrument #2 at 0:02 seconds, when we have ; two active instances of Instrument #1. i 2 2.0 0.1 e manual_src~dfsg/examples/ftsr.csd0000644000000000000000000000133312262561504016234 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftsr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 itab = p4 isr = ftsr(itab) prints "sampling-rate of table number %d = %d\n", itab, isr endin f 1 0 0 1 "kickroll.wav" 0 0 0 ;stereo file f 2 0 0 1 "ahh.aiff" 0 0 0 ;& different formats f 3 0 0 1 "beats.mp3" 0 0 0 f 4 0 0 1 "beats.ogg" 0 0 0 i 1 0 1 1 i 1 + 1 2 i 1 + 1 3 i 1 + 1 4 e manual_src~dfsg/examples/eqfil.csd0000644000000000000000000000136712262561504016365 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o eqfil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcf = p4 kfe expseg 10, p3*0.9, 1800, p3*0.1, 175 kenv linen .03, 0.05, p3, 0.05 ;low amplitude is needed to avoid clipping asig buzz kenv, kfe, sr/(2*kfe), 1 afil eqfil asig, kcf, 200, 10 outs afil*20, afil*20 endin ; a sine wave. f 1 0 16384 10 1 i 1 0 10 200 ;filter centre freq=200 i 1 + 10 1500 ;filter centre freq=1500 e manual_src~dfsg/examples/plltrack.csd0000644000000000000000000000126312262561504017074 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o plltrack.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kd = p4 a1 diskin2 "fox.wav", 1,0,1 apitch, aloc plltrack a1, kd krms rms a1 krms port krms, 0.01 asig buzz krms, apitch, 10, 1 outs asig, asig ;mix in some dry signal as well endin f1 0 65536 10 1 ;sine wave i 1 0 6 0.1 i 1 7 6 0.3 ;more feedback e manual_src~dfsg/examples/stix.csd0000644000000000000000000000107212262561504016245 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o stix.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 idamp = p4 ;vary damping amount asig stix .5, 0.01, 30, idamp outs asig, asig endin i1 0 1 .3 i1 + 1 > i1 + 1 > i1 + 1 .95 e manual_src~dfsg/examples/outq.csd0000644000000000000000000000234712262561504016254 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outq.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 asig vco2 .01, 110 ; sawtooth waveform at low volume ;filter the first channel kcut1 line 60, p3, 300 ; Vary cutoff frequency kresonance1 = 3 inumlayer1 = 3 asig1 lowresx asig, kcut1, kresonance1, inumlayer1 ;filter the second channel kcut2 line 300, p3, 60 ; Vary cutoff frequency kresonance2 = 3 inumlayer2 = 3 asig2 lowresx asig, kcut2, kresonance2, inumlayer2 ;filter the third channel kcut3 line 30, p3, 100; Vary cutoff frequency kresonance3 = 6 inumlayer3 = 3 asig3 lowresx asig, kcut3, kresonance3, inumlayer3 asig3 = asig3*.1 ; lower volume ;filter the fourth channel kcut4 line 100, p3, 30; Vary cutoff frequency kresonance4 = 6 inumlayer4 = 3 asig4 lowresx asig, kcut4, kresonance4, inumlayer4 asig4 = asig4*.1 ; lower volume outq asig1, asig2, asig3, asig4; output channels 1, 2, 3 & 4 endin i 1 0 5 e manual_src~dfsg/examples/polynomial.csd0000644000000000000000000000556312262561504017452 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o polynomial.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; The polynomial y=x^n where n is odd always produces a curve ; that traverses the range [-1, 1] when the input is within ; the same range. Therefore, we can use one of these curves ; to make a nonlinear phasor that repeatedly reads a table ; from beginning to end like a linear phasor (maintaining ; continuity) but that distorts the waveform in the table. instr 4 ; This instrument demonstrates phase distortion with x^3 idur = p3 iamp = p4 ifreq = p5 itable = p6 aenv linseg 0, .001, 1.0, idur - .051, 1.0, .05, 0 ; declicking envelope aosc phasor ifreq ; create a linear phasor apd polynomial aosc, 0, 0, 0, 1 ; distort the phasor with x^3 aout tablei apd, itable, 1 ; read a sine table with the nonlinear phasor outs aenv*aout*iamp, aenv*aout*iamp endin instr 5 ; This instrument demonstrates phase distortion with x^11 idur = p3 iamp = p4 ifreq = p5 itable = p6 aenv linseg 0, .001, 1.0, idur - .051, 1.0, .05, 0 ; declicking envelope aosc phasor ifreq ; create a linear phasor apd polynomial aosc, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1 ; distort the phasor with x^11 aout tablei apd, itable, 1 ; read a sine table with the nonlinear phasor outs aenv*aout*iamp, aenv*aout*iamp endin instr 6 ; This instrument crossfades between a pure sine and one distorted with x^11 idur = p3 iamp = p4 ifreq = p5 itable = p6 aenv linseg 0, .001, 1.0, idur - .051, 1.0, .05, 0 ; declicking envelope aosc phasor ifreq ; create a linear phasor aout3 tablei aosc, itable, 1 ; read a sine table without the linear phasor apd11 polynomial aosc, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1 ; distort the phasor with x^11 aout11 tablei apd11, itable, 1 ; read a sine table with the nonlinear phasor kamount linseg 1.0, 0.05, 0.9, 1.0, 0.0 ; crossfade between two outputs aout = aout3*kamount + aout11*(1.0 - kamount) outs aenv*aout*iamp, aenv*aout*iamp endin f1 0 16385 10 1 ; sine wave ; descending "just blues" scale t 0 100 i4 0 .333 .7 512 1 i. + . . 448 i. + . . 384 i. + . . 360 i. + . . 341.33 i. + . . 298.67 i. + 2 . 256 s t 0 100 i5 0 .333 .7 512 1 i. + . . 448 i. + . . 384 i. + . . 360 i. + . . 341.33 i. + . . 298.67 i. + 2 . 256 s t 0 100 i6 0 .333 .7 512 1 i. + . . 448 i. + . . 384 i. + . . 360 i. + . . 341.33 i. + . . 298.67 i. + 2 . 256 e manual_src~dfsg/examples/table.csd0000644000000000000000000000160612262561504016350 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o table.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Vary our index linearly from 0 to 1. kndx line 0, p3, 1 ; Read Table #1 with our index. ifn = 1 ixmode = 1 kfreq table kndx, ifn, ixmode ; Generate a sine waveform, use our table values ; to vary its frequency. a1 oscil 20000, kfreq, 2 out a1 endin ; Table #1, a line from 200 to 2,000. f 1 0 1025 -7 200 1024 2000 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e manual_src~dfsg/examples/gendyx-2.csd0000644000000000000000000000124512262561504016715 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kenv expseg 0.001, p3*0.05, 0.5, p3*0.9, 0.5, p3*0.05, 0.001 kc1 linseg 1, p3/2, 12, p3/2, 3 kc2 random 0, 4 seed 20120124 asig gendyx kenv, 1, 3, 0.7, 0.8, 120, 4300, 0.2, 0.7, kc1, kc2, 12, kc1 aout dcblock asig outs aout, aout endin i1 0 20 e manual_src~dfsg/examples/prints.csd0000644000000000000000000000125312262561504016576 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o prints.wav -W ;;; for file output any platform /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Init-time print. prints "%2.3f\\t%!%!%!%!%!%!semicolons! %%\\n", 1234.56789 endin /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Play instrument #1. i 1 0 0.004 manual_src~dfsg/examples/expsegb.csd0000644000000000000000000000167012262561504016717 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o expseg.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; p4 = frequency in pitch-class notation. kcps = cpspch(p4) ; Create an amplitude envelope. kenv expsegb 0.01, p3*0.25, 1, p3, 0.01 kamp = kenv * 30000 a1 oscil kamp, kcps, 1 out a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for a half-second, p4=8.00 i 1 0 0.5 8.00 ; Play Instrument #1 for a half-second, p4=8.01 i 1 1 0.5 8.01 ; Play Instrument #1 for a half-second, p4=8.02 i 1 2 0.5 8.02 ; Play Instrument #1 for a half-second, p4=8.03 i 1 3 0.5 8.03 e manual_src~dfsg/examples/trmix.csd0000644000000000000000000000153512262561504016425 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trmix.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "fox.wav", 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .003, 1, 3, 500 ; partial tracking fslo,fshi trsplit fst, 1000 ; split partial tracks at 1000 Hz fscl trscale fshi, 1.3 ; shift the upper tracks fmix trmix fslo,fscl ; mix the shifted and unshifted tracks aout tradsyn fmix, 1, 1, 500, 1 ; resynthesis of tracks outs aout, aout endin f1 0 8192 10 1 ;sine wave i 1 0 3 e manual_src~dfsg/examples/vdelayxws.csd0000644000000000000000000000157012262561504017307 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelayxws.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ims = .5 ;maximum delay time in seconds iws = 1024 ;window size adl = .5 ;delay time asig1, asig2 diskin2 "kickroll.wav", 1, 0, 1 ;loop stereo file kickroll.wav a2 poscil .2, .1, 1 ;make an LFO, 1 cycle per 10 seconds adl = a2 + ims/2 ;offset the LFO so that it is positive aoutL, aoutR vdelayxws asig1, asig2, adl, ims, iws ;use the LFO to control delay time outs aoutL, aoutR endin f1 0 8192 10 1 i 1 0 10 e manual_src~dfsg/examples/biquad-2.csd0000644000000000000000000000671212262561504016670 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o biquad-2.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 /* modal synthesis using biquad filters as oscillators Example by Scott Lindroth 2007 */ instr 1 ipi = 3.1415926 idenom = sr*0.5 ipulseSpd = p4 icps = p5 ipan = p6 iamp = p7 iModes = p8 apulse mpulse iamp, 0 icps = cpspch( icps ) ; filter gain iamp1 = 600 iamp2 = 1000 iamp3 = 1000 iamp4 = 1000 iamp5 = 1000 iamp6 = 1000 ; resonance irpole1 = 0.99999 irpole2 = irpole1 irpole3 = irpole1 irpole4 = irpole1 irpole5 = irpole1 irpole6 = irpole1 ; modal frequencies if (iModes == 1) goto modes1 if (iModes == 2) goto modes2 modes1: if1 = icps * 1 ;pot lid if2 = icps * 6.27 if3 = icps * 3.2 if4 = icps * 9.92 if5 = icps * 14.15 if6 = icps * 6.23 goto nextPart modes2: if1 = icps * 1 ;uniform wood bar if2 = icps * 2.572 if3 = icps * 4.644 if4 = icps * 6.984 if5 = icps * 9.723 if6 = icps * 12.0 goto nextPart nextPart: ; convert frequency to radian frequency itheta1 = (if1/idenom) * ipi itheta2 = (if2/idenom) * ipi itheta3 = (if3/idenom) * ipi itheta4 = (if4/idenom) * ipi itheta5 = (if5/idenom) * ipi itheta6 = (if6/idenom) * ipi ; calculate coefficients ib11 = -2 * irpole1 * cos(itheta1) ib21 = irpole1 * irpole1 ib12 = -2 * irpole2 * cos(itheta2) ib22 = irpole2 * irpole2 ib13 = -2 * irpole3 * cos(itheta3) ib23 = irpole3 * irpole3 ib14 = -2 * irpole4 * cos(itheta4) ib24 = irpole4 * irpole4 ib15 = -2 * irpole5 * cos(itheta5) ib25 = irpole5 * irpole5 ib16 = -2 * irpole6 * cos(itheta6) ib26 = irpole6 * irpole6 ;printk 1, ib 11 ;printk 1, ib 21 ; also try setting the -1 coeff. to 0, but be sure to scale down the amplitude! asin1 biquad apulse * iamp1, 1, 0, -1, 1, ib11, ib21 asin2 biquad apulse * iamp2, 1, 0, -1, 1, ib12, ib22 asin3 biquad apulse * iamp3, 1, 0, -1, 1, ib13, ib23 asin4 biquad apulse * iamp4, 1, 0, -1, 1, ib14, ib24 asin5 biquad apulse * iamp5, 1, 0, -1, 1, ib15, ib25 asin6 biquad apulse * iamp6, 1, 0, -1, 1, ib16, ib26 afin = (asin1 + asin2 + asin3 + asin4 + asin5 + asin6) outs afin * sqrt(p6), afin*sqrt(1-p6) endin ;ins st dur pulseSpd pch pan amp Modes i1 0 12 0 7.089 0 0.7 2 i1 . . . 7.09 1 . . i1 . . . 7.091 0.5 . . i1 0 12 0 8.039 0 0.7 2 i1 0 12 0 8.04 1 0.7 2 i1 0 12 0 8.041 0.5 0.7 2 i1 9 . . 7.089 0 . 2 i1 . . . 7.09 1 . . i1 . . . 7.091 0.5 . . i1 9 12 0 8.019 0 0.7 2 i1 9 12 0 8.02 1 0.7 2 i1 9 12 0 8.021 0.5 0.7 2 e manual_src~dfsg/examples/fmb3.csd0000644000000000000000000000120212262561504016100 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmb3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 220 kc1 = p4 kc2 = p5 kvrate = 6 kvdpth line 0, p3, p6 asig fmb3 .4, kfreq, kc1, kc2, kvdpth, kvrate outs asig, asig endin ;sine wave. f 1 0 32768 10 1 i 1 0 2 5 5 0.1 i 1 3 2 .5 .5 0.01 e manual_src~dfsg/examples/pvbufread.csd0000644000000000000000000000140512262561504017234 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvbufread.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; analyze "fox.wav" and "flute.aiff" with PVANAL first ktime1 line 0, p3, .8 ; use a part of "flute.pvx" file ktime2 line 0, p3, 1.2 ; use a part of "beats.pvx" file kcross expon .03, p3, 1 pvbufread ktime1, "flute.pvx" asig pvcross ktime2, 1, "beats.pvx", 1-kcross, kcross outs asig, asig endin i 1 0 3 i 1 + 10 e manual_src~dfsg/examples/pvsdiskin.csd0000644000000000000000000000121612262561504017270 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsdiskin.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; create a PVOC-EX (*.pvx) file with PVANAL first ktscale line 1, p3, .05 ;change speed fsigr pvsdiskin "fox.pvx", ktscale, 1 ;read PVOCEX file aout pvsynth fsigr ;resynthesise it outs aout, aout endin i 1 0 10 e manual_src~dfsg/examples/STKTubeBell.csd0000644000000000000000000000131612262561504017337 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKTubeBell.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv2 line p6, p3, p7 ;Crossfade of Outputs kv1 = p5 ;(FM) Modulator Index One asig STKTubeBell cpspch(p4), 1, 2, kv1, 4, kv2, 11, 10, 1, 70, 128,50 outs asig, asig endin i 1 0 2 7.05 0 100 100 i 1 + 4 9.00 127 127 30 i 1 + 1 10.00 127 12 30 i 1 + 3 6.08 127 1 100 e manual_src~dfsg/examples/distort1.csd0000644000000000000000000000163612262561504017035 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o distort1.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gadist init 0 instr 1 iamp = p4 ifqc = cpspch(p5) asig pluck iamp, ifqc, ifqc, 0, 1 gadist = gadist + asig endin instr 50 kpre init p4 kpost init p5 kshap1 init p6 kshap2 init p7 aout distort1 gadist, kpre, kpost, kshap1, kshap2, 1 outs aout, aout gadist = 0 endin ; Sta Dur Amp Pitch i1 0.0 3.0 0.5 6.00 i1 0.5 2.5 0.5 7.00 i1 1.0 2.0 0.5 7.07 i1 1.5 1.5 0.5 8.00 ; Sta Dur PreGain PostGain Shape1 Shape2 i50 0 4 2 .5 0 0 e manual_src~dfsg/examples/pluck.csd0000644000000000000000000000115412262561504016375 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pluck.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 220 icps = 220 ifn = 0 imeth = p4 asig pluck 0.7, 220, 220, ifn, imeth, .1, 10 outs asig, asig endin i 1 0 5 1 i 1 5 5 4 ;needs 2 extra parameters (iparm1, iparm2) i 1 10 5 6 e manual_src~dfsg/examples/delayw.csd0000644000000000000000000000216312262561504016545 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o delayw.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasig init 0 gidel = 1 ;delay time in seconds instr 1 ain pluck .7, 440, 1000, 0, 1 outs ain, ain vincr gasig, ain ;send to global delay endin instr 2 ifeedback = p4 abuf2 delayr gidel adelL deltap .4 ;first tap (on left channel) adelM deltap 1 ;second tap (on middle channel) delayw gasig + (adelL * ifeedback) abuf3 delayr gidel kdel line 1, p3, .01 ;vary delay time adelR deltap .65 * kdel ;one pitch changing tap (on the right chn.) delayw gasig + (adelR * ifeedback) ;make a mix of all deayed signals outs adelL + adelM, adelR + adelM clear gasig endin i 1 0 1 i 1 3 1 i 2 0 3 0 ;no feedback i 2 3 8 .8 ;lots of feedback e manual_src~dfsg/examples/sinh.csd0000644000000000000000000000103712262561504016220 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o sinh.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 1 i1 = sinh(irad) print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/gen15.csd0000644000000000000000000000614612262561504016204 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen15.wav -W ;;; for file output any platform ;example from the Csound Book, page 85 sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = p3 iamp = p4 ifrq = cpspch(p5) ;pitch iswp1 = p6 iswp2 = p7 kswp line iswp1, p3, iswp2 ;amplitude sweep values acosi oscili kswp*.5, ifrq, 2 ;f2=cosine wave asine oscili kswp, ifrq, 1 ;f1=sine wave atab1 tablei acosi, 33, 1, .5 ;tables a1 to GEN13 atab2 tablei acosi, 34, 1, .5 ;tables a1 to GEN14 knrm1 tablei kswp, 35, 1 ;normalizing f35 knrm2 tablei kswp, 36, 1 ;normalizing f36 anrm1 = atab1*knrm1 ;normalize GEN13 signal anrm2 = atab2*knrm2*asine ;normalize GEN14 signal amix = anrm1+anrm2 ;mix GEN13 and GEN14 kenv expseg .001, idur*.1, iamp, idur*.1, iamp*.8, idur*.8, .001 asig = amix*kenv outs asig, asig endin f 1 0 8193 10 1 ;sine wave f 2 0 8193 9 1 1 90 ;cosine wave ; Note that all the f33 tables in the following sections are defined with p4=-15, ; which means that tables 33 and 34 will not be normalized. Thus if we display ; tables when running this example, we'll get correct diagrams even if one table ; has very small values instead of 0 values, due to cpu approximations in processing ; sin(180), as in sections 2, 4, and 5. This has no consequence on the audio result, ; because of the use of amp normalization (tables 35 and 36). f 33 0 8193 -15 1 1 1 0 1 180 .8 45 .6 270 .5 90 .4 225 .2 135 .1 315 ;makes function tables 33 and 34 f 35 0 4097 4 33 1 ;amp normalization for f33 f 36 0 4097 4 34 1 ;amp normalization for f34 i 1 0 5 .6 8.00 0 1 i 1 + . .6 8.00 1 0 s ;even harmonics with no phase shift, odd harmonics with phase shift f 33 0 8193 -15 1 1 1 0 1 0 1 180 1 180 1 0 1 0 1 180 1 180 1 0 1 0 1 180 1 180 f 35 0 4097 4 33 1 ;amp normalization for f33 f 36 0 4097 4 34 1 ;amp normalization for f34 i 1 0 5 .6 8.00 0 1 i 1 + . .6 8.00 1 0 s ;different harmonic strenghts and phases f 33 0 8193 -15 1 1 1 0 1 0 .9 180 .5 270 .75 90 .4 45 .2 225 .1 0 f 35 0 4097 4 33 1 ;amp normalization for f33 f 36 0 4097 4 34 1 ;amp normalization for f34 i 1 0 5 .6 8.00 0 1 i 1 + . .6 8.00 1 0 s ;lower harmonics no phase shift, upper harmonics with phase shift f 33 0 8193 -15 1 1 1 0 1 0 .5 0 .9 0 .3 0 .75 0 .2 180 .6 180 .15 180 .5 180 .1 180 f 35 0 4097 4 33 1 ;amp normalization for f33 f 36 0 4097 4 34 1 ;amp normalization for f34 i 1 0 5 .6 8.00 0 1 i 1 + . .6 8.00 1 0 s ;lower harmonics with phase shift, upper harmonics no phase shift f 33 0 8193 -15 1 1 1 180 1 180 .5 180 .9 180 .3 180 .75 180 .2 0 .6 0 .15 0 .5 0 .1 0 f 35 0 4097 4 33 1 ;amp normalization for f33 f 36 0 4097 4 34 1 ;amp normalization for f34 i 1 0 5 .6 8.00 0 1 i 1 + . .6 8.00 1 0 e manual_src~dfsg/examples/cpstun.csd0000644000000000000000000000207212262561504016573 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cpstun.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Table #1, a normal 12-tone equal temperament scale. ; numgrades = 12 (twelve tones) ; interval = 2 (one octave) ; basefreq = 261.659 (Middle C) ; basekeymidi = 60 (Middle C) gitemp ftgen 1, 0, 64, -2, 12, 2, 261.659, 60, 1.00, \ 1.059, 1.122, 1.189, 1.260, 1.335, 1.414, \ 1.498, 1.588, 1.682, 1.782, 1.888, 2.000 ; Instrument #1. instr 1 ; Set the trigger. ktrig init 1 ; Use Table #1. kfn init 1 ; If the base key (note #60) is C, then 9 notes ; above it (note #60 + 9 = note #69) should be A. kindex init 69 k1 cpstun ktrig, kindex, kfn printk2 k1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/strcpyk.csd0000644000000000000000000000411312262561504016754 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o strcpyk.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 instr 1 ;get one element of the input string whenever the metro ;triggers, and call a subinstrument to play the file Smember strget p4 istrlen strlen Smember kprint init 0 ktrig metro .6 ;whenever the trigger gives signal if ktrig == 1 then ;choose a random element (0, 1 or 2) kel random 0, 3.9999 kel = int(kel) ;make a copy for leaving Smember intact Scopy strcpyk Smember ;set the initial index for reading substrings kndx = 0 ;set counter for searching the element kcount = 0 ;start looping over the elements in Smember loop: kdelim strindexk Scopy, ":" ;as long as ":" occurs in Scopy, do: if kdelim > 0 then ;if this is the element to get if kel == kcount then ;read it as substring Sfile strsubk Scopy, kndx, kdelim kprint = kprint+1 ;and jump out kgoto call ;if not else ;cut off this element from Scopy Scopy strsubk Scopy, kdelim+1, istrlen endif ;if no element has been found,go back to loop ;and look for the next element kcount = kcount+1 kgoto loop ;if there is no delimiter left, the rest is the searched element else Sfile strcpyk Scopy endif call: ;print the result, call the subinstrument and play the file printf "kel = %d, file = '%s'\n", ktrig+kprint, kel, Sfile S_call sprintfk {{i 2 0 1 "%s"}}, Sfile scoreline S_call, ktrig endif endin instr 2 ;play Sfile strget p4 ilen filelen Sfile p3 = ilen asig soundin Sfile outs asig, asig endin i 1 0 30 "mary.wav:fox.wav:beats.wav:flute.aiff" e manual_src~dfsg/examples/tambourine.csd0000644000000000000000000000104512262561504017423 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tambourine.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idamp = p4 asig tambourine .8, 0.01, 30, idamp outs asig, asig endin i 1 0 .2 0 i 1 + .2 > i 1 + 1 .7 e manual_src~dfsg/examples/STKBowed.csd0000644000000000000000000000142312262561504016700 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKBowed.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = p4 kv2 = p7 ;position on bow kv1 line p5, p3, p6 ;bow pressure kv4 line 0, p3, 7 ;depth of low-frequency oscillator asig STKBowed cpspch(ipch), 1, 2, kv1, 4, kv2, 11, 40, 1, kv4, 128, 100 asig = asig*2 ;amplify outs asig, asig endin i 1 0 15 6.00 20 100 115 i 1 17 3 7.00 120 0 0 i 1 21 3 7.09 120 0 30 i 1 21 4 7.03 50 0 0 e manual_src~dfsg/examples/hrtfmove.csd0000644000000000000000000000165212262561504017114 0ustar rootroot ; Select flags here ; realtime audio out -o dac ; For Non-realtime ouput leave only the line below: ;-o hrtf.wav sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 gasrc init 0 instr 1 ;a plucked string kamp = p4 kcps = cpspch(p5) icps = cpspch(p5) a1 pluck kamp, kcps, icps, 0, 1 gasrc = a1 endin instr 10 ;uses output from instr1 as source kaz linseg 0, p3, 720 ;2 full rotations aleft,aright hrtfmove gasrc, kaz,0, "hrtf-44100-left.dat","hrtf-44100-right.dat" outs aleft, aright endin ; Play Instrument 1: a simple arpeggio i1 0 .2 15000 8.00 i1 + .2 15000 8.04 i1 + .2 15000 8.07 i1 + .2 15000 8.11 i1 + .2 15000 9.02 i1 + 1.5 15000 8.11 i1 + 1.5 15000 8.07 i1 + 1.5 15000 8.04 i1 + 1.5 15000 8.00 i1 + 1.5 15000 7.09 i1 + 1.5 15000 8.00 ; Play Instrument 10 for 10 seconds. i10 0 10 manual_src~dfsg/examples/STKResonate.csd0000644000000000000000000000135012262561504017417 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKResonate.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; frequency and amplitude of STKResonate have no effect kv2 = p4 ;pole radii kv1 line 100, p3, 0 ;resonance freq + notch freq kv3 = kv1 asig STKResonate 1, 1, 2, kv1, 4, kv2, 1, 10, 11, kv3, 128, 100 asig = asig * .3 ;too loud outs asig, asig endin i 1 0 1 0 i 1 + . > i 1 + . > i 1 + . > i 1 + . 120 e manual_src~dfsg/examples/balance.csd0000644000000000000000000000132712262561504016646 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o balance.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Generate a band-limited pulse train. asrc buzz 0.9, 440, sr/440, 1 ; Send the source signal through 2 filters. a1 reson asrc, 1000, 100 a2 reson a1, 3000, 500 ; Balance the filtered signal with the source. afin balance a2, asrc outs afin, afin endin ;sine wave. f 1 0 16384 10 1 i 1 0 2 e manual_src~dfsg/examples/zir.csd0000644000000000000000000000156112262561504016065 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zir.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple instrument. instr 1 ; Set the zk variable #1 to 32.594. ziw 32.594, 1 endin ; Instrument #2 -- prints out zk variable #1. instr 2 ; Read the zk variable #1 at i-rate. i1 zir 1 ; Print out the value of zk variable #1. print i1 endin ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e manual_src~dfsg/examples/log2.csd0000644000000000000000000000102112262561504016113 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o log10.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = log2(8) print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/guiro.csd0000644000000000000000000000075612262561504016413 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o guiro.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 a1 guiro .8, p4 outs a1, a1 endin i1 0 1 1 i1 + 1 .01 e manual_src~dfsg/examples/dumpk3.csd0000644000000000000000000000255112262561504016464 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dumpk3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes three control signals to a file kfreq randh 100, 1, 2, 1, 500 ;generates one random number between 400 and 600 per second kdb randh 12, 1, 2, 1, -12 ;amplitudes in dB between -24 and 0 kpan randh .5, 1, 2, 1, .5 ;panning between 0 and 1 dumpk3 kfreq, kdb, kpan, "dumpk3.txt", 8, 1 ;writes the control signals prints "WRITING:\n" printks "kfreq = %f, kdb = %f, kpan = %f\n", 1, kfreq, kdb, kpan ;prints them endin instr 2 ;reads the file written by instr 1 kf,kdb,kp readk3 "dumpk3.txt", 8, 1 prints "READING:\n" printks "kfreq = %f, kdb = %f, kpan = %f\n", 1, kf, kdb, kp ;prints again kdb lineto kdb, .1 ;smoothing amp transition kp lineto kp, .1 ;smoothing pan transition aout poscil ampdb(kdb), kf, giSine aL, aR pan2 aout, kp outs aL, aR endin i 1 0 5 i 2 5 5 e manual_src~dfsg/examples/noteondur.csd0000644000000000000000000000136212262561504017275 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1;;;RT audio I/O with MIDI in sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 instr 1 ;Turned on by MIDI notes on channel 1 ifund notnum ivel veloc idur = 1 ;chord with single key noteondur 1, ifund, ivel, idur noteondur 1, ifund+3, ivel, idur noteondur 1, ifund+7, ivel, idur noteondur 1, ifund+9, ivel, idur endin ; Play Instrument #1 for 60 seconds. i1 0 60 manual_src~dfsg/examples/outq1.csd0000644000000000000000000000122312262561504016325 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outq1.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 asig vco2 .05, 30 ; sawtooth waveform at low volume kcut line 60, p3, 300 ; Vary cutoff frequency kresonance = 7 inumlayer = 2 asig lowresx asig, kcut, kresonance, inumlayer outq1 asig ; output channel 1 endin i 1 0 3 e manual_src~dfsg/examples/sfload.csd0000644000000000000000000000276212262561504016535 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out, virtual midi in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfload.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;load two soundfonts isf sfload "07AcousticGuitar.sf2" ir sfload "01hpschd.sf2" sfplist isf sfplist ir sfpassign 0, isf sfpassign 1, ir instr 1 ; play guitar from score and midi keyboard - preset index = 0 mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/3000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay3 ivel, inum, kamp*ivel, kfreq, 0 ;preset index = 0 outs a1, a2 endin instr 2 ; play harpsichord from score and midi keyboard - preset index = 1 mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/1000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay3 ivel, inum, kamp*ivel, kfreq, 1 ;preset index = 1 outs a1, a2 endin f0 60 ; stay active for 1 minute i1 0 1 60 100 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 i2 5 1 60 100 i2 + 1 62 < i2 7 1 65 < i2 7 1 69 10 e manual_src~dfsg/examples/ptable.csd0000644000000000000000000000160712262561504016531 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o table.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Vary our index linearly from 0 to 1. kndx line 0, p3, 1 ; Read Table #1 with our index. ifn = 1 ixmode = 1 kfreq ptable kndx, ifn, ixmode ; Generate a sine waveform, use our table values ; to vary its frequency. a1 oscil 20000, kfreq, 2 out a1 endin ; Table #1, a line from 200 to 2,000. f 1 0 1025 -7 200 1024 2000 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e manual_src~dfsg/examples/vtablewk.csd0000644000000000000000000000074412262561504017102 0ustar rootroot ;-ovtablewa.wav -W -b441 -B441 -odac -b441 -B441 sr=44100 kr=441 ksmps=100 nchnls=2 instr 1 ilen = ftlen(1) knew1 oscil 10000, 440, 3 knew2 oscil 15000, 440, 3, 0.5 kindex phasor 0.3 asig oscil 1, sr/ilen , 1 vtablewk kindex*ilen, 1, 0, knew1, knew2 out asig,asig endin f1 0 262144 -1 "beats.wav" 0 4 0 f2 0 262144 2 0 f3 0 1024 10 1 i1 0 10 manual_src~dfsg/examples/pvsifd.csd0000644000000000000000000000130212262561504016545 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsifd.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "beats.wav", 1, 0, 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; pvsifd analysis fst partials fs1, fsi2, .1, 1,3, 500 ; partial tracking aout resyn fst, 1, 1.5, 500, 1 ; resynthesis (up a 5th) outs aout, aout endin ;sine f1 0 4096 10 1 i 1 0 2 e manual_src~dfsg/examples/FLslidBnk2.csd0000644000000000000000000000257112262561504017155 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc -M0 ;;;RT audio I/O with MIDI in sr = 44100 ksmps = 100 nchnls = 2 ;Example by Gabriel Maldonado giElem init 8 giOutTab ftgen 1,0,128, 2, 0 ;min1, max1, exp1, type1, min2, max2, exp2, type2, min3, max3, exp3, type3 etc. giConfigTab ftgen 2,0,128,-2, .1, 1000, -1, 3, .1, 1000, -1, 3, .1, 1000, -1, 3, 30, 2000, -1, 3, \ .1, 5000, -1, 5, .1, 5000, -1, 5, .1, 5000, -1, 5, .1, 5000, -1, 5 giSine ftgen 3,0,256,10, 1 FLpanel "This Panel contains a Slider Bank",600,600 FLslidBnk2 "mod1@mod2@mod3@amp@freq1@freq2@freq3@freqPo", giElem, giOutTab, giConfigTab, 400, 500, 100, 10 FLpanel_end FLrun instr 1 kmodindex1 init 0 kmodindex2 init 0 kmodindex3 init 0 kamp init 0 kfreq1 init 0 kfreq2 init 0 kfreq3 init 0 kfreq4 init 0 vtable1k giOutTab, kmodindex1 , kmodindex2, kmodindex3, kamp, kfreq1, kfreq2 , kfreq3, kfreq4 amod1 oscili kmodindex1, kfreq1, giSine amod2 oscili kmodindex2, kfreq2, giSine amod3 oscili kmodindex3, kfreq3, giSine aout oscili kamp, kfreq4+amod1+amod2+amod3, giSine outs aout, aout endin i1 0 3600 f0 3600 manual_src~dfsg/examples/schedkwhen.csd0000644000000000000000000000217712262561504017410 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o schedkwhen.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1 - oscillator with a high note. instr 1 ; Use the fourth p-field as the trigger. ktrigger = p4 kmintim = 0 kmaxnum = 2 kinsnum = 2 kwhen = 0 kdur = 0.5 ; Play Instrument #2 at the same time, if the trigger is set. schedkwhen ktrigger, kmintim, kmaxnum, kinsnum, kwhen, kdur ; Play a high note. a1 oscils 10000, 880, 1 out a1 endin ; Instrument #2 - oscillator with a low note. instr 2 ; Play a low note. a1 oscils 10000, 220, 1 out a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; p4 = trigger for Instrument #2 (when p4 > 0). ; Play Instrument #1 for half a second, no trigger. i 1 0 0.5 0 ; Play Instrument #1 for half a second, trigger Instrument #2. i 1 1 0.5 1 e manual_src~dfsg/examples/zkr.csd0000644000000000000000000000224712262561504016071 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zkr.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Linearly vary a k-rate signal from 440 to 880. kline line 440, p3, 880 ; Add the linear signal to zk variable #1. zkw kline, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read zk variable #1. kfreq zkr 1 ; Use the value of zk variable #1 to vary ; the frequency of a sine waveform. a1 oscil 20000, kfreq, 1 ; Generate the audio output. out a1 ; Clear the zk variables, get them ready for ; another pass. zkcl 0, 1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e manual_src~dfsg/examples/ziwm.csd0000644000000000000000000000205012262561504016241 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ziwm.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple instrument. instr 1 ; Add 20.5 to zk variable #1. ziwm 20.5, 1 endin ; Instrument #2 -- another simple instrument. instr 2 ; Add 15.25 to zk variable #1. ziwm 15.25, 1 endin ; Instrument #3 -- prints out zk variable #1. instr 3 ; Read zk variable #1 at i-rate. i1 zir 1 ; Print out the value of zk variable #1. ; It should be 35.75 (20.5 + 15.25) print i1 endin ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 ; Play Instrument #3 for one second. i 3 0 1 e manual_src~dfsg/examples/marmstk1.wav0000644000000000000000000000105412262561504017041 0ustar rootrootRIFF$WAVEfmt "VDdata - i02 T|og;1sd h>'Q-;# Ғ(* ehO 5) AC E>/^+s cV tP5qC  R^d )Y   <F }ocC ` m ;XesEB to lw R#V N fWn<* LRmanual_src~dfsg/examples/pvoc.csd0000644000000000000000000000120412262561504016222 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvoc.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; analyze "fox.wav" with PVANAL first ispec = p4 ktime line 0, p3, 1.55 kfrq line .8, p3, 2 asig pvoc ktime, kfrq, "fox.pvx", ispec outs asig, asig endin i 1 0 6 0 i 1 + 6 1 ;preserve spectral envelope e manual_src~dfsg/examples/gain.csd0000644000000000000000000000137112262561504016176 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gain.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asrc buzz .8, 440, sr/440, 1 ; band-limited pulse train. a1 reson asrc, 1000, 100 ; Sent through a2 reson a1, 3000, 500 ; 2 filters krms rms asrc ; then balanced afin gain a2, krms ; with source outs afin, afin endin ;sine wave. f 1 0 16384 10 1 i 1 0 2 e manual_src~dfsg/examples/toot1.csd0000644000000000000000000000044612262561504016330 0ustar rootroot -odac instr 1 a1 oscil 10000, 440, 1 out a1 endin f1 0 4096 10 1 ; use GEN10 to compute a sine wave ;ins strt dur i1 0 4 e ; indicates the end of the score manual_src~dfsg/examples/lowres.csd0000644000000000000000000000131012262561504016564 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lowres.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kres = p4 asig vco .2, 220, 1 ;sawtooth kcut line 1000, p3, 10 ;note: kcut is not in Hz as lowres asig, kcut, kres ;note: kres is not in dB aout balance as, asig ;avoid very loud sounds outs aout, aout endin ; a sine f 1 0 16384 10 1 i 1 0 4 3 i 1 + 4 30 i 1 + 4 60 e manual_src~dfsg/examples/inleta.csd0000644000000000000000000001022012262561504016525 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o inleta.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 ; Connect up instruments and effects to create the signal flow graph. connect "SimpleSine", "leftout", "Reverberator", "leftin" connect "SimpleSine", "rightout","Reverberator", "rightin" connect "Moogy", "leftout", "Reverberator", "leftin" connect "Moogy", "rightout","Reverberator", "rightin" connect "Reverberator", "leftout", "Compressor", "leftin" connect "Reverberator", "rightout","Compressor", "rightin" connect "Compressor", "leftout", "Soundfile", "leftin" connect "Compressor", "rightout","Soundfile", "rightin" ; Turn on the "effect" units in the signal flow graph. alwayson "Reverberator", 0.91, 12000 alwayson "Compressor" alwayson "Soundfile" ; Define instruments and effects in order of signal flow. instr SimpleSine ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Default values: p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 pset 0, 0, 10, 0, 0, 0, 0.5 iattack = 0.015 idecay = 0.07 isustain = p3 irelease = 0.3 p3 = iattack + idecay + isustain + irelease adamping linsegr 0.0, iattack, 1.0, idecay + isustain, 1.0, irelease, 0.0 iHz = cpsmidinn(p4) ; Rescale MIDI velocity range to a musically usable range of dB. iamplitude = ampdb(p5 / 127 * 15.0 + 60.0) ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. icosine ftgenonce 0, 0, 65537, 11, 1 aoscili oscili iamplitude, iHz, icosine aadsr madsr iattack, idecay, 0.6, irelease asignal = aoscili * aadsr aleft, aright pan2 asignal, p7 ; Stereo audio output to be routed in the orchestra header. outleta "leftout", aleft outleta "rightout", aright endin instr Moogy ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Default values: p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 pset 0, 0, 10, 0, 0, 0, 0.5 iattack = 0.003 isustain = p3 irelease = 0.05 p3 = iattack + isustain + irelease adamping linsegr 0.0, iattack, 1.0, isustain, 1.0, irelease, 0.0 iHz = cpsmidinn(p4) ; Rescale MIDI velocity range to a musically usable range of dB. iamplitude = ampdb(p5 / 127 * 20.0 + 60.0) print iHz, iamplitude ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. isine ftgenonce 0, 0, 65537, 10, 1 asignal vco iamplitude, iHz, 1, 0.5, isine kfco line 2000, p3, 200 krez = 0.8 asignal moogvcf asignal, kfco, krez, 100000 asignal = asignal * adamping aleft, aright pan2 asignal, p7 ; Stereo audio output to be routed in the orchestra header. outleta "leftout", aleft outleta "rightout", aright endin instr Reverberator ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" idelay = p4 icutoff = p5 aleft, aright reverbsc aleftin, arightin, idelay, icutoff ; Stereo output. outleta "leftout", aleft outleta "rightout", aright endin instr Compressor ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" kthreshold = 25000 icomp1 = 0.5 icomp2 = 0.763 irtime = 0.1 iftime = 0.1 aleftout dam aleftin, kthreshold, icomp1, icomp2, irtime, iftime arightout dam arightin, kthreshold, icomp1, icomp2, irtime, iftime ; Stereo output. outleta "leftout", aleftout outleta "rightout", arightout endin instr Soundfile ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" outs aleftin, arightin endin ; It is not necessary to activate "effects" or create f-tables in the score! ; Overlapping notes create new instances of instruments with proper connections. i "SimpleSine" 1 5 60 85 i "SimpleSine" 2 5 64 80 i "Moogy" 3 5 67 75 i "Moogy" 4 5 71 70 ;1 extra second after the performance e 1 manual_src~dfsg/examples/seed.csd0000644000000000000000000000154312262561504016201 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o seed.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;same values every time seed 10 krnd randomh 100, 200, 5 printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin instr 2 ;different values every time - value is derived from system clock seed 0 ; seed from system clock krnd randomh 100, 200, 5 printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin f 1 0 16384 10 1 ;sine wave. i 1 0 1 i 2 2 1 e manual_src~dfsg/examples/FLknob.csd0000644000000000000000000000265212262561504016436 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLknob.wav -W ;;; for file output any platform ; A sine with oscillator with flknob controlled frequency sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Frequency Knob", 900, 400, 50, 50 ; Minimum value output by the knob imin = 200 ; Maximum value output by the knob imax = 5000 ; Logarithmic type knob selected iexp = -1 ; Knob graphic type (1=3D knob) itype = 1 ; Display handle (-1=not used) idisp = -1 ; Width of the knob in pixels iwidth = 70 ; Distance of the left edge of the knob ; from the left edge of the panel ix = 70 ; Distance of the top edge of the knob ; from the top of the panel iy = 125 gkfreq, ihandle FLknob "Frequency", imin, imax, iexp, itype, idisp, iwidth, ix, iy ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun ; Set the widget's initial value FLsetVal_i 300, ihandle instr 1 iamp = 15000 ifn = 1 asig oscili iamp, gkfreq, ifn out asig endin ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e manual_src~dfsg/examples/lorisread.csd0000644000000000000000000000327212262561504017246 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lorisread.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; Play the partials in clarinet.sdif from 0 to 3 sec with 1 ms fadetime ; and no frequency , amplitude, or bandwidth modification. instr 1 ktime linseg 0, p3, 3 ; linear time function from 0 to 3 seconds lorisread ktime, "clarinet.sdif", 1, 1, 1, 1, .001 asig lorisplay 1, 1, 1, 1 outs asig, asig endin ; Play the partials in clarinet.sdif from 0 to 3 sec with 1 ms fadetime ; adding tuning and vibrato, increasing the "breathiness" (noisiness) and overall ; amplitude, and adding a highpass filter. instr 2 ktime linseg 0, p3, 3 ; linear time function from 0 to 3 seconds ; compute frequency scale for tuning ifscale = cpspch(p4)/cpspch(8.08) ; (original pitch was G#4) ; make a vibrato envelope kvenv linseg 0, p3/6, 0, p3/6, .02, p3/3, .02, p3/6, 0, p3/6, 0 kvib oscil kvenv, 4, 1 ; table 1, sinusoid kbwenv linseg 1, p3/6, 1, p3/6, 2, 2*p3/3, 2 ;lots of noise lorisread ktime, "clarinet.sdif", 1, 1, 1, 1, .001 a1 lorisplay 1, ifscale+kvib, 2, kbwenv asig atone a1, 1000 ; highpass filter, cutoff 1000 Hz outs asig, asig endin ; a sinus f 1 0 4096 10 1 i 1 0 3 i 1 + 1 i 1 + 6 s ; pitch i 2 1 3 8.08 i 2 3.5 1 8.04 i 2 4 6 8.00 i 2 4 6 8.07 e manual_src~dfsg/examples/min.csd0000644000000000000000000000146212262561504016044 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o min.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 k1 oscili 1, 10.0, 1 ;combine 3 sinusses k2 oscili 1, 1.0, 1 ;at different rates k3 oscili 1, 3.0, 1 kmin min k1, k2, k3 kmin = kmin*250 ;scale kmin printk2 kmin ;check the values aout vco2 .5, 220, 6 ;sawtooth asig moogvcf2 aout, 600+kmin, .5 ;change filter around 600 Hz outs asig, asig endin f1 0 32768 10 1 i1 0 5 e manual_src~dfsg/examples/128,8-gridX0000755000000000000000000000557312262561504016254 0ustar rootroot 0 1 0 8 1 0 1 2 1 9 2 1 2 3 2 10 3 2 3 4 3 11 4 3 4 5 4 12 5 4 5 6 5 13 6 5 6 7 6 14 7 6 7 15 8 0 8 9 8 16 9 1 9 8 9 10 9 17 10 2 10 9 10 11 10 18 11 3 11 10 11 12 11 19 12 4 12 11 12 13 12 20 13 5 13 12 13 14 13 21 14 6 14 13 14 15 14 22 15 7 15 14 15 23 16 8 16 17 16 24 17 9 17 16 17 18 17 25 18 10 18 17 18 19 18 26 19 11 19 18 19 20 19 27 20 12 20 19 20 21 20 28 21 13 21 20 21 22 21 29 22 14 22 21 22 23 22 30 23 15 23 22 23 31 24 16 24 25 24 32 25 17 25 24 25 26 25 33 26 18 26 25 26 27 26 34 27 19 27 26 27 28 27 35 28 20 28 27 28 29 28 36 29 21 29 28 29 30 29 37 30 22 30 29 30 31 30 38 31 23 31 30 31 39 32 24 32 33 32 40 33 25 33 32 33 34 33 41 34 26 34 33 34 35 34 42 35 27 35 34 35 36 35 43 36 28 36 35 36 37 36 44 37 29 37 36 37 38 37 45 38 30 38 37 38 39 38 46 39 31 39 38 39 47 40 32 40 41 40 48 41 33 41 40 41 42 41 49 42 34 42 41 42 43 42 50 43 35 43 42 43 44 43 51 44 36 44 43 44 45 44 52 45 37 45 44 45 46 45 53 46 38 46 45 46 47 46 54 47 39 47 46 47 55 48 40 48 49 48 56 49 41 49 48 49 50 49 57 50 42 50 49 50 51 50 58 51 43 51 50 51 52 51 59 52 44 52 51 52 53 52 60 53 45 53 52 53 54 53 61 54 46 54 53 54 55 54 62 55 47 55 54 55 63 56 48 56 57 56 64 57 49 57 56 57 58 57 65 58 50 58 57 58 59 58 66 59 51 59 58 59 60 59 67 60 52 60 59 60 61 60 68 61 53 61 60 61 62 61 69 62 54 62 61 62 63 62 70 63 55 63 62 63 71 64 56 64 65 64 72 65 57 65 64 65 66 65 73 66 58 66 65 66 67 66 74 67 59 67 66 67 68 67 75 68 60 68 67 68 69 68 76 69 61 69 68 69 70 69 77 70 62 70 69 70 71 70 78 71 63 71 70 71 79 72 64 72 73 72 80 73 65 73 72 73 74 73 81 74 66 74 73 74 75 74 82 75 67 75 74 75 76 75 83 76 68 76 75 76 77 76 84 77 69 77 76 77 78 77 85 78 70 78 77 78 79 78 86 79 71 79 78 79 87 80 72 80 81 80 88 81 73 81 80 81 82 81 89 82 74 82 81 82 83 82 90 83 75 83 82 83 84 83 91 84 76 84 83 84 85 84 92 85 77 85 84 85 86 85 93 86 78 86 85 86 87 86 94 87 79 87 86 87 95 88 80 88 89 88 96 89 81 89 88 89 90 89 97 90 82 90 89 90 91 90 98 91 83 91 90 91 92 91 99 92 84 92 91 92 93 92 100 93 85 93 92 93 94 93 101 94 86 94 93 94 95 94 102 95 87 95 94 95 103 96 88 96 97 96 104 97 89 97 96 97 98 97 105 98 90 98 97 98 99 98 106 99 91 99 98 99 100 99 107 100 92 100 99 100 101 100 108 101 93 101 100 101 102 101 109 102 94 102 101 102 103 102 110 103 95 103 102 103 111 104 96 104 105 104 112 105 97 105 104 105 106 105 113 106 98 106 105 106 107 106 114 107 99 107 106 107 108 107 115 108 100 108 107 108 109 108 116 109 101 109 108 109 110 109 117 110 102 110 109 110 111 110 118 111 103 111 110 111 119 112 104 112 113 112 120 113 105 113 112 113 114 113 121 114 106 114 113 114 115 114 122 115 107 115 114 115 116 115 123 116 108 116 115 116 117 116 124 117 109 117 116 117 118 117 125 118 110 118 117 118 119 118 126 119 111 119 118 119 127 120 112 120 121 121 113 121 120 121 122 122 114 122 121 122 123 123 115 123 122 123 124 124 116 124 123 124 125 125 117 125 124 125 126 126 118 126 125 126 127 127 119 127 126 manual_src~dfsg/examples/looptseg.csd0000644000000000000000000000216112262561504017112 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o looptseg.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 1 ; frequency of loop repetition ktrig init 0 ; loop restart trigger (not used) iphase = 0 ; initial phase ktyp line 6,p3,-6 ; explore the useful range of curve types ; loop of filter cutoff values (oct format) ; value curve dur. kcfoct looptseg kfreq, ktrig, iphase,13, ktyp, 1, \ 4, ktyp, 0, \ 11, ktyp, 1, \ 4 asig vco2 0.2,cpsmidinn(48),0 ; a sawtooth asig moogladder asig,cpsoct(kcfoct),rnd(0.6) ; filter sawtooth outs asig, asig endin i 1 0 12 e manual_src~dfsg/examples/wterrain.csd0000644000000000000000000000165412262561504017117 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o wterrain.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 kdclk linseg 0, 0.01, 1, p3-0.02, 1, 0.01, 0 kcx line 0.1, p3, 1.9 krx linseg 0.1, p3/2, 0.5, p3/2, 0.1 kpch line cpspch(p4), p3, p5 * cpspch(p4) a1 wterrain 10000, kpch, kcx, kcx, -krx, krx, p6, p7 a1 dcblock a1 out a1*kdclk endin f1 0 8192 10 1 0 0.33 0 0.2 0 0.14 0 0.11 f2 0 4096 10 1 i1 0 4 7.00 1 1 1 i1 4 4 6.07 1 1 2 i1 8 8 6.00 1 2 2 e manual_src~dfsg/examples/miditempo.csd0000644000000000000000000000151512262561504017247 0ustar rootroot ; Select audio/midi flags here according to platform -odac -F Anna.mid ;;;realtime audio out and midi file input ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o miditempo.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 massign 0, 1 ; make sure that all channels pgmassign 0, 1 ; and programs are assigned to test instr instr 1 ksig miditempo prints "miditempo = %d\\n", ksig icps cpsmidi ; convert midi note to pitch kenv madsr 0.1, 0, 0.8, 0.3 asig pluck kenv*.15, icps, icps, 1, 1 ;low volume outs asig, asig endin f 0 200 ;stay active for 120 seconds f 1 0 4096 10 1 ;sine e manual_src~dfsg/examples/timout.csd0000644000000000000000000000164512262561504016605 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o timout.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 indx = 0 itim = p4 ;change time for one step clock: timout 0, itim, time reinit clock time: itmp table indx, 2, 0, 0, 1 if itmp == 1 then print itmp event_i "i",2, 0, .1 ;event has duration of .1 second endif indx = indx+1 endin instr 2 ;play it kenv transeg 0.01, p3*0.25, 1, 1, p3*0.75, .5, 0.01 asig oscili kenv*.4, 400, 1 outs asig, asig endin f 1 0 1024 10 1 ;sine f 2 0 16 2 1 0 0 1 0 1 0 1 0 1 0 0 0 0 0 0 ;the rythm table i1 0 10 .1 i1 + 10 .05 i1 + 10 .01 e manual_src~dfsg/examples/sekere.csd0000644000000000000000000000104612262561504016535 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sekere.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idamp = p4 ;vary damping amount asig sekere 1, 0.01, 64, idamp outs asig, asig endin i1 0 1 .1 i1 + 1 .9 e manual_src~dfsg/examples/babo_expert.csd0000644000000000000000000000734412262561504017560 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o babo_expert.wav -W ;;; for file output any platform /* Written by Nicola Bernardini */ sr = 44100 ksmps = 32 nchnls = 2 ; full blown babo instrument with movement ; instr 2 ixstart = p4 ; start x position of source (left-right) ixend = p7 ; end x position of source iystart = p5 ; start y position of source (front-back) iyend = p8 ; end y position of source izstart = p6 ; start z position of source (up-down) izend = p9 ; end z position of source ixsize = p10 ; width of the resonator iysize = p11 ; depth of the resonator izsize = p12 ; height of the resonator idiff = p13 ; diffusion coefficient iexpert = p14 ; power user values stored in this function ainput soundin "beats.wav" ksource_x line ixstart, p3, ixend ksource_y line iystart, p3, iyend ksource_z line izstart, p3, izend al,ar babo ainput*0.7, ksource_x, ksource_y, ksource_z, ixsize, iysize, izsize, idiff, iexpert outs al,ar endin /* Written by Nicola Bernardini */ ; full blown instrument ;p4 : start x position of source (left-right) ;p5 : end x position of source ;p6 : start y position of source (front-back) ;p7 : end y position of source ;p8 : start z position of source (up-down) ;p9 : end z position of source ;p10 : width of the resonator ;p11 : depth of the resonator ;p12 : height of the resonator ;p13 : diffusion coefficient ;p14 : power user values stored in this function ; decay hidecay rx ry rz rdistance direct early_diff f1 0 8 -2 0.95 0.95 0 0 0 0.3 0.5 0.8 ; brighter f2 0 8 -2 0.95 0.5 0 0 0 0.3 0.5 0.8 ; default (to be set as) f3 0 8 -2 0.95 0.01 0 0 0 0.3 0.5 0.8 ; darker f4 0 8 -2 0.95 0.7 0 0 0 0.3 0.1 0.4 ; to hear the effect of diffusion f5 0 8 -2 0.9 0.5 0 0 0 0.3 2.0 0.98 ; to hear the movement f6 0 8 -2 0.99 0.1 0 0 0 0.3 0.5 0.8 ; default vals ; ^ ; ----- gen. number: negative to avoid rescaling i2 0 10 6 4 3 6 4 3 14.39 11.86 10 1 6 ; defaults i2 + 4 6 4 3 6 4 3 14.39 11.86 10 1 1 ; hear brightness 1 i2 + 4 6 4 3 -6 -4 3 14.39 11.86 10 1 2 ; hear brightness 2 i2 + 4 6 4 3 -6 -4 3 14.39 11.86 10 1 3 ; hear brightness 3 i2 + 3 .6 .4 .3 -.6 -.4 .3 1.439 1.186 1.0 0.0 4 ; hear diffusion 1 i2 + 3 .6 .4 .3 -.6 -.4 .3 1.439 1.186 1.0 1.0 4 ; hear diffusion 2 i2 + 4 12 4 3 -12 -4 -3 24.39 21.86 20 1 5 ; hear movement ; i2 + 4 6 4 3 6 4 3 14.39 11.86 10 1 1 ; hear brightness 1 i2 + 4 6 4 3 -6 -4 3 14.39 11.86 10 1 2 ; hear brightness 2 i2 + 4 6 4 3 -6 -4 3 14.39 11.86 10 1 3 ; hear brightness 3 i2 + 3 .6 .4 .3 -.6 -.4 .3 1.439 1.186 1.0 0.0 4 ; hear diffusion 1 i2 + 3 .6 .4 .3 -.6 -.4 .3 1.439 1.186 1.0 1.0 4 ; hear diffusion 2 i2 + 4 12 4 3 -12 -4 -3 24.39 21.86 20 1 5 ; hear movement ; ^^^^^^^^^^^^^^^^^^^ ^^^^^^^^^^^^^^^^^ ^ ^ ; ||||||||||||||||||| ||||||||||||||||| | --: expert values function ; ||||||||||||||||||| ||||||||||||||||| +--: diffusion ; ||||||||||||||||||| ----------------: optimal room dims according to Milner and Bernard JASA 85(2), 1989 ; ||||||||||||||||||| ; --------------------: source position start and end e manual_src~dfsg/examples/endin.csd0000644000000000000000000000113312262561504016351 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o endin.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 iamp = 10000 icps = 440 iphs = 0 a1 oscils iamp, icps, iphs out a1 endin ; Play Instrument #1 for 2 seconds. i 1 0 2 e manual_src~dfsg/examples/chnmix.csd0000644000000000000000000000351712262561504016552 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chnmix.wav -W ;;; for file output any platform ;Example by Joachim Heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1; send i-values chnset 1, "sio" chnset -1, "non" endin instr 2; send k-values kfreq randomi 100, 300, 1 chnset kfreq, "cntrfreq" kbw = kfreq/10 chnset kbw, "bandw" endin instr 3; send a-values anois rand .1 chnset anois, "noise" loop: idur random .3, 1.5 timout 0, idur, do reinit loop do: ifreq random 400, 1200 iamp random .1, .3 asig oscils iamp, ifreq, 0 aenv transeg 1, idur, -10, 0 asine = asig * aenv chnset asine, "sine" endin instr 11; receive some chn values and send again ival1 chnget "sio" ival2 chnget "non" print ival1, ival2 kcntfreq chnget "cntrfreq" kbandw chnget "bandw" anoise chnget "noise" afilt reson anoise, kcntfreq, kbandw afilt balance afilt, anoise chnset afilt, "filtered" endin instr 12; mix the two audio signals amix1 chnget "sine" amix2 chnget "filtered" chnmix amix1, "mix" chnmix amix2, "mix" endin instr 20; receive and reverb amix chnget "mix" aL, aR freeverb amix, amix, .8, .5 outs aL, aR endin instr 100; clear chnclear "mix" endin i 1 0 20 i 2 0 20 i 3 0 20 i 11 0 20 i 12 0 20 i 20 0 20 i 100 0 20 manual_src~dfsg/examples/peak.csd0000644000000000000000000000132012262561504016172 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o peak.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1 - play an audio file. instr 1 ; Capture the highest amplitude in the "beats.wav" file. asig soundin "beats.wav" kp peak asig ; Print out the peak value once per second. printk 1, kp out asig endin ; Play Instrument #1, the audio file, for three seconds. i 1 0 3 e manual_src~dfsg/examples/gen20.csd0000644000000000000000000000226012262561504016171 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen20.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 insnd = 10 ;"fox.wav" ibasfrq = 44100 / ftlen(insnd) ;use original sample rate of insnd file kamp expseg .001, p3/2, .7, p3/2, .8 ;envelope kpitch line ibasfrq, p3, ibasfrq * .8 kdens line 600, p3, 10 kaoff line 0, p3, .1 kpoff line 0, p3, ibasfrq * .5 kgdur line .04, p3, .001 ;shorten duration of grain during note imaxgdur = .5 igfn = p4 ;different windows asigL grain kamp, kpitch, kdens, kaoff, kpoff, kgdur, insnd, igfn, imaxgdur, 0.0 asigR grain kamp, kpitch, kdens, kaoff, kpoff, kgdur, insnd, igfn, imaxgdur, 0.0 outs asigL, asigR endin f1 0 512 20 2 ;Hanning window f2 0 512 20 6 1 ;Gaussian window f10 0 16384 1 "fox.wav" 0 0 0 i1 0 5 1 ;use Hanning window i1 + 5 2 ;use Gaussian window e manual_src~dfsg/examples/lessequal.csd0000644000000000000000000000137612262561504017263 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o <=.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kval randomh 0, 1.2, 20 ;choose between 0 and 1.2 if kval >0 && kval<=.5 then ;3 possible outcomes kvl = 1 elseif kval >.5 && kval<=1 then kvl =2 elseif kval >1 then kvl =3 endif printks "random number = %f, result = %f\n", .1, kval, kvl asig poscil .7, 440*kvl, 1 outs asig, asig endin f1 0 16384 10 1 i 1 0 5 e manual_src~dfsg/examples/STKPercFlut.csd0000644000000000000000000000136212262561504017366 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKPercFlut.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;Total Modulator Index kv2 line p7, p3, 0 ;Modulator Crossfade asig STKPercFlut cpspch(ifrq), 1, 2, kv1, 4, kv2, 11, 0, 1, 100, 128, 40 outs asig, asig endin i 1 0 7 8.05 100 0 100 i 1 3 7 9.03 20 120 0 i 1 3 .5 8.05 20 120 0 i 1 4 .5 9.09 20 120 0 i 1 5 3 9.00 20 120 0 e manual_src~dfsg/examples/rnd.csd0000644000000000000000000000130512262561504016040 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd.wav -W ;;; for file output any platform ; Andres Cabrera 2010 sr = 44100 ksmps = 4410 nchnls = 1 0dbfs = 1 instr 1 ; Generate a random number from 0 to 10. irand = rnd(10) print irand endin instr 2 klimit init 10 krand = rnd(klimit) printk 0, krand endin i 1 0 1 ; Generate 1 number i 1 0 1 ; Generate another number i 1 0 1 ; yet another number i 2 2 1 ; 1 second prints 9 values (kr = 10) e manual_src~dfsg/examples/sfinstr3.csd0000644000000000000000000000256212262561504017036 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfinstr3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gi24 ftgen 1, 0, 32, -2, 24, 2, 261.626, 60, 1, 1.0293022, 1.059463, 1.0905076, 1.1224619, 1.1553525, 1.1892069, \ 1.2240532, 1.2599207, 1.2968391, 1.33483924, 1.3739531, 1.414213, 1.4556525, 1.4983063, 1.54221, 1.5874001, \ 1.6339145, 1.6817917, 1.73107, 1.7817962, 1.8340067, 1.8877471, 1.9430623, 2 ;table for microtuning, a 24 tone equal temperament giSF sfload "sf_GMbank.sf2" sfilist giSF instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey = p4 ivel = p5 aenv linsegr 1, 1, 1, 1, 0 ;envelope icps cpstuni ikey, 1 ;24 tones per octave iamp = 0.0002 ;scale amplitude iamp = iamp * ivel * 1/128 ;make velocity-dependent aL, aR sfinstr3 ivel, ikey, iamp, icps, 180, giSF, 1 ;= Slap Bass 3 aL = aL * aenv aR = aR * aenv outs aL, aR endin f0 60 ;play for 60 seconds i1 0 1 60 100 1 ;using ftable 1 i1 + 1 62 < . i1 + 1 65 < . i1 + 1 69 40 . e manual_src~dfsg/examples/loscil3.csd0000644000000000000000000000146012262561504016627 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loscil3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ichnls = ftchnls(p4) print ichnls if (ichnls == 1) then asigL loscil3 .8, 1, p4, 1 asigR = asigL elseif (ichnls == 2) then asigL, asigR loscil3 .8, 1, p4, 1 ;safety precaution if not mono or stereo else asigL = 0 asigR = 0 endif outs asigL, asigR endin f 1 0 0 1 "mary.wav" 0 0 0 f 2 0 0 1 "kickroll.wav" 0 0 0 i 1 0 3 1 ;mono file i 1 + 2 2 ;stereo file e manual_src~dfsg/examples/deltap.csd0000644000000000000000000000216312262561504016531 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o deltap.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasig init 0 gidel = 1 ;delay time in seconds instr 1 ain pluck .7, 440, 1000, 0, 1 outs ain, ain vincr gasig, ain ;send to global delay endin instr 2 ifeedback = p4 abuf2 delayr gidel adelL deltap .4 ;first tap (on left channel) adelM deltap 1 ;second tap (on middle channel) delayw gasig + (adelL * ifeedback) abuf3 delayr gidel kdel line 1, p3, .01 ;vary delay time adelR deltap .65 * kdel ;one pitch changing tap (on the right chn.) delayw gasig + (adelR * ifeedback) ;make a mix of all deayed signals outs adelL + adelM, adelR + adelM clear gasig endin i 1 0 1 i 1 3 1 i 2 0 3 0 ;no feedback i 2 3 8 .8 ;lots of feedback e manual_src~dfsg/examples/vdelayx.csd0000644000000000000000000000144212262561504016733 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelayx.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ims = .5 ;maximum delay time in seconds iws = 1024 ;window size adl = .5 ;delay time asig diskin2 "fox.wav", 1, 0, 1 ;loop fox.wav a2 poscil .2, .2, 1 ;make an LFO adl = a2 + ims/2 ;offset the LFO so that it is positive aout vdelayx asig, adl, ims, iws ;use the LFO to control delay time outs aout, aout endin f1 0 8192 10 1 i 1 0 10 e manual_src~dfsg/examples/vexpv.csd0000644000000000000000000000170512262561504016431 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ifn2 = p5 ielements = p6 idstoffset = p7 isrcoffset = p8 kval init 25 vexpv ifn1, ifn2, ielements, idstoffset, isrcoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 16 17 f 2 0 16 -7 0 16 1 i2 0.0 0.2 1 i2 0.2 0.2 2 i1 0.4 0.01 1 2 5 3 8 i2 0.8 0.2 1 i1 1.0 0.01 1 2 5 10 -2 i2 1.2 0.2 1 i1 1.4 0.01 1 2 8 14 0 i2 1.6 0.2 1 i1 1.8 0.002 1 2 8 0 14 i2 2.0 0.2 1 i1 2.2 0.002 1 1 8 5 2 i2 2.4 0.2 1 e manual_src~dfsg/examples/midichn.csd0000644000000000000000000000057212262561504016675 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M0 -+rtmidi=virtual ;;; midi file input sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;change channel on virtual midi keyboard i1 midichn print i1 endin f 0 20 ;runs for 20 seconds e manual_src~dfsg/examples/temposcal.csd0000644000000000000000000000124712262561504017251 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o temposcal.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ilock = p4 itab = 1 ipitch = 1 iamp = 0.8 ktime linseg 0.3, p3/2, 0.8, p3/2, 0.3 asig temposcal ktime, iamp, ipitch, itab, ilock outs asig, asig endin f 1 0 0 1 "fox.wav" 0 4 0 i 1 0 3.8 0 ;no locking i 1 4 3.8 1 ;locking e manual_src~dfsg/examples/deltapn.csd0000644000000000000000000000217512262561504016712 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o deltap3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasig init 0 gidel = 1 ;delay time in seconds instr 1 ain pluck .7, 440, 1000, 0, 1 outs ain, ain vincr gasig, ain ;send to global delay endin instr 2 ifeedback = p4 abuf2 delayr gidel adelL deltapn 4000 ;first tap (on left channel) adelM deltapn 44100 ;second tap (on middle channel) delayw gasig + (adelL * ifeedback) abuf3 delayr gidel kdel line 100, p3, 1 ;vary delay time adelR deltapn 100 * kdel ;one pitch changing tap (on the right chn.) delayw gasig + (adelR * ifeedback) ;make a mix of all deayed signals outs adelL + adelM, adelR + adelM clear gasig endin i 1 0 1 i 1 3 1 i 2 0 3 0 ;no feedback i 2 3 8 .8 ;lots of feedback e manual_src~dfsg/examples/ATSsinnoi-2.csd0000644000000000000000000000425012262561504017265 0ustar rootroot -odac -d -m128 ;example by joachim heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 1024, 10, 1 gSfile = "fox.ats" giNumParts ATSinfo gSfile, 3 ;overall number of partials giDur ATSinfo gSfile, 7 ;duration seed 0 instr PlayList event_i "i", "PlayAll", 0, 1, 1, 0, .5 ;sine only, half speed event_i "i", "PlayAll", giDur*2+1, 1, 0, 1, .5 ;noise only event_i "i", "PlayAll", giDur*4+2, 1, .5, .5, .5 ;half sine, half noise endin instr PlayAll iSinAmnt = p4 ;sinee amount (0-1) iNzAmnt = p5 ;noise amount (0-1) iSpeed = p6 ;speed p3 = giDur/iSpeed ktime line 0, giDur/iSpeed, giDur prints "Resynthesizing all partials with tone = %.1f and noise = %.1f.\n", iSinAmnt, iNzAmnt aOut ATSsinnoi ktime, iSinAmnt, iNzAmnt, 1, gSfile, giNumParts outs aOut, aOut endin instr PlayBand iOffset = p4 ;offset in partials iSpeed = p5 ;speed p3 = giDur/iSpeed ktime line 0, giDur/iSpeed, giDur prints "Resynthesizing partials %d to %d with related noise.\n", iOffset+1, iOffset+10 aOut ATSsinnoi ktime, 1, 1, 1, gSfile, 10, iOffset, 1 outs aOut, aOut ;call itself again if iOffset < giNumParts - 20 then event_i "i", "PlayBand", giDur/iSpeed+1, 1, iOffset+10, iSpeed endif endin instr PlayWeighted ;sine amount, noise amount and speeed are varying kSinAmnt randomi 0, 1, 1, 3 kNzAmnt = 1-kSinAmnt kSpeed randomi .01, .3, 1, 3 async init 0 atime, aEnd syncphasor kSpeed/giDur, async kTrig metro 100 kEnd max_k aEnd, kTrig, 1 ;1 if phasor signal crosses zero ktime downsamp atime aOut ATSsinnoi ktime*giDur, kSinAmnt, kNzAmnt, 1, gSfile, giNumParts outs aOut, aOut ;exit if file is at the end if kEnd == 1 then event "i", "End", 0, 1 endif endin instr End exitnow endin i "PlayList" 0 1 i "PlayBand" 20 1 0 .5 i "PlayWeighted" 110 100 manual_src~dfsg/examples/gendyc.csd0000644000000000000000000000077712262561504016542 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aout gendyc 0.5, 1, 1, 1, 1, 220, 440, 0.5, 0.5 outs aout, aout endin i1 0 10 e manual_src~dfsg/examples/changed.csd0000644000000000000000000000104312262561504016645 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o changed.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 ksig oscil 2, 0.5, 1 kint = int(ksig) ktrig changed kint printk 0.2, kint printk2 ktrig endin f 1 0 1024 10 1 i 1 0 20 e manual_src~dfsg/examples/fluidLoad.csd0000644000000000000000000000166412262561504017170 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidLoad.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "07AcousticGuitar.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 0 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 7 asigl, asigr fluidOut giengine outs asigl*imvol, asigr*imvol endin i 1 0 2 60 100 ;play one note from score and... i 99 0 60 ;play virtual keyboard for 60 sec. e manual_src~dfsg/examples/outic.csd0000644000000000000000000000120012262561504016372 0ustar rootroot ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out -+rtmidi=virtual ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outic.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 ikey notnum ivel veloc kbrt = 40 ;set controller 74 (=brightness) outic 1, 74, kbrt, 0, 127 ;so filter closes a bit midion 1, ikey, ivel ;play external synth endin f0 30 ;runs 30 seconds e manual_src~dfsg/examples/nlfilt.csd0000644000000000000000000000166412262561504016555 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o nlfilt.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;unfiltered noise asig rand .7 outs asig, asig endin instr 2 ;filtered noise ka = p4 kb = p5 kd = p6 kC = p7 kL = p8 asig rand .3 afilt nlfilt asig, ka, kb, kd, kC, kL asig clip afilt, 2, .9 outs asig, asig endin i 1 0 2 ; unfiltersd ; a b d C L i 2 2 2 0 0 0.8 0.5 20 ; non-linear effect i 2 + 2 .4 0.2 0.7 0.11 200 ; low=paas with non-linear i 2 + 2 0.35 -0.3 0.95 0.1 200 ; high-pass with non-linear i 2 + 2 0.7 -0.2 0.9 0.2 20 ; high-pass with non-linear e manual_src~dfsg/examples/interp.csd0000644000000000000000000000215112262561504016556 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o interp.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 8000 kr = 8 ksmps = 1000 nchnls = 1 ; Instrument #1 - a simple instrument. instr 1 ; Create an amplitude envelope. kamp linseg 0, p3/2, 20000, p3/2, 0 ; The amplitude envelope will sound rough because it ; jumps every ksmps period, 1000. a1 oscil kamp, 440, 1 out a1 endin ; Instrument #2 - a smoother sounding instrument. instr 2 ; Create an amplitude envelope. kamp linseg 0, p3/2, 25000, p3/2, 0 aamp interp kamp ; The amplitude envelope will sound smoother due to ; linear interpolation at the higher a-rate, 8000. a1 oscil aamp, 440, 1 out a1 endin ; Table #1, a sine wave. f 1 0 256 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 2 2 e manual_src~dfsg/examples/toot7.csd0000644000000000000000000000446012262561504016336 0ustar rootroot -odac instr 7 ifunc1 = p11 ; initial waveform ifunc2 = p12 ; crossfade waveform ifad1 = p3 * p13 ; calculate initial fade (% of dur) ifad2 = p3 - ifad1 ; calculate remaining duration irel = .01 ; set vibrato release time idel1 = p3 * p10 ; calculate initial delay (% of dur) isus = p3 - (idel1 + irel) ; calculate remaining duration iamp = ampdb(p4) iscale = iamp * .166 ; p4=amp inote = cpspch(p5) ; p5=freq k3 linseg 0, idel1, p9, isus, p9, irel, 0 ; p6=attack time k2 oscil k3, p8, 1 ; p7=release time k1 linen iscale, p6, p3, p7 ; p8=vib rate a6 oscil k1, inote*.998+k2, ifunc2 ; p9=vib depth a5 oscil k1, inote*1.002+k2, ifunc2 ; p10=vib delay (0-1) a4 oscil k1, inote+k2, ifunc2 ; p11=initial wave a3 oscil k1, inote*.997+k2, ifunc1 ; p12=cross wave a2 oscil k1, inote*1.003+k2, ifunc1 ; p13=fade time a1 oscil k1, inote+k2, ifunc1 kfade linseg 1, ifad1, 0, ifad2, 1 afunc1 = kfade * (a1+a2+a3) afunc2 = (1 - kfade) * (a4+a5+a6) out afunc1 + afunc2 endin f1 0 2048 10 1 ; Sine f2 0 2048 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth f3 0 2048 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square f4 0 2048 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse ;in st dur amp frq atk rel vbrt vbdp vbdl stw ndw crstm i7 0 5 96 8.07 .03 .1 5 6 .99 1 2 .1 i7 6 5 96 8.09 .03 .1 5 6 .99 1 3 .1 i7 12 8 96 8.07 .03 .1 5 6 .99 1 4 . e manual_src~dfsg/examples/flashtxt.csd0000644000000000000000000000116612262561504017117 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o flashtxt.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 instr 1 flashtxt 1, "Instr 1 live" ao oscil 4000, 440, 1 out ao endin ; Table 1: an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 e manual_src~dfsg/examples/randh.csd0000644000000000000000000000143612262561504016356 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o randh.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;same values every time krnd randh 100, 10 printk2 krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin instr 2 ;different values every time krnd randh 100, 10, 10 ; seed from system clock printk2 krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin f 1 0 16384 10 1 ;sine wave. i 1 0 1 i 2 2 1 e manual_src~dfsg/examples/oscil1i.csd0000644000000000000000000000216012262561504016620 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil1i.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipanfn = p4 asig vco2 .3, 220 kpan oscil1i 0, 1, p3, ipanfn ;create panning & kleft = sqrt(kpan) ;start right away kright = sqrt(1-kpan) outs kleft*asig, kright*asig endin f 1 0 3 -7 .5 3 .5 ;remain in center (.5 CONSTANT) f 2 0 129 7 1 129 0 ;left-->right f 3 0 129 7 .5 32 1 64 0 33 .5 ;center-->left-->right-->center i 1 0 2 1 ;use table 1 i 1 3 2 2 ;use table 2 i 1 6 2 3 ;use table 3 e manual_src~dfsg/examples/strfromurl.csd0000644000000000000000000000107012262561504017473 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o strget.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 #include "http://codemist.co.uk/jpff/test.in" instr 2 Sfile strget "http://codemist.co.uk/jpff/test.in" prints Sfile endin i 1 0 1 i 2 + . e manual_src~dfsg/examples/FLbutton.csd0000644000000000000000000000456612262561504017026 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLbutton.wav -W ;;; for file output any platform ; Using FLbuttons to create on screen controls for play, ; stop, fast forward and fast rewind of a sound file ; This example also makes use of a preset graphic for buttons. sr = 44100 kr = 44100 ksmps = 1 nchnls = 2 FLpanel "Buttons", 240, 400, 100, 100 ion = 0 ioff = 0 itype = 1 iwidth = 50 iheight = 50 ix = 10 iy = 10 iopcode = 0 istarttim = 0 idur = -1 ;Turn instruments on idefinitely ; Normal speed forwards gkplay, ihb1 FLbutton "@>", ion, ioff, itype, iwidth, iheight, ix, iy, iopcode, 1, istarttim, idur, 1 ; Stationary gkstop, ihb2 FLbutton "@square", ion,ioff, itype, iwidth, iheight, ix+55, iy, iopcode, 2, istarttim, idur ; Double speed backwards gkrew, ihb3 FLbutton "@<<", ion, ioff, itype, iwidth, iheight, ix + 110, iy, iopcode, 1, istarttim, idur, -2 ; Double speed forward gkff, ihb4 FLbutton "@>>", ion, ioff, itype, iwidth, iheight, ix+165, iy, iopcode, 1, istarttim, idur, 2 ; Type 1 gkt1, iht1 FLbutton "1-Normal Button", ion, ioff, 1, 200, 40, ix, iy + 65, -1 ; Type 2 gkt2, iht2 FLbutton "2-Light Button", ion, ioff, 2, 200, 40, ix, iy + 110, -1 ; Type 3 gkt3, iht3 FLbutton "3-Check Button", ion, ioff, 3, 200, 40, ix, iy + 155, -1 ; Type 4 gkt4, iht4 FLbutton "4-Round Button", ion, ioff, 4, 200, 40, ix, iy + 200, -1 ; Type 21 gkt5, iht5 FLbutton "21-Plastic Button", ion, ioff, 21, 200, 40, ix, iy + 245, -1 ; Type 22 gkt6, iht6 FLbutton "22-Plastic Light Button", ion, ioff, 22, 200, 40, ix, iy + 290, -1 ; Type 23 gkt7, iht7 FLbutton "23-Plastic Check Button", ion, ioff, 23, 200, 40, ix, iy + 335, -1 FLpanelEnd FLrun ; Ensure that only 1 instance of instr 1 ; plays even if the play button is clicked repeatedly insnum = 1 icount = 1 maxalloc insnum, icount instr 1 asig diskin2 "beats.wav", p4, 0, 1 outs asig, asig endin instr 2 turnoff2 1, 0, 0 ;Turn off instr 1 turnoff ;Turn off this instrument endin ; Real-time performance for 1 hour. f 0 3600 e manual_src~dfsg/examples/array.csd0000644000000000000000000000051512262561504016375 0ustar rootroot ; Select audio/midi flags here according to platform -n sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kS[] = array(1,7,5) printk2 kS[0] printk2 kS[1] printk2 kS[2] endin i 1 0 0 e manual_src~dfsg/examples/areson.csd0000644000000000000000000000126212262561504016546 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o areson.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; unfiltered noise asig rand 0.5 ; white noise signal. outs asig, asig endin instr 2 ; filtered noise kcf init 1000 kbw init 100 asig rand 0.5 afil areson asig, kcf, kbw afil balance afil,asig ; afil = very loud outs afil, afil endin i 1 0 2 i 2 2 2 e manual_src~dfsg/examples/pwd.csd0000644000000000000000000000106012262561504016045 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ; -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o system.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 ; Waits for command to execute before continuing Swd pwd puts Swd, 1 endin ; Play Instrument #1 for minimal time. i 1 0 0.1 e manual_src~dfsg/examples/foutir.csd0000644000000000000000000000152112262561504016565 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o foutir.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gihand fiopen "foutir.sco", 0 instr 1 ; play virtual midi keyboard inot notnum ;just for priting on screen icps cpsmidi iamp ampmidi 1 foutir gihand, 0, 1, icps, iamp prints "WRITING:\n" prints "note = %f,velocity = %f\n", icps, iamp ;prints them ficlose gihand asig pluck iamp, icps, 1000, 0, 1 outs asig, asig endin f 0 10 e manual_src~dfsg/examples/readscore.csd0000644000000000000000000000047712262561504017235 0ustar rootroot -d -o dac instr 1 a1 flooper2 1000,p4,2,5,0.1,1 out a1 endin instr 2 ires readscore {{ {12 COUNT i1 $COUNT 1 [1 + $COUNT/12] } }} endin f0 12 f 1 0 0 1 "fox.wav" 0 0 1 i2 0 1 manual_src~dfsg/examples/pow.csd0000644000000000000000000000145512262561504016070 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o pow.wav ; output to audio file sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Lo-Fi sound kpow = 10 ;exponent kbase line 1, p3, 1.4 ;vary the base kQuantize pow kbase, kpow kQuantize = kQuantize*0.5 ;half the number of steps for each side of a bipolar signal printk2 kQuantize asig diskin2 "fox.wav", 1, 0, 1 ;loop the fox asig = round(asig * kQuantize) / kQuantize ;quantize and scale audio signal outs asig, asig endin i1 0 19.2 e manual_src~dfsg/examples/dcblock2.csd0000644000000000000000000000124512262561504016743 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o dcblock2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 ;add DC to "beats.wav" asig soundin "beats.wav" asig = asig+5000 ;adds DC of 5000 outs asig, asig endin instr 2 ;dcblock audio asig soundin "beats.wav" asig = asig+5000 ;adds DC adc dcblock2 asig ;remove DC again outs adc, adc endin i 1 0 2 i 2 2 2 e manual_src~dfsg/examples/dust.csd0000644000000000000000000000077712262561504016250 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kdens expon 2, p3, 20000 aout dust 0.5, kdens outs aout, aout endin i1 0 10 e manual_src~dfsg/examples/system.csd0000644000000000000000000000163712262561504016611 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ; -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o system.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 ; Waits for command to execute before continuing ires system_i 1,{{ ps date cd ~/Desktop pwd ls -l whois csounds.com }} print ires turnoff endin instr 2 ; Runs command in a separate thread ires system_i 1,{{ ps date cd ~/Desktop pwd ls -l whois csounds.com }}, 1 print ires turnoff endin ; Play Instrument #1 for thirty seconds. i 1 0 1 i 2 5 1 e manual_src~dfsg/examples/cps2pch_19et.csd0000644000000000000000000000113112262561504017456 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cps2pch_19et.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use 19ET scale. ipch = 8.02 iequal = 19 icps cps2pch ipch, iequal print icps endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/outch-2.csd0000644000000000000000000000150212262561504016535 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outch-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 seed 0 instr 1 ;random movements between 4 speakers with outch ichn1 random 1, 4.999 ;channel to start ichn2 random 1, 4.999 ;channel to end prints "Moving from speaker %d to speaker %d%n", int(ichn1), int(ichn2) asamp soundin "fox.wav" kmov linseg 0, p3, 1 a1, a2 pan2 asamp, kmov outch int(ichn1), a1, int(ichn2), a2 endin r 5 i 1 0 3 e manual_src~dfsg/examples/tablefilter.csd0000644000000000000000000000434512262561504017561 0ustar rootroot sr=44100 ksmps=10 nchnls=1 gifarn init 8 ; initialise integer for Farey Sequence F_8 gires fareyleni gifarn ; calculate length of F_8, returns 23 ; the table length won't be a power of 2 ; (The length of a Farey Sequence with n > 1 is always odd) gilen init gires * -1 gifarey ftgen 200, 0, gilen, "farey", gifarn, 0 ; initialize destiniation table with 0s gifiltered ftgen 0, 0, gilen, 21, 1, 0 ; initialize second destiniation table with 0s gifiltered2 ftgen 0, 0, gilen, 21, 1, 0 ; table filtering opcode: dest. source, mode, threshold ginumpassed tablefilteri gifiltered, gifarey, 1, 6 ; the threshold parameter indicates that denominators whose weights are heavier ; than 6 are not passing through the filter. The weight is calculated using ; Clarence Barlow's function of indigestibility of a number. According to this function, ; higher prime numbers contribute to an increased weight of any natural integer they divide. ; ginumpassed is the number of elements from the source table 'gifarey' ; that have passed the test and which have been copied to the destination table 'gifiltered' ; apply a different filter: ginumpassed2 tablefilteri gifiltered2, gifarey, 2, 5 ; In mode=2 we again test the digestibility of the denominators of the ; fractions in the source table. ; The difference to mode=1 is that we now let pass only vaules from the ; source that are as heavy as the threshold or greater. instr 4 kndx init 0 ; read out elements of now filtered F_8 sequentially and print to file if (kndx < ginumpassed) then kelem tab kndx, gifiltered fprintks "fareyfilter_lp.txt", "%2.6f\\n", kelem kndx = kndx+1 endif endin instr 5 kndx init 0 ; read out elements and print to file if (kndx < ginumpassed2) then kelem tab kndx, gifiltered2 fprintks "fareyfilter_hp.txt", "%2.6f\\n", kelem kndx = kndx+1 endif endin i4 0 1 i5 0 1 e manual_src~dfsg/examples/notnum.csd0000644000000000000000000000107412262561504016600 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M0 -+rtmidi=virtual ;;;realtime audio out with virtual MIDI in sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 massign 1, 1 ; set MIDI channel 1 to play instr 1 instr 1 iNum notnum print iNum ; Convert MIDI note number to Hz iHz = (440.0*exp(log(2.0)*((iNum)-69.0)/12.0)) aosc oscil 0.6, iHz, 1 outs aosc, aosc endin f 1 0 16384 10 1 ;sine wave f 0 60 ;play 60 seconds e manual_src~dfsg/examples/expseg.csd0000644000000000000000000000167412262561504016561 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o expseg.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; p4 = frequency in pitch-class notation. kcps = cpspch(p4) ; Create an amplitude envelope. kenv expseg 0.01, p3*0.25, 1, p3*0.75, 0.01 kamp = kenv * 30000 a1 oscil kamp, kcps, 1 out a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for a half-second, p4=8.00 i 1 0 0.5 8.00 ; Play Instrument #1 for a half-second, p4=8.01 i 1 1 0.5 8.01 ; Play Instrument #1 for a half-second, p4=8.02 i 1 2 0.5 8.02 ; Play Instrument #1 for a half-second, p4=8.03 i 1 3 0.5 8.03 e manual_src~dfsg/examples/date.csd0000644000000000000000000000076112262561504016177 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o date.wav -W ;;; for file output any platform instr 1 ii date print ii Sa dates ii prints Sa Ss dates -1 prints Ss St dates 1 prints St endin i 1 0 1 e manual_src~dfsg/examples/moog.csd0000644000000000000000000000137012262561504016220 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o moog.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = cpspch(p4) kfiltq = p5 kfiltrate = 0.0002 kvibf = 5 kvamp = .01 ;low volume is needed asig moog .15, kfreq, kfiltq, kfiltrate, kvibf, kvamp, 1, 2, 3 outs asig, asig endin f 1 0 8192 1 "mandpluk.aiff" 0 0 0 f 2 0 256 1 "impuls20.aiff" 0 0 0 f 3 0 256 10 1 ; sine i 1 0 3 6.00 .1 i 1 + 3 6.05 .89 i 1 + 3 6.09 .50 e manual_src~dfsg/examples/FLexecButton.csd0000644000000000000000000000614212262561504017623 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No display -odac -iadc -d ;;;RT audio I/O sr = 44100 ksmps = 10 nchnls = 1 ; Example by Jonathan Murphy 2007 ;;; reset amplitude range 0dbfs = 1 ;;; set the base colour for the panel FLcolor 100, 0, 200 ;;; define the panel FLpanel "FLexecButton", 250, 100, 0, 0 ;;; sliders to control time stretch and pitch gkstr, gistretch FLslider "Time", 0.5, 1.5, 0, 6, -1, 10, 60, 150, 20 gkpch, gipitch FLslider "Pitch", 0.5, 1.5, 0, 6, -1, 10, 60, 200, 20 ;;; set FLexecButton colour FLcolor 255, 255, 0 ;;; when this button is pressed, fourier analysis is performed on the file ;;; "beats.wav", producing the analysis file "beats.pvx" gipvoc FLexecButton "csound -U pvanal beats.wav beats.pvx", 60, 20, 20, 20 ;;; set FLexecButton text FLsetText "PVOC", gipvoc ;;; when this button is pressed, instr 10000 is called, exiting ;;; Csound immediately ;;; cancel previous colour FLcolor -1 ;;; set colour for kill button FLcolor 255, 0, 0 gkkill, gikill FLbutton "X", 1, 1, 1, 20, 20, 100, 20, 0, 10000, 0, 0.1 ;;; cancel previous colour FLcolor -1 ;;; set colour for play/stop and pause buttons FLcolor 0, 200, 0 ;;; pause and play/stop buttons gkpause, gipause FLbutton "@||", 1, 0, 2, 40, 20, 20, 60, -1 gkplay, giplay FLbutton "@|>", 1, 0, 2, 40, 20, 80, 60, -1 ;;; end the panel FLpanelEnd ;;; set initial values for time stretch and pitch FLsetVal_i 1, gistretch FLsetVal_i 1, gipitch ;;; run the panel FLrun instr 1 ; trigger play/stop ;;; is the play/stop button on or off? ;;; either way we need to trigger something, ;;; so we can't just use the value of gkplay kon trigger gkplay, 0, 0 koff trigger gkplay, 1, 1 ;;; if on, start instr 2 schedkwhen kon, -1, -1, 2, 0, -1 ;;; if off, stop instr 2 schedkwhen koff, -1, -1, -2, 0, -1 endin instr 2 ;;; paused or playing? if (gkpause == 1) kgoto pause kgoto start pause: ;;; if the pause button is on, skip sound production kgoto end start: ;;; get the length of the analysis file in seconds ilen filelen "beats.pvx" ;;; determine base frequency of playback icps = 1/ilen ;;; create a table over the length of the file itpt ftgen 0, 0, 513, -7, 0, 512, ilen ;;; phasor for time control kphs phasor icps * gkstr ;;; use phasor as index into table kndx = kphs * 512 ;;; read table ktpt tablei kndx, itpt ;;; use value from table as time pointer into file fsig1 pvsfread ktpt, "beats.pvx" ;;; change playback pitch fsig2 pvscale fsig1, gkpch ;;; resynthesize aout pvsynth fsig2 ;;; envelope to avoid clicks and clipping aenv linsegr 0, 0.3, 0.75, 0.1, 0 aout = aout * aenv out aout end: endin instr 10000 ; kill exitnow endin i1 0 10000 e manual_src~dfsg/examples/FLtext.csd0000644000000000000000000000272212262561504016467 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLtext.wav -W ;;; for file output any platform ; A sine with oscillator with fltext box controlled ; frequency either click and drag or double click and ; type to change frequency value sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Frequency Text Box", 270, 600, 50, 50 ; Minimum value output by the text box imin = 200 ; Maximum value output by the text box imax = 5000 ; Step size istep = 1 ; Text box graphic type itype = 1 ; Width of the text box in pixels iwidth = 70 ; Height of the text box in pixels iheight = 30 ; Distance of the left edge of the text box ; from the left edge of the panel ix = 100 ; Distance of the top edge of the text box ; from the top edge of the panel iy = 300 gkfreq,ihandle FLtext "Enter the frequency", imin, imax, istep, itype, iwidth, iheight, ix, iy ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 iamp = 15000 ifn = 1 asig oscili iamp, gkfreq, ifn out asig endin ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e manual_src~dfsg/examples/timedseq.csd0000644000000000000000000000410012262561504017064 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o timedseq.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giseq ftgen 0,0,128,-2, 2, 0, 0.5, 8.00,\ ;first note 2, 1, 0.5, 8.02,\ ;second note 2, 2, 0.5, 8.04,\ ;third 2, 3, 0.5, 8.05,\ ;fourth 2, 4, 0.5, 8.07,\ ;fifth 2, 5, 0.5, 8.09,\ ;sixth 2, 6, 0.5, 8.11,\ ;seventh 2, 7, 0.5, 9.00,\ ;eight note 2, 8, 0.5, 8.00,\ ;due to a quirk in the opcode, it needs an extra note - a copy of the first note -1, 8, -1, -1 ;last line is a dummy event that indicates to timedseq when to loop back to the beginning instr 1 ibeats = 8 ;lengths of sequence in beats itempo = p4 ;tempo iBPS = itempo/60 ;beats per second kphase phasor iBPS/ibeats ;phasor to move through table kpointer = kphase*ibeats ;multiply phase (range 0 - 1) by the number of beats contained within the sequence kp1 init 0 kp2 init 0 kp3 init 0 kp4 init 0 ktrigger timedseq kpointer, giseq, kp1, kp2,kp3, kp4 schedkwhen ktrigger, 0, 0, 2, 0, kp3/abs(iBPS), kp4 ;p3 values have been scaled according to tempo so that they maesure beats rather than seconds endin ;abs(iBPS)(absolute value) is used because the tempo provided by the fourth note of the score is negative. ;Durations here should be positive, because negative values for duration would indicate a held note. instr 2 aenv linseg 0,0.01,1,p3-0.01,0 ;amplitude envelope asig vco2 0.4, cpspch(p4), 4, 0.5 outs asig*aenv, asig*aenv endin i 1 0 4 120 i 1 + . 240 i 1 + . 480 i 1 + . -480 ;when negative it plays backwards e manual_src~dfsg/examples/convolve.csd0000644000000000000000000000367312262561504017122 0ustar rootroot ; Select audio/midi flags here according to platform ;-odac ;;;RT audio out -iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o convolve.wav -W ;;; for file output any platform ; NB: 'Small' reverbs often require a much higher ; percentage of wet signal to sound interesting. 'Large' ; reverbs seem require less. Experiment! The wet/dry mix is ; very important - a small change can make a large difference. sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;The analysis file is not system independent! ; create "rv_mono.wav" and "rv_stereo.wav" with cvanal first! instr 1 imix = 0.25 ;wet/dry mix. Vary as desired. ivol = 1 ;Overall volume level of reverb. May need to adjust ;when wet/dry mix is changed, to avoid clipping. idel filelen p4 ;calculate length and number of channels of soundfile print idel ichnls filenchnls p4 print ichnls if (ichnls == 1) then adry soundin "fox.wav" ; input (dry) audio awet convolve adry,"rv_mono.cva" ; mono convolved (wet) audio awet diff awet ; brighten adrydel delay (1-imix)*adry, idel ; Delay dry signal to align it with convolved signal ; Apply level adjustment here too. outs ivol*(adrydel+imix*awet),ivol*(adrydel+imix*awet) ; Mix wet & dry else adry soundin "fox.wav" ; input (dry) audio awet1, awet2 convolve adry,"rv_stereo.cva" ; stereo convolved (wet) audio awet1 diff awet1 ; brighten left awet2 diff awet2 ; and brighten right adrydel delay (1-imix)*adry, idel ; Delay dry signal to align it with convolved signal ; Apply level adjustment here too. outs ivol*(adrydel+imix*awet1),ivol*(adrydel+imix*awet2) ; Mix wet & dry signals endif endin i 1 0 4 "rv_mono.wav" i 1 5 4 "rv_stereo.wav" e manual_src~dfsg/examples/outs.csd0000644000000000000000000000152712262561504016255 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outs.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig vco2 .01, 110 ; sawtooth waveform at low volume ;filter a channel kcut1 line 60, p3, 300 ; Vary cutoff frequency kresonance1 = 3 inumlayer1 = 3 asig1 lowresx asig, kcut1, kresonance1, inumlayer1 ;filter the other channel kcut2 line 300, p3, 60 ; Vary cutoff frequency kresonance2 = 3 inumlayer2 = 3 asig2 lowresx asig, kcut2, kresonance2, inumlayer2 outs asig1, asig2 ; output both channels 1 & 2 endin i 1 0 3 e manual_src~dfsg/examples/equals.csd0000644000000000000000000000144212262561504016551 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o equals.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ienv = p4 ;choose envelope in score if (ienv == 0) kthen kenv adsr 0.05, 0.05, 0.95, 0.05 ;sustained envelope elseif (ienv == 1) kthen kenv adsr 0.5, 1, 0.5, 0.5 ;triangular envelope elseif (ienv == 2) kthen kenv adsr 1, 1, 1, 0 ;ramp up endif aout vco2 .1, 110, 10 aout = aout * kenv outs aout, aout endin i1 0 2 0 i1 3 2 1 i1 6 2 2 e manual_src~dfsg/examples/linen.csd0000644000000000000000000000155312262561504016367 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o linen.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; p4=amp ; p5=freq ; p6=attack time ; p7=release time ares linen p4, p6, p3, p7 asig poscil ares, p5, 1 outs asig, asig endin f1 0 4096 10 1 ; sine wave ;ins strt dur amp freq attack release i1 0 1 .5 440 0.5 0.7 i1 1.5 1 .2 440 0.9 0.1 i1 3 1 .2 880 0.02 0.99 i1 4.5 1 .2 880 0.7 0.01 i1 6 3 .7 220 0.5 0.5 e manual_src~dfsg/examples/filenchnls.csd0000644000000000000000000000147212262561504017407 0ustar rootroot ; Select audio/midi flags here according to platform ;-odac ;;;realtime audio out -iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o filechnls.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; choose between mono or stereo file ilen filelen p4 ;calculate length of soundfile ;print ilen ichn filenchnls p4 ;check number of channels print ichn if (ichn == 1) then ;mono signal asig diskin2 p4, 1 outs asig, asig else ;stereo signal aL, aR diskin2 p4, .5, 0, 1 outs aL, aR endif endin i 1 0 3 "fox.wav" ;mono signal i 1 5 2 "kickroll.wav" ;stereo signal e manual_src~dfsg/examples/unirand.csd0000644000000000000000000000146112262561504016720 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o unirand.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values ktri unirand 100 printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 ktri unirand 100 printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/128-spiral-8,16,128,2,1over20000755000000000000000000000062712262561504020470 0ustar rootroot123 17 71 97 4 110 48 30 123 10 79 89 12 117 40 47 114 10 86 80 13 116 33 55 106 11 94 64 21 116 25 63 97 12 102 56 30 115 18 78 89 12 109 49 46 107 18 86 81 21 108 41 54 106 19 94 73 29 108 33 62 98 19 93 65 37 107 26 70 90 20 101 57 46 107 26 78 81 28 101 49 54 99 27 86 73 29 100 42 62 90 27 85 65 37 100 34 62 90 28 93 57 45 99 34 70 82 28 93 50 53 99 35 77 74 36 92 50 53 91 35 85 66 37 92 42 61 82 35 manual_src~dfsg/examples/loopxseg.csd0000644000000000000000000000202312262561504017113 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loopxseg.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq rspline 0.01,20,0.2,1 ; freq. of loop repetition created by random spline ktrig init 0 ; loop restart trigger (not used) iphase = 0 ; initial phase ; loop of filter cutoff values (oct format). Rescaled further down. kcfoct loopxseg kfreq, ktrig, iphase, 1,1,0,0 kenv linseg 0,0.01,1,p3-5.01,1,5,0 asig vco2 0.2*kenv,cpsmidinn(48),0 kdep rspline 5,8,0.2,1 ; filter depth created by a random spline kcf port cpsoct((kcfoct*kdep)+4), 0.001 ; smooth filter changes asig moogladder asig,kcf,rnd(0.6) aL,aR pan2 asig,rnd(1) outs aL, aR endin i 1 0 60 e manual_src~dfsg/examples/string-128.matrix0000644000000000000000000010000012262561504017616 0ustar rootroot0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 1 0 0 0 0 0 0 0 0 0 0 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0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 manual_src~dfsg/examples/fluid-2.csd0000644000000000000000000000477212262561504016532 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out No messages Look for midifile in folder manual/examples -odac -d -F midichn_advanced.mid ;;;reatime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fluid-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; LOAD SOUNDFONTS gienginenum1 fluidEngine gienginenum2 fluidEngine isfnum1 fluidLoad "Piano Steinway Grand Model C (21,738KB).sf2", gienginenum1, 1 ; Bright Steinway, program 1, channel 1 fluidProgramSelect gienginenum1, 1, isfnum1, 0, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidProgramSelect gienginenum1, 3, isfnum1, 0, 2 isfnum2 fluidLoad "63.3mg The Sound Site Album Bank V1.0.SF2", gienginenum2, 1 ; General MIDI, program 50, channel 2 fluidProgramSelect gienginenum2, 2, isfnum2, 0, 50 ; SEND NOTES TO STEINWAY SOUNDFONT instr 1 ; FluidSynth Steinway Rev ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 1 ikey = p4 ivelocity = p5 fluidNote gienginenum1, ichannel, ikey, ivelocity endin instr 2 ; GM soundfont ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 2 ikey = p4 ivelocity = p5 fluidNote gienginenum2, ichannel, ikey, ivelocity endin instr 3 ; FluidSynth Steinway Rev ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 3 ikey = p4 ivelocity = p5 fluidNote gienginenum1, ichannel, ikey, ivelocity endin ; COLLECT AUDIO FROM ALL SOUNDFONTS instr 100 ; Fluidsynth output iamplitude1 = 7 iamplitude2 = 7 ; AUDIO aleft1, aright1 fluidOut gienginenum1 aleft2, aright2 fluidOut gienginenum2 outs (aleft1 * iamplitude1) + (aleft2 * iamplitude2), \ (aright1 * iamplitude1) + (aright2 * iamplitude2) endin i 1 0 3 60 100 i 2 1 3 60 100 i 3 3 3 63 100 i 100 0 10 ;run for 10seconds e manual_src~dfsg/examples/vco.csd0000644000000000000000000000236212262561504016050 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vco.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1 instr 1 ; Set the amplitude. kamp = p4 ; Set the frequency. kcps = cpspch(p5) ; Select the wave form. iwave = p6 ; Set the pulse-width/saw-ramp character. kpw init 0.5 ; Use Table #1. ifn = 1 ; Generate the waveform. asig vco kamp, kcps, iwave, kpw, ifn ; Output and amplification. out asig endin ; Table #1, a sine wave. f 1 0 65536 10 1 ; Define the score. ; p4 = raw amplitude (0-32767) ; p5 = frequency, in pitch-class notation. ; p6 = the waveform (1=Saw, 2=Square/PWM, 3=Tri/Saw-Ramp-Mod) i 1 00 02 20000 05.00 1 i 1 02 02 20000 05.00 2 i 1 04 02 20000 05.00 3 i 1 06 02 20000 07.00 1 i 1 08 02 20000 07.00 2 i 1 10 02 20000 07.00 3 i 1 12 02 20000 09.00 1 i 1 14 02 20000 09.00 2 i 1 16 02 20000 09.00 3 i 1 18 02 20000 11.00 1 i 1 20 02 20000 11.00 2 i 1 22 02 20000 11.00 3 e manual_src~dfsg/examples/xtratim.csd0000644000000000000000000000206412262561504016750 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in Silent MIDI in -odac -iadc -d -M0 ;;;realtime I/O ;Simple usage of the xtratim opcode sr = 44100 ksmps = 10 nchnls = 2 ; sine wave for oscillators gisin ftgen 1, 0, 4096, 10, 1 instr 1 inum notnum icps cpsmidi iamp ampmidi 4000 ; ;------- complex envelope block ------ xtratim 1 ;extra-time, i.e. release dur krel init 0 krel release ;outputs release-stage flag (0 or 1 values) if (krel == 1) kgoto rel ;if in release-stage goto release section ; ;************ attack and sustain section *********** kmp1 linseg 0, .03, 1, .05, 1, .07, 0, .08, .5, 4, 1, 50, 1 kmp = kmp1*iamp kgoto done ; ;--------- release section -------- rel: kmp2 linseg 1, .3, .2, .7, 0 kmp = kmp1*kmp2*iamp done: ;------ a1 oscili kmp, icps, gisin outs a1, a1 endin f 0 3600 ;dummy table to wait for realtime MIDI events e manual_src~dfsg/examples/zkmod.csd0000644000000000000000000000300312262561504016376 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zkmod.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Initialize the ZAK space. ; Create 2 a-rate variables and 2 k-rate variables. zakinit 2, 2 ; Instrument #1 -- a signal with jitter. instr 1 ; Generate a k-rate signal goes from 30 to 2,000. kline line 30, p3, 2000 ; Add the signal into zk variable #1. zkw kline, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Create a k-rate signal modulated the jitter opcode. kamp init 20 kcpsmin init 40 kcpsmax init 60 kjtr jitter kamp, kcpsmin, kcpsmax ; Get the frequency values from zk variable #1. kfreq zkr 1 ; Add the the frequency values in zk variable #1 to ; the jitter signal. kjfreq zkmod kjtr, 1 ; Use a simple sine waveform for the left speaker. aleft oscil 20000, kfreq, 1 ; Use a sine waveform with jitter for the right speaker. aright oscil 20000, kjfreq, 1 ; Generate the audio output. outs aleft, aright ; Clear the zk variables, prepare them for ; another pass. zkcl 0, 2 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 ; Play Instrument #2 for 2 seconds. i 2 0 2 e manual_src~dfsg/examples/subinstr.csd0000644000000000000000000000223212262561504017126 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o subinstr.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - Creates a basic tone. instr 1 ; Print the value of p4, should be equal to ; Instrument #2's iamp field. print p4 ; Print the value of p5, should be equal to ; Instrument #2's ipitch field. print p5 ; Create a tone. asig oscils p4, p5, 0 out asig endin ; Instrument #2 - Demonstrates the subinstr opcode. instr 2 iamp = 20000 ipitch = 440 ; Use Instrument #1 to create a basic sine-wave tone. ; Its p4 parameter will be set using the iamp variable. ; Its p5 parameter will be set using the ipitch variable. abasic subinstr 1, iamp, ipitch ; Output the basic tone that we have created. out abasic endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #2 for one second. i 2 0 1 e manual_src~dfsg/examples/gen33.csd0000644000000000000000000000235412262561504016201 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen33.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; partials 1, 4, 7, 10, 13, 16, etc. with base frequency of 400 Hz ibsfrq = 400 inumh = int(1.5 + sr * 0.5 / (3 * ibsfrq)) ; estimate number of partials isrcln = int(0.5 + exp(log(2) * int(1.01 + log(inumh * 3) / log(2)))) ; source table length itmp ftgen 1, 0, isrcln, -2, 0 ; create empty source table ifpos = 0 ifrq = ibsfrq inumh = 0 l1: tableiw ibsfrq / ifrq, ifpos, 1 ; amplitude tableiw ifrq, ifpos + 1, 1 ; frequency tableiw 0, ifpos + 2, 1 ; phase ifpos = ifpos + 3 ifrq = ifrq + ibsfrq * 3 inumh = inumh + 1 if (ifrq < (sr * 0.5)) igoto l1 itemp ftgen 2, 0, 262144, -33, 1, inumh, 1, -1 ; store output in ftable 2 (size = 262144) asig poscil .5, ibsfrq, itemp outs asig, asig endin i 1 0 2 e manual_src~dfsg/examples/genfarey-2.csd0000644000000000000000000000600012262561504017211 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc for RT audio input as well ; For Non-realtime ouput leave only the line below: ; -o genfarey.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; GENfarey creates table gidelta. ; The table contains the delta values of Farey Sequence 7 (p5=7). ; They are used as Inter Onset Intervals (IOIs) or event durations. ; If p6 is set to 1 for IOI output then the length of the table (p3=-18) is -(|F_7| - 1) ; Remember that a negative sign is for non-power-of-2 table lengths. ; The negative sign in front of the GEN number prevents post-normalisation of its values. gidelta ftgen 0,0,-18,"farey",7,1 ; Use GENfarey with p6 set to 2 to generate the denominators of fractions of F_7 ; this is used in this example as factors to create a series of pitches: gimult ftgen 0,0,-18,"farey",7,2 ;-------- loop and trigger instrument 901 using a Farey Sequence polyrhythm instr 1 kindx init 0 kindx2 init 0 ktrigger init 0 ktime_unit init p6 kstart init p4 kloop init p5 kinitndx init 0 kfn_times init gidelta knote init 60 kbasenote init p8 ifundam init p7 ktrigger seqtime ktime_unit, kstart, kloop, kinitndx, kfn_times if (ktrigger > 0 ) then kpitch = cpspch(ifundam) kmult tab kindx2, gimult kpitch = kpitch * kmult knote = kbasenote + kmult event "i", 901, 0, .4, .10, kpitch, kpitch * .9, 0.4, 5, .75, .8, 1.0, .15, .0, .125, .125, .25, .5, 1.0, .0, .0, .0, .0, .125, .25, .25, .25, knote kindx = kindx + 1 kindx = kindx % kloop kindx2 = kindx2 + 1 kindx2 = kindx2 % kloop endif endin ;------ basic 2 Operators FM algorithm ---------------- instr 901 inotedur = p3 imaxamp = p4 ;ampdb(p4) icarrfreq = p5 imodfreq = p6 ilowndx = p7 indxdiff = p8-p7 knote = p27 aampenv linseg p9, p14*p3, p10, p15*p3, p11, p16*p3, p12, p17*p3, p13 adevenv linseg p18, p23*p3, p19, p24*p3, p20, p25*p3, p21, p26*p3, p22 amodosc oscili (ilowndx+indxdiff*adevenv)*imodfreq, imodfreq, 10 acarosc oscili imaxamp*aampenv, icarrfreq+amodosc, 10 outs acarosc, acarosc endin f10 0 4096 10 1 ;sine wave ; p4 kstart := index offset into the Farey Sequence ; p5 kloop := end index into Farey Seq. ; p6 timefac := time in seconds for one loop to complete ; p7 fundam := fundamental of the FM instrument ; p8 basenote:= root pitch of the midi voice output ; note that pitch structures of the midi file output are not equivalent to the ; ones used for the FM real-time synthesis. ; start dur kstart kloop timefac fundam. basenote i1 0.0 44 0 18 2 6.05 60 i1 4 30 0 18 3 7.05 72 i1 34 12 9 18 3 7.05 72 i1 10 12 0 18 1.5 8 84 i1 22 12 0 9 1.5 8 84 i1 15 16 0 18 1 5 48 i1 22 20 5 17 1.7 4 36 i1 46 20 3 11 2.5 7.04 71 i1 51 20 5 13 2.5 7.06 72 i1 73.5 1.5 11 18 1.5 5.05 48 i1 75 1 12 18 1 6.03 58 e manual_src~dfsg/examples/polyaft.csd0000644000000000000000000000155212262561504016737 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o polyaft.wav -W ;;; for file output any platform sr = 44100 ksmps = 10 nchnls = 1 massign 1, 1 itmp ftgen 1, 0, 1024, 10, 1 ; sine wave instr 1 kcps cpsmidib 2 ; note frequency inote notnum ; note number kaft polyaft inote, 0, 127 ; aftertouch ; interpolate aftertouch to eliminate clicks ktmp phasor 40 ktmp trigger 1 - ktmp, 0.5, 0 kaft tlineto kaft, 0.025, ktmp ; map to sine curve for crossfade kaft = sin(kaft * 3.14159 / 254) * 22000 asnd oscili kaft, kcps, 1 out asnd endin t 0 120 f 0 9 2 -2 0 e manual_src~dfsg/examples/ampdbfs.csd0000644000000000000000000000110612262561504016670 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ampdbfs.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 idb = p4 iamp = ampdbfs(idb) asig oscil iamp, 220, 1 print iamp outs asig, asig endin ;sine wave. f 1 0 16384 10 1 i 1 0 1 -1 i 1 + 1 -5 i 1 + 1 -6 i 1 + 1 -20 e manual_src~dfsg/examples/rnd31_seed7.csd0000644000000000000000000000132712262561504017277 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd31_seed7.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; i-rate random numbers with linear distribution, seed=7. ; (Note that the seed was used only in the first call.) i1 rnd31 1, 0.5, 7 i2 rnd31 1, 0.5 i3 rnd31 1, 0.5 print i1 print i2 print i3 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/FLroller.csd0000644000000000000000000000277712262561504017014 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLroller.wav -W ;;; for file output any platform ; A sine with oscillator with flroller controlled frequency sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Frequency Roller", 900, 400, 50, 50 ; Minimum value output by the roller imin = 200 ; Maximum value output by the roller imax = 5000 ; Increment with each pixel istep = 1 ; Logarithmic type roller selected iexp = -1 ; Roller graphic type (1=horizontal) itype = 1 ; Display handle (-1=not used) idisp = -1 ; Width of the roller in pixels iwidth = 300 ; Height of the roller in pixels iheight = 50 ; Distance of the left edge of the knob ; from the left edge of the panel ix = 300 ; Distance of the top edge of the knob ; from the top edge of the panel iy = 50 gkfreq, ihandle FLroller "Frequency", imin, imax, istep, iexp, itype, idisp, iwidth, iheight, ix, iy ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 iamp = 15000 ifn = 1 asig oscili iamp, gkfreq, ifn out asig endin ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e manual_src~dfsg/examples/crunch.csd0000644000000000000000000000076712262561504016552 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o crunch.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig crunch 0.8, 0.1, 7, p4 outs asig, asig endin i1 0 1 .9 i1 1 1 .1 e manual_src~dfsg/examples/polyaft.mid0000644000000000000000000000123212262561504016732 0ustar rootrootMThdxMTrk AB@AJAZ2@EQ @E@EE@~@|E@{E@z@wE @v@uE @t@qE E @n@mE@lE@k@iE@hE@g@fE@dE@cE@bE@aE@_E@^E@]E@[E@ZE@YE @XE!@VE"E$@U@TE%@SE&@Q@PE)E*@OE+@ME,@LE.@KE/@J@HE0@GE2@F@EE3@CE4@B@AE5E7@@E8@>@=E:@<E<@:E=@9E>@8E@@7EAEB@5@4EC@3EE@2EF@0EG@/EH@.@,EJ@+EK@*EL@)EO@'@&EP@%EQ@$ES@"ETEU@!EV@ @EX@EZ@E[E]@E^@E_Ea@Eb@Ec@Ed@EfEg@Eh@Ei@Ek@@ El@ Em@ En@ EpEq@Er@Et@EuEv@EwEy@Ez@E{E|E~E@E/manual_src~dfsg/examples/igoto.csd0000644000000000000000000000205412262561504016400 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o igoto.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Get the value of the 4th p-field from the score. iparam = p4 ; If iparam is 1 then play the high note. ; If not then play the low note. if (iparam == 1) igoto highnote igoto lownote highnote: ifreq = 880 goto playit lownote: ifreq = 440 goto playit playit: ; Print the values of iparam and ifreq. print iparam print ifreq a1 oscil 10000, ifreq, 1 out a1 endin ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; p4: 1 = high note, anything else = low note ; Play Instrument #1 for one second, a low note. i 1 0 1 0 ; Play a Instrument #1 for one second, a high note. i 1 1 1 1 e manual_src~dfsg/examples/signum.csd0000644000000000000000000000132712262561504016563 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 1 gaArr[] init 2 instr 1 kEnv transeg 1, p3, -3, 0 a_pi = 4 * taninv(1.0); a1 phasor 440; a2 = sin(2 * a_pi * 1/ksmps * a1); a3 dcblock2 a2 asig = signum(a3) gaArr[0] = a2 * 0.6 * kEnv gaArr[1] = asig * 0.6 * kEnv outs gaArr[0], gaArr[1] endin i 1 0 3 manual_src~dfsg/examples/lenarray.csd0000644000000000000000000000253512262561504017100 0ustar rootroot -n -m128 ;no sound output, reduced messages ;example by joachim heintz sr = 44100 ksmps = 32 nchnls = 1 0dbfs = 1 instr 1 ;simple example kArr[] fillarray 1, 2, 3, 4 ;fill array manually printks "Length of kArr = %d\n\n", 0, lenarray(kArr) ;print out its length turnoff ;only do this in the first k-cycle endin instr 2 ;random array length iNumEls random 1, 11 ;create random number between 1 and 10 kArr[] init int(iNumEls) ;create array of this length printks "Random length of kArr = %d\n", 0, lenarray(kArr) ;print out turnoff endin instr 3 ;fill random array length with random elements iNumEls random 1, 11 ;create random number between 1 and 10 kArr[] init int(iNumEls) ;create array of this length printks "Random length of kArr = %d\n", 0, lenarray(kArr) ;print out ;fill kIndx = 0 ;initialize index until kIndx == lenarray(kArr) do kArr[kIndx] rnd31 10, 0 ;set element to random value -10...10 kIndx += 1 ;increase index od ;print kIndx = 0 ;initialize index until kIndx == lenarray(kArr) do printf("kArr[%d] = %f\n", kIndx+1, kIndx, kArr[kIndx]) kIndx += 1 ;increase index od turnoff endin i 1 0 .1 i 2 0 .1 i 2 0 .1 i 3 0 .1 manual_src~dfsg/examples/strlower.csd0000644000000000000000000000155312262561504017143 0ustar rootroot ; Select audio/midi flags here according to platform -n ;;;no sound output ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ;example of Joachim Heintz opcode FilSuf, S, So ;returns the suffix of a filename or path, optional in lower case Spath,ilow xin ipos strrindex Spath, "." ;look for the rightmost '.' Suf strsub Spath, ipos+1 ;extract the substring after "." if ilow != 0 then ;if ilow input is not 0 then Suf strlower Suf ;convert to lower case endif xout Suf endop instr suff ilow = p4 prints "Printing suffix:\n" Suf FilSuf "/my/dir/my/file.WAV", ilow puts Suf, 1 endin i "suff" 0 1 0 ;do not convert to lower case i "suff" 3 1 1 ;convert to lower case e manual_src~dfsg/examples/space_quad.csd0000644000000000000000000000217512262561504017370 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o space_quad.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 4 ga1 init 0 ga2 init 0 ga3 init 0 ga4 init 0 instr 1 ;uses GEN28 file "move", as found in /manual/examples kx init 0 ky init 0 ktime line 0, 5, 5 ;same time as in table 1 (="move") asig diskin2 "beats.wav", 1, 0, 1 ;sound source is looped a1, a2, a3, a4 space asig, 1, ktime, .1, kx, ky ;use table 1 = GEN28 ar1, ar2, ar3, ar4 spsend ;send to reverb ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 a3 reverb2 ga3, 2.5, .5 a4 reverb2 ga4, 2.5, .5 outq a1, a2, a3, a4 ga1=0 ga2=0 ga3=0 ga4=0 endin f1 0 0 28 "move" i1 0 5 ;same time as ktime i 99 0 10 ;keep reverb active e manual_src~dfsg/examples/gen02.csd0000644000000000000000000000147712262561504016202 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen02.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ;choose different tables of GEN02 kcps init 1/p3 ;index over the length of entire note kndx phasor kcps ixmode = 1 ;normalize index data kamp tablei kndx, ifn, ixmode asig poscil kamp, 440, 1 ;use GEN02 as envelope for amplitude outs asig, asig endin f 1 0 8192 10 1 ;sine wave f 2 0 5 2 0 2 0 f 3 0 5 2 0 2 10 0 f 4 0 9 2 0 2 10 100 0 i 1 0 2 2 i 1 3 2 3 i 1 6 2 4 e manual_src~dfsg/examples/xyin.csd0000644000000000000000000000224412262561504016247 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc --displays ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o xyin.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 ksmps = 10 nchnls = 2 ; Instrument #1. instr 1 ; Print and capture values every 0.1 seconds. iprd = 0.1 ; The x values are from 1 to 30. ixmin = 1 ixmax = 30 ; The y values are from 1 to 30. iymin = 1 iymax = 30 ; The initial values for X and Y are both 15. ixinit = 15 iyinit = 15 ; Get the values kx and ky using the xyin opcode. kx, ky xyin iprd, ixmin, ixmax, iymin, iymax, ixinit, iyinit ; Print out the values of kx and ky. printks "kx=%f, ky=%f\\n", iprd, kx, ky ; Play an oscillator, use the x values for amplitude and ; the y values for frequency. kamp = kx * 1000 kcps = ky * 220 a1 poscil kamp, kcps, 1 outs a1, a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 30 seconds. i 1 0 30 e manual_src~dfsg/examples/line.csd0000644000000000000000000000152512262561504016210 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o line.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kp = p6 ;choose between expon or line if (kp == 0) then kpitch expon p4, p3, p5 elseif (kp == 1) then kpitch line p4, p3, p5 endif asig vco2 .6, kpitch outs asig, asig endin i 1 0 2 300 600 0 ;if p6=0 then expon is used i 1 3 2 300 600 1 ;if p6=1 then line is used i 1 6 2 600 1200 0 i 1 9 2 600 1200 1 i 1 12 2 1200 2400 0 i 1 15 2 1200 2400 1 i 1 18 2 2400 30 0 i 1 21 2 2400 30 1 e manual_src~dfsg/examples/pvsbuffer.csd0000644000000000000000000000236312262561504017264 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsbufread.wav -W ;;; for file output any platform sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 opcode FileToPvsBuf, iik, Siiii ;;writes an audio file at the first k-cycle to a fft-buffer (via pvsbuffer) Sfile, ifftsize, ioverlap, iwinsize, iwinshape xin ktimek timeinstk if ktimek == 1 then ilen filelen Sfile kcycles = ilen * kr; number of k-cycles to write the fft-buffer kcount init 0 loop: ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape ibuf, ktim pvsbuffer fftin, ilen + (ifftsize / sr) loop_lt kcount, 1, kcycles, loop xout ibuf, ilen, ktim endif endop instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window ibuffer, ilen, k0 FileToPvsBuf "fox.wav", ifftsize, ioverlap, iwinsize, iwinshape ktmpnt linseg ilen, p3, 0; reads the buffer backwards in p3 seconds fread pvsbufread ktmpnt, ibuffer aout pvsynth fread out aout endin i 1 0 5 e manual_src~dfsg/examples/alwayson.csd0000644000000000000000000001204612262561504017116 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-Wfo alwayson.wav ; Initialize the global variables. sr = 44100 ksmps = 32 nchnls = 2 ; Connect up instruments and effects to create the signal flow graph. connect "SimpleSine", "leftout", "Reverberator", "leftin" connect "SimpleSine", "rightout", "Reverberator", "rightin" connect "Moogy", "leftout", "Reverberator", "leftin" connect "Moogy", "rightout", "Reverberator", "rightin" connect "Reverberator", "leftout", "Compressor", "leftin" connect "Reverberator", "rightout", "Compressor", "rightin" connect "Compressor", "leftout", "Soundfile", "leftin" connect "Compressor", "rightout", "Soundfile", "rightin" ; Turn on the "effect" units in the signal flow graph. alwayson "Reverberator", 0.91, 12000 alwayson "Compressor" alwayson "Soundfile" ; Define instruments and effects in order of signal flow. instr SimpleSine ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Default values: p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 pset 0, 0, 10, 0, 0, 0, 0.5 iattack = 0.015 idecay = 0.07 isustain = p3 irelease = 0.3 p3 = iattack + idecay + isustain + irelease adamping linsegr 0.0, iattack, 1.0, idecay + isustain, 1.0, irelease, 0.0 iHz = cpsmidinn(p4) ; Rescale MIDI velocity range to a musically usable range of dB. iamplitude = ampdb(p5 / 127 * 15.0 + 60.0) ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. icosine ftgenonce 0, 0, 65537, 11, 1 aoscili oscili iamplitude, iHz, icosine aadsr madsr iattack, idecay, 0.6, irelease asignal = aoscili * aadsr aleft, aright pan2 asignal, p7 ; Stereo audio output to be routed in the orchestra header. outleta "leftout", aleft outleta "rightout", aright endin instr Moogy ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Default values: p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 pset 0, 0, 10, 0, 0, 0, 0.5 iattack = 0.003 isustain = p3 irelease = 0.05 p3 = iattack + isustain + irelease adamping linsegr 0.0, iattack, 1.0, isustain, 1.0, irelease, 0.0 iHz = cpsmidinn(p4) ; Rescale MIDI velocity range to a musically usable range of dB. iamplitude = ampdb(p5 / 127 * 20.0 + 60.0) print iHz, iamplitude ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. isine ftgenonce 0, 0, 65537, 10, 1 asignal vco iamplitude, iHz, 1, 0.5, isine kfco line 2000, p3, 200 krez = 0.8 asignal moogvcf asignal, kfco, krez, 100000 asignal = asignal * adamping aleft, aright pan2 asignal, p7 ; Stereo audio output to be routed in the orchestra header. outleta "leftout", aleft outleta "rightout", aright endin instr Reverberator ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" idelay = p4 icutoff = p5 aleft, aright reverbsc aleftin, arightin, idelay, icutoff ; Stereo output. outleta "leftout", aleft outleta "rightout", aright endin instr Compressor ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" kthreshold = 25000 icomp1 = 0.5 icomp2 = 0.763 irtime = 0.1 iftime = 0.1 aleftout dam aleftin, kthreshold, icomp1, icomp2, irtime, iftime arightout dam arightin, kthreshold, icomp1, icomp2, irtime, iftime ; Stereo output. outleta "leftout", aleftout outleta "rightout", arightout endin instr Soundfile ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" outs aleftin, arightin endin ; It is not necessary to activate "effects" or create f-tables in the score! ; Overlapping notes create new instances of instruments with proper connections. i "SimpleSine" 1 5 60 85 i "SimpleSine" 2 5 64 80 i "Moogy" 3 5 67 75 i "Moogy" 4 5 71 70 ; 1 extra second after the performance e 1 manual_src~dfsg/examples/schedkwhennamed.csd0000644000000000000000000000135012262561504020405 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ; For Non-realtime ouput leave only the line below: ; -o schedkwhennamed.wav -W ;;; for file output any platform sr = 48000 ksmps = 16 nchnls = 2 0dbfs = 1 ; Example by Jonathan Murphy 2007 gSinstr2 = "printer" instr 1 ktrig metro 1 if (ktrig == 1) then ;Call instrument "printer" once per second schedkwhennamed ktrig, 0, 1, gSinstr2, 0, 1 endif endin instr printer ktime timeinsts printk2 ktime endin i1 0 10 e manual_src~dfsg/examples/compilestr.csd0000644000000000000000000000107212262561504017437 0ustar rootroot -o dac -d sr = 44100 nchnls = 1 ksmps = 32 0dbfs = 1 instr 1 ;will fail because of wrong code ires compilestr {{ instr 2 a1 oscilb p4, p5, 0 out a1 endin }} print ires ; returns -1 because not successfull ;will compile ... ires compilestr {{ instr 2 a1 oscils p4, p5, 0 out a1 endin }} print ires ; ... and returns 0 ;call the new instrument ;(note that the overall performance is extended) scoreline_i "i 2 0 3 .2 415" endin i1 0 1 manual_src~dfsg/examples/ATSaddnz-2.csd0000644000000000000000000000166112262561504017071 0ustar rootroot -odac -d -m1 ;example by joachim heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr AllTheNoise Sfile = "fox.ats" prints "Resynthesizing with all the noise.\n" iDur ATSinfo Sfile, 7 p3 = iDur ktime line 0, iDur, iDur asig ATSaddnz ktime, Sfile, 25 outs asig, asig ;start next instr event_i "i", "NoiseInBandsOfFive", iDur+1, 1, 0 endin instr NoiseInBandsOfFive Sfile = "fox.ats" prints "Resynthesizing with noise bands %d - %d.\n", p4, p4+5 iDur ATSinfo Sfile, 7 p3 = iDur ktime line 0, iDur, iDur asig ATSaddnz ktime, Sfile, 5, p4 outs asig, asig ;start next instr if p4 < 20 then event_i "i", "NoiseInBandsOfFive", iDur+1, 1, p4+5 endif endin i "AllTheNoise" 0 1 e 25 manual_src~dfsg/examples/babo.csd0000644000000000000000000000254112262561504016163 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o babo.wav -W ;;; for file output any platform /* Written by Nicola Bernardini */ sr = 44100 ksmps = 32 nchnls = 2 ; minimal babo instrument ; instr 1 ix = p4 ; x position of source iy = p5 ; y position of source iz = p6 ; z position of source ixsize = p7 ; width of the resonator iysize = p8 ; depth of the resonator izsize = p9 ; height of the resonator ainput soundin "beats.wav" al,ar babo ainput*0.7, ix, iy, iz, ixsize, iysize, izsize outs al,ar endin /* Written by Nicola Bernardini */ ; simple babo usage: ; ;p4 : x position of source ;p5 : y position of source ;p6 : z position of source ;p7 : width of the resonator ;p8 : depth of the resonator ;p9 : height of the resonator ; i 1 0 20 6 4 3 14.39 11.86 10 ; ^^^^^^^ ^^^^^^^^^^^^^^ ; ||||||| ++++++++++++++: optimal room dims according to ; ||||||| Milner and Bernard JASA 85(2), 1989 ; +++++++++: source position e manual_src~dfsg/examples/phaser2.csd0000644000000000000000000000266012262561504016626 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o phaser2.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 2 ; demonstration of phase shifting abilities of phaser2. ; Input mixed with output of phaser2 to generate notches. ; Demonstrates the interaction of imode and ksep. idur = p3 iamp = p4 * .04 iorder = p5 ; number of 2nd-order stages in phaser2 network ifreq = p6 ; not used ifeed = p7 ; amount of feedback for phaser2 imode = p8 ; mode for frequency scaling isep = p9 ; used with imode to determine notch frequencies kamp linseg 0, .2, iamp, idur - .2, iamp, .2, 0 iharms = (sr*.4) / 100 ; "Sawtooth" waveform exponentially decaying function, to control notch frequencies asig gbuzz 1, 100, iharms, 1, .95, 2 kline expseg 1, idur, .005 aphs phaser2 asig, kline * 2000, .5, iorder, imode, isep, ifeed out (asig + aphs) * iamp endin ; cosine wave for gbuzz f2 0 8192 9 1 1 .25 ; phaser2, imode=1 i2 00 10 7000 8 .2 .9 1 .33 i2 11 10 7000 8 .2 .9 1 2 ; phaser2, imode=2 i2 22 10 7000 8 .2 .9 2 .33 i2 33 10 7000 8 .2 .9 2 2 e manual_src~dfsg/examples/tigoto.csd0000644000000000000000000000213712262561504016566 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tigoto.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = abs(p3) ;make p3 positive even if p3 is negative in score itiv tival i1 = -1 ;assume this is tied note, so keep fase of oscili tigoto slur ;no reinitialisation on tied notes i1 = 0 ;first note, so reset phase aatt line p4, idur, 0 ;primary envelope slur: if itiv==0 kgoto note ;no expression on first and second note aslur linseg 0, idur*.3, p4, idur*.7, 0 ;envelope for slurred note aatt = aatt + aslur note: asig oscili aatt, p5, 1, i1 outs asig, asig endin f1 0 4096 10 1 ;sine wave i1 0 -5 .8 451 ;p3 = 5 seconds i1 1.5 -1.5 .1 512 i1 3 2 .7 440 ;3 notes together--> duration = 5 seconds e manual_src~dfsg/examples/midiin.csd0000644000000000000000000000364612262561504016540 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -+rtmidi=virtual ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o midiin.wav -W ;;; for file output any platform sr = 44100 ksmps = 10 nchnls = 1 ; Example by schwaahed 2006 massign 0, 130 ; make sure that all channels pgmassign 0, 130 ; and programs are assigned to test instr instr 130 knotelength init 0 knoteontime init 0 kstatus, kchan, kdata1, kdata2 midiin if (kstatus == 128) then knoteofftime times knotelength = knoteofftime - knoteontime printks "kstatus= %d, kchan = %d, \\tnote# = %d, velocity = %d \\tNote OFF\\t%f %f\\n", 0, kstatus, kchan, kdata1,kdata2, knoteofftime, knotelength elseif (kstatus == 144) then knoteontime times printks "kstatus= %d, kchan = %d, \\tnote# = %d, velocity = %d \\tNote ON\\t%f\\n", 0, kstatus, kchan, kdata1, kdata2, knoteontime elseif (kstatus == 160) then printks "kstatus= %d, kchan = %d, \\tkdata1 = %d, kdata2 = %d \\tPolyphonic Aftertouch\\n", 0, kstatus, kchan, kdata1, kdata2 elseif (kstatus == 176) then printks "kstatus= %d, kchan = %d, \\t CC = %d, value = %d \\tControl Change\\n", 0, kstatus, kchan, kdata1, kdata2 elseif (kstatus == 192) then printks "kstatus= %d, kchan = %d, \\tkdata1 = %d, kdata2 = %d \\tProgram Change\\n", 0, kstatus, kchan, kdata1, kdata2 elseif (kstatus == 208) then printks "kstatus= %d, kchan = %d, \\tkdata1 = %d, kdata2 = %d \\tChannel Aftertouch\\n", 0, kstatus, kchan, kdata1, kdata2 elseif (kstatus == 224) then printks "kstatus= %d, kchan = %d, \\t ( data1 , kdata2 ) = ( %d, %d )\\tPitch Bend\\n", 0, kstatus, kchan, kdata1, kdata2 endif endin i130 0 3600 e manual_src~dfsg/examples/midipolyaftertouch.csd0000644000000000000000000000226412262561504021175 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midipolyaftertouch.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1;midi keyboard and score are used midinoteoncps p4, p5 ;puts MIDI key translated to cycles per second into p4, and MIDI velocity into p5 kafter init 127 ;full open midipolyaftertouch kafter, 1 printk2 kafter ;display the key value when it changes and when key is pressed kvel = p5/127 ;scale midi velocity to 0-1 kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by ain pluck kenv*kvel, p4, p4, 2, 1 ;velocity value asig moogvcf2 ain, 300+(kafter*100), .5 ;use value of aftertouch to control frequency of filter outs asig, asig ;base freq of filter = 300 Hz endin f 0 30 ;runs for 30 seconds f 2 0 4096 10 1 ;sine wave i 1 0 2 440 100 ;play these notes from score as well i 1 + 2 1440 100 e manual_src~dfsg/examples/oscil1.csd0000644000000000000000000000150112262561504016445 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil1.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giPan ftgen 0, 0, 8, -2, .5, .2, .8, .1, .9, 0, 1, .5 instr 1 istay = 2 ;how many seconds to stay on the first table value asig vco2 .3, 220 kpan oscil1 istay, 1, p3-istay, giPan ;create panning printk2 kpan ;print when new value aL, aR pan2 asig, kpan ;apply panning outs aL, aR endin i 1 0 10 e manual_src~dfsg/examples/fluidEngine.csd0000644000000000000000000000466012262561504017515 0ustar rootroot ; Select audio/midi flags here according to platform -odac -F midichn_advanced.mid ;;;reatime audio out and midifile in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fluidEngine.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; LOAD SOUNDFONTS gienginenum1 fluidEngine gienginenum2 fluidEngine isfnum1 fluidLoad "sf_GMbank.sf2", gienginenum1, 1 ; Piano 2, program 1, channel 1 fluidProgramSelect gienginenum1, 1, isfnum1, 0, 1 ; Piano 3, program 2, channel 2 fluidProgramSelect gienginenum1, 2, isfnum1, 0, 2 isfnum2 fluidLoad "19Trumpet.sf2", gienginenum2, 1 ; Trumpet, program 56, channel 3 fluidProgramSelect gienginenum2, 3, isfnum2, 0, 56 ;Look for midifile in folder manual/examples ;"midichn_advanced.mid" sends notes to the soundfonts instr 1 ; GM soundfont ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 1 ikey = p4 ivelocity = p5 fluidNote gienginenum1, ichannel, ikey, ivelocity endin instr 2 ; GM soundfont ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 2 ikey = p4 ivelocity = p5 fluidNote gienginenum1, ichannel, ikey, ivelocity endin instr 3 ; Trumpet ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 3 ikey = p4 ivelocity = p5 fluidNote gienginenum2, ichannel, ikey, ivelocity endin ; COLLECT AUDIO FROM ALL SOUNDFONTS instr 100 ; Fluidsynth output iamplitude1 = 7 iamplitude2 = 7 ; AUDIO aleft1, aright1 fluidOut gienginenum1 aleft2, aright2 fluidOut gienginenum2 outs (aleft1 * iamplitude1) + (aleft2 * iamplitude2), \ (aright1 * iamplitude1) + (aright2 * iamplitude2) endin i 1 0 3 60 100 i 2 1 3 60 100 i 3 3 3 63 100 i 100 0 10 ;run for 10 seconds e manual_src~dfsg/examples/gensone.csd0000644000000000000000000000204112262561504016711 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gensone.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; simple oscillator with loudness correction. kcps = cpspch(p4) kenv linseg 0, p3*0.25, 1, p3*0.75, 0 ;amplitude envelope kamp tablei 16384 *kenv, 2 asig oscil kamp, kcps, 1 outs asig, asig endin instr 2 ;neutral oscillator to compare with kcps = cpspch(p4) kenv linseg 0, p3*0.25, 1, p3*0.75, 0 ;amplitude envelope asig oscil kenv, kcps, 1 outs asig, asig endin f 1 0 16384 10 1 ;sine wave f 2 0 16385 "sone" 0 32000 32000 0 s f 0 1 ;1 second of silence before we start... s i 1 0 2 7.00 i 1 + . 7.01 i 1 + . 8.02 i 1 + . 8.03 s i 2 0 2 7.00 i 2 + . 7.01 i 2 + . 8.02 i 2 + . 8.03 e manual_src~dfsg/examples/FLpanel.csd0000644000000000000000000000151312262561504016577 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLpanel.wav -W ;;; for file output any platform ; Creates an empty window panel sr = 44100 kr = 441 ksmps = 100 nchnls = 1 ; Panel height in pixels ipanelheight = 900 ; Panel width in pixels ipanelwidth = 400 ; Horizontal position of the panel on screen in pixels ix = 50 ; Vertical position of the panel on screen in pixels iy = 50 FLpanel "A Window Panel", ipanelheight, ipanelwidth, ix, iy ; End of panel contents FLpanelEnd ;Run the widget thread! FLrun instr 1 endin ; 'Dummy' score event of 1 hour. f 0 3600 e manual_src~dfsg/examples/cps2pch.csd0000644000000000000000000000114412262561504016620 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cps2pch.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a normal twelve-tone scale. ipch = 8.02 iequal = 12 icps cps2pch ipch, iequal print icps endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/adsyn.csd0000644000000000000000000000114012262561504016370 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o adsyn.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; analyze the file "kickroll.wav" first kamod = 1 kfmod = p4 ksmod = p5 asig adsyn kamod, kfmod, ksmod, "kickroll.het" outs asig, asig endin i 1 0 4 1 .2 i 1 + 1 2 1 i 1 + 1 .3 1.5 e manual_src~dfsg/examples/mp3len.csd0000644000000000000000000000113312262561504016452 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o mp3len.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ilen mp3len p4 ;calculate length of mp3 file print ilen asigL, asigR mp3in p4 outs asigL, asigR endin i 1 0 30 "XORNOT_jul-14-05.mp3" ; long signal e manual_src~dfsg/examples/toot4.csd0000644000000000000000000000204012262561504016323 0ustar rootroot -odac instr 4 iamp = ampdb(p4) ; convert decibels to linear amp iscale = iamp * .333 ; scale the amp at initialization inote = cpspch(p5) ; convert octave.pitch to cps k1 linen iscale, p6, p3, p7 ; p4=amp a3 oscil k1, inote*.996, 1 ; p5=freq a2 oscil k1, inote*1.004, 1 ; p6=attack time a1 oscil k1, inote, 1 ; p7=release time a1 = a1+a2+a3 out a1 endin f1 0 4096 10 1 ; sine wave ;ins strt dur amp freq attack release i4 0 1 75 8.04 0.1 0.7 i4 1 1 70 8.02 0.07 0.6 i4 2 1 75 8.00 0.05 0.5 i4 3 1 70 8.02 0.05 0.4 i4 4 1 85 8.04 0.1 0.5 i4 5 1 80 8.04 0.05 0.5 i4 6 2 90 8.04 0.03 1. manual_src~dfsg/examples/tabsum.csd0000644000000000000000000000145112262561504016552 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ; For Non-realtime ouput leave only the line below: ; -o tab.wav -W ;;; for file output any platform sr = 44100 ksmps = 2205 nchnls = 1 0dbfs = 1 instr 1 ;;; Give a value to the increment kmax = 256 knorm tabsum 1, 0, kmax gkinc = knorm/10 endin instr 2 kmax = 256 kx = rnd(kmax) krnd tabsum 1, 0, kx knorm tabsum 1, 0, kmax kvar = krnd / knorm ;;; now n [0,1] range asig oscil kvar, p4, 2 out asig ;;; Make randomness give 1 more often kc tab 0, 1 tablew kc+gkinc, 0, 1 endin f1 0 256 21 1 f2 0 4096 10 1 i1 0 0.1 i2 0.1 3 440 e manual_src~dfsg/examples/vpow.csd0000644000000000000000000000146612262561504016260 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vpow ifn1, ival, ielements, idstoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e manual_src~dfsg/examples/virtual.csd0000644000000000000000000000133712262561504016750 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in Virtual MIDI -M0 is needed anyway -odac -iadc -+rtmidi=virtual -M0 ; By Mark Jamerson 2007 sr=44100 ksmps=10 nchnls=2 massign 1,1 prealloc 1,10 instr 1 ;Midi FM synth inote cpsmidi iveloc ampmidi 10000 idur = 2 xtratim 1 kgate oscil 1,10,2 anoise noise 100*inote,.99 acps samphold anoise,kgate aosc oscili 1000,acps,1 aout = aosc ; Use controller 7 to control volume kvol ctrl7 1, 7, 0.2, 1 outs kvol * aout, kvol * aout endin f0 3600 f1 0 1024 10 1 f2 0 16 7 1 8 0 8 f3 0 1024 10 1 .5 .6 .3 .2 .5 e manual_src~dfsg/examples/turnoff.csd0000644000000000000000000000134012262561504016737 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o turnoff.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 k1 expon 440, p3/10,880 ; begin gliss and continue if k1 < sr/2 kgoto contin ; until Nyquist detected turnoff ; then quit contin: a1 oscil 10000, k1, 1 out a1 endin ; Table #1: an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for 4 seconds. i 1 0 4 e manual_src~dfsg/examples/rnd31.csd0000644000000000000000000000144212262561504016206 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd31.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create random numbers at a-rate in the range -2 to 2 with ; a triangular distribution, seed from the current time. a31 rnd31 2, -0.5 ; Use the random numbers to choose a frequency. afreq = a31 * 500 + 100 a1 oscil 30000, afreq, 1 out a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/lphasor.csd0000644000000000000000000000203712262561504016730 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o lphashor.wav -W ;;; for file output any platform ; Example by Jonathan Murphy Dec 2006 sr = 44100 ksmps = 10 nchnls = 1 instr 1 ifn = 1 ; table number ilen = nsamp(ifn) ; return actual number of samples in table itrns = 1 ; no transposition ilps = 0 ; loop starts at index 0 ilpe = ilen ; ends at value returned by nsamp above imode = 3 ; loop forwards & backwards istrt = 10000 ; commence playback at index 10000 samples ; lphasor provides index into f1 alphs lphasor itrns, ilps, ilpe, imode, istrt atab tablei alphs, ifn ; amplify signal atab = atab * 10000 out atab endin f 1 0 262144 1 "beats.wav" 0 4 1 i1 0 60 e manual_src~dfsg/examples/e.csd0000644000000000000000000000136612262561504015510 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o e.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ga1 init 0 instr 1 aenv expseg .01, p3*0.1, 1, p3*0.9, 0.01 ga1 poscil3 .5*aenv, cpspch(p4), 1 outs ga1,ga1 endin instr 99 aL, aR reverbsc ga1, ga1, 0.85, 12000, sr, 0.5, 1 outs aL,aR ga1 = 0 endin f 1 0 128 10 1 ;sine wave i 1 1 0.1 8.00 i 1 2 0.1 8.02 i 1 3 0.1 8.04 i 1 4 0.1 9.06 i 99 0 6 ;remains active for 6 seconds e10 manual_src~dfsg/examples/maxaccum.csd0000644000000000000000000000161012262561504017052 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o maxaccum.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;saw asig vco2 .2, p4 outs asig, asig gasaw = asig endin instr 2 ;sine aout poscil .3, p4, 1 outs aout, aout gasin = aout endin instr 10 accum init 0 maxaccum accum, gasaw + gasin ;saw and sine accumulated outs accum, accum clear accum endin f 1 0 4096 10 1 i 1 0 7 330 i 2 3 3 440 i 1 10 7 330 ;same notes but without maxaccum, for comparison i 2 13 3 440 i 10 0 6 ;accumulation note stops after 6 seconds e manual_src~dfsg/examples/pchbend.csd0000644000000000000000000000133212262561504016660 0ustar rootroot ; Select audio/midi flags here according to platform -odac -Q1 -Ma ;;;realtime audio out and midi in (on all inputs) and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pchbend.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;use external midi keyboard icps cpsmidi kbnd pchbend 0, 100 ;one octave lower and higher kenv linsegr 0,.001, 1, .1, 0 ;amplitude envelope asig pluck .8 * kenv, icps+kbnd, 440, 0, 1 outs asig, asig endin f 0 30 ;runs 30 seconds manual_src~dfsg/examples/voice.csd0000644000000000000000000000166512262561504016373 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o voice.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = p4 kphon = p5 asig voice kamp, 200, kphon, 0.488, 0, 1, 1, 2 outs asig, asig endin f 1 0 256 1 "impuls20.aiff" 0 0 0 ;audio file for the carrier waveform f 2 0 256 10 1 ;sine wave for the vibrato waveform ; ampl phoneme i 1 0 2 0.8 1 i 1 + . 0.6 2 i 1 + . 1.8 3 i 1 + . 15.0 4 i 1 + . 0.05 5 i 1 + . 0.06 6 i 1 + . 0.03 7 i 1 + . 0.0002 8 i 1 + . 0.1 9 i 1 + . 0.5 10 i 1 + . 100 11 i 1 + . 0.03 12 i 1 + . 0.04 13 i 1 + . 0.04 14 i 1 + . 0.04 15 i 1 + . 0.05 16 e manual_src~dfsg/examples/grain2.csd0000644000000000000000000000435612262561504016450 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o grain2.wav -W ;;; for file output any platform sr = 48000 kr = 750 ksmps = 64 nchnls = 2 /* square wave */ i_ ftgen 1, 0, 4096, 7, 1, 2048, 1, 0, -1, 2048, -1 /* window */ i_ ftgen 2, 0, 16384, 7, 0, 4096, 1, 4096, 0.3333, 8192, 0 /* sine wave */ i_ ftgen 3, 0, 1024, 10, 1 /* room parameters */ i_ ftgen 7, 0, 64, -2, 4, 50, -1, -1, -1, 11, \ 1, 26.833, 0.05, 0.85, 10000, 0.8, 0.5, 2, \ 1, 1.753, 0.05, 0.85, 5000, 0.8, 0.5, 2, \ 1, 39.451, 0.05, 0.85, 7000, 0.8, 0.5, 2, \ 1, 33.503, 0.05, 0.85, 7000, 0.8, 0.5, 2, \ 1, 36.151, 0.05, 0.85, 7000, 0.8, 0.5, 2, \ 1, 29.633, 0.05, 0.85, 7000, 0.8, 0.5, 2 ga01 init 0 /* generate bandlimited square waves */ i0 = 0 loop1: imaxh = sr / (2 * 440.0 * exp(log(2.0) * (i0 - 69) / 12)) i_ ftgen i0 + 256, 0, 4096, -30, 1, 1, imaxh i0 = i0 + 1 if (i0 < 127.5) igoto loop1 instr 1 p3 = p3 + 0.2 /* note velocity */ iamp = 0.0039 + p5 * p5 / 16192 /* vibrato */ kcps oscili 1, 8, 3 kenv linseg 0, 0.05, 0, 0.1, 1, 1, 1 /* frequency */ kcps = (kcps * kenv * 0.01 + 1) * 440 * exp(log(2) * (p4 - 69) / 12) /* grain ftable */ kfn = int(256 + 69 + 0.5 + 12 * log(kcps / 440) / log(2)) /* grain duration */ kgdur port 100, 0.1, 20 kgdur = kgdur / kcps a1 grain2 kcps, kcps * 0.02, kgdur, 50, kfn, 2, -0.5, 22, 2 a1 butterlp a1, 3000 a2 grain2 kcps, kcps * 0.02, 4 / kcps, 50, kfn, 2, -0.5, 23, 2 a2 butterbp a2, 12000, 8000 a2 butterbp a2, 12000, 8000 aenv1 linseg 0, 0.01, 1, 1, 1 aenv2 linseg 3, 0.05, 1, 1, 1 aenv3 linseg 1, p3 - 0.2, 1, 0.07, 0, 1, 0 a1 = aenv1 * aenv3 * (a1 + a2 * 0.7 * aenv2) ga01 = ga01 + a1 * 10000 * iamp endin /* output instr */ instr 81 i1 = 0.000001 aLl, aLh, aRl, aRh spat3di ga01 + i1*i1*i1*i1, 3.0, 4.0, 0.0, 0.5, 7, 4 ga01 = 0 aLl butterlp aLl, 800.0 aRl butterlp aRl, 800.0 outs aLl + aLh, aRl + aRh endin t 0 60 i 1 0.0 1.3 60 127 i 1 2.0 1.3 67 127 i 1 4.0 1.3 64 112 i 1 4.0 1.3 72 112 i 81 0 6.4 e manual_src~dfsg/examples/lowresx.csd0000644000000000000000000000145012262561504016761 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lowresx.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kres = p4 inumlayer = 4 kenv linseg 0, p3*.1, 1, p3*.8, 1, p3*.1, 0 ;envelope asig vco .3 * kenv, 220, 1 ;sawtooth kcut line 30, p3, 1000 ;note: kcut is not in Hz alx lowresx asig, kcut, kres, inumlayer ;note: kres is not in dB aout balance alx, asig ;avoid very loud sounds outs aout, aout endin ;sine wave f 1 0 16384 10 1 i 1 0 5 1 i 1 + 5 3 i 1 + 5 20 e manual_src~dfsg/examples/modmatrix.csd0000644000000000000000000001001312262561504017255 0ustar rootroot ; Select audio flags here according to platform ; Audio out Audio in ;-odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -o modmatrix.wav -W ;;; for file output any platform sr = 44100 kr = 441 ksmps = 100 nchnls = 2 0dbfs = 1 ; basic waveforms giSine ftgen 0, 0, 65537, 10, 1 ; sine wave giSaw ftgen 0, 0, 4097, 7, 1, 4096, -1 ; saw (linear) giSoftSaw ftgen 0, 0, 65537, 30, giSaw, 1, 10 ; soft saw (only 10 first harmonics) ; modmatrix tables giMaxNumParam = 128 giMaxNumMod = 32 giParam_In ftgen 0, 0, giMaxNumParam, 2, 0 ; input parameters table ; output parameters table (parameter values with added modulators) giParam_Out ftgen 0, 0, giMaxNumParam, 2, 0 giModulators ftgen 0, 0, giMaxNumMod, 2, 0 ; modulators table ; modulation scaling and routing (mod matrix) table, start with empty table giModScale ftgen 0, 0, giMaxNumParam*giMaxNumMod, -2, 0 ;******************************************** ; generate the modulator signals ;******************************************** instr 1 ; LFO1, 1.5 Hz, normalized range (0.0 to 1.0) kLFO1 oscil 0.5, 1.5, giSine ; generate LFO signal kLFO1 = kLFO1+0.5 ; offset ; LFO2, 0.4 Hz, normalized range (0.0 to 1.0) kLFO2 oscil 0.5, 0.4, giSine ; generate LFO signal kLFO2 = kLFO2+0.5 ; offset ; write modulators to table tablew kLFO1, 0, giModulators tablew kLFO2, 1, giModulators endin ;******************************************** ; set parameter values ;******************************************** instr 2 ; Here we can set the parameter values icps1 = p4 icps2 = p5 icutoff = p6 ; write parameters to table tableiw icps1, 0, giParam_In tableiw icps2, 1, giParam_In tableiw icutoff, 2, giParam_In endin ;******************************************** ; mod matrix edit ;******************************************** instr 3 ; Here we can write to the modmatrix table by using tablew or tableiw iLfo1ToCps1 = p4 iLfo1ToCps2 = p5 iLfo1ToCutoff = p6 iLfo2ToCps1 = p7 iLfo2ToCps2 = p8 iLfo2ToCutoff = p9 tableiw iLfo1ToCps1, 0, giModScale tableiw iLfo1ToCps2, 1, giModScale tableiw iLfo1ToCutoff, 2, giModScale tableiw iLfo2ToCps1, 3, giModScale tableiw iLfo2ToCps2, 4, giModScale tableiw iLfo2ToCutoff, 5, giModScale ; and set the update flag for modulator matrix ; ***(must update to enable changes) ktrig init 1 chnset ktrig, "modulatorUpdateFlag" ktrig = 0 endin ;******************************************** ; mod matrix ;******************************************** instr 4 ; get the update flag kupdate chnget "modulatorUpdateFlag" ; run the mod matrix inum_mod = 2 inum_parm = 3 modmatrix giParam_Out, giModulators, giParam_In, \\ giModScale, inum_mod, inum_parm, kupdate ; and reset the update flag chnset 0, "modulatorUpdateFlag" ; reset the update flag endin ;******************************************** ; audio generator to test values ;******************************************** instr 5 ; basic parameters iamp = ampdbfs(-5) ; read modulated parameters from table kcps1 table 0, giParam_Out kcps2 table 1, giParam_Out kcutoff table 2, giParam_Out ; set filter parameters kCF_freq1 = kcps1*kcutoff kCF_freq2 = kcps2*kcutoff kReso = 0.7 kDist = 0.3 ; oscillators and filters a1 oscili iamp, kcps1, giSoftSaw a1 lpf18 a1, kCF_freq1, kReso, kDist a2 oscili iamp, kcps2, giSoftSaw a2 lpf18 a2, kCF_freq2, kReso, kDist outs a1, a2 endin ;******************************************** ; set initial parameters ; cps1 cps2 cutoff i2 0 1 400 800 3 ;******************************************** ; set modmatrix values ; lfo1ToCps1 lfo1ToCps2 lfo1ToCut lfo2ToCps1 lfo2ToCps2 lfo2ToCut i3 0 1 40 0 -2 -50 100 3 ;******************************************** ; start "always on" instruments #define SCORELEN # 20 # ; set length of score i1 0 $SCORELEN ; start modulators i4 0 $SCORELEN ; start mod matrix i5 0 $SCORELEN ; start audio oscillator e manual_src~dfsg/examples/adsr.csd0000644000000000000000000000135112262561504016207 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o adsr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iatt = p5 idec = p6 islev = p7 irel = p8 kenv adsr iatt, idec, islev, irel kcps = cpspch(p4) ;frequency asig vco2 kenv * 0.8, kcps outs asig, asig endin i 1 0 1 7.00 .0001 1 .01 .001 ; short attack i 1 2 1 7.02 1 .5 .01 .001 ; long attack i 1 4 2 6.09 .0001 1 .1 .7 ; long release e manual_src~dfsg/examples/table1.inc0000644000000000000000000000011412262561504016422 0ustar rootroot/* table1.inc */ ; Table #1, a sine wave. f 1 0 16384 10 1 /* table1.inc */ manual_src~dfsg/examples/FLpanel-sensekey2.csd0000644000000000000000000000213712262561504020510 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac ; -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLpanel-sensekey2.wav -W ;;; for file output any platform sr = 48000 ksmps = 32 nchnls = 1 ; Example by Istvan Varga ; if the FLTK opcodes are commented out, sensekey will read keyboard ; events from stdin FLpanel "", 150, 50, 100, 100, 0, 1 FLlabel 18, 10, 1, 0, 0, 0 FLgroup "Keyboard Input", 150, 50, 0, 0, 0 FLgroupEnd FLpanelEnd FLrun instr 1 ktrig1 init 1 ktrig2 init 1 nxtKey1: k1, k2 sensekey if (k1 != -1 || k2 != 0) then printf "Key code = %02X, state = %d\n", ktrig1, k1, k2 ktrig1 = 3 - ktrig1 kgoto nxtKey1 endif nxtKey2: k3 sensekey if (k3 != -1) then printf "Character = '%c'\n", ktrig2, k3 ktrig2 = 3 - ktrig2 kgoto nxtKey2 endif endin i 1 0 3600 e manual_src~dfsg/examples/rv_stereo.wav0000644000000000000000000020723612262561504017324 0ustar rootrootRIFFWAVEfmt DLISTINFOIARTmennodataX Ӿ < ،޶hg2M{C/hb5 Ġs  `#+NslZY  z@;B c'(,// .W -)I(%D5!%ۃҬElŋ *?`}؉ ܩ@ӈMu= 9<$/9"DRK.xPRRPnqLE"GAL;H4Q-e(&-0 W6E9-W71C', In  OfzB1ϲ"a&~(qy+e-Vp.ʹv._R.-f*K«&$ӫߓn "*g/{2K2І0<-̚(D$Ώ# % G&(.5t9|=/D' $ImKM=OP jOM KF@6+4 Bj2c벼 H[g N('S uvVJD  P0 V M܃!O%݃ӥL ՠV7G-. !Ye<>TR0%ZfNcOI2%"ce @ \!}"5$w'g+/"6-$=AD2fGDHC;!07p#,#t^̖{a YɽA>r 1jݹה`#ǥ'Y*j**$*)$ͷ(fS6C(! 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:4.%#,!5+>5DAFIHLHLDK9I+IG D@>:2)%382) )6;;:951/14:3BEHRKXNUPGS1TPJF@92( {~  %:ISURNHCDKUd{2J^pjM,|pdS@e0G#3.5Ii2N_d]K4|slhipyxrrnij'o6pHoYrfwszywy|hP30Oh~m&].P8BB9K1R+Y$]"Z'N/<="Pas}^>jP>/!#x3mGg_i{q/E8XNf_phvf|]}QxAo3](B% %<LX_^`ba f!r iDtaZ[]fqy1?FMRRNIE? :6./C(R'\&`#a!\SF3   4%J,b0u22+"xjT;#/8>EIJJJH IG@4%"- 4:=EJLMIE</$%-1,$    ##%(+.36442,#&,/1-($    "+ 3651-'   *17AHMNNNOOKH IE#<12;)?"<71 )  $%#                   "  !" ! !               manual_src~dfsg/examples/bbcuts.csd0000644000000000000000000000143612262561504016544 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o bbcuts.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 ;Play an audio file aleft, aright diskin2 "kickroll.wav", 1, 0 outs aleft, aright endin instr 2 ;Cut-up stereo audio file. ibps = 16 isubdiv = 2 ibarlength = 2 iphrasebars = 1 inumrepeats = 8 aleft, aright diskin2 "kickroll.wav", 1, 0 aleft, aright bbcuts aleft, aright, ibps, isubdiv, ibarlength, iphrasebars, inumrepeats outs aleft, aright endin i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/resony.csd0000644000000000000000000000156512262561504016604 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o resonx.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; unfiltered noise kenv linseg 0, p3*.5, 1, p3*.5, 0 ;envelope asig rand 0.7 ;white noise outs asig*kenv, asig*kenv endin instr 2 ; filtered noise ksep = p4 ;vary seperation of center frequency of filters in octaves kenv linseg 0, p3*.5, 1, p3*.5, 0 ;envelope asig rand 0.7 kbf line 300, p3, 2000 ;vary base frequency afilt resony asig, kbf, 300, 4, ksep asig balance afilt, asig outs asig*kenv, asig*kenv endin i 1 0 2 i 2 3 2 1 i 2 6 2 3 e manual_src~dfsg/examples/logcurve.csd0000644000000000000000000000071712262561504017111 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in Silent -odac -iadc -d ;;;realtime output sr = 48000 ksmps = 100 nchnls = 2 instr 1 ; logcurve test kmod phasor 1/p3 kout logcurve kmod, p4 printks "kmod = %f kout = %f\\n", 0.1, kmod, kout endin i1 0 10 2 i1 10 10 30 i1 20 10 0.5 e manual_src~dfsg/examples/cent.csd0000644000000000000000000000167212262561504016215 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cent.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; original pitch iroot = 440 ; root note = A (440 Hz) print iroot ;print out asig oscili 0.6, iroot, 1 outs asig, asig endin instr 2 iroot = 440 ; root note = A (440 Hz) icents = p4 ; change root note by 300 and 1200 cents ifactor = cent(icents) ; calculate new note inew = iroot * ifactor print iroot ; Print all print ifactor print inew asig oscili 0.6, inew, 1 outs asig, asig endin ; sine wave f1 0 32768 10 1 i 1 0 2 0 ;no change i 2 2.5 2 300 ;note = C above A i 2 5 2 1200 ;1 octave higher e manual_src~dfsg/examples/opcode1.xml0000644000000000000000000000326712262561504016647 0ustar rootroot opcode1 opcode1 This is an example DocBook XML file for an opcode. opcode1 Description This is an example DocBook XML file for an opcode. Syntax ar opcode1 insnum Initialization insnum -- instrument number. Equivalent to p1 in a score i statement. Performance They go beep. Examples Here is an example of the opcode1 opcode. It uses the files opcode1.orc and opcode1.sco. Example of the opcode1 opcode. Credits Author: Opcode Author New in version 5.00 manual_src~dfsg/examples/pgmassign.csd0000644000000000000000000000124312262561504017246 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o pgmassign.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Program 55 (synth vox) uses Instrument #10. pgmassign 55, 10 ; Instrument #10. instr 10 ; Just an example, no working code in here! endin ; Play Instrument #10 for one second. i 10 0 1 e manual_src~dfsg/examples/STKSimple.csd0000644000000000000000000000127712262561504017100 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKSimple.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;Filter Pole Position kv2 line 20, p3, 90 ;Noise/Pitched Cross-Fade asig STKSimple cpspch(p4), 1, 2, kv1, 4, kv2, 11, 10, 128, 120 outs asig, asig endin i 1 0 5 7.00 100 0 i 1 + . 7.05 10 127 i 1 + . 8.03 100 0 i 1 + . 5.00 10 10 e manual_src~dfsg/examples/pvsinfo.csd0000644000000000000000000000142612262561504016745 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsinfo.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; create a PVOC-EX (*.pvx) file with PVANAL first idur filelen "fox.pvx" ;find duration of analysis file kpos line 0,p3,idur ;to ensure we process whole file fsrc pvsfread kpos, "fox.pvx" ;create fsig from (mono) file iovl,inb,iws,ifmt pvsinfo fsrc ;get info print iovl,inb,iws,ifmt ;print info aout pvsynth fsrc outs aout, aout endin i 1 0 3 e manual_src~dfsg/examples/initc7.csd0000644000000000000000000000122112262561504016447 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M0 ;;;realtime audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if RT audio input is needed too sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; expects MIDI controller input on channel 1 ; run and move your midi controller to see result imax = 1 imin = 0 ichan = 1 ictlno = 7 initc7 1, 7, 1 ; start at max. volume kamp ctrl7 ichan, ictlno, imin, imax ; controller 7 printk2 kamp asig oscil kamp, 220, 1 outs asig, asig endin f 1 0 4096 10 1 i1 0 20 e manual_src~dfsg/examples/sin.csd0000644000000000000000000000243112262561504016047 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sin.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 isin1 = sin(0) ;sine of 0 is 0 isin2 = sin($M_PI_2) ;sine of pi/2 (1.5707...) is 1 isin3 = sin($M_PI) ;sine of pi (3.1415...) is 0 isin4 = sin($M_PI_2 * 3) ;sine of 3/2pi (4.7123...) is -1 isin5 = sin($M_PI * 2) ;sine of 2pi (6.2831...) is 0 isin6 = sin($M_PI * 4) ;sine of 4pi is also 0 print isin1, isin2, isin3, isin4, isin5, isin6 endin instr 2 ;sin used in panning, after an example from Hans Mikelson aout vco2 0.8, 220 ; sawtooth kpan linseg p4, p3, p5 ;0 = left, 1 = right kpan = kpan*$M_PI_2 ;range 0-1 becomes 0-pi/2 kpanl = cos(kpan) kpanr = sin(kpan) outs aout*kpanl, aout*kpanr endin i 1 0 0 i 2 0 5 0 1 ;move left to right i 2 5 5 1 0 ;move right to left e manual_src~dfsg/examples/subtracts.csd0000644000000000000000000000150312262561504017267 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; -o -.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = p3 iatt = p4 idec = p5 isus = p3-iatt-idec ;calculate sustain time from subtracting attack and decay printf_i "sutain time= note duration - attack - decay --> %.1f-%.1f-%.1f = %.1f\n", 1, idur, iatt, idec, isus kenv expseg 0.01, iatt, 1, isus, 1, idec, 0.01 ;envelope asig poscil 1*kenv, 200, 1 outs asig, asig endin f 1 0 4096 10 1 ;sine wave ; attack decay i 1 0 3 .1 .2 i 1 4 3 .5 1.5 i 1 8 5 4 .5 e manual_src~dfsg/examples/fluidNote.csd0000644000000000000000000000165612262561504017217 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidNote.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "19Trumpet.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 56 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 7 asigl, asigr fluidOut giengine outs asigl*imvol, asigr*imvol endin i 1 0 2 60 100 ;play one note from score and... i 99 0 60 ;play virtual keyboard for 60 sec. e manual_src~dfsg/examples/gausstrig-2.csd0000644000000000000000000000127212262561504017427 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfrq0 oscil 8, 0.25, 1 ktrig metro 1 kfrq samphold kfrq0+8, ktrig seed 20120125 aimp gausstrig 0.5, kfrq, 0.5, 1 aenv filter2 aimp, 1, 1, 0.993, 0.993 anoi fractalnoise 0.2, 1.7 al = anoi*aenv ar delay al, 0.02 outs al, ar endin f1 0 8192 10 1 i1 0 16 e manual_src~dfsg/examples/sprintf.csd0000644000000000000000000000123212262561504016741 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sprintf.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;the file "impuls20.aiff" can be found in /manual/examples instr 1 ifn = 20 Sname sprintf "impuls%02d.aiff", ifn Smsg sprintf "The file name is: '%s'", Sname puts Smsg, 1 asig soundin Sname outs asig, asig endin i 1 0 1 e manual_src~dfsg/examples/lineto.csd0000644000000000000000000000144512262561504016554 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lineto.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kfreq randh 1000, 20, 2, 1, 2000 ;generates ten random number between 100 and 300 per second kpan randh .5, 1, 2, 1, .5 ;panning between 0 and 1 kp lineto kpan, .5 ;smoothing pan transition aout poscil .4, kfreq, giSine aL, aR pan2 aout, kp outs aL, aR endin i 1 0 10 e manual_src~dfsg/examples/hvs1.csd0000644000000000000000000000272612262561504016146 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O sr=44100 ksmps=100 nchnls=2 0dbfs = 1 ; Example by Gabriel Maldonado and Andres Cabrera ginumLinesX init 16 ginumParms init 3 giOutTab ftgen 5,0,8, -2, 0 giPosTab ftgen 6,0,32, -2, 3,2,1,0,4,5,6,7,8,9,10, 11, 15, 14, 13, 12 giSnapTab ftgen 8,0,64, -2, 1,1,1, 2,0,0, 3,2,0, 2,2,2, 5,2,1, 2,3,4, 6,1,7, 0,0,0, \ 1,3,5, 3,4,4, 1,5,8, 1,1,5, 4,3,2, 3,4,5, 7,6,5, 7,8,9 tb0_init giOutTab FLpanel "hsv1",500,100,10,10,0 gk1,ih1 FLslider "X", 0,1, 0,5, -1, 400,30, 50,20 FLpanel_end FLrun instr 1 ; kx, inumParms, inumPointsX, iOutTab, iPosTab, iSnapTab [, iConfigTab] hvs1 gk1, ginumParms, ginumLinesX, giOutTab, giPosTab, giSnapTab ;, iConfigTab k0 init 0 k1 init 1 k2 init 2 printk2 tb0(k0) printk2 tb0(k1), 10 printk2 tb0(k2), 20 aosc1 oscil tb0(k0)/20, tb0(k1)*100 + 200, 1 aosc2 oscil tb0(k1)/20, tb0(k2)*100 + 200, 1 aosc3 oscil tb0(k2)/20, tb0(k0)*100 + 200, 1 aosc4 oscil tb0(k1)/20, tb0(k0)*100 + 200, 1 aosc5 oscil tb0(k2)/20, tb0(k1)*100 + 200, 1 aosc6 oscil tb0(k0)/20, tb0(k2)*100 + 200, 1 outs aosc1 + aosc2 + aosc3, aosc4 + aosc5 + aosc6 endin f1 0 1024 10 1 f0 3600 i1 0 3600 manual_src~dfsg/examples/nsamp.csd0000644000000000000000000000210012262561504016365 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o nsamp.wav -W ;;; for file output any platform ; slightly adapted example from Jonathan Murphy Dec 2006 sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ; table number ilen = nsamp(ifn) prints "actual numbers of samples = %d\n", ilen itrns = 1 ; no transposition ilps = 0 ; loop starts at index 0 ilpe = ilen ; ends at value returned by nsamp above imode = 1 ; loops forward istrt = 0 ; commence playback at index 0 samples ; lphasor provides index into f1 alphs lphasor itrns, ilps, ilpe, imode, istrt atab tablei alphs, ifn outs atab, atab endin f 1 0 262144 1 "kickroll.wav" 0 4 1 ;stereo file in table, with lots of zeroes f 2 0 262144 1 "fox.wav" 0 4 1 ;mono file in table, with lots of zeroes i1 0 10 1 i1 + 10 2 e manual_src~dfsg/examples/19Trumpet.sf20000644000000000000000000063466612262561504017036 0ustar rootrootRIFF9sfbkLISTINFOifilisng E-mu 10K1INAM 056 Florestan Trumpet metallicICRDJul 20, 2000IENGIPRDSBAWE32ICOPPublic DomainICMThttp://go.to/florestanISFTSFEDT v1.28:SWAMI v0.9.2LIST5sdtasmpl5a^"kA >jEF?-OqYMC<9'CK?JPbjl]3/e?s2,C8b]dZ q Q (NA^-uF=o iWCj}DG$qi!ug5QYhDP@&zMvZZU[TR&P D3c0T  - E }YRtH33?n-} 9En v>2XW)*Pmy W fO AE$xkA7"~/ 2=k"N'&K `.'tkN{8Az +"Q()'%3zJCx:pi! :f o!'E*%_2,I-Nx9nc*;`  O#Q#'$<.O#T|QQDSsW3* o (%$!; 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DpV[gh1[(4M(B]B(K"mPw1\UD|uyw^*t-`jX<^mBT-"8yJ3.KVT?1"= lG!4V>tE7C}5Wq]c_Pt.O`nNY"O02xcJTo8I]=phS%_3$X~8MPq  kE8pQeB([r^.5) .jbvim1,k]lN==)E)!= [QM2I;HU'cM_u]+Y?7]0Q K!7tsnc&U>SUR5q(A#Ans 5C.jJgQ$0PAxyz}T(g>vtRrjM{nF"G;_$`c:FYOf[SBN>D6^TT7i_#/"QMJsff2 R\(WYv z_c~Tm2vB8hJhWwAxOU$OU(=+rr@1NU;V 4 I6N6=$u ~SE/@rE Z{d&8 x [x>3wC*Z|/~L`*oi`6GRw3U;|8+v.M5X?;" GAjhC2*vDMDADm+eT~ry;r[ TvS,\R5KM+rr:~:`V@8]th%y-^KW@# oP&zW7od]3`Kg|)gH-]Pf9d ROD] 0"~`p.yEI?MrEavh]^ZL\+HV@# c:[H0}_y.XL]S0G#La5v_ v29TE Z-6j7Pf!%[ Vt%N! -(EWOE'v,Mg qq\fZAAvDOf*HSBQ /F)Tzp?f%J(- cy(~^7}DjA ?gd/lIQ; 1>cRXS=J4)+%c"H>QuOh4X)oQE~t}*dL4XwT %> $j$01'+ $[k AS1~JP8I'9\?#H|Yxs0OivT2M{J0i/tT 4&8n\0}JXUyZI&Cu d mc4 3MM{P7=z8?f ebK>LOj:qMSY tO4 _@ I i`t!BCC[B%"Ucu1P4 P37m7 [) 5O%O$5< _+.7&8]4J>]nG]~+O$+L]sO] Q"|- #\`]PrXG#;^VXq\q^:fdd "L8xOL1 mG`&jH%w(!4]qJ Ue iU]j@Ml.B=Ckq *~KJqwV5v<1c#;92L(>>9n '` 6ONc4{4oc=HQX*Ep76&=N-;5@J|[YY_}htu~T>E1Vs+-edvl_g  7LPM3Y^#yqz5>+' G8V!d}Sm! @a/8u rEndm@'?p a|urf[hH 7 QYMG2mstw!3DL)a bv >vNnW;wF\nsA%\hP_N8]*rez:!f-= 8QN2D* io&+e:a'S (YVzEyiyd ;wR`Se@+h,z$_pM"K|A),  8*R0M1%y8b y6&+ unYOWElIpW@a kWjba6d's ^iVWVX!P5~7 RdN3gm?0"U 5uc[1E4nl6Z~A!A-^E~'%jfGu::Hi6.n -dN-F [{2NP79}GnkAY15+F ++'j zt0s[R'a}F}El6e8-\.YIG\n~ =*G&VA%v`j=PS8N/A896 Jg98885==0A;GI %+{KX1$]t  (FW.DQ'q c}-oMp;E9n*8'V7,vu"dM *8p p0I9Hx zrm<}rD"C\ *" OLwCVMuR_jy wab. 9 jV# jzVS@3}I}Qv[UI,?TeeO33ZBpKmEWonL(N <h|rAl40U;Ihpt,E)e;Re!m#75SP4"FK\\[`H6.S]zZ@q(!pl [\y"_"wB 6889a *U&"p~7x0\39o<~`'1OTQ:| &1 l_sC)F ;j_}}U^x3x\IA:$ LaeF)#*?Ks5lwy5\k S-KU>! pW\](O 26?<de_HwQrrbg$e~Y4g]RF;cXxh43noK'6z8G-h4jug ,YJEU>" 6j1gR'6FL60XpoSdLB1 [F8Vpy< DRxf(4 rKXU%: / RFK@;+?o G`(E VCc'sRx[ R0[yk ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vpow_i ifn1, ival, ielements, idstoffset endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e manual_src~dfsg/examples/hsboscil_midi.csd0000644000000000000000000000155712262561504020076 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out -odac -M0 ;;;realtime audio out and realtime MIDI in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; -o hsboscil_midi.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; synth waveform giwave ftgen 1, 0, 1024, 10, 1, 1, 1, 1 ; blending window giblend ftgen 2, 0, 1024, -19, 1, 0.5, 270, 0.5 instr 1 ibase = cpsoct(6) ioctcnt = 5 ; all octaves sound alike. itona octmidi ; velocity is mapped to brightness ibrite ampmidi 4 ; Map an exponential envelope for the amplitude. kenv expon .8, 1, .01 asig hsboscil kenv, itona, ibrite, ibase, giwave, giblend, ioctcnt outs asig, asig endin i 1 0 30 ; play for 30 seconds e manual_src~dfsg/examples/lpsholdp.csd0000644000000000000000000000171212262561504017104 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpsholdp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipan = p4 ktrig = 0 ; (no retriggering) ; kphase - looping pointer - is generated using a random spline kphase rspline 0,1,0.5,5 ; a loop sequence of midi note numbers and durations ; val dur val dur etc... knote lpsholdp kphase, 60, 1, 59, 0.1, 62, 1, 64, 1, 67, 0.1, 65, 1 asig gbuzz 0.2, cpsmidinn(knote), 30, 1, 0.5, 1 outs asig*ipan, asig*(1-ipan) endin ; cosine wave. f 1 0 16384 11 1 ; 2 layers of the loop are played each with a different pan position (p4) i 1 0 60 0.25 i 1 0 60 0.75 e manual_src~dfsg/examples/downsamp.csd0000644000000000000000000000133012262561504017103 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o downsamp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = cpspch(p4) ain diskin2 "beats.wav", 1 aenv follow ain, .001 ;take the amplitude every 1/1000th of a second alow tone aenv, 25 ;lowpass-filter (25 Hz) for a clean signal kenv downsamp alow asig pluck kenv, ifrq, 15, 0, 1 outs asig, asig endin i 1 0 2 9 i 1 + . 7 i 1 + . 5 e manual_src~dfsg/examples/gen52.csd0000644000000000000000000000415512262561504016203 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen52.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 garvb init 0 gaW init 0 gaX init 0 gaY init 0 itmp ftgen 1, 0, 64, -2, 2, 40, -1, -1, -1, 123, \ 1, 13.000, 0.05, 0.85, 20000.0, 0.0, 0.50, 2, \ 1, 2.000, 0.05, 0.85, 20000.0, 0.0, 0.25, 2, \ 1, 16.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 9.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 12.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 8.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2 itmp ftgen 2, 0, 262144, -2, 0 spat3dt 2, -0.2, 1, 0, 1, 1, 2, 0.005 itmp ftgen 3, 0, 262144, -52, 3, 2, 0, 4, 2, 1, 4, 2, 2, 4 instr 1 a1 vco2 1, 440, 10 kfrq port 100, 0.008, 20000 a1 butterlp a1, kfrq a2 linseg 0, 0.003, 1, 0.01, 0.7, 0.005, 0, 1, 0 a1 = a1 * a2 * 2 denorm a1 vincr garvb, a1 aw, ax, ay, az spat3di a1, p4, p5, p6, 1, 1, 2 vincr gaW, aw vincr gaX, ax vincr gaY, ay endin instr 2 denorm garvb ; skip as many samples as possible without truncating the IR arW, arX, arY ftconv garvb, 3, 2048, 2048, (65536 - 2048) aW = gaW + arW aX = gaX + arX aY = gaY + arY garvb = 0 gaW = 0 gaX = 0 gaY = 0 aWre, aWim hilbert aW aXre, aXim hilbert aX aYre, aYim hilbert aY aWXr = 0.0928*aXre + 0.4699*aWre aWXiYr = 0.2550*aXim - 0.1710*aWim + 0.3277*aYre aL = aWXr + aWXiYr aR = aWXr - aWXiYr outs aL, aR endin i 1 0 0.5 0.0 2.0 -0.8 i 1 1 0.5 1.4 1.4 -0.6 i 1 2 0.5 2.0 0.0 -0.4 i 1 3 0.5 1.4 -1.4 -0.2 i 1 4 0.5 0.0 -2.0 0.0 i 1 5 0.5 -1.4 -1.4 0.2 i 1 6 0.5 -2.0 0.0 0.4 i 1 7 0.5 -1.4 1.4 0.6 i 1 8 0.5 0.0 2.0 0.8 i 2 0 10 e manual_src~dfsg/examples/subinstr_named.csd0000644000000000000000000000232712262561504020277 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o subinstr_named.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument "basic_tone" - Creates a basic tone. instr basic_tone ; Print the value of p4, should be equal to ; Instrument #2's iamp field. print p4 ; Print the value of p5, should be equal to ; Instrument #2's ipitch field. print p5 ; Create a tone. asig oscils p4, p5, 0 out asig endin ; Instrument #1 - Demonstrates the subinstr opcode. instr 1 iamp = 20000 ipitch = 440 ; Use the "basic_tone" named instrument to create a ; basic sine-wave tone. ; Its p4 parameter will be set using the iamp variable. ; Its p5 parameter will be set using the ipitch variable. abasic subinstr "basic_tone", iamp, ipitch ; Output the basic tone that we have created. out abasic endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/jspline.csd0000644000000000000000000000125212262561504016722 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o jspline.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp init p4 kcpsmin init 2 kcpsmax init 20 ksp jspline kamp, kcpsmin, kcpsmax aout pluck 1, 200+ksp, 1000, 0, 1 aout dcblock aout ;remove DC outs aout, aout endin i 1 0 10 2 ;a bit jitter i 1 8 10 10 ;some more i 1 16 10 20 ;lots more e manual_src~dfsg/examples/pdhalfy.csd0000644000000000000000000000342512262561504016711 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pdhalfy.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 5 idur = p3 iamp = p4 ifreq = p5 iamtinit = p6 ; initial amount of phase distortion iatt = p7 ; attack time isuslvl = p8 ; sustain amplitude idistdec = p9 ; time for distortion amount to reach zero itable = p10 idec = idistdec - iatt irel = .05 isus = idur - (idistdec + irel) aenv linseg 0, iatt, 1.0, idec, isuslvl, isus, isuslvl, irel, 0, 0, 0 kamount linseg -iamtinit, idistdec, 0.0, idur-idistdec, 0.0 aosc phasor ifreq apd pdhalfy aosc, kamount aout tablei apd, itable, 1 outs aenv*aout*iamp, aenv*aout*iamp endin f1 0 16385 10 1 ; sine f3 0 16385 9 1 1 270 ; inverted cosine ; descending "just blues" scale ; pdhalfy with cosine table t 0 100 i5 0 .333 .6 512 1.0 .02 0.5 .12 3 i. + . . 448 < i. + . . 384 < i. + . . 358.4 < i. + . . 341.33 < i. + . . 298.67 < i. + 2 . 256 0.5 s ; pdhalfy with sine table t 0 100 i5 0 .333 .6 512 1.0 .001 0.1 .07 1 i. + . . 448 < i. + . . 384 < i. + . . 358.4 < i. + . . 341.33 < i. + . . 298.67 < i. + 2 . 256 0.5 e manual_src~dfsg/examples/readk3.csd0000644000000000000000000000255112262561504016432 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o readk3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes three control signals to a file kfreq randh 100, 1, 2, 1, 500 ;generates one random number between 400 and 600 per second kdb randh 12, 1, 2, 1, -12 ;amplitudes in dB between -24 and 0 kpan randh .5, 1, 2, 1, .5 ;panning between 0 and 1 dumpk3 kfreq, kdb, kpan, "dumpk3.txt", 8, 1 ;writes the control signals prints "WRITING:\n" printks "kfreq = %f, kdb = %f, kpan = %f\n", 1, kfreq, kdb, kpan ;prints them endin instr 2 ;reads the file written by instr 1 kf,kdb,kp readk3 "dumpk3.txt", 8, 1 prints "READING:\n" printks "kfreq = %f, kdb = %f, kpan = %f\n", 1, kf, kdb, kp ;prints again kdb lineto kdb, .1 ;smoothing amp transition kp lineto kp, .1 ;smoothing pan transition aout poscil ampdb(kdb), kf, giSine aL, aR pan2 aout, kp outs aL, aR endin i 1 0 5 i 2 5 5 e manual_src~dfsg/examples/puts.csd0000644000000000000000000000121412262561504016247 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o puts.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcount init 440 ktrig metro 10 kcount = kcount + ktrig String sprintfk "frequency in Hertz : %d \n", kcount puts String, kcount asig poscil .7, kcount, 1 outs asig, asig endin f1 0 16384 10 1 i 1 0 10 e manual_src~dfsg/examples/vco2ift.csd0000644000000000000000000000146312262561504016636 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vco2ift.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; user defined waveform -2: fixed table size (64), number of partials ; multiplier is 1.4 itmp ftgen 2, 0, 64, 5, 1, 2, 120, 60, 1, 1, 0.001, 1 ift vco2init -2, 2, 1.4, 4096, 4096, 2 instr 1 icps = p4 ifn vco2ift icps, -2, 0.5 ;with user defined waveform print ifn asig oscili 1, 220, ifn ; (-2), and sr/2 bandwidth outs asig, asig endin i 1 0 2 20 i 1 3 2 2000 i 1 6 2 20000 e manual_src~dfsg/examples/fof2-2.csd0000644000000000000000000002142112262561504016251 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o fof2.wav -W ;;; for file output any platform sr = 44100 ksmps = 128 nchnls = 2 ; Example by Chuckk Hubbard 2007 instr 1 ;table-lookup vocal synthesis iolaps = 120 ifna = 1 ;f1 - sine wave ifnb = 2 ;f2 - linear rise shape itotdur = p3 iamp = p4 * 0dbfs ifreq1 = p5 ;starting frequency ifreq2 = p6 ;ending frequency kamp linseg 0, .003, iamp, itotdur-.007, iamp, .003, 0, .001, 0 kfund expseg ifreq1, itotdur, ifreq2 koct init 0 kris init .003 kdur init .02 kdec init .007 kphs init 0 kgliss init 0 iforma = p7 ;starting spectrum iformb = p8 ;ending spectrum iform1a tab_i 0, iforma ;read values of 5 formants for 1st spectrum iform2a tab_i 1, iforma iform3a tab_i 2, iforma iform4a tab_i 3, iforma iform5a tab_i 4, iforma idb1a tab_i 5, iforma ;read decibel levels for same 5 formants idb2a tab_i 6, iforma idb3a tab_i 7, iforma idb4a tab_i 8, iforma idb5a tab_i 9, iforma iband1a tab_i 10, iforma ;read bandwidths for same 5 formants iband2a tab_i 11, iforma iband3a tab_i 12, iforma iband4a tab_i 13, iforma iband5a tab_i 14, iforma iamp1a = ampdb(idb1a) ;convert db to linear multipliers iamp2a = ampdb(idb2a) iamp3a = ampdb(idb3a) iamp4a = ampdb(idb4a) iamp5a = ampdb(idb5a) iform1b tab_i 0, iformb ;values of 5 formants for 2nd spectrum iform2b tab_i 1, iformb iform3b tab_i 2, iformb iform4b tab_i 3, iformb iform5b tab_i 4, iformb idb1b tab_i 5, iformb ;decibel levels for 2nd set of formants idb2b tab_i 6, iformb idb3b tab_i 7, iformb idb4b tab_i 8, iformb idb5b tab_i 9, iformb iband1b tab_i 10, iformb ;bandwidths for 2nd set of formants iband2b tab_i 11, iformb iband3b tab_i 12, iformb iband4b tab_i 13, iformb iband5b tab_i 14, iformb iamp1b = ampdb(idb1b) ;convert db to linear multipliers iamp2b = ampdb(idb2b) iamp3b = ampdb(idb3b) iamp4b = ampdb(idb4b) iamp5b = ampdb(idb5b) kform1 line iform1a, itotdur, iform1b ;transition between formants kform2 line iform2a, itotdur, iform2b kform3 line iform3a, itotdur, iform3b kform4 line iform4a, itotdur, iform4b kform5 line iform5a, itotdur, iform5b kband1 line iband1a, itotdur, iband1b ;transition of bandwidths kband2 line iband2a, itotdur, iband2b kband3 line iband3a, itotdur, iband3b kband4 line iband4a, itotdur, iband4b kband5 line iband5a, itotdur, iband5b kamp1 line iamp1a, itotdur, iamp1b ;transition of amplitudes of formants kamp2 line iamp2a, itotdur, iamp2b kamp3 line iamp3a, itotdur, iamp3b kamp4 line iamp4a, itotdur, iamp4b kamp5 line iamp5a, itotdur, iamp5b ;5 formants for each spectrum a1 fof2 kamp1, kfund, kform1, koct, kband1, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss a2 fof2 kamp2, kfund, kform2, koct, kband2, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss a3 fof2 kamp3, kfund, kform3, koct, kband3, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss a4 fof2 kamp4, kfund, kform4, koct, kband4, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss a5 fof2 kamp5, kfund, kform5, koct, kband5, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss aout = (a1+a2+a3+a4+a5) * kamp/5 ;sum and scale aenv linen 1, 0.05, p3, 0.05 ;to avoid clicking outs aout*aenv, aout*aenv endin f1 0 8192 10 1 f2 0 4096 7 0 4096 1 ;**************************************************************** ; tables of formant values adapted from MiscFormants.html ; 100's: soprano 200's: alto 300's: countertenor 400's: tenor 500's: bass ; -01: "a" sound -02: "e" sound -03: "i" sound -04: "o" sound -05: "u" sound ; p-5 through p-9: frequencies of 5 formants ; p-10 through p-14: decibel levels of 5 formants ; p-15 through p-19: bandwidths of 5 formants ; formant frequencies decibel levels bandwidths ;soprano f101 0 16 -2 800 1150 2900 3900 4950 0.001 -6 -32 -20 -50 80 90 120 130 140 f102 0 16 -2 350 2000 2800 3600 4950 0.001 -20 -15 -40 -56 60 100 120 150 200 f103 0 16 -2 270 2140 2950 3900 4950 0.001 -12 -26 -26 -44 60 90 100 120 120 f104 0 16 -2 450 800 2830 3800 4950 0.001 -11 -22 -22 -50 40 80 100 120 120 f105 0 16 -2 325 700 2700 3800 4950 0.001 -16 -35 -40 -60 50 60 170 180 200 ;alto f201 0 16 -2 800 1150 2800 3500 4950 0.001 -4 -20 -36 -60 80 90 120 130 140 f202 0 16 -2 400 1600 2700 3300 4950 0.001 -24 -30 -35 -60 60 80 120 150 200 f203 0 16 -2 350 1700 2700 3700 4950 0.001 -20 -30 -36 -60 50 100 120 150 200 f204 0 16 -2 450 800 2830 3500 4950 0.001 -9 -16 -28 -55 70 80 100 130 135 f205 0 16 -2 325 700 2530 3500 4950 0.001 -12 -30 -40 -64 50 60 170 180 200 ;countertenor f301 0 16 -2 660 1120 2750 3000 3350 0.001 -6 -23 -24 -38 80 90 120 130 140 f302 0 16 -2 440 1800 2700 3000 3300 0.001 -14 -18 -20 -20 70 80 100 120 120 f303 0 16 -2 270 1850 2900 3350 3590 0.001 -24 -24 -36 -36 40 90 100 120 120 f304 0 16 -2 430 820 2700 3000 3300 0.001 -10 -26 -22 -34 40 80 100 120 120 f305 0 16 -2 370 630 2750 3000 3400 0.001 -20 -23 -30 -34 40 60 100 120 120 ;tenor f401 0 16 -2 650 1080 2650 2900 3250 0.001 -6 -7 -8 -22 80 90 120 130 140 f402 0 16 -2 400 1700 2600 3200 3580 0.001 -14 -12 -14 -20 70 80 100 120 120 f403 0 16 -2 290 1870 2800 3250 3540 0.001 -15 -18 -20 -30 40 90 100 120 120 f404 0 16 -2 400 800 2600 2800 3000 0.001 -10 -12 -12 -26 70 80 100 130 135 f405 0 16 -2 350 600 2700 2900 3300 0.001 -20 -17 -14 -26 40 60 100 120 120 ;bass f501 0 16 -2 600 1040 2250 2450 2750 0.001 -7 -9 -9 -20 60 70 110 120 130 f502 0 16 -2 400 1620 2400 2800 3100 0.001 -12 -9 -12 -18 40 80 100 120 120 f503 0 16 -2 250 1750 2600 3050 3340 0.001 -30 -16 -22 -28 60 90 100 120 120 f504 0 16 -2 400 750 2400 2600 2900 0.001 -11 -21 -20 -40 40 80 100 120 120 f505 0 16 -2 350 600 2400 2675 2950 0.001 -20 -32 -28 -36 40 80 100 120 120 ;**************************************************************** ; start dur amp start freq end freq start formant end formant i1 0 1 .8 440 412.5 201 203 i1 + . .8 412.5 550 201 204 i1 + . .8 495 330 202 205 i1 + . .8 110 103.125 501 503 i1 + . .8 103.125 137.5 501 504 i1 + . .8 123.75 82.5 502 505 i1 7 . .4 440 412.5 201 203 i1 8 . .4 412.5 550 201 204 i1 9 . .4 495 330 202 205 i1 7 . .4 110 103.125 501 503 i1 8 . .4 103.125 137.5 501 504 i1 9 . .4 123.75 82.5 502 505 i1 + . .4 440 412.5 101 103 i1 + . .4 412.5 550 101 104 i1 + . .4 495 330 102 105 e manual_src~dfsg/examples/pindex-2.csd0000644000000000000000000000454312262561504016712 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pindex-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; This UDO returns a pfield value but returns 0 if it does not exist. opcode mypvalue, i, i index xin inum pcount if (index > inum) then iout = 0.0 else iout pindex index endif xout iout endop ; Envelope UDO that reads parameters from a flexible number of pfields ; Syntax: kenv flexlinseg ipstart ; ipstart is the first pfield of the envelope ; parameters. Reads remaining pfields (up to 21 of them). ; kenv is the output envelope. opcode flexlinseg, k, i ipstart xin iep1 mypvalue ipstart iep2 mypvalue ipstart + 1 iep3 mypvalue ipstart + 2 iep4 mypvalue ipstart + 3 iep5 mypvalue ipstart + 4 iep6 mypvalue ipstart + 5 iep7 mypvalue ipstart + 6 iep8 mypvalue ipstart + 7 iep9 mypvalue ipstart + 8 iepa mypvalue ipstart + 9 iepb mypvalue ipstart + 10 iepc mypvalue ipstart + 11 iepd mypvalue ipstart + 12 iepe mypvalue ipstart + 13 iepf mypvalue ipstart + 14 iepg mypvalue ipstart + 15 ieph mypvalue ipstart + 16 iepi mypvalue ipstart + 17 iepj mypvalue ipstart + 18 iepk mypvalue ipstart + 19 iepl mypvalue ipstart + 20 kenv linseg iep1, iep2, iep3, iep4, iep5, iep6, iep7, iep8, \ iep9, iepa, iepb, iepc, iepd, iepe, iepf, iepg, \ ieph, iepi, iepj, iepk, iepl xout kenv endop instr 1 ; This instrument only requires 3 pfields but can accept up to 24. ; (You will still get warnings about more than 3). kenv flexlinseg 4 ; envelope params start at p4 aout oscili kenv*.5, 256, 1 outs aout, aout endin f1 0 8192 10 1 i1 0 2 0.0 1.0 1.0 1.0 0.0 i1 2 2 0.0 0.1 1.0 1.0 1.0 0.1 0.0 i1 4 2 0.0 0.5 0.0 ; one problem is that "missing" pfields carry i1 6 2 0.0 0.5 0.0 ! ; now we can fix this problem with ! i1 8 10 0.0 3.0 1.0 0.3 0.1 0.3 1.0 0.3 0.1 0.3 1.0 0.3 0.1 0.8 0.9 5.0 0.0 e manual_src~dfsg/examples/gogobel.csd0000644000000000000000000000132012262561504016670 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gogobel.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 200 ihrd = 0.5 ipos = p4 kvibf = 6.0 kvamp = 0.3 asig gogobel .9, kfreq, ihrd, ipos, 1, 6.0, 0.3, 2 outs asig, asig endin ;audio file f 1 0 256 1 "marmstk1.wav" 0 0 0 ;sine wave for the vibrato f 2 0 128 10 1 i 1 0.5 0.5 0.01 i 1 + 0.5 0.561 i 1 + 0.5 0.9 e manual_src~dfsg/examples/weibull.csd0000644000000000000000000000172112262561504016722 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o weibull.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values ktri weibull 100, 1 printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 ktri weibull 100, 1 printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin instr 3 ; every run time different values seed 0 ktri weibull 100, 10 ; closer to ksigma.. printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 i 3 6 2 e manual_src~dfsg/examples/spat3d_UHJ.csd0000644000000000000000000000377512262561504017176 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o spat3d_UHJ.wav -W ;;; for file output any platform /* Written by Istvan Varga */ sr = 48000 kr = 750 ksmps = 64 nchnls = 2 itmp ftgen 1, 0, 64, -2, \ /* depth1, depth2, max delay, IR length, idist, seed */ \ 3, 48, -1, 0.01, 0.25, 123, \ 1, 21.982, 0.05, 0.87, 4000.0, 0.6, 0.7, 2, /* ceil */ \ 1, 1.753, 0.05, 0.87, 3500.0, 0.5, 0.7, 2, /* floor */ \ 1, 15.220, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* front */ \ 1, 9.317, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* back */ \ 1, 17.545, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* right */ \ 1, 12.156, 0.05, 0.87, 5000.0, 0.8, 0.7, 2 /* left */ instr 1 p3 = p3 + 1.0 kazim line 0.0, 4.0, 360.0 ; azimuth kelev line 40, p3 - 1.0, -20 ; elevation kdist = 2.0 ; distance ; convert coordinates kX = kdist * cos(kelev * 0.01745329) * sin(kazim * 0.01745329) kY = kdist * cos(kelev * 0.01745329) * cos(kazim * 0.01745329) kZ = kdist * sin(kelev * 0.01745329) ; source signal a1 phasor 160.0 a2 delay1 a1 a1 = a1 - a2 kffrq1 port 200.0, 0.8, 12000.0 affrq upsamp kffrq1 affrq pareq affrq, 5.0, 0.0, 1.0, 2 kffrq downsamp affrq aenv4 phasor 3.0 aenv4 limit 2.0 - aenv4 * 8.0, 0.0, 1.0 a1 butterbp a1 * aenv4, kffrq, 160.0 aenv linseg 1.0, p3 - 1.0, 1.0, 0.04, 0.0, 1.0, 0.0 a_ = 4000000 * a1 * aenv + 0.00000001 ; spatialize a_W, a_X, a_Y, a_Z spat3d a_, kX, kY, kZ, 1.0, 1, 2, 2.0, 2 ; convert to UHJ format (stereo) aWre, aWim hilbert a_W aXre, aXim hilbert a_X aYre, aYim hilbert a_Y aWXre = 0.0928*aXre + 0.4699*aWre aWXim = 0.2550*aXim - 0.1710*aWim aL = aWXre + aWXim + 0.3277*aYre aR = aWXre - aWXim - 0.3277*aYre outs aL, aR endin /* Written by Istvan Varga */ t 0 60 i 1 0.0 8.0 e manual_src~dfsg/examples/marimba.csd0000644000000000000000000000217712262561504016675 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o marimba.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 ksmps = 128 nchnls = 2 ; Instrument #1. instr 1 ifreq = cpspch(p4) ihrd = 0.1 ipos = 0.561 imp = 1 kvibf = 6.0 kvamp = 0.05 ivibfn = 2 idec = 0.6 a1 marimba 20000, ifreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec, 20, 10 outs a1, a1 endin ; Table #1, the "marmstk1.wav" audio file. f 1 0 256 1 "marmstk1.wav" 0 0 0 ; Table #2, a sine wave for the vibrato. f 2 0 128 10 1 ; Play Instrument #1 for one second. i 1 0 1 8.09 i 1 + 0.5 8.00 i 1 + 0.5 7.00 i 1 + 0.25 8.02 i 1 + 0.25 8.01 i 1 + 0.25 7.09 i 1 + 0.25 8.02 i 1 + 0.25 8.01 i 1 + 0.25 7.09 i 1 + 0.3333 8.09 i 1 + 0.3333 8.02 i 1 + 0.3334 8.01 i 1 + 0.25 8.00 i 1 + 0.3333 8.09 i 1 + 0.3333 8.02 i 1 + 0.25 8.01 i 1 + 0.3333 7.00 i 1 + 0.3334 6.00 e manual_src~dfsg/examples/cngoto.csd0000644000000000000000000000162112262561504016547 0ustar rootroot ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; -o cngoto.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Change kval linearly from 0 to 2 over ; the period set by the third p-field. kval line 0, p3, 2 ; If kval *is not* greater than or equal to 1 then play ; the high note. Otherwise, play the low note. cngoto (kval >= 1), highnote kgoto lownote highnote: kfreq = 880 goto playit lownote: kfreq = 440 goto playit playit: ; Print the values of kval and kfreq. printks "kval = %f, kfreq = %f\\n", 1, kval, kfreq a1 oscil 10000, kfreq, 1 out a1 endin ; Table: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/test2.png0000644000000000000000000000725612262561504016344 0ustar rootrootPNG  IHDR{CsRGBbKGD pHYs  tIME $x?tEXtCommentCreated with GIMPW IDATxng;'0v+Qcw RofD6%-z fBu`}?_7`YnVݬߵ_O_vvmW^)q?ouvj2ڭ:'_ՍaZQ=uխڪ,GuonԿ?ګ{SlFsU`N<_՟S뺬9Tu^`N/~0NEԩS븞ՋzcF:SL:0. _ ?0`h 7Co Z0`h 7Co Z0`h 7Co Z0`h 7Co Z0`h 7Co Z0`h 7Co Z0`h 7h CoZ@?P7oNݫ:_HfwCUuk]:u4X~wv}]{_ ͺS:;Xxﮏ}zXmݯuK|I=Wڙ],?Q}n`߼:^ <X~WuYNտOFػY_]U}\uԏu<XPYu\'oqTmZ^O'>yU/Uo.S]ϨfG'^֛:C]NuYǺVL慟ձW`VϻcW@ 7oP@ 7oR@P7_ @1c[@ 7o_ oW@P @1c?P @ @P7_ o@1c?P @1cW c@P7?P @1c?P P?_1W c@?P @1cP?_1W ?P0`@?P @1W c@P ?P0`@ ~P?_1W oNݫ:P LA߬`OVm׭u밞֑c`ߩu~=:T ,7Nݯ֣Ԯc`>yXaԷuR ,_>='^}SkG1.PG>Ug{}7c`_e:U&0?g꺮~cY:D1~zU/zSb}^ԫ:m]ԧV ,_>=Y^R1.?z>XWguXx]? ?P0`@ ?P0`P?_W0@  ?P0`@?UW0@ `P?0?`P@ ^{ P0`@W0@ `P 0?`P@ 0W@ `P?0?`P@ 0?`PP ?`_W@A 0?`P@ ?`_W@0`?P@  0?`PP ?`_W@ 0`?P@  ?`_W@0`ک{W _fw0Cԭڮ[a=# SګzTGuXvo֝_A=G}=] w}:o~ݭX^|{XOQ=n׎`_]j}s0慟ձng WzWguۓZ?g꺮~c Y:D~zU/zSb}^ԫ:m]ԧV,_>=Y^R.?z>XWguXx] ?`?P@ 0`?P@&L80q`?P@&L80q`?P@&L80q`?P@&L80q`?P@&L80q`?P@&L80q`?P@fRZ`|S{=us`X`O={?ڭS7kKq\wnݪ3W;Umoa X\Du>?xR:SEڬ纬u^U}/ձ,~>ן^?`ue~Mׅ,x5IENDB`manual_src~dfsg/examples/midichn_advanced.csd0000644000000000000000000000245612262561504020525 0ustar rootroot ; Select audio/midi flags here according to platform -odac -F midichn_advanced.mid ;;;realtime audio out with MIDI file input ; For Non-realtime ouput leave only the line below: ; -o midichn_advanced.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 massign 1, 1 ; all channels use instr 1 massign 2, 1 massign 3, 1 massign 4, 1 massign 5, 1 massign 6, 1 massign 7, 1 massign 8, 1 massign 9, 1 massign 10, 1 massign 11, 1 massign 12, 1 massign 13, 1 massign 14, 1 massign 15, 1 massign 16, 1 gicnt = 0 ; note counter instr 1 gicnt = gicnt + 1 ; update note counter kcnt init gicnt ; copy to local variable ichn midichn ; get channel number istime times ; note-on time if (ichn > 0.5) goto l2 ; MIDI note printks "note %.0f (time = %.2f) was activated from the score\\n", \ 3600, kcnt, istime goto l1 l2: printks "note %.0f (time = %.2f) was activated from channel %.0f\\n", \ 3600, kcnt, istime, ichn l1: icps cpsmidi ; convert midi note to pitch kenv madsr 0.1, 0, 0.8, 0.9 asig pluck kenv, icps, icps, 1, 1 outs asig, asig endin t 0 60 ;beats per minute f 0 8 ;stay active for 8 seconds f 1 0 4096 10 1 ;sine e manual_src~dfsg/examples/reson.csd0000644000000000000000000000133512262561504016406 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o reson.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 asaw vco2 .2, 220 ;sawtooth kcf line 220, p3, 1760 ;vary cut-off frequency from 220 to 1280 Hz kbw = p4 ;vary bandwidth of filter too ares reson asaw, kcf, kbw asig balance ares, asaw outs asig, asig endin i 1 0 4 10 ;bandwidth of filter = 10 Hz i 1 + 4 50 ;50 Hz and i 1 + 4 200 ;200 Hz e manual_src~dfsg/examples/hrtfmove2.csd0000644000000000000000000000165112262561504017175 0ustar rootroot ; Select flags here ; realtime audio out -o dac ; For Non-realtime ouput leave only the line below: ; -o hrtf.wav sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 gasrc init 0 instr 1 ;a plucked string kamp = p4 kcps = cpspch(p5) icps = cpspch(p5) a1 pluck kamp, kcps, icps, 0, 1 gasrc = a1 endin instr 10 ;uses output from instr1 as source kaz linseg 0, p3, 720 ;2 full rotations aleft,aright hrtfmove2 gasrc, kaz,0, "hrtf-44100-left.dat","hrtf-44100-right.dat" outs aleft, aright endin ; Play Instrument 1: a simple arpeggio i1 0 .2 15000 8.00 i1 + .2 15000 8.04 i1 + .2 15000 8.07 i1 + .2 15000 8.11 i1 + .2 15000 9.02 i1 + 1.5 15000 8.11 i1 + 1.5 15000 8.07 i1 + 1.5 15000 8.04 i1 + 1.5 15000 8.00 i1 + 1.5 15000 7.09 i1 + 1.5 15000 8.00 ; Play Instrument 10 for 10 seconds. i10 0 10 manual_src~dfsg/examples/hrtfer.csd0000644000000000000000000000153312262561504016552 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o hrtfer.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 kaz linseg 0, p3, -360 ; move the sound in circle kel linseg -40, p3, 45 ; around the listener, changing ; elevation as its turning asrc soundin "beats.wav" aleft,aright hrtfer asrc, kaz, kel, "HRTFcompact" aleftscale = aleft * 200 arightscale = aright * 200 outs aleftscale, arightscale endin i 1 0 2 e manual_src~dfsg/examples/fluidCCk.csd0000644000000000000000000000175512262561504016752 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidCCk.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "22Bassoon.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 70 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 kpan line 0, p3, 127 ;panning from left to right fluidCCk giengine, 1, 10, kpan ;CC 10 = pan fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 7 asigl, asigr fluidOut giengine outs asigl*imvol, asigr*imvol endin i 1 0 4 48 100 i 1 4 2 50 120 i 1 6 1 53 80 i 1 7 1 45 70 i 1 8 1.5 48 80 i 99 0 10 ;keep instr 99 active e manual_src~dfsg/examples/gendyx.csd0000644000000000000000000000101012262561504016544 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aout gendyx 0.7, 1, 1, 1, 1, 20, 1000, 0.5, 0.5, 4, 0.13 outs aout, aout endin i1 0 10 e manual_src~dfsg/examples/readk.csd0000644000000000000000000000133012262561504016341 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o readk.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 0dbfs = 1 ; By Andres Cabrera 2008 instr 1 ; Read a number from the file every 0.5 seconds kfibo readk "fibonacci.txt", 7, 0.5 kpitchclass = 8 + ((kfibo % 12)/100) printk2 kpitchclass kcps = cpspch( kpitchclass ) printk2 kcps a1 oscil 0.5, kcps, 1 out a1 endin f 1 0 1024 10 1 i 1 0 10 e manual_src~dfsg/examples/outkpb.csd0000644000000000000000000000120112262561504016554 0ustar rootroot ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out and midi in and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outkpb.wav -W ;;; for file output any platform sr = 44100 ksmps = 10 nchnls = 2 instr 1 ikey notnum ivel veloc kpch linseg 100, 1, 0 ;vary in 1 second kpb = int(kpch) ;whole numbers only outkpb 1, kpb, 0, 127 ;(= pitchbend) midion 1, ikey, ivel ;of external synth endin f0 30 e manual_src~dfsg/examples/maxalloc.csd0000644000000000000000000000124112262561504017054 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o maxalloc.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 maxalloc 1, 3 ; Limit to three instances. instr 1 asig oscil .3, p4, 1 outs asig, asig endin ; sine f 1 0 32768 10 1 i 1 0 5 220 ;1 i 1 1 4 440 ;2 i 1 2 3 880 ;3, limit is reached i 1 3 2 1320 ;is not played i 1 4 1 1760 ;is not played e manual_src~dfsg/examples/log.csd0000644000000000000000000000101612262561504016035 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o log.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = log(8) print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/FLbutBank.csd0000644000000000000000000000160012262561504017063 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLbutton.wav -W ;;; for file output any platform sr = 44100 nchnls = 1 FLpanel "Button Bank", 520, 140, 100, 100 ;itype = 2 ;Light Buttons itype = 22 ;Plastic Light Buttons inumx = 10 inumy = 4 iwidth = 500 iheight = 120 ix = 10 iy = 10 iopcode = 0 istarttim = 0 idur = 1 gkbutton, ihbb FLbutBank itype, inumx, inumy, iwidth, iheight, ix, iy, iopcode, 1, istarttim, idur FLpanelEnd FLrun instr 1 ibutton = i(gkbutton) prints "Button %i pushed!\\n", ibutton endin ; Real-time performance for 1 hour. f 0 3600 e manual_src~dfsg/examples/JackoInfo.csd0000644000000000000000000000036112262561504017121 0ustar rootroot -n sr = 48000 ksmps = 128 nchnls = 2 0dbfs = 1 instr 1 JackoInit "default", "csound" JackoInfo endin i 1 0 0 e manual_src~dfsg/examples/ATSadd.csd0000644000000000000000000000113112262561504016352 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc for RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSadd.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 2 asig ATSadd ktime, 1, "beats.ats", 1, 20, 0, 2 outs asig*3, asig*3 ;amplify endin ;sine wave. f 1 0 16384 10 1 i 1 0 2 e manual_src~dfsg/examples/gen24.csd0000644000000000000000000000150612262561504016177 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen24.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ;choose between tables kcps init 1/p3 ;create index over duration of note. kndx phasor kcps ixmode = 1 ;normalize to 0-1 kval table kndx, ifn, ixmode asig poscil .7, 440 + kval, 1 ;add to frequency outs asig, asig endin f 1 0 16384 10 1 ;sine wave f 10 0 16384 -24 1 0 400;scale sine wave from table 1 from 0 to 400 f 11 0 16384 -24 1 0 50 ;and from 0 to 50 i 1 0 3 10 i 1 4 3 11 e manual_src~dfsg/examples/toot8.csd0000644000000000000000000000153612262561504016340 0ustar rootroot -odac instr 8 idur = p3 iamp = p4 iskiptime = p5 iattack = p6 irelease = p7 irvbtime = p8 irvbgain = p9 kamp linen iamp, iattack, idur, irelease asig soundin "fox.wav", iskiptime arampsig = kamp * asig aeffect reverb asig, irvbtime arvbretrn = aeffect * irvbgain out arampsig + arvbretrn endin ;ins strt dur amp skip atk rel rvbt rvbgain i8 0 2.28 .3 0 .03 .1 1.5 .3 i8 4 1.6 .3 1.6 .1 .1 1.1 .4 i8 5.5 2.28 .3 0 .5 .1 2.1 .2 i8 6.5 2.28 .4 0 .01 .1 1.1 .1 i8 8 2.28 .5 0.1 .01 .1 0.1 .1 e manual_src~dfsg/examples/vbap8.csd0000644000000000000000000000215112262561504016275 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in ;-odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -o vbap8.wav -W ;;; for file output any platform sr = 44100 kr = 441 ksmps = 100 nchnls = 4 vbaplsinit 2, 8, 0, 45, 90, 135, 200, 245, 290, 315 instr 1 asig oscil 20000, 440, 1 a1,a2,a3,a4,a5,a6,a7,a8 vbap8 asig, p4, 0, 20 ;p4 = azimuth ;render twice with alternate outq statements ; to obtain two 4 channel .wav files: outq a1,a2,a3,a4 ; outq a5,a6,a7,a8 ; or use an 8-channel output for realtime output (set nchnls to 8): ; outo a1,a2,a3,a4,a5,a6,a7,a8 endin f 1 0 8192 10 1 ; Play Instrument #1 for one second. ; azimuth i 1 0 1 20 i 1 + . 40 i 1 + . 60 i 1 + . 80 i 1 + . 100 i 1 + . 120 i 1 + . 140 i 1 + . 160 e manual_src~dfsg/examples/STKBrass.csd0000644000000000000000000000131112262561504016706 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKBrass.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;lip tension kv3 line p7, p3, p8 ;speed of low-frequency oscillator asig STKBrass cpspch(ifrq), 1, 2, kv1, 4, 100, 11, kv3, 1, 10, 128, 40 asig = asig *3 ;amplify outs asig, asig endin i 1 0 2 8.05 100 120 50 0 i 1 + 3 9.00 80 82 10 0 e manual_src~dfsg/examples/expcurve.csd0000644000000000000000000000074112262561504017121 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in Silent -odac -iadc -d ;;;realtime output sr = 48000 ksmps = 1000 nchnls = 2 instr 1 ; logcurve test kmod phasor 1/p3 kout expcurve kmod, p4 printks "mod = %f out= %f\\n", 0.5, kmod, kout endin /*--- ---*/ i1 0 5 2 i1 5 5 5 i1 10 5 30 i1 15 5 0.5 e manual_src~dfsg/examples/poscil3-file.csd0000644000000000000000000000172512262561504017554 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o poscil3-file.wav -W ;;; for file output any platform ; Example written by Joachim Heintz 07/2008 sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; non-normalized function table with a sample 1 giFile ftgen 1, 0, 0, -1, "fox.wav", 0, 0, 0 ; Instrument #1 - uses poscil3 for playing samples from a function table instr 1 kamp = p4 kspeed = p5 ifn = 1 iskip = p6 kcps = kspeed / (ftlen(ifn) / ftsr(ifn)); frequency of the oscillator iphs = iskip / (ftlen(ifn) / ftsr(ifn)); calculates skiptime to phase values (0-1) a1 poscil3 kamp, kcps, ifn, iphs out a1 endin i1 0 2.756 1 1 0 i1 3 2.756 1 -1 0 i1 6 1.378 1 .5 2.067 e manual_src~dfsg/examples/cpsmidinn2.csd0000644000000000000000000000135112262561504017324 0ustar rootroot ; Select audio/midi flags here according to platform ;;;RT audio out, midi in, note=p4 and velocity=p5 -odac -+rtmidi=virtual -M0d --midi-key=4 --midi-velocity-amp=5 ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpsmidinn.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 massign 0, 1 ;assign all midi to instr. 1 instr 1 ;play virtual keyboard inote = p4 icps = cpsmidinn(inote) asig oscil 0.6, icps, 1 print icps outs asig, asig endin f0 20 ;sine wave. f 1 0 16384 10 1 ;play note from score too i1 0 1 60 e manual_src~dfsg/examples/twopeaks.aiff0000644000000000000000000000107612262561504017253 0ustar rootrootFORM6AIFF(c) COMM@ DSSND 'Xbuu{ I 5Y0 !(#K%}'* ,d.1?36K8;>1@CFrIEL OQTWZ]`cgjm7pSssvy||yvsspSm7jgc`]ZWTQOL IEFrC@>1;86K31?.,d* '%}#K!( 0Y5 I {uubX' c +^~mBV`%nܵڃDӜ5Aɵ|ϿX ()% ɏlI&&Ilɕ %)( රX|ɵA5ӜDڃܵn%`VBm~^+ cmanual_src~dfsg/examples/OSCmidisend.csd0000644000000000000000000000122012262561504017412 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O sr = 44100 ksmps = 128 nchnls = 1 ; Example by David Akbari 2007 ; Modified by Jonathan Murphy ; Use this file to generate OSC events for OSCmidircv.csd #define IPADDRESS # "localhost" # #define PORT # 47120 # turnon 1000 instr 1000 kst, kch, kd1, kd2 midiin OSCsend kst+kch+kd1+kd2, $IPADDRESS, $PORT, "/midi", "iiii", kst, kch, kd1, kd2 endin f 0 3600 ;Dummy f-table e manual_src~dfsg/examples/i_statement.csd0000644000000000000000000000400712262561504017573 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o i_statement.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 10 icps init cpspch(p4) ; Get target pitch from score event iportime init abs(p3)/7 ; Portamento time dep on note length iamp0 init p5 ; Set default amps iamp1 init p5 iamp2 init p5 itie tival ; Check if this note is tied, if itie == 1 igoto nofadein ; if not fade in iamp0 init 0 nofadein: if p3 < 0 igoto nofadeout ; Check if this note is held, if not fade out iamp2 init 0 nofadeout: ; Now do amp from the set values: kamp linseg iamp0, .03, iamp1, abs(p3)-.03, iamp2 ; Skip rest of initialization on tied note: tigoto tieskip kcps init icps ; Init pitch for untied note kcps port icps, iportime, icps ; Drift towards target pitch kpw oscil .4, rnd(1), 1, rnd(.7) ; A simple triangle-saw oscil ar vco kamp, kcps, 3, kpw+.5, 1, 1/icps ; (Used in testing - one may set ipch to cpspch(p4+2) ; and view output spectrum) ; ar oscil kamp, kcps, 1 outs ar, ar tieskip: ; Skip some initialization on tied note endin f1 0 8192 10 1 ; Sine i10.1 0 -1 7.00 .15 i10.2 0 -1 7.04 i10.3 0 -1 7.07 i10.1 1 -1 8.00 i10.2 1 -1 8.04 i10.3 1 -1 8.07 i10.1 2 1 7.11 i10.2 2 1 8.04 i10.3 2 1 8.07 e manual_src~dfsg/examples/fractalnoise.csd0000644000000000000000000000106712262561504017734 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kbeta linseg 0, p3/4, 2, p3/4, 0, p3*0.1, 2, p3*0.15, 0 seed 20120124 aout fractalnoise 0.05, kbeta outs aout, aout endin i1 0 10 e manual_src~dfsg/examples/loopsegp.csd0000644000000000000000000000243112262561504017106 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loopsegp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktrig metro 22/8 ; triggers used to generate new direction values kdir trandom ktrig,-2.99,2.99 kdir = 0.5*int(kdir) ; kdir will either -1, -0.5, 0, 0.5 or 1 ; kphase - looping pointer kphase phasor kdir ; a loop sequence of midi note numbers and durations knote loopsegp kphase, 40,1,40,0, 43,1,43,0, 49,2,48,0, \ 47,1,47,0, 46,1,46,0, 46,1,47,0, 49,1,49,0, 43,1,43,0, 46,1,46,0, 40,1,39,0 kmul rspline 0.1,0.8,0.5,5 ; modulation of buzz tone asig gbuzz 0.2, cpsmidinn(knote), 30, 3, kmul, 1 ; buzz tone outs asig, asig schedkwhen ktrig,0,0,2,0,0.1 ; play metronome endin instr 2 ; metronome acps expon 180+rnd(40),p3,50 aamp expon 0.05+rnd(0.05),p3,0.001 asig poscil aamp-0.001,acps,2 outs asig,asig endin ; cosine wave. f 1 0 16384 11 1 ; sine wave. f 2 0 16384 10 1 i 1 0 360 0.25 e manual_src~dfsg/examples/strcmp.csd0000644000000000000000000000213212262561504016564 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o strcmp.wav -W ;;; for file output any platform ;modified example from Joachim Heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 opcode Triad, iii, S ;define UDO Sname xin iMaj strcmp "maj", Sname iMin strcmp "min", Sname iPrim = 8.00 ;notes in pitch notattion iQuint = 8.05 if iMaj == 0 then iTer = 8.03 elseif iMin == 0 then iTer = 8.02 endif xout iPrim, iTer, iQuint endop instr 1 Sname strget p4 ia, ib, ic Triad Sname ;apply UDO print ia, ib, ic asig1 pluck 0.7, cpspch(ia), 220, 0, 1 asig2 pluck 0.7, cpspch(ib), 220, 0, 1 asig3 pluck 0.7, cpspch(ic), 220, 0, 1 asig = (asig1+asig2+asig3)*.5 outs asig, asig endin i1 0 3 "maj" i1 4 3 "min" manual_src~dfsg/examples/dumpk-2.csd0000644000000000000000000000156612262561504016545 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dumpk-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes a control signal to a file kfreq randh 100, 1, 2, 1, 500 ;generates one random number between 400 and 600 per second dumpk kfreq, "dumpk.txt", 8, 1 ;writes the control signal printk 1, kfreq ;prints it endin instr 2 ;reads the file written by instr 1 kfreq readk "dumpk.txt", 8, 1 printk 1, kfreq ;prints it aout poscil .2, kfreq, giSine outs aout, aout endin i 1 0 5 i 2 5 5 e manual_src~dfsg/examples/sandpaper.csd0000644000000000000000000000125612262561504017237 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sandpaper.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idmp = p4 a1 line 2, p3, 2 ;preset amplitude increase a2 sandpaper 1, 0.01, 128, idmp ;sandpaper needs a little amp help at these settings asig product a1, a2 ;increase amplitude outs asig, asig endin i1 0 1 0.5 i1 + 1 0.95 e manual_src~dfsg/examples/vexpseg.csd0000644000000000000000000000135312262561504016741 0ustar rootroot -odac -B441 -b441 sr=44100 ksmps=10 nchnls=2 gilen init 32 gitable1 ftgen 0, 0, gilen, 10, 1 gitable2 ftgen 0, 0, gilen, 10, 1 gitable3 ftgen 0, 0, gilen, -7, 30, gilen, 35 gitable4 ftgen 0, 0, gilen, -7, 400, gilen, 450 gitable5 ftgen 0, 0, gilen, -7, 5000, gilen, 5500 instr 1 vcopy gitable2, gitable1, gilen turnoff endin instr 2 vexpseg gitable2, 16, gitable3, 2, gitable4, 2, gitable5 endin instr 3 kcount init 0 if kcount < 16 then kval table kcount, gitable2 printk 0,kval kcount = kcount +1 else turnoff endif endin i1 0 1 s i2 0 10 i3 0 1 i3 1 1 i3 1.5 1 i3 2 1 i3 2.5 1 i3 3 1 i3 3.5 1 i3 4 1 i3 4.5 1 manual_src~dfsg/examples/connect.csd0000644000000000000000000000607212262561504016714 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o connect.wav -W ;;; for file output any platform /* Written by Michael Gogins */ ; Initialize the global variables. sr = 44100 ksmps = 32 nchnls = 2 ; Connect up the instruments to create a signal flow graph. connect "SimpleSine", "leftout", "Reverberator", "leftin" connect "SimpleSine", "rightout", "Reverberator", "rightin" connect "Moogy", "leftout", "Reverberator", "leftin" connect "Moogy", "rightout", "Reverberator", "rightin" connect "Reverberator", "leftout", "Compressor", "leftin" connect "Reverberator", "rightout", "Compressor", "rightin" connect "Compressor", "leftout", "Soundfile", "leftin" connect "Compressor", "rightout", "Soundfile", "rightin" ; Turn on the "effect" units in the signal flow graph. alwayson "Reverberator", 0.91, 12000 alwayson "Compressor" alwayson "Soundfile" instr SimpleSine ihz = cpsmidinn(p4) iamplitude = ampdb(p5) print ihz, iamplitude ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. isine ftgenonce 0, 0, 4096, 10, 1 a1 oscili iamplitude, ihz, isine aenv madsr 0.05, 0.1, 0.5, 0.2 asignal = a1 * aenv ; Stereo audio outlet to be routed in the orchestra header. outleta "leftout", asignal * 0.25 outleta "rightout", asignal * 0.75 endin instr Moogy ihz = cpsmidinn(p4) iamplitude = ampdb(p5) ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. isine ftgenonce 0, 0, 4096, 10, 1 asignal vco iamplitude, ihz, 1, 0.5, isine kfco line 200, p3, 2000 krez init 0.9 asignal moogvcf asignal, kfco, krez, 100000 ; Stereo audio outlet to be routed in the orchestra header. outleta "leftout", asignal * 0.75 outleta "rightout", asignal * 0.25 endin instr Reverberator ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" idelay = p4 icutoff = p5 aleftout, arightout reverbsc aleftin, arightin, idelay, icutoff ; Stereo output. outleta "leftout", aleftout outleta "rightout", arightout endin instr Compressor ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" kthreshold = 25000 icomp1 = 0.5 icomp2 = 0.763 irtime = 0.1 iftime = 0.1 aleftout dam aleftin, kthreshold, icomp1, icomp2, irtime, iftime arightout dam arightin, kthreshold, icomp1, icomp2, irtime, iftime ; Stereo output. outleta "leftout", aleftout outleta "rightout", arightout endin instr Soundfile ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" outs aleftin, arightin endin ; Not necessary to activate "effects" or create f-tables in the score! ; Overlapping notes to create new instances of instruments. i "SimpleSine" 1 5 60 85 i "SimpleSine" 2 5 64 80 i "Moogy" 3 5 67 75 i "Moogy" 4 5 71 70 ;6 extra seconds after the performance e 12 manual_src~dfsg/examples/genarray.csd0000644000000000000000000000143712262561504017073 0ustar rootroot ; Select audio/midi flags here according to platform -n sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;create and fill two arrays iArr1[] genarray 1, 6 iArr2[] genarray 1, 6, 2/3 ;print the content of iArr1 printf "%s", 1, "iArr1: start=1, end=6, step=default\n" kndx = 0 until kndx == lenarray(iArr1) do printf "iArr[%d] = %f\n", kndx+1, kndx, iArr1[kndx] kndx += 1 od ;print the content of iArr2 printf "%s", 1, "iArr2: start=1, end=6, step=2/3\n" kndx = 0 until kndx == lenarray(iArr2) do printf "iArr[%d] = %f\n", kndx+1, kndx, iArr2[kndx] kndx += 1 od turnoff endin i 1 0 1 manual_src~dfsg/examples/midictrl.csd0000644000000000000000000000135112262561504017065 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midictrl.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .5 ips midictrl 9, 10, 500 ;controller 9 kenv madsr 0.5, 0, 1, 0.5 asig pluck kenv, icps, ips, 2, 1 ;change tone color outs asig, asig endin f 2 0 4096 10 1 ;sine wave ; no score events allowed f0 30 ;runs 30 seconds e manual_src~dfsg/examples/linrand.csd0000644000000000000000000000140712262561504016707 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o linrand.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values klin linrand 100 printk .2, klin ; look aout oscili 0.8, 440+klin, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 klin linrand 100 printk .2, klin ; look aout oscili 0.8, 440+klin, 1 ; & listen outs aout, aout endin ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/comb.csd0000644000000000000000000000133312262561504016176 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o comb.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gamix init 0 instr 1 kcps expon p5, p3, p4 asig vco2 0.3, kcps outs asig, asig gamix = gamix + asig endin instr 99 krvt = 3.5 ilpt = 0.1 aleft comb gamix, krvt, ilpt aright comb gamix, krvt, ilpt*.2 outs aleft, aright clear gamix ; clear mixer endin i 1 0 3 20 2000 i 1 5 .01 440 440 i 99 0 8 e manual_src~dfsg/examples/filelen.csd0000644000000000000000000000147012262561504016676 0ustar rootroot ; Select audio/midi flags here according to platform ;-odac ;;;realtime audio out -iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o filelen.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; choose between mono or stereo file ilen filelen p4 ;calculate length of soundfile print ilen ichn filenchnls p4 ;check number of channels ;print ichn if (ichn == 1) then ;mono signal asig diskin2 p4, 1 outs asig, asig else ;stereo signal aL, aR diskin2 p4, .5, 0, 1 outs aL, aR endif endin i 1 0 3 "fox.wav" ;mono signal i 1 5 2 "kickroll.wav" ;stereo signal e manual_src~dfsg/examples/rms.csd0000644000000000000000000000237112262561504016062 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -m0 ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rms.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 ksmps = 128 nchnls = 1 ;Example by Andres Cabrera 2007 0dbfs = 1 FLpanel "rms", 400, 100, 50, 50 gkrmstext, gihrmstext FLtext "Rms", -100, 0, 0.1, 3, 110, 30, 60, 50 gkihp, gihandle FLtext "ihp", 0, 10, 0.05, 1, 100, 30, 220, 50 gkrmsslider, gihrmsslider FLslider "", -60, -0.5, -1, 5, -1, 380, 20, 10, 10 FLpanelEnd FLrun FLsetVal_i 5, gihandle ; Instrument #1. instr 1 a1 inch 1 label: kval rms a1, i(gkihp) ;measures rms of input channel 1 rireturn kval = dbamp(kval) ; convert to db full scale printk 0.5, kval FLsetVal 1, kval, gihrmsslider ;update the slider and text values FLsetVal 1, kval, gihrmstext knewihp changed gkihp ; reinit when ihp text has changed if (knewihp == 1) then reinit label ;needed because ihp is an i-rate parameter endif endin ; Play Instrument #1 for one minute i 1 0 60 e manual_src~dfsg/examples/pan.csd0000644000000000000000000000166512262561504016044 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pan.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 kcps = p4 k1 phasor kcps ; "fraction" of circle - controls speed of rotation - can be negative k2 tablei k1, 1, 1 ; sin of angle (sinusoid in f1) k3 tablei k1, 1, 1, .25, 1 ; cos of angle (sin offset 1/4 circle) arnd randomi 400, 1000, 50 ; produce random values asig poscil .7, arnd, 1 ; audio signal.. a1,a2,a3,a4 pan asig, k2/2, k3/2, 2, 1, 1 ; sent in a circle (f2=1st quad sin) outq a1, a2, a3, a4 endin f1 0 8192 10 1 f2 0 8193 9 .25 1 0 i1 0 10 .2 ;move to the tight i1 11 10 -.2 ;move to the left e manual_src~dfsg/examples/urd.csd0000644000000000000000000000206012262561504016046 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o urd.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktab = 1 ;ftable 1 kurd = urd(ktab) ktrig metro 5 ;triggers 5 times per second kres samphold kurd, ktrig ;sample and hold value of kurd printk2 kres ;print it asig poscil .5, 220+kres, 2 outs asig, asig endin instr 2 seed 0 ;every run new values ktab = 1 ;ftable 1 kurd = urd(ktab) ktrig metro 5 ;triggers 5 times per second kres samphold kurd, ktrig ;sample and hold value of kurd printk2 kres ;print it asig poscil .5, 220+kres, 2 outs asig, asig endin f1 0 -20 -42 10 20 .3 100 200 .7 ;30% choose between 10 and 20 and 70% between 100 and 200 f2 0 8192 10 1 ;sine wave i 1 0 5 i 2 6 5 e manual_src~dfsg/examples/gen32.csd0000644000000000000000000000274412262561504016203 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen32.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 itmp ftgen 1, 0, 16384, 7, 1, 16384, -1 ; sawtooth itmp ftgen 2, 0, 8192, 10, 1 ; sine itmp ftgen 5, 0, 4096, -32, -2, 1.5, 1.0, 0.25, 1, 2, 0.5, 0, 1, 3, -0.25, 0.5 ; mix tables itmp ftgen 6, 0, 16384, 20, 3, 1 ; window ; generate band-limited waveforms inote = 0 loop0: icps = 440 * exp(log(2) * (inote - 69) / 12) ; one table for inumh = sr / (2 * icps) ; each MIDI note number ift = int(inote + 256.5) itmp ftgen ift, 0, 4096, -30, 5, 1, inumh inote = inote + 1 if (inote < 127.5) igoto loop0 instr 1 kcps expon 20, p3, 16000 kft = int(256.5 + 69 + 12 * log(kcps / 440) / log(2)) kft = (kft > 383 ? 383 : kft) a1 phasor kcps a1 tableikt a1, kft, 1, 0, 1 outs a1*.5, a1*.5 endin instr 2 kcps expon 20, p3, 16000 kft = int(256.5 + 69 + 12 * log(kcps / 440) / log(2)) kft = (kft > 383 ? 383 : kft) kgdur limit 10 / kcps, 0.1, 1 a1 grain2 kcps, 0.02, kgdur, 30, kft, 6, -0.5 outs a1*.08, a1*.08 endin t 0 60 i 1 0 10 i 2 12 10 e manual_src~dfsg/examples/spsend_quad.csd0000644000000000000000000000211712262561504017565 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o space_quad.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 4 ga1 init 0 ga2 init 0 ga3 init 0 ga4 init 0 instr 1 asig diskin2 "beats.wav", 1 a1, a2, a3, a4 space asig, 0, 0, .1, p4, p5 ; take position values from p4, p5 ar1, ar2, ar3, ar4 spsend ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 a3 reverb2 ga3, 2.5, .5 a4 reverb2 ga4, 2.5, .5 outq a1, a2, a3, a4 ga1=0 ga2=0 ga3=0 ga4=0 endin ;place the sound in the left speaker and near i1 0 1 -1 1 ;place the sound in the right speaker and far i1 1 1 45 45 ;place the sound equally between left and right and in the middle ground distance i1 2 1 0 12 i 99 0 5 e manual_src~dfsg/examples/ATSinterpread.csd0000644000000000000000000000120612262561504017762 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc for RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSinterpread.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 1.8 ATSbufread ktime, 1, "beats.ats", 42 kamp ATSinterpread p4 aosc oscili kamp, p4, 1 outs aosc * 25, aosc *25 endin ; sine wave. f 1 0 16384 10 1 i 1 0 2 100 e manual_src~dfsg/examples/atonek.csd0000644000000000000000000000134612262561504016543 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o atonek.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 0, 0, 2^10, 10, 1 instr 1 ksig randomh 400, 1800, 150 aout poscil .2, 1000+ksig, gisin outs aout, aout endin instr 2 ksig randomh 400, 1800, 150 khp line 1, p3, 400 ;vary high-pass ksig atonek ksig, khp aout poscil .2, 1000+ksig, gisin outs aout, aout endin i 1 0 5 i 2 5.5 5 e manual_src~dfsg/examples/strrindex.csd0000644000000000000000000000120012262561504017271 0ustar rootroot ; Select audio/midi flags here according to platform -n ;;;no sound output ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ;example by Joachim Heintz opcode FilNam, S, S ;returns the name of a file path Spath xin ipos strrindex Spath, "/" ;look for the rightmost '/' Snam strsub Spath, ipos+1 ;extract the substring xout Snam endop instr name prints "Printing name:\n" Snam FilNam "/my/dir/my/file.WAV" puts Snam, 1 endin i "name" 0 0 e manual_src~dfsg/examples/tab.csd0000644000000000000000000000125112262561504016023 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tab.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifn1 ftgen 1, 0, 0, 1, "flute.aiff", 0, 4, 0 ;deferred-size table instr 1 atab init 0 isize tableng 1 ;length of table? print isize andx phasor 1 / (isize / sr) asig tab andx, 1, 1 ;has a 0 to 1 range outs asig, asig endin i 1 0 2.3 e manual_src~dfsg/examples/outletk.csd0000644000000000000000000000137612262561504016754 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o inletk.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 connect "bend", "bendout", "guitar", "bendin" instr bend kbend line p4, p3, p5 outletk "bendout", kbend endin instr guitar kbend inletk "bendin" kpch pow 2, kbend/12 printk2 kpch asig oscili .4, 440*kpch, 1 outs asig, asig endin f1 0 1024 10 1 i"guitar" 0 5 8.00 i"bend" 3 .2 -12 12 i"bend" 4 .1 -17 40 e manual_src~dfsg/examples/fluid-2.orc0000644000000000000000000000425712262561504016542 0ustar rootrootsr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS gienginenum1 fluidEngine gienginenum2 fluidEngine isfnum1 fluidLoad "Piano Steinway Grand Model C (21,738KB).sf2", gienginenum1, 1 ; Bright Steinway, program 1, channel 1 fluidProgramSelect gienginenum1, 1, isfnum1, 0, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidProgramSelect gienginenum1, 3, isfnum1, 0, 2 isfnum2 fluidLoad "63.3mg The Sound Site Album Bank V1.0.SF2", gienginenum2, 1 ; General MIDI, program 50, channel 2 fluidProgramSelect gienginenum2, 2, isfnum2, 0, 50 ; SEND NOTES TO STEINWAY SOUNDFONT instr 1 ; FluidSynth Steinway Rev ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 1 ikey = p4 ivelocity = p5 fluidNote gienginenum1, ichannel, ikey, ivelocity endin instr 2 ; GM soundfont ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 2 ikey = p4 ivelocity = p5 fluidNote gienginenum2, ichannel, ikey, ivelocity endin instr 3 ; FluidSynth Steinway Rev ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 3 ikey = p4 ivelocity = p5 fluidNote gienginenum1, ichannel, ikey, ivelocity endin ; COLLECT AUDIO FROM ALL SOUNDFONTS instr 100 ; Fluidsynth output ; INITIALIZATION ; Normalize so iamplitude for p5 of 80 == ampdb(80). iamplitude1 = ampdb(p5) * (10000.0 / 0.1) iamplitude2 = ampdb(p6) * (10000.0 / 0.1) ; AUDIO aleft1, aright1 fluidOut gienginenum1 aleft2, aright2 fluidOut gienginenum2 outs (aleft1 * iamplitude1) + (aleft2 * iamplitude2), \ (aright1 * iamplitude1) + (aright2 * iamplitude2) endin manual_src~dfsg/examples/ATSreadnz.csd0000644000000000000000000000122212262561504017106 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSreadnz.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 2 kenergy ATSreadnz ktime, "beats.ats", 2 anoise randi kenergy, 500 aout oscili 0.005, 455, 1 aout = aout * anoise outs aout, aout endin ; cosine wave f 1 0 16384 11 1 1 i 1 0 2 e manual_src~dfsg/examples/serialBegin.csd0000644000000000000000000000171212262561504017503 0ustar rootroot ; Select audio/midi flags here according to platform -n ;;;no output ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o serialBegin.wav -W ;;; for file output any platform sr = 44100 ksmps = 500 ; the default krate can be too fast for the arduino to handle nchnls = 2 0dbfs = 1 instr 1 iPort serialBegin "/dev/cu.usbmodemfa131", 9600 ;connect to the arduino with baudrate = 9600 kGain init 16 ;read our knob value kVal serialRead iPort if (kVal != -1) then kGain = kVal/128 endif aSig in ;get our audio input and get its rms kRms rms aSig*kGain kRms = kRms*kRms*255 ;scale the rms to a good value for the LED and send it out serialWrite iPort, (kRms < 255 ? kRms : 255) ;must be in range: 0-255 endin f 1 0 1024 10 1 1 1 1 1 1 i 1 0 200 e manual_src~dfsg/examples/midion2.csd0000644000000000000000000000067112262561504016623 0ustar rootroot ; Select audio/midi flags here according to platform -M0 -Q1 ;;;midi in and midi out sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps line 3, p3, .1 klf lfo 1, kcps, 3 ;use a unipolar square to trigger ktr trigger klf, 1, 1 ;from 3 times to .1 time per sec. midion2 1, 60, 100, ktr endin i 1 0 20 e manual_src~dfsg/examples/aresonk.csd0000644000000000000000000000135512262561504016724 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o aresonk.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 0, 0, 2^10, 10, 1 instr 1 ksig randomh 400, 1800, 150 aout poscil .2, 1000+ksig, gisin outs aout, aout endin instr 2 ksig randomh 400, 1800, 150 kbw line 1, p3, 600 ; vary bandwith ksig aresonk ksig, 800, kbw aout poscil .2, 1000+ksig, gisin outs aout, aout endin i 1 0 5 i 2 5.5 5 e manual_src~dfsg/examples/fluid.orc0000644000000000000000000000076012262561504016376 0ustar rootrootsr = 44100 ksmps = 100 nchnls = 2 giengine fluidEngine isfnum fluidLoad "07AcousticGuitar.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 0 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 70000 asigl, asigr fluidOut giengine outs asigl * imvol, asigr * imvol endinmanual_src~dfsg/examples/rewindscore.csd0000644000000000000000000000133012262561504017577 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o rewindscore.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kenv expon 1, p3, 0.0001 aout poscil .5*kenv, cpspch(p4), 1 outs aout, aout endin instr 100 rewindscore endin f1 0 4096 10 1 ;sine wave i1 0 1 8.00 i1 + 1 8.03 i1 + 1 8.04 i1 + 1 8.07 i1 + 1 8.09 i1 + 1 8.10 i1 + 1 8.09 i1 + 1 8.07 i1 + 1 8.03 i100 9 1 ;rewind from 9th second e manual_src~dfsg/examples/kgoto.csd0000644000000000000000000000200312262561504016374 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o kgoto.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Change kval linearly from 0 to 2 over ; the period set by the third p-field. kval line 0, p3, 2 ; If kval is greater than or equal to 1 then play the high note. ; If not then play the low note. if (kval >= 1) kgoto highnote kgoto lownote highnote: kfreq = 880 goto playit lownote: kfreq = 440 goto playit playit: ; Print the values of kval and kfreq. printks "kval = %f, kfreq = %f\\n", 1, kval, kfreq a1 oscil 10000, kfreq, 1 out a1 endin ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/pvslock.csd0000644000000000000000000000132312262561504016736 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvslock.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifil ftgen 0, 0, 0, 1, "fox.wav", 0, 0, 1 instr 1 klock = p4 fsig pvstanal 1, 1, 1, gifil ; no further transformations fsigout pvslock fsig, klock ; lock frequency aout pvsynth fsigout outs aout, aout endin i 1 0 2.6 1 ; locked i 1 3 2.6 0 ; not locked e manual_src~dfsg/examples/zkwm.csd0000644000000000000000000000300712262561504016246 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zkwm.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a basic instrument. instr 1 ; Generate a k-rate signal. ; The signal goes from 30 to 20,000 then back to 30. kramp linseg 30, p3/2, 20000, p3/2, 30 ; Mix the signal into the zk variable #1. zkwm kramp, 1 endin ; Instrument #2 -- another basic instrument. instr 2 ; Generate another k-rate signal. ; This is a low frequency oscillator. klfo lfo 3500, 2 ; Mix this signal into the zk variable #1. zkwm klfo, 1 endin ; Instrument #3 -- generates audio output. instr 3 ; Read zk variable #1, containing a mix of both signals. kamp zkr 1 ; Create a sine waveform. Its amplitude will vary ; according to the values in zk variable #1. a1 oscil kamp, 880, 1 ; Generate the audio output. out a1 ; Clear the zk variable, get it ready for ; another pass. zkcl 0, 1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 5 seconds. i 1 0 5 ; Play Instrument #2 for 5 seconds. i 2 0 5 ; Play Instrument #3 for 5 seconds. i 3 0 5 e manual_src~dfsg/examples/dispfft.csd0000644000000000000000000000123612262561504016717 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dispfft.wav -W ;;; for file output any platform ;be sure to NOT have -d in the CsOptions... sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 110 ifn = 1 knh line p4, p3, p5 asig buzz 1, kcps, knh, ifn outs asig, asig dispfft asig, .1, 2048, 0, 1 endin ;sine wave. f 1 0 16384 10 1 i 1 0 3 20 20 i 1 + 3 3 3 i 1 + 3 150 1 e manual_src~dfsg/examples/pvscent.csd0000644000000000000000000000143712262561504016745 0ustar rootroot -odac sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 giSine ftgen 0, 0, 4096, 10, 1 instr 1 irefrtm = p4; time for generating new values for the spectral centroid ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window ;Sfile = "flute-C-octave0.wav" Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of the audio-signal ktrig metro 1 / irefrtm if ktrig == 1 then kcenter pvscent fftin; spectral center endif aout oscil .2, kcenter, giSine out aout endin i 1 0 2.757 .3 i 1 3 2.757 .05 i 1 6 2.757 .005 i 1 9 2.757 .001 e manual_src~dfsg/examples/STKFMVoices.csd0000644000000000000000000000126112262561504017313 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKFMVoices.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;vowel kv2 line p7, p3, p8 ;specral tilt asig STKFMVoices cpspch(ifrq), 1, 2, kv1, 4, kv2, 11, 10, 1, 10, 128, 50 asig = asig * 4 ;amplify outs asig, asig endin i 1 0 5 5.00 10 120 0 0 i 1 + 2 8.00 80 82 127 0 e manual_src~dfsg/examples/seqtime.csd0000644000000000000000000000231712262561504016730 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o seqtime.wav -W ;;; for file output any platform sr = 44100 ksmps = 64 nchnls = 1 ; By Tim Mortimer and Andres Cabrera 2007 0dbfs = 1 gisine ftgen 0, 0, 8192, 10, 1 ;;; table defining an integer pitch set gipset ftgen 0, 0, 4, -2, 8.00, 8.04, 8.07, 8.10 ;;;DELTA times for seqtime gidelta ftgen 0, 0, 4, -2, .5, 1, .25, 1.25 instr 1 kndx init 0 ktrigger init 0 ktime_unit init 1 kstart init p4 kloop init p5 kinitndx init 0 kfn_times init gidelta ktrigger seqtime ktime_unit, kstart, kloop, kinitndx, kfn_times printk2 ktrigger if (ktrigger > 0) then kpitch table kndx, gipset event "i", 2, 0, 1, kpitch kndx = kndx + 1 kndx = kndx % kloop endif endin instr 2 icps = cpspch (p4) a1 buzz 1, icps, 7, gisine aamp expseg 0.00003,.02,1,p3-.02,0.00003 a1 = a1 * aamp * 0.5 out a1 endin ; start dur kstart kloop i 1 0 7 0 4 i 1 8 10 0 3 i 1 19 10 4 4 manual_src~dfsg/examples/vco2.csd0000644000000000000000000000351612262561504016134 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vco2.wav -W ;;; for file output any platform sr = 44100 ksmps = 10 nchnls = 1 ; user defined waveform -1: trapezoid wave with default parameters (can be ; accessed at ftables starting from 10000) itmp ftgen 1, 0, 16384, 7, 0, 2048, 1, 4096, 1, 4096, -1, 4096, -1, 2048, 0 ift vco2init -1, 10000, 0, 0, 0, 1 ; user defined waveform -2: fixed table size (4096), number of partials ; multiplier is 1.02 (~238 tables) itmp ftgen 2, 0, 16384, 7, 1, 4095, 1, 1, -1, 4095, -1, 1, 0, 8192, 0 ift vco2init -2, ift, 1.02, 4096, 4096, 2 instr 1 kcps expon p4, p3, p5 ; instr 1: basic vco2 example a1 vco2 12000, kcps ; (sawtooth wave with default out a1 ; parameters) endin instr 2 kcps expon p4, p3, p5 ; instr 2: kpw linseg 0.1, p3/2, 0.9, p3/2, 0.1 ; PWM example a1 vco2 10000, kcps, 2, kpw out a1 endin instr 3 kcps expon p4, p3, p5 ; instr 3: vco2 with user a1 vco2 14000, kcps, 14 ; defined waveform (-1) aenv linseg 1, p3 - 0.1, 1, 0.1, 0 ; de-click envelope out a1 * aenv endin instr 4 kcps expon p4, p3, p5 ; instr 4: vco2ft example, kfn vco2ft kcps, -2, 0.25 ; with user defined waveform a1 oscilikt 12000, kcps, kfn ; (-2), and sr/4 bandwidth out a1 endin i 1 0 3 20 2000 i 2 4 2 200 400 i 3 7 3 400 20 i 4 11 2 100 200 f 0 14 e manual_src~dfsg/examples/cuserrnd.csd0000644000000000000000000000155712262561504017113 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cuserrnd.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values kuser cuserrnd 0, 100, 1 printk .2, kuser asig poscil .5, 220+kuser, 3 outs asig, asig endin instr 2 ; every run time different values seed 0 kuser cuserrnd 0, 100, 1 printk .2, kuser asig poscil .5, 220+kuser, 3 outs asig, asig endin f 1 0 16 -7 1 4 0 8 0 4 1 ;distrubution using GEN07 f 2 0 16384 40 1 ;GEN40 is to be used with cuserrnd f 3 0 8192 10 1 ;sine i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/lpf18.csd0000644000000000000000000000174512262561504016217 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o lpf18.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a sine waveform. ; Note that its amplitude (kamp) ranges from 0 to 1. kamp init 1 kcps init 440 knh init 3 ifn = 1 asine buzz kamp, kcps, knh, ifn ; Filter the sine waveform. ; Vary the cutoff frequency (kfco) from 300 to 3,000 Hz. kfco line 300, p3, 3000 kres init 0.8 kdist = p4 ivol = p5 aout lpf18 asine, kfco, kres, kdist out aout * ivol endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; different distortion and volumes to compensate i 1 0 4 0.2 30000 i 1 4.5 4 0.9 27000 e manual_src~dfsg/examples/cggoto.csd0000644000000000000000000000133612262561504016543 0ustar rootroot ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; -o cggoto.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = p4 ; If i1 is equal to one, play a high note. ; Otherwise play a low note. cggoto (i1 == 1), highnote lownote: a1 oscil 10000, 220, 1 goto playit highnote: a1 oscil 10000, 440, 1 goto playit playit: out a1 endin ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; Play lownote for one second. i 1 0 1 1 ; Play highnote for one second. i 1 1 1 2 e manual_src~dfsg/examples/trcross.csd0000644000000000000000000000171412262561504016760 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trcross.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain1 diskin2 "beats.wav", 1, 0, 1 ain2 diskin2 "fox.wav", 1 imode = p4 fs1,fsi2 pvsifd ain1, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .01, 1, 3, 500 ; partial tracking fs11,fsi12 pvsifd ain2, 2048, 512, 1 ; ifd analysis (second input) fst1 partials fs11, fsi12, .01, 1, 3, 500 ; partial tracking (second input fcr trcross fst, fst1, 1.05, 1, imode ; cross-synthesis (mode 0 and mode 1) aout tradsyn fcr, 1, 1, 500, 1 ; resynthesis of tracks outs aout*3, aout*3 endin f1 0 8192 10 1 i 1 0 3 0 i 1 5 3 1 e manual_src~dfsg/examples/linenr.csd0000644000000000000000000000375612262561504016560 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc -M0 ;;;RT audio I/O with MIDI in ; Example by Jonathan Murphy and Charles Gran 2007 sr = 44100 ksmps = 10 nchnls = 2 ; new, and important. Make sure that midi note events are only ; received by instruments that actually need them. ; turn default midi routing off massign 0, 0 ; route note events on channel 1 to instr 1 massign 1, 1 ; Define your midi controllers #define C1 #21# #define C2 #22# #define C3 #23# ; Initialize MIDI controllers initc7 1, 21, 0.5 ;delay send initc7 1, 22, 0.5 ;delay: time to zero initc7 1, 23, 0.5 ;delay: rate gaosc init 0 ; Define an opcode to "smooth" the MIDI controller signal opcode smooth, k, k kin xin kport linseg 0, 0.0001, 0.01, 1, 0.01 kin portk kin, kport xout kin endop instr 1 ; Generate a sine wave at the frequency of the MIDI note that triggered the intrument ifqc cpsmidi iamp ampmidi 10000 aenv linenr iamp, .01, .1, .01 ;envelope a1 oscil aenv, ifqc, 1 ; All sound goes to the global variable gaosc gaosc = gaosc + a1 endin instr 198 ; ECHO kcmbsnd ctrl7 1, 21, 0, 1 ;delay send ktime ctrl7 1, 22, 0.01, 6 ;time loop fades out kloop ctrl7 1, 23, 0.01, 1 ;loop speed ; Receive MIDI controller values and then smooth them kcmbsnd smooth kcmbsnd ktime smooth ktime kloop smooth kloop imaxlpt = 1 ;max loop time ; Create a variable reverberation (delay) of the gaosc signal acomb vcomb gaosc, ktime, kloop, imaxlpt, 1 aout = (acomb * kcmbsnd) + gaosc * (1 - kcmbsnd) outs aout, aout gaosc = 0 endin f1 0 16384 10 1 i198 0 10000 e manual_src~dfsg/examples/pdclip.csd0000644000000000000000000000151112262561504016527 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o abs.wav -W ;;; for file output any platform ; test instrument for pdclip opcode instr 3 idur = p3 iamp = p4 ifreq = p5 ifn = p6 kenv linseg 0, .05, 1.0, idur - .1, 1.0, .05, 0 aosc oscil 1.0, ifreq, ifn kmod expseg 0.00001, idur, 1.0 aout pdclip aosc, kmod, 0.0, 1.0 out kenv*aout*iamp endin f1 0 16385 10 1 f2 0 16385 10 1 .5 .3333 .25 .5 ; pdclipped sine wave i3 0 3 15000 440 1 i3 + 3 15000 330 1 i3 + 3 15000 220 1 s ; pdclipped composite wave i3 0 3 15000 440 2 i3 + 3 15000 330 2 i3 + 3 15000 220 2 e manual_src~dfsg/examples/powershape.csd0000644000000000000000000000144112262561504017433 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o abs.wav -W ;;; for file output any platform instr 1 imaxamp = 10000 kshapeamt line p5, p3, p6 aosc oscili 1.0, cpspch(p4), 1 aout powershape aosc, kshapeamt adeclick linseg 0.0, 0.01, 1.0, p3 - 0.06, 1.0, 0.05, 0.0 out aout * adeclick * imaxamp endin f1 0 32768 10 1 i1 0 1 7.00 0.000001 0.8 i1 + 0.5 7.02 0.01 1.0 i1 + . 7.05 0.5 1.0 i1 + . 7.07 4.0 1.0 i1 + . 7.09 1.0 10.0 i1 + 2 7.06 1.0 25.0 e manual_src~dfsg/examples/FLbox.csd0000644000000000000000000000207312262561504016272 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLbox.wav -W ;;; for file output any platform sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Text Box", 700, 400, 50, 50 ; Box border type (7=embossed box) itype = 7 ; Font type (10='Times Bold') ifont = 10 ; Font size isize = 20 ; Width of the flbox iwidth = 400 ; Height of the flbox iheight = 30 ; Distance of the left edge of the flbox ; from the left edge of the panel ix = 150 ; Distance of the upper edge of the flbox ; from the upper edge of the panel iy = 100 ih3 FLbox "Use Text Boxes For Labelling", itype, ifont, isize, iwidth, iheight, ix, iy ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 endin ; Real-time performance for 1 hour. f 0 3600 e manual_src~dfsg/examples/random.csd0000644000000000000000000000145612262561504016544 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o random.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;same values every time krnd random 100, 1000 printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin instr 2 ;different values every time seed 0 krnd random 100, 1000 ; seed from system clock printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin f 1 0 16384 10 1 ;sine wave. i 1 0 1 i 2 2 1 e manual_src~dfsg/examples/midinoteonpch.csd0000644000000000000000000000204412262561504020116 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midinoteonpch.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 midinoteonpch p4, p5 ;gets a MIDI note number value as octave-point-decimal value into p4, and MIDI velocity into p5 print p4 ;display the pitch value when it changes and when key is pressed kvel = kvel/127 ;scale midi velocity to 0-1 ipch = p4 icps = cpspch(ipch) ;convert octave-point-decimal value into Hz kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by asig pluck kenv*kvel, icps, icps, 2, 1 ;velocity value outs asig, asig endin f 0 30 ;runs for 30 seconds f 2 0 4096 10 1 i 1 0 2 8.09 100 ; play these notes from score as well i 1 + 2 9.05 100 e manual_src~dfsg/examples/pvsbufread.csd0000644000000000000000000000236312262561504017423 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsbufread.wav -W ;;; for file output any platform sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 opcode FileToPvsBuf, iik, Siiii ;;writes an audio file at the first k-cycle to a fft-buffer (via pvsbuffer) Sfile, ifftsize, ioverlap, iwinsize, iwinshape xin ktimek timeinstk if ktimek == 1 then ilen filelen Sfile kcycles = ilen * kr; number of k-cycles to write the fft-buffer kcount init 0 loop: ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape ibuf, ktim pvsbuffer fftin, ilen + (ifftsize / sr) loop_lt kcount, 1, kcycles, loop xout ibuf, ilen, ktim endif endop instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window ibuffer, ilen, k0 FileToPvsBuf "fox.wav", ifftsize, ioverlap, iwinsize, iwinshape ktmpnt linseg ilen, p3, 0; reads the buffer backwards in p3 seconds fread pvsbufread ktmpnt, ibuffer aout pvsynth fread out aout endin i 1 0 5 e manual_src~dfsg/examples/trirand.csd0000644000000000000000000000146112262561504016723 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trirand.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values ktri trirand 100 printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 ktri trirand 100 printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/FLtabs.csd0000644000000000000000000000320712262561504016433 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLtabs.wav -W ;;; for file output any platform ; A single oscillator with frequency, amplitude and ; panning controls on separate file tab cards sr = 44100 kr = 441 ksmps = 100 nchnls = 2 FLpanel "Tabs", 300, 350, 100, 100 itabswidth = 280 itabsheight = 330 ix = 5 iy = 5 FLtabs itabswidth,itabsheight, ix,iy itab1width = 280 itab1height = 300 itab1x = 10 itab1y = 40 FLgroup "Tab 1", itab1width, itab1height, itab1x, itab1y gkfreq, i1 FLknob "Frequency", 200, 5000, -1, 1, -1, 70, 70, 130 FLsetVal_i 400, i1 FLgroupEnd itab2width = 280 itab2height = 300 itab2x = 10 itab2y = 40 FLgroup "Tab 2", itab2width, itab2height, itab2x, itab2y gkamp, i2 FLknob "Amplitude", 0, 15000, 0, 1, -1, 70, 70, 130 FLsetVal_i 15000, i2 FLgroupEnd itab3width = 280 itab3height = 300 itab3x = 10 itab3y = 40 FLgroup "Tab 3", itab3width, itab3height, itab3x, itab3y gkpan, i3 FLknob "Pan position", 0, 1, 0, 1, -1, 70, 70, 130 FLsetVal_i 0.5, i3 FLgroupEnd FLtabsEnd FLpanelEnd ; Run the widget thread! FLrun instr 1 ifn = 1 asig oscili gkamp, gkfreq, ifn outs asig*(1-gkpan), asig*gkpan endin ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e manual_src~dfsg/examples/rand.csd0000644000000000000000000000143212262561504016202 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o rand.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;same values every time krnd rand 100 printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin instr 2 ;different values every time krnd rand 100, 10 ; seed from system clock printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin f 1 0 16384 10 1 ;sine wave. i 1 0 1 i 2 2 1 e manual_src~dfsg/examples/include.csd0000644000000000000000000000115512262561504016703 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o include.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a basic oscillator. instr 1 kamp = 10000 kcps = 440 ifn = 1 a1 oscil kamp, kcps, ifn out a1 endin ; Include the file for Table #1. #include "table1.inc" ; Play Instrument #1 for 2 seconds. i 1 0 2 e manual_src~dfsg/examples/freeverb.csd0000644000000000000000000000131112262561504017052 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o freeverb.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 a1 vco2 0.75, 440, 10 kfrq port 100, 0.008, 20000 a1 butterlp a1, kfrq a2 linseg 0, 0.003, 1, 0.01, 0.7, 0.005, 0, 1, 0 a1 = a1 * a2 denorm a1 aL, aR freeverb a1, a1, 0.9, 0.35, sr, 0 outs a1 + aL, a1 + aR endin i 1 0 5 e manual_src~dfsg/examples/eee.aiff0000644000000000000000000000035612262561504016154 0ustar rootrootFORMAIFFCOMM\@ DSSND1)@&=Q#=M > %p3N$C]K!! AF }LcP] i2mrܴv6ޣ~ޣs/w1manual_src~dfsg/examples/adsynt2.csd0000644000000000000000000000435512262561504016651 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o adsynt2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; Generate a sinewave table. giwave ftgen 1, 0, 1024, 10, 1 ; Generate two empty tables for adsynt2. gifrqs ftgen 2, 0, 32, 7, 0, 32, 0 ; A table for freqency and amp parameters. giamps ftgen 3, 0, 32, 7, 0, 32, 0 ; Generates parameters at init time instr 1 ; Generate 10 voices. icnt = 10 ; Init loop index. index = 0 ; Loop only executed at init time. loop: ; Define non-harmonic partials. ifreq pow index + 1, 1.5 ; Define amplitudes. iamp = 1 / (index+1) ; Write to tables. tableiw ifreq, index, gifrqs ; Used by adsynt2. tableiw iamp, index, giamps index = index + 1 ; Do loop/ if (index < icnt) igoto loop asig adsynt2 0.4, 150, giwave, gifrqs, giamps, icnt outs asig, asig endin ; Generates parameters every k-cycle. instr 2 ; Generate 10 voices. icnt = 10 ; Reset loop index. kindex = 0 ; Loop executed every k-cycle. loop: ; Generate lfo for frequencies. kspeed pow kindex + 1, 1.6 ; Individual phase for each voice. kphas phasorbnk kspeed * 0.7, kindex, icnt klfo table kphas, giwave, 1 ; Arbitrary parameter twiddling... kdepth pow 1.4, kindex kfreq pow kindex + 1, 1.5 kfreq = kfreq + klfo*0.006*kdepth ; Write freqs to table for adsynt2. tablew kfreq, kindex, gifrqs ; Generate lfo for amplitudes. kspeed pow kindex + 1, 0.8 ; Individual phase for each voice. kphas phasorbnk kspeed*0.13, kindex, icnt, 2 klfo table kphas, giwave, 1 ; Arbitrary parameter twiddling... kamp pow 1 / (kindex + 1), 0.4 kamp = kamp * (0.3+0.35*(klfo+1)) ; Write amps to table for adsynt2. tablew kamp, kindex, giamps kindex = kindex + 1 ; Do loop. if (kindex < icnt) kgoto loop asig adsynt2 0.25, 150, giwave, gifrqs, giamps, icnt outs asig, asig endin ; Play Instrument #1 for 2.5 seconds. i 1 0 2.5 ; Play Instrument #2 for 2.5 seconds. i 2 3 2.5 e manual_src~dfsg/examples/scans-2.csd0000644000000000000000000000271212262561504016526 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o scans-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 strset 1, "mary.wav" strset 2, "fox.wav" instr 2 ;show 2 different trajectories, with samples as excitation signal ismp = p6 iamp = p7 itrj = p8 aout soundin p6 ;choose wave file scanu ismp, .01, 6, 2, 33, 44, 5, 2, .01, .05, -.05, .1, .5, 0, 0, aout, 1, 0 asig scans iamp, cpspch(p5), itrj , 0 outs asig, asig endin f1 0 128 7 0 64 1 64 0 ; Initial condition f2 0 128 -7 1 128 0.3 ; Masses f33 0 16384 -23 "cylinder-128,8" ; Spring matrices f44 0 128 -7 2 4 0 124 2 ; Centering force f5 0 128 -7 1 128 0 ; Damping f6 0 128 -7 -.0 128 0 ; Initial velocity f7 0 128 -5 .001 128 128 ; Trajectories f77 0 128 -23 "spiral-8,16,128,2,1over2" s i2 0 5 63 6.00 1 .9 7 ;"mary.wav" & i2 6 5 60 7.00 ;trajectory table 7 i2 10 5 60 8.00 s i2 0 5 63 6.00 2 .08 7 ;"fox.wav", at much lower volume i2 6 5 60 7.00 i2 10 5 60 8.00 s i2 0 5 63 6.00 1 .9 77 ;"mary.wav" & i2 6 5 60 7.00 ;trajectory table 77 i2 10 5 60 8.00 s i2 0 5 63 6.00 2 .08 77 ;"fox.wav", at much lower volume i2 6 5 60 7.00 i2 10 5 60 8.00 e manual_src~dfsg/examples/midinoteonkey.csd0000644000000000000000000000154112262561504020135 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M1 -Q1 ;;;realtime audio out and midi in and midi out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midinoteonkey.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;uses external midi device kkey init 0 ;initialize key number kvel init 0 ;initialize velocity midinoteonkey kkey, kvel ;MIDI note number value printk2 kkey ;display the key value when it changes and when key is pressed midion 1, kkey, kvel ;sent note to external device endin f 0 30 ;runs for 30 seconds i 1 0 2 62 ; play these notes from score as well i 1 + 2 65 e manual_src~dfsg/examples/fof2.csd0000644000000000000000000000377212262561504016123 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o fof2.wav -W ;;; for file output any platform sr = 44100 ksmps = 128 nchnls = 2 ;By Andres Cabrera 2007 instr 1 ;table-lookup vocal synthesis kris init p12 kdur init p13 kdec init p14 iolaps init p15 ifna init 1 ; Sine wave ifnb init 2 ; Straight line rise shape itotdur init p3 kphs init 0 ; No phase modulation (constant kphs) kfund line p4, p3, p5 kform line p6, p3, p7 koct line p8, p3, p9 kband line p10, p3, p11 kgliss line p16, p3, p17 kenv linen 5000, 0.03, p3, 0.03 ;to avoid clicking aout fof2 kenv, kfund, kform, koct, kband, kris, kdur, kdec, iolaps, \ ifna, ifnb, itotdur, kphs, kgliss outs aout, aout endin f1 0 8192 10 1 f2 0 4096 7 0 4096 1 ; kfund1 kfund2 kform1 kform2 koct1 koct2 kband1 kband2 kris kdur kdec iolaps kgliss1 kgliss2 i1 0 4 220 220 510 510 0 0 30 30 0.01 0.03 0.01 20 1 1 i1 + . 220 220 510 910 0 0 30 30 0.01 0.03 0.01 20 1 1 i1 + . 220 220 510 510 0 0 100 30 0.01 0.03 0.01 20 1 1 i1 + . 220 220 510 510 0 1 30 30 0.01 0.03 0.01 20 1 1 i1 + . 220 220 510 510 0 0 30 30 0.01 0.03 0.01 20 1 2 i1 + . 220 220 510 510 0 0 30 30 0.01 0.03 0.01 20 0.5 1 i1 + . 220 220 510 510 0 0 30 30 0.01 0.05 0.01 100 1 1 i1 + . 220 440 510 510 0 0 30 30 0.01 0.05 0.01 100 1 1 e manual_src~dfsg/examples/rtclock.csd0000644000000000000000000000257412262561504016727 0ustar rootroot ; Select audio/midi flags here according to platform -n ;;;no sound ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o rtclock.wav -W ;;; for file output any platform ;after an example by Iain McCurdy sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 FLcolor 200, 200, 200, 0, 0, 0 ; LABEL | WIDTH | HEIGHT | X | Y FLpanel "rtclock", 500, 130, 0, 0 ; ON,OFF,TYPE,WIDTH, HEIGHT, X, Y, OPCODE, INS,START,IDUR gkOnOff,ihOnOff FLbutton "On/Off", 1, 0, 22, 150, 25, 5, 5, 0, 1, 0, 3600 gkExit,ihExit FLbutton "exitnow",1, 0, 21, 150, 25, 345, 5, 0, 999, 0, 0.001 FLsetColor2 255, 0, 50, ihOnOff ;reddish color ;VALUE DISPLAY BOXES WIDTH,HEIGHT,X, Y gidclock FLvalue "clock", 100, 25, 200, 60 FLsetVal_i 1, ihOnOff FLpanel_end FLrun instr 1 if gkOnOff !=0 kgoto CONTINUE ;sense if FLTK on/off switch is not off (in which case skip the next line) turnoff ;turn this instr. off now CONTINUE: ktime rtclock ;clock continues to run even FLprintk2 ktime, gidclock ;after the on/off button was used to stop endin instr 999 exitnow ;exit Csound as fast as possible endin f 0 60 ;runs 60 seconds e manual_src~dfsg/examples/mode.csd0000644000000000000000000000336012262561504016204 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o moogvcf.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1; 2 modes excitator idur init p3 ifreq11 init p4 ifreq12 init p5 iQ11 init p6 iQ12 init p7 iamp init ampdb(p8) ifreq21 init p9 ifreq22 init p10 iQ21 init p11 iQ22 init p12 ; to simulate the shock between the excitator and the resonator ashock mpulse 3,0 aexc1 mode ashock,ifreq11,iQ11 aexc1 = aexc1*iamp aexc2 mode ashock,ifreq12,iQ12 aexc2 = aexc2*iamp aexc = (aexc1+aexc2)/2 ;"Contact" condition : when aexc reaches 0, the excitator looses ;contact with the resonator, and stops "pushing it" aexc limit aexc,0,3*iamp ; 2modes resonator ares1 mode aexc,ifreq21,iQ21 ares2 mode aexc,ifreq22,iQ22 ares = (ares1+ares2)/2 display aexc+ares,p3 outs aexc+ares,aexc+ares endin ;wooden excitator against glass resonator i1 0 8 1000 3000 12 8 70 440 888 500 420 ;felt against glass i1 4 8 80 188 8 3 70 440 888 500 420 ;wood against wood i1 8 8 1000 3000 12 8 70 440 630 60 53 ;felt against wood i1 12 8 80 180 8 3 70 440 630 60 53 i1 16 8 1000 3000 12 8 70 440 888 2000 1630 i1 23 8 80 180 8 3 70 440 888 2000 1630 ;With a metallic excitator i1 33 8 1000 1800 1000 720 70 440 882 500 500 i1 37 8 1000 1800 1000 850 70 440 630 60 53 i1 42 8 1000 1800 2000 1720 70 440 442 500 500 manual_src~dfsg/examples/tbvcf.csd0000644000000000000000000000520512262561504016364 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tbvcf.wav -W ;;; for file output any platform ;--------------------------------------------------------- ; TBVCF Test ; Coded by Hans Mikelson December, 2000 ;--------------------------------------------------------- sr = 44100 ; Sample rate ksmps = 10 ; Samples/Kontrol period nchnls = 2 ; Normal stereo 0dbfs = 1 instr 10 idur = p3 ; Duration iamp = p4 ; Amplitude ifqc = cpspch(p5) ; Pitch to frequency ipanl = sqrt(p6) ; Pan left ipanr = sqrt(1-p6) ; Pan right iq = p7 idist = p8 iasym = p9 kdclck linseg 0, .002, 1, idur-.004, 1, .002, 0 ; Declick envelope kfco expseg 10000, idur, 1000 ; Frequency envelope ax vco 1, ifqc, 2, 0.5 ; Square wave ay tbvcf ax, kfco, iq, idist, iasym ; TB-VCF ay buthp ay/1, 100 ; Hi-pass outs ay*iamp*ipanl*kdclck, ay*iamp*ipanr*kdclck endin f1 0 65536 10 1 ; TeeBee Test ; Sta Dur Amp Pitch Pan Q Dist1 Asym i10 0 0.2 0.5 7.00 .5 0.0 2.0 0.0 i10 0.3 0.2 0.5 7.00 .5 0.8 2.0 0.0 i10 0.6 0.2 0.5 7.00 .5 1.6 2.0 0.0 i10 0.9 0.2 0.5 7.00 .5 2.4 2.0 0.0 i10 1.2 0.2 0.5 7.00 .5 3.2 2.0 0.0 ;i10 1.5 0.2 0.5 7.00 .5 4.0 2.0 0.0 i10 1.8 0.2 0.5 7.00 .5 0.0 2.0 0.25 i10 2.1 0.2 0.5 7.00 .5 0.8 2.0 0.25 i10 2.4 0.2 0.5 7.00 .5 1.6 2.0 0.25 i10 2.7 0.2 0.5 7.00 .5 2.4 2.0 0.25 i10 3.0 0.2 0.5 7.00 .5 3.2 2.0 0.25 i10 3.3 0.2 0.5 7.00 .5 4.0 2.0 0.25 i10 3.6 0.2 0.5 7.00 .5 0.0 2.0 0.5 i10 3.9 0.2 0.5 7.00 .5 0.8 2.0 0.5 i10 4.2 0.2 0.5 7.00 .5 1.6 2.0 0.5 i10 4.5 0.2 0.5 7.00 .5 2.4 2.0 0.5 i10 4.8 0.2 0.5 7.00 .5 3.2 2.0 0.5 i10 5.1 0.2 0.5 7.00 .5 4.0 2.0 0.5 i10 5.4 0.2 0.5 7.00 .5 0.0 2.0 0.75 i10 5.7 0.2 0.5 7.00 .5 0.8 2.0 0.75 i10 6.0 0.2 0.5 7.00 .5 1.6 2.0 0.75 i10 6.3 0.2 0.5 7.00 .5 2.4 2.0 0.75 i10 6.6 0.2 0.5 7.00 .5 3.2 2.0 0.75 i10 6.9 0.2 0.5 7.00 .5 4.0 2.0 0.75 i10 7.2 0.2 0.5 7.00 .5 0.0 2.0 1.0 i10 7.5 0.2 0.5 7.00 .5 0.8 2.0 1.0 i10 7.8 0.2 0.5 7.00 .5 1.6 2.0 1.0 i10 8.1 0.2 0.5 7.00 .5 2.4 2.0 1.0 i10 8.4 0.2 0.5 7.00 .5 3.2 2.0 1.0 i10 8.7 0.2 0.5 7.00 .5 4.0 2.0 1.0 e manual_src~dfsg/examples/gen23.csd0000644000000000000000000000223012262561504016171 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen23.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;"spectrum.txt" is created by the spectrum plotter of Audacity (set at size 128), using "fox.wav". instr 1 ;performs additive synthesis based on spectrum.txt indx =0 ;start reading at first value loop: ifreq tab_i indx, 2 ;take odd values of list (= frequency) iamp tab_i indx+1, 2 ;take even values of list (= amplitude) event_i "i", 10, 0, p3, iamp, ifreq ;use "event_i" to trigger instr. 10 loop_lt indx, 2, 128, loop ;use all 128 frequency and amplitude values endin instr 10 ;generate sound iamp = p4 ifreq = p5 asig poscil ampdb(iamp), ifreq, 1 asig linen asig, .01, p3, p2 outs asig, asig endin f 1 0 16384 10 1 ;sine wave f 2 0 128 -23 "spectrum.txt" ;"spectrum.txt" can be found in /manual/examples i1 0 2 e manual_src~dfsg/examples/until.csd0000644000000000000000000000105512262561504016412 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ifthen.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 lab99: if p4<0 goto lab100 p4 = p4-1 print p4 goto lab99 lab100: endin instr 2 until p4<0 do p4 = p4-1 print p4 od endin i 1 1 1 4 i 2 2 1 4 e manual_src~dfsg/examples/cauchyi.csd0000644000000000000000000000111412262561504016700 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o exprand.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 klambda cauchyi 100, 1, 3 printk2 klambda ; look aout oscili 0.8, 440+klambda, 1 ; & listen outs aout, aout endin ; sine wave f 1 0 16384 10 1 i 1 0 4 e manual_src~dfsg/examples/duserrnd.csd0000644000000000000000000000120612262561504017103 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o duserrnd.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 k1 duserrnd 1 printk 0, k1 asig poscil .5, 220*k1, 2 ;multiply frequency with random value outs asig, asig endin f1 0 -20 -41 2 .1 8 .9 ;choose 2 at 10% probability, and 8 at 90% f2 0 8192 10 1 i1 0 2 e manual_src~dfsg/examples/pop.csd0000644000000000000000000000140112262561504016050 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pop.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 stack 100000 instr 1 a1 oscils 0.7, 220, 0 k1 line 0, p3, 1 push "blah", 123.45, a1, k1 push rnd(k1) k_rnd pop S01, i01, a01, k01 pop printf_i "S01 = '%s', i01 = %g\n", 1, S01, i01 ktrig metro 5.0 printf "k01 = %.3f, k_rnd = %.3f\n", ktrig, k01, k_rnd outs a01, a01 endin i 1 0 5 e manual_src~dfsg/examples/biquad.csd0000644000000000000000000000252612262561504016530 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o biquad.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Instrument #1. instr 1 ; Get the values from the score. idur = p3 iamp = p4 icps = cpspch(p5) kfco = p6 krez = p7 ; Calculate the biquadratic filter's coefficients kfcon = 2*3.14159265*kfco/sr kalpha = 1-2*krez*cos(kfcon)*cos(kfcon)+krez*krez*cos(2*kfcon) kbeta = krez*krez*sin(2*kfcon)-2*krez*cos(kfcon)*sin(kfcon) kgama = 1+cos(kfcon) km1 = kalpha*kgama+kbeta*sin(kfcon) km2 = kalpha*kgama-kbeta*sin(kfcon) kden = sqrt(km1*km1+km2*km2) kb0 = 1.5*(kalpha*kalpha+kbeta*kbeta)/kden kb1 = kb0 kb2 = 0 ka0 = 1 ka1 = -2*krez*cos(kfcon) ka2 = krez*krez ; Generate an input signal. axn vco 1, icps, 1 ; Filter the input signal. ayn biquad axn, kb0, kb1, kb2, ka0, ka1, ka2 outs ayn*iamp/2, ayn*iamp/2 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Sta Dur Amp Pitch Fco Rez i 1 0.0 1.0 20000 6.00 1000 .8 i 1 1.0 1.0 20000 6.03 2000 .95 e manual_src~dfsg/examples/outs2.csd0000644000000000000000000000123612262561504016334 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outs2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig vco2 .01, 110 ; sawtooth waveform at low volume kcut line 300, p3, 60 ; Vary cutoff frequency kresonance = 3 inumlayer = 3 asig lowresx asig, kcut, kresonance, inumlayer outs2 asig ; output stereo channel 2 only endin i 1 0 3 e manual_src~dfsg/examples/vdivv.csd0000644000000000000000000000170512262561504016417 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ifn2 = p5 ielements = p6 idstoffset = p7 isrcoffset = p8 kval init 25 vdivv ifn1, ifn2, ielements, idstoffset, isrcoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 15 16 f 2 0 16 -7 1 15 2 i2 0.0 0.2 1 i2 0.2 0.2 2 i1 0.4 0.01 1 2 5 3 8 i2 0.8 0.2 1 i1 1.0 0.01 1 2 5 10 -2 i2 1.2 0.2 1 i1 1.4 0.01 1 2 8 14 0 i2 1.6 0.2 1 i1 1.8 0.01 1 2 8 0 14 i2 2.0 0.2 1 i1 2.2 0.002 1 1 8 5 2 i2 2.4 0.2 1 e manual_src~dfsg/examples/oscili.csd0000644000000000000000000000206112262561504016537 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscili.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = .6 kcps = 440 ifn = p4 asig oscil kamp, kcps, ifn outs asig,asig endin instr 2 kamp = .6 kcps = 440 ifn = p4 asig oscili kamp, kcps, ifn outs asig,asig endin f1 0 128 10 1 ; Sine with a small amount of data f2 0 128 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth with a small amount of data f3 0 128 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square with a small amount of data f4 0 128 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse with a small amount of data i 1 0 2 1 i 2 3 2 1 i 1 6 2 2 i 2 9 2 2 i 1 12 2 3 i 2 15 2 3 i 1 18 2 4 i 2 21 2 4 e manual_src~dfsg/examples/phasor.csd0000644000000000000000000000140112262561504016546 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o phasor.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = 1 ;read table 1 with our index ixmode = 1 kndx phasor p4 kfrq table kndx, ifn, ixmode asig poscil .6, kfrq, 2 ;re-synthesize with sine outs asig, asig endin f 1 0 1025 -7 200 1024 2000 ;a line from 200 to 2,000 f 2 0 16384 10 1;sine wave i 1 0 1 1 ;once per second i 1 2 2 .5 ;once per 2 seconds i 1 5 1 2 ;twice per second e manual_src~dfsg/examples/joystick.csd0000644000000000000000000000305512262561504017120 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o joystick.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 ;0dbfs = 1 instr 1 ; gives information about your joystick in real time kmask joystick 0, 1 kidx = 2 kaxes tab 0, 1 ; number of axes has been stored in position 0 kbuttons tab 1, 1 ; number of buttons has been stored in position 1 printf "this joystick has %d axes, %d buttons\n", kidx, kaxes, kbuttons kuniq init 0 reportaxis: ; first we see if we have any x/y input kcheck = kmask & (1< f1 0 32 7 0 7 0 ; will hold the joystick data i1 0 60000 e manual_src~dfsg/examples/multiplies.csd0000644000000000000000000000123512262561504017446 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o multiplies.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kenv expon 0.01, p3, 1 aout poscil 0.8*kenv, 440, 1 ;multiply amplitude from 0 to 1 * 0.8 printks "base amplitude * rising expon output = 0.8 * %f\n", .1, kenv outs aout, aout endin f 1 0 16384 10 1 ; sine wave i 1 0 2 e manual_src~dfsg/examples/genwave.csd0000644000000000000000000000630012262561504016711 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o 0dbfs.wav -W ;;; for file output any platform sr = 44100 kr = 4410 nchnls = 1 0dbfs = 1 zakinit 3,1 instr 1 ; wavelet synth instrument iamp = p4; scaling factor of wavelets ifreq = p5; frequency of wavelets itab = p6; selected wavelet function inum = p7; number of wavelets to be created a1 osciln p4, p5, p6, p7 out a1 endin instr 2 ; wavelet analysis intrument a1 soundin "fox.wav" ; Decomposition Structure: ; 1 LEVEL 2 LEVEL ; HP->ah1 ; a1->| HP(up2)->ah2 ; LP->al1->| ; LP(up2)->al2 ; ain = a1*.5; attenuate input signal ; since wavelet coefficients ; could reach big values ah1 dconv ain,ftlen(8),8 al1 dconv ain,ftlen(7),7 ah2 dconv al1,ftlen(10),10 al2 dconv al1,ftlen(9),9 zaw ah1,0 zaw al1,1 zaw ah2,2 zaw al2,3 aout zar p4 out aout zacl 0,3 endin ; First of all, we need several FIR filters which are capable ; to produce wavelet families. ; One can input filter coefficients manualy using GEN02 ; or read them from text file. ; Most of compact-supported wavelet coefficients can be obtained from ; Wavelet Browser by PyWavelets wavelets.pybytes.com ; You can select family and order of filter ; then copy desired coefficients into txt file. ; Notice that for correct interpretation of results you should use ; coeffs of Decomposition low-pass filter. ; Daubechies 2 f 1 0 4 -2 -0.1294095226 0.2241438680 0.8365163037 0.4829629131 ; Symlet 10 f 2 0 0 -23 "sym10.txt" ; Now we want to produce some wavelet granules. ; They can be used in wavelet synthesis etc. ; Tables of large sizes should produce smoother wavelets. ; We take array of filter coefficients from ftable 1 ; and deconvolve it until output length of 16384. ; The order of filters through the deconvolution process ; is given by 14 which is 1110 in binary. ; So the first filter is LP ('0') and others are HP ('1'). f 3 0 16384 "wave" 1 14 0 f 4 0 16384 "wave" 2 1 0 f 5 0 16384 "wave" 2 7 0 f 6 0 16384 "wave" 2 6 0 ; The main purpose of using wavelets is wavelet transform. ; It is not that easy to perform a classic DWT in Csound since downsampling ; of audio signal is needed at each step of wavelet decomposition. ; Anyway, using GENwave it is possible to create a number of upsampled ; wavelets and perform a so-called undecimated wavelet transform ; aka stationary wavelet transform (and it is even better). ; So we need some upsampled childs of mother wavelet. f 7 0 16 "wave" 1 0 -1 ;db2 scaling function for 1st iteration f 8 0 16 "wave" 1 1 -1 ;db2 wavelet function for 1st iteration f 9 0 32 "wave" 1 0 -1 ;db2 scaling function for 2nd iteration f 10 0 32 "wave" 1 1 -1 ;db2 wavelet function for 2nd iteration ; Let's hear how some wavelets could sound.. ; amp frq wave times i 1 0 1 0.6 15 3 8 i 1 0.5 . 0.9 20 4 5 i 1 0.9 . 0.7 8 5 . i 1 1.1 . 0.4 30 6 9 ; Now try to decompose input file using wavelets i 2 2 4 1; approximation 1st level i 2 5 . 2; details 2nd level manual_src~dfsg/examples/pvsfilter.csd0000644000000000000000000000233612262561504017300 0ustar rootroot -odac sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 giSine ftgen 0, 0, 4096, 10, 1 giBell ftgen 0, 0, 4096, 9, .56, 1, 0, .57, .67, 0, .92, 1.8, 0, .93, 1.8, 0, 1.19, 2.67, 0, 1.7, 1.67, 0, 2, 1.46, 0, 2.74, 1.33, 0, 3, 1.33, 0, 3.76, 1, 0, 4.07, 1.33, 0; bell-like (after Risset) instr 1 ipermut = p4; 1 = change order of soundfiles ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "fox.wav" ain1 soundin Sfile1 kfreq randomi 200, 300, 3 ain2 oscili .2, kfreq, giBell ;ain2 oscili .2, kfreq, giSine; try also this fftin1 pvsanal ain1, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file 1 fftin2 pvsanal ain2, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file 2 if ipermut == 1 then fcross pvsfilter fftin2, fftin1, 1 else fcross pvsfilter fftin1, fftin2, 1 endif aout pvsynth fcross out aout * 20 endin i 1 0 2.757 0; frequencies from fox.wav, amplitudes multiplied by amplitudes of giBell i 1 3 2.757 1; frequencies from giBell, amplitudes multiplied by amplitudes of fox.wav e manual_src~dfsg/examples/invalue.csd0000644000000000000000000000102012262561504016712 0ustar rootroot ;run this example in CsoundQt, a Csound editor that provides widgets ;make the Widgets-panel visible, by clicking the Widgets symbol in the menu or pressing (Alt+1). sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; written by Andres Cabrera instr 1 kfreq invalue "freq" ; Quotes are needed here asig oscil 0.1, kfreq, 1 outs asig, asig endin f 1 0 1024 10 1 ;sine i 1 0 300 ;play for 300 seconds e manual_src~dfsg/examples/cpsxpch_105et.csd0000644000000000000000000000122512262561504017644 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cpsxpch_105et.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a 10.5ET scale. ipch = 4.02 iequal = 21 irepeat = 4 ibase = 16.35160062496 icps cpsxpch ipch, iequal, irepeat, ibase print icps endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/gen05.csd0000644000000000000000000000151012262561504016171 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen05.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ;choose different tables for GEN05 kcps init 1/p3 ;index over the length of entire note kndx phasor kcps ixmode = 1 ;normalize index data kamp tablei kndx, ifn, ixmode asig poscil kamp, 440, 1 outs asig, asig endin f 1 0 8192 10 1 ;sine wave f 2 0 129 5 1 100 0.0001 29 ;short attack f 3 0 129 5 0.00001 87 1 22 .5 20 0.0001 ;long attack i 1 0 2 2 i 1 3 2 3 e manual_src~dfsg/examples/follow.csd0000644000000000000000000000153612262561504016565 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o follow.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig soundin "beats.wav" outs asig, asig endin instr 2 ;envelope follower as soundin "beats.wav" as = as*.7 ;reduce volume a bit at tone as, 500 ;smooth estimated envelope af follow at, p4 asin poscil3 .5, 440, 1 ; "beats.wav" provides amplitude for poscil asig balance asin, af outs asig, asig endin ;sine wave. f 1 0 32768 10 1 i 1 0 2 i 2 2 2 0.001 ;follow quickly i 2 5 3 0.2 ;follow slowly e manual_src~dfsg/examples/vsubv.csd0000644000000000000000000000170512262561504016426 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ifn2 = p5 ielements = p6 idstoffset = p7 isrcoffset = p8 kval init 25 vsubv ifn1, ifn2, ielements, idstoffset, isrcoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 15 16 f 2 0 16 -7 1 15 2 i2 0.0 0.2 1 i2 0.2 0.2 2 i1 0.4 0.01 1 2 5 3 8 i2 0.8 0.2 1 i1 1.0 0.01 1 2 5 10 -2 i2 1.2 0.2 1 i1 1.4 0.01 1 2 8 14 0 i2 1.6 0.2 1 i1 1.8 0.01 1 2 8 0 14 i2 2.0 0.2 1 i1 2.2 0.002 1 1 8 5 2 i2 2.4 0.2 1 e manual_src~dfsg/examples/pcauchy.csd0000644000000000000000000000142212262561504016711 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pcauchy.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values kalpha pcauchy 1000 printk .2, kalpha ; look aout oscili 0.8, 440+kalpha, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 kalpha pcauchy 1000 printk .2, kalpha ; look aout oscili 0.8, 440+kalpha, 1 ; & listen outs aout, aout endin ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/foscil.csd0000644000000000000000000000125712262561504016542 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o foscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 440 kcar = 1 kmod = p4 kndx line 0, p3, 20 ;intensivy sidebands asig foscil .5, kcps, kcar, kmod, kndx, 1 outs asig, asig endin ; sine f 1 0 16384 10 1 i 1 0 9 .01 ;vibrato i 1 10 . 1 i 1 20 . 1.414 ;gong-ish i 1 30 5 2.05 ;with "beat" e manual_src~dfsg/examples/mp3in.csd0000644000000000000000000000104612262561504016305 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mp3in.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iskptim = .3 ibufsize = 64 ar1, ar2 mp3in "beats.mp3", iskptim, 0, 0, ibufsize outs ar1, ar2 endin i 1 0 2 e manual_src~dfsg/examples/trshift.csd0000644000000000000000000000153112262561504016741 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trshift.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kpsft = p4 ain diskin2 "fox.wav", 1 fs1, fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, 0.003, 1, 3, 500 ; partial tracking fscl trshift fst, kpsft ; frequency shift aout tradsyn fscl, 1, 1, 500, 1 ; resynthesis outs aout, aout endin f1 0 8192 10 1 ;sine i 1 0 3 150 ;adds 150Hz to all tracks i 1 + 3 500 ;adds 500Hz to all tracks e manual_src~dfsg/examples/Mixer.csd0000644000000000000000000000217012262561504016342 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ; -iadc ;;;RT audio out ; For Non-realtime ouput leave only the line below: ; -o Mixer.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 katt expon 0.01, p3, 1 ;create an attack aout poscil .7, 440,1 MixerSetLevel 1, 3, katt ;impose attack on the gain level MixerSend aout, 1, 3, 0 ;send to channel 0 endin instr 2 aout vco2 .5, 110 ;saw wave MixerSetLevel 2, 3, .25 ;set level to .25 of vco2 MixerSend aout, 2, 3, 1 ;send to channel 1 endin instr 3 ;mix instr.1 and 2 with reverb kgain1 MixerGetLevel 1,3 ;get level form buss 3 kgain2 MixerGetLevel 2,3 ;get level form buss 3 a1 MixerReceive 3,0 ;receive channel 0 a2 MixerReceive 3,1 ;receive channel 1 aout = a1*kgain1+a2*kgain2 ;mix them aoutL, aoutR reverbsc aout, aout, 0.85, 12000 ;add a nice reverb outs aoutL, aoutR MixerClear endin f1 0 4096 10 1 i1 0 2 i2 0 2 i3 0 8 ;reverb stays on for 8 sec. e manual_src~dfsg/examples/port.csd0000644000000000000000000000136612262561504016250 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o port.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 aout diskin2 "fox.wav",1, 0, 1 kf,ka ptrack aout, 512 ; pitch track with winsize=1024 kcps port kf, 0.01 ; smooth freq kamp port ka, 0.01 ; smooth amp ; drive an oscillator asig poscil ampdb(kamp)*0dbfs, kcps, 1 outs asig, asig endin ; simple sine wave f 1 0 4096 10 1 i 1 0 5 e manual_src~dfsg/examples/notequal.csd0000644000000000000000000000132412262561504017106 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o notequal.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = cpspch(p4) iprint = p5 if (iprint != 1) igoto skipPrint print ipch asig vco2 .7, ipch outs asig, asig skipPrint: endin f 1 0 65536 10 1 ;sine wave i1 0 .5 8.00 0 i1 0 .5 8.01 1 ; this note will print it's ipch value and only this one will be played i1 0 .5 8.02 2 e manual_src~dfsg/examples/clockon.csd0000644000000000000000000000166012262561504016711 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o clockon.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Start clock #1. clockon 1 ; Do something that keeps Csound busy. a1 oscili 10000, 440, 1 out a1 ; Stop clock #1. clockoff 1 ; Print the time accumulated in clock #1. i1 readclock 1 print i1 endin ; Initialize the function tables. ; Table 1: an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for one second starting at 0:00. i 1 0 1 ; Play Instrument #1 for one second starting at 0:01. i 1 1 1 ; Play Instrument #1 for one second starting at 0:02. i 1 2 1 e manual_src~dfsg/examples/sfplay.csd0000644000000000000000000000170312262561504016555 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfplay.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 isf sfload "07AcousticGuitar.sf2" sfplist isf sfpassign 0, isf instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/5000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay ivel, inum, kamp*ivel, kfreq, 0 ;preset index = 0 outs a1, a2 endin f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e manual_src~dfsg/examples/moogvcf.csd0000644000000000000000000000136012262561504016716 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o moogvcf.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; iscale does not need to be set here because already 0dbfs = 1 aout vco .3, 220, 1 ; Use a nice sawtooth waveform. kfco line 200, p3, 2000 ; filter-cutoff frequency from .2 to 2 KHz krez init p4 asig moogvcf aout, kfco, krez outs asig, asig endin ;a sine wave f 1 0 16384 10 1 i 1 0 3 .1 i 1 + 3 .7 i 1 + 3 .95 e manual_src~dfsg/examples/sfinstr3m.csd0000644000000000000000000000165512262561504017215 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfinstr3m.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf sfload "07AcousticGuitar.sf2" sfilist isf instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/7000 ;scale amplitude kfreq init 1 ;do not change freq from sf aout sfinstr3m ivel, inum, kamp*ivel, kfreq, 0, gisf outs aout, aout endin f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e manual_src~dfsg/examples/spdist.csd0000644000000000000000000000251612262561504016570 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o spdist.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 4 ga1 init 0 ga2 init 0 ga3 init 0 ga4 init 0 instr 1 ;uses GEN28 file "move2", as found in /manual/examples ifreq = 1 kx init 0 ky init 0 ktime line 0, 5.6, 5.6 ;same time as in table 1 (="move2") kdist spdist 1, ktime, kx, ky kfreq = (ifreq*340) / (340 + kdist) ;calculate doppler shift printk2 kdist ;print distance values asig diskin2 "flute.aiff", kfreq, 0, 1 ;sound source is looped a1, a2, a3, a4 space asig, 1, ktime, .1, kx, ky ;use table 1 = GEN28 ar1, ar2, ar3, ar4 spsend ;send to reverb ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 a3 reverb2 ga3, 2.5, .5 a4 reverb2 ga4, 2.5, .5 outq a1, a2, a3, a4 ga1=0 ga2=0 ga3=0 ga4=0 endin f1 0 0 28 "move2" ;from left front and left rear to the middle in front i 1 0 5.6 ;same time as ktime i 99 0 10 ;keep reverb active e manual_src~dfsg/examples/OSCmidircv.csd0000644000000000000000000000365412262561504017270 0ustar rootroot ; network_recv.csd ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O sr = 44100 ksmps = 128 nchnls = 1 ; Example by Jonathan Murphy and Andres Cabrera 2007 ; Use file OSCmidisend.csd to generate OSC events for this file 0dbfs = 1 gilisten OSCinit 47120 gisin ftgen 1, 0, 16384, 10, 1 givel ftgen 2, 0, 128, -2, 0 gicc ftgen 3, 0, 128, -7, 100, 128, 100 ;Default all controllers to 100 ;Define scale tuning giji_12 ftgen 202, 0, 32, -2, 12, 2, 256, 60, 1, 16/15, 9/8, 6/5, 5/4, 4/3, 7/5, \ 3/2, 8/5, 5/3, 9/5, 15/8, 2 #define DEST #"/midi"# ; Use controller number 7 for volume #define VOL #7# turnon 1000 instr 1000 kst init 0 kch init 0 kd1 init 0 kd2 init 0 next: kk OSClisten gilisten, $DEST, "iiii", kst, kch, kd1, kd2 if (kk == 0) goto done printks "kst = %i, kch = %i, kd1 = %i, kd2 = %i\\n", \ 0, kst, kch, kd1, kd2 if (kst == 176) then ;Store controller information in a table tablew kd2, kd1, gicc endif if (kst == 144) then ;Process noteon and noteoff messages. kkey = kd1 kvel = kd2 kcps cpstun kvel, kkey, giji_12 kamp = kvel/127 if (kvel == 0) then turnoff2 1001, 4, 1 elseif (kvel > 0) then event "i", 1001, 0, -1, kcps, kamp endif endif kgoto next ;Process all events in queue done: endin instr 1001 ;Simple instrument icps init p4 kvol table $VOL, gicc ;Read MIDI volume from controller table kvol = kvol/127 aenv linsegr 0, .003, p5, 0.03, p5 * 0.5, 0.3, 0 aosc oscil aenv, icps, gisin out aosc * kvol endin f 0 3600 ;Dummy f-table e manual_src~dfsg/examples/ziw.csd0000644000000000000000000000155112262561504016071 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ziw.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple instrument. instr 1 ; Set zk variable #1 to 64.182. ziw 64.182, 1 endin ; Instrument #2 -- prints out zk variable #1. instr 2 ; Read zk variable #1 at i-rate. i1 zir 1 ; Print out the value of zk variable #1. print i1 endin ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e manual_src~dfsg/examples/vco2init.csd0000644000000000000000000000170712262561504017020 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vco2init.wav -W ;;; for file output any platform sr=44100 ksmps=1 nchnls=2 ; create waveform with discontinuities, so it has a lot of high freq content gitable ftgen 0, 0, 2^16+1, 7, -1, 2^14, 1, 0, -1, 2^14, 1, 0, -1, 2^15, 1 ; make bandlimited tables of the waveform gi_nextfree vco2init -gitable, gitable+1, 1.05, 128, 2^16, gitable gitable_bl = -gitable instr 1 kfreq expon 14000, p3, 500 kfn vco2ft kfreq, gitable_bl asig oscilikt 5000, kfreq, kfn printk 0.1, kfn ; remove semicolon on next line to hear original waveform, demonstrating the aliasing ;asig oscili 5000, kfreq, gitable outs asig, asig endin i1 0 5 e manual_src~dfsg/examples/fluidAllOut.csd0000644000000000000000000000236412262561504017507 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidAllOut.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine1 fluidEngine isfnum1 fluidLoad "sf_GMbank.sf2", giengine1, 1 fluidProgramSelect giengine1, 1, isfnum1, 0, 0 giengine2 fluidEngine ; soundfont path to manual/examples isfnum2 fluidLoad "22Bassoon.sf2", giengine2, 1 fluidProgramSelect giengine2, 1, isfnum2, 0, 70 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine1, 1, ikey, ivel endin instr 2 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine2, 1, ikey, ivel endin instr 100 imvol init 7 ;amplify a bit asigl, asigr fluidAllOut outs asigl*imvol, asigr*imvol endin i 1 0 2 60 127 ;play one note on instr 1 i 2 2 2 60 127 ;play another note on instr 2 and... i 100 0 60 ;play virtual midi keyboard e manual_src~dfsg/examples/statvar.csd0000644000000000000000000000127612262561504016750 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 ;;;RT audio I/O, midi in ; For Non-realtime ouput leave only the line below: ; -o statevar.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 kenv linseg 0,0.1,1, p3-0.2,1, 0.1, 0 asig buzz 16000*kenv, 100, 100, 1 kf expseg 100, p3/2, 5000, p3/2, 1000 ahp,alp,abp,abr statevar asig, kf, 200 outs alp,ahp ; lowpass left, highpass right endin ; Table #1, a sine wave. f 1 0 16384 10 1 i1 0 2 manual_src~dfsg/examples/envlpxr.csd0000644000000000000000000000124712262561504016760 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o envlpxr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .5 kenv envlpxr iamp, 0.2, 1, 1, 1, .01 asig pluck kenv, icps, 200, 2, 1 outs asig, asig endin f 1 0 129 -7 0 128 1 f 2 0 4096 10 1 f0 30 ;runs 30 seconds e manual_src~dfsg/examples/gen07.csd0000644000000000000000000000171512262561504016202 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen07.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;use GEN07 to alter frequency ifn = p4 ;use different GEN07 tables kcps init 10/p3 ;index ftable 10 times over the duration of entire note kndx phasor kcps ixmode = 1 ;normalize index data kfrq tablei kndx, ifn, ixmode kfrq = kfrq*1000 ;scale asig poscil .8, 1220+kfrq, 1 ;add to frequency outs asig, asig endin f 1 0 8192 10 1 ;sine wave f 2 0 1024 7 0 512 1 0 -1 512 0 ;sawtooth up and down f 3 0 1024 7 1 512 1 0 -1 512 -1 ;square f 4 0 1024 7 1 1024 -1 ;saw down i 1 0 2 2 i 1 + 2 3 i 1 + 1 4 e manual_src~dfsg/examples/ftfree.csd0000644000000000000000000000116712262561504016536 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftfree.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gitempTable ftgen 0, 0, 65537, 10, 1 instr 1 aout oscili .5, 440, gitempTable outs aout, aout ;free temp table at deinit time ftfree gitempTable, 1 print gitempTable endin f 0 5 i 1 0 .1 i 1 3 1 e manual_src~dfsg/examples/imageopcodesdemo2.csd0000644000000000000000000000267612262561504020657 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o imageopcodesdemo2.wav -W ;;; for file output any platform sr = 48000 ksmps = 100 nchnls = 2 ;By Cesare Marilungo 2008 zakinit 10,1 ;Load the image - should be 512x512 pixels giimage imageload "test1.png" ;giimage imageload "test2.png" ;--try this too giimagew, giimageh imagesize giimage giwave ftgen 1, 0, 1024, 10, 1 gifrqs ftgen 2,0,512,-5, 1,512,10 giamps ftgen 3, 0, 512, 10, 1 instr 100 kindex = 0 icnt = giimageh kx_ linseg 0, p3, 1 kenv linseg 0, .2, 500, p3 - .4, 500, .2, 0 ; Read a column of pixels and store the red values ; inside the table 'giamps' loop: ky_ = kindex/giimageh ;Get the pixel color values at kx_, ky_ kred, kgreen, kblue imagegetpixel giimage, kx_, ky_ ;Write the red values inside the table 'giamps' tablew kred, kindex, giamps kindex = kindex+1 if (kindex < icnt) kgoto loop ; Use an oscillator bank (additive synthesis) to generate sound ; setting amplitudes for each partial according to the image asig adsynt kenv, 220, giwave, gifrqs, giamps, icnt, 2 outs asig, asig endin instr 101 ; Free memory used by the image imagefree giimage endin t 0 60 i100 1 20 i101 21 1 e manual_src~dfsg/examples/qnan.csd0000644000000000000000000000147412262561504016221 0ustar rootroot -+rtaudio=alsa -o dac:hw:0 nchnls = 2 ksmps = 400 #define WII_B #3# #define WII_A #4# #define WII_R_A #304# #define WII_PITCH #20# #define WII_ROLL #21# gkcnt init 1 instr 1 i1 wiiconnect 3,1 wiirange $WII_PITCH., -20, 0 kt wiidata $WII_B. ka wiidata $WII_A. kra wiidata $WII_R_A. gka wiidata $WII_PITCH. gkp wiidata $WII_ROLL. ; If the B (trigger) button is pressed then activate a note if (kt==0) goto ee if (qnan(gka)) goto ee if (qnan(gkp)) goto ee event "i", 2, 0, 5 gkcnt = gkcnt + 1 printk2 kb endin instr 2 a1 oscil ampdbfs(gka), 440+gkp, 1 outs a1, a1 endin f1 0 4096 10 1 i1 0 300 manual_src~dfsg/examples/tablemix.csd0000644000000000000000000000144412262561504017066 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tablemix.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisinoid ftgen 1, 0, 256, 10, 1, 0, 0, .4 ;sinoid gisaw ftgen 2, 0, 1024, 7, 0, 256, 1 ;saw gimix ftgen 100, 0, 256, 7, 0, 256, 1 ;destination table instr 1 kgain linseg 0, p3*.5, .5, p3*.5, 0 tablemix 100, 0, 256, 1, 0, 1, 2, 0, kgain asig poscil .5, 110, gimix ;mix table 1 & 2 outs asig, asig endin i1 0 10 e manual_src~dfsg/examples/lposcila.csd0000644000000000000000000000163312262561504017067 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lposcila.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 1.3 ;a 3d up kloop = 0 ;loop start time in samples kend = 10000 ;loop end time in samples aenv expsega 0.01, 0.1, 1, 0.1, 0.5, 0.5, 0.01 ;envelope with fast and short segment asig lposcila aenv, kcps, kloop, kend, 1 ;use it for amplitude outs asig, asig endin ; Its table size is deferred, ; and format taken from the soundfile header. f 1 0 0 1 "beats.wav" 0 0 0 ; Play Instrument #1 for 6 seconds. ; This will loop the drum pattern several times. i 1 0 2 e manual_src~dfsg/examples/fluidProgramSelect.csd0000644000000000000000000000166712262561504021063 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidProgramSelect.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "19Trumpet.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 56 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 7 asigl, asigr fluidOut giengine outs asigl*imvol, asigr*imvol endin i 1 0 2 60 100 ;play one note from score and... i 99 0 60 ;play virtual keyboard for 60 sec. e manual_src~dfsg/examples/pitchamdf.csd0000644000000000000000000000155312262561504017221 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pitchamdf.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;clean audio asig soundin p4 outs asig, asig endin instr 2 ;use pitch asig soundin p4 asig tone asig, 1000 ;lowpass-filter kcps, krms pitchamdf asig, 100, 500, 200 asig poscil krms, kcps, 1 ;re-synthesize with sawtooth outs asig, asig endin f1 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ;sawtooth i 1 0 3 "fox.wav" i 2 3 3 "fox.wav" i 1 6 3 "mary.wav" i 2 9 3 "mary.wav" i 1 12 3 "beats.wav" i 2 15 3 "beats.wav" e manual_src~dfsg/examples/midinoteonoct.csd0000644000000000000000000000203512262561504020131 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midinoteonoct.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 midinoteonoct p4, p5 ;gets a MIDI note number value as octave-point-decimal value into p4, and MIDI velocity into p5 print p4 ;display the key value when it changes and when key is pressed kvel = p5/127 ;scale midi velocity to 0-1 ioct = p4 icps = cpsoct(ioct) ;convert octave-point-decimal value into Hz kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by asig pluck kenv*kvel, icps, icps, 2, 1 ;velocity value outs asig, asig endin f 0 30 ;runs for 30 seconds f 2 0 4096 10 1 i 1 0 2 8.000 100 ; play these notes from score as well i 1 + 2 8.917 100 e manual_src~dfsg/examples/wgflute.csd0000644000000000000000000000130412262561504016731 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o wgflute.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 440 kjet init p4 ;vary air jet iatt = 0.1 idetk = 0.1 kngain = 0.15 kvibf = 5.925 kvamp = 0.05 asig wgflute .8, kfreq, kjet, iatt, idetk, kngain, kvibf, kvamp, 1 outs asig, asig endin f 1 0 16384 10 1 ;sine wave i 1 0 2 0.02 ;more air jet i 1 + 2 0.32 e manual_src~dfsg/examples/tradsyn.csd0000644000000000000000000000131112262561504016736 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tradsyn.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = p4 ain diskin2 "beats.wav", 1 fs1,fsi2 pvsifd ain,2048,512,1 ; ifd analysis fst partials fs1,fsi2,.003,1,3,500 ; partial tracking aout tradsyn fst, 1, ipch, 500, 1 ; resynthesis outs aout, aout endin f1 0 8192 10 1 i 1 0 2 1.5 ;up a 5th i 1 + 2 .5 ;octave down e manual_src~dfsg/examples/ampdb.csd0000644000000000000000000000110112262561504016332 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ampdb.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 idb = p4 iamp = ampdb(idb) asig oscil iamp, 220, 1 print iamp outs asig, asig endin ;sine wave. f 1 0 16384 10 1 i 1 0 1 50 i 1 + 1 90 i 1 + 1 68 i 1 + 1 80 e manual_src~dfsg/examples/mxadsr.csd0000644000000000000000000000121112262561504016547 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mxadsr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .5 kenv mxadsr 0.5, 0, 1, 0.5 asig pluck kenv, icps, icps, 2, 1 outs asig, asig endin f 2 0 4096 10 1 f0 30 ;runs 30 seconds e manual_src~dfsg/examples/divz.csd0000644000000000000000000000151212262561504016231 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o divz.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Define the numbers to be divided. ka init 200 ; Linearly change the value of kb from 200 to 0. kb line 0, p3, 200 ; If a "divide by zero" error occurs, substitute -1. ksubst init -1 ; Safely divide the numbers. kresults divz ka, kb, ksubst ; Print out the results. printks "%f / %f = %f\\n", 0.1, ka, kb, kresults endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/outipc.csd0000644000000000000000000000111412262561504016556 0ustar rootroot ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out and midi in and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outipc.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 outipc 1, 80, 0, 127 ;program change --> 80 ikey notnum ivel veloc midion 1, ikey, ivel ;play external synth endin f0 30 ;runs 30 seconds e manual_src~dfsg/examples/zkw.csd0000644000000000000000000000225412262561504016074 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zkw.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Linearly vary a k-rate signal from 100 to 1,000. kline line 100, p3, 1000 ; Add the linear signal to zk variable #1. zkw kline, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read zk variable #1. kfreq zkr 1 ; Use the value of zk variable #1 to vary ; the frequency of a sine waveform. a1 oscil 20000, kfreq, 1 ; Generate the audio output. out a1 ; Clear the zk variables, get them ready for ; another pass. zkcl 0, 1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 0 2 e manual_src~dfsg/examples/zar.csd0000644000000000000000000000204412262561504016052 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zar.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Generate a simple sine waveform. asin oscil 20000, 440, 1 ; Send the sine waveform to za variable #1. zaw asin, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read za variable #1. a1 zar 1 ; Generate audio output. out a1 ; Clear the za variables, get them ready for ; another pass. zacl 0, 1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e manual_src~dfsg/examples/array_scalar_math.csd0000644000000000000000000000270412262561504020735 0ustar rootroot -n -m128 instr 1 ;create array and fill with numbers 1..10 kArr1[] fillarray 1, 2, 3, 4, 5, 6, 7, 8, 9, 10 ;print content printf "%s", 1, "\nInitial content:\n" kndx = 0 until kndx == lenarray(kArr1) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr1[kndx] kndx += 1 od ;add 10 kArr2[] = kArr1 + 10 ;print content printf "%s", 1, "\nAfter adding 10:\n" kndx = 0 until kndx == lenarray(kArr2) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr2[kndx] kndx += 1 od ;subtract 5 kArr3[] = kArr2 - 5 ;print content printf "%s", 1, "\nAfter subtracting 5:\n" kndx = 0 until kndx == lenarray(kArr3) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr3[kndx] kndx += 1 od ;multiply by -1.5 kArr4[] = kArr3 * -1.5 ;print content printf "%s", 1, "\nAfter multiplying by -1.5:\n" kndx = 0 until kndx == lenarray(kArr4) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr4[kndx] kndx += 1 od ;divide by -3/2 kArr5[] = kArr4 / -(3/2) ;print content printf "%s", 1, "\nAfter dividing by -3/2:\n" kndx = 0 until kndx == lenarray(kArr5) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr5[kndx] kndx += 1 od ;turnoff turnoff endin i 1 0 .1 manual_src~dfsg/examples/gen30.csd0000644000000000000000000000322612262561504016175 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac ;;;realtime audio out ; For Non-realtime ouput leave only the line below: ; -o gen30.wav -W ;;; for file output any platform ;a simplified example of Istvan Varga sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 isaw ftgen 1, 0, 16384, 7, 1, 16384, -1 ;sawtooth wave iFM ftgen 3, 0, 4096, 7, 0, 512, 0.25, 512, 1, 512, 0.25, 512, \ 0, 512, -0.25, 512, -1, 512, -0.25, 512, 0 ;FM waveform iAM ftgen 4, 0, 4096, 5, 1, 4096, 0.01 ;AM waveform iEQ ftgen 5, 0, 1024, 5, 1, 512, 32, 512, 1 ;FM to EQ isine ftgen 6, 0, 1024, 10, 1 ;sine wave /* generate bandlimited sawtooth waves */ i0 = 0 loop1: imaxh = sr / (2 * 440.0 * exp(log(2.0) * (i0 - 69) / 12)) i_ ftgen i0 + 10, 0, 4096, -30, 1, 1, imaxh ;use gen 30 i0 = i0 + 1 if (i0 < 127.5) igoto loop1 instr 1 kcps = 440.0 * exp(log(2.0) * (p4 - 69) / 12) ;note frequency klpmaxf limit p5 * kcps, 1000.0, 12000.0 ;lowpass max. frequency kfmd1 = 0.03 * kcps ;FM depth in Hz kamfr = kcps * 0.02 ;AM frequency kamfr2 = kcps * 0.1 kfnum = (10 + 69 + 0.5 + 12 * log(kcps / 440.0) / log(2.0)) ;table number aenv linseg 0, p3*0.25, 1, p3*0.75, 0 ;amp. envelope asig oscbnk kcps, 0.0, kfmd1, 0.0, 40, 200, 0.1, 0.2, 0, 0, 144, \ 0.0, klpmaxf, 0.0, 0.0, 1.5, 1.5, 2, kfnum, 3, 0, 5, 5, 5 asig = asig * aenv*.03 outs asig, asig endin s i 1 0 6 41 10 i 1 0 6 60 i 1 0 6 65 i 1 0 6 69 s i 1 0 6 41 64 i 1 0 6 60 i 1 0 6 65 i 1 0 6 69 e manual_src~dfsg/examples/gen42.csd0000644000000000000000000000206212262561504016175 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o GEN42.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktab = 1 ;ftable 1 kurd = urd(ktab) ktrig metro 5 ;triggers 5 times per second kres samphold kurd, ktrig ;sample and hold value of kurd printk2 kres ;print it asig poscil .5, 220+kres, 2 outs asig, asig endin instr 2 seed 0 ;every run new values ktab = 1 ;ftable 1 kurd = urd(ktab) ktrig metro 5 ;triggers 5 times per second kres samphold kurd, ktrig ;sample and hold value of kurd printk2 kres ;print it asig poscil .5, 220+kres, 2 outs asig, asig endin f1 0 -20 -42 10 20 .3 100 200 .7 ;30% choose between 10 and 20 and 70% between 100 and 200 f2 0 8192 10 1 ;sine wave i 1 0 5 i 2 6 5 e manual_src~dfsg/examples/fox.wav0000644000000000000000000073275612262561504016122 0ustar rootrootRIFFWAVEfmt DXdataµ --7-*'+*++0*96,/%',$#&++756+('/,-*@%#(0::GTLUFOPadppty}syowhlr`mffTZWWVKLaXL;0$ )*(!$*#$28#.8+(1$%2/55GLF<0&+638L>3-' -F(+. =42:*"  &( $ ';),7' 2+%B%6-'6A5817+ ,50.%# -(4 (!)"5'     {t %(   ""&*,2 !!'#-##$8.1C #% $    ',#5HV4,:"+ *# "-+$V")*+ 2$2">8&2<26E94Y9lQ.7<2G/-9-S=l#V '4Y0#P!7!  1/3  L#&"4z +^mUtt&Ywu4s09O., 1;6%m _W874iVR<Iu89}=-Y4v&%MG BvU]o/";ej*#Sy=Kk'M]7(<YioDAMBf"n'pj*U<R>!ekD_eyfb/mS'jN8FTn-gMP(XdtY}~ff#[6 >(u6CUs3MY\JM&v:BoyNh)o?A.H6~MYJ\vRO0Kx`xh_ shc[1hskn Gg9V"\:E\QpG(];3N9<02C`!WZCZ1lD v.q#\y 2%Y'7V~q=doW@K3wVxQ]Ke(a^i+c/TLIJ ;ff\,Z}G|adDlX(xP()_MK9|l8kCCJ2:]XlCWeE ] X$6  r*Zhy+X#Wht.naibh.93"P`_T9_|>I|3\U3nXe=#YQL9\2X~ap)>\b?#JVD%/w{ {K{T$8vy)Q`XXha yo)m#(!8%"'zp+$.d;(tzI[Hg\# _ "  \ n ( s Y hUI [  $ r } V N * v W  w  O8-)\H'=@|.s.4ktiLR]=B 3H)A?vt(Mg^uEW-y~p @ UCl-TBagIo~c,li9AMNq#Kpsa|)-*y+t0"kET2)$8~U }dI[uY\ l503J"T?1 ML rGi>hE01HY+brJ e%x`MYX<o/7U#,i E&K};+B5H;$wf }BL<xo>ClU 2G|y P=Lxqru?uagj5P@qx=-g*>#M.u "fTD)A,^b 9g5G5flIG\hF$nK l3wuS28p=a`lR_\UQ%q8(@|CKf#/ -jx#?j2Q'Am2tx&Qy^b+hgPk*m(/ kJknmmiVAL("HK.,ZZLH[Qkw]NrrN($!Q)g3<"VM3g5,@).?@98GI[teOdV+2H$Puvw @HQN00GD;OPa|}yzhxeDHUQEBSdZV_oo!LNW| ,BA& 0_Qd{y{ | 1dZVz`]ZS]xokpx}b3.>oH(I&{>oeS5PRC{F =<^^[X_SGq`IH FheSL9 T .-Rz8T{}tL^p^B@mx[R&[GX`k '`UC.F]awB37<>{9$%E/_F/- CHF-t^T/* )2%3Nfx~wtcMC+:=18KHQ\Za {/1|~jhUHGBLHYQ[VQbzx|ynWRR@UZcer ,)4*-/<KJSWGJH>GHFG<=A>P>;?1/*   *"  }uzy   (.2,  $6QNFR??<IEJJUB(",($(,$1%0+/=<GfiwtuUZc}~xmgV^f[VXYMNHNVS`lYSXR^TXQS`eun[M>7<AMMYQK[K?9%3!*5528* #6PYMTZGKOQ\NM<.)" 7L^\ZMTCJcy~mtwktkitt||kjaMhhToudONFPNHZadVJY[aY\\TJMEQ=XT<?50@)6*$#*&-!/.;)   ,6%"+!$+ $ $$! >4. .ENJ\6/+2&=6!  .1;3)/) '94RJF>%(+??O?R@-5%;8:/.9*8/B590*"            % -30#%    >81 (EHIM3"+$01,<E/131668JN7@'' &CJWX\mdoluT^^FE: "% /2AAA;$$ * )82%$$$&20;15BMLNED;?NUED?E:--$9B8*   $ #  "0.,5FXNVRZ_h``TOX\ebspXJOSY[_c[==*(06&     ). %.58$CCGN@;3( '2*-0)#&57+DB>B:*%"#-"#" $%--2%    ##! 32@F85/2PVp4]=|q'   v6.UW1(}q(mp[mJ2Uf[W ? b 1 EIT6Xac:Q)p!CR-e6AevMfk%8]o;xH{: bSdZ3$RZ{1jii  (@X\L?7;Y3M6Ukgss oZ-0k+rb"quJB0h F=O2i;P0GW.6 by>XWGG/pn^VlAZ: Y=7o=!B50[ KK{EV, wUY. )yl }F\H=&Hatp6W'G hiNg-lWrZMt>\$FbzVu l -   $ y5 \  ytm 78'4 ~X6\  T\1r8D6hZ1gxp( _ a w v &vJ)Edvk[aZ<-Ov N$:IyhA%W{yIAI(+!a<3oS1$upbaA'@6!`!x1Q<|Jn[SJ05[p s#IAE2Y0 61 Fu!R %)p# )dyt<AqR3$Ll*iL54aQO@  2ym - 1sk}6Nk)8ja)Cw.!n\S),R9JB?^H-O<,fwg |M , } ^ ; 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For Non-realtime ouput leave only the line below: ; -o fog.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;p4 = transposition factor ;p5 = speed factor ;p6 = function table for grain data i1 = sr/ftlen(1) ;scaling to reflect sample rate and table length a1 phasor i1*p5 ;index for speed asigl fog .5, 15, p4, a1, 1, 0, .01, .5, .01, 30, 1, 2, p3 ;left channel asigr fog .4, 25, p4+.2, a1, 1, 0, .01, .5, .01, 30, 1, 2, p3, .5 ;right channel outs asigl, asigr endin f 1 0 131072 1 "fox.wav" 0 0 0 f 2 0 1024 19 .5 .5 270 .5 i 1 0 10 .7 .1 i 1 + 4 1.2 2 e manual_src~dfsg/examples/alpass.csd0000644000000000000000000000140312262561504016537 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o alpass.wav -W ;;; for file output any platform sr = 44100 ksmps = 100 nchnls = 2 0dbfs = 1 gamix init 0 instr 1 acps expon p4, p3, p5 asig vco 0.6, acps, 1 outs asig, asig gamix = gamix + asig endin instr 99 arvt1 line 3.5*1.5, p3, 6 arvt2 line 3.5, p3, 4 ilpt = 0.1 aleft alpass gamix, arvt1, ilpt aright alpass gamix, arvt2, ilpt*2 outs aleft, aright gamix = 0 ; clear mixer endin f1 0 4096 10 1 i 1 0 3 20 2000 i 99 0 8 e manual_src~dfsg/examples/reverb.csd0000644000000000000000000000145712262561504016552 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o reverb.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 ga1 init 0 instr 1 asig poscil .2, cpspch(p4), 1 outs asig, asig ga1 += asig ;add direct signal to global reverb endin instr 99 ;(highest instr number executed last) arev reverb ga1, 1.5 outs arev, arev ga1 = 0 ;clear endin f 1 0 128 10 1 ;sine i 1 0 0.1 7.00 ;short sounds i 1 1 0.1 8.02 i 1 2 0.1 8.04 i 1 3 0.1 8.06 i 99 0 6 ;reverb runs for 6 seconds e manual_src~dfsg/examples/xscanu.csd0000644000000000000000000000502712262561504016563 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o xscanu.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;the matrices can be found in /manual/examples instr 1 ; Plain scanned syntnesis ; Note Also that I am using quadratic interpolation on these. a0 = 0 xscanu 1, .01, 6, 2, "128,8-gridX", 4, 5, 2, .1, .1, -.01, .1, .5, 0, 0, a0, 0, 0 a1 xscans .5, cpspch(p4), 333, 0, p6 ; NOTE LEFT RIGHT TRAJECTORY (f333) IS CLEAN! a1 dcblock a1 outs a1, a1 endin instr 2 ; Scan synthesis with audio injection and dual scan paths a0 diskin2 "fox.wav",1,0,1 ; a0,aa ins a0 = a0/.8 xscanu 1, .01, 6, 2, "128,8-torusX", 14, 5, 2, .01, .05, -.05, .1, .5, 0, 0, a0, 0, 0 a1 xscans .3, cpspch(7.00), 333, 0, 2 ; NOTE LEFT RIGHT TRAJECTORY (f333) IS CLEAN! a2 xscans .3, cpspch(6.00), 77, 0, 2 a1 dcblock a1 a2 dcblock a2 outs a1*.5,a2*.1 endin ; Initial condition ;f1 0 16 7 0 8 1 8 0 f1 0 128 7 0 64 1 64 0 ; Masses f2 0 128 -7 1 128 1 ; Centering force f4 0 128 -7 0 128 2 f14 0 128 -7 2 64 0 64 2 ; Damping f5 0 128 -7 1 128 1 ; Initial velocity f6 0 128 -7 -.0 128 .0 ; Trajectories f7 0 128 -5 .001 128 128 f77 0 128 -23 "128-spiral-8,16,128,2,1over2" f777 0 128 -23 "128,8-torusX" ; Spring matrices f3 0 128 -23 "128-stringX" f33 0 128 -23 "128-stringcircularX" f333 0 128 -23 "128-left_rightX" f3333 0 128 -23 "128,8-torusX" f33333 0 128 -23 "128,8-cylinderX" f333333 0 128 -23 "128,8-gridX" ; Sine f9 0 1024 10 1 ; Pitch tables f100 0 1024 -7 +3 128 +3 128 -2 128 -2 128 +0 128 +0 128 -4 128 -4 128 +3 f101 0 1024 -7 -2 128 -2 128 -2 128 -2 128 -5 128 -5 128 -4 128 -4 128 -2 f102 0 1024 -7 +3 128 +3 128 +2 128 +2 128 +0 128 +0 128 +0 128 +0 128 +3 f103 0 1024 -7 +7 128 +7 128 +5 128 +5 128 +3 128 +3 128 +3 128 +3 128 +7 ; Amplitude tables f200 0 1024 7 1 128 0 128 0 127 0 1 1 128 0 128 0 127 0 1 1 128 0 127 0 1 1 f201 0 1024 7 0 127 0 1 1 127 0 1 1 128 0 127 0 1 1 127 0 1 1 128 0 127 0 1 1 127 0 1 1 f202 0 1024 7 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 f203 0 1024 7 1 1024 0 ;;f204 0 1024 7 1 512 0 511 0 1 1 ;----------------------- ; Note list i1 0 10 6.00 1 2 s i2 0 15 e manual_src~dfsg/examples/gen34.csd0000644000000000000000000000235312262561504016201 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen34.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; partials 1, 4, 7, 10, 13, 16, etc. with base frequency of 400 Hz ibsfrq = 400 inumh = int(1.5 + sr * 0.5 / (3 * ibsfrq)) ; estimate number of partials isrcln = int(0.5 + exp(log(2) * int(1.01 + log(inumh * 3) / log(2)))) ; source table length itmp ftgen 1, 0, isrcln, -2, 0 ; create empty source table ifpos = 0 ifrq = ibsfrq inumh = 0 l1: tableiw ibsfrq / ifrq, ifpos, 1 ; amplitude tableiw ifrq, ifpos + 1, 1 ; frequency tableiw 0, ifpos + 2, 1 ; phase ifpos = ifpos + 3 ifrq = ifrq + ibsfrq * 3 inumh = inumh + 1 if (ifrq < (sr * 0.5)) igoto l1 itmp ftgen 2, 0, 262144, -34, 1, inumh, 1, -1 ; store output in ftable 2 (size = 262144) asig poscil .5, ibsfrq, itmp outs asig, asig endin i 1 0 2 e manual_src~dfsg/examples/pchmidi.csd0000644000000000000000000000122012262561504016666 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o pchmidi.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; This example expects MIDI note inputs on channel 1 i1 pchmidi print i1 endin ;Dummy f-table to give time for real-time MIDI events f 0 8000 e manual_src~dfsg/examples/outkpc.csd0000644000000000000000000000132512262561504016564 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1 ;;;RT audio I/O with MIDI in sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 kprogram init 0 instr 1 ;Triggered by MIDI notes on channel 1 ifund notnum ivel veloc idur = 1 ; Sends a MIDI program change message according to ; the triggering note's velocity outkpc 1 ,ivel ,0 ,127 noteondur 1 ,ifund ,ivel ,idur endin ; Dummy ftable f 0 60 manual_src~dfsg/examples/gbuzz.csd0000644000000000000000000000152212262561504016417 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gbuzz.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 instr 1 kcps = 220 knh = p4 ;total no. of harmonics klh = p5 ;lowest harmonic kmul line 0, p3, 1 ;increase amplitude of ;higer partials asig gbuzz .6, kcps, knh, klh, kmul, 1 outs asig, asig endin ; a cosine wave f 1 0 16384 11 1 i 1 0 3 3 1 ;3 harmonics, lowest harmonic=1 i 1 + 3 30 1 ;30 harmonics, lowest harmonic=1 i 1 + 3 3 2 ;3 harmonics, lowest harmonic=3 i 1 + 3 30 2 ;30 harmonics, lowest harmonic=3 e manual_src~dfsg/examples/octcps.csd0000644000000000000000000000115212262561504016550 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o octcps.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Convert a cycles-per-second value into an ; octave value. icps = 440 ioct = octcps(icps) print ioct endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/ckgoto.csd0000644000000000000000000000202712262561504016545 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ckgoto.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Change kval linearly from 0 to 2 over ; the period set by the third p-field. kval line 0, p3, 2 ; If kval is greater than or equal to 1 then play the high note. ; If not then play the low note. ckgoto (kval >= 1), highnote kgoto lownote highnote: kfreq = 880 goto playit lownote: kfreq = 440 goto playit playit: ; Print the values of kval and kfreq. printks "kval = %f, kfreq = %f\\n", 1, kval, kfreq a1 oscil 10000, kfreq, 1 out a1 endin ; Table: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/noise-2.csd0000644000000000000000000000130712262561504016533 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac ; -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o noise.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 FLpanel "noise", 200, 50, -1 , -1 gkbeta, gislider1 FLslider "kbeta", -1, 1, 0, 5, -1, 180, 20, 10, 10 FLpanelEnd FLrun instr 1 iamp = 0dbfs / 4 ; Peaks 12 dB below 0dbfs print iamp a1 noise iamp, gkbeta printk2 gkbeta outs a1,a1 endin ; Play Instrument #1 for one minute. i 1 0 60 e manual_src~dfsg/examples/outq3.csd0000644000000000000000000000122312262561504016327 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outq3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 asig vco2 .05, 30 ; sawtooth waveform at low volume kcut line 30, p3, 100 ; Vary cutoff frequency kresonance = 7 inumlayer = 2 asig lowresx asig, kcut, kresonance, inumlayer outq3 asig ; output channel 3 endin i 1 0 3 e manual_src~dfsg/examples/vrandh.csd0000644000000000000000000000223212262561504016537 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vranh.wav -W ;;; for file output any platform ;Example by Andres Cabrera sr=44100 ksmps=128 nchnls=2 gitab ftgen 0, 0, 16, -7, 0, 128, 0 instr 1 krange init p4 kcps init p5 ioffset init p6 kav1 init 0 kav2 init 0 kcount init 0 ; table krange kcps ielements idstoffset iseed isize ioffset vrandh gitab, krange, kcps, 3, 3, 2, 0, ioffset kfreq1 table 3, gitab kfreq2 table 4, gitab kfreq3 table 5, gitab ;Change the frequency of three oscillators according to the random values aosc1 oscili 4000, kfreq1, 1 aosc2 oscili 2000, kfreq2, 1 aosc3 oscili 4000, kfreq3, 1 outs aosc1+aosc2, aosc3+aosc2 endin f 1 0 1024 10 1 ; krange kcps ioffset i 1 0 5 100 1 300 i 1 5 5 300 1 400 i 1 10 5 100 2 1000 i 1 15 5 400 4 1000 i 1 20 5 1000 8 2000 i 1 25 5 250 16 300 e manual_src~dfsg/examples/cpstmid.csd0000644000000000000000000000166012262561504016724 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M0 ;;;RT audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpstmid.wav -W ;;; for file output any platform ; after an example from Kevin Conder sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; Table #1, a normal 12-tone equal temperament scale. ; numgrades = 12 (twelve tones) ; interval = 2 (one octave) ; basefreq = 261.659 (Middle C) ; basekeymidi = 60 (Middle C) gitemp ftgen 1, 0, 64, -2, 12, 2, 261.659, 60, 1.00, \ 1.059, 1.122, 1.189, 1.260, 1.335, 1.414, \ 1.498, 1.588, 1.682, 1.782, 1.888, 2.000 instr 1 ifn = 1 icps cpstmid ifn print icps asig oscil 0.6, icps, 2 outs asig, asig endin f 0 20 ;sine wave. f 2 0 16384 10 1 e manual_src~dfsg/examples/filter2.csd0000644000000000000000000000170112262561504016624 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o filter2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; 2 saw waves of which one is slightly detuned ib1 = p5 ivol = p6 ;volume to compensate kcps init cpspch(p4) asig1 vco2 .05, kcps ;saw 1 asaw1 filter2 asig1, 1, 1, 1, ib1 ;filter 1 asig2 vco2 .05, kcps+1 ;saw 2 asaw2 filter2 asig2, 1, 1, 1, ib1 ;filter 2 aout = (asaw1+asaw2)*ivol ;mix outs aout, aout endin i 1 0 4 6.00 -.001 5 ;different filter values i 1 + 4 6.00 -.6 2 ;and different volumes i 1 + 4 6.00 -.95 .3 ;to compensate e manual_src~dfsg/examples/qinf.csd0000644000000000000000000000147412262561504016221 0ustar rootroot -+rtaudio=alsa -o dac:hw:0 nchnls = 2 ksmps = 400 #define WII_B #3# #define WII_A #4# #define WII_R_A #304# #define WII_PITCH #20# #define WII_ROLL #21# gkcnt init 1 instr 1 i1 wiiconnect 3,1 wiirange $WII_PITCH., -20, 0 kt wiidata $WII_B. ka wiidata $WII_A. kra wiidata $WII_R_A. gka wiidata $WII_PITCH. gkp wiidata $WII_ROLL. ; If the B (trigger) button is pressed then activate a note if (kt==0) goto ee if (qinf(gka)) goto ee if (qinf(gkp)) goto ee event "i", 2, 0, 5 gkcnt = gkcnt + 1 printk2 kb endin instr 2 a1 oscil ampdbfs(gka), 440+gkp, 1 outs a1, a1 endin f1 0 4096 10 1 i1 0 300 manual_src~dfsg/examples/trandom.csd0000644000000000000000000000142612262561504016725 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trandom.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 ; every run time different values instr 1 kmin init 0 ;random number between 0 and 220 kmax init 220 ktrig = p4 k1 trandom ktrig, kmin, kmax printk2 k1 ;print when k1 changes asig poscil .4, 220+k1, 1 ;if triggered, add random values to frequency outs asig, asig endin f1 0 4096 10 1 i 1 0 2 0 ;not triggered i 1 + 2 1 ;triggered e manual_src~dfsg/examples/gendy-2.csd0000644000000000000000000000143712262561504016530 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kenv expseg 0.01, p3*0.1, 1, p3*0.8, 1, p3*0.1, 0.001 kosc oscil 0.1, 3/p3, 1 seed 20120123 kdis bexprnd kosc knum linseg 3, p3*0.75, 10, p3*0.20, 12, p3*0.05, 5 asig gendy 0.2, kosc*60, 6, 0.7, kdis, 500*kenv, 4800, 0.23, 0.3, 12, knum aflt resonz asig, 1400, 400 aout comb kenv*aflt*0.1, 0.9, 0.1 outs aout, aout endin f1 0 8192 10 1 0 .8 0 0 .3 0 0 0 .1 i1 0 20 e manual_src~dfsg/examples/greaterthan.csd0000644000000000000000000000145312262561504017565 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o greaterthan.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Get the 4th p-field from the score. k1 = p4 ; Is it greater than 3? (1 = true, 0 = false) k2 = (p4 > 3 ? 1 : 0) ; Print the values of k1 and k2. printks "k1 = %f, k2 = %f\\n", 1, k1, k2 endin ; Call Instrument #1 with a p4 = 2. i 1 0 0.5 2 ; Call Instrument #1 with a p4 = 3. i 1 1 0.5 3 ; Call Instrument #1 with a p4 = 4. i 1 2 0.5 4 e manual_src~dfsg/examples/FLxyin.csd0000644000000000000000000000103312262561504016464 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O sr=48000 ksmps=128 nchnls=2 ; Example by Andres Cabrera 2007 FLpanel "FLxyin", 200, 100, -1, -1, 3 FLpanelEnd FLrun instr 1 koutx, kouty, kinside FLxyin 0, 10, 100, 1000, 10, 190, 10, 90 aout buzz 10000, kouty, koutx, 1 printk2 koutx outs aout, aout endin f 1 0 1024 10 1 i 1 0 3600 e manual_src~dfsg/examples/logbtwo.csd0000644000000000000000000000102612262561504016732 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o logbtwo.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = logbtwo(3) print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/cpsmidib.csd0000644000000000000000000000112012262561504017042 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M0 ;;;RT audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpsmidi.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; move pitch bend wheel while you play kcps cpsmidib asig oscil 0.6, kcps, 1 printk2 kcps outs asig, asig endin f0 20 ;sine wave. f 1 0 16384 10 1 e manual_src~dfsg/examples/syncphasor.csd0000644000000000000000000000236412262561504017454 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o abs.wav -W ;;; for file output any platform instr 1 ; Use two syncphasors - one is the "master", ; the other the "slave" ; master's frequency determines pitch imastercps = cpspch(p4) imaxamp = 10000 ; the slave's frequency affects the timbre kslavecps line imastercps, p3, imastercps * 3 ; the master "oscillator" ; the master has no sync input anosync init 0.0 am, async syncphasor imastercps, anosync ; the slave "oscillator" aout, as syncphasor kslavecps, async adeclick linseg 0.0, 0.05, 1.0, p3 - 0.1, 1.0, 0.05, 0.0 ; Output the slave's phase value which is a rising ; sawtooth wave. This produces aliasing, but hey, this ; this is just an example ;) out aout * adeclick * imaxamp endin i1 0 1 7.00 i1 + 0.5 7.02 i1 + . 7.05 i1 + . 7.07 i1 + . 7.09 i1 + 2 7.06 e manual_src~dfsg/examples/wgpluck_brighter.csd0000644000000000000000000000137112262561504020622 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o wgpluck_brighter.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 icps = 220 iamp = 20000 kpick = 0.5 iplk = 0 idamp = 30 ifilt = 10 axcite oscil 1, 1, 1 apluck wgpluck icps, iamp, kpick, iplk, idamp, ifilt, axcite out apluck endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/sfplaym.csd0000644000000000000000000000171412262561504016734 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfplaym.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf sfload "07AcousticGuitar.sf2" sfplist gisf sfpassign 100, gisf instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/7000 ;scale amplitude kfreq init 1 ;do not change freq from sf aout sfplaym ivel, inum, kamp*ivel, kfreq, 100 ;preset index = 100 outs aout, aout endin f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e manual_src~dfsg/examples/gauss.csd0000644000000000000000000000174012262561504016402 0ustar rootroot -d -o dac instr 1 irange = p4 imu = p5 isamples = p6 indx = 0 icount = 1 ix = 0.0 ix2 = 0.0 loop: i1 gauss irange i1 = i1 + imu ix = ix + i1 ix2 = ix2 + i1*i1 if i1 >= -(irange+imu) && i1 <= (irange+imu) then icount = icount+1 endif loop_lt indx, 1, isamples, loop imean = ix / isamples ;mean value istd = sqrt(ix2/isamples - imean*imean) ;standard deviation prints "mean = %3.3f, std = %3.3f, ", imean, istd prints "samples inside the given range: %3.3f\%\n", icount*100.0/isamples endin i 1 0 0.1 1.0 0 100000 ; range = 1, mu = 0.0, sigma = 1/3.83 = 0.261 i 1 0.1 0.1 3.83 0 100000 ; range = 3.83, mu = 0.0, sigma = 1 i 1 0.2 0.1 5.745 2.7 100000 ; range = 5.745, mu = 2.7, sigma = 5.745/3.83 = 1.5 manual_src~dfsg/examples/tabmorphi.csd0000644000000000000000000000215012262561504017241 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tabmorphi.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 8193, 10, 1 ;sine wave giSquare ftgen 0, 0, 8193, 7, 1, 4096, 1, 0, -1, 4096, -1 ;square wave giTri ftgen 0, 0, 8193, 7, 0, 2048, 1, 4096, -1, 2048, 0 ;triangle wave giSaw ftgen 0, 0, 8193, 7, 1, 8192, -1 ;sawtooth wave, downward slope instr 1 iamp = .7 kindex phasor 440 ;read table value at this index kindex = kindex*8192 ;for all 8192 index points kweightpoint = 0.5 ;set weightpoint ktabnum1 line 0, p3, 3 ;morph through all tables ktabnum2 = 2 ;set to triangle wave ksig tabmorphi kindex, kweightpoint, ktabnum1, ktabnum2, giSine, giSquare, giTri, giSaw ksig = ksig*iamp asig interp ksig ;convert to audio signal outs asig, asig endin i1 0 5 e manual_src~dfsg/examples/schottstaedt.orc0000644000000000000000000000366712262561504020015 0ustar rootroot ;=========================================================== ; schottstaedt.orc ;=========================================================== sr = 44100 kr = 2205 ksmps = 20 nchnls = 2 ; functions for the schottstaedt string giSineWave ftgen 1, 0, 512, 10, 1 giTable1 ftgen 2, 0, 129, 7, 1, 129, 0 giVibEnvlp ftgen 3, 0, 129, 5, .01, 129, 1 giPanLeft ftgen 10, 0, 129, 9, .25, 1, 90 giPanRight ftgen 11, 0, 129, 9, .25, 1, 0 instr 1 ; a schottstaedt string instrument ;=========================================================== ; p4=amp(0-1) p5=pitch(in oct) p6=pan(0=l, 1=r) ;=========================================================== ; set constants iamp = 32767*p4 icps = cpsoct(p5) ; check to see if pan is out of our range ipanfac = (p6 > 1 ? 1 : p6) ipanfac = (p6 < 0 ? 0 : ipanfac) ; map linear pan factor to more natural sqrt function ilpan tablei ipanfac,10,1 irpan tablei ipanfac,11,1 indx1 = 7.5/log(icps) indx2 = (8.5-log(icps))/(3+icps*1000) indx3 = 1.25/sqrt(icps) ;vibrato unit krnd randi .0075,15 kvibe oscil1i 0,.015,p3/4,3 ;envlp for vibrato kvib oscili krnd+kvibe,5.5*kvibe,1 ;attack noise kgaten oscil1i 0,iamp/5,.2,2 krnd randi kgaten,.2*icps anoise oscili krnd,2000,1 ;main unit kgate oscil1i 0,1,.2,2 ;envlp for indicies to make ;chiff sound amod1 oscili (kgate+indx1)*icps,icps,1 amod2 oscili (kgate+indx2)*3*icps,3*icps,1 ;modulators amod3 oscili (kgate+indx3)*4*icps,4*icps,1 amods = icps+amod1+amod2+amod3 asig oscili iamp,amods*(kvib+1),1 ;carrier oscil with vibrato asigs linen asig+anoise,.2,p3,.2 ;final envlp for oscil+noise outs asigs*ilpan,asigs*irpan endin manual_src~dfsg/examples/pchmidib.csd0000644000000000000000000000122212262561504017032 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o pchmidib.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; This example expects MIDI note inputs on channel 1 i1 pchmidib print i1 endin ;Dummy f-table to give time for real-time MIDI events f 0 8000 e manual_src~dfsg/examples/csbeats.csd0000644000000000000000000000464712262561504016715 0ustar rootroot sr = 44100 nchnls = 2 gi1 ftgen 1, 0, 4096, 10, 1 gi2 ftgen 2, 0, 4096, 7, -1, 4096, 1 ; sawtooth gi3 ftgen 3, 0, 4096, 7, 0, 1024, 1, 2048, -1, 1024, 0 ;triangle instr 101,102,103 iamp = ampdbfs(p5) a1 oscil iamp, p4, p1-100 kenv expseg 1, p3, .01 a1 = a1 * kenv outs a1, a1 endin ; by Brian Baughn 3-14-05 ; bbaughn@berklee.net beats = 100 permeasure = 4 i101 m1 b1 q mp D3 i101 F3 i101 D3 i101 m2 b1 D3 i101 b3 D3 i101 m3 b1 D3 i101 F3 i101 D3 i101 m4 b1 D3 i101 b3 D3 i101 m5 b1 D3 i101 b4 G5 i101 m6 b1 E5 i101 b2 F5 i101 b3 e Eb5 i101 b3.5 e i101 b4 q i101 m7 b1 e D5 i101 i101 q i101 e Db5 i101 i101 q i101 m8 b1 q D5 i101 E5 i101 D5 i102 m1 b2 q D4 i102 b4 E4 i102 b4 Bb3 i102 m2 b2 F4 i102 b2 B3 i102 b4 C#4 i102 b4 Bb3 i102 m3 b2 q D4 i102 b4 E4 i102 b4 Bb3 i102 m4 b2 F4 i102 b2 B3 i102 b4 C#4 i102 b4 Bb3 i103 m5 b2 e F6 i103 b2 e A5 i103 b2.5 e D6 i103 b3 e F6 i103 b3 e A5 i103 b4 e E6 i103 m6 b1 q C#6 i103 q D6 i103 e C6 i103 i103 q i103 m7 b1 e B5 i103 i103 q i103 e Bb5 i103 i103 q i103 m8 b1 e F5 i103 b1 e A5 i103 b1.5 e D6 i103 b2 e Bb5 i103 b2.5 e D6 i103 b3 q F5 i103 b3 A5 end manual_src~dfsg/examples/lfo.csd0000644000000000000000000000124112262561504016034 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lfo.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 5 itype = p4 ;lfo type klfo line 0, p3, 20 al lfo klfo, kcps, itype asig poscil .5, 220+al, 1 outs asig, asig endin ; sine wave. f 1 0 32768 10 1 i 1 0 3 0 ;lfo = sine i 1 + 3 2 ;lfo = square i 1 + 3 5 ;lfo = saw-tooth down e manual_src~dfsg/examples/scale.csd0000644000000000000000000000066712262561504016356 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in Silent -odac -iadc -d ;;;realtime output sr = 22050 ksmps = 10 nchnls = 2 /*--- ---*/ instr 1 ; scale test kmod ctrl7 1, 1, 0, 1 printk2 kmod kout scale kmod, 0, -127 printk2 kout endin /*--- ---*/ i1 0 8888 e manual_src~dfsg/examples/cpuprc.csd0000644000000000000000000000157412262561504016561 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpuprc.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 cpuprc 1, 2 cpuprc 2, 30 instr 1 ;cpu processing-time percent usage is set to 2% for each note asig oscil 0.2, 440, 1 outs asig, asig endin instr 2 ;cpu processing-time percent usage is set to 30% for each note ;so the 4 notes of the score exceeds 100% by far asig oscil 0.2, 440, 1 outs asig, asig endin f 1 0 32768 10 1 ; sine wave i 1 0 1 i 1 0 1 i 1 0 1 i 1 0 1 ;too many notes to process, ;check Csound output! i 2 3 1 i 2 3 1 i 2 3 1 i 2 3 1 e manual_src~dfsg/examples/STKStifKarp.csd0000644000000000000000000000123512262561504017364 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKStifKarp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p6, p3, p7 ;Pickup Position kv2 = p5 ;String Sustain asig STKStifKarp cpspch(p4), 1, 4, kv1, 11, kv2, 1, 10 outs asig, asig endin i 1 0 2 5.00 0 100 100 i 1 + 40 5.00 127 1 127 i 1 10 32 5.00 127 1 10 e manual_src~dfsg/examples/chebyshevpoly.csd0000644000000000000000000000246512262561504020151 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chebyshevpoly.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 ; time-varying mixture of first six harmonics instr 1 ; According to the GEN13 manual entry, ; the pattern + - - + + - - for the signs of ; the chebyshev coefficients has nice properties. ; these six lines control the relative powers of the harmonics k1 line 1.0, p3, 0.0 k2 line -0.5, p3, 0.0 k3 line -0.333, p3, -1.0 k4 line 0.0, p3, 0.5 k5 line 0.0, p3, 0.7 k6 line 0.0, p3, -1.0 ; play the sine wave at a frequency of 256 Hz with amplitude = 1.0 ax oscili 1, 256, 1 ; waveshape it ay chebyshevpoly ax, 0, k1, k2, k3, k4, k5, k6 ; avoid clicks, scale final amplitude, and output adeclick linseg 0.0, 0.05, 1.0, p3 - 0.1, 1.0, 0.05, 0.0 outs ay * adeclick * 10000, ay * adeclick * 10000 endin f1 0 32768 10 1 ; a sine wave i1 0 5 e manual_src~dfsg/examples/PartikkelExample1.csd0000644000000000000000000001141312262561504020601 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out -odac ;;;RT audio ; For Non-realtime ouput leave only the line below: ; -o partikkel.wav -W ;;; for file output any platform sr = 44100 ksmps = 20 nchnls = 2 ; Example by Joachim Heintz and Oeyvind Brandtsegg 2008 giCosine ftgen 0, 0, 8193, 9, 1, 1, 90 ; cosine giDisttab ftgen 0, 0, 32768, 7, 0, 32768, 1 ; for kdistribution giFile ftgen 0, 0, 0, 1, "fox.wav", 0, 0, 0 ; soundfile for source waveform giWin ftgen 0, 0, 4096, 20, 9, 1 ; grain envelope giPan ftgen 0, 0, 32768, -21, 1 ; for panning (random values between 0 and 1) ; ************************************************* ; partikkel example, processing of soundfile ; uses the file "fox.wav" ; ************************************************* instr 1 /*score parameters*/ ispeed = p4 ; 1 = original speed igrainrate = p5 ; grain rate igrainsize = p6 ; grain size in ms icent = p7 ; transposition in cent iposrand = p8 ; time position randomness (offset) of the pointer in ms icentrand = p9 ; transposition randomness in cents ipan = p10 ; panning narrow (0) to wide (1) idist = p11 ; grain distribution (0=periodic, 1=scattered) /*get length of source wave file, needed for both transposition and time pointer*/ ifilen tableng giFile ifildur = ifilen / sr /*sync input (disabled)*/ async = 0 /*grain envelope*/ kenv2amt = 1 ; use only secondary envelope ienv2tab = giWin ; grain (secondary) envelope ienv_attack = -1 ; default attack envelope (flat) ienv_decay = -1 ; default decay envelope (flat) ksustain_amount = 0.5 ; no meaning in this case (use only secondary envelope, ienv2tab) ka_d_ratio = 0.5 ; no meaning in this case (use only secondary envelope, ienv2tab) /*amplitude*/ kamp = 0.4*0dbfs ; grain amplitude igainmasks = -1 ; (default) no gain masking /*transposition*/ kcentrand rand icentrand ; random transposition iorig = 1 / ifildur ; original pitch kwavfreq = iorig * cent(icent + kcentrand) /*other pitch related (disabled)*/ ksweepshape = 0 ; no frequency sweep iwavfreqstarttab = -1 ; default frequency sweep start iwavfreqendtab = -1 ; default frequency sweep end awavfm = 0 ; no FM input ifmamptab = -1 ; default FM scaling (=1) kfmenv = -1 ; default FM envelope (flat) /*trainlet related (disabled)*/ icosine = giCosine ; cosine ftable kTrainCps = igrainrate ; set trainlet cps equal to grain rate for single-cycle trainlet in each grain knumpartials = 1 ; number of partials in trainlet kchroma = 1 ; balance of partials in trainlet /*panning, using channel masks*/ imid = .5; center ileftmost = imid - ipan/2 irightmost = imid + ipan/2 giPanthis ftgen 0, 0, 32768, -24, giPan, ileftmost, irightmost ; rescales giPan according to ipan tableiw 0, 0, giPanthis ; change index 0 ... tableiw 32766, 1, giPanthis ; ... and 1 for ichannelmasks ichannelmasks = giPanthis ; ftable for panning /*random gain masking (disabled)*/ krandommask = 0 /*source waveforms*/ kwaveform1 = giFile ; source waveform kwaveform2 = giFile ; all 4 sources are the same kwaveform3 = giFile kwaveform4 = giFile iwaveamptab = -1 ; (default) equal mix of source waveforms and no amplitude for trainlets /*time pointer*/ afilposphas phasor ispeed / ifildur /*generate random deviation of the time pointer*/ iposrandsec = iposrand / 1000 ; ms -> sec iposrand = iposrandsec / ifildur ; phase values (0-1) krndpos linrand iposrand ; random offset in phase values /*add random deviation to the time pointer*/ asamplepos1 = afilposphas + krndpos; resulting phase values (0-1) asamplepos2 = asamplepos1 asamplepos3 = asamplepos1 asamplepos4 = asamplepos1 /*original key for each source waveform*/ kwavekey1 = 1 kwavekey2 = kwavekey1 kwavekey3 = kwavekey1 kwavekey4 = kwavekey1 /* maximum number of grains per k-period*/ imax_grains = 100 aL, aR partikkel igrainrate, idist, giDisttab, async, kenv2amt, ienv2tab, \ ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, igrainsize, kamp, igainmasks, \ kwavfreq, ksweepshape, iwavfreqstarttab, iwavfreqendtab, awavfm, \ ifmamptab, kfmenv, icosine, kTrainCps, knumpartials, \ kchroma, ichannelmasks, krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, asamplepos4, \ kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains outs aL, aR endin ;i1 st dur speed grate gsize cent posrnd cntrnd pan dist i1 0 2.757 1 200 15 0 0 0 0 0 s i1 0 2.757 1 200 15 400 0 0 0 0 s i1 0 2.757 1 15 450 400 0 0 0 0 s i1 0 2.757 1 15 450 400 0 0 0 0.4 s i1 0 2.757 1 200 15 0 400 0 0 1 s i1 0 5.514 .5 200 20 0 0 600 .5 1 s i1 0 11.028 .25 200 15 0 1000 400 1 1 manual_src~dfsg/examples/pvread.csd0000644000000000000000000000137012262561504016540 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvread.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; analyze "fox.wav" with PVANAL first ibin = p4 ktime line 0, p3, 2.8 kfreq, kamp pvread ktime, "fox.pvx", ibin ;read data from 7th analysis bin. asig poscil kamp, kfreq, 1 ;function 1 is a stored sine outs asig*5, asig*5 ;compensate loss of volume endin ;sine wave f1 0 4096 10 1 i 1 0 6 7 i 1 + 6 15 i 1 + 2 25 e manual_src~dfsg/examples/pdhalf.csd0000644000000000000000000000311612262561504016515 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pdhalf.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 4 idur = p3 iamp = p4 ifreq = p5 itable = p6 aenv linseg 0, .001, 1.0, idur - .051, 1.0, .05, 0 aosc phasor ifreq kamount linseg 0.0, 0.02, -0.99, 0.05, -0.9, idur-0.06, 0.0 apd pdhalf aosc, kamount aout tablei apd, itable, 1 outs aenv*aout*iamp, aenv*aout*iamp endin f1 0 16385 10 1 f2 0 16385 10 1 .5 .3333 .25 .5 f3 0 16385 9 1 1 270 ; inverted cosine ; descending "just blues" scale ; pdhalf with cosine table ; (imitates the CZ-101 "sawtooth waveform") t 0 100 i4 0 3 .6 512 3 i. + . . 448 i. + . . 384 i. + . . 358.4 i. + . . 341.33 i. + . . 298.67 i. + 5 . 256 s ; pdhalf with a sine table t 0 120 i4 0 3 .6 512 1 i. + . . 448 i. + . . 384 i. + . . 358.4 i. + . . 341.33 i. + . . 298.67 i. + 5 . 256 s ; pdhalf with a sawtooth-like table t 0 150 i4 0 3 .6 512 2 i. + . . 448 i. + . . 384 i. + . . 358.4 i. + . . 341.33 i. + . . 298.67 i. + 5 . 256 e manual_src~dfsg/examples/printks.csd0000644000000000000000000000145612262561504016756 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o printks.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Change a value linearly from 0 to 100, ; over the period defined by p3. kup line 0, p3, 100 ; Change a value linearly from 30 to 10, ; over the period defined by p3. kdown line 30, p3, 10 ; Print the value of kup and kdown, once per second. printks "kup = %f, kdown = %f\\n", 1, kup, kdown endin ; Play Instrument #1 for 5 seconds. i 1 0 5 e manual_src~dfsg/examples/resonk.csd0000644000000000000000000000137212262561504016562 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o resonk.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 0, 0, 2^10, 10, 1 instr 1 ksig randomh 400, 1800, 150 aout poscil .2, 1000+ksig, gisin outs aout, aout endin instr 2 ksig randomh 400, 1800, 150 khp line 1, p3, 400 ;vary high-pass ksig resonk ksig, khp, 50 aout poscil .2, 1000+ksig, gisin outs aout, aout endin i 1 0 5 i 2 5.5 5 e manual_src~dfsg/examples/gen16.csd0000644000000000000000000000146312262561504016202 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o gen16.wav -W ;;; for file output any platform sr = 44100 ksmps = 128 nchnls = 1 instr 1 kcps init 1/p3 kndx phasor kcps ifn = p4 ixmode = 1 kval table kndx, ifn, ixmode ibasefreq = 440 kfreq = kval * ibasefreq a1 oscil 20000, ibasefreq + kfreq, 1 out a1 endin f 1 0 16384 10 1 f 2 0 1024 16 1 1024 1 0 f 3 0 1024 16 1 1024 2 0 f 4 0 1024 16 1 1024 10 0 f 5 0 1024 16 1 1024 -1 0 f 6 0 1024 16 1 1024 -2 0 f 7 0 1024 16 1 1024 -10 0 i 1 0 2 2 i 1 + . 3 i 1 + . 4 i 1 + . 5 i 1 + . 6 i 1 + . 7 e manual_src~dfsg/examples/fluidAllOut.orc0000644000000000000000000000406512262561504017521 0ustar rootrootsr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS gienginenum1 fluidEngine gienginenum2 fluidEngine isfnum1 fluidLoad "Piano Steinway Grand Model C (21,738KB).sf2", gienginenum1, 1 ; Bright Steinway, program 1, channel 1 fluidProgramSelect gienginenum1, 1, isfnum1, 0, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidProgramSelect gienginenum1, 3, isfnum1, 0, 2 isfnum2 fluidLoad "63.3mg The Sound Site Album Bank V1.0.SF2", gienginenum2, 1 ; General MIDI, program 50, channel 2 fluidProgramSelect gienginenum2, 2, isfnum2, 0, 50 ; SEND NOTES TO STEINWAY SOUNDFONT instr 1 ; FluidSynth Steinway Rev ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 1 ikey = p4 ivelocity = p5 istatus = 144 fluidControl gienginenum1, istatus, ichannel, ikey, ivelocity endin instr 2 ; GM soundfont ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 2 ikey = p4 ivelocity = p5 istatus = 144 fluidNote gienginenum2, ichannel, ikey, ivelocity endin instr 3 ; FluidSynth Steinway Rev ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 3 ikey = p4 ivelocity = p5 istatus = 144 fluidNote gienginenum1, ichannel, ikey, ivelocity endin ; COLLECT AUDIO FROM ALL SOUNDFONTS instr 100 ; Fluidsynth output ; INITIALIZATION ; Normalize so iamplitude for p5 of 80 == ampdb(80). iamplitude = ampdb(p5) * (10000.0 / 0.1) ; AUDIO aleft, aright fluidAllOut outs aleft * iamplitude, aright * iamplitude endin manual_src~dfsg/examples/birnd.csd0000644000000000000000000000066512262561504016363 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 ; Generate a random number from -1 to 1. kbin = birnd(1) printk .2, kbin endin i 1 0 1 i 1 + . i 1 + . i 1 + . i 1 + . i 1 + . i 1 + . e manual_src~dfsg/examples/nestedap.csd0000644000000000000000000000271312262561504017064 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o nestedap.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 5 insnd = p4 gasig diskin2 insnd, 1 endin instr 10 imax = 1 idel1 = p4/1000 igain1 = p5 idel2 = p6/1000 igain2 = p7 idel3 = p8/1000 igain3 = p9 idel4 = p10/1000 igain4 = p11 idel5 = p12/1000 igain5 = p13 idel6 = p14/1000 igain6 = p15 afdbk init 0 aout1 nestedap gasig+afdbk*.4, 3, imax, idel1, igain1, idel2, igain2, idel3, igain3 aout2 nestedap aout1, 2, imax, idel4, igain4, idel5, igain5 aout nestedap aout2, 1, imax, idel6, igain6 afdbk butterlp aout, 1000 outs gasig+(aout+aout1)/2, gasig-(aout+aout1)/2 gasig = 0 endin f1 0 8192 10 1 ; Diskin ; Sta Dur Soundin i5 0 3 "beats.wav" ; Reverb ; St Dur Del1 Gn1 Del2 Gn2 Del3 Gn3 Del4 Gn4 Del5 Gn5 Del6 Gn6 i10 0 4 97 .11 23 .07 43 .09 72 .2 53 .2 119 .3 e manual_src~dfsg/examples/cabasa.csd0000644000000000000000000000104512262561504016470 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cabasa.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 inum = p4 idamp = p5 asig cabasa 0.9, 0.01, inum, idamp outs asig, asig endin i1 1 1 48 .95 i1 + 1 1000 .5 e manual_src~dfsg/examples/expsegr.csd0000644000000000000000000000123212262561504016731 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o expsegr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .3 kenv expsegr 1, .05, 0.5, 1, .01 asig pluck kenv, icps, 200, 1, 1 outs asig, asig endin f 1 0 4096 10 1 ;sine wave f0 30 ;runs 30 seconds e manual_src~dfsg/examples/oscil.csd0000644000000000000000000000146712262561504016377 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = .6 kcps = 440 ifn = p4 asig oscil kamp, kcps, ifn outs asig,asig endin f1 0 16384 10 1 ; Sine f2 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth f3 0 16384 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square f4 0 16384 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse i 1 0 2 1 i 1 3 2 2 i 1 6 2 3 i 1 9 2 4 e manual_src~dfsg/examples/midic7.csd0000644000000000000000000000152212262561504016432 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midic7.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; This example expects MIDI controller input on channel 1 ; run, play a note and move your midi controller 7 to see result imax = 1 imin = 0 ichan = 1 ictlno= 7 ; = midi volume kamp midic7 ictlno, imin, imax printk2 kamp asig oscili kamp, 220, 1 outs asig, asig endin ; no score events allowed f 0 20 ;20 sec. for real-time MIDI events f 1 0 4096 10 1 ;sine wave e manual_src~dfsg/examples/gen08.csd0000644000000000000000000000152312262561504016200 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen08.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ;choose between tables kcps init 1/p3 ;create index over duration of note. kndx phasor kcps ixmode = 1 kval table kndx, 2, ixmode ;normalize index data ibasefreq = 440 kfreq = kval * 100 ;scale asig poscil .7, ibasefreq + kfreq, 1 ;and add to frequency outs asig, asig endin f 1 0 16384 10 1 ;sine wave. f 2 0 65 8 0 16 1 16 1 16 0 17 0 f 3 0 65 8 -1 32 1 2 0 14 0 17 0 i 1 0 2 1 i 1 3 2 2 e manual_src~dfsg/examples/128-left_rightX0000755000000000000000000000313312262561504017300 0ustar rootroot 0 1 0 127 1 0 1 2 2 1 2 3 3 2 3 4 4 3 4 5 5 4 5 6 6 5 6 7 7 6 7 8 8 7 8 9 9 8 9 10 10 9 10 11 11 10 11 12 12 11 12 13 13 12 13 14 14 13 14 15 15 14 15 16 16 15 16 17 17 16 17 18 18 17 18 19 19 18 19 20 20 19 20 21 21 20 21 22 22 21 22 23 23 22 23 24 24 23 24 25 25 24 25 26 26 25 26 27 27 26 27 28 28 27 28 29 29 28 29 30 30 29 30 31 31 30 31 32 32 31 32 33 33 32 33 34 34 33 34 35 35 34 35 36 36 35 36 37 37 36 37 38 38 37 38 39 39 38 39 40 40 39 40 41 41 40 41 42 42 41 42 43 43 42 43 44 44 43 44 45 45 44 45 46 46 45 46 47 47 46 47 48 48 47 48 49 49 48 49 50 50 49 50 51 51 50 51 52 52 51 52 53 53 52 53 54 54 53 54 55 55 54 55 56 56 55 56 57 57 56 57 58 58 57 58 59 59 58 59 60 60 59 60 61 61 60 61 62 62 61 62 63 63 62 63 64 64 63 64 65 65 64 65 66 66 65 66 67 67 66 67 68 68 67 68 69 69 68 69 70 70 69 70 71 71 70 71 72 72 71 72 73 73 72 73 74 74 73 74 75 75 74 75 76 76 75 76 77 77 76 77 78 78 77 78 79 79 78 79 80 80 79 80 81 81 80 81 82 82 81 82 83 83 82 83 84 84 83 84 85 85 84 85 86 86 85 86 87 87 86 87 88 88 87 88 89 89 88 89 90 90 89 90 91 91 90 91 92 92 91 92 93 93 92 93 94 94 93 94 95 95 94 95 96 96 95 96 97 97 96 97 98 98 97 98 99 99 98 99 100 100 99 100 101 101 100 101 102 102 101 102 103 103 102 103 104 104 103 104 105 105 104 105 106 106 105 106 107 107 106 107 108 108 107 108 109 109 108 109 110 110 109 110 111 111 110 111 112 112 111 112 113 113 112 113 114 114 113 114 115 115 114 115 116 116 115 116 117 117 116 117 118 118 117 118 119 119 118 119 120 120 119 120 121 121 120 121 122 122 121 122 123 123 122 123 124 124 123 124 125 125 124 125 126 126 125 126 127 127 0 127 126 manual_src~dfsg/examples/dripwater.csd0000644000000000000000000000131612262561504017260 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dripwater.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 imaxshake = p4 ifreq = p5 ifreq1 = p6 ifreq2 = p7 ;low amplitude adrp dripwater .1, 0.09, 10, .9, imaxshake, ifreq, ifreq1, ifreq2 asig clip adrp, 2, 0.9 ; avoid drips that drip too loud outs asig, asig endin {100 CNT i1 [0.1 * $CNT] 0.5 0.5 430 1000 800 } e manual_src~dfsg/examples/FLsavesnap_simple.csd0000644000000000000000000000352512262561504020676 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O sr=48000 ksmps=128 nchnls=2 ; Example by Hector Centeno and Andres Cabrera 2007 ; giSWMtab4 ftgen 0, 0, 513, 21, 10, 1, .3 ; giSWMtab4M ftgen 0, 0, 64, 7, 1, 50, 1 FLpanel "Snapshots", 530, 190, 40, 410, 3 FLcolor 100, 118 ,140 ivalSM1 FLvalue "", 70, 20, 270, 20 gksliderA, gislidSM1 FLslider "Slider", -4, 4, 0, 3, ivalSM1, 250, 20, 20, 20 itext1 FLbox "store", 1, 1, 14, 50, 25, 355, 15 itext2 FLbox "load", 1, 1, 14, 50, 25, 415, 15 gksnap, ibuttn1 FLbutton "1", 1, 0, 11, 25, 25, 364, 45, 0, 3, 0, 3, 1 gksnap, ibuttn2 FLbutton "2", 1, 0, 11, 25, 25, 364, 75, 0, 3, 0, 3, 2 gksnap, ibuttn3 FLbutton "3", 1, 0, 11, 25, 25, 364, 105, 0, 3, 0, 3, 3 gksnap, ibuttn4 FLbutton "4", 1, 0, 11, 25, 25, 364, 135, 0, 3, 0, 3, 4 gkload, ibuttn1 FLbutton "1", 1, 0, 11, 25, 25, 424, 45, 0, 4, 0, 3, 1 gkload, ibuttn2 FLbutton "2", 1, 0, 11, 25, 25, 424, 75, 0, 4, 0, 3, 2 gkload, ibuttn3 FLbutton "3", 1, 0, 11, 25, 25, 424, 105, 0, 4, 0, 3, 3 gkload, ibuttn4 FLbutton "4", 1, 0, 11, 25, 25, 424, 135, 0, 4, 0, 3, 4 ivalSM2 FLvalue "", 70, 20, 270, 80 gkknobA, gislidSM2 FLknob "Knob", -4, 4, 0, 3, ivalSM2, 60, 120, 60 FLpanelEnd FLsetVal_i 1, gislidSM1 FLsetVal_i 1, gislidSM2 FLrun instr 1 endin instr 3 ; Save snapshot index init 0 ipstno = p4 Sfile sprintf "snapshot_simple.%d.snap", ipstno inumsnap, inumval FLsetsnap index ;, -1, igroup FLsavesnap Sfile endin instr 4 ;Load snapshot index init 0 ipstno = p4 Sfile sprintf "snapshot_simple.%d.snap", ipstno FLloadsnap Sfile inumload FLgetsnap index ;, igroup endin f 0 3600 e manual_src~dfsg/examples/envlpx.csd0000644000000000000000000000160612262561504016575 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o envlpx.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 irise = 0.2 idec = 0.5 idur = p3 - idec ifn = 1 iatss = p5 iatdec = 0.01 kenv envlpx .6, irise, idur, idec, ifn, iatss, iatdec kcps = cpspch(p4) asig vco2 kenv, kcps ;apply envlpx to the filter cut-off frequency asig moogvcf asig, kcps + (kenv * 8 * kcps) , .5 ;the higher the pitch, the higher the filter cut-off frequency outs asig, asig endin ; a linear rising envelope f 1 0 129 -7 0 128 1 i 1 0 2 7.00 .1 i 1 + 2 7.02 1 i 1 + 2 7.03 2 i 1 + 2 7.05 3 e manual_src~dfsg/examples/ntrpol.csd0000644000000000000000000000125612262561504016600 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ntrpol.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSin ftgen 1, 0, 1024, 10, 1 instr 1 avco vco2 .5, 110 ;sawtootyh wave asin poscil .5, 220, giSin ;sine wave but octave higher kx linseg 0, p3*.4, 1, p3*.6, 1 ;crossfade between saw and sine asig ntrpol avco, asin, kx outs asig, asig endin i 1 0 5 e manual_src~dfsg/examples/ins.csd0000644000000000000000000000124512262561504016051 0ustar rootroot ; Select audio/midi flags here according to platform -odac -idac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; ins.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 ;two channels out 0dbfs = 1 instr 1 ain1, ainr ins ;grab your mic and sing adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time = 20ms aoutl flanger ain1, adel, .7 aoutr flanger ain1, adel*2, .8 fout "in_s.wav", 14, aoutl, aoutr ;write to stereo file, outs aoutl, aoutr ;16 bits with header endin i 1 0 10 e manual_src~dfsg/examples/tablecopy.csd0000644000000000000000000000245312262561504017244 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tablecopy.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;after an example from Jonathan Murphy gilen = 7 gilist ftgen 1, 0, gilen + 1, -2, 2, 3, 4, 5, 6, 7 ;only 6 elements, so 1 is doubled gitmp ftgen 2, 0, gilen + 1, -2, 0 ;empty table gkmax init gilen seed 0 ;each time different instr 1 ktrig metro 3 ;trigger values krnd random 0, gkmax if (ktrig == 1) then kval table krnd, gilist tablew 0, krnd, gilist kread = 0 kwrite = 0 start: knew table kread, gilist if (knew != 0) then tablew knew, kwrite, gitmp kwrite = kwrite + 1 endif kread = kread + 1 if (kread <= gilen) kgoto start tablecopy gilist, gitmp ;fill with zeroes gkmax = gkmax - 1 endif printk2 kval if (gkmax < 0) then event "i", 2, 0, 1/kr ;when ready, then stop endif asig vco2 .5, 40*kval ;sound generation outs asig, asig endin instr 2 exitnow endin i1 0 5 e manual_src~dfsg/examples/clip.csd0000644000000000000000000000141512262561504016206 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o clip.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; white noise arnd rand 1 ; full amlitude ; Clip the noisy waveform's amplitude to 0.5 a1 clip arnd, 2, 0.5 outs a1, a1 endin instr 2 ; white noise arnd rand 1 ; full amlitude ; Clip the noisy waveform's amplitude to 0.1 a1 clip arnd, 2, 0.1 outs a1, a1 endin ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 1 1 e manual_src~dfsg/examples/passign.csd0000644000000000000000000000210712262561504016722 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o passign.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 8 idur,iamp,iskiptime,iattack,irelease,irvbtime,irvbgain passign 3 kamp linen iamp, iattack, idur, irelease asig soundin "fox.wav", iskiptime arampsig = kamp * asig aeffect reverb asig, irvbtime arvbretrn = aeffect * irvbgain ;mix dry & wet signals outs arampsig + arvbretrn, arampsig + arvbretrn endin ;ins strt dur amp skip atk rel rvbt rvbgain i8 0 4 .3 0 .03 .1 1.5 .3 i8 4 4 .3 1.6 .1 .1 3.1 .7 i8 8 4 .3 0 .5 .1 2.1 .2 i8 12 4 .4 0 .01 .1 1.1 .1 i8 16 4 .5 0.1 .01 .1 0.1 .1 e manual_src~dfsg/examples/rbjeq.csd0000644000000000000000000000126412262561504016364 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o rbjeq.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 imode = p4 a1 vco2 .3, 155.6 ; sawtooth wave kfco expon 8000, p3, 200 ; filter frequency asig rbjeq a1, kfco, 1, kfco * 0.005, 1, imode outs asig, asig endin i 1 0 5 0 ;lowpass i 1 6 5 2 ;highpass i 1 12 5 4 ;bandpass i 1 18 5 8 ;equalizer e manual_src~dfsg/examples/expsegba.csd0000644000000000000000000000155512262561504017062 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o expsega.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Define a short percussive amplitude envelope that ; goes from 0.01 to 20,000 and back. aenv expsegba 0.01, 0.1, 20000, 0.2, 0.01 a1 oscil aenv, 440, 1 out a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #1 for one second. i 1 1 1 ; Play Instrument #1 for one second. i 1 2 1 ; Play Instrument #1 for one second. i 1 3 1 e manual_src~dfsg/examples/vlinseg.csd0000644000000000000000000000135312262561504016727 0ustar rootroot -odac -B441 -b441 sr=44100 ksmps=10 nchnls=2 gilen init 32 gitable1 ftgen 0, 0, gilen, 10, 1 gitable2 ftgen 0, 0, gilen, 10, 1 gitable3 ftgen 0, 0, gilen, -7, 30, gilen, 35 gitable4 ftgen 0, 0, gilen, -7, 400, gilen, 450 gitable5 ftgen 0, 0, gilen, -7, 5000, gilen, 5500 instr 1 vcopy gitable2, gitable1, gilen turnoff endin instr 2 vlinseg gitable2, 16, gitable3, 2, gitable4, 2, gitable5 endin instr 3 kcount init 0 if kcount < 16 then kval table kcount, gitable2 printk 0,kval kcount = kcount +1 else turnoff endif endin i1 0 1 s i2 0 10 i3 0 1 i3 1 1 i3 1.5 1 i3 2 1 i3 2.5 1 i3 3 1 i3 3.5 1 i3 4 1 i3 4.5 1 manual_src~dfsg/examples/modulus.csd0000644000000000000000000000205212262561504016745 0ustar rootroot ; Select audio/midi flags here according to platform -odac -Ma --midi-key=4 --midi-velocity-amp=5 -m0 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o %.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giScale ftgen 1, 0, -12, -2, 0, -34, -7, 10, -14, 3, -21, -3, -27, -10, 7, -17 ;12 note scale with detuned keys instr 1 ikey = p4 ivel = p5 indx = ikey % 12 ;work on the twelftone scale icent tab_i indx, giScale ;load the scale ifreqeq = cpsmidinn(ikey) ifreq = ifreqeq * cent(icent) ;change frequency by cents from table prints "Key %d modulus 12 = %d. ", ikey, indx prints "Equal-tempered frequency of this key = %f,", ifreqeq prints " but here with cent deviation %d = %f%n", icent, ifreq asig vco2 ivel*.5, ifreq outs asig, asig endin f 0 60 ;run for 60 seconds e manual_src~dfsg/examples/cell.csd0000644000000000000000000000515412262561504016202 0ustar rootroot ; Select audio/midi flags here according to platform ;-odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cell.wav -W ;;; for file output any platform ; cell.csd by Gleb Rogozinsky sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 0dbfs = 1 ; Cellular automaton-driven synthesis in spectral domain instr 1 iatt = p4 ; envelope attack time isus = p5 ; envelope sustain time irel = p6 ; envelope release time ivol = p7 ; overall volume ; create some white noise asig rand 0.8 ; spectral analysis of asig fsig pvsanal asig, 2048, 1024, 2048, 0 ; get a vector of magnitudes ; calculate cellular automaton state kfreq line 50, 5, 1 ; variable CA triggering frequency ktrig metro kfreq ; trigger the CA to update cells cell ktrig, 0, 3, 1, 2, 2048 ; cells are written into ftable 3 ; use current row of cells in spectral domain fmas pvstencil fsig, 0, 1, 3 ; apply spectral mask aout pvsynth fmas ; reconstruct time signal ; apply envelope and out signal kenv expseg .001, iatt, 1, isus, 1, irel, .001 out aout*kenv*ivol endin ; This example uses one-dimensional cellular automaton ; to produce structures in spectral domain ; We have to prepare initial row of cells. ; One alive cell is enough to produce a simple fractal, ; so two alivee cells will make structure more sophisticated f1 0 2048 7 0 150 0 0 1 1 1 0 0 45 0 0 1 1 1 0 0 ; The CA rule is used as follows: ; the states (values) of each cell are summed with their neighboring cells. ; Each sum is used as an index to read a next state of cell ; from the rule table. ; Let us try rule # 129 (LSB binary 1 0 0 0 0 0 0 1). ; This rule will produce a fractal structure for single active cell ; For more rules see http://mathworld.wolfram.com/ElementaryCellularAutomaton.html f2 0 8 -2 1 0 0 0 0 0 0 1 ; Try some different rules i.E. 254 (0 1 1 1 1 1 1 1) or 169 (1 0 0 1 0 1 0 1) ; Prepare the output table of ielements size f3 0 2048 10 0 ; Time to make it sound! i1 0 13 0.3 7 3 1 e manual_src~dfsg/examples/median.csd0000644000000000000000000000036212262561504016514 0ustar rootroot sr = 44100 kr = 441 ksmps = 100 nchnls = 1 instr 1 a1 oscil 30000, 10, 1 a2 median a1, 5, 8 out a2 endin f1 0 4096 10 1 i 1 0 0.1 e manual_src~dfsg/examples/sininv.csd0000644000000000000000000000104512262561504016564 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o sininv.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 0.5 i1 = sininv(irad) print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/mediank.csd0000644000000000000000000000042012262561504016662 0ustar rootroot -n sr = 44100 kr = 147 ksmps = 300 nchnls = 1 instr 1 k1 oscil 100, 10, 1 k2 mediank k1, 5, 8 printk 0, k2 endin f1 0 4096 10 1 i 1 0 1 e manual_src~dfsg/examples/wgbrass.csd0000644000000000000000000000155012262561504016727 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o wgbrass.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 ksmps = 10 nchnls = 1 0dbfs = 1 ; Instrument #1. instr 1 kamp = 0.7 kfreq = p4 ktens = p5 iatt = p6 kvibf = p7 ifn = 1 ; Create an amplitude envelope for the vibrato. kvamp line 0, p3, 0.5 a1 wgbrass kamp, kfreq, ktens, iatt, kvibf, kvamp, ifn out a1 endin ; Table #1, a sine wave. f 1 0 1024 10 1 ; freq tens att vibf i 1 0 4 440 0.4 0.1 6.137 i 1 4 4 440 0.4 0.01 0.137 i 1 8 4 880 0.4 0.1 6.137 e manual_src~dfsg/examples/sum.csd0000644000000000000000000000141412262561504016062 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sum.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisine ftgen 0, 0, 2^10, 10, 1 instr 1 a1 oscili 1, 10.0, gisine ;combine 3 sinusses a2 oscili 1, 1.0, gisine ;at different rates a3 oscili 1, 3.0, gisine ares sum a1, a2, a3 ;sum them ares = ares*100 ;scale result and asig poscil .5, ares+110, gisine ;add to frequency outs asig, asig endin i1 0 5 e manual_src~dfsg/examples/rezzy.csd0000644000000000000000000000126712262561504016447 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o rezzy.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 asaw vco2 .3, 110 ;sawtooth kcf line 1760, p3, 220 ;vary cut-off frequency from 220 to 1280 Hz kres = p4 ;vary resonance too ares rezzy asaw, kcf, kres asig balance ares, asaw outs asig, asig endin i 1 0 4 10 i 1 + 4 30 i 1 + 4 120 ;lots of resonance e manual_src~dfsg/examples/leftbrace.csd0000644000000000000000000000203412262561504017204 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o abs.wav -W ;;; for file output any platform nchnls = 2 gaReverbSend init 0 ; a simple sine wave partial instr 1 idur = p3 iamp = p4 ifreq = p5 aenv linseg 0.0, 0.1*idur, iamp, 0.6*idur, iamp, 0.3*idur, 0.0 aosc oscili aenv, ifreq, 1 vincr gaReverbSend, aosc endin ; global reverb instrument instr 2 al, ar reverbsc gaReverbSend, gaReverbSend, 0.85, 12000 outs gaReverbSend+al, gaReverbSend+ar clear gaReverbSend endin f1 0 4096 10 1 { 4 CNT { 8 PARTIAL ; start time duration amplitude frequency i1 [0.5 * $CNT.] [1 + ($CNT * 0.2)] [500 + (~ * 200)] [800 + (200 * $CNT.) + ($PARTIAL. * 20)] } } i2 0 6 e manual_src~dfsg/examples/gen19.csd0000644000000000000000000000162612262561504016206 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen19.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;p4 = transposition factor ;p5 = speed factor ;p6 = function table for grain data i1 = sr/ftlen(1) ;scaling to reflect sample rate and table length a1 phasor i1*p5 ;index for speed asigl fog .5, 15, p4, a1, 1, 0, .01, .5, .01, 30, 1, 2, p3 ;left channel asigr fog .4, 25, p4+.2, a1, 1, 0, .01, .5, .01, 30, 1, 2, p3, .5 ;right channel outs asigl, asigr endin f 1 0 131072 1 "fox.wav" 0 0 0 f 2 0 1024 19 .5 .5 270 .5 i 1 0 10 .7 .1 i 1 + 4 1.2 2 e manual_src~dfsg/examples/fprintks-2.csd0000644000000000000000000000157012262561504017260 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages ; -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -n -Fmidichn_advanced.mid ;Don't write audio ouput to disk and use the file midichn_advanced.mid as MIDI input sr = 48000 ksmps = 16 nchnls = 2 ;Example by Jonathan Murphy 2007 ; assign all midi events to instr 1000 massign 0, 1000 pgmassign 0, 1000 instr 1000 ktim timeinsts kst, kch, kd1, kd2 midiin if (kst != 0) then ; p4 = MIDI event type p5 = channel p6= data1 p7= data2 fprintks "MIDI2cs.sco", "i1\\t%f\\t%f\\t%d\\t%d\\t%d\\t%d\\n", ktim, 1/kr, kst, kch, kd1, kd2 endif endin i1000 0 10000 e manual_src~dfsg/examples/vibrato.csd0000644000000000000000000000161112262561504016723 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vibrato.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kaverageamp init .5 kaveragefreq init 5 krandamountamp line p4, p3, p5 ;increase random amplitude of vibrato krandamountfreq init .3 kampminrate init 3 kampmaxrate init 5 kcpsminrate init 3 kcpsmaxrate init 5 kvib vibrato kaverageamp, kaveragefreq, krandamountamp, krandamountfreq, kampminrate, kampmaxrate, kcpsminrate, kcpsmaxrate, 1 asig poscil .8, 220+kvib, 1 ;add vibrato outs asig, asig endin f 1 0 16384 10 1 ;sine wave i 1 0 15 .01 20 e manual_src~dfsg/examples/ATSsinnoi.csd0000644000000000000000000000115212262561504017124 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSsinnoi.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 2 knzfade expon 0.001, p3, 2 aout ATSsinnoi ktime, 1, knzfade, 1, "beats.ats", 150 outs aout*2, aout*2 ;amplify some more endin i 1 0 2 e manual_src~dfsg/examples/resonz.csd0000644000000000000000000000522412262561504016601 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o resonz.wav -W ;;; for file output any platform /* Written by Sean Costello */ ; Orchestra file for resonant filter sweep of a sawtooth-like waveform. ; The outputs of reson, resonr, and resonz are scaled by coefficients ; specified in the score, so that each filter can be heard on its own ; from the same instrument. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 idur = p3 ibegfreq = p4 ; beginning of sweep frequency iendfreq = p5 ; ending of sweep frequency ibw = p6 ; bandwidth of filters in Hz ifreq = p7 ; frequency of gbuzz that is to be filtered iamp = p8 ; amplitude to scale output by ires = p9 ; coefficient to scale amount of reson in output iresr = p10 ; coefficient to scale amount of resonr in output iresz = p11 ; coefficient to scale amount of resonz in output ; Frequency envelope for reson cutoff kfreq linseg ibegfreq, idur * .5, iendfreq, idur * .5, ibegfreq ; Amplitude envelope to prevent clicking kenv linseg 0, .1, iamp, idur - .2, iamp, .1, 0 ; Number of harmonics for gbuzz scaled to avoid aliasing iharms = (sr*.4)/ifreq asig gbuzz 1, ifreq, iharms, 1, .9, 1 ; "Sawtooth" waveform ain = kenv * asig ; output scaled by amp envelope ares reson ain, kfreq, ibw, 1 aresr resonr ain, kfreq, ibw, 1 aresz resonz ain, kfreq, ibw, 1 out ares * ires + aresr * iresr + aresz * iresz endin /* Written by Sean Costello */ f1 0 8192 9 1 1 .25 ; cosine table for gbuzz generator i1 0 10 1 3000 200 100 4000 1 0 0 ; reson output with bw = 200 i1 10 10 1 3000 200 100 4000 0 1 0 ; resonr output with bw = 200 i1 20 10 1 3000 200 100 4000 0 0 1 ; resonz output with bw = 200 i1 30 10 1 3000 50 200 8000 1 0 0 ; reson output with bw = 50 i1 40 10 1 3000 50 200 8000 0 1 0 ; resonr output with bw = 50 i1 50 10 1 3000 50 200 8000 0 0 1 ; resonz output with bw = 50 e manual_src~dfsg/examples/linseg.csd0000644000000000000000000000204112262561504016534 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o linseg.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kcps = cpspch(p4) kenv linseg 0, 0.25, 1, 0.75, 0 ; together = 1 sec asig poscil kenv, kcps, giSine outs asig, asig endin instr 2 ; scaling to duration kcps = cpspch(p4) kenv linseg 0, p3*0.25, 1, p3*0.75, 0 asig poscil kenv, kcps, giSine outs asig, asig endin instr 3 ; with negative value kcps = cpspch(p4) aenv linseg 0, 0.1, 1, 0.5, -0.9, 0.4, 0 asig poscil aenv, kcps, giSine outs asig, asig endin i 1 0 1 7.00 ; = 1 sec, p3 fits exactly i 1 2 2 7.00 ; = 2 sec, p3 truncated at 1 sec i 2 4 1 7.00 ; scales to duration i 2 6 2 7.00 ; of p3 i 3 9 2 7.00 e manual_src~dfsg/examples/oscilikt.csd0000644000000000000000000000160612262561504017102 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o oscilikt.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a uni-polar (0-1) square wave. kamp1 init 1 kcps1 init 2 itype = 3 ksquare lfo kamp1, kcps1, itype ; Use the square wave to switch between Tables #1 and #2. kamp2 init 20000 kcps2 init 220 kfn = ksquare + 1 a1 oscilikt kamp2, kcps2, kfn out a1 endin ; Table #1, a sine waveform. f 1 0 4096 10 0 1 ; Table #2: a sawtooth wave f 2 0 3 -2 1 0 -1 ; Play Instrument #1 for two seconds. i 1 0 2 manual_src~dfsg/examples/randomh.csd0000644000000000000000000000155112262561504016710 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o randomh.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 seed 0 ; Instrument #1. instr 1 ; Choose a random frequency between 220 and 440 Hz. ; Generate new random numbers at 10 Hz. kmin init 220 kmax init 440 kcps init 10 imode = p4 ifstval = p5 printf_i "\nMode: %d\n", 1, imode k1 randomh kmin, kmax, kcps, imode, ifstval printk2 k1 endin ; Play Instrument #1 for one second, ; each time with a different mode. i 1 0 1 i 1 1 1 2 330 i 1 2 1 3 e manual_src~dfsg/examples/sfilist.csd0000644000000000000000000000173212262561504016736 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out, virtual midi in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfilist.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf sfload "sf_GMbank.sf2" sfilist gisf ;list all instruments instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/3000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1, a2 sfinstr3 ivel, inum, kamp*ivel, kfreq, 100, gisf ;choose Halo Pad outs a1, a2 endin f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e manual_src~dfsg/examples/fluid.csd0000644000000000000000000000122012262561504016354 0ustar rootroot ; Audio out Audio in -odac -iadc -M0 ;;;RT audio I/O and RT midi sr = 44100 ksmps = 100 nchnls = 2 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "07AcousticGuitar.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 0 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 120000 asigl, asigr fluidOut giengine outs asigl * imvol, asigr * imvol endin i 1 0 3600 60 100 i 99 0 3600 e manual_src~dfsg/examples/insremot.csd0000644000000000000000000000076312262561504017124 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o insremot.wav -W ;;; for file output any platform nchnls = 1 insremot "192.168.1.100", "192.168.1.101", 1 instr 1 aq barmodel 1, 1, p4, 0.001, 0.23, 5, p5, p6, p7 out aq endin f0 360 e manual_src~dfsg/examples/fmvoice.csd0000644000000000000000000000155212262561504016711 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmvoice.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 110 kvowel = p4 ; p4 = vowel (0 - 64) ktilt = p5 kvibamt = 0.005 kvibrate = 6 asig fmvoice .5, kfreq, kvowel, ktilt, kvibamt, kvibrate outs asig, asig endin ; sine wave. f 1 0 16384 10 1 i 1 0 1 1 0 ; tilt=0 i 1 1 1 > . i 1 2 1 > . i 1 3 1 > . i 1 4 1 > . i 1 5 1 > . i 1 6 1 > . i 1 7 1 12 . i 1 10 1 1 90 ; tilt=90 i 1 11 1 > . i 1 12 1 > . i 1 13 1 > . i 1 14 1 > . i 1 15 1 > . i 1 16 1 > . i 1 17 1 12 . e manual_src~dfsg/examples/128-stringX0000755000000000000000000000311712262561504016461 0ustar rootroot 0 1 1 0 1 2 2 1 2 3 3 2 3 4 4 3 4 5 5 4 5 6 6 5 6 7 7 6 7 8 8 7 8 9 9 8 9 10 10 9 10 11 11 10 11 12 12 11 12 13 13 12 13 14 14 13 14 15 15 14 15 16 16 15 16 17 17 16 17 18 18 17 18 19 19 18 19 20 20 19 20 21 21 20 21 22 22 21 22 23 23 22 23 24 24 23 24 25 25 24 25 26 26 25 26 27 27 26 27 28 28 27 28 29 29 28 29 30 30 29 30 31 31 30 31 32 32 31 32 33 33 32 33 34 34 33 34 35 35 34 35 36 36 35 36 37 37 36 37 38 38 37 38 39 39 38 39 40 40 39 40 41 41 40 41 42 42 41 42 43 43 42 43 44 44 43 44 45 45 44 45 46 46 45 46 47 47 46 47 48 48 47 48 49 49 48 49 50 50 49 50 51 51 50 51 52 52 51 52 53 53 52 53 54 54 53 54 55 55 54 55 56 56 55 56 57 57 56 57 58 58 57 58 59 59 58 59 60 60 59 60 61 61 60 61 62 62 61 62 63 63 62 63 64 64 63 64 65 65 64 65 66 66 65 66 67 67 66 67 68 68 67 68 69 69 68 69 70 70 69 70 71 71 70 71 72 72 71 72 73 73 72 73 74 74 73 74 75 75 74 75 76 76 75 76 77 77 76 77 78 78 77 78 79 79 78 79 80 80 79 80 81 81 80 81 82 82 81 82 83 83 82 83 84 84 83 84 85 85 84 85 86 86 85 86 87 87 86 87 88 88 87 88 89 89 88 89 90 90 89 90 91 91 90 91 92 92 91 92 93 93 92 93 94 94 93 94 95 95 94 95 96 96 95 96 97 97 96 97 98 98 97 98 99 99 98 99 100 100 99 100 101 101 100 101 102 102 101 102 103 103 102 103 104 104 103 104 105 105 104 105 106 106 105 106 107 107 106 107 108 108 107 108 109 109 108 109 110 110 109 110 111 111 110 111 112 112 111 112 113 113 112 113 114 114 113 114 115 115 114 115 116 116 115 116 117 117 116 117 118 118 117 118 119 119 118 119 120 120 119 120 121 121 120 121 122 122 121 122 123 123 122 123 124 124 123 124 125 125 124 125 126 126 125 126 127 127 126 manual_src~dfsg/examples/gen13.csd0000644000000000000000000001374212262561504016202 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen13.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;example by Russell Pinkston - Univ. of Texas (but slightly modified) gisine ftgen 0, 0, 16384, 10, 1 ;sine wave instr 1 ihertz = cpspch(p4) ipkamp = p5 iwsfn = p6 ;waveshaping function inmfn = p7 ;normalization function agate linen 1, .01, p3, .1 ;overall amp envelope kctrl linen .99, 2, p3, 2 ;waveshaping index control aindex poscil kctrl/2, ihertz, gisine ;sine wave to be distorted asignal tablei .5+aindex, iwsfn, 1 ;waveshaping knormal tablei kctrl, inmfn, 1 ;amplitude normalization asig = asignal*knormal*ipkamp*agate outs asig, asig endin ; This proves the statement in Dodge (p. 147) that Chebyshev polynomials ; of order K have "only the kth harmonic." This is only true when the ; waveshaping index is at the maximum - i.e., when the entire transfer ; function is being accessed. RP. ;-------------------------------------------------------------------------------------------------------------------------------------------- ; quasi sawtooth transfer function: ; h0 h1 h2 h3 h4 h5 h6 h7 h8 h9 h10 h11 h12 h13 h14 h15 h16 h17 h18 h19 h20 f1 0 513 13 1 1 0 100 -50 -33 25 20 -16.7 -14.2 12.5 11.1 -10 -9.09 8.333 7.69 -7.14 -6.67 6.25 5.88 -5.55 -5.26 5 f2 0 257 4 1 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 0 4 6.00 .7 1 2 i1 4 . 7.00 . i1 8 . 8.00 . ;-------------------------------------------------------------------------------------------------------------------------------------------- ; quasi square wave transfer function: ; h0 h1 h2 h3 h4 h5 h6 h7 h8 h9 h10 h11 h12 h13 h14 h15 h16 h17 h18 h19 f3 0 513 13 1 1 0 100 0 -33 0 20 0 -14.2 0 11.1 0 -9.09 0 7.69 0 -6.67 0 5.88 0 -5.26 f4 0 257 4 3 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 16 4 6.00 .7 3 4 i1 20 . 7.00 . i1 24 . 8.00 . ;-------------------------------------------------------------------------------------------------------------------------------------------- ; quasi triangle wave transfer function: ; h0 h1 h2 h3 h4 h5 h6 h7 h8 h9 h10 h11 h12 h13 h14 h15 h16 h17 h18 h19 f5 0 513 13 1 1 0 100 0 -11.11 0 4 0 -2.04 0 1.23 0 -.826 0 .59 0 -.444 0 .346 0 -.277 f6 0 257 4 5 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 32 4 6.00 .7 5 6 i1 36 . 7.00 . i1 40 . 8.00 . ;-------------------------------------------------------------------------------------------------------------------------------------------- ; transfer function1: h0 h1 h2 h3 h4 h5 h6 h7 h8 h9 h10 h11 h12 h13 h14 h15 h16 f7 0 513 13 1 1 0 1 -.8 0 .6 0 0 0 .4 0 0 0 0 .1 -.2 -.3 .5 f8 0 257 4 7 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 48 4 5.00 .7 7 8 i1 52 . 6.00 . i1 56 . 7.00 . ;-------------------------------------------------------------------------------------------------------------------------------------------- ;=========================================================================; ; This demonstrates the use of high partials, sometimes without a ; ; fundamental, to get quasi-inharmonic spectra from waveshaping. ; ;=========================================================================; ; transfer function2: h0 h1 h2 h3 h4 h5 h6 h7 h8 h9 h10 h11 h12 h13 h14 h15 h16 f9 0 513 13 1 1 0 0 0 -.1 0 .3 0 -.5 0 .7 0 -.9 0 1 0 -1 0 f10 0 257 4 9 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 64 4 5.00 .7 9 10 i1 68 . 6.00 . i1 72 . 7.00 . ;-------------------------------------------------------------------------------------------------------------------------------------------- ; transfer function3: h0 h1 h2 h3 h4 h5 h6 h7 h8 h9 h10 h11 h12 h13 h14 h15 h16 h17 h18 h19 h17 h18 h19 h20 f11 0 513 13 1 1 0 0 0 0 0 0 0 -1 0 1 0 0 -.1 0 .1 0 -.2 .3 0 -.7 0 .2 0 -.1 f12 0 257 4 11 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 80 4 5.00 .7 11 12 i1 84 . 5.06 . i1 88 . 6.00 . ;-------------------------------------------------------------------------------------------------------------------------------------------- ;=========================================================================; ; split a sinusoid into 3 odd-harmonic partials of relative strength 5:3:1 ;=========================================================================; ;-------------------------------------------------------------------------------------------------------------------------------------------- ; transfer function4: h0 h1 h2 h3 h4 h5 f13 0 513 13 1 1 0 5 0 3 0 1 f14 0 257 4 13 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 96 4 5.00 .7 13 14 i1 100 . 5.06 . i1 104 . 6.00 . e manual_src~dfsg/examples/zamod.csd0000644000000000000000000000224312262561504016371 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zamod.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 2 a-rate variables and 2 k-rate variables. zakinit 2, 2 ; Instrument #1 -- a simple waveform. instr 1 ; Vary an a-rate signal linearly from 20,000 to 0. asig line 20000, p3, 0 ; Send the signal to za variable #1. zaw asig, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Generate a simple sine wave. asin oscil 1, 440, 1 ; Modify the sine wave, multiply its amplitude by ; za variable #1. a1 zamod asin, -1 ; Generate the audio output. out a1 ; Clear the za variables, prepare them for ; another pass. zacl 0, 2 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 ; Play Instrument #2 for 2 seconds. i 2 0 2 e manual_src~dfsg/examples/valpass-2.csd0000644000000000000000000000146712262561504017076 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o valpass-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 65536, 10, 1 ;sine wave instr 1 asig diskin2 "beats.wav", 1, 0, 1 krvt line 0.01, p3, p3 ;reverb time adepth = p4 ;sine depth krate = 0.3 ;sine rate (speed) adel oscil 0.5, krate, giSine ;delay time oscillator (LFO) adel = ((adel+0.5)*adepth) ;scale and offset LFO aout valpass asig, krvt, adel*0.01, 0.5 outs aout, aout endin i1 0 10 1 i1 11 10 5 e manual_src~dfsg/examples/ATSaddnz.csd0000644000000000000000000000107012262561504016724 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc for RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSaddnzwav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 2 asig ATSaddnz ktime, "beats.ats", 1, 24 outs asig*10, asig*10 ;amplify endin i 1 0 2 e manual_src~dfsg/examples/outq4.csd0000644000000000000000000000122312262561504016330 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outq4.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 asig vco2 .05, 30 ; sawtooth waveform at low volume kcut line 100, p3, 30 ; Vary cutoff frequency kresonance = 7 inumlayer = 2 asig lowresx asig, kcut, kresonance, inumlayer outq4 asig ; output channel 4 endin i 1 0 3 e manual_src~dfsg/examples/bqrez.csd0000644000000000000000000000142312262561504016401 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o bqrez.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;sawtooth waveform. kfco line 200, p3, 2000;filter-cutoff frequency from .2 to 5 KHz. kres = p4 ;resonance imode = p5 ;mode asig vco 0.2, 220, 1 afilt bqrez asig, kfco, kres, imode asig balance afilt, asig outs asig, asig endin ;sine wave f 1 0 16384 10 1 i 1 0 3 1 0 ; low pass i 1 + 3 30 0 ; low pass i 1 + 3 1 1 ; high pass i 1 + 3 30 1 ; high pass e manual_src~dfsg/examples/portk.csd0000644000000000000000000000166612262561504016426 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ; -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o portk.wav -W ;;; for file output any platform sr = 44100 ksmps = 128 nchnls = 1 ;Example by Andres Cabrera 2007 FLpanel "Slider", 650, 140, 50, 50 gkval1, gislider1 FLslider "Watch me", 0, 127, 0, 5, -1, 580, 30, 25, 20 gkval2, gislider2 FLslider "Move me", 0, 127, 0, 5, -1, 580, 30, 25, 80 gkhtim, gislider3 FLslider "khtim", 0.1, 1, 0, 6, -1, 30, 100, 610, 10 FLpanelEnd FLrun FLsetVal_i 0.1, gislider3 ;set initial time to 0.1 instr 1 kval portk gkval2, gkhtim ; take the value of slider 2 and apply portamento FLsetVal 1, kval, gislider1 ;set the value of slider 1 to kval endin ; Play Instrument #1 for one minute. i 1 0 60 e manual_src~dfsg/examples/osciliktp.csd0000644000000000000000000000131112262561504017253 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o osciliktp.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1: osciliktp example instr 1 kphs line 0, p3, 4 a1x osciliktp 220.5, 1, 0 a1y osciliktp 220.5, 1, -kphs a1 = a1x - a1y out a1 * 14000 endin ; Table #1: Sawtooth wave f 1 0 3 -2 1 0 -1 ; Play Instrument #1 for four seconds. i 1 0 4 e manual_src~dfsg/examples/svfilter.csd0000644000000000000000000000360312262561504017116 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o svfilter.wav -W ;;; for file output any platform ; Orchestra file for resonant filter sweep of a sawtooth-like waveform. ; The seperate outputs of the filter are scaled by values from the score, ; and are mixed together. sr = 44100 kr = 2205 ksmps = 20 nchnls = 1 instr 1 idur = p3 ifreq = p4 iamp = p5 ilowamp = p6 ; determines amount of lowpass output in signal ihighamp = p7 ; determines amount of highpass output in signal ibandamp = p8 ; determines amount of bandpass output in signal iq = p9 ; value of q iharms = (sr*.4) / ifreq asig gbuzz 1, ifreq, iharms, 1, .9, 1 ; Sawtooth-like waveform kfreq linseg 1, idur * 0.5, 4000, idur * 0.5, 1 ; Envelope to control filter cutoff alow, ahigh, aband svfilter asig, kfreq, iq aout1 = alow * ilowamp aout2 = ahigh * ihighamp aout3 = aband * ibandamp asum = aout1 + aout2 + aout3 kenv linseg 0, .1, iamp, idur -.2, iamp, .1, 0 ; Simple amplitude envelope out asum * kenv endin f1 0 8192 9 1 1 .25 i1 0 5 100 1000 1 0 0 5 ; lowpass sweep i1 5 5 200 1000 1 0 0 30 ; lowpass sweep, octave higher, higher q i1 10 5 100 1000 0 1 0 5 ; highpass sweep i1 15 5 200 1000 0 1 0 30 ; highpass sweep, octave higher, higher q i1 20 5 100 1000 0 0 1 5 ; bandpass sweep i1 25 5 200 1000 0 0 1 30 ; bandpass sweep, octave higher, higher q i1 30 5 200 2000 .4 .6 0 ; notch sweep - notch formed by combining highpass and lowpass outputs e manual_src~dfsg/examples/pvstanal.csd0000644000000000000000000000177112262561504017114 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvstanal.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifil ftgen 0, 0, 0, 1, "fox.wav", 0, 0, 1 instr 1 fsig pvstanal p4, 1, p5, gifil, p6, p7 aout pvsynth fsig outs aout, aout endin instr 2 kspeed randi 2, 2, 2 ;speed randomly between -2 and 2 kpitch randi 2, 2, 2 ;pitch between 2 octaves lower or higher fsig pvstanal kspeed, 1, octave(kpitch), gifil aout pvsynth fsig outs aout, aout endin ; speed pch det wrap i 1 0 2.757 1 1 0 0 i 1 3 . 2 1 0 0 i 1 6 . 2 1 0 1 i 1 9 . 1 .75 i 2 12 10 ;random scratching e manual_src~dfsg/examples/integ.csd0000644000000000000000000000151212262561504016363 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o integ.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig diskin2 "fox.wav", 1 outs asig, asig endin instr 2 ; with diff asig diskin2 "fox.wav", 1 ares diff asig outs ares, ares endin instr 3 ; with integ asig diskin2 "fox.wav", 1 aint integ asig aint = aint*.05 ;way too loud outs aint, aint endin instr 4 ; with diff and integ asig diskin2 "fox.wav", 1 ares diff asig aint integ ares outs aint, aint endin i 1 0 1 i 2 1 1 i 3 2 1 i 4 3 1 e manual_src~dfsg/examples/vtablewa.csd0000644000000000000000000000052112262561504017061 0ustar rootroot -odac -b441 -B441 sr=44100 kr=4410 ksmps=10 nchnls=2 0dbfs = 1 instr 1 vcopy 2, 1, 262144 ar random 0, 1 vtablewa ar,2,1,ar out ar,ar endin f1 0 262144 -1 "beats.wav" 0 4 0 f2 0 262144 2 0 i1 0 4 manual_src~dfsg/examples/toot5.csd0000644000000000000000000000243712262561504016336 0ustar rootroot -odac instr 5 irel = 0.01 ; set vibrato release time idel1 = p3 * p10 ; calculate initial delay (% of dur) isus = p3 - (idel1 + irel) ; calculate remaining duration iamp = ampdb(p4) iscale = iamp * .333 ; p4=amp inote = cpspch(p5) ; p5=freq k3 linseg 0, idel1, p9, isus, p9, irel, 0 ; p6=attack time k2 oscil k3, p8, 1 ; p7=release time k1 linen iscale, p6, p3, p7 ; p8=vib rate a3 oscil k1, inote*.995+k2, 1 ; p9=vib depth a2 oscil k1, inote*1.005+k2, 1 ; p10=vib delay (0-1) a1 oscil k1, inote+k2, 1 out a1+a2+a3 endin f1 0 4096 10 1 ; sine wave ;ins strt dur amp freq atk rel vibrt vbdpt vbdel i5 0 3 86 10.00 0.1 0.7 7 6 .4 i5 4 3 86 10.02 1 0.2 6 6 .4 i5 8 4 86 10.04 2 1 5 6 .4 manual_src~dfsg/examples/nreverb.csd0000644000000000000000000000133612262561504016724 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o nreverb.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 gaout init 0 instr 1 a1 oscil 15000, 440, 1 out a1 gaout = gaout+a1 endin instr 99 a2 nreverb gaout, 2, .3 out a2*.15 ;volume of reverb gaout = 0 endin ; Table 1: an ordinary sine wave. f 1 0 32768 10 1 i 1 0 .5 i 1 1 .5 i 1 2 .5 i 1 3 .5 i 1 4 .5 i 99 0 9 e manual_src~dfsg/examples/trfilter.csd0000644000000000000000000000162112262561504017111 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trfilter.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifn ftgen 2, 0, -22050, 5, 1, 1000, 1, 4000, 0.000001, 17050, 0.000001 ; low-pass filter curve of 22050 points instr 1 kam line 1, p3, p4 ain diskin2 "beats.wav", 1, 0, 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .003, 1, 3, 500 ; partial tracking fscl trfilter fst, kam, gifn ; filtering using function table 2 aout tradsyn fscl, 1, 1, 500, 1 ; resynthesis outs aout, aout endin f1 0 8192 10 1 i 1 0 4 1 i 1 5 4 0 ;reduce filter effect e manual_src~dfsg/examples/moogladder.csd0000644000000000000000000000117012262561504017372 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o moogladder.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfe expseg 500, p3*0.9, 1800, p3*0.1, 3000 asig buzz 1, 100, 20, 1 kres line .1, p3, .99 ;increase resonance afil moogladder asig, kfe, kres outs afil, afil endin f 1 0 4096 10 1 i 1 0 10 e manual_src~dfsg/examples/atonex.csd0000644000000000000000000000127212262561504016556 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o atonex.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; unfiltered noise asig rand 0.7 ; white noise outs asig, asig endin instr 2 ; filtered noise asig rand 0.7 khp line 100, p3, 3000 afilt atonex asig, khp, 32 ; Clip the filtered signal's amplitude to 85 dB. a1 clip afilt, 2, ampdb(85) outs a1, a1 endin i 1 0 2 i 2 2 2 e manual_src~dfsg/examples/chnset.csd0000644000000000000000000000351712262561504016550 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chnset.wav -W ;;; for file output any platform ;Example by Joachim Heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1; send i-values chnset 1, "sio" chnset -1, "non" endin instr 2; send k-values kfreq randomi 100, 300, 1 chnset kfreq, "cntrfreq" kbw = kfreq/10 chnset kbw, "bandw" endin instr 3; send a-values anois rand .1 chnset anois, "noise" loop: idur random .3, 1.5 timout 0, idur, do reinit loop do: ifreq random 400, 1200 iamp random .1, .3 asig oscils iamp, ifreq, 0 aenv transeg 1, idur, -10, 0 asine = asig * aenv chnset asine, "sine" endin instr 11; receive some chn values and send again ival1 chnget "sio" ival2 chnget "non" print ival1, ival2 kcntfreq chnget "cntrfreq" kbandw chnget "bandw" anoise chnget "noise" afilt reson anoise, kcntfreq, kbandw afilt balance afilt, anoise chnset afilt, "filtered" endin instr 12; mix the two audio signals amix1 chnget "sine" amix2 chnget "filtered" chnmix amix1, "mix" chnmix amix2, "mix" endin instr 20; receive and reverb amix chnget "mix" aL, aR freeverb amix, amix, .8, .5 outs aL, aR endin instr 100; clear chnclear "mix" endin i 1 0 20 i 2 0 20 i 3 0 20 i 11 0 20 i 12 0 20 i 20 0 20 i 100 0 20 manual_src~dfsg/examples/midinoteoncps.csd0000644000000000000000000000167312262561504020140 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midinoteoncps.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 midinoteoncps p4, p5 ;puts MIDI key translated to cycles per second into p4, and MIDI velocity into p5 print p4 ;display the key value when it changes and when key is pressed kvel = p5/127 ;scale midi velocity to 0-1 kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by asig pluck kenv*kvel, p4, p4, 2, 1 ;velocity value outs asig, asig endin f 0 30 ;runs for 30 seconds f 2 0 4096 10 1 i 1 0 2 440 100 ; play these notes from score as well i 1 + 2 1440 100 e manual_src~dfsg/examples/spat3d_stereo.csd0000644000000000000000000000533612262561504020044 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o spat3d_stereo.wav -W ;;; for file output any platform /* Written by Istvan Varga */ sr = 48000 kr = 1000 ksmps = 48 nchnls = 2 /* room parameters */ idep = 3 /* early reflection depth */ itmp ftgen 1, 0, 64, -2, \ /* depth1, depth2, max delay, IR length, idist, seed */ \ idep, 48, -1, 0.01, 0.25, 123, \ 1, 21.982, 0.05, 0.87, 4000.0, 0.6, 0.7, 2, /* ceil */ \ 1, 1.753, 0.05, 0.87, 3500.0, 0.5, 0.7, 2, /* floor */ \ 1, 15.220, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* front */ \ 1, 9.317, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* back */ \ 1, 17.545, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* right */ \ 1, 12.156, 0.05, 0.87, 5000.0, 0.8, 0.7, 2 /* left */ instr 1 /* some source signal */ a1 phasor 150 ; oscillator a1 butterbp a1, 500, 200 ; filter a1 = taninv(a1 * 100) a2 phasor 3 ; envelope a2 mirror 40*a2, -100, 5 a2 limit a2, 0, 1 a1 = a1 * a2 * 9000 kazim line 0, 2.5, 360 ; move sound source around kdist line 1, 10, 4 ; distance ; convert polar coordinates kX = sin(kazim * 3.14159 / 180) * kdist kY = cos(kazim * 3.14159 / 180) * kdist kZ = 0 a1 = a1 + 0.000001 * 0.000001 ; avoid underflows imode = 1 ; change this to 3 for 8 spk in a cube, ; or 1 for simple stereo aW, aX, aY, aZ spat3d a1, kX, kY, kZ, 1.0, 1, imode, 2, 2 aW = aW * 1.4142 ; stereo ; aL = aW + aY /* left */ aR = aW - aY /* right */ ; quad (square) ; ;aFL = aW + aX + aY /* front left */ ;aFR = aW + aX - aY /* front right */ ;aRL = aW - aX + aY /* rear left */ ;aRR = aW - aX - aY /* rear right */ ; eight channels (cube) ; ;aUFL = aW + aX + aY + aZ /* upper front left */ ;aUFR = aW + aX - aY + aZ /* upper front right */ ;aURL = aW - aX + aY + aZ /* upper rear left */ ;aURR = aW - aX - aY + aZ /* upper rear right */ ;aLFL = aW + aX + aY - aZ /* lower front left */ ;aLFR = aW + aX - aY - aZ /* lower front right */ ;aLRL = aW - aX + aY - aZ /* lower rear left */ ;aLRR = aW - aX - aY - aZ /* lower rear right */ outs aL, aR endin /* Written by Istvan Varga */ i 1 0 10 e manual_src~dfsg/examples/pvsblur.csd0000644000000000000000000000107612262561504016757 0ustar rootroot -odac sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of the audio-signal fftblur pvsblur fftin, p4, 1; blur aout pvsynth fftblur; resynthesis out aout endin i 1 0 3 0 i 1 3 3 .1 i 1 6 3 .5 e manual_src~dfsg/examples/fillarray.csd0000644000000000000000000000051612262561504017245 0ustar rootroot ; Select audio/midi flags here according to platform -n sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kS[] fillarray 1,7,5 printk2 kS[0] printk2 kS[1] printk2 kS[2] endin i 1 0 0 e manual_src~dfsg/examples/xtratim-2.csd0000644000000000000000000000401212262561504017102 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in Silent MIDI in -odac -iadc -d -M0 ;;;realtime I/O ;xtratim example by Jonathan Murphy Dec. 2006 sr = 44100 ksmps = 10 nchnls = 2 ; sine wave for oscillators gisin ftgen 1, 0, 4096, 10, 1 ; set volume initially to midpoint ctrlinit 1, 7,64 ;;; simple two oscil, two envelope synth instr 1 ; frequency kcps cpsmidib ; initial velocity (noteon) ivel veloc ; master volume kamp ctrl7 1, 7, 0, 127 kamp = kamp * ivel ; parameters for aenv1 iatt1 = 0.03 idec1 = 1 isus1 = 0.25 irel1 = 1 ; parameters for aenv2 iatt2 = 0.06 idec2 = 2 isus2 = 0.5 irel2 = 2 ; extra (release) time allocated xtratim (irel1>irel2 ? irel1 : irel2) ; krel is used to trigger envelope release krel init 0 krel release ; if noteoff received, krel == 1, otherwise krel == 0 if (krel == 1) kgoto rel ; attack, decay, sustain segments atmp1 linseg 0, iatt1, 1, idec1, isus1 , 1, isus1 atmp2 linseg 0, iatt2, 1, idec2, isus2 , 1, isus2 aenv1 = atmp1 aenv2 = atmp2 kgoto done ; release segment rel: atmp3 linseg 1, irel1, 0, 1, 0 atmp4 linseg 1, irel2, 0, 1, 0 aenv1 = atmp1 * atmp3 ;to go from the current value (in case aenv2 = atmp2 * atmp4 ;the attack hasn't finished) to the release. ; control oscillator amplitude using envelopes done: aosc1 oscil aenv1, kcps, gisin aosc2 oscil aenv2, kcps * 1.5, gisin aosc1 = aosc1 * kamp aosc2 = aosc2 * kamp ; send aosc1 to left channel, aosc2 to right, ; release times are noticably different outs aosc1, aosc2 endin f 0 3600 ;dummy table to wait for realtime MIDI events manual_src~dfsg/examples/sfpassign.csd0000644000000000000000000000277312262561504017264 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out, virtual midi in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfpassign.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;load two soundfonts gisf sfload "07AcousticGuitar.sf2" gir sfload "01hpschd.sf2" sfplist gisf sfplist gir sfpassign 0, gisf sfpassign 1, gir instr 1 ; play guitar from score and midi keyboard - preset index = 0 mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/5000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay3 ivel, inum, kamp*ivel, kfreq, 0 ;preset index = 0 outs a1, a2 endin instr 2 ; play harpsichord from score and midi keyboard - preset index = 1 mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/1000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay3 ivel, inum, kamp*ivel, kfreq, 1 ;preset index = 1 outs a1, a2 endin f0 60 ; stay active for 1 minute i1 0 1 60 100 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 i2 5 1 60 100 i2 + 1 62 < i2 7 1 65 < i2 7 1 69 10 e manual_src~dfsg/examples/timeinsts.csd0000644000000000000000000000133512262561504017277 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o timeinsts.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kvib init 1 ktim timeinsts ;read time if ktim > 2 then ;do something after 2 seconds kvib oscili 2, 3, giSine ;make a vibrato endif asig poscil .5, 600+kvib, giSine ;add vibrato outs asig, asig endin i 1 0 5 e manual_src~dfsg/examples/sprintfk.csd0000644000000000000000000000134012262561504017114 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o sprintfk.wav -W ;;; for file output any platform sr = 48000 ksmps = 16 nchnls = 2 0dbfs = 1 ; Example by Jonathan Murphy 2007 instr 1 S1 = "1" S2 = " + 1" ktrig init 0 kval init 2 if (ktrig == 1) then S1 strcatk S1, S2 kval = kval + 1 endif String sprintfk "%s = %d", S1, kval puts String, kval ktrig metro 1 endin i1 0 10 e manual_src~dfsg/examples/STKFlute.csd0000644000000000000000000000120612262561504016716 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKFlute.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;jet delay kv4 line 0, p3, 100 ;vibrato depth asig STKFlute cpspch(ifrq), 1, 2, kv1, 4, 100, 11, 100, 1, kv4, 128, 100 outs asig, asig endin i 1 0 5 8.00 0 0 i 1 7 3 9.00 20 120 e manual_src~dfsg/examples/resonx.csd0000644000000000000000000000140512262561504016574 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o resonx.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; unfiltered noise kenv linseg 0, p3*.5, 1, p3*.5, 0 ;envelope asig rand 0.7 ;white noise outs asig*kenv, asig*kenv endin instr 2 ; filtered noise kenv linseg 0, p3*.5, 1, p3*.5, 0 ;envelope asig rand 0.7 kcf line 300, p3, 2000 afilt resonx asig, kcf, 300, 4 asig balance afilt, asig outs asig*kenv, asig*kenv endin i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/fout.csd0000644000000000000000000000200712262561504016232 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fout.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 asig poscil3 .5, 880, giSine ;write a raw file: 32 bits with header fout "fout_880.wav", 15, asig outs asig, asig endin instr 2 klfo lfo 1, 2, 0 asig poscil3 .5*klfo, 220, giSine ;write an aiff file: 32 bits with header fout "fout_aif.aiff", 25, asig ; fout "fout_all3.wav", 14, asig outs asig, asig endin instr 99 ;read the stereo csound output buffer allL, allR monitor ;write the output of csound to an audio file ;to a wav file: 16 bits with header fout "fout_all.wav", 14, allL, allR endin i 1 0 2 i 2 0 3 i 99 0 3 e manual_src~dfsg/examples/round.csd0000644000000000000000000000116312262561504016406 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o round.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idiv init 1 loop: inumber = 9 i1 = inumber / idiv iro = round(i1) print inumber, idiv, iro ;print number / idiv = result using round idiv = idiv + 1 if (idiv <= 10) igoto loop endin i 1 0 0 e manual_src~dfsg/examples/fluidcomplex.csd0000644000000000000000000000332112262561504017750 0ustar rootroot ; Select audio/midi flags here according to platform ;Load external midi file from manual/examples -odac -T -F Anna.mid;;;realtime audio I/O and midifile in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fluidcomplex.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; Example by Istvan Varga ; disable triggering of instruments by MIDI events ichn = 1 lp1: massign ichn, 0 loop_le ichn, 1, 16, lp1 pgmassign 0, 0 ; initialize FluidSynth gifld fluidEngine gisf2 fluidLoad "sf_GMbank.sf2", gifld, 1 ; k-rate version of fluidProgramSelect opcode fluidProgramSelect_k, 0, kkkkk keng, kchn, ksf2, kbnk, kpre xin igoto skipInit doInit: fluidProgramSelect i(keng), i(kchn), i(ksf2), i(kbnk), i(kpre) reinit doInit rireturn skipInit: endop instr 1 ; initialize channels kchn init 1 if (kchn == 1) then lp2: fluidControl gifld, 192, kchn - 1, 0, 0 fluidControl gifld, 176, kchn - 1, 7, 100 fluidControl gifld, 176, kchn - 1, 10, 64 loop_le kchn, 1, 16, lp2 endif ; send any MIDI events received to FluidSynth nxt: kst, kch, kd1, kd2 midiin if (kst != 0) then if (kst != 192) then fluidControl gifld, kst, kch - 1, kd1, kd2 else fluidProgramSelect_k gifld, kch - 1, gisf2, 0, kd1 endif kgoto nxt endif ; get audio output from FluidSynth ivol init 3 ;a bit louder aL, aR fluidOut gifld outs aL*ivol, aR*ivol endin i 1 0 3600 e manual_src~dfsg/examples/minabs.csd0000644000000000000000000000146512262561504016535 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o minabs.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 k1 oscili 1, 10.0, 1 ;combine 3 sinusses k2 oscili 1, 1.0, 1 ;at different rates k3 oscili 1, 3.0, 1 kmin minabs k1, k2, k3 kmin = kmin*250 ;scale kmin printk2 kmin ;check the values aout vco2 .5, 220, 6 ;sawtooth asig moogvcf2 aout, 600+kmin, .5 ;change filter above 600 Hz outs asig, asig endin f1 0 32768 10 1 i1 0 5 e manual_src~dfsg/examples/else.csd0000644000000000000000000000131412262561504016205 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o else.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = cpspch(p4) ienv = p5 if (ienv == 0) then kenv adsr 0.01, 0.95, .7, .5 else kenv linseg 0, p3 * .5, 1, p3 * .5, 0 endif aout vco2 .8, ipch, 10 aout moogvcf aout, ipch + (kenv * 6 * ipch) , .5 aout = aout * kenv outs aout, aout endin i 1 0 2 8.00 0 i 1 3 2 8.00 1 e manual_src~dfsg/examples/max_k.csd0000644000000000000000000000161012262561504016353 0ustar rootroot ; Select audio/midi flags here according to platform -odac -Ma ;;;realtime audio out and midi in (on all inputs) ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o max_k.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 FLpanel "This Panel contains VU-meter",300,100 gk1,gih1 FLslider "VU-meter", 0,1,0,1, -1, 250,30, 30,30 FLsetColor2 50, 50, 255, gih1 FLpanel_end FLrun ga1 init 0 instr 1 kenv linsegr 0,.5,.7,.5,.5,.2,0 ifreq cpsmidi a1 poscil 0dbfs*kenv, ifreq, 1 ga1 = ga1+a1 endin instr 2 outs ga1, ga1 ktrig metro 25 ;refresh 25 times per second kval max_k ga1, ktrig, 1 FLsetVal ktrig, kval, gih1 ga1 = 0 endin f1 0 1024 10 1 i2 0 3600 f0 3600 e manual_src~dfsg/examples/setscorepos.csd0000644000000000000000000000133412262561504017630 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o setscorepos.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 8192, 10, 1 instr 1 asig poscil 0.5, p4, giSine ;play something outs asig, asig endin instr 11 setscorepos 8.5 endin i1 0 2 220 ;this one will be played i11 2.5 1 ;start setscorepos now i1 3 2 330 ;skip this note i1 6 2 440 ;and this one i1 9 2 550 ;play this one manual_src~dfsg/examples/clarinet.sdif0000644000000000000000001504355012262561504017250 0ustar rootrootSDIFRBELpRBEL5??@@@@@@@@@@@@@ @ @"@"@$@$@&@&@(@(@*@*@,@,@.@.@0@0@1@1@2@2@3@3@4@4@5@5@6@6@7@7@8@8@9@9@:@:@;@;@<@<@=@=@>@>@?@?@@@@@@@@@A@A@A@A@B@B@B@B@C@C@C@C@D@D@D@D@E@E@E@E@F@F@F@F@G@G@G@G@H@H@H@H@I@I@I@I@J@J@JRBEPh?PbMRBEP@|ݡ4a>3~ 0@/?Wa@@8y?gg?xN?\kC0?N;g8@@<8n>?@ v[? ?I_vzRBEP?eL_RBEP@O>׋FF%@h?T#@@0h?ĵAD@3 ӦH?͢?Cb( @@<8n@> ?(ӠLP@(@)0Šj?_pCw@=?t?@BؠRBEP?o䎊qRBEP@O?ʤӊ?N3h@@̾f?m^wd@ /sǴ?vnc?K7.e1@@l4鴍W>ú?@ Wv?2!?M&%lC@@{~? 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Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trigseq.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giTimes ftgen 91, 0, 128, -2, 1, 1/2, 1/2, 1/8, 1/8, 1/2,1/2, 1/16, 1/16, 1/16, 1/16, 1/16, 1/16, 1/16, 1/16; times giSeq ftgen 90, 0, 128, -2, 1, 2, .5, 3, .25, 4, .10, 5, .05, 6 ;** sequence amplitude and freq-ratio bins instr 1 icps init p4 iamp init .3 kloop init p5 initndx init p6 kloop2 init p7 initndx2 init p8 kdur init p9 iminTime init p10 imaxTime init p11 kampratio init 1 kfreqratio init 1 ktime_unit expseg iminTime,p3/8,iminTime,p3* 3/4,imaxTime,p3/8,imaxTime ;**ktrig seqtime ktime_unit, kstart, kloop, initndx, kfn_times ;ktrig seqtime 1/ktime_unit, 0, 15, 0, giTimes ktrig metro ktime_unit ;**** trigseq ktrig_in, kstart, kloop, initndx, kfn_values, kout1 [, kout2, kout3, ...., koutN] trigseq ktrig, 0, kloop2,initndx2, giSeq, kampratio, kfreqratio ;atrig = ktrig*10000 schedkwhen ktrig, -1, -1, 3, 0, kdur, kampratio*iamp, kfreqratio*icps ; schedkwhen ktrig, -1, -1, 2, 0, ktrig, kampratio*iamp, kfreqratio*icps endin instr 2 icps init p4 iamp init 20000 kloop init p5 initndx init p6 kloop2 init p7 initndx2 init p8 kdur init p9 iminTime init p10 imaxTime init p11 kampratio init 1 kfreqratio init 1 ktime_unit expseg iminTime,p3/8,iminTime,p3* 3/4,imaxTime,p3/8,imaxTime ;**ktrig seqtime ktime_unit, kstart, kloop, initndx, kfn_times ktrig seqtime 1/ktime_unit, 0, 15, 0, giTimes ;ktrig metro ktime_unit ;**** trigseq ktrig_in, kstart, kloop, initndx, kfn_values, kout1 [, kout2, kout3, ...., koutN] trigseq ktrig, 0, kloop2, initndx2, giSeq, kampratio, kfreqratio printk2 ktrig ;atrig = ktrig*10000 ; schedkwhen ktrig, -1, -1, 2, 0, kdur, kampratio*iamp, kfreqratio*icps schedkwhen ktrig, -1, -1, 3, 0, ktrig, kampratio*iamp, kfreqratio*icps endin instr 3 print p3 kenv expseg 1.04, p3,.04 a1 foscili p4*a(kenv-0.04), p5,1,1,kenv*5, 2 outs a1, a1 endin f2 0 8192 10 1 ; icps unused unused kloop2 initndx2 kdur iminTime imaxTime s i1 0 6 100 0 0 5 0 .2 3 15 i1 8 6 150 0 0 4 1 .1 4 30 i1 16 6 200 0 0 5 3 .25 8 50 i1 24 6 300 0 0 3 0 .1 1 30 i2 32 6 100 0 0 5 0 .2 1 1 i2 40 6 150 0 0 4 1 .1 .5 .5 i2 48 6 200 0 0 5 3 .25 3 .5 i2 56 6 300 0 0 5 0 .1 1 8 e manual_src~dfsg/examples/array_array_math.csd0000644000000000000000000000322712262561504020607 0ustar rootroot -n -m128 instr 1 ;create array and fill with numbers 1..10 resp .1..1 kArr1[] fillarray 1, 2, 3, 4, 5, 6, 7, 8, 9, 10 kArr2[] fillarray 1, 2, 3, 5, 8, 13, 21, 34, 55, 89 ;print contents printf "%s", 1, "\nkArr1:\n" kndx = 0 until kndx == lenarray(kArr1) do printf "kArr1[%d] = %f\n", kndx+1, kndx, kArr1[kndx] kndx += 1 od printf "%s", 1, "\nkArr2:\n" kndx = 0 until kndx == lenarray(kArr2) do printf "kArr2[%d] = %f\n", kndx+1, kndx, kArr2[kndx] kndx += 1 od ;add arrays kArr3[] = kArr1 + kArr2 ;print content printf "%s", 1, "\nkArr1 + kArr2:\n" kndx = 0 until kndx == lenarray(kArr3) do printf "kArr3[%d] = %f\n", kndx+1, kndx, kArr3[kndx] kndx += 1 od ;subtract arrays kArr4[] = kArr1 - kArr2 ;print content printf "%s", 1, "\nkArr1 - kArr2:\n" kndx = 0 until kndx == lenarray(kArr4) do printf "kArr4[%d] = %f\n", kndx+1, kndx, kArr4[kndx] kndx += 1 od ;multiply arrays kArr5[] = kArr1 * kArr2 ;print content printf "%s", 1, "\nkArr1 * kArr2:\n" kndx = 0 until kndx == lenarray(kArr5) do printf "kArr5[%d] = %f\n", kndx+1, kndx, kArr5[kndx] kndx += 1 od ;divide arrays kArr6[] = kArr1 / kArr2 ;print content printf "%s", 1, "\nkArr1 / kArr2:\n" kndx = 0 until kndx == lenarray(kArr6) do printf "kArr5[%d] = %f\n", kndx+1, kndx, kArr6[kndx] kndx += 1 od ;turnoff turnoff endin i 1 0 .1 manual_src~dfsg/examples/ATSbufread.csd0000644000000000000000000000137112262561504017240 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc for RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSbufread.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" and "fox.ats" are created by atsa ktime line 0, p3, 4 ktime2 line 0, p3, 4 kline expseg 0.001, .3, 1, p3-.3, 1 kline2 expseg 0.001, p3, 3 ATSbufread ktime2, 1, "fox.ats", 20 aout ATScross ktime, 2, "beats.ats", 1, kline, 0.001 * (4 - kline2), 180 outs aout*2, aout*2 endin ; sine wave. f 1 0 16384 10 1 i 1 0 4 e manual_src~dfsg/examples/dates.csd0000644000000000000000000000233412262561504016360 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dates.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 ;each time different seed instr 1 ;;generating a different filename each time csound renders itim date Stim dates itim Syear strsub Stim, 20, 24 Smonth strsub Stim, 4, 7 Sday strsub Stim, 8, 10 iday strtod Sday Shor strsub Stim, 11, 13 Smin strsub Stim, 14, 16 Ssec strsub Stim, 17, 19 Sfilnam sprintf "%s_%s_%02d_%s_%s_%s.wav", Syear, Smonth, iday, Shor,Smin, Ssec ;;rendering with random frequency, amp and pan, and writing to disk ifreq random 400, 1000 iamp random .1, 1 ipan random 0, 1 asin oscils iamp, ifreq, 0 aL, aR pan2 asin, ipan fout Sfilnam, 14, aL, aR outs aL, aR printf_i "File '%s' written to the same directory as this CSD file is!\n", 1, Sfilnam endin i 1 0 1 manual_src~dfsg/examples/wgbowedbar.csd0000644000000000000000000000122612262561504017402 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o wgbowedbar.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kp = p6 asig wgbowedbar p4, cpspch(p5), 1, kp, 0.995 outs asig, asig endin s i1 0 .5 .5 7.00 .1 ;short sound i1 + . .3 8.00 .1 i1 + . .5 9.00 .1 s i1 0 .5 .5 7.00 1 ;longer sound i1 + . .3 8.00 1 i1 + . .5 9.00 1 e manual_src~dfsg/examples/vrandi.csd0000644000000000000000000000216112262561504016541 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vrandi.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 ;Example by Andres Cabrera gitab ftgen 0, 0, 16, -7, 0, 128, 0 instr 1 krange init p4 kcps init p5 ioffset init p6 ; table krange kcps ielements idstoffset iseed isize ioffset vrandi gitab, krange, kcps, 3, 3, 2, 1, ioffset kfreq1 table 3, gitab kfreq2 table 4, gitab kfreq3 table 5, gitab ;Change the frequency of three oscillators according to the random values aosc1 oscili 4000, kfreq1, 1 aosc2 oscili 2000, kfreq2, 1 aosc3 oscili 4000, kfreq3, 1 outs aosc1+aosc2, aosc3+aosc2 endin f 1 0 2048 10 1 ; krange kcps ioffset i 1 0 5 100 1 300 i 1 5 5 5 1 400 i 1 10 5 100 2 1000 i 1 15 5 400 4 1000 i 1 20 5 1000 8 2000 i 1 20 5 300 32 350 e manual_src~dfsg/examples/pcount.csd0000644000000000000000000000454112262561504016572 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pcount.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; This UDO returns a pfield value but returns 0 if it does not exist. opcode mypvalue, i, i index xin inum pcount if (index > inum) then iout = 0.0 else iout pindex index endif xout iout endop ; Envelope UDO that reads parameters from a flexible number of pfields ; Syntax: kenv flexlinseg ipstart ; ipstart is the first pfield of the envelope ; parameters. Reads remaining pfields (up to 21 of them). ; kenv is the output envelope. opcode flexlinseg, k, i ipstart xin iep1 mypvalue ipstart iep2 mypvalue ipstart + 1 iep3 mypvalue ipstart + 2 iep4 mypvalue ipstart + 3 iep5 mypvalue ipstart + 4 iep6 mypvalue ipstart + 5 iep7 mypvalue ipstart + 6 iep8 mypvalue ipstart + 7 iep9 mypvalue ipstart + 8 iepa mypvalue ipstart + 9 iepb mypvalue ipstart + 10 iepc mypvalue ipstart + 11 iepd mypvalue ipstart + 12 iepe mypvalue ipstart + 13 iepf mypvalue ipstart + 14 iepg mypvalue ipstart + 15 ieph mypvalue ipstart + 16 iepi mypvalue ipstart + 17 iepj mypvalue ipstart + 18 iepk mypvalue ipstart + 19 iepl mypvalue ipstart + 20 kenv linseg iep1, iep2, iep3, iep4, iep5, iep6, iep7, iep8, \ iep9, iepa, iepb, iepc, iepd, iepe, iepf, iepg, \ ieph, iepi, iepj, iepk, iepl xout kenv endop instr 1 ; This instrument only requires 3 pfields but can accept up to 24. ; (You will still get warnings about more than 3). kenv flexlinseg 4 ; envelope params start at p4 aout oscili kenv*.5, 256, 1 outs aout, aout endin f1 0 8192 10 1 i1 0 2 0.0 1.0 1.0 1.0 0.0 i1 2 2 0.0 0.1 1.0 1.0 1.0 0.1 0.0 i1 4 2 0.0 0.5 0.0 ; one problem is that "missing" pfields carry i1 6 2 0.0 0.5 0.0 ! ; now we can fix this problem with ! i1 8 10 0.0 3.0 1.0 0.3 0.1 0.3 1.0 0.3 0.1 0.3 1.0 0.3 0.1 0.8 0.9 5.0 0.0 e manual_src~dfsg/examples/outq2.csd0000644000000000000000000000122312262561504016326 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outq2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 asig vco2 .05, 30 ; sawtooth waveform at low volume kcut line 300, p3, 60 ; Vary cutoff frequency kresonance = 7 inumlayer = 2 asig lowresx asig, kcut, kresonance, inumlayer outq2 asig ; output channel 2 endin i 1 0 3 e manual_src~dfsg/examples/define_args.csd0000644000000000000000000000152512262561504017527 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o define_args.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Define the oscillator macro. #define OSCMACRO(VOLUME'FREQ'TABLE) #oscil $VOLUME, $FREQ, $TABLE# ; Instrument #1 instr 1 ; Use the oscillator macro. ; This will be expanded to "a1 oscil 5000, 440, 1". a1 $OSCMACRO(5000'440'1) ; Send it to the output. out a1 endin ; Define Table #1 with an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/HRTFcompact0000644000000000000000000056000012262561504016621 0ustar rootroot ZZyy"~"~ (r(r ,, QQrr 44KK&&ss ..--pp[[<<::XX$$88//$$QQ]]$$  ((55KK1144oo00AA HHNN--::^^gg99hhEE**kk  "&65Bٖg!%h +}}tT(4f \. _ W$TP*sg/^6|c:DC'Y5 lcywd 39 d> R":'bbo)I AMvd->9{ mZx9`N^^xdHyY"I`Tq*7FUOje*'FbHk?;:ng &km>C[,_=1h *" $k +X,* A= .  *+U, b? -  yrEt,6aBSBIvY]0+^)1;<xw%N!5k)<y',%b(9fi-]tz<.+%%PeT&kR7I+SV3--{3K)>j0 ,5PE_s AM #/]|< /Of- } k zߊ0 62h%%Y<KlyumT#NEP {D Y    5 K fe%G@5|#x\J]#mMYf9@#WK w`/M74Bh@WeAxr(rztY.1ra9HN[hKXy"~V@M3ugFM_tH*¼% !#T)i - (u=&5@8 ' tm0 %$XDdB/e[$N[3U01Qcp]4_ jagv~W jm4wc@gt]B Xo_4PpWY)tM)cysy/ZVVWXx3%"4 #I4W  3CR]`(eq@T32OY &*-{X#wa]N$aL"`<R-iBt/uT_$Bzk}Eom2yq&J`X3&'@c'Fjoo %skRn@.P>xILvF,  }$ _ilDLųݓ:81W61Vb ] uNz+x|zN?qXmW:#rWJ6NJ&X /luquWBX?`xvO"Dx A@I" U/|Xpm"{]VjK`99+;hD{/}cH,N+ K%C. `u=2W_ NwPo&^wias@ | .ES,-8 VsB~K#HvN[n0>ZgGlffuXL(scJ[dX 9L&m/!]U(,)nnnrx6tsC0:RPpEW/ub1 5F$L=KŁxҩu5 NB @  b 4 +JN'6u3q&u8C),38bQYY aq #jKvtO anFoViLdg? ,VL MXK$44c|bxsArHyfaRsCBvA*?g6#I?ہ%zzF@G  E &)  I r-IkDt!G'm$ &M[9tr _zsCn=n;=IrInwt8X/=amqhaZ10$)J(b 8AoaI * wpM]?hs{1\g}8!0}aM"bm] CoE1~p =$pT5BMA @ Y  /^fs5ZTn( #: zn2 ^[Z\' Rs!Hp8aJM?%@N3txiYgx9(Bnq}|Yo}~RUXEynY G*juuSBT5bw{_s?O>+r 1^ 4"|9I{yÇiebe^9_1+` 9 ;AF]hL"2|k04MGXuh\O8jnT7%4o>Y@KK* O+1J?RQA/x7}`afF5s, }P"?oVt*PQ+)65V=tmqxs`i )5|#oFM~)J9jg $%F]{4mJ*Zz: VmHnF mBV}h8M 0&WT <nAB5ck *ѿɍ'*"z ;8 E3;nn^4#y %o Gj0Y=n *l;n;Drg^iAR]G)%/6:E}87u#^|>XYVk8&?2~>lUa)5 8[$cKB4f)f62bh\B FJ+ML$Pr *$ ]x -y= ܷG l cձ=qtF9 r g c-)c=z.I4A[SMl>XCw$p4w0:!lc!UNo=}; x:\N4atdVX/+"`M RfMBa'{VNFY~)*zEp.RC4>"D(05 !R.<L/e  QvO!SL}:3ugh-e"94{749?~">H' qsWr@T]g%eu)qXY_J#|A1]R }uvBKG;M`lu:|n} v&ZE%aDT)3Z4Rxm !dڌ D$">2,& >9!<ؼ   "Bh  F il0IIyQ6P_{gs V!yp },3K@]|.#v  B9Zne) l{k+='&+*n [z# m9s:zL GzpF=tRqpa=iAR'MMF> +G#Ef9P9[($ 0 d? 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EEnn 55uu ##SS,,''!!qqbb''!!QQyy XX..kkCCRRssvvtt$$55^^xxQQEEcc22hhuuss]]TT??wwyyCC^^#x#x  ``??''88||{{""PPcc##ssNNiiKKyy\\**""VV;;55==;;}}HHXX33]]OO""nnvv::99FFEE&&||ssmanual_src~dfsg/examples/jitter2.csd0000644000000000000000000000137112262561504016643 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o jitter2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktotamp init p4 kamp1 init .5 kcps1 init 10 kamp2 init .5 kcps2 init 2 kamp3 init .5 kcps3 init 3 kj2 jitter2 ktotamp, kamp1, kcps1, kamp2, kcps2, kamp3, kcps3 aout pluck 1, 200+kj2, 1000, 0, 1 aout dcblock aout outs aout, aout endin i 1 0 15 2 ;a bit jitter i 1 8 15 10 ;some more i 1 16 15 20 ;lots more e manual_src~dfsg/examples/plate.csd0000644000000000000000000000132712262561504016366 0ustar rootroot ; Select audio/midi flags here according to platform ;-odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: -o plate.wav -W ;;; for file output any platform ; Note: this example is heavy on CPU sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ab diskin2 "beats.wav", 1 al,ar platerev 1, 2, p4, 0.73, 1.0, 5.0, 0.001, ab,ab outs al*.25,ar*.25 endin f1 0 8 -2 0.3 0.3875 0.39274 0.32 0.85714 0.78548 f2 0 8 -2 0.2 0.666667 1.57097 0.24 0.75 0.78548 i1 0 4 1 i1 + 4 2 e manual_src~dfsg/examples/vtable1k.csd0000644000000000000000000000236112262561504016771 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O sr = 44100 ksmps = 100 nchnls = 2 giElem init 13 giOutTab ftgen 1,0,128, 2, 0 giFreqTab ftgen 2,0,128,-7, 1,giElem, giElem+1 giSine ftgen 3,0,256,10, 1 FLpanel "This Panel contains a Slider Bank",500,400 FLslidBnk "mod1@mod2@mod3@amp@freq1@freq2@freq3@freqPo", giElem, giOutTab, 360, 600, 100, 10 FLpanel_end FLrun instr 1 kout1 init 0 kout2 init 0 kout3 init 0 kout4 init 0 kout5 init 0 kout6 init 0 kout7 init 0 kout8 init 0 vtable1k giOutTab, kout1 , kout2, kout3, kout4, kout5 , kout6, kout7, kout8 kmodindex1= 2 * db(kout1 * 80 ) kmodindex2= 2 * db(kout2 * 80 ) kmodindex3= 2 * db(kout3 * 80 ) kamp = 50 * db(kout4 * 70 ) kfreq1 = 1.1 * octave(kout5 * 10) kfreq2 = 1.1 * octave(kout6 * 10) kfreq3 = 1.1 * octave(kout7 * 10) kfreq4 = 30 * octave(kout8 * 8) amod1 oscili kmodindex1, kfreq1, giSine amod2 oscili kmodindex2, kfreq2, giSine amod3 oscili kmodindex3, kfreq3, giSine aout oscili kamp, kfreq4+amod1+amod2+amod3, giSine outs aout, aout endin i1 0 3600 f0 3600 manual_src~dfsg/examples/noteondur2.csd0000644000000000000000000000127412262561504017361 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1;;;RT audio I/O with MIDI in sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 instr 1 ifund notnum ivel veloc idur = 1 ;chord with single key noteondur2 1, ifund, ivel, idur noteondur2 1, ifund+3, ivel, idur noteondur2 1, ifund+7, ivel, idur noteondur2 1, ifund+9, ivel, idur endin ; Dummy ftable f 0 60 manual_src~dfsg/examples/dust2.csd0000644000000000000000000000100012262561504016306 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kdens expon 2, p3, 20000 aout dust2 0.5, kdens outs aout, aout endin i1 0 10 e manual_src~dfsg/examples/stack.csd0000644000000000000000000000140312262561504016361 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o stack.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 stack 100000 instr 1 a1 oscils 0.7, 220, 0 k1 line 0, p3, 1 push "blah", 123.45, a1, k1 push rnd(k1) k_rnd pop S01, i01, a01, k01 pop printf_i "S01 = '%s', i01 = %g\n", 1, S01, i01 ktrig metro 5.0 printf "k01 = %.3f, k_rnd = %.3f\n", ktrig, k01, k_rnd outs a01, a01 endin i 1 0 5 e manual_src~dfsg/examples/cosinv.csd0000644000000000000000000000104512262561504016557 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cosinv.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 0.5 i1 = cosinv(irad) print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/in.csd0000644000000000000000000000117512262561504015670 0ustar rootroot ; Select audio/midi flags here according to platform -odac -idac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; in.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;1 channel in, two channels out ain1 in ;grab your mic and sing adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time = 20ms aout flanger ain1, adel, .7 fout "in_1.wav", 14, aout, aout ;write to stereo file, outs aout, aout ;16 bits with header endin i 1 0 10 e manual_src~dfsg/examples/pvsfreeze.csd0000644000000000000000000000130212262561504017263 0ustar rootroot -odac sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 seed 0 instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "fox.wav" ain soundin Sfile1 kfreq randomh .7, 1.1, 3; probability of freezing freqs: 1/4 kamp randomh .7, 1.1, 3; idem for amplitudes fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file freeze pvsfreeze fftin, kamp, kfreq; freeze amps or freqs independently aout pvsynth freeze; resynthesize out aout endin r 10 i 1 0 2.757 e manual_src~dfsg/examples/exp.csd0000644000000000000000000000247712262561504016064 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o exp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 gisine ftgen 0, 0, 2^10, 10, 1 ;table for a sine wave instr 1 ;master instrument koct linseg 6, p3, 12 ; octave register straight rising from 6 to 12 kexp linseg 0, p3/3, 5, p3/3, 5, p3/3, 0 ;exponent goes from 0 to 5 and back kdens = exp(kexp) ;density is e to the power of kexp printks "Generated events per second: %d\n", 1, kdens ktrig metro kdens ;trigger single notes in kdens frequency if ktrig == 1 then ;call instr 10 for 1/kdens duration, .5 amplitude and koct register event "i", 10, 0, 1/kdens, .5, koct endif endin instr 10 ;performs one tone ioct rnd31 1, 0 ;random deviation maximum one octave plus/minus aenv transeg p4, p3, -6, 0 ;fast decaying envelope for p4 amplitude asin poscil aenv, cpsoct(p5+ioct), gisine ;sine for p5 octave register plus random deviation outs asin, asin endin i 1 0 30 e manual_src~dfsg/examples/resonxk.csd0000644000000000000000000000166412262561504016756 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o resonxk.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 0, 0, 2^10, 10, 1 instr 1 ksig randomh 400, 1800, 150 aout poscil .2, 1000+ksig, gisin outs aout, aout endin instr 2 ksig randomh 400, 1800, 150 kcf line 1, p3, 1000 ;vary high-pass ilay = p4 ksig resonxk ksig, kcf, 100, ilay aout poscil .2, 1000+ksig, gisin asig interp ksig ;convert k-rate to a-rate aout balance asig, aout ;avoid getting asig out of range outs aout, aout endin i 1 0 5 i 2 6 5 1 ;number of filter stack = 1 i 2 12 5 5 ;number of filter stack = 5 e manual_src~dfsg/examples/denorm.csd0000644000000000000000000000154412262561504016546 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o denorm.wav -W ;;; for file output any platform ; slightly simplified example from Istvan Varga 2006 sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 garvb init 0 instr 1 a1 oscils 0.6, 440, 0 a2 linsegr 0, 0.005, 1, 3600, 1, 0.08, 0 a1 = a1 * a2 vincr garvb, a1 outs a1, a1 endin instr 99 ;"Always on" denorm garvb aL, aR reverbsc garvb * 0.5, garvb * 0.5, 0.92, 10000 clear garvb outs aL, aR endin i 99 0 -1 ;held by a negative p3, means "always on" i 1 0 0.5 i 1 4 0.5 e 8 ;8 extra seconds after the performance manual_src~dfsg/examples/v.csd0000644000000000000000000000144112262561504015523 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o v.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .4*aenv, p4, 1 outs asig, asig endin f 1 0 16384 10 1 ;sine wave ;because note 3 and 5 are played simultaneously and are nearly of the same frequency, ;played together they will create a "beating" sound. i 1 0 2 110 ; note1 v2 i 1 3 . 220 ; note2 i 1 6 . 110 ; note3 v1 i 1 9 . 880 ; note4 i 1 12 . 100 ; note5 e manual_src~dfsg/examples/my.orc0000644000000000000000000000005512262561504015715 0ustar rootrootinstr 2 a1 oscils p4, p5, 0 out a1 endin manual_src~dfsg/examples/delay.csd0000644000000000000000000000150412262561504016354 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o delay.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 adel init 0 ilev = p4 ;level of direct sound idelay = p5 *.001 ;Delay in ms ifd = p6 ;feedback ain diskin2 "fox.wav", 1, 1 adel delay ain + (adel*ifd), idelay;ifd = amount of feedback asig moogvcf adel, 1500, .6, 1 ;color feedback outs asig*ilev, ain endin ;Delay is in ms i 1 0 15 2 200 .95 ;with feedback i 1 4 5 2 20 .95 i 1 + 3 2 5 .95 i 1 + 3 3 5 0 ;no feedback e manual_src~dfsg/examples/FLkeyIn.csd0000644000000000000000000000110012262561504016547 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O sr=44100 ksmps=128 nchnls=2 ;Example by Andres Cabrera 2007 FLpanel "FLkeyIn", 400, 300, -1, -1, 5, 1, 1 FLpanelEnd FLrun 0dbfs = 1 instr 1 kascii FLkeyIn ktrig changed kascii if (kascii > 0) then printf "Key Down: %i\n", ktrig, kascii else printf "Key Up: %i\n", ktrig, -kascii endif endin i 1 0 120 e manual_src~dfsg/examples/chuap.csd0000644000000000000000000000425412262561504016363 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chuas_oscillator.wav.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gibuzztable ftgen 1, 0, 16384, 10, 1 instr 1 ; sys_variables = system_vars(5:12); % L,R0,C2,G,Ga,Gb,E,C1 or p8:p15 ; integ_variables = [system_vars(14:16),system_vars(1:2)]; % x0,y0,z0,dataset_size,step_size or p17:p19, p4:p5 istep_size = p5 iL = p8 iR0 = p9 iC2 = p10 iG = p11 iGa = p12 iGb = p13 iE = p14 iC1 = p15 iI3 = p17 iV2 = p18 iV1 = p19 iattack = 0.02 isustain = p3 irelease = 0.02 p3 = iattack + isustain + irelease iscale = 1.0 adamping linseg 0.0, iattack, iscale, isustain, iscale, irelease, 0.0 aguide buzz 0.5, 440, sr/440, gibuzztable aI3, aV2, aV1 chuap iL, iR0, iC2, iG, iGa, iGb, iE, iC1, iI3, iV2, iV1, istep_size asignal balance aV2, aguide outs adamping * asignal, adamping * asignal endin ; Adapted from ABC++ MATLAB example data. i 1 0 20 1500 .1 -1 -1 -0.00707925 0.00001647 100 1 -.99955324 -1.00028375 1 -.00222159 204.8 -2.36201596260071 3.08917625807226e-03 3.87075614929199 7 .4 .004 1 86 30; torus attractor ( gallery of attractors ) i 1 + 20 1500 .425 0 -1 1.3506168 0 -4.50746268737 -1 2.4924 .93 1 1 0 -22.28662665 .009506608 -22.2861576 32 10 2 20 86 30 ; heteroclinic orbit i 1 + 20 1024 .05 -1 -1 0.00667 0.000651 10 -1 .856 1.1 1 .06 51.2 -20.200590133667 .172539323568344 -4.07686233520508 2.5 10 .2 1 66 81 ; periodic attractor (torus breakdown route) i 1 + 20 1024 0.05 -1 -1 0.00667 0.000651 10 -1 0.856 1.1 1 0.1 153.6 21.12496758 0.03001749 0.515828669 2.5 10 0.2 1 66 81 ; torus attractor (torus breakdown route)' manual_src~dfsg/examples/lpread.csd0000644000000000000000000000164712262561504016535 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpread.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; works with or without -a option when analyzing "fox.wav" from the manual ilen filelen "fox.wav" ; length of soundfile prints "fox.wav = %f seconds\\n",ilen ktime line 0, p3, p4 krmsr,krmso,kerr,kcps lpread ktime,"fox_poles.lpc" krmso = krmso*.00007 ; low volume aout buzz krmso, kcps, 15, 1 krmsr = krmsr*.0001 ; low volume asig rand krmsr outs (aout*2)+asig, (aout*2)+asig ; mix buzz and rand endin ; sine f1 0 4096 10 1 i 1 0 2.8 1 ; first words only i 1 4 2.8 2.8 ; whole sentence e manual_src~dfsg/examples/vmap.csd0000644000000000000000000000231712262561504016224 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vmap.wav -W ;;; for file output any platform ksmps = 256 nchnls = 2 gisize = 64 gitable ftgen 0, 0, gisize, 10, 1 ;Table to be processed gimap1 ftgen 0, 0, gisize, -7, gisize-1, gisize-1, 0 ; Mapping function to reverse table gimap2 ftgen 0, 0, gisize, -5, 1, gisize-1, gisize-1 ; Mapping function for PWM gimap3 ftgen 0, 0, gisize, -7, 1, (gisize/2)-1, gisize-1, 1, 1, (gisize/2)-1, gisize-1 ; Double frequency instr 1 ;Hear an oscillator using gitable asig oscil 10000, 440, gitable outs asig,asig endin instr 2 ;Reverse the table (no sound change, except for a single click vmap gimap1, gitable, gisize vcopy_i gitable, gimap1, gisize turnoff endin instr 3 ;Non-interpolated PWM (or phase waveshaping) vmap gimap2, gitable, gisize vcopy_i gitable, gimap2, gisize turnoff endin instr 4 ;Double frequency vmap gimap3, gitable, gisize vcopy_i gitable, gimap3, gisize turnoff endin i 1 0 8 i 2 2 1 i 3 4 1 i 4 6 1 e manual_src~dfsg/examples/ATSpartialtap.csd0000644000000000000000000000145312262561504017772 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc for RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSpartialtap.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 2 ATSbufread ktime, 1, "beats.ats", 30 kfreq1, kam1 ATSpartialtap 5 kfreq2, kam2 ATSpartialtap 20 kfreq3, kam3 ATSpartialtap 30 aout1 oscil kam1, kfreq1, 1 aout2 oscil kam2, kfreq2, 1 aout3 oscil kam3, kfreq3, 1 aout = (aout1+aout2+aout3)*10 ; amplify some more outs aout, aout endin ; sine wave. f 1 0 16384 10 1 i 1 0 2 e manual_src~dfsg/examples/pvcross.csd0000644000000000000000000000176012262561504016761 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvcross.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; analyze "beats.wav", "flute.aiff" and "mary.wav" with PVANAL first ktime1 line 0, p3, 2 ; used as index in the "beats.pvx" file ktime2 line 0, p3, 2.6 ; used as index in the "flute.pvx" or "mary.pvx" pvbufread ktime1, "beats.pvx" ;take only amplitude from "beats.pvx" if p4 = 0 then asig pvcross ktime2, 1, "flute.pvx", 1, 0 ;and keep freqs of "flute.aiff" asig = asig*.8 ;scale output else asig pvcross ktime2, 1, "mary.pvx", 1, 0 ;and keep freqs of "mary.wav" asig = asig*.4 ;scale output endif outs asig, asig endin i 1 0 3 0 i 1 + 3 1 e manual_src~dfsg/examples/resonr.csd0000644000000000000000000000522412262561504016571 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o resonr.wav -W ;;; for file output any platform /* Written by Sean Costello */ ; Orchestra file for resonant filter sweep of a sawtooth-like waveform. ; The outputs of reson, resonr, and resonz are scaled by coefficients ; specified in the score, so that each filter can be heard on its own ; from the same instrument. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 idur = p3 ibegfreq = p4 ; beginning of sweep frequency iendfreq = p5 ; ending of sweep frequency ibw = p6 ; bandwidth of filters in Hz ifreq = p7 ; frequency of gbuzz that is to be filtered iamp = p8 ; amplitude to scale output by ires = p9 ; coefficient to scale amount of reson in output iresr = p10 ; coefficient to scale amount of resonr in output iresz = p11 ; coefficient to scale amount of resonz in output ; Frequency envelope for reson cutoff kfreq linseg ibegfreq, idur * .5, iendfreq, idur * .5, ibegfreq ; Amplitude envelope to prevent clicking kenv linseg 0, .1, iamp, idur - .2, iamp, .1, 0 ; Number of harmonics for gbuzz scaled to avoid aliasing iharms = (sr*.4)/ifreq asig gbuzz 1, ifreq, iharms, 1, .9, 1 ; "Sawtooth" waveform ain = kenv * asig ; output scaled by amp envelope ares reson ain, kfreq, ibw, 1 aresr resonr ain, kfreq, ibw, 1 aresz resonz ain, kfreq, ibw, 1 out ares * ires + aresr * iresr + aresz * iresz endin /* Written by Sean Costello */ f1 0 8192 9 1 1 .25 ; cosine table for gbuzz generator i1 0 10 1 3000 200 100 4000 1 0 0 ; reson output with bw = 200 i1 10 10 1 3000 200 100 4000 0 1 0 ; resonr output with bw = 200 i1 20 10 1 3000 200 100 4000 0 0 1 ; resonz output with bw = 200 i1 30 10 1 3000 50 200 8000 1 0 0 ; reson output with bw = 50 i1 40 10 1 3000 50 200 8000 0 1 0 ; resonr output with bw = 50 i1 50 10 1 3000 50 200 8000 0 0 1 ; resonz output with bw = 50 e manual_src~dfsg/examples/jitter.csd0000644000000000000000000000124712262561504016563 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o jitter.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp init p4 kcpsmin init 4 kcpsmax init 8 kj2 jitter kamp, kcpsmin, kcpsmax aout pluck 1, 200+kj2, 1000, 0, 1 aout dcblock aout ;remove DC outs aout, aout endin i 1 0 15 2 ;a bit jitter i 1 8 15 10 ;some more i 1 16 15 20 ;lots more e manual_src~dfsg/examples/impuls20.aiff0000644000000000000000000000111212262561504017060 0ustar rootrootFORMJAIFF(c) |0COMM@ DSSNDzjS7^C w c,,I vKJV2hrLQB 4l6>!DUf b@Cm4`N( m'*'+64oo46+'*'m (N`4mC@b fUD!>6l4 BQLrh2VJKv I,,c w C^7Sjzmanual_src~dfsg/examples/vdelayxw.csd0000644000000000000000000000152112262561504017120 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelayxw.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ims = .5 ;maximum delay time in seconds iws = 1024 ;best quality adl = .5 ;delay time asig diskin2 "flute.aiff", .5, 0, 1 ;loop flute.aiff at half speed a2 poscil3 .2, .1, 1 ;make an LFO, 1 cycle per 2 seconds adl = a2 + ims/2 ;offset the LFO so that it is positive aout vdelayxw asig, adl, ims, iws ;use the LFO to control delay time outs aout, aout endin f1 0 8192 10 1 i 1 0 10 e manual_src~dfsg/examples/tonek.csd0000644000000000000000000000136612262561504016404 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tonek.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 0, 0, 2^10, 10, 1 instr 1 ksig randomh 400, 1800, 150 aout poscil .2, 100+ksig, gisin outs aout, aout endin instr 2 ksig randomh 400, 1800, 150 khp line 1, p3, 100 ;vary high-pass ksig tonek ksig, khp aout poscil .2, 100+ksig, gisin outs aout, aout endin i 1 0 5 i 2 5.5 5 e manual_src~dfsg/examples/dam.csd0000644000000000000000000000133512262561504016021 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dam.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;normal audio asig diskin2 "beats.wav", 1, 0, 1 outs asig, asig endin instr 2 ; compressed audio kthreshold = 0.2 icomp1 = 0.8 icomp2 = 0.2 irtime = 0.01 iftime = 0.5 asig diskin2 "beats.wav", 1, 0, 1 asig dam asig, kthreshold, icomp1, icomp2, irtime, iftime outs asig, asig endin i 1 0 2 i 2 2.5 8.5 e manual_src~dfsg/examples/tabmorphak.csd0000644000000000000000000000221012262561504017401 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tabmorphak.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 giSine ftgen 0, 0, 8193, 10, 1 ; sine wave giSquare ftgen 0, 0, 8193, 7, 1, 4096, 1, 0, -1, 4096, -1 ; square wave giTri ftgen 0, 0, 8193, 7, 0, 2048, 1, 4096, -1, 2048, 0 ; triangle wave giSaw ftgen 0, 0, 8193, 7, 1, 8192, -1 ; sawtooth wave, downward slope instr 1 iamp = .7 aindex phasor 110 ; read table value at this index kweightpoint expon 0.001, p3, 1 ; using the weightpoint to morph between two tables exponentially ktabnum1 = p4 ; first wave, it morphs to ktabnum2 = p5 ; the second wave asig tabmorphak aindex, kweightpoint, ktabnum1,ktabnum2, giSine, giSquare, giTri, giSaw asig = asig*iamp outs asig, asig endin i1 0 5 0 1 ;from sine to square wave i1 6 5 2 3 ;from triangle to saw e manual_src~dfsg/examples/beats.mp30000644000000000000000000004051512262561504016307 0ustar rootrootPInfoOAM  #&&*-0337:==@DGJJMQTWWZ]addgjnnqtw{{~9LAME3.96r.H@$B@AMu \P V r ` 1r1Y:4hB!ww-"""!uxI$TBB}[t\E߾lE~OԵ$ۻ]=V5ʜseӧ7 5˛ȴ,( 0AoGAK~[ci;aR q19w.tIF;$(q}UBߓe}viwoOc#iO'r|Jk gbDUu˴FLXC)es@FvhpE$B6ZVfc-`MEFJLe+r1a1ǚ{XMW3kɞLEm9R H5d{'@_6ІƱU Lirb@iq֮o%C({S2ZL6o|䀵A ߿րjt0[h <"33pk:N ɑmLV[392_G,zshP%k1*{w9߈]R ;{`u'o4ЌOTyC9 1*4*lDj['$VLܶ$Ǧ_iM8#Ww<'\m5 Vr~zueۭF >sHv)PFNCΖs5/˫HlLr r*T{Dr8q.;|^pΞpVYr?^^bOjdzWhR ioaag1px ͘IvLs)_"q `BȮNEL.d TjM!^4{asG/G#LY7de V &(CaV\GU^<܇4dS9QJ2;LNG'%˩F(J*7]i8h╮R 8i`Q_0|anKyc }œ*vt̩"(}#O38xAX -J=z,(C0R@ZLW:4YFӷ^w~el1~;;0XgJ'>w|ƐHuvJ\^ kOtm"hLg,BEwersЍ0_xE[#ǜȔn#1vF73H:Y7 3m͏54(afcWs{Vųlƾ3f}QOͤP&  L8 R E\E4B߼T/ !hcDD>* U b#rX& *@P(@n'' ?'%E~? 96Odw@АsO@$ńf11eW7k]_R9!Pj70\U.;$5v?HnS|'H$dD:I `2xQ%DRm q!vF0FiZyvж<̊AokWk%dq t3{._.J݃W:X>Aۗ1'}6/wL2"1hیZ"i*mS#"< Ie%Hϒ'uyNa47t6ƹmQwjkRx omqJ! xD"$ Lu^^[5 䣴7=3 sT]N>G\auUEC7" CFVwiILY3 cĠ1Q9.c_5'~O|ai_ovyo}>pƙD˙PrFK$ Rą $mi=-1yt7x QAAzx0B=݈QC U&9Dim#|8wBTŴ] =YK1O8Diymn@*D>diQC q#Jyup*7ͥY:*!8iäԂa<-1uj2>9q&$c;VϹ!~lmK˄d|;8,ή L[BKۛ5 y?G;wL{QF{{HY:y~`%19+L(qSU=p˃RgbRĞ tebKaxÈ$ѓ+PЎ|U PdR4CIkIIjW,{cw X[* Gh5>ϙg=XMv߷N{J〞)w13sV%Iv>fgF#=$I%]cUw$kq"HH"D^g^fg?yF"FEp$P^Rĩ |_a ^ 𠠚LAME3.96.1LAME3.96.1Rč̑2c41 V2и "Pl>x0p1X.0# p_@P @X8(0o"àR 8l N&\"`4)ARĸ 4`X4E^Nv&BoDWu(}z. 8UL2@>qA8V(2Lw9FMߢh0N@!N4gD``oH)u i@3  #I0 \@aR 4l- !A@pl a"LhE%̟@p ({!&Or$0xC:L10$\5&* Cek>:쬧+,oS#,XV>lоFUĮH`i>#fh(RZC3$rJz-5vfʔaQU)4?fT`!u_Sٷr_3dHdO.1U疨ԾR1~YWXf ܅S>LZ5deTB3J|6G|m\R6]58F  q؁921B1GS[~S2|ͫ|IDǍHɂh*? 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񢛝&zUӕe4z ie NȂIi?SHEJ #FMR5 !aqjR.`aamNB MW"uy(sfV2l p4uP̠p[~|ub^2eR< maT<`sa '=3TZ wê{ ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o scantable.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gi1 ftgen 1, 0, 128, 7, 0, 64, 1, 64, 0 ; initial position gi2 ftgen 2, 0, 128, -7, 1, 128, 1 ; masses gi3 ftgen 3, 0, 128, -7, 0, 64, 100, 64, 0 ; stiffness gi4 ftgen 4, 0, 128, -7, 1, 128, 1 ; damping gi5 ftgen 5, 0, 128, -7, 0, 128, 0.5 ; initial velocity instr 1 iamp ampmidi .5 ipch cpsmidi kenv madsr .1, .1, .8, .3 asig scantable iamp, ipch, 1, 2, 3, 4, 5 asig dcblock asig outs asig*kenv, asig*kenv endin f0 60 ; play for 60 seconds e manual_src~dfsg/examples/vbapmove.csd0000644000000000000000000000123412262561504017075 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbapmove.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 vbaplsinit 2, 4, 0, 90, 180, 270 instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav a1,a2,a3,a4 vbapmove asig, p3, 1, 2, 310, 180 ;change movement of soundsource in outq a1,a2,a3,a4 ;the rear speakers endin i 1 0 5 e manual_src~dfsg/examples/tablekt.csd0000644000000000000000000000174512262561504016713 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tablekt.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 gift1 ftgen 1, 0, 1024, 10, 1 ;sine wave gift2 ftgen 2, 0, 1024, 10, 1, 1, 1, 1, 1, 1, 1, 1, 1 ;pulse instr 1 andx phasor 400 ;phasor for reading the index kfn init 1 ;initialize the choice of the function table kmetro init 1 ;initialize the frequency of the metro knewft metro kmetro ;make a new choice for selecting the function table once a second if knewft == 1 then kfn = (kfn == 1 ? 2 : 1) ;switch between 1 and 2 kmetro random .5, 2 ;create new metro frequency printk2 kfn endif ares tablekt andx, kfn, 1 outs ares, ares endin i 1 0 10 e manual_src~dfsg/examples/FLknob-2.csd0000644000000000000000000000347512262561504016601 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLknob.wav -W ;;; for file output any platform sr = 44100 kr = 441 ksmps = 100 nchnls = 1 ;By Andres Cabrera 2007 FLpanel "Knob Types", 330, 230, 50, 50 ; Distance of the left edge of the knob ; from the left edge of the panel ix = 20 ; Distance of the top edge of the knob ; from the top of the panel iy = 20 ;Create boxes that display a widget's value ihandleA FLvalue "A", 60, 20, ix + 130, iy + 110 ihandleB FLvalue "B", 60, 20, ix + 220, iy + 110 ihandleC FLvalue "C", 60, 20, ix + 130, iy + 160 ihandleD FLvalue "D", 60, 20, ix + 220, iy + 160 ; The foru types of FLknobs gkdummy1, ihandle1 FLknob "Type 1", 200, 5000, -1, 1, ihandleA, 70, ix, iy, 90 gkdummy2, ihandle2 FLknob "Type 2", 200, 5000, -1, 2, ihandleB, 70, ix + 100, iy gkdummy3, ihandle3 FLknob "Type 3", 200, 5000, -1, 3, ihandleC, 70, ix + 200, iy gkdummy4, ihandle4 FLknob "Type 4", 200, 5000, -1, 4, ihandleD, 70, ix , iy + 100 ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun ; Set the color of widgets FLsetColor 20, 23, 100, ihandle1 FLsetColor 0, 123, 100, ihandle2 FLsetColor 180, 23, 12, ihandle3 FLsetColor 10, 230, 0, ihandle4 FLsetColor2 200, 230, 0, ihandle1 FLsetColor2 200,0 ,123 , ihandle2 FLsetColor2 180, 180, 100, ihandle3 FLsetColor2 180, 23, 12, ihandle4 ; Set the initial value of the widget FLsetVal_i 300, ihandle1 FLsetVal_i 1000, ihandle2 instr 1 ; Nothing here for now endin f 0 3600 ;Dumy table to make csound wait for realtime events e manual_src~dfsg/examples/ftgentmp.csd0000644000000000000000000000051012262561504017076 0ustar rootroot sr = 44100 ksmps = 128 nchnls = 2 0dbfs = 1 instr 1 ifno ftgentmp 0, 0, 512, 10, 1 print ifno endin instr 2 print ftlen(p4) endin i 1 0 10 i 2 2 1 101 i 1 5 10 i 2 7 1 102 i 2 12 1 101 i 2 17 1 102 e manual_src~dfsg/examples/mandel.csd0000644000000000000000000000570712262561504016527 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mandel.wav -W ;;; for file output any platform ;example by Brian Evans sr = 44100 ksmps = 32 nchnls = 2 instr 1 ipitchtable = 1 ; pitch table in score ipitchndx = p5 ; p5=pitch index from table ipitch table ipitchndx, ipitchtable kenv expseg 1.0, 1.0, 1.0, 11.5, .0001 asig pluck ampdb(p4)*kenv, cpspch(ipitch), cpspch(ipitch), 0, 1 outs asig, asig endin f1 0 32 -2 6.00 6.02 6.04 6.05 6.07 6.09 6.11 ; f1 is a pitch table defining a four octave C major scale starting 7.00 7.02 7.04 7.05 7.07 7.09 7.11 ; on C two octaves below middle C 8.00 8.02 8.04 8.05 8.07 8.09 8.11 9.00 9.02 9.04 9.05 9.07 9.09 9.11 ;ins start dur ampdb(p4) pitchndx(p5) i1 0 12.0 75 3 i1 1.5999 12.0 75 4 i1 3.4000 12.0 75 5 i1 4.2000 12.0 75 6 i1 4.4000 12.0 75 7 i1 4.6000 12.0 75 9 i1 4.8000 12.0 75 10 i1 5.0000 12.0 75 5 i1 5.2000 12.0 75 27 i1 5.4000 12.0 75 5 i1 5.6000 12.0 75 20 i1 6.0000 12.0 75 24 i1 6.2000 12.0 75 2 i1 6.4000 12.0 75 27 i1 6.6000 12.0 75 20 i1 6.8000 12.0 75 15 i1 7.0000 12.0 75 3 i1 7.2000 12.0 75 3 i1 7.4000 12.0 75 23 i1 7.6000 12.0 75 9 i1 7.8000 12.0 75 17 i1 8.0000 12.0 75 18 i1 8.2000 12.0 75 3 i1 8.4000 12.0 75 26 i1 8.6000 12.0 75 15 i1 8.8 12.0 75 2 i1 9 12.0 75 26 i1 9.2 12.0 75 8 i1 9.3999 12.0 75 22 i1 9.5999 12.0 75 22 i1 9.7999 12.0 75 20 i1 9.9999 12.0 75 19 i1 10.399 12.0 75 20 i1 10.799 12.0 75 22 i1 10.999 12.0 75 27 i1 11.199 12.0 75 25 i1 11.399 12.0 75 20 i1 11.599 12.0 75 21 i1 11.799 12.0 75 24 i1 11.999 12.0 75 24 i1 12.199 12.0 75 4 i1 12.399 12.0 75 13 i1 12.599 12.0 75 15 i1 12.799 12.0 75 14 i1 12.999 12.0 75 3 i1 13.199 12.0 75 21 i1 13.399 12.0 75 6 i1 13.599 12.0 75 3 i1 13.799 12.0 75 10 i1 13.999 12.0 75 25 i1 14.199 12.0 75 21 i1 14.399 12.0 75 20 i1 14.599 12.0 75 19 i1 14.799 12.0 75 18 i1 15.199 12.0 75 17 i1 15.599 12.0 75 16 i1 15.999 12.0 75 15 i1 16.599 12.0 75 14 i1 17.199 12.0 75 13 i1 18.399 12.0 75 12 i1 18.599 12.0 75 11 i1 19.199 12.0 75 10 i1 19.799 12.0 75 9 e manual_src~dfsg/examples/fluidOut.csd0000644000000000000000000000307012262561504017051 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidOut.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine1 fluidEngine ; soundfont path to manual/examples isfnum1 fluidLoad "01hpschd.sf2", giengine1, 1 fluidProgramSelect giengine1, 1, isfnum1, 0, 0 giengine2 fluidEngine ; soundfont path to manual/examples isfnum2 fluidLoad "22Bassoon.sf2", giengine2, 1 fluidProgramSelect giengine2, 1, isfnum2, 0, 70 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine1, 1, ikey, ivel endin instr 2 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine2, 1, ikey, ivel endin instr 98 imvol init 7 asigl, asigr fluidOut giengine1 outs asigl*imvol, asigr*imvol endin instr 99 imvol init 4 asigl, asigr fluidOut giengine2 ;add a stereo flanger adelL linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time =20ms adelR linseg 0.02, p3*.5, 0, p3*.5, 0.02 ;max delay time =20ms asigL flanger asigl, adelL, .6 asigR flanger asigr, adelR, .6 outs asigL*imvol, asigR*imvol endin i 1 0 2 60 100 ;play one note of instr 1 i 2 2 2 60 100 ;play another note of instr 2 and... i 98 0 60 ;play virtual keyboard for 60 sec. i 99 0 60 e manual_src~dfsg/examples/active_k.csd0000644000000000000000000000225512262561504017047 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o active_k.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a noisy waveform. instr 1 ; Generate a really noisy waveform. anoisy rand 44100 ; Turn down its amplitude. aoutput gain anoisy, 2500 ; Send it to the output. out aoutput endin ; Instrument #2 - counts active instruments at k-rate. instr 2 ; Count the active instances of Instrument #1. kcount active 1 ; Print the number of active instances. printk2 kcount endin ; Start the first instance of Instrument #1 at 0:00 seconds. i 1 0.0 3.0 ; Start the second instance of Instrument #1 at 0:015 seconds. i 1 1.5 1.5 ; Play Instrument #2 at 0:01 seconds, when we have only ; one active instance of Instrument #1. i 2 1.0 0.1 ; Play Instrument #2 at 0:02 seconds, when we have ; two active instances of Instrument #1. i 2 2.0 0.1 e manual_src~dfsg/examples/tabmorph.csd0000644000000000000000000000214612262561504017075 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tabmorph.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 8193, 10, 1 ;sine wave giSquare ftgen 0, 0, 8193, 7, 1, 4096, 1, 0, -1, 4096, -1 ;square wave giTri ftgen 0, 0, 8193, 7, 0, 2048, 1, 4096, -1, 2048, 0 ;triangle wave giSaw ftgen 0, 0, 8193, 7, 1, 8192, -1 ;sawtooth wave, downward slope instr 1 iamp = .7 kindex phasor 440 ;read table value at this index kindex = kindex*8192 ;for all 8192 index points kweightpoint = 0.5 ;set weightpoint ktabnum1 line 0, p3, 3 ;morph through all tables ktabnum2 = 2 ;set to triangle wave ksig tabmorph kindex, kweightpoint, ktabnum1, ktabnum2, giSine, giSquare, giTri, giSaw ksig = ksig*iamp asig interp ksig ;convert to audio signal outs asig, asig endin i1 0 5 e manual_src~dfsg/examples/readscratch.csd0000644000000000000000000000063312262561504017543 0ustar rootroot ; Select audio/midi flags here according to platform -n sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ilast readscratch writescratch p2 ilen readscratch 1 writescratch p3, 1 printf_i "last run at %f for %f\n", ilen, ilast, ilen endin i 1 0 1 i 1 2 3 i 1 6 10 e manual_src~dfsg/examples/rspline.csd0000644000000000000000000000133112262561504016730 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o jspline.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 krangeMin init p4 krangeMax init p5 kcpsmin init 2 kcpsmax init 3 ksp rspline krangeMin, krangeMax, kcpsmin, kcpsmax aout pluck 1, 200+ksp, 1000, 0, 1 aout dcblock aout ;remove DC outs aout, aout endin i 1 0 10 2 5 ;a bit jitter i 1 8 10 10 20 ;some more i 1 16 10 20 50 ;lots more e manual_src~dfsg/examples/lorenz.csd0000644000000000000000000000234512262561504016573 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o lorenz.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 2 ; Instrument #1 - a lorenz system in 3D space. instr 1 ; Create a basic tone. kamp init 25000 kcps init 1000 ifn = 1 asnd oscil kamp, kcps, ifn ; Figure out its X, Y, Z coordinates. ksv init 10 krv init 28 kbv init 2.667 kh init 0.0003 ix = 0.6 iy = 0.6 iz = 0.6 iskip = 1 ax1, ay1, az1 lorenz ksv, krv, kbv, kh, ix, iy, iz, iskip ; Place the basic tone within 3D space. kx downsamp ax1 ky downsamp ay1 kz downsamp az1 idist = 1 ift = 0 imode = 1 imdel = 1.018853416 iovr = 2 aw2, ax2, ay2, az2 spat3d asnd, kx, ky, kz, idist, \ ift, imode, imdel, iovr ; Convert the 3D sound to stereo. aleft = aw2 + ay2 aright = aw2 - ay2 outs aleft, aright endin ; Table #1 a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 5 seconds. i 1 0 5 e manual_src~dfsg/examples/ftgen-2.csd0000644000000000000000000000160712262561504016524 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ftgen-2.wav -W ;;; for file output any platform sr = 48000 ksmps = 16 nchnls = 2 ;Example by Jonathan Murphy 2007 0dbfs = 1 instr 1 Sfile = "beats.wav" ilen filelen Sfile ; Find length isr filesr Sfile ; Find sample rate isamps = ilen * isr ; Total number of samples isize init 1 loop: isize = isize * 2 ; Loop until isize is greater than number of samples if (isize < isamps) igoto loop itab ftgen 0, 0, isize, 1, Sfile, 0, 0, 0 print isize print isamps turnoff endin i1 0 10 e manual_src~dfsg/examples/vpvoc.csd0000644000000000000000000000130312262561504016410 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vpvoc.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; analyze "fox.wav" with PVANAL first iend = p4 ktime line 0, p3, iend tablexseg p5, p3, p6 ;morph from table 1 asig vpvoc ktime, 1, "fox.pvx" ;to table 2 outs asig*3, asig*3 endin f 1 0 512 9 .5 1 0 f 2 0 512 5 1 60 0.01 390 0.01 62 1 i 1 0 5 2.7 1 2 e manual_src~dfsg/examples/ampmidid.csd0000644000000000000000000000165312262561504017047 0ustar rootroot ; Select audio/midi flags here according to platform ;;;RT audio out, note=p4 and velocity=p5 -odac --midi-key=4 --midi-velocity-amp=5 ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ampmidid.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 massign 0, 1 ;assign all midi to instr. 1 instr 1 isine ftgenonce 0, 0, 4096, 10, 1 ;sine wave ihz = cpsmidinn(p4) ivelocity = p5 idb ampmidid ivelocity, 20 ;map to dynamic range of 20 dB. idb = idb + 60 ;limit range to 60 to 80 decibels iamplitude = ampdb(idb) ;loudness in dB to signal amplitude a1 oscili iamplitude, ihz, isine aenv madsr 0.05, 0.1, 0.5, 0.2 asig = a1 * aenv outs asig, asig endin ; note velocity i 1 0 2 61 100 i 1 + 2 65 10 e manual_src~dfsg/examples/endif.csd0000644000000000000000000000210712262561504016343 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o endif.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Get the note value from the fourth p-field. knote = p4 ; Does the user want a low note? if (knote == 0) then kcps = 220 ; Does the user want a middle note? elseif (knote == 1) then kcps = 440 ; Does the user want a high note? elseif (knote == 2) then kcps = 880 endif ; Create the note. kamp init .8 ifn = 1 a1 oscili kamp, kcps, ifn outs a1, a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; p4: 0=low note, 1=middle note, 2=high note. ; Play Instrument #1 for one second, low note. i 1 0 1 0 ; Play Instrument #1 for one second, middle note. i 1 1 1 1 ; Play Instrument #1 for one second, high note. i 1 2 1 2 e manual_src~dfsg/examples/FLsetcolor.csd0000644000000000000000000000226612262561504017340 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLsetcolor.wav -W ;;; for file output any platform ; Using the opcode flsetcolor to change from the ; default colours for widgets sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Coloured Sliders", 900, 360, 50, 50 gkfreq, ihandle FLslider "A Red Slider", 200, 5000, -1, 5, -1, 750, 30, 85, 50 ired1 = 255 igreen1 = 0 iblue1 = 0 FLsetColor ired1, igreen1, iblue1, ihandle gkfreq, ihandle FLslider "A Green Slider", 200, 5000, -1, 5, -1, 750, 30, 85, 150 ired1 = 0 igreen1 = 255 iblue1 = 0 FLsetColor ired1, igreen1, iblue1, ihandle gkfreq, ihandle FLslider "A Blue Slider", 200, 5000, -1, 5, -1, 750, 30, 85, 250 ired1 = 0 igreen1 = 0 iblue1 = 255 FLsetColor ired1, igreen1, iblue1, ihandle ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 endin ; 'Dummy' score event for 1 hour. f 0 3600 e manual_src~dfsg/examples/printf.csd0000644000000000000000000000133512262561504016562 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o printf.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 Sfile strget p4 ivld filevalid Sfile if ivld=0 then printf_i "Audiofile '%s' does not exist!\n", 1, Sfile else asig diskin2 Sfile, 1 outs asig, asig endif endin i 1 0 3 "frox.wav";file does not exist!!! i 1 + 3 "fox.wav";but this one certainly does... e manual_src~dfsg/examples/planet.csd0000644000000000000000000000250612262561504016544 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o planet.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 2 ; Instrument #1 - a planet oribiting in 3D space. instr 1 ; Create a basic tone. kamp init 5000 kcps init 440 ifn = 1 asnd oscil kamp, kcps, ifn ; Figure out its X, Y, Z coordinates. km1 init 0.5 km2 init 0.35 ksep init 2.2 ix = 0 iy = 0.1 iz = 0 ivx = 0.5 ivy = 0 ivz = 0 ih = 0.0003 ifric = -0.1 ax1, ay1, az1 planet km1, km2, ksep, ix, iy, iz, \ ivx, ivy, ivz, ih, ifric ; Place the basic tone within 3D space. kx downsamp ax1 ky downsamp ay1 kz downsamp az1 idist = 1 ift = 0 imode = 1 imdel = 1.018853416 iovr = 2 aw2, ax2, ay2, az2 spat3d asnd, kx, ky, kz, idist, \ ift, imode, imdel, iovr ; Convert the 3D sound to stereo. aleft = aw2 + ay2 aright = aw2 - ay2 outs aleft, aright endin ; Table #1 a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 10 seconds. i 1 0 10 e manual_src~dfsg/examples/flooper2.csd0000644000000000000000000000130612262561504017006 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o flooper2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; looping back and forth, 0.05 crossfade kst line .2, p3, 2 ;vary loopstartpoint aout flooper2 .8, 1, 0, kst, 0.05, 1, 0, 2 outs aout, aout endin ; Its table size is deferred, ; and format taken from the soundfile header f 1 0 0 1 "fox.wav" 0 0 0 i 1 0 12 e manual_src~dfsg/examples/tlineto.csd0000644000000000000000000000145712262561504016743 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tlineto.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kmtr lfo 1, .5, 1 ;produce trigger signal ktr trigger kmtr, .5, 0 ;with triangle wave ktime = p4 kfreq randh 1000, 3, .2, 0, 500 ;generate random values kfreq tlineto kfreq, ktime, ktr ;different glissando times aout poscil .4, kfreq, giSine outs aout, aout endin i 1 0 10 .2 ;short glissando i 1 11 10 .8 ;longer glissande e manual_src~dfsg/examples/0dbfs.csd0000644000000000000000000000167712262561504016267 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o 0dbfs.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 ksmps = 10 nchnls = 2 ; Set the 0dbfs to 1. 0dbfs = 1 instr 1 ; from linear amplitude (0-1 range) print p4 a1 oscil p4, 440, 1 outs a1, a1 endin instr 2 ; from linear amplitude (0-32767 range) iamp = p4 / 32767 print iamp a1 oscil iamp, 440, 1 outs a1, a1 endin instr 3 ; from dB FS iamp = ampdbfs(p4) print iamp a1 oscil iamp, 440, 1 outs a1, a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 i 1 0 1 1 i 1 + 1 0.5 i 1 + 1 0.1 s i 2 0 1 32767 i 2 + 1 [32767/2] i 2 + 1 [3276.7] s i 3 0 1 0 i 3 + 1 -6 i 3 + 1 -20 e manual_src~dfsg/examples/goto.csd0000644000000000000000000000130412262561504016224 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o goto.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 a1 oscil 10000, 440, 1 goto playit ; The goto will go to the playit label. ; It will skip any code in between like this comment. playit: out a1 endin ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/outch.csd0000644000000000000000000000135412262561504016403 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outch.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 5 0dbfs = 1 instr 1 asig vco2 .05, 100 ; sawtooth waveform at low volume kcut line 100, p3, 30 ; Vary cutoff frequency kresonance = .7 inumlayer = 3 asig lowresx asig, kcut, kresonance, inumlayer klfo lfo 4, .5, 4 klfo = klfo+1 ; offset of 1 printks "signal is sent to channel %d\\n", .1, klfo outch klfo,asig endin i 1 0 30 e manual_src~dfsg/examples/pvsbandp.csd0000644000000000000000000000133412262561504017074 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsbandp.wav -W ;;; for file output any platform sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 Sfile = "fox.wav" klowcut = 100 klowfull = 200 khighfull = 1900 khighcut = 2000 ain soundin Sfile fftin pvsanal ain, 1024, 256, 1024, 1; fft-analysis of the audio-signal fftbp pvsbandp fftin, klowcut, klowfull, khighfull, khighcut ; band pass abp pvsynth fftbp; resynthesis out abp endin i 1 0 3 e manual_src~dfsg/examples/poscil3.csd0000644000000000000000000000123212262561504016630 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o poscil3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 krnd randomh 40, 440, 1 ; produce random values ain poscil3 .6, krnd, giSine kline line 1, p3, 0 ; straight line aL,aR pan2 ain, kline ; sent across image outs aL, aR endin i1 0 10 e manual_src~dfsg/examples/vbapg.csd0000644000000000000000000000142512262561504016357 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbap4.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 vbaplsinit 2.01, 4, 0, 90, 180, 270 instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav kaz line 0, p3, p4 ;come from right rear speaker & k1,k2,k3,k4 vbapg 180, 100, kaz, 1 ;change spread of soundsource printks "spread of source = %d\n", 1, kaz ;print spread value outq asig*k1,asig*k2,asig*k3,asig*k4 endin i 1 0 12 100 e manual_src~dfsg/examples/noise.csd0000644000000000000000000000110412262561504016367 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o noise.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kbeta line -1, p3, 1 ;change beta value from -1 to 1 asig noise .3, kbeta asig clip asig, 2, .9 ;clip signal outs asig, asig endin i 1 0 10 e manual_src~dfsg/examples/pvinterp.csd0000644000000000000000000000141712262561504017130 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvinterp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; analyze "fox.wav" and "flute.aiff" with PVANAL first ktime1 line 0, p3, 2.8 ; used as index in the "fox.pvx" file ktime2 line 0, p3, 3 ; used as index in the "flute.pvx" file kinterp line 1, p3, 0 pvbufread ktime1, "fox.pvx" asig pvinterp ktime2,1,"flute.pvx",.9, 3, .6, 1, kinterp,1-kinterp outs asig, asig endin i 1 0 3 i 1 + 10 e manual_src~dfsg/examples/dcblock.csd0000644000000000000000000000124312262561504016657 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o dcblock.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 ;add DC to "beats.wav" asig soundin "beats.wav" asig = asig+5000 ;adds DC of 5000 outs asig, asig endin instr 2 ;dcblock audio asig soundin "beats.wav" asig = asig+5000 ;adds DC adc dcblock asig ;remove DC again outs adc, adc endin i 1 0 2 i 2 2 2 e manual_src~dfsg/examples/fprintks.csd0000644000000000000000000000154412262561504017122 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o fprintks.wav -W ;;; for file output any platform /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a score generator example. instr 1 ; K-rate stuff. kstart init 0 kdur linrand 10 kpitch linrand 8 ; Printing to to a file called "my.sco". fprintks "my.sco", "i1\\t%2.2f\\t%2.2f\\t%2.2f\\n", kstart, kdur, 4+kpitch knext linrand 1 kstart = kstart + knext endin /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Play Instrument #1. i 1 0 0.001 manual_src~dfsg/examples/cpsoct.csd0000644000000000000000000000120112262561504016543 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpsoct.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Convert octave-point-decimal value into Hz ioct = p4 icps = cpsoct(ioct) print icps asig oscil 0.7, icps, 1 outs asig, asig endin ;sine wave. f 1 0 16384 10 1 i 1 0 1 8.75 i 1 + 1 8.77 i 1 + 1 8.79 i 1 + .5 6.30 e manual_src~dfsg/examples/vibr.csd0000644000000000000000000000116512262561504016223 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vibr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kaverageamp init 500 kaveragefreq init 4 kvib vibr kaverageamp, kaveragefreq, 1 asig poscil .8, 220+kvib, 1 ;add vibrato outs asig, asig endin f 1 0 16384 10 1 ;sine wave i 1 0 10 e manual_src~dfsg/examples/clfilt_highpass.csd0000644000000000000000000000127312262561504020424 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o clfilt_highpass.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; white noise asig rand 0.6 outs asig, asig endin instr 2 ;filtered noise asig rand 0.7 ; Highpass filter signal asig with a 6-pole Chebyshev ; Type I at 20 Hz with 3 dB of passband ripple. a1 clfilt asig, 20, 1, 6, 1, 3 outs a1, a1 endin i 1 0 2 i 2 2 2 e manual_src~dfsg/examples/jacktransport.csd0000644000000000000000000000066112262561504020146 0ustar rootroot -+rtaudio=JACK -b 64 --sched -o dac:system:playback_ sr = 44100 ksmps = 16 nchnls = 2 instr 1 jacktransport p4, p5 endin instr 2 jacktransport p4 endin i2 0 5 1; play i2 5 1 0; stop i1 6 5 1 2 ; move at 2 seconds and start playing back i1 11 1 0 0 ; stop and rewind e manual_src~dfsg/examples/cpspch.csd0000644000000000000000000000117312262561504016540 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpspch.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Convert pitch-class value into Hz ipch = p4 icps = cpspch(ipch) print icps asig oscil 0.7, icps, 1 outs asig, asig endin ;sine wave. f 1 0 16384 10 1 i 1 0 1 8.01 i 1 + 1 8.02 i 1 + 1 8.03 i 1 + .5 5.09 e manual_src~dfsg/examples/wrap.csd0000644000000000000000000000204412262561504016227 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o wrap.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; Limit / Mirror / Wrap igain = p4 ;gain ilevl1 = p5 ; + level ilevl2 = p6 ; - level imode = p7 ;1 = limit, 2 = mirror, 3 = wrap ain soundin "fox.wav" ain = ain*igain if imode = 1 goto limit if imode = 2 goto mirror asig wrap ain, ilevl2, ilevl1 goto outsignal limit: asig limit ain, ilevl2, ilevl1 goto outsignal mirror: asig mirror ain, ilevl2, ilevl1 outsignal: outs asig*.5, asig*.5 ;mind your speakers endin ; Gain +Levl -Levl Mode i1 0 3 4.00 .25 -1.00 1 ;limit i1 4 3 4.00 .25 -1.00 2 ;mirror i1 8 3 4.00 .25 -1.00 3 ;wrap e manual_src~dfsg/examples/madsr.csd0000644000000000000000000000160212262561504016363 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o madsr.wav -W ;;; for file output any platform /* Written by Iain McCurdy */ ; Initialize the global variables. sr = 44100 kr = 441 ksmps = 100 nchnls = 1 ; Instrument #1. instr 1 ; Attack time. iattack = 0.5 ; Decay time. idecay = 0 ; Sustain level. isustain = 1 ; Release time. irelease = 0.5 aenv madsr iattack, idecay, isustain, irelease a1 oscili 10000, 440, 1 out a1*aenv endin /* Written by Iain McCurdy */ ; Table #1, a sine wave. f 1 0 1024 10 1 ; Leave the score running for 6 seconds. f 0 6 ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/t.csd0000644000000000000000000000147512262561504015530 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o t.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .8*aenv, p4, 1 outs asig, asig endin f 1 0 16384 10 1 ;sine wave t 0 240 12 30 15 240 ;start tempo = 240 i 1 0 2 110 ;tempo = 240 i 1 3 2 220 ;slow down & i 1 6 2 440 ;slow down & i 1 9 2 880 ;slow down & i 1 12 2 110 ;slow down to 30 at 12 seconds i 1 15 2 220 ;speed up to 240 again i 1 18 2 440 ;stay at tempo 240 i 1 21 2 880 e manual_src~dfsg/examples/timek.csd0000644000000000000000000000114112262561504016364 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o timek.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the value from timek every half-second. k1 timek printks "k1 = %f samples\\n", 0.5, k1 endin ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/clockoff.csd0000644000000000000000000000166112262561504017050 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o clockoff.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Start clock #1. clockon 1 ; Do something that keeps Csound busy. a1 oscili 10000, 440, 1 out a1 ; Stop clock #1. clockoff 1 ; Print the time accumulated in clock #1. i1 readclock 1 print i1 endin ; Initialize the function tables. ; Table 1: an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for one second starting at 0:00. i 1 0 1 ; Play Instrument #1 for one second starting at 0:01. i 1 1 1 ; Play Instrument #1 for one second starting at 0:02. i 1 2 1 e manual_src~dfsg/examples/vcella.csd0000644000000000000000000000657512262561504016541 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vcella.wav -W ;;; for file output any platform ; vcella.csd ; by Anthony Kozar ; This file demonstrates some of the new opcodes available in ; Csound 5 that come from Gabriel Maldonado's CsoundAV. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Cellular automata-driven oscillator bank using vcella and adsynt instr 1 idur = p3 iCArate = p4 ; number of times per second the CA calculates new values ; f-tables for CA parameters iCAinit = p5 ; CA initial states iCArule = p6 ; CA rule values ; The rule is used as follows: ; the states (values) of each cell are summed with their neighboring cells within ; the specied radius (+/- 1 or 2 cells). Each sum is used as an index to read a ; value from the rule table which becomes the new state value for its cell. ; All new states are calculated first, then the new values are all applied ; simultaneously. ielements = ftlen(iCAinit) inumrules = ftlen(iCArule) iradius = 1 ; create some needed tables iCAstate ftgen 0, 0, ielements, -2, 0 ; will hold the current CA states ifreqs ftgen 0, 0, ielements, -2, 0 ; will hold the oscillator frequency for each cell iamps ftgen 0, 0, ielements, -2, 0 ; will hold the amplitude for each cell ; calculate cellular automata state ktrig metro iCArate ; trigger the CA to update iCArate times per second vcella ktrig, 0, iCAstate, iCAinit, iCArule, ielements, inumrules, iradius ; scale CA state for use as amplitudes of the oscillator bank vcopy iamps, iCAstate, ielements vmult iamps, (1/3), ielements ; divide by 3 since state values are 0-3 vport iamps, .01, ielements ; need to smooth the amplitude changes for adsynt ; we could use adsynt2 instead of adsynt, but it does not seem to be working ; i-time loop for calculating frequencies index = 0 inew = 1 iratio = 1.125 ; just major second (creating a whole tone scale) loop1: tableiw inew, index, ifreqs, 0 ; 0 indicates integer indices inew = inew * iratio index = index + 1 if (index < ielements) igoto loop1 ; create sound with additive oscillator bank ifreqbase = 64 iwavefn = 1 iphs = 2 ; random oscillator phases kenv linseg 0.0, 0.5, 1.0, idur - 1.0, 1.0, 0.5, 0.0 aosc adsynt kenv, ifreqbase, iwavefn, ifreqs, iamps, ielements, iphs out aosc * ampdb(68) endin f1 0 16384 10 1 ; This example uses a 4-state cellular automata ; Possible state values are 0, 1, 2, and 3 ; CA initial state ; We have 16 cells in our CA, so the initial state table is size 16 f10 0 16 -2 0 1 0 0 1 0 0 2 2 0 0 1 0 0 1 0 ; CA rule ; The maximum sum with radius 1 (3 cells) is 9, so we need 10 values in the rule (0-9) f11 0 16 -2 1 0 3 2 1 0 0 2 1 0 ; Here is our one and only note! i1 0 20 4 10 11 e manual_src~dfsg/examples/beats.ats0000644000000000000000000073374012262561504016410 0ustar rootroot^@@:@:@v@E@d?aCw%@@?H+ :@R@&wGjfAV@jBOSV@Qĝh?fЃ^X@rP ?hAU^@0CJG`@yk^i9Cb@7K|޸?dXd@ccxd@1|zkd@iYkd@įjf@p-^h@y hj@N-fj@u*k@O*?+l@~W~l@&~n@Min@LGq@m#r@坋|0r@6Gs@9pt@t.6t@k? u@ڹ'v@uLj?t_vw@3px@a0y@ 2y@+z@y{@m7:#>|@A-|@PJ'}@~@&8~@H߂ @'x?mR@J3v}?Gx@P/.@$ʡTk?D@Dﶂ@@a`[m?sKC@)}jc@@[s?GIz@u(h]@*@2l@ii@۠އ@|u]q?|'F@(@bPn@\1/zl?Lـ~@ĀI@f܋@E.:-ƌ@ST,ʚ@m8RK@T*@de@¤@H%@N"k@KJG8@ypA@V@(@V刓@o6~*g@f@#q)͔@Yd;@2@T#QJ|@Bؗ@!B&@@(Ř@8@Vz@o]6@eϒC-@5cEؚ@>ƘQ?quoC@Ce }@Xț@Wn^ʜ@|3̞@{p>k.@:;@srYzܟ@[HU @׵@DWH@`@/x@f[@,6@}rѣ@1W#@D`gv@.ei@`ќ@RZ$ؤ@|zk@T@NG@qå@x QW@C*@k@4L @O@y@gG\@0i@,Iw@2L@F3~@Dӕ@@K@=^@No_@Vh6 @HAΫ@EjvJ@^j_?f~zz@A7ɯAS@:I\?;ҽ@{[G@{o[@a<= @b@G@%7@8 @kQ@jX6@\kt9T?"zSw@;bj`?.>@uv#ް@TAI@{@z{Z?W4@/C-T?9KNUX@O;g>p@CXHb?| w@AZ{l?e ±@ L1|d?~|D@IR@'i?%)@toR?#[@l5@ʼnVa?ͱs@\:Ii?ED@N@JSTY_?ԉ@ZI4d?K5P@[?Y)F@~gNR@q%] U?ZJ@ډrlz@{@5T?uKԝ=Ĵ@\Dua?^t@s%޾@Oi-`? 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0p@܆\֍@$@@8_@6@-{RD@7@&@2(X@^!<=;@1YO@uJ˅@>9@Bk@8@Ꞇ @ h @|f|@ 2=@^ƽ3@H"S@D>y@گ@8d@g_)A@uuqH@&Փ@Un@#"0f@o@7#C@y@ @g@_@ٿj:@k8H@RiZ|@@'º@GP@AIt@@manual_src~dfsg/examples/b.csd0000644000000000000000000000176512262561504015510 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o b.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .4*aenv, 220, 1 outs asig, asig endin instr 2 asig pluck 0.7, p4, 220, 0, 1 outs asig, asig endin instr 3 asig loscil .8, 1, 2, 1 outs asig, asig endin instr 4 asig bamboo .8, 0.01 outs asig, asig endin f 1 0 16384 10 1 ;sine wave f 2 0 0 1 "fox.wav" 0 0 0 ;sample i1 0 2 i1 10 2 b 5 ; set the clock "forward" i2 1 2 220 ; start time = 6 i2 2 2 110 ; start time = 7 b -1 i3 3 2 ; start time = 2 i3 5.5 1 ; start time 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manual_src~dfsg/examples/STKVoicForm.csd0000644000000000000000000000122112262561504017360 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKVoicForm.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv2 line p5, p3, p6 ;Vowel/Phoneme Selection asig STKVoicForm cpspch(p4), 1, 2, 1, 4, kv2, 128, 100, 1, 10, 11, 100 asig = asig * .5 ;too loud outs asig, asig endin i 1 0 5 7.00 100 0 i 1 + 10 7.00 1 50 e manual_src~dfsg/examples/pconvolve.csd0000644000000000000000000000425612262561504017300 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out -iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pconvolve.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kmix = .5 ; Wet/dry mix. Vary as desired. kvol = .05*kmix ; Overall volume level of reverb. May need to adjust ; when wet/dry mix is changed, to avoid clipping. ; do some safety checking to make sure we the parameters a good kmix = (kmix < 0 || kmix > 1 ? .5 : kmix) kvol = (kvol < 0 ? 0 : .5*kvol*kmix) ; size of each convolution partion -- for best performance, this parameter needs to be tweaked ipartitionsize = p4 ; calculate latency of pconvolve opcode idel = (ksmps < ipartitionsize ? ipartitionsize + ksmps : ipartitionsize)/sr prints "Convolving with a latency of %f seconds%n", idel ; actual processing al, ar ins ;get live input awetl, awetr pconvolve kvol*(al+ar), "kickroll.wav", ipartitionsize ; Delay dry signal, to align it with the convoled sig adryl delay (1-kmix)*al, idel adryr delay (1-kmix)*al, idel outs adryl+awetl, adryr+awetr endin instr 2 imix = 0.5 ; Wet/dry mix. Vary as desired. ivol = .05*imix ; Overall volume level of reverb. May need to adjust ; when wet/dry mix is changed, to avoid clipping. ipartitionsize = 1024 ; size of each convolution partion idel = (ksmps < ipartitionsize ? ipartitionsize + ksmps : ipartitionsize)/sr ; latency of pconvolve opcode kcount init idel*kr ; since we are using a soundin [instead of ins] we can ; do a kind of "look ahead" by looping during one k-pass ; without output, creating zero-latency loop: asig soundin p4, 0 awetl, awetr pconvolve ivol*(asig),"rv_stereo.wav", ipartitionsize adry delay (1-imix)*asig,idel ; Delay dry signal, to align it with kcount = kcount - 1 if kcount > 0 kgoto loop outs awetl+adry, awetr+adry endin i 1 0 20 1024 ;play live for 20 seconds i 2 20 5 "fox.wav" i 2 25 5 "flute.aiff" e manual_src~dfsg/examples/vpow_i-2.csd0000644000000000000000000000142412262561504016721 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vpow_i-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifil ftgen 1, 0, 0, 1, "fox.wav", 0, 0, 1 instr 1 ival = p4 ;different distortion settings ielements = p5 idstoffset = p6 ;index offset vpow_i 1, ival, ielements, idstoffset asig lposcil 1, 1, 0, 0, 1 outs asig, asig endin i1 0 2.7 .5 70000 0 ;no offset i1 3 2.7 .01 50000 70000 ;add another period of distortion, starting at sample 70000 e manual_src~dfsg/examples/cigoto.csd0000644000000000000000000000205412262561504016543 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Get the value of the 4th p-field from the score. iparam = p4 ; If iparam is 1 then play the high note. ; If not then play the low note. cigoto (iparam ==1), highnote igoto lownote highnote: ifreq = 880 goto playit lownote: ifreq = 440 goto playit playit: ; Print the values of iparam and ifreq. print iparam print ifreq a1 oscil 10000, ifreq, 1 out a1 endin ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; p4: 1 = high note, anything else = low note ; Play Instrument #1 for one second, a low note. i 1 0 1 0 ; Play a Instrument #1 for one second, a high note. i 1 1 1 1 e manual_src~dfsg/examples/partikkel.csd0000644000000000000000000000604612262561504017252 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o partikkel.wav -W ;;; for file output any platform sr = 44100 ksmps = 20 nchnls = 2 giSine ftgen 0, 0, 65537, 10, 1 giCosine ftgen 0, 0, 8193, 9, 1, 1, 90 instr 1 kgrainfreq = 200 ; 4 grains per second kdistribution = 0 ; periodic grain distribution idisttab = -1 ; (default) flat distribution used for grain distribution async = 0 ; no sync input kenv2amt = 0 ; no secondary enveloping ienv2tab = -1 ; default secondary envelope (flat) ienv_attack = -1 ; ; default attack envelope (flat) ienv_decay = -1 ; ; default decay envelope (flat) ksustain_amount = 0.5 ; time (in fraction of grain dur) at sustain level for each grain ka_d_ratio = 0.5 ; balance between attack and decay time kduration = (0.5/kgrainfreq)*1000 ; set grain duration relative to grain rate kamp = 5000 ; amp igainmasks = -1 ; (default) no gain masking kwavfreq = 440 ; fundamental frequency of source waveform ksweepshape = 0 ; shape of frequency sweep (0=no sweep) iwavfreqstarttab = -1 ; default frequency sweep start (value in table = 1, which give no frequency modification) iwavfreqendtab = -1 ; default frequency sweep end (value in table = 1, which give no frequency modification) awavfm = 0 ; no FM input ifmamptab = -1 ; default FM scaling (=1) kfmenv = -1 ; default FM envelope (flat) icosine = giCosine ; cosine ftable kTrainCps = kgrainfreq ; set trainlet cps equal to grain rate for single-cycle trainlet in each grain knumpartials = 3 ; number of partials in trainlet kchroma = 1 ; balance of partials in trainlet ichannelmasks = -1 ; (default) no channel masking, all grains to output 1 krandommask = 0 ; no random grain masking kwaveform1 = giSine ; source waveforms kwaveform2 = giSine ; kwaveform3 = giSine ; kwaveform4 = giSine ; iwaveamptab = -1 ; (default) equal mix of all 4 sourcve waveforms and no amp for trainlets asamplepos1 = 0 ; phase offset for reading source waveform asamplepos2 = 0 ; asamplepos3 = 0 ; asamplepos4 = 0 ; kwavekey1 = 1 ; original key for source waveform kwavekey2 = 1 ; kwavekey3 = 1 ; kwavekey4 = 1 ; imax_grains = 100 ; max grains per k period asig partikkel kgrainfreq, kdistribution, idisttab, async, kenv2amt, ienv2tab, \ ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, kduration, kamp, igainmasks, \ kwavfreq, ksweepshape, iwavfreqstarttab, iwavfreqendtab, awavfm, \ ifmamptab, kfmenv, icosine, kTrainCps, knumpartials, \ kchroma, ichannelmasks, krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, asamplepos4, \ kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains outs asig, asig endin i1 0 5 ; partikkel e manual_src~dfsg/examples/gausstrig.csd0000644000000000000000000000116112262561504017265 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kdev line 0, p3, 0.9 seed 20120125 aimp gausstrig 0.5, 10, kdev aenv filter2 aimp, 1, 1, 0.993, 0.993 anoi fractalnoise 0.2, 1.7 al = anoi*aenv ar delay al, 0.02 outs al, ar endin i1 0 10 e manual_src~dfsg/examples/STKBandedWG.csd0000644000000000000000000000132512262561504017254 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKBandedWG.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;pressure of bow kenv line 1, p3, 0 asig STKBandedWG cpspch(ifrq), 1, 2, kv1, 4, 100, 11, 0, 1, 0, 64, 100, 128, 120, 16, 2 asig = asig * kenv ;simple envelope outs asig, asig endin i 1 0 10 5.00 100 0 i 1 10 8 6.03 10 . i 1 20 5 7.05 50 127 e manual_src~dfsg/examples/madsr-2.csd0000644000000000000000000000121112262561504016516 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o madsr-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .5 kenv madsr 0.5, 0, 1, 0.5 asig pluck kenv, icps, icps, 2, 1 outs asig, asig endin f 2 0 4096 10 1 f0 30 ;runs 30 seconds e manual_src~dfsg/examples/cpstuni.csd0000644000000000000000000000200612262561504016741 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cpstuni.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Table #1, a normal 12-tone equal temperament scale. ; numgrades = 12 (twelve tones) ; interval = 2 (one octave) ; basefreq = 261.659 (Middle C) ; basekeymidi = 60 (Middle C) gitemp ftgen 1, 0, 64, -2, 12, 2, 261.659, 60, 1.00, \ 1.059, 1.122, 1.189, 1.260, 1.335, 1.414, \ 1.498, 1.588, 1.682, 1.782, 1.888, 2.000 ; Instrument #1. instr 1 ; Use Table #1. ifn = 1 ; If the base key (note #60) is C, then 9 notes ; above it (note #60 + 9 = note #69) should be A. index = 69 i1 cpstuni index, ifn print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/lpfreson.csd0000644000000000000000000000174312262561504017113 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpfreson.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;do not use -a option when analyzing "fox.wav" with lpfreson, ;it needs a filter coefficient type of file instr 1 ilen filelen "fox.wav" ; length of soundfile prints "fox.wav = %f seconds\\n",ilen ktime line 0, p3, p4 krmsr,krmso,kerr,kcps lpread ktime,"fox_nopoles.lpc" krmso = krmso*.00001 ; low volume asig buzz krmso, kcps, int(sr/2/kcps), 1 ; max harmonics without aliasing aout lpfreson asig, 1.2 asig clip aout, 0, 1 ; prevents distortion outs asig, asig endin ; sine f1 0 4096 10 1 i 1 0 2.8 1 ; first words only i 1 4 2.8 2.8 ; whole sentence e manual_src~dfsg/examples/gen51.csd0000644000000000000000000000411212262561504016173 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M0 ;;;realtime audio out and midi input ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen51.wav -W ;;; for file output any platform ;example by Iain McCurdy sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giEqTmp12 ftgen 1,0,128,-51,12,2,cpsoct(8),60,1,2^(1/12),2^(2/12),2^(3/12),2^(4/12),2^(5/12),2^(6/12),2^(7/12),2^(8/12),2^(9/12),2^(10/12),2^(11/12),2^(12/12) giEqTmp10 ftgen 2,0,128,-51,10,2,cpsoct(8),60,1,2^(1/10),2^(2/10),2^(3/10),2^(4/10),2^(5/10),2^(6/10),2^(7/10),2^(8/10),2^(9/10),2^(10/10) giEqTmp24 ftgen 3,0,128,-51,24,2,cpsoct(8),60,1,2^(1/24),2^(2/24),2^(3/24),2^(4/24),2^(5/24),2^(6/24),2^(7/24),2^(8/24),2^(9/24),2^(10/24),2^(11/24), \ 2^(12/24),2^(13/24),2^(14/24),2^(15/24),2^(16/24),2^(17/24),2^(18/24),2^(19/24),2^(20/24),2^(21/24),2^(22/24),2^(23/24),2^(24/24) instr 1 ;midi input instrument /*USE PITCH BEND TO MODULATE NOTE NUMBER UP OR DOWN ONE STEP - ACTUAL INTERVAL IT WILL MODULATE BY WILL BE DEPENDENT UPON THE GEN51 SCALE USED*/ ;kbend pchbend 0,2 /*ALTERNATIVELY IF USING VIRTUAL MIDI DEVICE OR A KEYBOARD WITH NO PITCH BEND WHEEL, USE CONTROLLERS 1 AND 2 TO MODULATE PITCH UP OR DOWN 1 STEP*/ kup ctrl7 1, 1, 0, 1 kdown ctrl7 1, 2, 0, -1 kbend = kup+kdown inum notnum kcps tablei inum+kbend, giEqTmp24 ;read cps values from GEN51, scale table using a combination of note played and pitch bend/controllers 1 and 2 a1 vco2 0.2, kcps, 4, 0.5 outs a1, a1 endin instr 2 ;score input instrument knum line p4, p3, p5 ;gliss using a straight line bewteen p4 and p5 for the entire note duration kcps tablei knum, giEqTmp24 ;read cps values from GEN51 scale table a1 vco2 0.2, kcps, 4, 0.5 outs a1, a1 endin f 0 3600 ;instr 2. Score input. Gliss from step number p4 to step number p5 ;p4 - starting note number ;p5 - ending note number i 2 0 2 60 61 i 2 + 2 70 58 i 2 + 2 66 66.5 i 2 + 2 71.25 71 e manual_src~dfsg/examples/STKHevyMetl.csd0000644000000000000000000000136212262561504017377 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKHevyMetl.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;Total Modulator Index kv2 line p7, p3, 0 ;Modulator Crossfade asig STKHevyMetl cpspch(ifrq), 1, 2, kv1, 4, kv2, 11, 0, 1, 100, 128, 40 outs asig, asig endin i 1 0 7 8.05 100 0 100 i 1 3 7 9.03 20 120 0 i 1 3 .5 8.05 20 120 0 i 1 4 .5 9.09 20 120 0 i 1 5 3 9.00 20 120 0 e manual_src~dfsg/examples/pvswarp.csd0000644000000000000000000000131112262561504016754 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvswarp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kscal = p4 asig soundin "fox.wav" ; get the signal in fsig pvsanal asig, 1024, 256, 1024, 1 ; analyse it ftps pvswarp fsig, kscal, 0 ; warp it atps pvsynth ftps ; synthesise it outs atps, atps endin i 1 0 3 1 i 1 + 3 1.5 i 1 + 3 3 i 1 + 3 .25 e manual_src~dfsg/examples/cauchy.csd0000644000000000000000000000140312262561504016530 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cauchy.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values kalpha cauchy 100 printk .2, kalpha ; look aout oscili 0.8, 440+kalpha, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 kalpha cauchy 100 printk .2, kalpha ; look aout oscili 0.8, 440+kalpha, 1 ; & listen outs aout, aout endin ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/partials.csd0000644000000000000000000000123712262561504017100 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o partials.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "fox.wav", 1 fs1,fsi2 pvsifd ain,2048,512,1 ; ifd analysis fst partials fs1,fsi2,.003,1,3,500 ; partial tracking aout resyn fst, 1, 1.5, 500, 1 ; resynthesis (up a 5th) outs aout, aout endin f 1 0 4096 10 1 i 1 0 2.8 e manual_src~dfsg/examples/octmidib.csd0000644000000000000000000000122212262561504017045 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o octmidib.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; This example expects MIDI note inputs on channel 1 i1 octmidib print i1 endin ;Dummy f-table to give time for real-time MIDI events f 0 8000 e manual_src~dfsg/examples/deltapxw.csd0000644000000000000000000000144512262561504017112 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o deltapxw.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 a1 phasor 300 a1 = a1 - 0.5 a_ delayr 1 adel phasor 4 adel = sin(2 * 3.14159265 * adel) * 0.01 + 0.2 deltapxw a1, adel, 32 adel phasor 2 adel = sin(2 * 3.14159265 * adel) * 0.01 + 0.2 deltapxw a1, adel, 32 adel = 0.3 a2 deltapx adel, 32 a1 = 0 delayw a1 outs a2*.7, a2*.7 endin i1 0 5 e manual_src~dfsg/examples/strchar.csd0000644000000000000000000000235512262561504016731 0ustar rootroot -ndm0 ;example by joachim heintz 2013 opcode ToAscii, S, S ;returns the ASCII numbers of the input string as string Sin xin ;input string ilen strlen Sin ;its length ipos = 0 ;set counter to zero Sres = "" ;initialize output string loop: ;for all characters in input string: ichr strchar Sin, ipos ;get its ascii code number Snew sprintf "%d ", ichr ;put this number into a new string Sres strcat Sres, Snew ;append this to the output string loop_lt ipos, 1, ilen, loop ;see comment for 'loop:' xout Sres ;return output string endop instr Characters printf_i "\nCharacters:\n given as single strings: %s%s%s%s%s%s\n", 1, "c", "s", "o", "u", "n", "d" printf_i " but can also be given as numbers: %c%c%c%c%c%c\n", 1, 99, 115, 111, 117, 110, 100 Scsound ToAscii "csound" printf_i " in csound, the ASCII code of a character can be accessed with the opcode strchar.%s", 1, "\n" printf_i " the name 'csound' returns the numbers %s\n\n", 1, Scsound endin i "Characters" 0 0 e manual_src~dfsg/examples/midichn_advanced.mid0000644000000000000000000000056412262561504020523 0ustar rootrootMThdMTrkv#Copyright (c) xxxx Copyright HolderCreated by Rosegardenhttp://www.rosegardenmusic.com/QB@X/MTrk; Imported MIDId @][ AB@A<@</MTrk/ Imported MIDId @][@>@>/MTrk/ Imported MIDId @][ 7@7/MTrk/ Imported MIDId @][`H@H/manual_src~dfsg/examples/vibes.csd0000644000000000000000000000155112262561504016370 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vibes.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 ksmps = 10 nchnls = 2 ; Instrument #1. instr 1 ; kamp = 20000 ; kfreq = 440 ; ihrd = 0.5 ; ipos = p4 ; imp = 1 ; kvibf = 6.0 ; kvamp = 0.05 ; ivibfn = 2 ; idec = 0.1 asig vibes 20000, 440, .5, p4 , 1, 6.0, 0.05, 2, .1 outs asig, asig endin ; Table #1, the "marmstk1.wav" audio file. f 1 0 256 1 "marmstk1.wav" 0 0 0 ; Table #2, a sine wave for the vibrato. f 2 0 128 10 1 ; Play Instrument #1 for four seconds. i 1 0 4 0.561 i 1 + 4 1 e manual_src~dfsg/examples/p.csd0000644000000000000000000000115012262561504015512 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o p.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Get the value in the fourth p-field, p4. i1 = p(4) print i1 endin ; p4 = value to be printed. ; Play Instrument #1 for one second, p4 = 50.375. i 1 0 1 50.375 e manual_src~dfsg/examples/trscale.csd0000644000000000000000000000141012262561504016707 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trscale.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kpitch = p4 ain diskin2 "fox.wav", 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .003, 1, 3, 500 ; partial tracking fscl trscale fst, kpitch ; frequency scale aout tradsyn fscl, 1, 1, 500, 1 ; resynthesis outs aout, aout endin f1 0 8192 10 1 i 1 0 3 1.5 ;up a 5th i 1 3 3 3 ;two octaves higher e manual_src~dfsg/examples/maxabs.csd0000644000000000000000000000146512262561504016537 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o maxabs.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 k1 oscili 1, 10.0, 1 ;combine 3 sinusses k2 oscili 1, 1.0, 1 ;at different rates k3 oscili 1, 3.0, 1 kmax maxabs k1, k2, k3 kmax = kmax*250 ;scale kmax printk2 kmax ;check the values aout vco2 .5, 220, 6 ;sawtooth asig moogvcf2 aout, 600+kmax, .5 ;change filter above 600 Hz outs asig, asig endin f1 0 32768 10 1 i1 0 5 e manual_src~dfsg/examples/gaussi.csd0000644000000000000000000000111312262561504016545 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o exprand.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 klambda gaussi 100, 1, 3 printk2 klambda ; look aout oscili 0.8, 440+klambda, 1 ; & listen outs aout, aout endin ; sine wave f 1 0 16384 10 1 i 1 0 4 e manual_src~dfsg/examples/FLvalue.csd0000644000000000000000000000245112262561504016616 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLvalue.wav -W ;;; for file output any platform ; Using the opcode flvalue to display the output of a slider sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Value Display Box", 900, 200, 50, 50 ; Width of the value display box in pixels iwidth = 50 ; Height of the value display box in pixels iheight = 20 ; Distance of the left edge of the value display ; box from the left edge of the panel ix = 65 ; Distance of the top edge of the value display ; box from the top edge of the panel iy = 55 idisp FLvalue "Hertz", iwidth, iheight, ix, iy gkfreq, ihandle FLslider "Frequency", 200, 5000, -1, 5, idisp, 750, 30, 125, 50 FLsetVal_i 500, ihandle ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 iamp = 15000 ifn = 1 asig oscili iamp, gkfreq, ifn out asig endin ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e manual_src~dfsg/examples/nchnls_i.csd0000644000000000000000000000165412262561504017061 0ustar rootroot ; Select audio/midi flags here according to platform -odac -idac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; nchnls_i.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 ;2 channels out 0dbfs = 1 nchnls_i = 4 ;4 channels in instr 1 ;4 channels in, two channels out ain1, ain2, ain3, ain4 inq ;grab your mics and sing adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time = 20ms adel2 linseg 0.02, p3*.5, 0, p3*.5, 0.02 ;max delay time = 20ms aoutl flanger ain1, adel, .7 aoutr flanger ain2, adel*2, .8 aoutla flanger ain3, adel2, .9 aoutra flanger ain4, adel2*2, .5 ;write to quad file, 16 bits with header fout "in_4.wav", 14, aoutl, aoutr, aoutla, aoutra outs (aoutl+aoutla)*.5, (aoutr+aoutra)*.5 ;stereo out endin i 1 0 10 e manual_src~dfsg/examples/pvshift.csd0000644000000000000000000000207612262561504016746 0ustar rootroot -odac sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 ishift = p4; shift amount in Hz ilowest = p5; lowest frequency to be shifted ikeepform = p6; 0=no formant keeping, 1=keep by amps, 2=keep by spectral envelope ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file fshift pvshift fftin, ishift, ilowest, ikeepform; shift frequencies aout pvsynth fshift; resynthesize out aout endin i 1 0 2.757 0 0 0; no shift at all i 1 3 2.757 100 0 0; shift all frequencies by 100 Hz i 1 6 2.757 200 0 0; by 200 Hz i 1 9 2.757 200 0 1; keep formants by method 1 i 1 12 2.757 200 0 2; by method 2 i 1 15 2.757 200 1000 0; shift by 200 Hz but just above 1000 Hz i 1 18 2.757 1000 500 0; shift by 1000 Hz above 500 Hz i 1 21 2.757 1000 300 0; above 300 Hz e manual_src~dfsg/examples/tan.csd0000644000000000000000000000103612262561504016040 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tan.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 25 i1 = tan(irad) print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/wgpluck2.csd0000644000000000000000000000123112262561504017011 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o wgpluck2.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 iplk = 0.75 kamp = 30000 icps = 220 kpick = 0.75 krefl = 0.5 apluck wgpluck2 iplk, kamp, icps, kpick, krefl out apluck endin ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/adds.csd0000644000000000000000000000131012262561504016164 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o adds.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; add unipolar square to oscil kamp = p4 kcps = 1 itype = 3 klfo lfo kamp, kcps, itype printk2 klfo asig oscil 0.7, 440+klfo, 1 outs asig, asig endin ;sine wave. f 1 0 32768 10 1 i 1 0 2 1 ;adds 1 Hz to frequency i 1 + 2 10 ;adds 10 Hz to frequency i 1 + 2 220 ;adds 220 Hz to frequency e manual_src~dfsg/examples/gen09.csd0000644000000000000000000000251212262561504016200 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen09.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gi1 ftgen 1,0,2^10,9, 1,3,0, 3,1,0, 9,0.333,180 ;an approximation of a square wave gi2 ftgen 2,0,2^10,9, 1,3,180, 3,1,0, 9,0.333,0 ;same values as gi1, except some phase values gi3 ftgen 3,0,2^10,9, 1, .4, 0, 2.2, .5, 0, 3.8, 1, 0 ;inharmonic, but does not sound well --> wave does not end at same point as where it begins --> artefacts gi4 ftgen 4,0,2^10,9, 10, .4, 0, 22, .5, 0, 38, 1, 0 ;the same proportions, but value of partial number is multiplied 10 times -->the sound is much clearer, ;because the sudden "jump" like the one in gi3 will pop up only once in 10 repetitions instr 1 kamp = .6 kcps = 220 ifn = p4 asig poscil kamp, kcps*p5, ifn outs asig,asig endin i 1 0 2 1 1 ;subtle difference between table 1 and 2 i 1 3 2 2 1 i 1 7 2 3 1 ;big difference between table 3 and 4 i 1 10 2 4 .1 ;p5 has to compensate for the 10 repetitions of gi4 as opposed to gi3 to get the same pitch e manual_src~dfsg/examples/delayk.csd0000644000000000000000000000174312262561504016534 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o delayk.wav -W ;;; for file output any platform ;example shows "delayk" for fm index and ;a second "delayk" for panning sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 0, 0, 2^10, 10, 1 instr 1 kenv1 transeg 0, .02, 0, 1, 3.98, -6, 0 ;envelope kenv2 delayk kenv1, 2 ;delayed by two seconds kindx expon 5, p3, 1 ;fm index decreasing over p3 asig foscili .6, 400, 1, 11/4, kindx, gisin kpan1 linseg 0, 4, 1 ;panning for first sound kpan2 linseg 1, 4, 0 ;panning for second sound ... kpan2 delayk kpan2, 2 ;delayed by two seconds a1 = asig * kenv1 a2 = asig * kenv2 aL1,aR1 pan2 a1, kpan1 aL2,aR2 pan2 a2, kpan2 outs aL1+aL2, aR1+aR2 endin i 1 0 6 e manual_src~dfsg/examples/pvsynth.csd0000644000000000000000000000131212262561504016766 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsynth.wav -W ;;; for file output any platform sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1 ; von-Hann window Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of the audio-signal aout pvsynth fftin; resynthesis out aout endin i 1 0 3 e manual_src~dfsg/examples/bbcutm.csd0000644000000000000000000000160012262561504016527 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o bbcutm.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - Play an audio file normally. instr 1 asource soundin "beats.wav" out asource endin ; Instrument #2 - Cut-up an audio file. instr 2 asource soundin "beats.wav" ibps = 4 isubdiv = 8 ibarlength = 4 iphrasebars = 1 inumrepeats = 2 a1 bbcutm asource, ibps, isubdiv, ibarlength, iphrasebars, inumrepeats out a1 endin ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 3 2 e manual_src~dfsg/examples/dbamp.csd0000644000000000000000000000106712262561504016345 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dbamp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 iamp = p4 idb = dbamp(iamp) print idb asig vco2 iamp, 110 ;sawtooth outs asig, asig endin i 1 0 1 100 i 1 + 1 1000 i 1 + 1 10000 i 1 + 1 20000 e manual_src~dfsg/examples/space_stereo.csd0000644000000000000000000000202412262561504017730 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o space_stereo.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 ga1 init 0 ga2 init 0 instr 1 kx = p4 ky = p5 asig diskin2 "beats.wav", 1 a1, a2, a3, a4 space asig, 0, 0, .1, kx, ky ;take position values from p4, p5 ar1, ar2, ar3, ar4 spsend ;send to reverb ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a2 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 outs a1, a2 ga1=0 ga2=0 endin ;place the sound in the left speaker and near i1 0 1 -1 1 ;place the sound in the right speaker and far i1 1 1 45 45 ;place the sound equally between left and right and in the middle ground distance i1 2 1 0 12 i 99 0 7 ;keep reverb active e manual_src~dfsg/examples/hrtfearly.csd0000644000000000000000000000254312262561504017262 0ustar rootroot ; Select flags here ; realtime audio out -o dac ; file ouput ; -o hrtf.wav nchnls = 2 gasrc init 0 ;global instr 1 ;a plucked string, distorted and filtered iamp = 15000 icps = cpspch(p4) a1 pluck iamp, icps, icps, 0, 1 adist distort1 a1, 10, .5, 0, 0 afilt moogvcf2 adist, 8000, .5 aout linen afilt, 0, p3, .01 gasrc = gasrc + aout endin instr 10 ;uses output from instr1 as source ;simple path for source kx line 2, p3, 9 ;early reflections, room default 1 aearlyl,aearlyr, irt60low, irt60high, imfp hrtfearly gasrc, kx, 5, 1, 5, 1, 1, "hrtf-44100-left.dat", "hrtf-44100-right.dat", 1 ;later reverb, uses outputs from above arevl, arevr, idel hrtfreverb gasrc, irt60low, irt60high, "hrtf-44100-left.dat", "hrtf-44100-right.dat", 44100, imfp ;delayed and scaled alatel delay arevl * .1, idel alater delay arevr * .1, idel outs aearlyl + alatel, aearlyr + alater gasrc = 0 endin ; Play Instrument 1: a simple arpeggio i1 0 .2 8.00 i1 + .2 8.04 i1 + .2 8.07 i1 + .2 8.11 i1 + .2 9.02 i1 + 1.5 8.11 i1 + 1.5 8.07 i1 + 1.5 8.04 i1 + 1.5 8.00 i1 + 1.5 7.09 i1 + 4 8.00 ; Play Instrument 10 for 13 seconds. i10 0 13 manual_src~dfsg/examples/outrg.csd0000644000000000000000000000152212262561504016416 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outrg.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 instr 1 kleft init 1 asig vco2 .5, 220 ;sawtooth idur = p3/(nchnls-1) knext init idur kpos init 0 krate init 1/idur kbase init 0 ktime timeinsts if ktime>=knext then kleft = kleft + 1 knext = knext + idur kpos = 0 kbase = ktime else kpos = (ktime-kbase)/idur endif printks "speaker %d position %f\n", 0, kleft, kpos a1,a2 pan2 asig, kpos outrg kleft, a1, a2 kpos = kbase/idur endin i 1 0 10 e manual_src~dfsg/examples/STKSitar.csd0000644000000000000000000000111712262561504016722 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKSitar.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;STKSitar - has no controllers ifrq = p4 asig STKSitar cpspch(p4), 1 asig = asig * 2 ;amplify outs asig, asig endin i 1 0 4 6.00 i 1 + 2 7.05 i 1 + 7 5.05 e manual_src~dfsg/examples/elseif.csd0000644000000000000000000000152712262561504016532 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o elseif.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = cpspch(p4) ienv = p5 if (ienv == 0) then ;ADSR kenv adsr 0.05, 0.05, .95, .05 elseif (ienv == 1) then ;Linear Triangular Envelope kenv linseg 0, p3 * .5, 1, p3 * .5, 0 elseif (ienv == 2) then ;Ramp Up kenv linseg 0, p3 - .01, 1, .01, 0 endif aout vco2 .8, ipch, 10 aout moogvcf aout, ipch + (kenv * 5 * ipch) , .5 aout = aout * kenv outs aout, aout endin i 1 0 2 8.00 0 i 1 3 2 8.00 1 i 1 6 2 8.00 2 e manual_src~dfsg/examples/deltapx.csd0000644000000000000000000000144412262561504016722 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o deltapx.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 a1 phasor 300 a1 = a1 - 0.5 a_ delayr 1 adel phasor 4 adel = sin(2 * 3.14159265 * adel) * 0.01 + 0.2 deltapxw a1, adel, 32 adel phasor 2 adel = sin(2 * 3.14159265 * adel) * 0.01 + 0.2 deltapxw a1, adel, 32 adel = 0.3 a2 deltapx adel, 32 a1 = 0 delayw a1 outs a2*.7, a2*.7 endin i1 0 5 e manual_src~dfsg/examples/osciln.csd0000644000000000000000000000206212262561504016545 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o osciln.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gione ftgen 1, 0, 1024, 7, 0, 1, 1, 1024, 0 gitwo ftgen 2, 0, 1024, 7, 0, 512, 1, 512, 0 instr 1 ;very simple waveguide system ifn = p4 ipitch = p5 itimes = p6 iperiod = 1000/ipitch afeed init 0 aimpl osciln 1, ipitch, ifn, itimes ;use as excitation signal arefl tone aimpl + afeed, 4000 aout atone arefl, 5000 afeed vdelay arefl, iperiod, 10 outs aout*3, aout*3 endin i 1 0 4 1 110 1 ;use different tables, i 1 5 4 2 110 1 ;& different pitch i 1 10 4 1 110 10 ;& different number of times the table is read i 1 15 4 2 110 10 i 1 20 6 1 880 1 i 1 25 3 2 880 1 i 1 30 3 1 880 10 i 1 35 3 2 880 10 e manual_src~dfsg/examples/minarray.csd0000644000000000000000000000133512262561504017102 0ustar rootroot ; Select audio/midi flags here according to platform -n ;example by joachim heintz seed 0 instr 1 ;create an array with 10 elements kArr[] init 10 ;fill in random numbers and print them out kIndx = 0 until kIndx == 10 do kNum random -100, 100 kArr[kIndx] = kNum printf "kArr[%d] = %10f\n", kIndx+1, kIndx, kNum kIndx += 1 od ;investigate minimum number and print it out kMin, kMinIndx minarray kArr printf "Minimum of kArr = %f at index %d\n", kIndx+1, kMin, kMinIndx turnoff endin i1 0 0.1 e manual_src~dfsg/examples/vincr.csd0000644000000000000000000000127712262561504016406 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vincr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gaReverb init 0 instr 1 idur = p3 kpitch = p4 a1 diskin2 "fox.wav", kpitch a1 = a1*.5 ;reduce volume vincr gaReverb, a1 endin instr 99 ; global reverb al, ar reverbsc gaReverb, gaReverb, .8, 10000 outs gaReverb+al, gaReverb+ar clear gaReverb endin i1 0 3 1 i99 0 5 e manual_src~dfsg/examples/shaker.csd0000644000000000000000000000110612262561504016531 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o shaker.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 instr 1 kfreq line p4, p3, 440 a1 shaker 10000, kfreq, 8, 0.999, 100, 0 out a1 endin i 1 0 1 440 i 1 + 1 4000 e manual_src~dfsg/examples/butterhp.csd0000644000000000000000000000115412262561504017114 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o butterhp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; White noise asig rand 0.5 outs asig, asig endin instr 2 ; filtered noise asig rand 0.6 ahp butterhp asig, 500 ;pass frequencies above 500 Hz outs ahp, ahp endin i 1 0 2 i 2 2.5 2 e manual_src~dfsg/examples/out.csd0000644000000000000000000000124512262561504016067 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o out.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 1 0dbfs = 1 instr 1 kamp = .6 kcps = 440 ifn = p4 asig oscil kamp, kcps, ifn out asig ;one channel endin f1 0 16384 10 1 ; Sine f2 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth i 1 0 2 1 i 1 3 2 2 e manual_src~dfsg/examples/equal.csd0000644000000000000000000000115112262561504016363 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o equal.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 instr 1 ; Assign a value to the variable i1. i1 = 1234 ; Print the value of the i1 variable. print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/fprints.csd0000644000000000000000000000131512262561504016743 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o fprints.wav -W ;;; for file output any platform /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a score generator example. instr 1 ; Print to the file "my.sco". fprints "my.sco", "%!Generated score by ma++\\n \\n" endin /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Play Instrument #1. i 1 0 0.001 manual_src~dfsg/examples/times_complex.csd0000644000000000000000000000453612262561504020136 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o times_complex.wav -W ;;; for file output any platform ;by joachim heintz and rory walsh sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giWave ftgen 0, 0, 1024, 10, 1, .5, .25 instr again instance = p4 ;reset the duration of this instance iDur rnd31 5, 3 ;shorter values are more probable iDur = abs(iDur) + 0.2 p3 = iDur ;trigger the effect instrument of this instance event_i "i", "fx_processor", 0, iDur, instance ;print the status quo kTime times prints "instance = %d, start = %f, duration = %f\n", instance, i(kTime), iDur ;make sound iamp active 1 ;scale amplitudes iOct random 5, 10 ;find pitch aEnv transeg 0, 0.02, 0, 1/iamp, p3-0.02, -6, 0 ;output envelope aSend poscil aEnv, cpsoct(iOct), giWave ;audio signal ;send signal to effect instrument Sbus sprintf "audio_%d", instance ;create unique software bus chnset aSend/2, Sbus ;send audio on this bus ;get the last k-cycle of this instance and trigger the successor in it kLast release schedkwhen kLast, 0, 0, "again", 0, 1, instance+1 endin instr fx_processor ;apply feedback delay to the above instrument iwhich = p4 ;receive instance number ... Sbus sprintf "audio_%d", iwhich ; ... and related software bus audio chnget Sbus ;receive audio on this bus irvbtim random 1, 5 ;find reverb time p3 = p3+irvbtim ;adjust instrument duration iltptmL random .1, .5 ;find looptime left ... iltptmR random .1, .5 ;...and right ipan random 0, 1 ; pan and ... imix random 0, 1 ;... mix audio aL,aR pan2 audio, ipan ;create stereo awetL comb aL, irvbtim, iltptmL ;comb filter awetR comb aR, irvbtim, iltptmR aoutL ntrpol aL, awetL, imix ;wet-dry mix aoutR ntrpol aR, awetR, imix outs aoutL/2, aoutR/2 endin i "again" 0 1 1 e 3600 manual_src~dfsg/examples/waveset.csd0000644000000000000000000000113512262561504016734 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o waveset.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 instr 1 krep init p4 asig soundin "flute.aiff" aout waveset asig, krep outs aout, aout endin i 1 0 3 1 ;no repetitions i 1 + 10 3 ;stretching 3 times i 1 + 14 6 ;6 times e manual_src~dfsg/examples/spectrum.txt0000644000000000000000000000274312262561504017174 0ustar rootroot#Frequency (Hz) Level (dB) 344.531250 -9.408175 689.062500 -11.542959 1033.593750 -21.892752 1378.125000 -24.258373 1722.656250 -18.601250 2067.187500 -20.766144 2411.718750 -25.426712 2756.250000 -20.080441 3100.781250 -26.096043 3445.312500 -26.792006 3789.843750 -25.379452 4134.375000 -34.031952 4478.906250 -41.497753 4823.437500 -46.513386 5167.968750 -32.469891 5512.500000 -26.959146 5857.031250 -33.118156 6201.562500 -52.667919 6546.093750 -50.139523 6890.625000 -44.757580 7235.156250 -38.376083 7579.687500 -40.954887 7924.218750 -42.764221 8268.750000 -37.079933 8613.281250 -42.961800 8957.812500 -55.379814 9302.343750 -54.114830 9646.875000 -55.103600 9991.406250 -58.144154 10335.937500 -60.414536 10680.468750 -52.401680 11025.000000 -47.045467 11369.531250 -50.944092 11714.062500 -56.675507 12058.593750 -59.445457 12403.125000 -55.266907 12747.656250 -56.959026 13092.187500 -60.342400 13436.718750 -57.675564 13781.250000 -52.839638 14125.781250 -59.135277 14470.312500 -65.163216 14814.843750 -64.589638 15159.375000 -64.101051 15503.906250 -68.578819 15848.437500 -71.754791 16192.968750 -55.440922 16537.500000 -49.826622 16882.031250 -55.727921 17226.562500 -70.721367 17571.093750 -67.187195 17915.625000 -62.886978 18260.156250 -64.825745 18604.687500 -68.334122 18949.218750 -66.449799 19293.750000 -61.810974 19638.281250 -67.812668 19982.812500 -73.615036 20327.343750 -72.949310 20671.875000 -76.245728 21016.406250 -80.125305 21360.937500 -83.319687 21705.468750 -81.967491 manual_src~dfsg/examples/p5g.csd0000644000000000000000000000257212262561504015757 0ustar rootroot -+rtaudio=alsa -o dac:hw:0 nchnls = 1 ksmps = 1000 #define P5G_BUTTONS #0# #define P5G_BUTTON_A #1# #define P5G_BUTTON_B #2# #define P5G_BUTTON_C #4# #define P5G_JUSTPUSH #8# #define P5G_JUSTPU_A #9# #define P5G_JUSTPU_B #10# #define P5G_JUSTPU_C #12# #define P5G_RELEASED #16# #define P5G_RELSED_A #17# #define P5G_RELSED_B #18# #define P5G_RELSED_C #20# #define P5G_FINGER_INDEX #32# #define P5G_FINGER_MIDDLE #33# #define P5G_FINGER_RING #34# #define P5G_FINGER_PINKY #35# #define P5G_FINGER_THUMB #36# #define P5G_DELTA_X #37# #define P5G_DELTA_Y #38# #define P5G_DELTA_Z #39# #define P5G_DELTA_XR #40# #define P5G_DELTA_YR #41# #define P5G_DELTA_ZR #42# #define P5G_ANGLES #43# gka init 0 gkp init 0 instr 1 p5gconnect ka p5gdata $P5G_JUSTPU_A. kc p5gdata $P5G_BUTTON_C. ; If the A button is just pressed then activate a note if (ka==0) goto ee event "i", 2, 0, 2 ee: gka p5gdata $P5G_DELTA_X. gkp p5gdata $P5G_DELTA_Y. printk2 gka printk2 gkp if (kc==0) goto ff printks "turning off (%d)\n", 0, kc turnoff ff: endin instr 2 a1 oscil ampdbfs(gkp), 440+100*gka, 1 ;; a1 oscil 10000, 440, 1 out a1 endin f1 0 4096 10 1 i1 0 300 manual_src~dfsg/examples/ATSadd-2.csd0000644000000000000000000000254512262561504016523 0ustar rootroot -odac -d -m1 ;example by joachim heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 1024, 10, 1 instr AllTheTones Sfile = "fox.ats" prints "Resynthesizing with all the tones.\n" iDur ATSinfo Sfile, 7 p3 = iDur iNumParts ATSinfo Sfile, 3 prints "Overall number of partials = %d\n", iNumParts ktime line 0, iDur, iDur asig ATSadd ktime, 1, Sfile, giSine, iNumParts outs asig, asig ;start next instr event_i "i", "TonesInBandsOfTen", iDur+1, iDur, 0, iNumParts endin instr TonesInBandsOfTen Sfile = "fox.ats" iOffset = p4 ;start at this partial iNumParts = p5 ;overall number of partials prints "Resynthesizing with partials %d - %d.\n", iOffset+1, iOffset+10 ktime line 0, p3, p3 asig ATSadd ktime, 1, Sfile, giSine, 10, iOffset outs asig, asig ;start next instance until there are enough partials left if iOffset+20 < iNumParts then event_i "i", "TonesInBandsOfTen", p3+1, p3, iOffset+10, iNumParts else event_i "i", "End", p3, 1 endif endin instr End exitnow endin i "AllTheTones" 0 1 e 999 manual_src~dfsg/examples/poscil.csd0000644000000000000000000000127512262561504016554 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o poscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 gisine ftgen 0, 0, 2^10, 10, 1 instr 1 ipeak random 0, 1 ;where is the envelope peak asig poscil .8, 220, gisine aenv transeg 0, p3*ipeak, 6, 1, p3-p3*ipeak, -6, 0 aL,aR pan2 asig*aenv, ipeak ;pan according to random value outs aL, aR endin i1 0 5 i1 4 5 i1 8 5 e manual_src~dfsg/examples/makecsd.py0000644000000000000000000000220512262561504016543 0ustar rootrootimport glob import os, re files = glob.glob('*.orc') files.sort() files.remove('ambisonic.orc') files.remove('fluid.orc') files.remove('fluidAllOut.orc') for file in files: #print file handleorc = open(file, 'r') handlesco = open(file[:file.index('.orc')]+'.sco') #print file[:file.index('.sco')]+'.csd' newtext = ''' ; Select audio/midi flags here according to platform ; Audio out Audio in Silent MIDI in -odac -idac -d -M0 ;;;realtime output ; For Non-realtime ouput leave only the line below: ; -o ''' + file[:file.index('.orc')] + '''.wav -W ;;; for file output any platform ''' commentorc = re.compile('/.*[0-9a-z]*\.orc.*/[\N\R]*') commentsco = re.compile('/.*[0-9a-z]*\.sco.*/[\N\R]*') orc = handleorc.read() orc = commentorc.sub('',orc) sco = handlesco.read() sco = commentsco.sub('',sco) newtext += orc + '\n\n\n' + sco + '\n\n\n' #print newtext dest = open(file[:file.index('.orc')]+'.csd','w') dest.write(newtext) dest.close() handleorc.close() handlesco.close() manual_src~dfsg/examples/crossfm.csd0000644000000000000000000000153612262561504016737 0ustar rootroot -d -o dac sr = 96000 ksmps = 10 nchnls = 2 0dbfs = 1 FLpanel "crossfmForm", 600, 400, 0, 0 gkfrq1, ihfrq1 FLcount "Freq #1", 0, 20000, 0.001, 1, 1, 200, 30, 20, 50, -1 gkfrq2, ihfrq2 FLcount "Freq #2", 0, 20000, 0.001, 1, 1, 200, 30, 20, 130, -1 gkndx1, gkndx2, ihndx1, ihndx2 FLjoy "Indexes", 0, 10, 0, 10, 0, 0, -1, -1, 200, 200, 300, 50 FLsetVal_i 164.5, ihfrq1 FLsetVal_i 263.712, ihfrq2 FLsetVal_i 1.5, ihndx1 FLsetVal_i 3, ihndx2 FLpanelEnd FLrun maxalloc 1, 2 instr 1 kamp linen 0.5, 0.01, p3, 0.5 a1,a2 crossfm gkfrq1, gkfrq2, gkndx1, gkndx2, 1, 1, 1 outs a1*kamp, a2*kamp endin f1 0 16384 10 1 0 i1 0 60 e manual_src~dfsg/examples/dumpk2.csd0000644000000000000000000000226612262561504016466 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dumpk2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes two control signals to a file kfreq randh 100, 1, 2, 1, 500 ;generates one random number between 400 and 600 per second kdb randh 12, 1, 2, 1, -12 ;amplitudes in dB between -24 and 0 dumpk2 kfreq, kdb, "dumpk2.txt", 8, 1 ;writes the control signals prints "WRITING:\n" printks "kfreq = %f, kdb = %f\n", 1, kfreq, kdb ;prints them endin instr 2 ;reads the file written by instr 1 kf,kdb readk2 "dumpk2.txt", 8, 1 prints "READING:\n" printks "kfreq = %f, kdb = %f\n", 1, kf, kdb ;prints again kdb lineto kdb, .1 ;smoothing amp transition aout poscil ampdb(kdb), kf, giSine outs aout, aout endin i 1 0 5 i 2 5 5 e manual_src~dfsg/examples/toot6.csd0000644000000000000000000000355512262561504016341 0ustar rootroot -odac instr 6 ifunc = p11 ; select the basic waveform irel = 0.01 ; set vibrato release time idel1 = p3 * p10 ; calculate initial delay (% of dur) isus = p3 - (idel1 + irel) ; calculate remaining duration iamp = ampdb(p4) iscale = iamp * .333 ; p4=amp inote = cpspch(p5) ; p5=freq k3 linseg 0, idel1, p9, isus, p9, irel, 0 ; p6=attack time k2 oscil k3, p8, 1 ; p7=release time k1 linen iscale, p6, p3, p7 ; p8=vib rate a3 oscil k1, inote*.999+k2, ifunc ; p9=vib depth a2 oscil k1, inote*1.001+k2, ifunc ; p10=vib delay (0-1) a1 oscil k1, inote+k2, ifunc out a1+a2+a3 endin f1 0 2048 10 1 ; Sine f2 0 2048 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth f3 0 2048 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square f4 0 2048 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse ;ins strt dur amp frq atk rel vbrt vbdpt vibdl waveform i6 0 2 86 8.00 .03 .7 6 9 .8 1 i6 3 2 86 8.02 .03 .7 6 9 .8 2 i6 6 2 86 8.04 .03 .7 6 9 .8 3 i6 9 3 86 8.05 .03 .7 6 9 .8 manual_src~dfsg/examples/locsig_quad.csd0000644000000000000000000000204012262561504017544 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o locsig_quad.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 ga1 init 0 ga2 init 0 ga3 init 0 ga4 init 0 instr 1 krevsend = p4 aout diskin2 "beats.wav", 1, 0, 1 kdegree line 0, p3, 360 ;full circle kdistance line 1, p3 , 1 a1, a2, a3, a4 locsig aout, kdegree, kdistance, krevsend ar1, ar2, ar3, ar4 locsend ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 3.5, .5 a2 reverb2 ga2, 3.5, .5 a3 reverb2 ga3, 3.5, .5 a4 reverb2 ga4, 3.5, .5 outq a1, a2, a3, a4 ga1 = 0 ga2 = 0 ga3 = 0 ga4 = 0 endin ; sine wave. f 1 0 16384 10 1 i 1 0 14 .1 ;with reverb i 1 14 14 0 ;no reverb i99 0 36 e manual_src~dfsg/examples/prealloc.csd0000644000000000000000000000157612262561504017070 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o prealloc.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Pre-allocate memory for five instances of Instrument #1. prealloc 1, 5 ; Instrument #1 instr 1 ; Generate a waveform, get the cycles per second from the 4th p-field. a1 oscil 6500, p4, 1 out a1 endin ; Just generate a nice, ordinary sine wave. f 1 0 32768 10 1 ; Play five instances of Instrument #1 for one second. ; Note that 4th p-field contains cycles per second. i 1 0 1 220 i 1 0 1 440 i 1 0 1 880 i 1 0 1 1320 i 1 0 1 1760 e manual_src~dfsg/examples/compileorc.csd0000644000000000000000000000057312262561504017417 0ustar rootroot -o dac -d sr = 44100 nchnls = 1 ksmps = 32 0dbfs = 1 instr 1 ires compileorc "does_not_exist.orc" print ires ; -1 as could not compile ires compileorc "my.orc" print ires ; 0 as compiled successfully event_i "i", 2, 0, 3, .2, 465 ;send event endin i1 0 1 manual_src~dfsg/examples/chnclear.csd0000644000000000000000000000352112262561504017036 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chnclear.wav -W ;;; for file output any platform ;Example by Joachim Heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1; send i-values chnset 1, "sio" chnset -1, "non" endin instr 2; send k-values kfreq randomi 100, 300, 1 chnset kfreq, "cntrfreq" kbw = kfreq/10 chnset kbw, "bandw" endin instr 3; send a-values anois rand .1 chnset anois, "noise" loop: idur random .3, 1.5 timout 0, idur, do reinit loop do: ifreq random 400, 1200 iamp random .1, .3 asig oscils iamp, ifreq, 0 aenv transeg 1, idur, -10, 0 asine = asig * aenv chnset asine, "sine" endin instr 11; receive some chn values and send again ival1 chnget "sio" ival2 chnget "non" print ival1, ival2 kcntfreq chnget "cntrfreq" kbandw chnget "bandw" anoise chnget "noise" afilt reson anoise, kcntfreq, kbandw afilt balance afilt, anoise chnset afilt, "filtered" endin instr 12; mix the two audio signals amix1 chnget "sine" amix2 chnget "filtered" chnmix amix1, "mix" chnmix amix2, "mix" endin instr 20; receive and reverb amix chnget "mix" aL, aR freeverb amix, amix, .8, .5 outs aL, aR endin instr 100; clear chnclear "mix" endin i 1 0 20 i 2 0 20 i 3 0 20 i 11 0 20 i 12 0 20 i 20 0 20 i 100 0 20 manual_src~dfsg/examples/midremot.csd0000644000000000000000000000350312262561504017077 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o midremot.wav -W ;;; for file output any platform sr = 44100 kr = 441 ksmps = 100 nchnls = 2 massign 1,1 massign 2,2 massign 3,3 massign 4,4 massign 5,5 ga1 init 0 ga2 init 0 gi1 sfload "19Trumpet.sf2" gi2 sfload "01hpschd.sf2" gi3 sfload "07AcousticGuitar.sf2" gi4 sfload "22Bassoon.sf2" gitab ftgen 1,0,1024,10,1 midremot "192.168.1.100", "192.168.1.101", 1 midremot "192.168.1.100", "192.168.1.102", 2 midremot "192.168.1.100", "192.168.1.103", 3 midglobal "192.168.1.100", 5 instr 1 sfpassign 0, gi1 ifreq cpsmidi iamp ampmidi 10 inum notnum ivel veloc kamp linsegr 1,1,1,.1,0 kfreq init 1 a1,a2 sfplay ivel,inum,kamp*iamp,kfreq,0,0 outs a1,a2 vincr ga1, a1*.5 vincr ga2, a2*.5 endin instr 2 sfpassign 0, gi2 ifreq cpsmidi iamp ampmidi 15 inum notnum ivel veloc kamp linsegr 1,1,1,.1,0 kfreq init 1 a1,a2 sfplay ivel,inum,kamp*iamp,kfreq,0,0 outs a1,a2 vincr ga1, a1*.4 vincr ga2, a2*.4 endin instr 3 sfpassign 0, gi3 ifreq cpsmidi iamp ampmidi 10 inum notnum ivel veloc kamp linsegr 1,1,1,.1,0 kfreq init 1 a1,a2 sfplay ivel,inum,kamp*iamp,kfreq,0,0 outs a1,a2 vincr ga1, a1*.5 vincr ga2, a2*.5 endin instr 4 sfpassign 0, gi4 ifreq cpsmidi iamp ampmidi 15 inum notnum ivel veloc kamp linsegr 1,1,1,.1,0 kfreq init 1 a1,a2 sfplay ivel,inum,kamp*iamp,kfreq,0,0 outs a1,a2 vincr ga1, a1*.5 vincr ga2, a2*.5 endin instr 5 kamp midic7 1,0,1 denorm ga1 denorm ga2 aL, aR reverbsc ga1, ga2, .9, 16000, sr, 0.5 outs aL, aR ga1 = 0 ga2 = 0 endin ; Score f0 160 manual_src~dfsg/examples/FLjoy.csd0000644000000000000000000000353512262561504016307 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLjoy.wav -W ;;; for file output any platform ; Demonstration of the flpanel opcode ; Horizontal click-dragging controls the frequency of the oscillator ; Vertical click-dragging controls the amplitude of the oscillator sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "X Y Panel", 900, 400, 50, 50 ; Minimum value output by x movement (frequency) iminx = 200 ; Maximum value output by x movement (frequency) imaxx = 5000 ; Minimum value output by y movement (amplitude) iminy = 0 ; Maximum value output by y movement (amplitude) imaxy = 15000 ; Logarithmic change in x direction iexpx = -1 ; Linear change in y direction iexpy = 0 ; Display handle x direction (-1=not used) idispx = -1 ; Display handle y direction (-1=not used) idispy = -1 ; Width of the x y panel in pixels iwidth = 800 ; Height of the x y panel in pixels iheight = 300 ; Distance of the left edge of the x y panel from ; the left edge of the panel ix = 50 ; Distance of the top edge of the x y ; panel from the top edge of the panel iy = 50 gkfreqx, gkampy, ihandlex, ihandley FLjoy "X - Frequency Y - Amplitude", iminx, imaxx, iminy, imaxy, iexpx, iexpy, idispx, idispy, iwidth, iheight, ix, iy ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 ifn = 1 asig oscili gkampy, gkfreqx, ifn out asig endin ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e manual_src~dfsg/examples/powoftwo.csd0000644000000000000000000000163212262561504017144 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o poweroftwo.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; bit reduction for Lo-Fi sound iBits = p5 ;bit depth iQuantize = powoftwo(iBits) ;find number of discrete steps for this bit depth iQuantize = iQuantize*0.5 ;half the number of steps for each side of a bipolar signal print iQuantize asig soundin "fox.wav" asig = round(asig * iQuantize) / iQuantize ;quantize audio signal (bit reduce) outs asig, asig endin ; bits i1 0 3 16 i1 ^+3 . 12 i1 ^+3 . 8 i1 ^+3 . 4 i1 ^+3 . 2 i1 ^+3 . 1 e manual_src~dfsg/examples/STKBlowHole.csd0000644000000000000000000000123512262561504017354 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKBlowHole.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = p4 kv1 = p7 ;stiffness of reed kv3 line p5, p3, p6 ;state of tonehole asig STKBlowHole cpspch(ipch), 1, 2, kv1, 4, 100, 11, kv3, 1, 10, 128, 100 outs asig, asig endin i 1 0 4 10.00 20 127 100 i 1 + 7 6.09 120 0 10 e manual_src~dfsg/examples/dssiplay_stereo.csd0000644000000000000000000000220512262561504020466 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o DSSIplay_stereo.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 gihandle dssiinit "g2reverb.so", 0, 1 gaout init 0 instr 1 ; activate DSSI dssiactivate gihandle, 1 endin instr 2 ain1 diskin2 "beats.wav", 1 gaout = gaout+(ain1*.3) endin instr 3 dssictls gihandle, 4, 100, 1 ; room 10 to 150 dssictls gihandle, 5, 10, 1 ; reverb time 1 to 20 dssictls gihandle, 6, .5, 1 ; input bandwith 0 to 1 dssictls gihandle, 7, .25, 1 ; damping 0 to 1 dssictls gihandle, 8, 0, 1 ; dry -80 to 0 dssictls gihandle, 9, -10, 1 ; reflections -80 to 0 dssictls gihandle, 10, -15, 1 ; rev. tail -80 to 0 endin instr 4 aout1, aout2 dssiaudio gihandle, gaout, gaout ;get beats.wav and outs aout1,aout2 ; stereo DSSI plugin gaout = 0 endin i 1 0 2 i 2 1 10 i 3 1 10 i 4 0 10 e manual_src~dfsg/examples/FLpanel-sensekey.csd0000644000000000000000000000141112262561504020420 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLpanel-sensekey.wav -W ;;; for file output any platform ; Example by Johnathan Murphy sr = 44100 ksmps = 128 nchnls = 2 ; ikbdcapture flag set to 1 ikey init 1 FLpanel "sensekey", 740, 340, 100, 250, 2, ikey gkasc, giasc FLbutBank 2, 16, 8, 700, 300, 20, 20, -1 FLpanelEnd FLrun instr 1 kkey sensekey kprint changed kkey FLsetVal kprint, kkey, giasc endin i1 0 60 e manual_src~dfsg/examples/zkcl.csd0000644000000000000000000000226012262561504016221 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zkcl.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Linearly vary a k-rate signal from 220 to 1760. kline line 220, p3, 1760 ; Add the linear signal to zk variable #1. zkw kline, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read zk variable #1. kfreq zkr 1 ; Use the value of zk variable #1 to vary ; the frequency of a sine waveform. a1 oscil 20000, kfreq, 1 ; Generate the audio output. out a1 ; Clear the zk variables, get them ready for ; another pass. zkcl 0, 1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 ; Play Instrument #2 for three seconds. i 2 0 3 e manual_src~dfsg/examples/gen40.csd0000644000000000000000000000155412262561504016200 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o GEN40.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values kuser cuserrnd 0, 100, 1 printk .2, kuser asig poscil .5, 220+kuser, 3 outs asig, asig endin instr 2 ; every run time different values seed 0 kuser cuserrnd 0, 100, 1 printk .2, kuser asig poscil .5, 220+kuser, 3 outs asig, asig endin f 1 0 16 -7 1 4 0 8 0 4 1 ;distrubution using GEN07 f 2 0 16384 40 1 ;GEN40 is to be used with cuserrnd f 3 0 8192 10 1 ;sine i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/instr.csd0000644000000000000000000000110412262561504016411 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o instr.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 iamp = 10000 icps = 440 iphs = 0 a1 oscils iamp, icps, iphs out a1 endin ; Play Instrument #1 for 2 seconds. i 1 0 2 e manual_src~dfsg/examples/ftlen.csd0000644000000000000000000000123712262561504016371 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftlen.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 instr 1 ift = ftlen(p4) print ift aout loscil3 .8, 4, p4 outs aout, aout endin f 1 0 0 1 "fox.wav" 0 0 0 ;Csound computes tablesize f 2 0 0 1 "beats.wav" 0 0 0 ;Csound computes tablesize i 1 0 3 1 ;"fox.wav" i 1 3 3 2 ;"beats.wav" e manual_src~dfsg/examples/phaser1.csd0000644000000000000000000000271212262561504016623 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o phaser1.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; demonstration of phase shifting abilities of phaser1. instr 1 ; Input mixed with output of phaser1 to generate notches. ; Shows the effects of different iorder values on the sound idur = p3 iamp = p4 * .05 iorder = p5 ; number of 1st-order stages in phaser1 network. ; Divide iorder by 2 to get the number of notches. ifreq = p6 ; frequency of modulation of phaser1 ifeed = p7 ; amount of feedback for phaser1 kamp linseg 0, .2, iamp, idur - .2, iamp, .2, 0 iharms = (sr*.4) / 100 asig gbuzz 1, 100, iharms, 1, .95, 2 ; "Sawtooth" waveform modulation oscillator for phaser1 ugen. kfreq oscili 5500, ifreq, 1 kmod = kfreq + 5600 aphs phaser1 asig, kmod, iorder, ifeed out (asig + aphs) * iamp endin ; inverted half-sine, used for modulating phaser1 frequency f1 0 16384 9 .5 -1 0 ; cosine wave for gbuzz f2 0 8192 9 1 1 .25 ; phaser1 i1 0 5 7000 4 .2 .9 i1 6 5 7000 6 .2 .9 i1 12 5 7000 8 .2 .9 i1 18 5 7000 16 .2 .9 i1 24 5 7000 32 .2 .9 i1 30 5 7000 64 .2 .9 e manual_src~dfsg/examples/tablei.csd0000644000000000000000000000227612262561504016525 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tablei.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 ;generate new values every time the instr is played instr 1 ifn = p4 isize = p5 ithresh = 0.5 itemp ftgen ifn, 0, isize, 21, 2 iwrite_value = 0 i_index = 0 loop_start: iread_value tablei i_index, ifn if iread_value > ithresh then iwrite_value = 1 else iwrite_value = -1 endif tableiw iwrite_value, i_index, ifn loop_lt i_index, 1, isize, loop_start turnoff endin instr 2 ifn = p4 isize = ftlen(ifn) prints "Index\tValue\n" i_index = 0 loop_start: ivalue tablei i_index, ifn prints "%d:\t%f\n", i_index, ivalue loop_lt i_index, 1, isize, loop_start ;read table 1 with our index aout oscili .5, 100, ifn ;use table to play the polypulse outs aout, aout endin i 1 0 1 100 16 i 2 0 2 100 e manual_src~dfsg/examples/lpreson.csd0000644000000000000000000000174312262561504016745 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpreson.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; works with or without -a option when analyzing "fox.wav" from the manual ;both options sound a little different instr 1 ilen filelen "fox.wav" ; length of soundfile prints "fox.wav = %f seconds\\n",ilen ktime line 0, p3, p4 krmsr,krmso,kerr,kcps lpread ktime,"fox_poles.lpc" krmso = krmso*.00001 ; low volume asig buzz krmso, kcps, int(sr/2/kcps), 1 ; max harmonics without aliasing aout lpreson asig asig clip aout, 0, 1 ; prevents distortion outs asig, asig endin ; sine f1 0 4096 10 1 i 1 0 2.8 1 ; first words only i 1 4 2.8 2.8 ; whole sentence e manual_src~dfsg/examples/lpreson-2.csd0000644000000000000000000000147712262561504017110 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpreson-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; works with or without -a option when analyzing "fox.wav" from the manual ;both options sound a little different instr 1 ilen filelen "fox.wav" ;length of soundfile 1 prints "fox.wav = %f seconds\\n",ilen ktime line 0, p3, ilen krmsr,krmso,kerr,kcps lpread ktime,"fox_nopoles.lpc" asig diskin2 "flute.aiff", 1 aout lpreson asig ares balance aout, asig outs ares, ares endin i 1 0 2.8 e manual_src~dfsg/examples/midion_scale.csd0000644000000000000000000000147312262561504017711 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1 ;;;RT audio I/O with MIDI in sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 instr 1 ; Triggered by MIDI notes on channel 1 ivel veloc krate = 8 iscale = 100 ;f ; Random sequence from table f100 krnd randh int(14),krate,-1 knote table abs(krnd),iscale ; Generates random notes from the scale on ftable 100 ; on channel 1 of csound's MIDI output midion 1,knote,ivel endin f100 0 32 -2 40 50 60 70 80 44 54 65 74 84 39 49 69 69 ; Dummy ftable f0 60 manual_src~dfsg/examples/display.csd0000644000000000000000000000122512262561504016723 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o display.wav -W ;;; for file output any platform ;be sure to NOT have -d in the CsOptions... sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 110 ifn = 1 knh line p4, p3, p5 asig buzz 1, kcps, knh, ifn outs asig, asig display asig, .1, 3 endin ;sine wave. f 1 0 16384 10 1 i 1 0 3 20 20 i 1 + 3 3 3 i 1 + 3 150 1 e manual_src~dfsg/examples/slicearray.csd0000644000000000000000000000214412262561504017415 0ustar rootroot ; Select audio/midi flags here according to platform -n sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;create and fill an array kArr[] genarray_i 1, 9 ;print the content printf "%s", 1, "kArr = whole array\n" kndx = 0 until kndx == lenarray(kArr) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr[kndx] kndx += 1 od ;build new arrays for the slices kArr1[] init 5 kArr2[] init 4 ;put in first five and last four elements kArr1 slicearray kArr, 0, 4 kArr2 slicearray kArr, 5, 8 ;print the content printf "%s", 1, "\nkArr1 = slice from index 0 to index 4\n" kndx = 0 until kndx == lenarray(kArr1) do printf "kArr1[%d] = %f\n", kndx+1, kndx, kArr1[kndx] kndx += 1 od printf "%s", 1, "\nkArr2 = slice from index 5 to index 8\n" kndx = 0 until kndx == lenarray(kArr2) do printf "kArr2[%d] = %f\n", kndx+1, kndx, kArr2[kndx] kndx += 1 od turnoff endin i 1 0 1 manual_src~dfsg/examples/outkat.csd0000644000000000000000000000136212262561504016567 0ustar rootroot ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out and midi in and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outkat.wav -W ;;; for file output any platform sr = 44100 ksmps = 10 nchnls = 2 instr 1 ikey notnum ivel veloc kvib linseg 100, .5, 120 ;vary aftertouch in .5 second kvbr = int(kvib) ;whole numbers only outkat 1, kvbr, 0, 127 ;assign aftertouch on printk2 kvbr ;external synth for example to midion 1, ikey, ivel ;change depth of filter modulation endin f0 30 ;play for 30 seconds e manual_src~dfsg/examples/strset.csd0000644000000000000000000000100012262561504016571 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O sr = 44100 ksmps = 1 nchnls = 1 ;Example by Andres Cabrera 2008 ; \\n is used to denote "new line" strset 1, "String 1\\n" strset 2, "String 2\\n" instr 1 Str strget p4 prints Str endin ; p4 is used to select string i 1 0 1 1 i 1 3 1 2 manual_src~dfsg/examples/lorismorph.csd0000644000000000000000000000555612262561504017467 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lorismorph.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; clarinet.sdif and meow.sdif can be found in /manual/examples ; Morph the partials in meow.sdif into the partials in clarinet.sdif over the duration of ; the sustained portion of the two tones (from .2 to 2.0 seconds in the meow, and from ; .5 to 2.1 seconds in the clarinet). The onset and decay portions in the morphed sound are ; specified by parameters p4 and p5, respectively. The morphing time is the time between the ; onset and the decay. The meow partials are shfited in pitch to match the pitch of the clarinet ; tone (D above middle C). instr 1 ionset = p4 idecay = p5 itmorph = p3 - (ionset + idecay) ipshift = cpspch(8.02)/cpspch(8.08) ktme linseg 0, ionset, .2, itmorph, 2.0, idecay, 2.1 ; meow time function, morph from .2 to 2.0 seconds ktcl linseg 0, ionset, .5, itmorph, 2.1, idecay, 2.3 ; clarinet time function, morph from .5 to 2.1 seconds kmurph linseg 0, ionset, 0, itmorph, 1, idecay, 1 lorisread ktme, "meow.sdif", 1, ipshift, 2, 1, .001 lorisread ktcl, "clarinet.sdif", 2, 1, 1, 1, .001 lorismorph 1, 2, 3, kmurph, kmurph, kmurph asig lorisplay 3, 1, 1, 1 outs asig, asig endin ; Morph the partials in clarinet.sdif into the partials in meow.sdif. The start and end times ; for the morph are specified by parameters p4 and p5, respectively. The morph occurs over the ; second of four pitches in each of the sounds, from .75 to 1.2 seconds in the flutter-tongued ; clarinet tone, and from 1.7 to 2.2 seconds in the cat's meow. Different morphing functions are ; used for the frequency and amplitude envelopes, so that the partial amplitudes make a faster ; transition from clarinet to cat than the frequencies. (The bandwidth envelopes use the same ; morphing function as the amplitudes.) instr 2 ionset = p4 imorph = p5 - p4 irelease = p3 - p5 ktclar linseg 0, ionset, .75, imorph, 1.2, irelease, 2.4 ktmeow linseg 0, ionset, 1.7, imorph, 2.2, irelease, 3.4 kmfreq linseg 0, ionset, 0, .75*imorph, .25, .25*imorph, 1, irelease, 1 kmamp linseg 0, ionset, 0, .75*imorph, .9, .25*imorph, 1, irelease, 1 lorisread ktclar, "clarinet.sdif", 1, 1, 1, 1, .001 lorisread ktmeow, "meow.sdif", 2, 1, 1, 1, .001 lorismorph 1, 2, 3, kmfreq, kmamp, kmamp asig lorisplay 3, 1, 1, 1 outs asig, asig endin ; strt dur onset decay i 1 0 3 .25 .15 i 1 + 1 .10 .10 i 1 + 6 1. 1. ; strt dur morph_start morph_end i 2 9 4 .75 2.75 e manual_src~dfsg/examples/tableseg.csd0000644000000000000000000000130612262561504017044 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tableseg.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; analyze "fox.wav" with PVANAL first iend = p4 ktime line 0, p3, iend tableseg p5, p3, p6 ;morph from table 1 asig vpvoc ktime, 1, "fox.pvx" ;to table 2 outs asig*3, asig*3 endin f 1 0 512 9 .5 1 0 f 2 0 512 7 0 20 1 30 0 230 0 232 1 i 1 0 10 2.7 1 2 e manual_src~dfsg/examples/endop.csd0000644000000000000000000000401212262561504016360 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o endop.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 /* example opcode 1: simple oscillator */ opcode Oscillator, a, kk kamp, kcps xin ; read input parameters a1 vco2 kamp, kcps ; sawtooth oscillator xout a1 ; write output endop /* example opcode 2: lowpass filter with local ksmps */ opcode Lowpass, a, akk setksmps 1 ; need sr=kr ain, ka1, ka2 xin ; read input parameters aout init 0 ; initialize output aout = ain*ka1 + aout*ka2 ; simple tone-like filter xout aout ; write output endop /* example opcode 3: recursive call */ opcode RecursiveLowpass, a, akkpp ain, ka1, ka2, idep, icnt xin ; read input parameters if (icnt >= idep) goto skip1 ; check if max depth reached ain RecursiveLowpass ain, ka1, ka2, idep, icnt + 1 skip1: aout Lowpass ain, ka1, ka2 ; call filter xout aout ; write output endop /* example opcode 4: de-click envelope */ opcode DeClick, a, a ain xin aenv linseg 0, 0.02, 1, p3 - 0.05, 1, 0.02, 0, 0.01, 0 xout ain * aenv ; apply envelope and write output endop /* instr 1 uses the example opcodes */ instr 1 kamp = .6 ; amplitude kcps expon 50, p3, 500 ; pitch a1 Oscillator kamp, kcps ; call oscillator kflt linseg 0.4, 1.5, 0.4, 1, 0.8, 1.5, 0.8 ; filter envelope a1 RecursiveLowpass a1, kflt, 1 - kflt, 10 ; 10th order lowpass a1 DeClick a1 outs a1, a1 endin i 1 0 4 e manual_src~dfsg/examples/0dbfs-1.csd0000644000000000000000000000162312262561504016414 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o 0dbfs.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Set the 0dbfs to 1. 0dbfs = 1 ; Instrument #1. instr 1 ; Linearly increase the amplitude value "kamp" from ; -90 to p4 (in dBfs) over the duration defined by p3. kamp line -90, p3, p4 print ampdbfs(p4) ; Generate a basic tone using our amplitude value. a1 oscil ampdbfs(kamp), 440, 1 ; Since 0dbfs = 1 we don't need to multiply the output out a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 -6 e manual_src~dfsg/examples/FLsavesnap.csd0000644000000000000000000000735212262561504017327 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O sr=48000 ksmps=128 nchnls=2 ; Example by Hector Centeno and Andres Cabrera 2007 ; giSWMtab4 ftgen 0, 0, 513, 21, 10, 1, .3 ; giSWMtab4M ftgen 0, 0, 64, 7, 1, 50, 1 FLpanel "Snapshots", 530, 350, 40, 410, 3 FLcolor 100, 118 ,140 FLsetSnapGroup 0 ivalSM1 FLvalue "", 70, 20, 270, 20 ivalSM2 FLvalue "", 70, 20, 270, 60 ivalSM3 FLvalue "", 70, 20, 270, 100 ivalSM4 FLvalue "", 70, 20, 270, 140 gksliderA, gislidSM1 FLslider "Slider A", -4, 4, 0, 3, ivalSM1, 250, 20, 20, 20 gksliderB, gislidSM2 FLslider "Slider B", 1, 10, 0, 3, ivalSM2, 250, 20, 20, 60 gksliderC, gislidSM3 FLslider "Slider C", 0, 1, 0, 3, ivalSM3, 250, 20, 20, 100 gksliderD, gislidSM4 FLslider "Slider D", 0, 1, 0, 3, ivalSM4, 250, 20, 20, 140 itext1 FLbox "store", 1, 1, 14, 50, 25, 355, 15 itext2 FLbox "load", 1, 1, 14, 50, 25, 415, 15 itext3 FLbox "Group 1", 1, 1, 14, 30, 145, 485, 15 gksnap, ibuttn1 FLbutton "1", 1, 0, 11, 25, 25, 364, 45, 0, 3, 0, 3, 1 gksnap, ibuttn2 FLbutton "2", 1, 0, 11, 25, 25, 364, 75, 0, 3, 0, 3, 2 gksnap, ibuttn3 FLbutton "3", 1, 0, 11, 25, 25, 364, 105, 0, 3, 0, 3, 3 gksnap, ibuttn4 FLbutton "4", 1, 0, 11, 25, 25, 364, 135, 0, 3, 0, 3, 4 gkload, ibuttn1 FLbutton "1", 1, 0, 11, 25, 25, 424, 45, 0, 4, 0, 3, 1 gkload, ibuttn2 FLbutton "2", 1, 0, 11, 25, 25, 424, 75, 0, 4, 0, 3, 2 gkload, ibuttn3 FLbutton "3", 1, 0, 11, 25, 25, 424, 105, 0, 4, 0, 3, 3 gkload, ibuttn4 FLbutton "4", 1, 0, 11, 25, 25, 424, 135, 0, 4, 0, 3, 4 FLcolor 100, 140 ,118 FLsetSnapGroup 1 ivalSM5 FLvalue "", 70, 20, 270, 190 ivalSM6 FLvalue "", 70, 20, 270, 230 ivalSM7 FLvalue "", 70, 20, 270, 270 ivalSM8 FLvalue "", 70, 20, 270, 310 gkknobA, gislidSM5 FLknob "Knob A", -4, 4, 0, 3, ivalSM5, 45, 10, 230 gkknobB, gislidSM6 FLknob "Knob B", 1, 10, 0, 3, ivalSM6, 45, 75, 230 gkknobC, gislidSM7 FLknob "Knob C", 0, 1, 0, 3, ivalSM7, 45, 140, 230 gkknobD, gislidSM8 FLknob "Knob D", 0, 1, 0, 3, ivalSM8, 45, 205, 230 itext4 FLbox "store", 1, 1, 14, 50, 25, 355, 185 itext5 FLbox "load", 1, 1, 14, 50, 25, 415, 185 itext6 FLbox "Group 2", 1, 1, 14, 30, 145, 485, 185 gksnap, ibuttn1 FLbutton "5", 1, 0, 11, 25, 25, 364, 215, 0, 3, 0, 3, 5 gksnap, ibuttn2 FLbutton "6", 1, 0, 11, 25, 25, 364, 245, 0, 3, 0, 3, 6 gksnap, ibuttn3 FLbutton "7", 1, 0, 11, 25, 25, 364, 275, 0, 3, 0, 3, 7 gksnap, ibuttn4 FLbutton "8", 1, 0, 11, 25, 25, 364, 305, 0, 3, 0, 3, 8 gkload, ibuttn1 FLbutton "5", 1, 0, 11, 25, 25, 424, 215, 0, 4, 0, 3, 5 gkload, ibuttn2 FLbutton "6", 1, 0, 11, 25, 25, 424, 245, 0, 4, 0, 3, 6 gkload, ibuttn3 FLbutton "7", 1, 0, 11, 25, 25, 424, 275, 0, 4, 0, 3, 7 gkload, ibuttn4 FLbutton "8", 1, 0, 11, 25, 25, 424, 305, 0, 4, 0, 3, 8 FLpanelEnd FLsetVal_i 1, gislidSM1 FLsetVal_i 1, gislidSM2 FLsetVal_i 0, gislidSM3 FLsetVal_i 0, gislidSM4 FLsetVal_i 1, gislidSM5 FLsetVal_i 1, gislidSM6 FLsetVal_i 0, gislidSM7 FLsetVal_i 0, gislidSM8 FLrun instr 1 endin instr 3 ; Save snapshot index init 0 ipstno = p4 igroup = 0 Sfile sprintf "PVCsynth.%d.snap", ipstno if ipstno > 4 then igroup = 1 endif inumsnap, inumval FLsetsnap index , -1, igroup FLsavesnap Sfile endin instr 4 ;Load snapshot index init 0 ipstno = p4 igroup = 0 Sfile sprintf "PVCsynth.%d.snap", ipstno if ipstno > 4 then igroup = 1 endif FLloadsnap Sfile inumload FLgetsnap index , igroup endin ;Dummy table for FLgetsnap ; f 1 0 1024 10 1 f 0 3600 e manual_src~dfsg/examples/pvsbufread2.csd0000644000000000000000000000126712262561504017507 0ustar rootroot ksmps = 64 0dbfs = 1 nchnls = 2 instr 1 kcnt init 0 ifftsize = 2048 ihop = ifftsize/4 a1 diskin2 "beats.wav", 1, 0, 1 a1 = a1*0.5 fsig1 pvsanal a1, ifftsize, ihop, ifftsize, 1 ih, kt pvsbuffer fsig1, 10 fsig2 pvsbufread2 kt, ih, 1, 1 fsig3 pvsbufread2 kt, ih, 2, 2 a2 pvsynth fsig3 a3 pvsynth fsig2 outs a2, a3 endin f1 0 2048 -7 0 128 1.1 128 0.5 256 1.8 512 1.1 1024 0.1 f2 0 2048 -7 1 128 0.2 128 0.05 256 0.5 512 0.9 1024 0.1 i1 0 60 manual_src~dfsg/examples/mandol.csd0000644000000000000000000000145612262561504016536 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mandol.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = p4 ksize = p5 kdetune = p6 asig mandol kamp, 880, .4, kdetune, 0.99, ksize, 1, 220 outs asig, asig endin ; "mandpluk.aiff" audio file f 1 0 8192 1 "mandpluk.aiff" 0 0 0 i 1 .5 1 1 2 .99 i 1 + 1 .5 1 .99 ;lower volume to compensate i 1 + 3 .3 .3 .99 ;lower volume to compensate i 1 4 1 1 2 .39 ;change detune value i 1 + 1 .5 1 .39 i 1 + 3 .3 .3 .39 e manual_src~dfsg/examples/sndwarp.csd0000644000000000000000000000171112262561504016734 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sndwarp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ktimewarp line 0, p3, 2.7 ;length of "fox.wav" kresample init 1 ;do not change pitch ibeg = 0 ;start at beginning iwsize = 4410 ;window size in samples with irandw = 882 ;bandwidth of a random number generator itimemode = 1 ;ktimewarp is "time" pointer ioverlap = p4 asig sndwarp .5, ktimewarp, kresample, 1, ibeg, iwsize, irandw, ioverlap, 2, itimemode outs asig, asig endin f 1 0 131072 1 "fox.wav" 0 0 0 ; audio file f 2 0 1024 9 0.5 1 0 ; half of a sine wave i 1 0 7 2 ;different overlaps i 1 + 7 5 i 1 + 7 15 e manual_src~dfsg/examples/zaw.csd0000644000000000000000000000205012262561504016054 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zaw.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Generate a simple sine waveform. asin oscil 20000, 440, 1 ; Send the sine waveform to za variable #1. zaw asin, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read za variable #1. a1 zar 1 ; Generate the audio output. out a1 ; Clear the za variables, get them ready for ; another pass. zacl 0, 1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e manual_src~dfsg/examples/oscils.csd0000644000000000000000000000103212262561504016546 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscils.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iflg = p4 asig oscils .7, 220, 0, iflg outs asig, asig endin i 1 0 2 0 i 1 3 2 2 ;double precision e manual_src~dfsg/examples/128-stringcircularX0000755000000000000000000000313312262561504020204 0ustar rootroot 0 1 0 127 1 0 1 2 2 1 2 3 3 2 3 4 4 3 4 5 5 4 5 6 6 5 6 7 7 6 7 8 8 7 8 9 9 8 9 10 10 9 10 11 11 10 11 12 12 11 12 13 13 12 13 14 14 13 14 15 15 14 15 16 16 15 16 17 17 16 17 18 18 17 18 19 19 18 19 20 20 19 20 21 21 20 21 22 22 21 22 23 23 22 23 24 24 23 24 25 25 24 25 26 26 25 26 27 27 26 27 28 28 27 28 29 29 28 29 30 30 29 30 31 31 30 31 32 32 31 32 33 33 32 33 34 34 33 34 35 35 34 35 36 36 35 36 37 37 36 37 38 38 37 38 39 39 38 39 40 40 39 40 41 41 40 41 42 42 41 42 43 43 42 43 44 44 43 44 45 45 44 45 46 46 45 46 47 47 46 47 48 48 47 48 49 49 48 49 50 50 49 50 51 51 50 51 52 52 51 52 53 53 52 53 54 54 53 54 55 55 54 55 56 56 55 56 57 57 56 57 58 58 57 58 59 59 58 59 60 60 59 60 61 61 60 61 62 62 61 62 63 63 62 63 64 64 63 64 65 65 64 65 66 66 65 66 67 67 66 67 68 68 67 68 69 69 68 69 70 70 69 70 71 71 70 71 72 72 71 72 73 73 72 73 74 74 73 74 75 75 74 75 76 76 75 76 77 77 76 77 78 78 77 78 79 79 78 79 80 80 79 80 81 81 80 81 82 82 81 82 83 83 82 83 84 84 83 84 85 85 84 85 86 86 85 86 87 87 86 87 88 88 87 88 89 89 88 89 90 90 89 90 91 91 90 91 92 92 91 92 93 93 92 93 94 94 93 94 95 95 94 95 96 96 95 96 97 97 96 97 98 98 97 98 99 99 98 99 100 100 99 100 101 101 100 101 102 102 101 102 103 103 102 103 104 104 103 104 105 105 104 105 106 106 105 106 107 107 106 107 108 108 107 108 109 109 108 109 110 110 109 110 111 111 110 111 112 112 111 112 113 113 112 113 114 114 113 114 115 115 114 115 116 116 115 116 117 117 116 117 118 118 117 118 119 119 118 119 120 120 119 120 121 121 120 121 122 122 121 122 123 123 122 123 124 124 123 124 125 125 124 125 126 126 125 126 127 127 0 127 126 manual_src~dfsg/examples/lposcil3.csd0000644000000000000000000000150112262561504017003 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lposcil3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 1.5 ; a fifth up kloop = 0 ; loop start time (in samples) kend line 45000, p3, 10000 ; vary loop end time (in samples) asig lposcil3 1, kcps, kloop, kend, 1 outs asig, asig endin ; Its table size is deferred, ; and format taken from the soundfile header. f 1 0 0 1 "beats.wav" 0 0 0 ; Play Instrument #1 for 6 seconds. ; This will loop the drum pattern several times. i 1 0 6 e manual_src~dfsg/examples/delay1.csd0000644000000000000000000000154512262561504016442 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o delay.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Instrument #1 -- Silence on one channel instr 1 ; Make a basic sound. abeep vco 20000, 440, 1 ; Delay the beep by 1 sample. idlt = 1/sr adel delay abeep, idlt adel1 delay1 abeep ; Send the beep to the left speaker and ; the difference in the delayes to the right speaker. outs abeep, adel-adel1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1. i 1 0.0 1 e manual_src~dfsg/examples/tablexseg.csd0000644000000000000000000000130712262561504017235 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tablexseg.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; analyze "fox.wav" with PVANAL first iend = p4 ktime line 0, p3, iend tablexseg p5, p3, p6 ;morph from table 1 asig vpvoc ktime, 1, "fox.pvx" ;to table 2 outs asig*3, asig*3 endin f 1 0 512 9 .5 1 0 f 2 0 512 5 1 60 0.01 390 0.01 62 1 i 1 0 5 2.7 1 2 e manual_src~dfsg/examples/partikkel_softsync.csd0000644000000000000000000001435512262561504021204 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out -odac ;;;RT audio ; For Non-realtime ouput leave only the line below: ; -o partikkel_softsync.wav -W ;;; for file output any platform sr = 44100 ksmps = 20 nchnls = 2 ; Example by Oeyvind Brandtsegg 2007, revised 2008 giSine ftgen 0, 0, 65537, 10, 1 giCosine ftgen 0, 0, 8193, 9, 1, 1, 90 giSigmoRise ftgen 0, 0, 8193, 19, 0.5, 1, 270, 1 ; rising sigmoid giSigmoFall ftgen 0, 0, 8193, 19, 0.5, 1, 90, 1 ; falling sigmoid ; ************************************************* ; example of soft synchronization of two partikkel instances ; ************************************************* instr 1 /*score parameters*/ igrainrate = p4 ; grain rate igrainsize = p5 ; grain size in ms igrainFreq = p6 ; fundamental frequency of source waveform iosc2Dev = p7 ; partikkel instance 2 grain rate deviation factor iMaxSync = p8 ; max soft sync amount (increasing to this value during length of note) /*overall envelope*/ iattack = 0.001 idecay = 0.2 isustain = 0.7 irelease = 0.2 amp linsegr 0, iattack, 1, idecay, isustain, 1, isustain, irelease, 0 kgrainfreq = igrainrate ; grains per second kdistribution = 0 ; periodic grain distribution idisttab = -1 ; (default) flat distribution used ; for grain distribution async = 0 ; no sync input kenv2amt = 0 ; no secondary enveloping ienv2tab = -1 ; default secondary envelope (flat) ienv_attack = giSigmoRise ; default attack envelope (flat) ienv_decay = giSigmoFall ; default decay envelope (flat) ksustain_amount = 0.3 ; time (in fraction of grain dur) at ; sustain level for each grain ka_d_ratio = 0.2 ; balance between attack and decay time kduration = igrainsize ; set grain duration in ms kamp = 0.2*0dbfs ; amp igainmasks = -1 ; (default) no gain masking kwavfreq = igrainFreq ; fundamental frequency of source waveform ksweepshape = 0 ; shape of frequency sweep (0=no sweep) iwavfreqstarttab = -1 ; default frequency sweep start ; (value in table = 1, which give ; no frequency modification) iwavfreqendtab = -1 ; default frequency sweep end ; (value in table = 1, which give ; no frequency modification) awavfm = 0 ; no FM input ifmamptab = -1 ; default FM scaling (=1) kfmenv = -1 ; default FM envelope (flat) icosine = giCosine ; cosine ftable kTrainCps = kgrainfreq ; set trainlet cps equal to grain ; rate for single-cycle trainlet in ; each grain knumpartials = 3 ; number of partials in trainlet kchroma = 1 ; balance of partials in trainlet ichannelmasks = -1 ; (default) no channel masking, ; all grains to output 1 krandommask = 0 ; no random grain masking kwaveform1 = giSine ; source waveforms kwaveform2 = giSine ; kwaveform3 = giSine ; kwaveform4 = giSine ; iwaveamptab = -1 ; mix of 4 source waveforms and ; trainlets (set to default) asamplepos1 = 0 ; phase offset for reading source waveform asamplepos2 = 0 ; asamplepos3 = 0 ; asamplepos4 = 0 ; kwavekey1 = 1 ; original key for source waveform kwavekey2 = 1 ; kwavekey3 = 1 ; kwavekey4 = 1 ; imax_grains = 100 ; max grains per k period iopcode_id = 1 ; id of opcode, linking partikkel ; to partikkelsync a1 partikkel kgrainfreq, kdistribution, idisttab, async, kenv2amt, \ ienv2tab,ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, \ kduration, kamp, igainmasks, kwavfreq, ksweepshape, \ iwavfreqstarttab, iwavfreqendtab, awavfm, ifmamptab, kfmenv, \ icosine, kTrainCps, knumpartials, kchroma, ichannelmasks, \ krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, asamplepos4, \ kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains, iopcode_id async1 partikkelsync iopcode_id ; clock pulse output of the ; partikkel instance above ksyncGravity line 0, p3, iMaxSync ; strength of synchronization aphase2 init 0 asyncPolarity limit (int(aphase2*2)*2)-1, -1, 1 ; use the phase of partikkelsync instance 2 to find sync ; polarity for partikkel instance 2. ; If the phase of instance 2 is less than 0.5, we want to ; nudge it down when synchronizing, ; and if the phase is > 0.5 we want to nudge it upwards. async1 = async1*ksyncGravity*asyncPolarity ; prepare sync signal ; with polarity and strength kgrainfreq2 = igrainrate * iosc2Dev ; grains per second for second partikkel instance iopcode_id2 = 2 a2 partikkel kgrainfreq2, kdistribution, idisttab, async1, kenv2amt, \ ienv2tab, ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, \ kduration, kamp, igainmasks, kwavfreq, ksweepshape, \ iwavfreqstarttab, iwavfreqendtab, awavfm, ifmamptab, kfmenv, \ icosine, kTrainCps, knumpartials, kchroma, ichannelmasks, \ krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, \ asamplepos4, kwavekey1, kwavekey2, kwavekey3, kwavekey4, \ imax_grains, iopcode_id2 async2, aphase2 partikkelsync iopcode_id2 ; clock pulse and phase ; output of the partikkel instance above, ; we will only use the phase outs a1*amp, a2*amp endin /*score parameters igrainrate = p4 ; grain rate igrainsize = p5 ; grain size in ms igrainFreq = p6 ; frequency of source wave within grain iosc2Dev = p7 ; partikkel instance 2 grain rate deviation factor iMaxSync = p8 ; max soft sync amount (increasing to this value during length of note) */ ; GrRate GrSize GrFund Osc2Dev MaxSync i1 0 10 2 20 880 1.3 0.3 s i1 0 10 5 20 440 0.8 0.3 s i1 0 6 55 15 660 1.8 0.45 s i1 0 6 110 10 440 0.6 0.6 s i1 0 6 220 3 660 2.6 0.45 s i1 0 6 220 3 660 2.1 0.45 s i1 0 6 440 3 660 0.8 0.22 s e e manual_src~dfsg/examples/array_udo.csd0000644000000000000000000000054012262561504017242 0ustar rootroot -nm128 opcode FirstEl, k, k[] ;returns the first element of vector kArr kArr[] xin xout kArr[0] endop instr 1 kArr[] fillarray 6, 3, 9, 5, 1 kFirst FirstEl kArr printf "kFirst = %d\n", 1, kFirst turnoff endin i 1 0 .1 manual_src~dfsg/examples/greaterequal.csd0000644000000000000000000000147112262561504017742 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o greaterequal.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Get the 4th p-field from the score. k1 = p4 ; Is it greater than or equal to 3? (1 = true, 0 = false) k2 = (p4 >= 3 ? 1 : 0) ; Print the values of k1 and k2. printks "k1 = %f, k2 = %f\\n", 1, k1, k2 endin ; Call Instrument #1 with a p4 = 2. i 1 0 0.5 2 ; Call Instrument #1 with a p4 = 3. i 1 1 0.5 3 ; Call Instrument #1 with a p4 = 4. i 1 2 0.5 4 e manual_src~dfsg/examples/vmult.csd0000644000000000000000000000150512262561504016426 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 kr=4410 ksmps=10 nchnls=2 instr 1 ;table playback ar lposcil 1, 1, 0, 262144, 1 outs ar,ar endin instr 2 vcopy 2, 1, 40000 ;copy vector from sample to empty table vmult 5, 10000, 262144 ;scale noise to make it audible vcopy 1, 5, 40000 ;put noise into sample turnoff endin instr 3 vcopy 1, 2, 40000 ;put original information back in turnoff endin f1 0 262144 -1 "beats.wav" 0 4 0 f2 0 262144 2 0 f5 0 262144 21 3 30000 i1 0 4 i2 3 1 s i1 0 4 i3 3 1 s i1 0 4 manual_src~dfsg/examples/farey7shuffled.csd0000644000000000000000000000433212262561504020176 0ustar rootroot -odac -+rtaudio=alsa --midioutfile=farey7.mid sr=48000 ksmps=10 nchnls=1 0dbfs = 1 gidelta init 100 gimult init 101 ;-------- loop and trigger instrument 901 using a Farey Sequence polyrhythm instr 1 kindx init 0 kindx2 init 0 ktrigger init 0 ktime_unit init p6 kstart init p4 kloop init p5 kinitndx init 0 kfn_times init gidelta knote init 60 kbasenote init p8 ifundam init p7 ktrigger seqtime ktime_unit, kstart, kloop, kinitndx, kfn_times if (ktrigger > 0 ) then kpitch = cpspch(ifundam) kmult tab kindx2, gimult kpitch = kpitch * kmult knote = kbasenote + kmult event "i", 901, 0, .4, .1, kpitch, kpitch * .9, .4, 5, .75, .8, 1.0, .15, .0, .125, .125, .25, .5, 1.0, .0, .0, .0, .0, .125, .25, .25, .25, knote kindx = kindx + 1 kindx = kindx % kloop kindx2 = kindx2 + 1 kindx2 = kindx2 % kloop if (kindx2 == 0) then tableshuffle gimult endif endif endin ; 1 ;------ basic 2 Operators FM algorithm ---------------- instr 901 inotedur = p3 imaxamp = p4 ;ampdb(p4) icarrfreq = p5 imodfreq = p6 ilowndx = p7 indxdiff = p8-p7 knote = p27 aampenv linseg p9, p14*p3, p10, p15*p3, p11, p16*p3, p12, p17*p3, p13 adevenv linseg p18, p23*p3, p19, p24*p3, p20, p25*p3, p21, p26*p3, p22 amodosc oscili (ilowndx+indxdiff*adevenv)*imodfreq, imodfreq, 10 acarosc oscili imaxamp*aampenv, icarrfreq+amodosc, 10 out acarosc ;------ we also write down a midi track here ---------- midion 1, knote, 100 endin ; 901 f10 0 4096 10 1 f100 0 -18 "farey" 7 1 f101 0 -18 "farey" 7 2 ; p4 kstart := index offset into the Farey Sequence ; p5 kloop := end index into Farey Seq. ; p6 timefac := time in seconds for one loop to complete ; p7 fundam := fundamental of the FM instrument ; p8 basenote:= root pitch of the midi voice output ; note that pitch structures of the midi file output are not equivalent to the ; ones used for the FM real-time synthesis. ; start dur kstart kloop timefac fundam. basenote i1 0.0 44 0 18 1 6.05 60 i1 4 40 0 18 3 7.05 72 i1 10 38 0 18 1.5 8 84 i1 15 50 0 18 1 5 48 i1 22 75 5 17 1.7 4 36 e manual_src~dfsg/examples/ifthen.csd0000644000000000000000000000206412262561504016535 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ifthen.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Get the note value from the fourth p-field. knote = p4 ; Does the user want a low note? if (knote == 0) then kcps = 220 ; Does the user want a middle note? elseif (knote == 1) then kcps = 440 ; Does the user want a high note? elseif (knote == 2) then kcps = 880 endif ; Create the note. kamp init 25000 ifn = 1 a1 oscili kamp, kcps, ifn out a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; p4: 0=low note, 1=middle note, 2=high note. ; Play Instrument #1 for one second, low note. i 1 0 1 0 ; Play Instrument #1 for one second, middle note. i 1 1 1 1 ; Play Instrument #1 for one second, high note. i 1 2 1 2 e manual_src~dfsg/examples/Church.wav0000644000000000000000000002334012262561504016520 0ustar rootrootRIFF&WAVEfmt DXsmpl<X</datap&I1!_(v.m/\Z Cp{a &+kGRb@(~LLiB#[G+.3D 6Gh-^MPOwky0C8e)J}1@5r,6ge OHK5xC4 +FxJ q*.u#Uh{.f\$"Tb*  x5zl0=-}d, ) ! 8XDrSk_ u b HLJ~]=@$aBOvKnyZ&"-Fi Cpr\X/5r MRX < % "-q?"OHeF@e1 )r7   :;T4h(&>0tl[q } k jL$C 0> qRc/_O, T"Q0 iqys='*\#fg7tpq1yz' j %N aV\~"B2W7paH)&&'! w,0l5yB Gs: dOxe  ,G/y|+j }v,`O %=:۪ڡܹ(>m>y#i&?!L}J gB; BO=~x` X6  C $ >iV!#%$;9_1#)6^dWE !~ f)\ \,  lhw DuLvv f TZ@9p v` i$ w/ n Hej .bH bI}/p:uGp1u    yr\}C | q  0d**; vJ!&H$sR* E?+\ q 2l^ۄ @*. 6T %d; J>h PFLQ s05ҜA@% CP RXa _ E Nawk!,GE58CsA1Li+) d #hTW!SyݯD v)/(/$_% +M474[/%: (-p+$z7~} Dt,[ . 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cq(d>}se/+VT՜W}qڥ5  &/226 :94*%4&(*&y##!'0;@>|<4z)L~S<[2 PLa1 ǟƲyȲȼρ̵ƿ]LʹLHԌ㱃|Zߝ\r[ wMZ-q?ORUOFJ@Q==<;?YDhJL^KI7C:7y& kh 5E;{GHJێw+ϟJȅD̫ղFEռ˶d <[È9НeМM35{]r5sIH o0BuMqLEI?(< =o=<@FMXR*RQLA1# !?b 8h$E'['$#l"\4;!iw/S+F܋6Ag͕$f/23:h@?6)%*X0s0,'h! %"+6+?JkW1eic]ZYpXbZcYXv\`8g3jjhVaXT$AS, x (%,.l+&z%G#7% Aّ#ץACt²Ȼ{oC~,l<𭃿"3s aWBtBz.<@]=~8435520 3u6;|=i>2?~8j+\Z: &^ O=_$H0:+1@6ݑ3ˈ!XţX9ij%Xl 3hU+  WW<)6CBUEC@>4?yBhA}?s@CJMNYPxJ<)R H B a a lBۄfݼӶlɱ*hԀvث]ώcıӵP[)ƚ`?ؠҧ2j Bd [S )}$ n!-:N?t>>M?@#CAB??]A~GKtNPXL@,Y ?  pB <V!IUl;l K@)Y9+ ׯ ӼQ?. ܥ8J:%]?,2Y7A0HtH?1+-|36,n#&2=HQZT]Z_YZ@\`^[O[:\`cfni&d(XD/#L#%k#/!"!!!0oQm۠Ҩoݡ/֔͑Ƽ%$N_ėŲ=S|Uŭ[ٺ U 2 'o)J354h579A<9'5314.5;697=.?-g2'38I 5 `"$YKٷەVMȞDдɰPž4oĞɍɶ粮dԓr W k$@d v[>j(2;<9F:3=>@D!EBIBADjFGJIA0tK~T <#kɥmqG*D˛̠2eѼ$3kJ`ݜKQhʋZ8)lNXn5$/9$=;7<??BaF)FCCBAD+DDGG@1pYp3J qu@{YEWYo:9}#ޝQC׶=13ro`*3>TFIVDf7-S*+H.)#B#0?LV_q`]]^`Vcla!^\E[[NZZ\ZR@Q+ %~'z"~{s#R$$:%#*K7*ۯߪmݦ̱˂ʆ 0&FfY#MmÇ8ھ(ty6 rO%_X",5+6446U9<9?7*655S4a573*rYgj_rZ ѿtǧyayչP3ogͥEqչj"a#% 80 u6<%2 c1c |ewj38%<ƶZ;9eϺX՞/'xGW[  /eHWH+6L<;Z;=@DF,DCCCjD>D%GGqC7#fc&/'On[i׎U,h_ZUk#q4manual_src~dfsg/examples/fluidSetInterpMethod.csd0000644000000000000000000000203312262561504021356 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidSetInterpMethod.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "07AcousticGuitar.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 0 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 iInterpMethod = p6 fluidSetInterpMethod giengine, 1, iInterpMethod fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 7 asigl, asigr fluidOut giengine outs asigl*imvol, asigr*imvol endin ;hear the difference i 1 0 2 60 120 0 ;no interpolation i 1 3 2 72 120 0 i 1 6 2 60 120 7 ;7th order interpolation i 1 9 2 72 120 7 i 99 0 12 e manual_src~dfsg/examples/lpshold.csd0000644000000000000000000000240012262561504016717 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpshold.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfrq init p4 ; frequency of the loop ifrac = p5 ; fraction of frequency at which to force retrigger ipan = p6 ; pan position ktrig metro kfrq * ifrac ; trigger to force restart the loop iphs = 0 ; initial phase of the loop ; a loop of midi note numbers: note duration etc... knote lpshold kfrq, ktrig, iphs, 61, 0.0625, 60, 0.9375, 61, 1, 58, 1, \ 63, 2, 65, 3 aenv linseg 0,0.01,1,p3-0.11,1,0.1,0 ; amplitude envelope krnd rspline -0.05,0.05,0.5,1 ; random detune asig gbuzz 0.2*aenv, cpsmidinn(knote+krnd), 30, 1, 0.5, 1 outs asig*ipan, asig*(1-ipan) endin ; cosine wave. f 1 0 16384 11 1 ; 3 layers of the loop are played, each at a different speed, ; - with different retriggering rate, and pan location. i 1 0 60 0.5 [8/10] 0.5 i 1 0 60 0.375 [8/11] 0.1 i 1 0 60 0.25 [8/13] 0.9 e manual_src~dfsg/examples/readk-2.csd0000644000000000000000000000156612262561504016513 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o readk-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes a control signal to a file kfreq randh 100, 1, 2, 1, 500 ;generates one random number between 400 and 600 per second dumpk kfreq, "dumpk.txt", 8, 1 ;writes the control signal printk 1, kfreq ;prints it endin instr 2 ;reads the file written by instr 1 kfreq readk "dumpk.txt", 8, 1 printk 1, kfreq ;prints it aout poscil .2, kfreq, giSine outs aout, aout endin i 1 0 5 i 2 5 5 e manual_src~dfsg/examples/dam_expanded.csd0000644000000000000000000000137112262561504017671 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dam_expanded.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig diskin2 "beats.wav", 1, 0, 1 outs asig, asig endin instr 2 ;expanded audio kthreshold = .5 icomp1 = 2 icomp2 = 3 irtime = 0.01 iftime = 0.1 asig diskin2 "beats.wav", 1, 0, 1 asig dam asig, kthreshold, icomp1, icomp2, irtime, iftime outs asig*.2, asig*.2 ;adjust volume of expanded beat endin i 1 0 2 i 2 2.5 6.5 e manual_src~dfsg/examples/resyn.csd0000644000000000000000000000140212262561504016413 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o resyn.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktracks = p4 ain diskin2 "fox.wav", 1, 0, 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; pvsifd analysis fst partials fs1, fsi2, .1, 1,3, 500 ; partial tracking aout resyn fst, 1, 1.5, ktracks, 1 ; resynthesis (up a 5th) outs aout, aout endin ;sine f1 0 4096 10 1 i 1 0 2.7 500 i 1 3 2.7 10 ;non-linear filtering effect e manual_src~dfsg/examples/binit.csd0000644000000000000000000000133012262561504016360 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o binit.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 ;ain inch 1 ; for live input ain diskin "beats.wav", 1 ; input signal fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .003, 1, 3, 500 ; partial tracking fbins binit fst, 2048 ; convert it back to bins aout pvsynth fbins ; overlap-add resynthesis outs aout, aout endin i 1 0 2 e manual_src~dfsg/examples/loop_le.csd0000644000000000000000000000162612262561504016714 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loop_le.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 gisine ftgen 0, 0, 2^10, 10, 1 instr 1 ;master instrument ininstr = 5 ;number of called instances indx = 0 loop: prints "play instance %d\\n", indx ipan random 0, 1 ifreq random 100, 1000 iamp = 1/ininstr event_i "i", 10, 0, p3, iamp, ifreq, ipan loop_le indx, 1, ininstr, loop endin instr 10 ipeak random 0, 1 ;where is the envelope peak asig poscil3 p4, p5, gisine aenv transeg 0, p3*ipeak, 6, 1, p3-p3*ipeak, -6, 0 aL,aR pan2 asig*aenv, p6 outs aL, aR endin i1 0 10 e manual_src~dfsg/examples/midiprogramchange.csd0000644000000000000000000000164112262561504020740 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M1 -Q1 ;;;realtime audio out and midi in and midi out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midiprogramchange.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;uses external midi device kkey init 0 ;initialize key number kvel init 0 ;initialize velocity midiprogramchange 52 ;choose GM program number 52 = string pad midinoteonkey kkey, kvel ;MIDI note number value printk2 kkey ;display the key value when it changes and when key is pressed midion 1, kkey, kvel ;sent note to external device endin f 0 30 ;runs for 30 seconds i 1 0 2 62 ; play these notes from score as well i 1 + 2 65 e manual_src~dfsg/examples/abs.csd0000644000000000000000000000074512262561504016031 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o abs.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 ix = p4 iabs = abs(ix) print iabs endin i 1 0 1 0 i 1 + 1 -.15 i 1 + 1 -13 e manual_src~dfsg/examples/ftlptim.csd0000644000000000000000000000111012262561504016726 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftlptim.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 instr 1 itim = ftlptim(1) print itim aout loscil3 .8, 40, 1 outs aout, aout endin f 1 0 0 1 "Church.wav" 0 0 0 ;Csound computes tablesize i 1 0 5 e manual_src~dfsg/examples/midichannelaftertouch.csd0000644000000000000000000000220412262561504021614 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midichannelaftertouch.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 midinoteoncps p4, p5 ;puts MIDI key translated to cycles per second into p4, and MIDI velocity into p5 kafter init 127 ;full open midichannelaftertouch kafter printk2 kafter ;display the key value when it changes and when key is pressed kvel = p5/127 ;scale midi velocity to 0-1 kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by ain pluck kenv*kvel, p4, p4, 2, 1 ;velocity value asig moogvcf2 ain, 200+(kafter*40), .5 ;scale value of aftertouch and control filter frequency outs asig, asig ;base freq of filter = 200 Hz endin f 0 30 ;runs 30 seconds f 2 0 4096 10 1 i 1 0 2 440 100 ; play these notes from score as well i 1 + 2 1440 100 e manual_src~dfsg/examples/printk.csd0000644000000000000000000000123412262561504016565 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o printk.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Change a value linearly from 0 to 100, ; over the period defined by p3. kval line 0, p3, 100 ; Print the value of kval, once per second. printk 1, kval endin ; Play Instrument #1 for 5 seconds. i 1 0 5 e manual_src~dfsg/examples/outs1.csd0000644000000000000000000000123612262561504016333 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outs1.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig vco2 .01, 110 ; sawtooth waveform at low volume kcut line 60, p3, 300 ; Vary cutoff frequency kresonance = 3 inumlayer = 3 asig lowresx asig, kcut, kresonance, inumlayer outs1 asig ; output stereo channel 1 only endin i 1 0 3 e manual_src~dfsg/examples/samphold.csd0000644000000000000000000000132312262561504017064 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o samphold.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kx line -1, p3, 1 ; between -1 and +1 ktrig metro 1 ; triggers 1 time per second kval samphold kx, ktrig ; change value whenever ktrig = 1 printk2 kval ; will print every time kval changes asig diskin2 "flute.aiff", 1+kval, 0, 1 outs asig, asig endin i 1 0 11 e manual_src~dfsg/examples/FLscroll.csd0000644000000000000000000000166312262561504017004 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLscroll.wav -W ;;; for file output any platform ; Demonstration of the flscroll opcode which enables ; the use of widget sizes and placings beyond the ; dimensions of the containing panel sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Text Box", 420, 200, 50, 50 iwidth = 420 iheight = 200 ix = 0 iy = 0 FLscroll iwidth, iheight, ix, iy ih3 FLbox "DRAG THE SCROLL BAR TO THE RIGHT IN ORDER TO READ THE REST OF THIS TEXT!", 1, 10, 20, 870, 30, 10, 100 FLscrollEnd ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 endin ; 'Dummy' score event of 1 hour. f 0 3600 e manual_src~dfsg/examples/pvsarp2.csd0000644000000000000000000000206612262561504016657 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsarp2.wav -W ;;; for file output any platform sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "fox.wav" ain1 soundin Sfile1 fftin pvsanal ain1, ifftsize, ioverlap, iwinsize, iwinshape ;make 3 independently moving accentuations in the spectrum kbin1 linseg 0.05, p3/2, .05, p3/2, .05 farp1 pvsarp fftin, kbin1, .9, 10 kbin2 linseg 0.075, p3/2, .1, p3/2, .075 farp2 pvsarp fftin, kbin2, .9, 10 kbin3 linseg 0.02, p3/2, .03, p3/2, .04 farp3 pvsarp fftin, kbin3, .9, 10 ;resynthesize and add them aout1 pvsynth farp1 aout2 pvsynth farp2 aout3 pvsynth farp3 aout = aout1*.3 + aout2*.3 + aout3*.3 out aout endin i 1 0 3 e manual_src~dfsg/examples/int.csd0000644000000000000000000000110212262561504016042 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o int.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icount init 0 loop: inum = icount / 3 inm = int(inum) prints "integer (%f/3) = %f\\n", icount, inm loop_lt icount, 1, 10, loop endin i 1 0 0 e manual_src~dfsg/examples/diff.csd0000644000000000000000000000151112262561504016164 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o diff.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig diskin2 "fox.wav", 1 outs asig, asig endin instr 2 ; with diff asig diskin2 "fox.wav", 1 ares diff asig outs ares, ares endin instr 3 ; with integ asig diskin2 "fox.wav", 1 aint integ asig aint = aint*.05 ;way too loud outs aint, aint endin instr 4 ; with diff and integ asig diskin2 "fox.wav", 1 ares diff asig aint integ ares outs aint, aint endin i 1 0 1 i 2 1 1 i 3 2 1 i 4 3 1 e manual_src~dfsg/examples/exitnow.csd0000644000000000000000000000257412262561504016763 0ustar rootroot ; Select audio/midi flags here according to platform -n ;;;no sound ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o exitnow.wav -W ;;; for file output any platform ;after an example by Iain McCurdy sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 FLcolor 200, 200, 200, 0, 0, 0 ; LABEL | WIDTH | HEIGHT | X | Y FLpanel "rtclock", 500, 130, 0, 0 ; ON,OFF,TYPE,WIDTH, HEIGHT, X, Y, OPCODE, INS,START,IDUR gkOnOff,ihOnOff FLbutton "On/Off", 1, 0, 22, 150, 25, 5, 5, 0, 1, 0, 3600 gkExit,ihExit FLbutton "exitnow",1, 0, 21, 150, 25, 345, 5, 0, 999, 0, 0.001 FLsetColor2 255, 0, 50, ihOnOff ;reddish color ;VALUE DISPLAY BOXES WIDTH,HEIGHT,X, Y gidclock FLvalue "clock", 100, 25, 200, 60 FLsetVal_i 1, ihOnOff FLpanel_end FLrun instr 1 if gkOnOff !=0 kgoto CONTINUE ;sense if FLTK on/off switch is not off (in which case skip the next line) turnoff ;turn this instr. off now CONTINUE: ktime rtclock ;clock continues to run even FLprintk2 ktime, gidclock ;after the on/off button was used to stop endin instr 999 exitnow ;exit Csound as fast as possible endin f 0 60 ;runs 60 seconds e manual_src~dfsg/examples/cosh.csd0000644000000000000000000000103712262561504016213 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cosh.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 1 i1 = cosh(irad) print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/rv_mono.wav0000644000000000000000000010356212262561504016770 0ustar rootrootRIFFjWAVEfmt DXLISTINFOIARTmennodata,By5!YT@l[gG$>  U!Q##!*@ sZlyFٶ֢Vbۃ\ $ \!\"7$')*s++j*&g!C )8n [i1 e 7 )-&e r%"+{.701R0-(|"Fkq _R," F $o<=T9ج8}n ;B, F H  $ Y O0 ) ~a R V!#"6h/س؍iޡsAJB+xlVPoh<b}yi IBQb P 1 #G<{ *2[ d$t܊ 0ԍFhէG^.E M!3&+T.E.,+*s'n"I? 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Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o /.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idiv = 1/p3 * p4 ktrm oscil 1, idiv, 1 ;use oscil as an envelope printf "retrigger rate per note duration = %f\n",1, idiv kndx line 5, p3, 1 ;vary index of FM asig foscil ktrm, 200, 1, 1.4, kndx, 1 outs asig, asig endin f 1 0 4096 10 1 ;sine wave i 1 0 3 10 i 1 4 3 15 i 1 8 3 2 e manual_src~dfsg/examples/barmodel.csd0000644000000000000000000000137712262561504017053 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o barmodel.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 aq barmodel 1, 1, p4, 0.001, 0.23, 5, p5, p6, p7 out aq endin i1 0.0 0.5 3 0.2 500 0.05 i1 0.5 0.5 -3 0.3 1000 0.05 i1 1.0 0.5 -3 0.4 1000 0.1 i1 1.5 4.0 -3 0.5 800 0.05 e /* barmodel */ manual_src~dfsg/examples/vadd.csd0000644000000000000000000000146212262561504016177 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vadd ifn1, ival, ielements, idstoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 5 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 8 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1 10 12 i2 1.6 0.2 1 e manual_src~dfsg/examples/FLsetText.csd0000644000000000000000000000172212262561504017142 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLsetText.wav -W ;;; for file output any platform sr = 44100 ksmps = 128 nchnls = 2 ; Example by Giorgio Zucco and Andres Cabrera 2007 FLpanel "FLsetText",250,100,50,50 gk1,giha FLcount "", 1, 20, 1, 20, 1, 200, 40, 20, 20, 0, 1, 0, 1 FLpanelEnd FLrun instr 1 ; This instrument is triggered by FLcount above each time ; its value changes iname = i(gk1) print iname ; Must use FLsetText on the init pass! if (iname == 1) igoto text1 if (iname == 2) igoto text2 if (iname == 3) igoto text3 igoto end text1: FLsetText "FM",giha igoto end text2: FLsetText "GRANUL",giha igoto end text3: FLsetText "PLUCK",giha igoto end end: endin f 0 3600 manual_src~dfsg/examples/STKShakers.csd0000644000000000000000000000112712262561504017241 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKShakers.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 asig STKShakers cpspch(p4), 1, 2, 10, 4, 10, 11, 10, 1, 112, 128, 80, 1071, 5 asig = asig *2 ;amplify outs asig, asig endin i 1 0.2 .5 7.00 75 0 20 e manual_src~dfsg/examples/print.csd0000644000000000000000000000115512262561504016414 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o print.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 irand = rnd(3) ;generate a random number from 0 to 3 print irand ;print it asig poscil .7, 440*irand, 1 outs asig, asig endin f1 0 16384 10 1 ;sine wave i 1 0 1 i 1 2 1 i 1 4 1 e manual_src~dfsg/examples/fmpercfl.csd0000644000000000000000000000117412262561504017057 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmpercfl.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 220 kc1 = 5 kvdepth = .01 kvrate = 6 kc2 line 5, p3, p4 asig fmpercfl .5, kfreq, kc1, kc2, kvdepth, kvrate outs asig, asig endin ; sine wave. f 1 0 32768 10 1 i 1 0 4 5 i 1 5 8 .1 e manual_src~dfsg/examples/fofilter.csd0000644000000000000000000000121212262561504017064 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fofilter.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfe expseg 10, p3*0.9, 180, p3*0.1, 175 kenv linen .1, 0.05, p3, 0.05 asig buzz kenv, kfe, sr/(2*kfe), 1 afil fofilter asig, 900, 0.007, 0.04 outs afil, afil endin ; sine wave f 1 0 16384 10 1 i 1 0 10 e manual_src~dfsg/examples/reinit.csd0000644000000000000000000000113712262561504016552 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o reinit.wav -W ;;; for file output any platform sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 instr 1 reset: timout 0, p3/10, contin reinit reset contin: kLine expon 440, p3/10, 880 aSig oscil 10000, kLine, 1 out aSig rireturn endin f1 0 4096 10 1 i1 0 10 e manual_src~dfsg/examples/limit.csd0000644000000000000000000000204412262561504016374 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o limit.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; Limit / Mirror / Wrap igain = p4 ;gain ilevl1 = p5 ; + level ilevl2 = p6 ; - level imode = p7 ;1 = limit, 2 = mirror, 3 = wrap ain soundin "fox.wav" ain = ain*igain if imode = 1 goto limit if imode = 2 goto mirror asig wrap ain, ilevl2, ilevl1 goto outsignal limit: asig limit ain, ilevl2, ilevl1 goto outsignal mirror: asig mirror ain, ilevl2, ilevl1 outsignal: outs asig*.5, asig*.5 ;mind your speakers endin ; Gain +Levl -Levl Mode i1 0 3 4.00 .25 -1.00 1 ;limit i1 4 3 4.00 .25 -1.00 2 ;mirror i1 8 3 4.00 .25 -1.00 3 ;wrap e manual_src~dfsg/examples/dssiactivate.csd0000644000000000000000000000214212262561504017740 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dssiactivate.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 gihandle dssiinit "caps.so", 19, 1 ; = mono phaser and gaout init 0 ; verbose about all ports instr 1 ; activate DSSI ktoggle = p4 dssiactivate gihandle, ktoggle endin instr 2 ain1 diskin2 "beats.wav", 1,0,1 ; loop ain1 = ain1*.5 outs ain1, ain1 gaout = gaout+ain1 endin instr 3 dssictls gihandle, 0, 1, 1 ; range -1 to 1 dssictls gihandle, 1, 2, 1 ; rate 0 to 10 dssictls gihandle, 2, .8, 1 ; depth 0 to 1 dssictls gihandle, 3, 3, 1 ; spread 0 to 3.14 dssictls gihandle, 4, .9, 1 ; feedback 0 to 0.999 endin instr 4 aout1 dssiaudio gihandle, gaout ;get beats.wav, mono out outs aout1,aout1 gaout = 0 endin i 1 0 4 1 i 1 + . 0 i 1 + . 1 i 1 + . 0 i 1 + . 1 i 2 1 20 i 3 1 20 i 4 0 20 e manual_src~dfsg/examples/fiopen.csd0000644000000000000000000000107212262561504016536 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fiopen.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gihand fiopen "test1.txt", 0 instr 1 ires random 0, 100 fouti gihand, 0, 1, ires ficlose gihand endin i 1 0 1 e manual_src~dfsg/examples/setctrl.csd0000644000000000000000000000162612262561504016743 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o setctrl.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Display the label "Volume" on Slider #1. setctrl 1, "Volume", 4 ; Set Slider #1's initial value to 20. setctrl 1, 20, 1 ; Capture and display the values for Slider #1. k1 control 1 printk2 k1 ; Play a simple oscillator. ; Use the values from Slider #1 for amplitude. kamp = k1 * 128 a1 oscil kamp, 440, 1 out a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for thirty seconds. i 1 0 30 e manual_src~dfsg/examples/pvsmix.csd0000644000000000000000000000134212262561504016604 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsmix.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 1, 0, 2048, 10, 1 instr 1 asig1 diskin2 "fox.wav", 1 ;signal in 1 asig2 oscil .3, 100, gisin ;signal in 2 fsig1 pvsanal asig1,1024,256,1024,0 ;pvoc analysis fsig2 pvsanal asig2,1024,256,1024,0 ;of both signals fsall pvsmix fsig1, fsig2 asig pvsynth fsall outs asig, asig endin i 1 0 3 e manual_src~dfsg/examples/strsub.csd0000644000000000000000000000132512262561504016601 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc ;;;-d RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o strsub.wav -W ;;; for file output any platform ; By: Jonathan Murphy 2007 instr 1 Smember strget p4 ; Parse Smember istrlen strlen Smember idelimiter strindex Smember, ":" S1 strsub Smember, 0, idelimiter ; "String1" S2 strsub Smember, idelimiter + 1, istrlen ; "String2" printf "First string: %s\nSecond string: %s\n", 1, S1, S2 endin i 1 0 1 "String1:String2" manual_src~dfsg/examples/taninv2.csd0000644000000000000000000000107312262561504016640 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o taninv2.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Returns the arctangent for 1/2. i1 taninv2 1, 2 print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/active_scale.csd0000644000000000000000000000120212262561504017673 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o atone.wav -W ;;; for file output any platform sr= 44100 ksmps = 64 nchnls = 1 0dbfs = 1 ;by Victor Lazzarini 2008 instr 1 kscal active 1 kamp port 1/kscal, 0.01 asig oscili kamp, p4, 1 kenv linseg 0, 0.1,1,p3-0.2,1,0.1, 0 out asig*kenv endin f1 0 16384 10 1 i1 0 10 440 i1 1 3 220 i1 2 5 350 i1 4 3 700 e manual_src~dfsg/examples/genarray_i.csd0000644000000000000000000000144312262561504017400 0ustar rootroot ; Select audio/midi flags here according to platform -n sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;create and fill two arrays kArr1[] genarray_i 1, 6 kArr2[] genarray_i 1, 6, 2/3 ;print the content of kArr1 printf "%s", 1, "kArr1: start=1, end=6, step=default\n" kndx = 0 until kndx == lenarray(kArr1) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr1[kndx] kndx += 1 od ;print the content of kArr2 printf "%s", 1, "kArr2: start=1, end=6, step=2/3\n" kndx = 0 until kndx == lenarray(kArr2) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr2[kndx] kndx += 1 od turnoff endin i 1 0 1 manual_src~dfsg/examples/oscilikts.csd0000644000000000000000000000155512262561504017270 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o oscilikts.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1: oscilikts example. instr 1 ; Frequency envelope. kfrq expon 400, p3, 1200 ; Phase. kphs line 0.1, p3, 0.9 ; Sync 1 atmp1 phasor 100 ; Sync 2 atmp2 phasor 150 async diff 1 - (atmp1 + atmp2) a1 oscilikts 14000, kfrq, 1, async, 0 a2 oscilikts 14000, kfrq, 1, async, -kphs out a1 - a2 endin ; Table #1: Sawtooth wave f 1 0 3 -2 1 0 -1 ; Play Instrument #1 for four seconds. i 1 0 4 e manual_src~dfsg/examples/dssilist.csd0000644000000000000000000000045612262561504017121 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out -odac sr = 44100 ksmps = 32 nchnls = 2 instr 1 ; list all DSSI and LADSPA plugins dssilist endin i 1 0 0 e manual_src~dfsg/examples/max.csd0000644000000000000000000000146012262561504016044 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o max.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 k1 oscili 1, 10.0, 1 ;combine 3 sinusses k2 oscili 1, 1.0, 1 ;at different rates k3 oscili 1, 3.0, 1 kmax max k1, k2, k3 kmax = kmax*250 ;scale kmax printk2 kmax ;check the values aout vco2 .5, 220, 6 ;sawtooth asig moogvcf2 aout, 600+kmax, .5 ;change filter around 600 Hz outs asig, asig endin f1 0 32768 10 1 i1 0 5 e manual_src~dfsg/examples/imageopcodes.csd0000644000000000000000000000156512262561504017724 0ustar rootroot ; Select audio/midi flags here according to platform -n ;no sound output sr=48000 ksmps=1 nchnls=2 ; this test .csd copies image.png into a new file 'imageout.png' giimage1 imageload "image.png" giimagew, giimageh imagesize giimage1 giimage2 imagecreate giimagew,giimageh instr 1 kndx = 0 kx_ linseg 0, p3, 1 ;print imagewidth and imageheigth of image.png prints "imagewidth = %f pixels, imageheigth = %f pixels\\n", giimagew, giimageh myloop: ky_ = kndx/(giimageh) kr_, kg_, kb_ imagegetpixel giimage1, kx_, ky_ imagesetpixel giimage2, kx_, ky_, kr_, kg_, kb_ loop_lt kndx, 0.5, giimageh, myloop endin instr 2 imagesave giimage2, "imageout.png" endin instr 3 imagefree giimage1 imagefree giimage2 endin i1 1 1 i2 2 1 i3 3 1 e manual_src~dfsg/examples/s.csd0000644000000000000000000000125412262561504015522 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o s.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .8*aenv, p4, 1 outs asig, asig endin f 1 0 16384 10 1;sine wave s ;first section i 1 0 2 110 i 1 3 2 220 i 1 6 2 440 i 1 9 2 880 s ;second section i 1 0 2 880 i 1 3 2 440 i 1 6 2 220 i 1 9 2 110 e manual_src~dfsg/examples/deltap3.csd0000644000000000000000000000216612262561504016617 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o deltap3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasig init 0 gidel = 1 ;delay time in seconds instr 1 ain pluck .7, 440, 1000, 0, 1 outs ain, ain vincr gasig, ain ;send to global delay endin instr 2 ifeedback = p4 abuf2 delayr gidel adelL deltap3 .4 ;first tap (on left channel) adelM deltap3 1 ;second tap (on middle channel) delayw gasig + (adelL * ifeedback) abuf3 delayr gidel kdel line 1, p3, .01 ;vary delay time adelR deltap3 .65 * kdel ;one pitch changing tap (on the right chn.) delayw gasig + (adelR * ifeedback) ;make a mix of all deayed signals outs adelL + adelM, adelR + adelM clear gasig endin i 1 0 1 i 1 3 1 i 2 0 3 0 ;no feedback i 2 3 8 .8 ;lots of feedback e manual_src~dfsg/examples/harmon3.csd0000644000000000000000000000132312262561504016624 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o harmon3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ilow = p4 ;lowest value to harmonize aout diskin2 "fox.wav", 1, 0, 1 koct, kamp pitch aout, .01, 6, 10, 10 ;track pitch asig harmon3 aout, koct, .9, 1.5, 0.7, 0, ilow outs (asig + aout)*.4, (asig + aout)*.4 ;mix dry&wet signal endin i1 0 2.7 8.8 i1 3 2.7 8.2 i1 6 2.7 7.0 e manual_src~dfsg/examples/diskin.csd0000644000000000000000000000103412262561504016535 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o diskin.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktrans linseg 1, 5, 2, 10, -2 a1 diskin "beats.wav", ktrans, 0, 1, 0, 32 outs a1, a1 endin i 1 0 15 e manual_src~dfsg/examples/vst4cs.csd0000644000000000000000000000441312262561504016506 0ustar rootroot ; Credits: Adapted by Michael Gogins ; from code by David Horowitz and Lian Cheung. ; The "--displays" option is required in order for ; the Pianoteq GUI to dispatch events and display properly. -m3 --displays -odac sr = 44100 ksmps = 20 nchnls = 2 ; Load the Pianoteq into memory. gipianoteq vstinit "C:\\Program Files\\Steinberg\\VstPlugins\\Pianoteq 3.0 Trial\\Pianoteq30 Trial.dll", 1 ; Print information about the Pianoteq, such as parameter names and numbers. vstinfo gipianoteq ; Open the Pianoteq's GUI. vstedit gipianoteq ; Send notes from the score to the Pianoteq. instr 1 ; MIDI channels are numbered starting at 0. ; p3 always contains the duration of the note. ; p4 contains the MIDI key number (pitch), ; p5 contains the MIDI velocity number (loudness), imidichannel init 0 vstnote gipianoteq, imidichannel, p4, p5, p3 endin ; Send parameter changes to the Pianoteq. instr 2 ; p4 is the parameter number. ; p5 is the parameter value. vstparamset gipianoteq, p4, p5 endin ; Send audio from the Pianoteq to the output. instr 3 ablankinput init 0 aleft, aright vstaudio gipianoteq, ablankinput, ablankinput outs aleft, aright endin ; Turn on the instrument that receives audio from the Pianoteq indefinitely. i 3 0 -1 ; Send parameter changes to Pianoteq before sending any notes. ; NOTE: All parameters must be between 0.0 and 1.0. ; Length of piano strings: i 2 0 1 33 0.5 ; Hammer noise: i 2 0 1 25 0.1 ; Send a C major 7th arpeggio to the Pianoteq. i 1 1 10 60 76 i 1 2 10 64 73 i 1 3 10 67 70 i 1 4 10 71 67 ; End the performance, leaving some time ; for the Pianoteq to finish sending out its audio, ; or for the user to play with the Pianoteq virtual keyboard. e 20 manual_src~dfsg/examples/FLslidBnk2Setk.csd0000644000000000000000000000400412262561504017775 0ustar rootroot ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O sr = 44100 ksmps = 10 nchnls = 2 ;Example by Gabriel Maldonado 2007 giElem init 8 giOutTab ftgen 1,0,128, 2, 0 giSine ftgen 3,0,256,10, 1 giOutTab2 ftgen 4,0,128, 2, 0 itab ftgen 29, 0, 129, 5, .002, 128, 1 ;** exponential ascending curve for slider mapping giExpTab ftgen 30, 0, 129, -24, itab, 0, 1 ;** rescaled curve for slider mapping giConfigTab ftgen 2,0,128,-2, 1, 500, -1, 13, \ 1, 500, -1, 13, \ 1, 500, -1, 13, \ 1, 5000, -1, 13, \ \ 1, 1000, -1, 5, \ 1, 1000, -1, 5, \ 1, 1000, -1, 5, \ 1, 5000, -1, 5 FLpanel "Multiple FM",600,600 FLslidBnk2 "mod1@mod2@mod3@amp@freq1@freq2@freq3@freqPo", giElem, giOutTab2, giConfigTab, 400, 500, 100, 10 giHandle FLslidBnkGetHandle FLpanel_end FLrun instr 1 ktrig slider8table 1, giOutTab, 0, \ ; ctl min max init func 27, 1, 500, 3, -1, \ ;1 repeat rate 28, 1, 500, 4, -1, \ ;2 random freq. amount 29, 1, 500, 1, -1, \ ;3 random amp. amount 30, 1, 5000, 1, -1, \ ;4 number of concurrent loop points \ 31, 1, 1000, 1, -1, \;5 kloop1 32, 1, 1000, 1, -1, \;6 kloop2 33, 1, 1000, 1, -1, \;7 kloop3 34, 1, 1000, 1, -1 ;8 kloop4 kmodindex1 init 0 kmodindex2 init 0 kmodindex3 init 0 kamp init 0 kfreq1 init 0 kfreq2 init 0 kfreq3 init 0 kfreq4 init 0 vtable1k giOutTab2, kmodindex1, kmodindex2, kmodindex3, kamp, kfreq1, kfreq2, kfreq3, kfreq4 ; *kflag, *ihandle, *ifn, *startInd, *startSlid, *numSlid; FLslidBnk2Setk ktrig, giHandle, giOutTab, 0, 0, giElem printk2 kmodindex1 printk2 kmodindex2,10 printk2 kmodindex3,20 printk2 kamp,30 amod1 oscili kmodindex1, kfreq1, giSine amod2 oscili kmodindex2, kfreq2, giSine amod3 oscili kmodindex3, kfreq3, giSine aout oscili kamp, kfreq4+amod1+amod2+amod3, giSine outs aout, aout endin i1 0 3600 manual_src~dfsg/examples/vmult_i.csd0000644000000000000000000000146612262561504016744 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vmult_i ifn1, ival, ielements, idstoffset endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e manual_src~dfsg/examples/streson.csd0000644000000000000000000000121312262561504016750 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o streson.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 asig diskin2 "fox.wav", 1, 0, 1 kfr = p4 ifdbgain = 0.90 astr streson asig, kfr, ifdbgain asig clip astr, 0, 1 outs asig, asig endin i 1 0 1 20 i 1 + . > i 1 + . > i 1 + . > i 1 + . > i 1 + . > i 1 + . 1000 e manual_src~dfsg/examples/outkc.csd0000644000000000000000000000126512262561504016407 0ustar rootroot ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out and midi in and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outkc.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 ikey notnum ivel veloc kcut linseg 100, .5, 20 ;vary controller in .5 second kbrt = int(kcut) ;whole numbers only outkc 1, 74, kbrt, 0, 127 ;controller 74 (= brightness) midion 1, ikey, ivel ;of external synth endin f0 30 ;runs 30 seconds e manual_src~dfsg/examples/ftgenonce.csd0000644000000000000000000000127312262561504017231 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftgenonce.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Use ftgenonce instead of ftgen, ftgentmp, or f statement iHz = p4 isine ftgenonce 0, 0, 1024, 10, 1 aoscili pluck .7, iHz, 100, isine, 1 aadsr adsr 0.015, 0.07, 0.6, 0.3 asig = aoscili * aadsr outs asig, asig endin i 1 0 1 220 i 1 2 1 261 e manual_src~dfsg/examples/betarand.csd0000644000000000000000000000142212262561504017035 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o betarand.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values kbeta betarand 100, 1, 1 printk .2, kbeta ; look aout oscili 0.8, 440+kbeta, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 kbeta betarand 100, 1, 1 printk .2, kbeta ; look aout oscili 0.8, 440+kbeta, 1 ; & listen outs aout, aout endin ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/adsynt.csd0000644000000000000000000000434612262561504016567 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o adsynt.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; Generate a sinewave table. giwave ftgen 1, 0, 1024, 10, 1 ; Generate two empty tables for adsynt. gifrqs ftgen 2, 0, 32, 7, 0, 32, 0 ; A table for freqency and amp parameters. giamps ftgen 3, 0, 32, 7, 0, 32, 0 ; Generates parameters at init time instr 1 ; Generate 10 voices. icnt = 10 ; Init loop index. index = 0 ; Loop only executed at init time. loop: ; Define non-harmonic partials. ifreq pow index + 1, 1.5 ; Define amplitudes. iamp = 1 / (index+1) ; Write to tables. tableiw ifreq, index, gifrqs ; Used by adsynt. tableiw iamp, index, giamps index = index + 1 ; Do loop/ if (index < icnt) igoto loop asig adsynt 0.3, 150, giwave, gifrqs, giamps, icnt outs asig, asig endin ; Generates parameters every k-cycle. instr 2 ; Generate 10 voices. icnt = 10 ; Reset loop index. kindex = 0 ; Loop executed every k-cycle. loop: ; Generate lfo for frequencies. kspeed pow kindex + 1, 1.6 ; Individual phase for each voice. kphas phasorbnk kspeed * 0.7, kindex, icnt klfo table kphas, giwave, 1 ; Arbitrary parameter twiddling... kdepth pow 1.4, kindex kfreq pow kindex + 1, 1.5 kfreq = kfreq + klfo*0.006*kdepth ; Write freqs to table for adsynt. tablew kfreq, kindex, gifrqs ; Generate lfo for amplitudes. kspeed pow kindex + 1, 0.8 ; Individual phase for each voice. kphas phasorbnk kspeed*0.13, kindex, icnt, 2 klfo table kphas, giwave, 1 ; Arbitrary parameter twiddling... kamp pow 1 / (kindex + 1), 0.4 kamp = kamp * (0.3+0.35*(klfo+1)) ; Write amps to table for adsynt. tablew kamp, kindex, giamps kindex = kindex + 1 ; Do loop. if (kindex < icnt) kgoto loop asig adsynt 0.25, 150, giwave, gifrqs, giamps, icnt outs asig, asig endin ; Play Instrument #1 for 2.5 seconds. i 1 0 2.5 ; Play Instrument #2 for 2.5 seconds. i 2 3 2.5 e manual_src~dfsg/examples/dssiplay_mono.csd0000644000000000000000000000203512262561504020136 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o DSSIplay_mono.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 gihandle dssiinit "caps.so", 19, 1 ; = mono phaser and gaout init 0 ; verbose about all ports instr 1 ; activate DSSI dssiactivate gihandle, 1 endin instr 2 ain1 diskin2 "beats.wav", 1,0,1 ; loop gaout = gaout+(ain1*.5) endin instr 3 dssictls gihandle, 0, .8, 1 ; range -1 to 1 dssictls gihandle, 1, .05, 1 ; rate 0 to 10 dssictls gihandle, 2, .8, 1 ; depth 0 to 1 dssictls gihandle, 3, 2, 1 ; spread 0 to 3.14 dssictls gihandle, 4, .7, 1 ; feedback 0 to 0.999 endin instr 4 aout1 dssiaudio gihandle, gaout ;get beats.wav, mono out outs aout1,aout1 gaout = 0 endin i 1 0 20 i 2 1 20 i 3 1 20 i 4 0 20 e manual_src~dfsg/examples/gen09square.csd0000644000000000000000000000132612262561504017423 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o gen09square.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 kcps = 440 ifn = 1 ; Play the waveform stored in Table #1. a1 oscil kamp, kcps, ifn out a1 endin ; Table #1: an approximation of a square wave (using GEN09). f 1 0 16384 9 1 3 0 3 1 0 9 0.3333 180 ; Play Instrument #1 for 2 seconds. i 1 0 2 e manual_src~dfsg/examples/ATSread-2.csd0000644000000000000000000000672612262561504016713 0ustar rootroot -odac -d -m128 ;example by joachim heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 1024, 10, 1 gSfile = "fox.ats" giNumParts ATSinfo gSfile, 3 ;overall number of partials giDur ATSinfo gSfile, 7 ;duration seed 0 instr ReadOnePartial iPartial = p4 p3 = giDur ktime line 0, giDur, giDur prints "Resynthesizing partial number %d.\n", iPartial kFq,kAmp ATSread ktime, gSfile, iPartial kAmp port kAmp, .1 ;smooth amplitudes - still not satisfactoring aOut poscil kAmp, kFq, giSine aOut linen aOut, 0, p3, .01 ;anti-click outs aOut*10, aOut*10 ;start next instr: normal speed, three loops, pause between loops one second event_i "i", "MasterRand", giDur+3, 1, 1, 3, 2 endin instr MasterRand ;random selections of 10 partials per second, overlapping iSpeed = p4 ;speed of reading / playing iNumLoops = p5 ;number of loops iPause = p6 ;length of pause between loops prints "Resynthesizing random partials.\n" p3 = (giDur/iSpeed+iPause) * iNumLoops ;start next instr: half speed, three loops, three seonds pause between loops event_i "i", "MasterArp", p3+3, 1, .5, 3, 3 ;loop over duration plus pause loop: timout 0, giDur/iSpeed+iPause, play reinit loop play: gkTime line 0, giDur/iSpeed, giDur ;start time from 0 in each loop kTrig metro 10 ;10 new partials per second ;call subinstrument if trigger and no pause if kTrig == 1 && gkTime < giDur then kPart random 1, giNumParts+.999 event "i", "PlayRand", 0, 1, int(kPart) endif endin instr MasterArp ;argeggio-like reading and playing of partials iSpeed = p4 ;speed of reading / playing iNumLoops = p5 ;number of loops iPause = p6 ;length of pause between loops prints "Arpeggiating partials.\n" p3 = (giDur/iSpeed+iPause) * iNumLoops loop: timout 0, giDur/iSpeed+iPause, play reinit loop play: gkTime line 0, giDur/iSpeed, giDur kArp linseg 1, (giDur/iSpeed)/2, giNumParts, (giDur/iSpeed)/2, 1 ;arp up and down kTrig metro 10 ;10 new partials per second if kTrig == 1 && gkTime < giDur then event "i", "PlayArp", 0, 5, int(kArp) endif ;exit csound when finished event_i "i", "End", p3+5, 1 endin instr PlayRand iPartial = p4 kFq,kAmp ATSread gkTime, gSfile, iPartial kamp port kAmp, .15 ;smooth amplitudes aOut poscil kAmp, kFq, giSine aOut linen aOut, .01, p3, .01 outs aOut, aOut endin instr PlayArp kCount init 1 ;k-cycle iPartial = p4 kFq,kAmp ATSread gkTime, gSfile, iPartial if kCount == 1 then ;get freq from first k-cycle kModFq = kFq ;avoid to go with 0 Hz as this blocks the mode filter if kModFq == 0 then turnoff endif endif iVol random -42, -12 ;db iOffset random .01, .1 ;no too regularily ... aImp mpulse ampdb(iVol), p3, iOffset iQ random 500, 5000 aOut mode aImp, kModFq, iQ aOut linen aOut, 0, p3, p3/3 outs aOut, aOut kCount = 2 endin instr End exitnow endin i "ReadOnePartial" 0 1 10 e 999 manual_src~dfsg/examples/inq.csd0000644000000000000000000000164712262561504016055 0ustar rootroot ; Select audio/midi flags here according to platform -odac -idac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; inq.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 ;2 channels out 0dbfs = 1 nchnls_i = 4 ;4 channels in instr 1 ;4 channels in, two channels out ain1, ain2, ain3, ain4 inq ;grab your mics and sing adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time = 20ms adel2 linseg 0.02, p3*.5, 0, p3*.5, 0.02 ;max delay time = 20ms aoutl flanger ain1, adel, .7 aoutr flanger ain2, adel*2, .8 aoutla flanger ain3, adel2, .9 aoutra flanger ain4, adel2*2, .5 ;write to quad file, 16 bits with header fout "in_4.wav", 14, aoutl, aoutr, aoutla, aoutra outs (aoutl+aoutla)*.5, (aoutr+aoutra)*.5 ;stereo out endin i 1 0 10 e manual_src~dfsg/examples/clear.csd0000644000000000000000000000127612262561504016352 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o clear.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gaReverb init 0 instr 1 idur = p3 kpitch = p4 a1 diskin2 "fox.wav", kpitch a1 = a1*.5 ;reduce volume vincr gaReverb, a1 endin instr 99 ; global reverb al, ar reverbsc gaReverb, gaReverb, .8, 10000 outs gaReverb+al, gaReverb+ar clear gaReverb endin i1 0 3 1 i99 0 5 e manual_src~dfsg/examples/gen10.csd0000644000000000000000000000146712262561504016200 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen10.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = .6 kcps = 440 ifn = p4 asig oscil kamp, kcps, ifn outs asig,asig endin f1 0 16384 10 1 ; Sine f2 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth f3 0 16384 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square f4 0 16384 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse i 1 0 2 1 i 1 3 2 2 i 1 6 2 3 i 1 9 2 4 e manual_src~dfsg/examples/readk4.csd0000644000000000000000000000313312262561504016430 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o readk4.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes four control signals to a file kcf randh 950, 1, 2, 1, 1050 ;generates one random number between 100 and 2000 per second kq randh 10, 1, 2, 1, 11 ;generates another random number between 1 and 21 per second kdb randh 9, 1, 2, 1, -15 ;amplitudes in dB between -24 and -6 kpan randh .5, 1, 2, 1, .5 ;panning between 0 and 1 dumpk4 kcf, kq, kdb, kpan, "dumpk4.txt", 8, 1 ;writes the control signals prints "WRITING:\n" printks "kcf = %f, kq = %f, kdb = %f, kpan = %f\n", 1, kcf, kq, kdb, kpan ;prints them endin instr 2 ;reads the file written by instr 1 kcf,kq,kdb,kp readk4 "dumpk4.txt", 8, 1 prints "READING:\n" printks "kcf = %f, kq = %f, kdb = %f, kpan = %f\n", 1, kcf, kq, kdb, kp ;prints values kdb lineto kdb, .1 ;smoothing amp transition kp lineto kp, .1 ;smoothing pan transition anoise rand ampdb(kdb), 2, 1 kbw = kcf/kq ;bandwidth of resonant filter abp reson anoise, kcf, kbw aout balance abp, anoise aL, aR pan2 aout, kp outs aL, aR endin i 1 0 5 i 2 5 5 e manual_src~dfsg/examples/test1.png0000644000000000000000000010672712262561504016346 0ustar rootrootPNG  IHDR{CsRGB pHYs  tIME `tEXtCommentCreated with GIMPW IDATxu6;\J(*(˱Nl AO7^|IĊ,Qd(.ܝ~(T˩s:nOOOOS O(K+) 9OiS!?= Q9_ }_I\9H!a-jN""_ozb__*zɳ\H!p&S{9GBߦƬdPUHi(뗰 j=΃L!a Ȃˢߨ&%6KgsދnzPH?"C? 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L ng#X67;.G!FY+%Hg+ O]Q' /yfe19p 0^Pž0yا@THt/!LH;;{ɹD.#F\Xy@O?J=y3?>S^*KCh}B*{ TE/]QGL !IGQigmW1}R4)¿q/P[|^]ӚGyz];IwYV(loz+l<඼Wu&'$Q!C+`=v]TC>k+ՋA$W(b|'j@Dg$Tџ4%r)'i>ajFBMi" 1e=]cͣakQr_h .qZDosd tV;TY?,wd=D_9DTMM~J=/.oWY*08&O.ѕ@oMIe0Z@'GS%QQWbS _Tⷩ]|p-5>+~#E`Lf<؃Y7f|<;TɮaNvD+?`ݎnW怩!r|$]6_c/dӡꣿ+f0WeU%@ hnb7{hpWK-yqI7u'Mt^vz6χ#'ܹG%؛VU^m֏SuGF6ɡ<[} 0WKAra}kGr/̞zmѷ.ؙqp5/ c_‰>7z M.sI]=tVjr*CYwQDjІr:vUb"h'CTe@AzG+迕{z)H;kHqe=B Q(=(RwzHi 'Eg#IENDB`manual_src~dfsg/examples/vexp_i.csd0000644000000000000000000000146512262561504016556 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vexp_i ifn1, ival, ielements, idstoffset endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e manual_src~dfsg/examples/maxarray.csd0000644000000000000000000000133712262561504017106 0ustar rootroot ; Select audio/midi flags here according to platform -n ;example by joachim heintz seed 0 instr 1 ;create an array with 10 elements kArr[] init 10 ;fill in random numbers and print them out kIndx = 0 until kIndx == 10 do kNum random -100, 100 kArr[kIndx] = kNum printf "kArr[%d] = %10f\n", kIndx+1, kIndx, kNum kIndx += 1 od ;investigate maximum number and print it out kMax, kMaxIndx maxarray kArr printf "Maximum of kArr = %f at index %d\n", kIndx+1, kMax, kMaxIndx turnoff endin i1 0 0.1 e manual_src~dfsg/examples/pvsbin.csd0000644000000000000000000000137112262561504016561 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsbin.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 ifftsize = 1024 iwtype = 1 /* cleaner with hanning window */ ;a1 soundin "input.wav" ;select a soundifle a1 inch 1 ;Use realtime input fsig pvsanal a1, ifftsize, ifftsize/4, ifftsize, iwtype kamp, kfr pvsbin fsig, 10 adm oscil kamp, kfr, 1 out adm endin ; sine wave f 1 0 4096 10 1 i 1 0 30 e manual_src~dfsg/examples/mclock.csd0000644000000000000000000000047212262561504016531 0ustar rootroot ; Select audio/midi flags here according to platform -Q0 ;;;midi out sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;let csound synchronize a sequencer mclock 24 endin i 1 0 30 e manual_src~dfsg/examples/init.csd0000644000000000000000000000102212262561504016214 0ustar rootroot ; Select audio/midi flags here according to platform -n ;no sound output sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;shows what init does kinit init 0 kinit = kinit + 1 printk .1, kinit endin instr 2 ;shows what an assignment does knoinit = 0 knoinit = knoinit + 1 printk .1, knoinit endin ;play one second each i1 0 1 i2 2 1 e manual_src~dfsg/examples/grain3.csd0000644000000000000000000000506412262561504016446 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o grain3.wav -W ;;; for file output any platform sr = 48000 kr = 1000 ksmps = 48 nchnls = 1 /* Bartlett window */ itmp ftgen 1, 0, 16384, 20, 3, 1 /* sawtooth wave */ itmp ftgen 2, 0, 16384, 7, 1, 16384, -1 /* sine */ itmp ftgen 4, 0, 1024, 10, 1 /* window for "soft sync" with 1/32 overlap */ itmp ftgen 5, 0, 16384, 7, 0, 256, 1, 7936, 1, 256, 0, 7936, 0 /* generate bandlimited sawtooth waves */ itmp ftgen 3, 0, 4096, -30, 2, 1, 2048 icnt = 0 loop01: ; 100 tables for 8 octaves from 30 Hz ifrq = 30 * exp(log(2) * 8 * icnt / 100) itmp ftgen icnt + 100, 0, 4096, -30, 3, 1, sr / (2 * ifrq) icnt = icnt + 1 if (icnt < 99.5) igoto loop01 /* convert frequency to table number */ #define FRQ2FNUM(xout'xcps'xbsfn) # $xout = int(($xbsfn) + 0.5 + (100 / 8) * log(($xcps) / 30) / log(2)) $xout limit $xout, $xbsfn, $xbsfn + 99 # /* instr 1: pulse width modulated grains */ instr 1 kfrq = 523.25 ; frequency $FRQ2FNUM(kfnum'kfrq'100) ; table number kfmd = kfrq * 0.02 ; random variation in frequency kgdur = 0.2 ; grain duration kdens = 200 ; density iseed = 1 ; random seed kphs oscili 0.45, 1, 4 ; phase a1 grain3 kfrq, 0, kfmd, 0.5, kgdur, kdens, 100, \ kfnum, 1, -0.5, 0, iseed, 2 a2 grain3 kfrq, 0.5 + kphs, kfmd, 0.5, kgdur, kdens, 100, \ kfnum, 1, -0.5, 0, iseed, 2 ; de-click aenv linseg 0, 0.01, 1, p3 - 0.05, 1, 0.04, 0, 1, 0 out aenv * 2250 * (a1 - a2) endin /* instr 2: phase variation */ instr 2 kfrq = 220 ; frequency $FRQ2FNUM(kfnum'kfrq'100) ; table number kgdur = 0.2 ; grain duration kdens = 200 ; density iseed = 2 ; random seed kprdst expon 0.5, p3, 0.02 ; distribution a1 grain3 kfrq, 0.5, 0, 0.5, kgdur, kdens, 100, \ kfnum, 1, 0, -kprdst, iseed, 64 ; de-click aenv linseg 0, 0.01, 1, p3 - 0.05, 1, 0.04, 0, 1, 0 out aenv * 1500 * a1 endin /* instr 3: "soft sync" */ instr 3 kdens = 130.8 ; base frequency kgdur = 2 / kdens ; grain duration kfrq expon 880, p3, 220 ; oscillator frequency $FRQ2FNUM(kfnum'kfrq'100) ; table number a1 grain3 kfrq, 0, 0, 0, kgdur, kdens, 3, kfnum, 5, 0, 0, 0, 2 a2 grain3 kfrq, 0.667, 0, 0, kgdur, kdens, 3, kfnum, 5, 0, 0, 0, 2 ; de-click aenv linseg 0, 0.01, 1, p3 - 0.05, 1, 0.04, 0, 1, 0 out aenv * 10000 * (a1 - a2) endin t 0 60 i 1 0 3 i 2 4 3 i 3 8 3 e manual_src~dfsg/examples/fibonacci.txt0000644000000000000000000000010312262561504017233 0ustar rootroot1 1 2 3 5 8 13 21 34 55 89 144 233 377 610 987 1597 2584 4181 6765 manual_src~dfsg/examples/vbap.csd0000644000000000000000000000137012262561504016207 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbap.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 vbaplsinit 2, 4, 0, 90, 180, 270 instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav kaz line 0, p3, p4 ;come from right rear speaker & a1,a2,a3,a4 vbap asig, 180, 100, kaz ;change spread of soundsource printks "spread of source = %d\n", 1, kaz ;print spread value outq a1,a2,a3,a4 endin i 1 0 12 100 e manual_src~dfsg/examples/toot9.csd0000644000000000000000000000262312262561504016337 0ustar rootroot -odac nchnls = 2 ; stereo output garvbsig init 0 ; global "a" variable initialized to 0 instr 9 idur = p3 iamp = p4 iskiptime = p5 iattack = p6 irelease = p7 ibalance = p8 ; 1 = left, .5 = center, 0 = right irvbgain = p9 kamp linen iamp, iattack, idur, irelease asig soundin "fox.wav", iskiptime arampsig = kamp * asig outs arampsig * ibalance, arampsig * (1 - ibalance) garvbsig = garvbsig + arampsig * irvbgain endin instr 99 irvbtime = p4 asig reverb garvbsig, irvbtime ; put global sig into reverb outs asig, asig garvbsig = 0 ; then clear it endin ;ins strt dur rvbtime i99 0 15 2.2 ;ins strt dur amp skip atk rel blnce(0-1) rvbsend i9 0 1.2 .5 0 .02 .1 1 .2 i9 2 1.4 .5 0 .03 .1 0 .3 i9 3.5 2.28 .5 0 .9 .1 .5 .1 i9 4.5 2.28 .5 0 1.2 .1 0 .2 i9 5 2.28 .5 0 .2 .1 1 .3 i9 9 2.28 .7 0 .1 .1 .5 .03 e manual_src~dfsg/examples/reverbsc.csd0000644000000000000000000000150112262561504017066 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o reverbsc.wav -W ;;; for file output any platform sr = 48000 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 a1 vco2 0.85, 440, 10 kfrq port 100, 0.004, 20000 a1 butterlp a1, kfrq a2 linseg 0, 0.003, 1, 0.01, 0.7, 0.005, 0, 1, 0 a1 = a1 * a2 a2 = a1 * p5 a1 = a1 * p4 denorm a1, a2 aL, aR reverbsc a1, a2, 0.85, 12000, sr, 0.5, 1 outs a1 + aL, a2 + aR endin i 1 0 1 0.71 0.71 i 1 1 1 0 1 i 1 2 1 -0.71 0.71 i 1 3 1 1 0 i 1 4 4 0.71 0.71 e manual_src~dfsg/examples/pvsmooth.csd0000644000000000000000000000175712262561504017147 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsmooth.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kacf = p4 kfcf = p5 asig soundin "fox.wav" fsig pvsanal asig, 1024, 256, 1024, 1 ; analyse it ftps pvsmooth fsig, kacf, kfcf atps pvsynth ftps ; synthesise it outs atps*3, atps*3 endin ; amp freq i 1 0 3 0.01 0.01 ;smooth amplitude and frequency with cutoff frequency of filter at 1% of 1/2 frame-rate (ca 0.86 Hz) i 1 + 3 1 0.01 ;no smoothing on amplitude, but frequency with cf at 1% of 1/2 frame-rate (ca 0.86 Hz) i 1 + 10 .001 1 ;smooth amplitude with cf at 0.1% of 1/2 frame-rate (ca 0.086 Hz) ;and no smoothing of frequency e manual_src~dfsg/examples/multitap.csd0000644000000000000000000000121712262561504017116 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o multitap.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ga1 init 0 instr 1 asig diskin2 "beats.wav", 1,0 outs asig, asig ga1 = ga1+asig endin instr 2 asig multitap ga1, 1.2, .5, 1.4, .2 outs asig, asig ga1 = 0 endin i 1 .5 .2 ; short sound i 2 0 3 ; echoes e manual_src~dfsg/examples/mandpluk.aiff0000644000000000000000000002141212262561504017225 0ustar rootrootFORM# AIFF(c)  ECOMMb@ DSSND" e  Kr sٶަ7mP6L`[nt[p-eW?H8!?ýק Z\¼q=0*Qy678[s|yqji>knlbP_;R)} #. 8<5(&~7 xQ : ZB$&[¦^ ? 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Bc`P#(-//.,)e& " L, ~?)2 2>+XZ!gjsoN>ks:U`~H? \ @ U\pr8z#!=ne 1A=UT!lbaeySTKMsmChU=]9"]5- t y= \eX}uvz<4M`J < "U:@sQ"cC?*}b\y_OFIW;.YG q ` 00wc*j6  .|0pcg1@JvFdP0R7Z@Fz,l^ Z V 8 { - C7  _I3J!5YqS3^]kvvYDDaSTDu71TU k-KlC7mQ%0bhQcc&H[^ZROF, Nm#xMmE3[ /, Z9j3c!xC>l GW0htrncZcyQ)Yu=/geJ605;8'S,~OA] 0j*G Fi VYJNNJ%-V4gB_XA,! uf} b}%c. }ML~ZXdgPu ]&(}dsMVMNkO)hxjVN%.'mxul66OUD!@UO/yf-YS_9|8jOW>mvgl(SxMwy]3[|b9I2Tt+`=)2&lO8,0Fl/bSkVNUbh[8l@Z)>oi&^0B(-gAZ MP0Sd^E'  \'5jp&xa`~l%%j8]qkT%zL=jqN;83Y) W qhkx ;u"m+:|(BUrLvup[-h`[L21w &^\!<b KN*e5>sG $Do xN(B4?pDK  dGcjo~M{mW- `3s0>i0~r]=##C$j8d3Xj_<Jd}5}X0dk0_iL:5ASj 6GB!fP:lXXo8B0et.~^?j4OSP~U#r>ZsAay{_;uP&AvcR`#j"8;'uP&udWPRXcr ,Syx[D?FOP>iF kK5),>d2'm%0.$p\O?'}} 6Sm~{U j~kc^ao R#//&!06,p^PJJMOJ>-$8XD9[qvhJ'i0~Z45Mg?VXJ9//6;6gI&*Rr 3VuyV#m[G, F{Bb|o<U/8o!,3ARj^85m5^~bMDGLJ<lU@2-2@P^gmr,000232* xaM=/))-4;FWm5Xs~R&`q [W$Ll~pd[VV[agllia\SF2 '?[us<f5!>]*Uig\RMORSSOLD6P/iGxrU;!mL4# -=LZi 6RYUPUi{[;#pD-UzJiy~||{uiS>0/9LVUD$iF#2Oo&Oro> gF#0Pjusld`co 0IasyuoigaO,udXM@7=U}!Bi~_A' U& -S-D\vjI!l^`o 3Ldx|jP- {L# $Do&29?DIP[gsyviO0iUMORUSUXdxFjydFU2 #$!#8Xz;XppBoS8$*Dg ,?B<0))-5;93'^C>Zmanual_src~dfsg/examples/vdelayxs.csd0000644000000000000000000000156612262561504017125 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelayxs.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ims = .5 ;maximum delay time in seconds iws = 1024 ;window size adl = .5 ;delay time asig1, asig2 diskin2 "kickroll.wav", 1, 0, 1 ;loop stereo file kickroll.wav a2 poscil .25, .1, 1 ;make an LFO, 1 cycle per 2 seconds adl = a2 + ims/2 ;offset the LFO so that it is positive aoutL, aoutR vdelayxs asig1, asig2, adl, ims, iws ;use the LFO to control delay time outs aoutL, aoutR endin f1 0 8192 10 1 i 1 0 10 e manual_src~dfsg/examples/wguide1.csd0000644000000000000000000000162612262561504016630 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o wguide1.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a simple noise waveform. instr 1 ; Generate some noise. asig noise 20000, 0.5 out asig endin ; Instrument #2 - a waveguide example. instr 2 ; Generate some noise. asig noise 20000, 0.5 ; Run it through a wave-guide model. kfreq init 200 kcutoff init 3000 kfeedback init 0.8 awg1 wguide1 asig, kfreq, kcutoff, kfeedback out awg1 endin ; Play Instrument #1 for 2 seconds. i 1 0 2 ; Play Instrument #2 for 2 seconds. i 2 2 2 e manual_src~dfsg/examples/octpch.csd0000644000000000000000000000116312262561504016537 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o octpch.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Convert a pitch-class value into an ; octave-point-decimal value. ipch = 8.09 ioct = octpch(ipch) print ioct endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/ctrl7.csd0000644000000000000000000000121312262561504016306 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M0 ;;;RT audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if RT audio input is needed too sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; expects MIDI controller input on channel 1 ; run and move your midi controller to see result imax = 1 imin = 0 ichan = 1 ictlno = 7 initc7 1, 7, 1 ; start at max. volume kamp ctrl7 ichan, ictlno, imin, imax ; controller 7 printk2 kamp asig oscil kamp, 220, 1 outs asig, asig endin f 1 0 4096 10 1 i1 0 20 e manual_src~dfsg/examples/tempoval.csd0000644000000000000000000000122112262561504017101 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc -t60 ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tempoval.wav -W -t60 ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Adjust the tempo to 120 beats per minute. tempo 120, 60 ; Get the tempo value. kval tempoval printks "kval = %f\\n", 0.1, kval endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/hilbert_barberpole.csd0000644000000000000000000000325212262561504021106 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o hilbert_barberpole.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 ; kr must equal sr for the barberpole effect to work. kr = 44100 ksmps = 1 nchnls = 2 ; Instrument #1 instr 1 idur = p3 ibegshift = p4 iendshift = p5 ; sawtooth wave, not bandlimited asaw phasor 100 ; add offset to center phasor amplitude between -.5 and .5 asaw = asaw - .5 ; sawtooth wave, with amplitude of 10000 ain = asaw * 20000 ; The envelope of the frequency shift. kfreq linseg ibegshift, idur, iendshift ; Phase quadrature output derived from input signal. areal, aimag hilbert ain ; The quadrature oscillator. asin oscili 1, kfreq, 1 acos oscili 1, kfreq, 1, .25 ; Based on trignometric identities. amod1 = areal * acos amod2 = aimag * asin ; Calculate the up-shift and down-shift. aupshift = (amod1 + amod2) * 0.7 adownshift = (amod1 - amod2) * 0.7 ; Mix in the original signal to achieve the barberpole effect. amix1 = aupshift + ain amix2 = aupshift + ain ; Make sure the output doesn't get louder than the original signal. aout1 balance amix1, ain aout2 balance amix2, ain outs aout1, aout2 endin ; Table 1: A sine wave for the quadrature oscillator. f 1 0 16384 10 1 ; The score. ; p4 = frequency shifter, starting frequency. ; p5 = frequency shifter, ending frequency. i 1 0 6 -10 10 e manual_src~dfsg/examples/frac.csd0000644000000000000000000000104012262561504016164 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o frac.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = 16 / 5 i2 = frac(i1) print i2 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/pvsfread.csd0000644000000000000000000000140712262561504017072 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsfread.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; create a PVOC-EX (*.pvx) file with PVANAL first idur filelen "kickroll.pvx" ;find duration of (stereo) analysis file kpos line 0,p3,idur ;to ensure we process whole file fsigr pvsfread kpos,"kickroll.pvx", 1 ;create fsig from right channel aout pvsynth fsigr ;resynthesise it outs aout, aout endin i 1 0 10 i 1 11 1 e manual_src~dfsg/examples/zacl.csd0000644000000000000000000000205112262561504016205 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zacl.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Generate a simple sine waveform. asin oscil 20000, 440, 1 ; Send the sine waveform to za variable #1. zaw asin, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read za variable #1. a1 zar 1 ; Generate the audio output. out a1 ; Clear the za variables, get them ready for ; another pass. zacl 0, 1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e manual_src~dfsg/examples/vbaplsinit.csd0000644000000000000000000000162112262561504017431 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbaplsinit.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 8 0dbfs = 1 vbaplsinit 2, 8, 0, 45, 90, 180, 270, 0, 0, 0 ;5 speakers for 5.1 amps instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav kazim line 1, p3, 355 a1,a2,a3,a4,a5,a6,a7,a8 vbap8 asig, kazim, 0, 1 ;change azimuth of soundsource ; Speaker mapping aFL = a1 ; Front Left aMF = a5 ; Mid Front aFR = a2 ; Front Right aBL = a3 ; Back Left aBR = a4 ; Back Right outo aFL,aFR,aBL,aBR,aMF,a6,a7,a8 ;a6, a7 and a8 are dummies endin i 1 0 5 e manual_src~dfsg/examples/delayr.csd0000644000000000000000000000216312262561504016540 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o delayr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasig init 0 gidel = 1 ;delay time in seconds instr 1 ain pluck .7, 440, 1000, 0, 1 outs ain, ain vincr gasig, ain ;send to global delay endin instr 2 ifeedback = p4 abuf2 delayr gidel adelL deltap .4 ;first tap (on left channel) adelM deltap 1 ;second tap (on middle channel) delayw gasig + (adelL * ifeedback) abuf3 delayr gidel kdel line 1, p3, .01 ;vary delay time adelR deltap .65 * kdel ;one pitch changing tap (on the right chn.) delayw gasig + (adelR * ifeedback) ;make a mix of all deayed signals outs adelL + adelM, adelR + adelM clear gasig endin i 1 0 1 i 1 3 1 i 2 0 3 0 ;no feedback i 2 3 8 .8 ;lots of feedback e manual_src~dfsg/examples/cpsxpch.csd0000644000000000000000000000123212262561504016724 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cpsxpch.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a normal twelve-tone scale. ipch = 8.02 iequal = 12 irepeat = 2 ibase = 1.02197503906 icps cpsxpch ipch, iequal, irepeat, ibase print icps endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/tone.csd0000644000000000000000000000124212262561504016222 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tone.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 asig diskin2 "beats.wav", 1 outs asig, asig endin instr 2 kton line 10000, p3, 0 ;all the way down to 0 Hz asig diskin2 "beats.wav", 1 asig tone asig, kton ;half-power point at 500 Hz outs asig, asig endin i 1 0 2 i 2 2 2 e manual_src~dfsg/examples/vtablek.csd0000644000000000000000000000105512262561504016707 0ustar rootroot -odac -B441 -b441 sr = 44100 kr = 100 ksmps = 441 nchnls = 2 gkindx init -1 instr 1 kindex init 0 ktrig metro 0.5 if ktrig = 0 goto noevent gkindx = gkindx + 1 noevent: endin instr 2 kout1 init 0 kout2 init 0 kout3 init 0 kout4 init 0 vtablek gkindx, 1, 1, 0, kout1,kout2, kout3, kout4 printk2 kout1 printk2 kout2 printk2 kout3 printk2 kout4 endin f 1 0 32 10 1 i 1 0 20 i 2 0 20 manual_src~dfsg/examples/FLvslidBnk.csd0000644000000000000000000000242012262561504017252 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O sr = 44100 kr = 441 ksmps = 100 nchnls = 1 gitypetable ftgen 0, 0, 8, -2, 1, 1, 3, 3, 5, 5, 7, 7 giouttable ftgen 0, 0, 8, -2, 0, 0.2, 0.3, 0.4, 0.5, 0.6, 0.8, 1 FLpanel "Slider Bank", 400, 400, 50, 50 ;Number of sliders inum = 8 ; Table to store output iouttable = giouttable ; Width of the slider bank in pixels iwidth = 350 ; Height of the slider in pixels iheight = 160 ; Distance of the left edge of the slider ; from the left edge of the panel ix = 30 ; Distance of the top edge of the slider ; from the top edge of the panel iy = 10 ; Table containing fader types itypetable = gitypetable FLvslidBnk "1@2@3@4@5@6@7@8@9@10@11@12@13@14@15@16", 16 , iouttable , iwidth , iheight , ix \ , iy , itypetable FLvslidBnk " ", inum , iouttable , iwidth , iheight , ix \ , iy + 200 , -23 ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 ;Dummy instrument endin ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e manual_src~dfsg/examples/pvsarp.csd0000644000000000000000000000162112262561504016571 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc ;;;-d RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsarp.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 ksmps = 100 nchnls = 1 0dbfs = 1 instr 1 asig in ; get the signal in idepth = p4 fsig pvsanal asig, 1024, 256, 1024, 1 ; analyse it kbin oscili 0.1, 0.5, 1 ; ftable 1 in the 0-1 range ftps pvsarp fsig, kbin+0.01, idepth, 2 ; arpeggiate it (range 220.5 - 2425.5) atps pvsynth ftps ; synthesise it out atps endin f 1 0 4096 10 1 ;sine wave i 1 0 10 0.9 i 1 + 10 0.5 e manual_src~dfsg/examples/mincer.csd0000644000000000000000000000123712262561504016536 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mincer.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = p3 ilock = p4 ipitch = 1 itimescale = 0.5 iamp = 0.8 atime line 0,idur,idur*itimescale asig mincer atime, iamp, ipitch, 1, ilock outs asig, asig endin f 1 0 0 1 "fox.wav" 0 0 0 i 1 0 5 0 ;not locked i 1 6 5 1 ;locked e manual_src~dfsg/examples/r.csd0000644000000000000000000000222212262561504015515 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o r.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; The score's p4 parameter has the number of repeats. kreps = p4 ; The score's p5 parameter has our note's frequency. kcps = p5 ; Print the number of repeats. printks "Repeated %i time(s).\\n", 1, kreps ; Generate a nice beep. a1 oscil 20000, kcps, 1 out a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; We'll repeat this section 6 times. Each time it ; is repeated, its macro REPS_MACRO is incremented. r6 REPS_MACRO ; Play Instrument #1. ; p4 = the r statement's macro, REPS_MACRO. ; p5 = the frequency in cycles per second. i 1 00.10 00.10 $REPS_MACRO 1760 i 1 00.30 00.10 $REPS_MACRO 880 i 1 00.50 00.10 $REPS_MACRO 440 i 1 00.70 00.10 $REPS_MACRO 220 ; Marks the end of the section. s e manual_src~dfsg/examples/vco2ft.csd0000644000000000000000000000352012262561504016461 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vco2ft.wav -W ;;; for file output any platform sr = 44100 ksmps = 10 nchnls = 1 ; user defined waveform -1: trapezoid wave with default parameters (can be ; accessed at ftables starting from 10000) itmp ftgen 1, 0, 16384, 7, 0, 2048, 1, 4096, 1, 4096, -1, 4096, -1, 2048, 0 ift vco2init -1, 10000, 0, 0, 0, 1 ; user defined waveform -2: fixed table size (4096), number of partials ; multiplier is 1.02 (~238 tables) itmp ftgen 2, 0, 16384, 7, 1, 4095, 1, 1, -1, 4095, -1, 1, 0, 8192, 0 ift vco2init -2, ift, 1.02, 4096, 4096, 2 instr 1 kcps expon p4, p3, p5 ; instr 1: basic vco2 example a1 vco2 12000, kcps ; (sawtooth wave with default out a1 ; parameters) endin instr 2 kcps expon p4, p3, p5 ; instr 2: kpw linseg 0.1, p3/2, 0.9, p3/2, 0.1 ; PWM example a1 vco2 10000, kcps, 2, kpw out a1 endin instr 3 kcps expon p4, p3, p5 ; instr 3: vco2 with user a1 vco2 14000, kcps, 14 ; defined waveform (-1) aenv linseg 1, p3 - 0.1, 1, 0.1, 0 ; de-click envelope out a1 * aenv endin instr 4 kcps expon p4, p3, p5 ; instr 4: vco2ft example, kfn vco2ft kcps, -2, 0.25 ; with user defined waveform a1 oscilikt 12000, kcps, kfn ; (-2), and sr/4 bandwidth out a1 endin i 1 0 3 20 2000 i 2 4 2 200 400 i 3 7 3 400 20 i 4 11 2 100 200 f 0 14 e manual_src~dfsg/examples/cpsxpch_pierce.csd0000644000000000000000000000124612262561504020260 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cpsxpch_pierce.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a Pierce scale centered on middle A. ipch = 2.02 iequal = 12 irepeat = 3 ibase = 261.62561 icps cpsxpch ipch, iequal, irepeat, ibase print icps endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/locsend_stereo.csd0000644000000000000000000000160412262561504020267 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o locsend_stereo.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ga1 init 0 ga2 init 0 instr 1 krevsend = p4 aout diskin2 "beats.wav", 1, 0, 1 kdegree line 0, p3, 180 ;left to right kdistance line 1, p3, 30 a1, a2 locsig aout, kdegree, kdistance, p4 ar1, ar2 locsend ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a2 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 outs a1, a2 ga1 = 0 ga2 = 0 endin ; sine wave. f 1 0 16384 10 1 i 1 0 4 .1 ;with reverb i 1 + 4 0 ;no reverb i99 0 7 e manual_src~dfsg/examples/STKMoog.csd0000644000000000000000000000124412262561504016542 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKMoog.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;filter Q asig STKMoog cpspch(ifrq), 1, 2,kv1, 4, 120, 11, 40, 1, 1, 128, 120 asig = asig * .3 ;too loud outs asig, asig endin i 1 0 .5 6.00 100 0 i 1 + . 5.05 10 127 i 1 + . 7.06 100 0 i 1 + 3 7.00 10 10 e manual_src~dfsg/examples/xscanmap.csd0000644000000000000000000000337212262561504017075 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o xscanmap.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;the matrices can be found in /manual/examples instr 1 ; Plain scanned syntnesis ; note - scanu display is turned off a0 = 0 xscanu 1, .01, 6, 2, "128-stringcircularX", 4, 5, 2, .1, .1, -.01, .1, .5, 0, 0, a0, 0, 0 a1 xscans p4, cpspch(p5), 7, 0, 3 k1,k2 xscanmap 0, 1000, 1000, 64 display k1, .25 ; note - display is updated every second outs a1, a1 endin instr 2 ; Scan synthesis with audio injection and dual scan paths ; note - scanu display is turned off ain diskin2 "fox.wav",1,0,1 ain in a0 = ain/10000 xscanu 1, .01, 6, 2, "128,8-gridX", 14, 5, 2, .01, .05, -.05, .1, .5, 0, 0, a0, 0, 0 a1 xscans p4, cpspch(p5), 7, 0, 2 a2 xscans p4, cpspch(p6), 77, 0, 3 k1,k2 xscanmap 0, 1000, 1000, 127 display k2, .5 ; note - display is updated ever 500ms outs a1,a2 endin ; Initial condition ;f1 0 16 7 0 8 1 8 0 f1 0 128 7 0 64 1 64 0 ; Masses f2 0 128 -7 1 128 1 ; Centering force f4 0 128 -7 0 128 2 f14 0 128 -7 2 64 0 64 2 ; Damping f5 0 128 -7 1 128 1 ; Initial velocity f6 0 128 -7 -.0 128 .0 ; Trajectories f7 0 128 -5 .001 128 128 f77 0 128 -23 "128-spiral-8,16,128,2,1over2" ; Sine f9 0 1024 10 1 ;-------------------------------- ; Note list i1 0 10 .9 7.00 s i2 0 10 1 8.00 6.00 i2 0 10 1 7.00 8.05 e manual_src~dfsg/examples/chnget.csd0000644000000000000000000000351712262561504016534 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chnget.wav -W ;;; for file output any platform ;Example by Joachim Heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1; send i-values chnset 1, "sio" chnset -1, "non" endin instr 2; send k-values kfreq randomi 100, 300, 1 chnset kfreq, "cntrfreq" kbw = kfreq/10 chnset kbw, "bandw" endin instr 3; send a-values anois rand .1 chnset anois, "noise" loop: idur random .3, 1.5 timout 0, idur, do reinit loop do: ifreq random 400, 1200 iamp random .1, .3 asig oscils iamp, ifreq, 0 aenv transeg 1, idur, -10, 0 asine = asig * aenv chnset asine, "sine" endin instr 11; receive some chn values and send again ival1 chnget "sio" ival2 chnget "non" print ival1, ival2 kcntfreq chnget "cntrfreq" kbandw chnget "bandw" anoise chnget "noise" afilt reson anoise, kcntfreq, kbandw afilt balance afilt, anoise chnset afilt, "filtered" endin instr 12; mix the two audio signals amix1 chnget "sine" amix2 chnget "filtered" chnmix amix1, "mix" chnmix amix2, "mix" endin instr 20; receive and reverb amix chnget "mix" aL, aR freeverb amix, amix, .8, .5 outs aL, aR endin instr 100; clear chnclear "mix" endin i 1 0 20 i 2 0 20 i 3 0 20 i 11 0 20 i 12 0 20 i 20 0 20 i 100 0 20 manual_src~dfsg/examples/fof.csd0000644000000000000000000000363112262561504016033 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fof.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 ; Combine five formants together to create ; a transformation from an alto-"a" sound ; to an alto-"i" sound. ; Values common to all of the formants. kfund init 261.659 koct init 0 kris init 0.003 kdur init 0.02 kdec init 0.007 iolaps = 100 ifna = 1 ifnb = 2 itotdur = p3 ; First formant. k1amp = ampdb(0) k1form line 800, p3, 350 k1band line 80, p3, 50 ; Second formant. k2amp line ampdb(-4), p3, ampdb(-20) k2form line 1150, p3, 1700 k2band line 90, p3, 100 ; Third formant. k3amp line ampdb(-20), p3, ampdb(-30) k3form line 2800, p3, 2700 k3band init 120 ; Fourth formant. k4amp init ampdb(-36) k4form line 3500, p3, 3700 k4band line 130, p3, 150 ; Fifth formant. k5amp init ampdb(-60) k5form init 4950 k5band line 140, p3, 200 a1 fof k1amp, kfund, k1form, koct, k1band, kris, \ kdur, kdec, iolaps, ifna, ifnb, itotdur a2 fof k2amp, kfund, k2form, koct, k2band, kris, \ kdur, kdec, iolaps, ifna, ifnb, itotdur a3 fof k3amp, kfund, k3form, koct, k3band, kris, \ kdur, kdec, iolaps, ifna, ifnb, itotdur a4 fof k4amp, kfund, k4form, koct, k4band, kris, \ kdur, kdec, iolaps, ifna, ifnb, itotdur a5 fof k5amp, kfund, k5form, koct, k5band, kris, \ kdur, kdec, iolaps, ifna, ifnb, itotdur ; Combine all of the formants together asig sum (a1+a2+a3+a4+a5) * 13000 outs asig, asig endin ; sine wave f 1 0 4096 10 1 ; sigmoid wave f 2 0 1024 19 0.5 0.5 270 0.5 i 1 0 1 i 1 2 5 ; same but slower e manual_src~dfsg/examples/ptablei.csd0000644000000000000000000000227012262561504016677 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tablei.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 ;generate new values every time the instr is played instr 1 ifn = p4 isize = p5 ithresh = 0.5 itemp ftgen ifn, 0, isize, 21, 2 iwrite_value = 0 i_index = 0 loop_start: iread_value ptablei i_index, ifn if iread_value > ithresh then iwrite_value = 1 else iwrite_value = -1 endif ptableiw iwrite_value, i_index, ifn loop_lt i_index, 1, isize, loop_start turnoff endin instr 2 ifn = p4 isize = ftlen(ifn) prints "Index\tValue\n" i_index = 0 loop_start: ivalue tablei i_index, ifn prints "%d:\t%f\n", i_index, ivalue loop_lt i_index, 1, isize, loop_start ;read table 1 with our index aout oscili .5, 100, ifn ;use table to play the polypulse outs aout, aout endin i 1 0 1 100 16 i 2 0 2 100 e manual_src~dfsg/examples/floor.csd0000644000000000000000000000116212262561504016377 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o floor.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idiv init 1 loop: inumber = 9 i1 = inumber / idiv ifl = floor(i1) print inumber, idiv, ifl ;print number / idiv = result using floor idiv = idiv + 1 if (idiv <= 10) igoto loop endin i 1 0 0 e manual_src~dfsg/examples/tonex.csd0000644000000000000000000000122312262561504016411 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tonex.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 asig diskin2 "beats.wav", 1 outs asig, asig endin instr 2 kton line 10000, p3, 0 ;all the way down to 0 Hz asig diskin2 "beats.wav", 1 asig tonex asig, kton, 8 ;8 filters outs asig, asig endin i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/event_i.csd0000644000000000000000000000162012262561504016706 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o event_i.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 gisine ftgen 0, 0, 2^10, 10, 1 instr 1 ;master instrument ininstr = 10 ;number of called instances indx = 0 loop: ipan random 0, 1 ifreq random 100, 1000 iamp = 1/ininstr event_i "i", 10, 0, p3, iamp, ifreq, ipan loop_lt indx, 1, ininstr, loop endin instr 10 print p4, p5, p6 ipeak random 0, 1 ;where is the envelope peak asig poscil3 p4, p5, gisine aenv transeg 0, p3*ipeak, 6, 1, p3-p3*ipeak, -6, 0 aL,aR pan2 asig*aenv, p6 outs aL, aR endin i1 0 10 i1 8 10 i1 16 15 e manual_src~dfsg/examples/wii.csd0000644000000000000000000000204112262561504016043 0ustar rootroot -+rtaudio=alsa -o dac:hw:0 nchnls = 2 ksmps = 400 #define WII_B #3# #define WII_A #4# #define WII_R_A #304# #define WII_PITCH #20# #define WII_ROLL #21# #define WII_BATTERY #27# #define WII_RUMBLE #3# #define WII_SET_LEDS #4# gkcnt init 1 instr 1 i1 wiiconnect 3,1 wiirange $WII_PITCH., -20, 0 kb wiidata $WII_BATTERY. kt wiidata $WII_B. ka wiidata $WII_A. kra wiidata $WII_R_A. gka wiidata $WII_PITCH. gkp wiidata $WII_ROLL. ; If the B (trigger) button is pressed then activate a note if (kt==0) goto ee event "i", 2, 0, 5 gkcnt = gkcnt + 1 wiisend $WII_SET_LEDS., gkcnt ee: if (ka==0) goto ff wiisend $WII_RUMBLE., 1 ff: if (kra==0) goto gg wiisend $WII_RUMBLE., 0 gg: printk2 kb endin instr 2 a1 oscil ampdbfs(gka), 440+gkp, 1 outs a1, a1 endin f1 0 4096 10 1 i1 0 300 manual_src~dfsg/examples/moogvcf2.csd0000644000000000000000000000115012262561504016775 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o moogvcf2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aout diskin2 "beats.wav", 1, 0, 1 kfco line 100, p3, 10000 ;filter-cutoff krez init p4 asig moogvcf2 aout, kfco, krez outs asig, asig endin i 1 0 4 .1 i 1 + 4 .6 i 1 + 4 .9 e manual_src~dfsg/examples/outvalue.csd0000644000000000000000000000072112262561504017122 0ustar rootroot ;run this example in CsoundQt, a Csound editor that provides widgets ;make the Widgets-panel visible, by clicking the Widgets symbol in the menu or pressing (Alt+1). sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; written by Alex Hofmann instr 1 ;move fader kMoveUp linseg 0, 3, 1, 1, 1, 0.5, 0 outvalue "movefader", kMoveUp endin i 1 0 5 e manual_src~dfsg/examples/pset-midi.csd0000644000000000000000000000170612262561504017155 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 --midi-key-oct=4 --midi-velocity=5 ;;;realtime audio out and virtual midi keyboard ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pset-midi.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 pset 0, 0, 3600, 0, 0, 0 iinstrument = p1 istarttime = p2 iattack = 0.005 isustain = p3 irelease = 0.06 p3 = isustain + iattack + irelease kdamping linsegr 0.0, iattack, 1.0, isustain, 1.0, irelease, 0.0 ioctave = p4 ifrequency = cpsoct(ioctave) iamplitude = p5*.15 ;lower volume print p1, p2, p3, p4, p5 asig STKBandedWG ifrequency, iamplitude outs asig, asig endin f 0 60 ; runs 69 seconds e manual_src~dfsg/examples/schedwhen.csd0000644000000000000000000000140112262561504017222 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o schedwhen.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kmtr metro 100 ;produce 100 triggers per second schedwhen kmtr, 2, 1, .3 ;but schedwhen plays instr. 2 only once endin instr 2 aenv linseg 0, p3*.1, 1, p3*.3, 1, p3*.6, 0 ;envelope a1 poscil .3*aenv, 1000, 1 outs a1, a1 endin f 1 0 16384 10 1 ;sine i 1 0 3 i 1 3 5 e manual_src~dfsg/examples/poisson.csd0000644000000000000000000000151312262561504016750 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o poisson.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 ksmps = 441 ;ksmps set deliberately high to have few k-periods per second nchnls = 1 ; Instrument #1. instr 1 ; Generates a random number in a poisson distribution. ; klambda = 1 i1 poisson 1 print i1 endin instr 2 kres poisson p4 printk (ksmps/sr),kres ;prints every k-period endin ; Play Instrument #1 for one second. i 1 0 1 i 2 1 0.2 0.5 i 2 2 0.2 4 ;average 4 events per k-period i 2 3 0.2 20 ;average 20 events per k-period e manual_src~dfsg/examples/sfplay3m.csd0000644000000000000000000000171312262561504017016 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfplay3m.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf sfload "07AcousticGuitar.sf2" sfplist gisf sfpassign 10, gisf instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/7000 ;scale amplitude kfreq init 1 ;do not change freq from sf aout sfplay3m ivel, inum, kamp*ivel, kfreq, 10 ;preset index = 10 outs aout, aout endin f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e manual_src~dfsg/examples/log10.csd0000644000000000000000000000102212262561504016173 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o log10.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = log10(8) print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/taninv.csd0000644000000000000000000000104512262561504016555 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o taninv.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 0.5 i1 = taninv(irad) print i1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/timeinstk.csd0000644000000000000000000000115512262561504017267 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o timeinstk.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the value from timeinstk every half-second. k1 timeinstk printks "k1 = %f samples\\n", 0.5, k1 endin ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/scans.csd0000644000000000000000000000226412262561504016371 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o scans.wav -W ;;; for file output any platform sr = 44100 ksmps = 128 nchnls = 1 instr 1 a0 = 0 ; scanu init, irate, ifnvel, ifnmass, ifnstif, ifncentr, ifndamp, kmass, kstif, kcentr, kdamp, ileft, iright, kpos, kstrngth, ain, idisp, id scanu 1, .01, 6, 2, 3, 4, 5, 2, .1, .1, -.01, .1, .5, 0, 0, a0, 1, 2 ;ar scans kamp, kfreq, ifntraj, id a1 scans ampdb(p4), cpspch(p5), 7, 2 out a1 endin ; Initial condition f1 0 128 7 0 64 1 64 0 ; Masses f2 0 128 -7 1 128 1 ; Spring matrices f3 0 16384 -23 "string-128.matrix" ; Centering force f4 0 128 -7 0 128 2 ; Damping f5 0 128 -7 1 128 1 ; Initial velocity f6 0 128 -7 0 128 0 ; Trajectories f7 0 128 -5 .001 128 128 ; Note list i1 0 10 86 6.00 i1 11 14 86 7.00 i1 15 20 86 5.00 e manual_src~dfsg/examples/pvsftr.csd0000644000000000000000000000206612262561504016606 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsftr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifil ftgen 0, 0, 0, -1, "fox.wav", 0, 0, 1 instr 1 ifftsize = 1024 ;fft size ioverlap = 256 ;overlap knewamp = 0 ;new value for amplitudes ;create fsig stream from function table fsrc pvstanal 1, 1, 1, gifil, 0, 0, 0, ifftsize, ioverlap, 0 ifn ftgen 0, 0, ifftsize/2, 2, 0 ;create empty function table kflag pvsftw fsrc,ifn ;export amps to table ;overwrite the first 10 bins each time the table has been filled new if kflag == 1 then kndx = 0 kmaxbin = 10 loop: tablew knewamp, kndx, ifn loop_le kndx, 1, kmaxbin, loop endif pvsftr fsrc,ifn ;read modified data back to fsrc aout pvsynth fsrc ;and resynth outs aout, aout endin i 1 0 4 e manual_src~dfsg/examples/strcat.csd0000644000000000000000000000103012262561504016550 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o strcat.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 Sname = "beats" Sname strcat Sname, ".wav" asig soundin Sname outs asig, asig endin i 1 0 2 e manual_src~dfsg/examples/fdl.txt0000644000000000000000000004766212262561504016110 0ustar rootroot GNU Free Documentation License Version 1.2, November 2002 Copyright (C) 2000,2001,2002 Free Software Foundation, Inc. 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA Everyone is permitted to copy and distribute verbatim copies of this license document, but changing it is not allowed. 0. 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If your document contains nontrivial examples of program code, we recommend releasing these examples in parallel under your choice of free software license, such as the GNU General Public License, to permit their use in free software. manual_src~dfsg/examples/joystick-2.csd0000644000000000000000000000317012262561504017255 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o joystick-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 ;0dbfs = 1 instr 1 kmask joystick 0, 1 kaxes init 0 kbuttons init 0 kx0 init 0 ; first two entries are # of axes and # of buttons, ky0 init 0 ; then axes, then buttons vtabk 0, 1, kaxes, kbuttons, kx0, ky0 kidx = 2+kaxes buttons: kcheck = kmask & 1< 0 kgoto noexit turnoff noexit: endin f1 0 32 7 0 7 0 ; will hold the joystick data i1 0 60000 e manual_src~dfsg/examples/syncgrain.csd0000644000000000000000000000137712262561504017263 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o syncgrain.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 iolaps = 2 igrsize = 0.04 ifreq = iolaps/igrsize ips = 1/iolaps istr = .3 /* timescale */ ipitch = p4 /* pitchscale */ asig syncgrain 1, ifreq, ipitch, igrsize, ips*istr, 1, 2, iolaps outs asig, asig endin f1 0 0 1 "fox.wav" 0 0 0 ;deferred table f2 0 8192 20 2 1 i1 0 5 1 i1 + 5 4 i1 + 5 .8 e manual_src~dfsg/examples/pchoct.csd0000644000000000000000000000116312262561504016537 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pchoct.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Convert an octave-point-decimal value into a ; pitch-class value. ioct = 8.75 ipch = pchoct(ioct) print ipch endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/gainslider.csd0000644000000000000000000000114412262561504017377 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in Silent -odac -iadc -d ;;;realtime output sr = 48000 ksmps = 100 nchnls = 2 instr 1 ; gainslider test ; uncomment for realtime midi ;kmod ctrl7 1, 1, 0, 127 ; uncomment for non realtime km0d phasor 1/10 kmod scale km0d, 127, 0 kout gainslider kmod printks "kmod = %f kout = %f\\n", 0.1, kmod, kout aout diskin2 "fox.wav", 1, 0, 1 aout = aout*kout outs aout, aout endin i1 0 30 e manual_src~dfsg/examples/gen28.csd0000644000000000000000000000217012262561504016201 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen28.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 4 ga1 init 0 ga2 init 0 ga3 init 0 ga4 init 0 instr 1 ;uses GEN28 file "move", as found in /manual/examples kx init 0 ky init 0 ktime line 0, 5, 5 ;same time as in table 1 (="move") asig diskin2 "beats.wav", 1, 0, 1 ;sound source is looped a1, a2, a3, a4 space asig, 1, ktime, .1, kx, ky ;use table 1 = GEN28 ar1, ar2, ar3, ar4 spsend ;send to reverb ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 a3 reverb2 ga3, 2.5, .5 a4 reverb2 ga4, 2.5, .5 outq a1, a2, a3, a4 ga1=0 ga2=0 ga3=0 ga4=0 endin f1 0 0 28 "move" i1 0 5 ;same time as ktime i 99 0 10 ;keep reverb active e manual_src~dfsg/examples/dssictls.csd0000644000000000000000000000261012262561504017105 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dssictls.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 gihandle dssiinit "caps.so", 1, 1 ; = equaliser and gaoutl init 0 ; verbose about all ports gaoutr init 0 instr 1 ; activate DSSI dssiactivate gihandle, 1 endin instr 2 ain1 diskin2 "beats.wav", 1,0,1 ; loop gaoutl = gaoutl+(ain1*.1) ; temper input gaoutr = gaoutr+(ain1*.1) endin instr 3 dssictls gihandle, 2, -48, 1 ; 31 Hz range -48 to 24 dssictls gihandle, 3, -48, 1 ; 63 Hz range -48 to 24 dssictls gihandle, 4, -48, 1 ; 125 Hz range -48 to 24 dssictls gihandle, 5, 20, 1 ; 250 Hz range -48 to 24 dssictls gihandle, 6, -48, 1 ; 500 Hz range -48 to 24 dssictls gihandle, 7, -48, 1 ; 1 kHz Hz range -48 to 24 dssictls gihandle, 8, -48, 1 ; 2 kHz range -48 to 24 dssictls gihandle, 9, 24, 1 ; 4 kHz range -48 to 24 dssictls gihandle, 10, 24, 1 ; 8 kHz range -48 to 24 dssictls gihandle, 11, 24, 1 ; 16 kHz range -48 to 24 endin instr 4 aout1, aout2 dssiaudio gihandle, gaoutl, gaoutr ;get beats.wav, mono out outs aout1,aout2 gaoutl = 0 gaoutr = 0 endin i 1 0 20 i 2 1 20 i 3 1 20 i 4 0 20 e manual_src~dfsg/examples/bitshift.csd0000644000000000000000000000154012262561504017072 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in ;-odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -o bitshift.wav -W --nosound ;;; for file output any platform sr = 44100 ksmps = 128 nchnls = 2 instr 1 ;bit shift right ival = p4>>p5 printf_i "%i>>%i = %i\n", 1, p4, p5, ival endin instr 2 ;bit shift left ival = p4< i 1 0 0.1 2 1 i 1 + . 3 1 i 1 + . 7 2 i 1 + . 16 1 i 1 + . 16 2 i 1 + . 16 3 i 2 5 0.1 1 1 i 2 + . 1 2 i 2 + . 1 3 i 2 + . 1 4 i 2 + . 2 1 i 2 + . 2 2 i 2 + . 2 3 i 2 + . 3 2 e manual_src~dfsg/examples/vdelayxwq.csd0000644000000000000000000000203512262561504017302 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelayxwq.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 ims = .5 ;maximum delay time in seconds iws = 1024 ;best quality adl = .5 ;delay time aout1 diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav aout2 diskin2 "fox.wav", 1, 0, 1 ;loop fox.wav aout3 diskin2 "Church.wav", 1, 0, 1 ;loop Church.wav aout4 diskin2 "flute.aiff", 1, 0, 1 ;loop flute.aiff a2 poscil3 .2, .1, 1 ;make an LFO, 1 cycle per 2 seconds adl = a2 + ims/2 ;offset the LFO so that it is positive aout1, aout2, aout3, aout4 vdelayxwq aout1, aout2, aout3, aout4, adl, ims, iws ;use the LFO to control delay time outq aout1, aout2, aout3, aout4 endin f1 0 8192 10 1 i 1 0 10 e manual_src~dfsg/examples/vaget.csd0000644000000000000000000000200612262561504016362 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o avarget.wav -W ;;; for file output any platform sr=44100 ksmps=16 nchnls=2 instr 1 ; Sqrt Signal ifreq = (p4 > 15 ? p4 : cpspch(p4)) iamp = ampdb(p5) aout init 0 ksampnum init 0 kenv linseg 0, p3 * .5, 1, p3 * .5, 0 aout1 vco2 1, ifreq aout2 vco2 .5, ifreq * 2 aout3 vco2 .2, ifreq * 4 aout sum aout1, aout2, aout3 ;Take Sqrt of signal, checking for negatives kcount = 0 loopStart: kval vaget kcount,aout if (kval > .0) then kval = sqrt(kval) elseif (kval < 0) then kval = sqrt(-kval) * -1 else kval = 0 endif vaset kval, kcount,aout loop_lt kcount, 1, ksmps, loopStart aout = aout * kenv aout moogladder aout, 8000, .1 aout = aout * iamp outs aout, aout endin i1 0.0 2 440 80 e manual_src~dfsg/examples/transegr.csd0000644000000000000000000000132512262561504017104 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o transegr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .2 ; st,dur1,typ1,val,dur2,typ2,end kenv transegr 0, .2, 2, .5, 1, - 3, 0 asig pluck kenv*iamp, icps, icps, 1, 1 outs asig, asig endin f1 0 4096 10 1 ;sine f0 30 ;runs 30 seconds e manual_src~dfsg/examples/x.csd0000644000000000000000000000144312262561504015527 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o x.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .8*aenv, p4, 1 outs asig, asig endin f 1 0 16384 10 1 ;sine wave s ;first section i 1 0 2 110 i 1 3 2 220 i 1 6 2 440 i 1 9 2 880 s ;second section x ;skip the rest i 1 0 2 110 ;of this section i 1 3 2 220 i 1 6 2 440 i 1 9 2 880 s ;but continue with this one i 1 0 2 880 i 1 3 2 440 i 1 6 2 220 i 1 9 2 110 e manual_src~dfsg/examples/mary.wav0000644000000000000000000113316012262561504016257 0ustar rootrootRIFFhWAVEfmt DXPEAKZ<5">data,   #'*.269=EHKNQTVX[\efhijjkkkkqponmkifdada]ZVQMID?>83-'" ~{xurpnfddccccdegbdgjmqty} &.7?GOW_jry{ri_VLB8/%}tk^VNF?82,'"   !(/7?HQ[dov"/<HUap} %(*,,-,+)'+'# zl]O@1"}oaPC6* ,;IXhw1BUfw+6AJS\jqw|yrkb_UJ?2% ubO;)ydQ?- vnf_SNJGEDDEGJGLQX`ir} 1BVk,CXn&7HXgvueYG4! u]E-lU>'qaRD7+ )7FVgy !9Rk ;Uo  7Mcx#3:@EHKKKIFID=5+!~hQ9! hL/hL0|eP;''7Md|8Vt ,Kk9Uo(9I_mysl]L:&hK.rQ0jJ(rW<"yk^KA81,(&%&)&+3t 9e9Rhz}iS9#k>x@OJ~8j%XVl6pE|fSD7.(&'++6DUiGsw=g*=MZchjhi`SC/vKH Q 7V n 6On&Yf/jBskfdgmw@j)b\.u ZG7(x`< K1c6UpjT3\(~?r*GZclu%:XJj:nR:& *B]xCw%c/yaXWZ\[N.t<}Kv!5DPX[[[RE:% ~U%G|3KTM?0|#vx%Cy6\0vgTNKMS]l{>nN`NO\u0MclhOS#Sx Y.b$X rp]=zR+l VTeTT. ,UX#odp0X&],]&FN;QFijCBt,5}X"Ou+PzCz` fG nD&O~Z63L}V6nT:[&ED[ + Y y o I  Am!sS>Vhw 3Jh.l`z8`3CqV8?jD* xl[C LU7 }  H l  _ 5  X  |'qAfyyob[X`qI6MZ0 #Q<$y;~f^]i|1F[ceY?> 0 u " N t u K  l % ( k+=<6$Z:s`VXm*in)yM%:f P?g?.(4Nl#Lz4PcliV, H 3 s  > ^ v z b H ! ? 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"$%')*,-./012334444444332210/.-+*)'%#!  !#%&()+,-./011223333322210//.-+*('%$"    "#%&')*+,-../000111000//.--+*)('%$"   !"$%'()*+,-..//000000//..-,+*)('%$"   !"#%&'()*+,,--.......--,,+*)('&%#"    "#$%&'())*++++,+++++*))('&%$#"!   !"#$%&'(())))***)))((''&%$#"!   !"#$$%%&&&''''&&&&%%$#"!    !!""#########"""!    !!!!""""!!!!                                                                                                               manual_src~dfsg/examples/ooo.aiff0000644000000000000000000000035612262561504016212 0ustar rootrootFORMAIFFCOMM\@ DSSNDa.d;;[/[TIKB ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbapgmove.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 vbaplsinit 2, 4, 0, 90, 180, 270 instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav k0,k1,k2,k3 vbapgmove p3, 1, 2, 310, 180 ;change movement of soundsource in outq k0*asig,k1*asig,k2*asig,k3*asig ;the rear speakers endin i 1 0 5 e manual_src~dfsg/examples/veloc.csd0000644000000000000000000000110312262561504016361 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisine ftgen 0, 0, 1024, 10, 1 instr 1 ivel veloc 0, 1 ;scale 0 - 1 print ivel ;print velocity asig poscil .5*ivel, 220, gisine ;and use it as amplitude outs asig, asig endin f 0 30 ;runs 30 seconds e manual_src~dfsg/examples/wgbow.csd0000644000000000000000000000141312262561504016402 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o wgbow.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kpres = p4 ;pressure value krat = p5 ;position along string kvibf = 6.12723 kvib linseg 0, 0.5, 0, 1, 1, p3-0.5, 1 ; amplitude envelope for the vibrato. kvamp = kvib * 0.01 asig wgbow .7, 55, kpres, krat, kvibf, kvamp, 1 outs asig, asig endin f 1 0 2048 10 1 ;sine wave i 1 0 3 3 0.127236 i 1 + 3 5 0.127236 i 1 + 3 5 0.23 e manual_src~dfsg/examples/toot2.csd0000644000000000000000000000075612262561504016335 0ustar rootroot -odac instr 2 a1 oscil p4, p5, 1 ; p4=amp out a1 ; p5=freq endin f1 0 4096 10 1 ; sine wave ;ins strt dur amp(p4) freq(p5) i2 0 1 2000 880 i2 1.5 1 4000 440 i2 3 1 8000 523.251 i2 4.5 1 16000 659.255 i2 6 1 32000 783.991 e manual_src~dfsg/examples/STKSaxofony.csd0000644000000000000000000000140712262561504017450 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKSaxofony.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifreq = p4 kv1 = p5 ;reed stiffness kv3 line p6, p3, p7 ;blow position kv6 line 0, p3, 127 ;depth of low-frequency oscillator asig STKSaxofony cpspch(p4), 1, 2, kv1, 4, 100, 26, 70, 11, kv3, 1, kv6, 29, 100 asig = asig * .5 ;too loud outs asig, asig endin i 1 0 3 6.00 30 100 10 i 1 + . 8.00 30 100 100 i 1 + . 7.00 90 127 30 e manual_src~dfsg/examples/cosseg.csd0000644000000000000000000000204112262561504016536 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o linseg.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kcps = cpspch(p4) kenv cosseg 0, 0.25, 1, 0.75, 0 ; together = 1 sec asig poscil kenv, kcps, giSine outs asig, asig endin instr 2 ; scaling to duration kcps = cpspch(p4) kenv cosseg 0, p3*0.25, 1, p3*0.75, 0 asig poscil kenv, kcps, giSine outs asig, asig endin instr 3 ; with negative value kcps = cpspch(p4) aenv cosseg 0, 0.1, 1, 0.5, -0.9, 0.4, 0 asig poscil aenv, kcps, giSine outs asig, asig endin i 1 0 1 7.00 ; = 1 sec, p3 fits exactly i 1 2 2 7.00 ; = 2 sec, p3 truncated at 1 sec i 2 4 1 7.00 ; scales to duration i 2 6 2 7.00 ; of p3 i 3 9 2 7.00 e manual_src~dfsg/examples/vexp.csd0000644000000000000000000000146612262561504016247 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vexp ifn1, ival, ielements, idstoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e manual_src~dfsg/examples/mpulse.csd0000644000000000000000000000136512262561504016570 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o mpulse.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 gkfreq init 0.1 instr 1 kamp = 10000 a1 mpulse kamp, gkfreq out a1 endin instr 2 ; Assign the value of p4 to gkfreq gkfreq init p4 endin ; Play Instrument #1 for one second. i 1 0 11 i 2 2 1 0.05 i 2 4 1 0.01 i 2 6 1 0.005 ; only last notes are audible i 2 8 1 0.003 i 2 10 1 0.002 e manual_src~dfsg/examples/minaccum.csd0000644000000000000000000000161012262561504017050 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o minaccum.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;saw asig vco2 .2, p4 outs asig, asig gasaw = asig endin instr 2 ;sine aout poscil .3, p4, 1 outs aout, aout gasin = aout endin instr 10 accum init 0 minaccum accum, gasaw + gasin ;saw and sine accumulated outs accum, accum clear accum endin f 1 0 4096 10 1 i 1 0 7 330 i 2 3 3 440 i 1 10 7 330 ;same notes but without minaccum, for comparison i 2 13 3 440 i 10 0 6 ;accumulation note stops after 6 seconds e manual_src~dfsg/examples/ftchnls.csd0000644000000000000000000000145212262561504016721 0ustar rootroot ; Select audio/midi flags here according to platform ;-odac ;;;realtime audio out -iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftchnls.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ichnls = ftchnls(p4) print ichnls if (ichnls == 1) then asigL loscil3 .8, 4, p4 asigR = asigL elseif (ichnls == 2) then asigL, asigR loscil3 .8, 4, p4 ;safety precaution if not mono or stereo else asigL = 0 asigR = 0 endif outs asigL, asigR endin f 1 0 0 1 "mary.wav" 0 0 0 f 2 0 0 1 "kickroll.wav" 0 0 0 i 1 0 3 1 ;mono file i 1 + 2 2 ;stereo file e manual_src~dfsg/examples/urandom_krate.csd0000644000000000000000000000121412262561504020107 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd31_krate.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create random numbers at k-rate in the range -1 to 1 ; with a uniform distribution. k1 urandom printks "k1=%f\\n", 0.1, k1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/hsboscil.csd0000644000000000000000000000153012262561504017063 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o hsboscil.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; synth waveform giwave ftgen 1, 0, 1024, 10, 1, 1, 1, 1 ; blending window giblend ftgen 2, 0, 1024, -19, 1, 0.5, 270, 0.5 instr 1 ; produces Risset's glissando. kamp = .4 kbrite = 0.3 ibasfreq = 200 ioctcnt = 5 ; Change ktone linearly from 0 to 1, ; over the period defined by p3. ktone line 0, p3, 1 asig hsboscil kamp, ktone, kbrite, ibasfreq, giwave, giblend, ioctcnt outs asig, asig endin i 1 0 10 e manual_src~dfsg/examples/sfplay3.csd0000644000000000000000000000323312262561504016640 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfplay3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gitwelve ftgen 1, 0, 16, -2, 12, 2, 440, 69, 1, 16/15, 9/8, 6/5, 5/4, 4/3, 7/5, 3/2, 8/5, 5/3, 9/5, 15/8, 2 givife ftgen 2, 0, 16, -2, 5, 2, 261.659, 60, 1, 1.1486, 1.3195, 1.5157, 1.7411, 2.00 giSF sfload "01hpschd.sf2" sfplist giSF gipre sfpreset 0, 0, giSF, 0 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey = p4 ivel = p5 aenv linsegr 1, 1, 1, 1, 0 ;envelope icps cpstuni ikey, gitwelve ;12 tones per octave iamp = 0.0004 ;scale amplitude iamp = iamp * ivel * 1/128 ;make velocity-dependent aL, aR sfplay3 ivel, ikey, iamp, icps, gipre, 1 aL = aL * aenv aR = aR * aenv outs aL, aR endin instr 2 mididefault 60, p3 midinoteonkey p4, p5 ikey = p4 ivel = p5 aenv linsegr 1, 1, 1, 1, 0 ;envelope icps cpstuni ikey, givife ;5 tones per octave iamp = 0.0004 ;scale amplitude iamp = iamp * ivel * 1/128 ;make velocity-dependent aL, aR sfplay3 ivel, ikey, iamp, icps, gipre, 1 aL = aL * aenv aR = aR * aenv outs aL, aR endin f0 60 ;play for 60 seconds ;instr.1 using ftable 1 i1 0 1 60 100 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 40 ;instr.2 using ftable 2 i2 5 1 60 100 i2 + 1 62 < i2 + 1 65 < i2 + 1 69 40 e manual_src~dfsg/examples/gen18.csd0000644000000000000000000000222712262561504016203 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen18.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ilen = ftlen(ifn) aphase phasor 220 asig tablei aphase*ilen, ifn outs asig, asig endin f 1 0 4096 10 1 ;sine f 2 0 4096 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ;sawtooth f 3 0 4096 11 10 5 2 ;cosine partials f 11 0 8192 18 1 1 0 4095 2 1 4096 8191 ;sine+sawtooth f 12 0 8192 18 1 1 0 4095 3 1 4096 8191 ;sine+cosine partials f 13 0 1024 18 1 0.7 0 767 3 0.7 512 1023 ;sine+cosine partials overlapped, shorter table i 1 0 2 2 ;play sawtooth i 1 + 2 3 ;then cosine partials i 1 5 2 11 ;now sine+sawtooth i 1 + 2 12 ;and sine+cosine partials i 1 + 2 13 ;and sine+cosine partials overlapped, shorter table e manual_src~dfsg/examples/gen14.csd0000644000000000000000000000321712262561504016177 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen14.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;after the example from The Csound Book, page 83 instr 1 ;compare results from GEN13 & GEN14 iwshpfn = p6 inrmfn = p7 aswp linseg 0.01, p3*.5, .49, p3*.5, 0.01 ;index sweep function aindex poscil aswp, p5, 2 ;sound to waveshape atable tablei aindex, iwshpfn, 1, .5 ;waveshape index anrm tablei aswp*2, inrmfn, 1 ;normalization aenv linen p4, .01, p3, .02 ;amplitude envelope asig = (atable*anrm)*aenv ;normalize and impose envelope asig dcblock2 asig ;get rid of DC outs asig, asig endin f 2 0 8192 10 1 ;sine wave f 28 0 4097 13 1 1 1 0 .8 0 .5 0 .2 ;waveshaping function: GEN13 - odd harmonics f 280 0 2049 4 28 1 ;normalization function for f28 f 29 0 4097 14 1 1 1 0 .8 0 .5 0 .2 ;waveshaping function: GEN14 - same harmonics f 290 0 2049 4 29 1 ;normalization function for f29 f 30 0 4097 13 1 1 0 1 0 .6 0 .4 0 .1 ;waveshaping function: GEN13 - even harmonics f 301 0 2049 4 30 1 ;normalization function for f30 f 31 0 4097 14 1 1 0 1 0 .6 0 .4 0 .1 ;waveshaping function: GEN13 - even harmonics f 310 0 2049 4 31 1 ;normalization function for 31 s i1 0 3 .7 440 28 280 i1 4 . .7 . 29 290 i1 8 . .7 . 30 301 i1 12 3 .7 . 31 310 e manual_src~dfsg/examples/urandom.csd0000644000000000000000000000134212262561504016723 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd31.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create random numbers at a-rate in the range -2 to 2 aur urandom -2, 2 ; Use the random numbers to choose a frequency. afreq = aur * 500 + 100 a1 oscil 30000, afreq, 1 out a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/seqtime2.csd0000644000000000000000000000217012262561504017007 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o seqtime2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gitabMap2 ftgen 57,0,512,-2, 1,1/4,1/4,1/8,1/8,1/8,1/8,1/4,1/4,.5,1/4,1/4,1/16,1/16,1/16,1/16,1/16,1/16,1/16,1/16 gisine ftgen 1,0,512,10, 1 instr 1 ktrigin metro .333333333333 ktrig2 metro 1 schedkwhen ktrig2, 0,0, 2, 0, .1 ; just to set the metronome! kspeed init 1 ; ktime_unit, kstart, kloop, initndx, kfn_times ktrig seqtime2 ktrigin, kspeed, 0, 20, 2, gitabMap2 ;ktrig seqtime kspeed, 0, 20, 0, gitabMap2 ; try with seqtime too... schedkwhen ktrig, 0, 0, 3, 0, ktrig ; the duration is got from seqtime2 output! endin instr 2 a1 line 1,p3,0 aout oscili 0.7*a1,500,gisine outs1 aout endin instr 3 a1 line 1,p3,0 aout oscili 0.7*a1,1000,gisine outs2 aout endin i1 0 20 ;f0 3600 manual_src~dfsg/examples/gen06.csd0000644000000000000000000000163412262561504016201 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen06.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ;choose between tables kcps init 1/p3 ;create index over duration of note. kndx phasor kcps ixmode = 1 kval table kndx, ifn, ixmode ;normalize mode kfreq = kval * 30 ;scale frequency to emphasixe effect asig poscil .7, 220 + kfreq, 1 ;add to frequency outs asig, asig endin f 1 0 16384 10 1 ;sine wave. f 2 0 513 6 1 128 -1 128 1 64 -.5 64 .5 16 -.5 8 1 16 -.5 8 1 16 -.5 84 1 16 -.5 8 .1 16 -.1 17 0 f 3 0 513 6 0 128 0.5 128 1 128 0 129 -1 i 1 0 3 2 i 1 4 3 3 e manual_src~dfsg/examples/massign.csd0000644000000000000000000000275412262561504016727 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o massign.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "19Trumpet.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 56 massign 0,0 ;disable triggering of all instruments on all channels, but massign 12,10 ;assign instr. 10 to midi channel 12 massign 3,30 ;assign instr. 30 to midi channel 3 instr 10 ; soundfont only on midi channel 12 mididefault 60, p3 midinoteonkey p4, p5 ; in midi notes ikey init p4 ivel init p5 fluidNote giengine, 1, ikey, ivel endin instr 30 ; FM-oscilator only on midi channel 3 mididefault 60, p3 midinoteoncps p4, p5 ; in Hertz icps init p4 iamp init p5 iamp = iamp/127 kenv madsr 0.5, 0, 1, 0.5 asig foscil iamp*kenv, icps, 1, 1.414, 2, 1 outs asig, asig endin instr 99 ; output sound from fluidengine imvol init 7 aL, aR fluidOut giengine outs aL*imvol, aR*imvol endin ; sine f 1 0 16384 10 1 i 10 0 2 60 100 ;one note on the trumpet in midi and... i 30 2 2 220 80 ;one FM note in Hz i 99 0 60 ;stay active for 60 sec. e manual_src~dfsg/examples/loopseg.csd0000644000000000000000000000317712262561504016736 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loopseg.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq init p4 ; frequency of loop repetition ifrac = p5 ; frequency ratio of restart triggers ktrig metro kfreq * ifrac ; triggers to restart loop iphase = 0 ; initial phase ; loop of note values (some glissandi) ; val dur val dur etc... knote loopseg kfreq, ktrig, iphase, 40, 1, 40, 0, 43,1,43,0, 49,2,48,0, \ 47,1,47,0, 46,1,46,0, 46,1,47,0, 49,1,49,0, 43,1,43,0, 46,1,46,0, 40,1,39,0 ; loop of filter cutoff values (oct format). This loop, half speed of note loop. kcfoct loopseg kfreq*0.5, ktrig, iphase, 11,2,4,0, 12,1,4,0, 13,1,4,0, \ 11.5,3,4,0, 12.5,1,4,0, 13,2,4,0, 12.5,1,4,0 kenv linseg 0,0.01,1,p3-5.01,1,5,0 ioct = int((rnd(0.999)*4)-2) ; random value either -1, 0 or 1 asig vco2 0.2*kenv,cpsmidinn(knote)*octave(ioct),0 ; sawtooth asig moogladder asig,cpsoct(kcfoct),rnd(0.6) ; filter sawtooth aL,aR pan2 asig,rnd(1) ; random static pan location outs aL, aR endin ; 4 layers, each with a different frequency of loop repetition (p4), ; frequency ratio of restart triggers (p5) and pan position (p6). i 1 0 30 0.5 [11/19] i 1 6 30 0.25 [11/13] i 1 12 30 0.125 [11/16] i 1 18 30 1 [11/12] e manual_src~dfsg/examples/butterbr.csd0000644000000000000000000000123012262561504017103 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o butterbr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; White noise asig rand 0.5 outs asig, asig endin instr 2 ; filtered noise asig rand 0.7 abr butterbr asig, 3000, 2000 ;center frequency = 3000, bandwidth = +/- (2000)/2, so 2000-4000 outs abr, abr endin i 1 0 2 i 2 2.5 2 e manual_src~dfsg/examples/pvsgain.csd0000644000000000000000000000122712262561504016727 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsgain.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kgain = p4 asig diskin2 "beats.wav", 1 fsig pvsanal asig, 1024, 256, 1024, 1; analyse it ftps pvsgain fsig, kgain ; amplify it atps pvsynth ftps ; synthesise it outs atps, atps endin i1 0 2 .5 i1 + 2 1 e manual_src~dfsg/examples/foscili.csd0000644000000000000000000000126112262561504016706 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o foscili.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 440 kcar = 1 kmod = p4 kndx line 0, p3, 20 ;intensivy sidebands asig foscili .5, kcps, kcar, kmod, kndx, 1 outs asig, asig endin ; sine f 1 0 16384 10 1 i 1 0 9 .01 ;vibrato i 1 10 . 1 i 1 20 . 1.414 ;gong-ish i 1 30 5 2.05 ;with "beat" e manual_src~dfsg/examples/clfilt_lowpass.csd0000644000000000000000000000122312262561504020301 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o clfilt_lowpass.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; white noise asig rand 0.5 outs asig, asig endin instr 2 ; filtered noise asig rand 0.9 ; Lowpass filter signal asig with a ; 10-pole Butterworth at 500 Hz. a1 clfilt asig, 500, 0, 10 outs a1, a1 endin i 1 0 2 i 2 2 2 e manual_src~dfsg/examples/zakinit.csd0000644000000000000000000000245412262561504016734 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zakinit.wav -W ;;; for file output any platform sr = 44100 ksmps = 4410 nchnls = 1 ; Initialize the ZAK space. ; Create 3 a-rate variables and 5 k-rate variables. zakinit 2, 3 instr 1 ;a simple waveform. ; Generate a simple sine waveform. asin oscil 20000, 440, 1 ; Send the sine waveform to za variable #1. zaw asin, 1 endin instr 2 ;generates audio output. ; Read za variable #1. a1 zar 1 ; Generate audio output. out a1 ; Clear the za variables, get them ready for ; another pass. zacl 0, 3 endin instr 3 ;increments k-type channels k0 zkr 0 k1 zkr 1 k2 zkr 2 zkw k0+1, 0 zkw k1+5, 1 zkw k2+10, 2 endin instr 4 ;displays values from k-type channels k0 zkr 0 k1 zkr 1 k2 zkr 2 ; The total count for k0 is 30, since there are 10 ; control blocks per second and intruments 3 and 4 ; are on for 3 seconds. printf "k0 = %i\n",k0, k0 printf "k1 = %i\n",k1, k1 printf "k2 = %i\n",k2, k2 endin ; Table #1, a sine wave. f 1 0 16384 10 1 i 1 0 1 i 2 0 1 i 3 0 3 i 4 0 3 e manual_src~dfsg/examples/sfplist.csd0000644000000000000000000000174712262561504016753 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out, virtual midi in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfplist.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf sfload "sf_GMbank.sf2" sfplist gisf ;list all presets gir sfpreset 125, 3, gisf, 0 ;preset = Car Pass instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/6000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay3 ivel, inum, kamp, kfreq, gir outs a1, a2 endin f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e manual_src~dfsg/examples/sr.csd0000644000000000000000000000137312262561504015706 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ; -iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;use sr to find maximum harmonics ihar = int(sr/2/p4) ; maximum possible number of harmonics w/o aliasing prints "maximum number of harmonics = %d \\n", ihar kenv linen .5, 1, p3, .2 ; envelope asig buzz kenv, p4, ihar, 1 outs asig, asig endin f1 0 4096 10 1 ;sine wave i 1 0 3 100 ;different frequencies i 1 + 3 1000 i 1 + 3 10000 e manual_src~dfsg/examples/vpow-2.csd0000644000000000000000000000235412262561504016414 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vpow-2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "fox.wav", 1 ;soundfile fsrc pvsanal ain, 1024, 256, 1024, 1 ifn ftgen 0, 0, 1024/2, 2, 0 ;create empty function table for the 513 bins kflag pvsftw fsrc,ifn ;export only amplitudes to table kval line .001, p3, 1 ;start with big distortion, cahnge over note duration to clean sound kbin line p4, p3, p5 ;vary the bins vpow ifn, kval, kbin, 0 ;note that this operation is applied each k-cycle! ;vpow ifn, kval, kbin, 10 ;if you set kdstoffset to 10 it will affect bins 10+(kbin line p4, p3, p5) pvsftr fsrc,ifn ;read modified data back to fsrc aout pvsynth fsrc ;and resynth outs aout*p6, aout*p6 ;adjust volume to compensate endin i 1 0 4 100 100 .02 ;first 100 bins are affected i 1 + 4 10 10 .1 ;first 10 bins i 1 + 4 1 400 .05 ;sweep from bin 1 to 400 e manual_src~dfsg/examples/gen27.csd0000644000000000000000000000137212262561504016203 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen27.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 1, 0, 32768, 10, 1 gienv ftgen 2, 0, 1025, 27, 0, 0,200, 1, 400, -1, 513, 0 instr 1 kcps init 3/p3 ;play 3x over duration of note kndx phasor kcps ixmode = 1 ;normalize to 0-1 kval table kndx, gienv, ixmode kval = kval*100 ;scale 0-100 asig poscil 1, 220+kval, 1 ;add to 220 Hz outs asig, asig endin i 1 0 4 e manual_src~dfsg/examples/FLsetVal.csd0000644000000000000000000000154612262561504016744 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLsetVal.wav -W ;;; for file output any platform sr = 44100 kr = 441 ksmps = 100 nchnls = 1 ; By Andres Cabrera 2007 FLpanel "Frequency Slider", 340, 100, 50, 50 gkvalue1, gihandle1 FLslider "Slider A", 200, 5000, -1, 1, -1, 320, 20, 10, 10 gkvalue2, gihandle2 FLslider "Slider B", 200, 5000, -1, 1, -1, 320, 20, 10, 50 ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun ;Set the widget's initial value FLsetVal_i 300, gihandle1 instr 1 endin ; Function table that defines a single cycle ; of a sine wave. f 0 3600 e manual_src~dfsg/examples/sensekey.csd0000644000000000000000000000103212262561504017100 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o sensekey.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 k1 sensekey printk2 k1 endin ; Play Instrument #1 for thirty seconds. i 1 0 30 e manual_src~dfsg/examples/128,8-torusX0000755000000000000000000000624312262561504016476 0ustar rootroot 0 1 0 7 0 8 0 120 1 0 1 2 1 9 1 121 2 1 2 3 2 10 2 122 3 2 3 4 3 11 3 123 4 3 4 5 4 12 4 124 5 4 5 6 5 13 5 125 6 5 6 7 6 14 6 126 7 0 7 6 7 15 7 127 8 0 8 9 8 15 8 16 9 1 9 8 9 10 9 17 10 2 10 9 10 11 10 18 11 3 11 10 11 12 11 19 12 4 12 11 12 13 12 20 13 5 13 12 13 14 13 21 14 6 14 13 14 15 14 22 15 7 15 8 15 14 15 23 16 8 16 17 16 23 16 24 17 9 17 16 17 18 17 25 18 10 18 17 18 19 18 26 19 11 19 18 19 20 19 27 20 12 20 19 20 21 20 28 21 13 21 20 21 22 21 29 22 14 22 21 22 23 22 30 23 15 23 16 23 22 23 31 24 16 24 25 24 31 24 32 25 17 25 24 25 26 25 33 26 18 26 25 26 27 26 34 27 19 27 26 27 28 27 35 28 20 28 27 28 29 28 36 29 21 29 28 29 30 29 37 30 22 30 29 30 31 30 38 31 23 31 24 31 30 31 39 32 24 32 33 32 39 32 40 33 25 33 32 33 34 33 41 34 26 34 33 34 35 34 42 35 27 35 34 35 36 35 43 36 28 36 35 36 37 36 44 37 29 37 36 37 38 37 45 38 30 38 37 38 39 38 46 39 31 39 32 39 38 39 47 40 32 40 41 40 47 40 48 41 33 41 40 41 42 41 49 42 34 42 41 42 43 42 50 43 35 43 42 43 44 43 51 44 36 44 43 44 45 44 52 45 37 45 44 45 46 45 53 46 38 46 45 46 47 46 54 47 39 47 40 47 46 47 55 48 40 48 49 48 55 48 56 49 41 49 48 49 50 49 57 50 42 50 49 50 51 50 58 51 43 51 50 51 52 51 59 52 44 52 51 52 53 52 60 53 45 53 52 53 54 53 61 54 46 54 53 54 55 54 62 55 47 55 48 55 54 55 63 56 48 56 57 56 63 56 64 57 49 57 56 57 58 57 65 58 50 58 57 58 59 58 66 59 51 59 58 59 60 59 67 60 52 60 59 60 61 60 68 61 53 61 60 61 62 61 69 62 54 62 61 62 63 62 70 63 55 63 56 63 62 63 71 64 56 64 65 64 71 64 72 65 57 65 64 65 66 65 73 66 58 66 65 66 67 66 74 67 59 67 66 67 68 67 75 68 60 68 67 68 69 68 76 69 61 69 68 69 70 69 77 70 62 70 69 70 71 70 78 71 63 71 64 71 70 71 79 72 64 72 73 72 79 72 80 73 65 73 72 73 74 73 81 74 66 74 73 74 75 74 82 75 67 75 74 75 76 75 83 76 68 76 75 76 77 76 84 77 69 77 76 77 78 77 85 78 70 78 77 78 79 78 86 79 71 79 72 79 78 79 87 80 72 80 81 80 87 80 88 81 73 81 80 81 82 81 89 82 74 82 81 82 83 82 90 83 75 83 82 83 84 83 91 84 76 84 83 84 85 84 92 85 77 85 84 85 86 85 93 86 78 86 85 86 87 86 94 87 79 87 80 87 86 87 95 88 80 88 89 88 95 88 96 89 81 89 88 89 90 89 97 90 82 90 89 90 91 90 98 91 83 91 90 91 92 91 99 92 84 92 91 92 93 92 100 93 85 93 92 93 94 93 101 94 86 94 93 94 95 94 102 95 87 95 88 95 94 95 103 96 88 96 97 96 103 96 104 97 89 97 96 97 98 97 105 98 90 98 97 98 99 98 106 99 91 99 98 99 100 99 107 100 92 100 99 100 101 100 108 101 93 101 100 101 102 101 109 102 94 102 101 102 103 102 110 103 95 103 96 103 102 103 111 104 96 104 105 104 111 104 112 105 97 105 104 105 106 105 113 106 98 106 105 106 107 106 114 107 99 107 106 107 108 107 115 108 100 108 107 108 109 108 116 109 101 109 108 109 110 109 117 110 102 110 109 110 111 110 118 111 103 111 104 111 110 111 119 112 104 112 113 112 119 112 120 113 105 113 112 113 114 113 121 114 106 114 113 114 115 114 122 115 107 115 114 115 116 115 123 116 108 116 115 116 117 116 124 117 109 117 116 117 118 117 125 118 110 118 117 118 119 118 126 119 111 119 112 119 118 119 127 120 0 120 112 120 121 120 127 121 1 121 113 121 120 121 122 122 2 122 114 122 121 122 123 123 3 123 115 123 122 123 124 124 4 124 116 124 123 124 125 125 5 125 117 125 124 125 126 126 6 126 118 126 125 126 127 127 7 127 119 127 120 127 126 manual_src~dfsg/examples/gentanh.csd0000644000000000000000000000173212262561504016705 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gentanh.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aswp linseg 0.01, p3*.5, .49, p3*.5, 0.01 ;index sweep function aindex poscil aswp, 110, 1 ;sound to waveshape atable tablei aindex, p4, 1, .5 ;waveshape index aenv linen 0.8, .01, p3, .02 ;amplitude envelope asig = (atable*aenv)*p5 ;impose envelope and scale asig dcblock2 asig ;get rid of DC outs asig, asig endin f 1 0 8192 10 1 ;sine wave f 2 0 8192 "tanh" -100 100 0 ;symmetrical transfer fuction f 3 0 8192 "tanh" -10 10 0 ;symmetrical f 4 0 8192 "tanh" 0 10 0 ;not symmetrical i1 0 3 2 1 i1 + 3 3 1 i1 + 3 4 2 e manual_src~dfsg/examples/luaopcode.csd0000644000000000000000000000340212262561504017230 0ustar rootroot lua_opdef "luatest", {{ local ffi = require("ffi") local string = require("string") local csoundLibrary = ffi.load('csound64.dll.5.2') ffi.cdef[[ int csoundGetKsmps(void *); double csoundGetSr(void *); struct luatest_arguments {double *out; double *stringout; char *stringin; double *in1; double *in2; double sr; int ksmps; }; ]] function luatest_init(csound, opcode, carguments) local arguments = ffi.cast("struct luatest_arguments *", carguments) arguments.sr = csoundLibrary.csoundGetSr(csound) print(string.format('stringin: %s', ffi.string(arguments.stringin))) print(string.format('sr: %f', arguments.sr)) arguments.ksmps = csoundLibrary.csoundGetKsmps(csound) print(string.format('ksmps: %d', arguments.ksmps)) arguments.out[0] = arguments.in1[0] * arguments.in2[0] ffi.copy(arguments.stringout, 'Hello, world!') return 0 end function luatest_kontrol(csound, opcode, carguments) local arguments = ffi.cast("struct luatest_arguments *", carguments) arguments.out[0] = arguments.in1[0] * arguments.in2[0] return 0 end function luatest_noteoff(csound, opcode, carguments) local arguments = ffi.cast("struct luatest_arguments *", carguments) arguments.out[0] = arguments.in1[0] * arguments.in2[0] print('off') return 0 end }} instr 1 iresult = 0 Stringin = "stringin" Stringout = "stringout" lua_iopcall "luatest", iresult, Stringout, Stringin, p2, p3 prints Stringout endin instr 2 iresult = 0 Stringin = "stringin" Stringout = "initial value" lua_iopcall_off "luatest", iresult, Stringout, Stringin, p2, p3 print iresult prints Stringout endin i 1 1 2 i 2 2 2 i 1 3 2 e manual_src~dfsg/examples/diskin2.csd0000644000000000000000000000103612262561504016621 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o diskin2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktrans linseg 1, 5, 2, 10, -2 a1 diskin2 "beats.wav", ktrans, 0, 1, 0, 32 outs a1, a1 endin i 1 0 15 e manual_src~dfsg/examples/tanh.csd0000644000000000000000000000127512262561504016215 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tanh.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 0dbfs = 1 ; Instrument #1. instr 1 asig1 vco 1, 440, 2, 0.4, 1 asig2 vco 1, 800, 3, 0.5, 1 asig = asig1+asig2 ;; will go out of range out tanh(asig) ;; but tanh is a limiter endin f1 0 65536 10 1 ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/rnd31_krate.csd0000644000000000000000000000123412262561504017373 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd31_krate.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create random numbers at k-rate in the range -1 to 1 ; with a uniform distribution, seed=10. k1 rnd31 1, 0, 10 printks "k1=%f\\n", 0.1, k1 endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/chanctrl.csd0000644000000000000000000000155012262561504017055 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out No messages MIDI in -odac -d -M0 ;;;RT audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chanctrl.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; press your midi keyboard and move your midi controller to see result ichnl = 1 ;MIDI note inputs on channel 1 ictlno = 7 ;use midi volume controller kch chanctrl ichnl, 7 ;to control amplitude of oscil printk2 kch asig oscil kch*(1/127), 220, 1 outs asig, asig endin ;Dummy f-table to give time for real-time MIDI events f 0 30 ;sine wave. f 1 0 16384 10 1 e manual_src~dfsg/examples/lposcilsa.csd0000644000000000000000000000165312262561504017254 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lposcilsa.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 1.3 ;a 3th up kloop = 0 ;loop start time in samples kend = 45000 ;loop end time in samples aenv expsega 0.01, 0.1, 1, 0.1, 0.5, 0.5, 0.01 ;envelope with fast and short segment aL, aR lposcilsa aenv, kcps, kloop, kend, 1 ;use it for amplitude outs aL, aR endin ; table size of stereo file is deferred, ; and format taken from the soundfile header. f 1 0 0 1 "kickroll.wav" 0 0 0 ; Play Instrument #1 for 6 seconds. ; This will loop the drum pattern several times. i 1 0 2 e manual_src~dfsg/examples/oscil3.csd0000644000000000000000000000206112262561504016451 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil3.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = .6 kcps = 440 ifn = p4 asig oscil kamp, kcps, ifn outs asig,asig endin instr 2 kamp = .6 kcps = 440 ifn = p4 asig oscil3 kamp, kcps, ifn outs asig,asig endin f1 0 128 10 1 ; Sine with a small amount of data f2 0 128 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth with a small amount of data f3 0 128 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square with a small amount of data f4 0 128 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse with a small amount of data i 1 0 2 1 i 2 3 2 1 i 1 6 2 2 i 2 9 2 2 i 1 12 2 3 i 2 15 2 3 i 1 18 2 4 i 2 21 2 4 e manual_src~dfsg/examples/pindex.csd0000644000000000000000000000137112262561504016547 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc ; -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ;-o pindex.wav -W ;;; for file output any platform ;Example by Anthony Kozar Dec 2006 instr 1 inum pcount index init 1 loop1: ivalue pindex index printf_i "p%d = %f\n", 1, index, ivalue index = index + 1 if (index <= inum) igoto loop1 print inum endin i1 0 3 40 50 ; has 5 pfields i1 1 2 80 ; has 5 due to carry i1 2 1 40 50 60 70 ; has 7 e manual_src~dfsg/examples/cross2.csd0000644000000000000000000000221012262561504016464 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cross2.wav -W ;;; for file output any platform ; after example from Kevin Conder sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;play audio file aout soundin "fox.wav" outs aout, aout endin instr 2 ;cross-synthesize icps = p4 ifn = p5 ; Use the "ahh.aiff" sound and "eee.aiff" ain1 oscil 0.6, p4, ifn ain2 soundin "fox.wav" ; Use the "fox.wav" as modulator isize = 4096 ioverlap = 2 iwin = 3 kbias init 1 aout cross2 ain1, ain2, isize, ioverlap, iwin, kbias outs aout, aout endin ;audio files f 1 0 128 1 "ahh.aiff" 0 4 0 f 2 0 128 1 "eee.aiff" 0 4 0 f 3 0 2048 20 2 ;windowing function i 1 0 3 i 2 3 3 50 1 ;"eee.aiff" i 2 + 3 50 2 ;"ahh.aiff" i 2 + 3 100 1 ;"eee.aiff" i 2 + 3 100 2 ;"ahh.aiff" i 2 + 3 250 1 ;"eee.aiff" i 2 + 3 250 2 ;"ahh.aiff" i 2 + 3 20 1 ;"eee.aiff" i 2 + 3 20 2 ;"ahh.aiff" e manual_src~dfsg/examples/trsplit.csd0000644000000000000000000000140112262561504016753 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trsplit.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "beats.wav", 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .003, 1, 3, 500 ; partial tracking fslo,fshi trsplit fst, 1500 ; split partial tracks at 1500 Hz aout tradsyn fshi, 1, 1, 500, 1 ; resynthesis of tracks above 1500Hz outs aout, aout endin f1 0 8192 10 1 ;sine i 1 0 2 e manual_src~dfsg/examples/gen17.csd0000644000000000000000000000176312262561504016206 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M0 -+rtmidi=virtual ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen17.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 inote cpsmidi iveloc ampmidi .5 ictl midictrl 5 ;move slider of controller 5 to change ftable itab table ictl, 2 aout poscil iveloc, inote, itab outs aout, aout endin f 1 0 8193 10 1 f 2 0 128 -17 0 10 32 20 64 30 96 40 ;inhibit rescaling f 10 0 16384 10 1 ; Sine f 20 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111; Sawtooth f 30 0 16384 10 1 0 0.3 0 0.2 0 0.14 0 .111; Square f 40 0 16384 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse f 0 30 ;run for 30 seconds e manual_src~dfsg/examples/valpass.csd0000644000000000000000000000115012262561504016724 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o valpass.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 krvt = 1.5 klpt line p4, p3, p5 imaxlpt = .1 a1 diskin2 "fox.wav", 1 a1 valpass a1, krvt, klpt, imaxlpt a2 valpass a1, krvt, klpt*.5, imaxlpt outs a1, a2 endin i 1 0 5 .01 .2 e manual_src~dfsg/examples/nreverb_ftable.csd0000644000000000000000000000146012262561504020237 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o nreverb_ftable.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 a1 soundin "beats.wav" a2 nreverb a1, 1.5, .75, 0, 8, 71, 4, 72 out a1 + a2 * .4 endin ; freeverb time constants, as direct (negative) sample, with arbitrary gains f71 0 16 -2 -1116 -1188 -1277 -1356 -1422 -1491 -1557 -1617 0.8 0.79 0.78 0.77 0.76 0.75 0.74 0.73 f72 0 16 -2 -556 -441 -341 -225 0.7 0.72 0.74 0.76 i1 0 3 e manual_src~dfsg/examples/vbap8move.csd0000644000000000000000000000211212262561504017161 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vbap4move.wav -W ;;; for file output any platform sr = 48000 ksmps = 10 nchnls = 8 ;Example by Hector Centeno 2007 vbaplsinit 2, 8, 15, 65, 115, 165, 195, 245, 295, 345 instr 1 ifldnum = 9 ispread = 30 idur = p3 ;; Generate a sound source kenv loopseg 10, 0, 0, 0, 0.5, 1, 10, 0 a1 pinkish 3000*kenv ;; Move circling around once all the speakers aout1, aout2, aout3, aout4, aout5, aout6, aout7, aout8 vbap8move a1, idur, ispread, ifldnum, 15, 65, 115, 165, 195, 245, 295, 345, 15 ;; Speaker mapping aFL = aout8 ; Front Left aFR = aout1 ; Front Right aMFL = aout7 ; Mid Front Left aMFR = aout2 ; Mid Front Right aMBL = aout6 ; Mid Back Left aMBR = aout3 ; Mid Back Right aBL = aout5 ; Back Left aBR = aout4 ; Back Right outo aFL, aFR, aMFL, aMFR, aMBL, aMBR, aBL, aBR endin i1 0 30 e manual_src~dfsg/examples/pvsosc.csd0000644000000000000000000000220612262561504016573 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsosc.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 ; a band-limited sawtooth-wave oscillator fsig pvsosc 10000, 440, 1, 1024 ; generate wave spectral signal asig pvsynth fsig ; resynthesise it out asig endin instr 2 ; a band-limited square-wave oscillator fsig pvsosc 10000, 440, 2, 1024 ; generate wave spectral signal asig pvsynth fsig ; resynthesise it out asig endin instr 3 ; a pulse oscillator fsig pvsosc 10000, 440, 3, 1024 ; generate wave spectral signal asig pvsynth fsig ; resynthesise it out asig endin instr 4 ; a cosine-wave oscillator fsig pvsosc 10000, 440, 4, 1024 ; generate wave spectral signal asig pvsynth fsig ; resynthesise it out asig endin i 1 0 1 i 2 2 1 i 3 4 1 i 4 6 1 e manual_src~dfsg/examples/diskgrain.csd0000644000000000000000000000156212262561504017235 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o diskgrain.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iolaps = 2 igrsize = 0.04 ifreq = iolaps/igrsize ips = 1/iolaps istr = p4 /* timescale */ ipitch = p5 /* pitchscale */ a1 diskgrain "mary.wav", 1, ifreq, ipitch, igrsize, ips*istr, 1, iolaps outs a1, a1 endin f 1 0 8192 20 2 1 ;Hanning function ; timescale pitchscale i 1 0 5 1 1 i 1 + 5 2 1 i 1 + 5 1 0.75 i 1 + 5 1.5 1.5 i 1 + 5 0.5 1.5 e manual_src~dfsg/examples/deltapi.csd0000644000000000000000000000216612262561504016705 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o deltapi.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasig init 0 gidel = 1 ;delay time in seconds instr 1 ain pluck .7, 440, 1000, 0, 1 outs ain, ain vincr gasig, ain ;send to global delay endin instr 2 ifeedback = p4 abuf2 delayr gidel adelL deltapi .4 ;first tap (on left channel) adelM deltapi 1 ;second tap (on middle channel) delayw gasig + (adelL * ifeedback) abuf3 delayr gidel kdel line 1, p3, .01 ;vary delay time adelR deltapi .65 * kdel ;one pitch changing tap (on the right chn.) delayw gasig + (adelR * ifeedback) ;make a mix of all deayed signals outs adelL + adelM, adelR + adelM clear gasig endin i 1 0 1 i 1 3 1 i 2 0 3 0 ;no feedback i 2 3 8 .8 ;lots of feedback e manual_src~dfsg/examples/pareq.csd0000644000000000000000000000234212262561504016367 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pareq.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 15 ifc = p4 ; Center / Shelf kq = p5 ; Quality factor sqrt(.5) is no resonance kv = ampdb(p6) ; Volume Boost/Cut imode = p7 ; Mode 0=Peaking EQ, 1=Low Shelf, 2=High Shelf kfc linseg ifc*2, p3, ifc/2 asig rand 5000 ; Random number source for testing aout pareq asig, kfc, kv, kq, imode ; Parmetric equalization outs aout, aout ; Output the results endin ; SCORE: ; Sta Dur Fcenter Q Boost/Cut(dB) Mode i15 0 1 10000 .2 12 1 i15 + . 5000 .2 12 1 i15 . . 1000 .707 -12 2 i15 . . 5000 .1 -12 0 e manual_src~dfsg/examples/gen41.csd0000644000000000000000000000113112262561504016170 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o GEN41.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 k1 duserrnd 1 printk 0, k1 asig poscil .5, 220*k1, 2 outs asig, asig endin f1 0 -20 -41 2 .1 8 .9 ;choose 2 at 10% probability, 8 at 90% f2 0 8192 10 1 i1 0 2 e manual_src~dfsg/examples/sflooper.csd0000644000000000000000000000213512262561504017110 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -+rtmidi=virtual -M0 ;;;realtime audio in, midi in ; For Non-realtime ouput leave only the line below: ; -o sflooper.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 isf sfload "07AcousticGuitar.sf2" sfpassign 0, isf instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 print ivel ivel init ivel/127 ;velocity dependent kamp linsegr 1,1,1,.1,0 ;envelope kamp = kamp * .0002 ;scale amplitude (= kamp/5000) kfreq init 1 ;do not change freq from sf ;"07AcousticGuitar.sf2" contains 2 samples, on notes E1 and C#4 ;start loop from beginning, loop .2 seconds - on the root key of these samples aL,aR sflooper ivel, inum, kamp*ivel, kfreq, 0, 0, .2, .05 outs aL, aR endin f0 60 ; stay active for 1 minute i1 0 1 60 100 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e manual_src~dfsg/examples/combinv.csd0000644000000000000000000000111012262561504016704 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o comb.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps expon p5, p3, p4 asig oscil3 0.3, kcps, 1 krvt = 3.5 ilpt = 0.1 aleft combinv asig, krvt, ilpt outs aleft, asig endin f1 0 4096 10 1 i 1 0 3 20 2000 e manual_src~dfsg/examples/FLmouse.csd0000644000000000000000000000161712262561504016635 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O sr=44100 ksmps=128 nchnls=2 ;Example by Andres Cabrera 2007 giwidth = 400 giheight = 300 FLpanel "FLmouse", giwidth, giheight, 10, 10 FLpanelEnd FLrun 0dbfs = 1 instr 1 kx, ky, kb1, kb2, kb3 FLmouse 2 ktrig changed kx, ky ;Print only if coordinates have changed printf "kx = %f ky = %f \n", ktrig, kx, ky kfreq = ((giwidth - ky)*1000/giwidth) + 300 ; y coordinate determines frequency, x coordinate determines amplitude ; Left mouse button (kb1) doubles the frequency ; Right mouse button (kb3) activates sound on channel 2 aout oscil kx /giwidth , kfreq * (kb1 + 1), 1 outs aout, aout * kb3 endin f 1 0 1024 10 1 i 1 0 120 e manual_src~dfsg/examples/midiout.csd0000644000000000000000000000073712262561504016737 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -Ma -Q1 ;;;realtime audio out and midi out and midi in (all midi inputs) sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 midiout 192, 1, 21, 0 ;program change to instr. 21 inum notnum ivel veloc midion 1, inum, ivel endin i 1 0 3 80 100 ;play note for 3 seconds e manual_src~dfsg/examples/genfarey.csd0000644000000000000000000000107612262561504017062 0ustar rootroot sr=44100 ksmps=10 nchnls=1 instr 4 kndx init 0 ; read out elements of F_8 one by one and print to file if (kndx < 23) then kelem tab kndx, 1 fprintks "farey8table.txt", "%2.6f\\n", kelem kndx = kndx+1 endif endin ; initialise integer for Farey Sequence F_8 f1 0 -23 "farey" 8 0 ; if mode=0 then the table stores all elements of the Farey Sequence ; as fractions in the range [0,1] i4 0 1 e manual_src~dfsg/examples/exprandi.csd0000644000000000000000000000111412262561504017065 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o exprand.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 klambda exprandi 20, 1, 3 printk2 klambda ; look aout oscili 0.8, 440+klambda, 1 ; & listen outs aout, aout endin ; sine wave f 1 0 16384 10 1 i 1 0 4 e manual_src~dfsg/examples/pgmassign_advanced.csd0000644000000000000000000000276412262561504021104 0ustar rootroot ; Select audio/midi flags here according to platform -odac -F pgmassign_advanced.mid ;;;realtime audio out with midifile in ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pgmassign_advanced.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 massign 1, 1 ; channels 1 to 4 use instr 1 by default massign 2, 1 massign 3, 1 massign 4, 1 ; pgmassign_advanced.mid can be found in /manual/examples ; pgmassign.mid has 4 notes with these parameters: ; ; Start time Channel Program ; ; note 1 0.5 1 10 ; note 2 1.5 2 11 ; note 3 2.5 3 12 ; note 4 3.5 4 13 pgmassign 0, 0 ; disable program changes pgmassign 11, 3 ; program 11 uses instr 3 pgmassign 12, 2 ; program 12 uses instr 2 ; waveforms for instruments itmp ftgen 1, 0, 1024, 10, 1 itmp ftgen 2, 0, 1024, 10, 1, 0.5, 0.3333, 0.25, 0.2, 0.1667, 0.1429, 0.125 itmp ftgen 3, 0, 1024, 10, 1, 0, 0.3333, 0, 0.2, 0, 0.1429, 0, 0.10101 instr 1 /* sine */ kcps cpsmidib 2 ; note frequency asnd oscili .6, kcps, 1 outs asnd, asnd endin instr 2 /* band-limited sawtooth */ kcps cpsmidib 2 ; note frequency asnd oscili .6, kcps, 2 outs asnd, asnd endin instr 3 /* band-limited square */ kcps cpsmidib 2 ; note frequency asnd oscili .6, kcps, 3 outs asnd, asnd endin t 0 120 f 0 8.5 2 -2 0 e manual_src~dfsg/examples/ftconv.csd0000644000000000000000000000415712262561504016564 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ftconv.wav -W ;;; for file output any platform sr = 48000 ksmps = 32 nchnls = 2 0dbfs = 1 garvb init 0 gaW init 0 gaX init 0 gaY init 0 itmp ftgen 1, 0, 64, -2, 2, 40, -1, -1, -1, 123, \ 1, 13.000, 0.05, 0.85, 20000.0, 0.0, 0.50, 2, \ 1, 2.000, 0.05, 0.85, 20000.0, 0.0, 0.25, 2, \ 1, 16.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 9.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 12.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 8.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2 itmp ftgen 2, 0, 262144, -2, 0 spat3dt 2, -0.2, 1, 0, 1, 1, 2, 0.005 itmp ftgen 3, 0, 262144, -52, 3, 2, 0, 4, 2, 1, 4, 2, 2, 4 instr 1 a1 vco2 1, 440, 10 kfrq port 100, 0.008, 20000 a1 butterlp a1, kfrq a2 linseg 0, 0.003, 1, 0.01, 0.7, 0.005, 0, 1, 0 a1 = a1 * a2 * 2 denorm a1 vincr garvb, a1 aw, ax, ay, az spat3di a1, p4, p5, p6, 1, 1, 2 vincr gaW, aw vincr gaX, ax vincr gaY, ay endin instr 2 denorm garvb ; skip as many samples as possible without truncating the IR arW, arX, arY ftconv garvb, 3, 2048, 2048, (65536 - 2048) aW = gaW + arW aX = gaX + arX aY = gaY + arY garvb = 0 gaW = 0 gaX = 0 gaY = 0 aWre, aWim hilbert aW aXre, aXim hilbert aX aYre, aYim hilbert aY aWXr = 0.0928*aXre + 0.4699*aWre aWXiYr = 0.2550*aXim - 0.1710*aWim + 0.3277*aYre aL = aWXr + aWXiYr aR = aWXr - aWXiYr outs aL, aR endin i 1 0 0.5 0.0 2.0 -0.8 i 1 1 0.5 1.4 1.4 -0.6 i 1 2 0.5 2.0 0.0 -0.4 i 1 3 0.5 1.4 -1.4 -0.2 i 1 4 0.5 0.0 -2.0 0.0 i 1 5 0.5 -1.4 -1.4 0.2 i 1 6 0.5 -2.0 0.0 0.4 i 1 7 0.5 -1.4 1.4 0.6 i 1 8 0.5 0.0 2.0 0.8 i 2 0 10 e manual_src~dfsg/examples/vmult-2.csd0000644000000000000000000000146712262561504016574 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vmult ifn1, ival, ielements, idstoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e manual_src~dfsg/examples/sndloop.csd0000644000000000000000000000157612262561504016745 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sndloop.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 itrig = p4 asig diskin2 "beats.wav", 1, 0, 1 ;get the signal in, loop it ktrig line 0, itrig, 1 ;when to trigger signal = p4 kpitch line 1.2, p3, .5 ;vary pitch of recorded signal aout,krec sndloop asig, kpitch, ktrig, .4, 0.05 ;rec starts at p4 sec, for .4 secs and 0.05 crossfade printk2 krec ; prints the recording signal outs aout, aout endin i1 0 5 .5 ;trigger in seconds (=p4) i1 + 5 .8 i1 + 10 1.2 e manual_src~dfsg/examples/strindexk.csd0000644000000000000000000000410212262561504017266 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o strindexk.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 instr 1 ;get one element of the input string whenever the metro ;triggers, and call a subinstrument to play the file Smember strget p4 istrlen strlen Smember kprint init 0 ktrig metro .5 ;whenever the trigger gives signal if ktrig == 1 then ;choose a random element (0, 1 or 2) kel random 0, 2.9999 kel = int(kel) ;make a copy for leaving Smember intact Scopy strcpyk Smember ;set the initial index for reading substrings kndx = 0 ;set counter for searching the element kcount = 0 ;start looping over the elements in Smember loop: kdelim strindexk Scopy, ":" ;as long as ":" occurs in Scopy, do: if kdelim > 0 then ;if this is the element to get if kel == kcount then ;read it as substring Sfile strsubk Scopy, kndx, kdelim kprint = kprint+1 ;and jump out kgoto call ;if not else ;cut off this element from Scopy Scopy strsubk Scopy, kdelim+1, istrlen endif ;if no element has been found,go back to loop ;and look for the next element kcount = kcount+1 kgoto loop ;if there is no delimiter left, the rest is the searched element else Sfile strcpyk Scopy endif call: ;print the result, call the subinstrument and play the file printf "kel = %d, file = '%s'\n", ktrig+kprint, kel, Sfile S_call sprintfk {{i 2 0 1 "%s"}}, Sfile scoreline S_call, ktrig endif endin instr 2 ;play Sfile strget p4 ilen filelen Sfile p3 = ilen asig soundin Sfile outs asig, asig endin i 1 0 30 "mary.wav:fox.wav:beats.wav" e manual_src~dfsg/examples/pvsmorph.csd0000644000000000000000000000201112262561504017126 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsmorph.wav -W ;;; for file output any platform sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 giSine ftgen 0, 0, 4096, 10, 1 instr 1 iampint1 = p4 iampint2 = p5 ifrqint1 = p6 ifrqint2 = p7 kampint linseg iampint1, p3, iampint2 kfrqint linseg ifrqint1, p3, ifrqint2 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "fox.wav" ain1 soundin Sfile1 ain2 buzz .2, 50, 100, giSine fftin1 pvsanal ain1, ifftsize, ioverlap, iwinsize, iwinshape fftin2 pvsanal ain2, ifftsize, ioverlap, iwinsize, iwinshape fmorph pvsmorph fftin1, fftin2, kampint, kfrqint aout pvsynth fmorph out aout * .5 endin i 1 0 3 0 0 1 1 i 1 3 3 1 0 1 0 i 1 6 3 0 1 0 1 e manual_src~dfsg/examples/event_named.csd0000644000000000000000000000232712262561504017547 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o event_named.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - an oscillator with a high note. instr 1 ; Create a trigger and set its initial value to 1. ktrigger init 1 ; If the trigger is equal to 0, continue playing. ; If not, schedule another event. if (ktrigger == 0) goto contin ; kscoreop="i", an i-statement. ; kinsnum="low_note", instrument named "low_note". ; kwhen=1, start at 1 second. ; kdur=0.5, play for a half-second. event "i", "low_note", 1, 0.5 ; Make sure the event isn't triggered again. ktrigger = 0 contin: a1 oscils 10000, 440, 1 out a1 endin ; Instrument "low_note" - an oscillator with a low note. instr low_note a1 oscils 10000, 220, 1 out a1 endin ; Make sure the score plays for two seconds. f 0 2 ; Play Instrument #1 for a half-second. i 1 0 0.5 e manual_src~dfsg/examples/linsegr.csd0000644000000000000000000000123112262561504016716 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o linsegr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .3 kenv linsegr 1, .05, 0.5, 1, 0 asig pluck kenv, icps, 200, 1, 1 outs asig, asig endin f 1 0 4096 10 1 ;sine wave f0 30 ;runs 30 seconds e manual_src~dfsg/examples/fmrhode.csd0000644000000000000000000000137312262561504016706 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmrhode.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 220 kc1 = p4 kc2 = p5 kvdepth = 0.01 kvrate = 3 ifn1 = 1 ifn2 = 1 ifn3 = 1 ifn4 = 2 ivfn = 1 asig fmrhode .5, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn outs asig, asig endin ; sine wave. f 1 0 32768 10 1 ; audio file. f 2 0 256 1 "fwavblnk.aiff" 0 0 0 i 1 0 3 6 0 i 1 + . 6 3 i 1 + . 20 0 e manual_src~dfsg/examples/toot3.csd0000644000000000000000000000135512262561504016332 0ustar rootroot -odac instr 3 ; p3=duration of note k1 linen p4, p6, p3, p7 ; p4=amp a1 oscil k1, p5, 1 ; p5=freq out a1 ; p6=attack time endin f1 0 4096 10 1 ; sine wave ;ins strt dur amp(p4) freq(p5) attack(p6) release(p7) i3 0 1 10000 440 0.5 0.7 i3 1.5 1 10000 440 0.9 0.1 i3 3 1 5000 880 0.02 0.99 i3 4.5 1 5000 880 0.7 0.01 i3 6 2 20000 220 0.5 0.5 e manual_src~dfsg/examples/inch.csd0000644000000000000000000000127412262561504016203 0ustar rootroot ; Select audio/midi flags here according to platform -odac -idac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; inch.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;nchnls channels in, two channels out ain1, ainr inch 1, 2 ;grab your mic and sing adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time = 20ms aoutl flanger ain1, adel, .7 aoutr flanger ain1, adel*2, .8 fout "in_ch.wav", 14, aoutl, aoutr ;write to stereo file, outs aoutl, aoutr ;16 bits with header endin i 1 0 10 e manual_src~dfsg/examples/spsend_stereo.csd0000644000000000000000000000176312262561504020142 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o spsend_stereo.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ga1 init 0 ga2 init 0 instr 1 asig diskin2 "beats.wav", 1 a1, a2, a3, a4 space asig, 0, 0, .1, p4, p5 ; take position values from p4, p5 ar1, ar2, ar3, ar4 spsend ; a3, a4, ar3, ar4 are dummies ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a2 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 outs a1, a2 ga1=0 ga2=0 endin ;place the sound in the left speaker and near i1 0 1 -1 1 ;place the sound in the right speaker and far i1 1 1 45 45 ;place the sound equally between left and right and in the middle ground distance i1 2 1 0 12 i 99 0 5 e manual_src~dfsg/examples/expon.csd0000644000000000000000000000154312262561504016412 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o expon.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kpitch = p6 ;choose between expon or line if (kpitch == 0) then kpitch expon p4, p3, p5 elseif (kpitch == 1) then kpitch line p4, p3, p5 endif asig vco2 .6, kpitch outs asig, asig endin i 1 0 2 300 600 0 ;if p6=0 then expon is used i 1 3 2 300 600 1 ;if p6=1 then line is used i 1 6 2 600 1200 0 i 1 9 2 600 1200 1 i 1 12 2 1200 2400 0 i 1 15 2 1200 2400 1 i 1 18 2 2400 30 0 i 1 21 2 2400 30 1 e manual_src~dfsg/examples/zawm.csd0000644000000000000000000000250512262561504016236 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zawm.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a basic instrument. instr 1 ; Generate a simple sine waveform. asin oscil 15000, 440, 1 ; Mix the sine waveform with za variable #1. zawm asin, 1 endin ; Instrument #2 -- another basic instrument. instr 2 ; Generate another waveform with a different frequency. asin oscil 15000, 880, 1 ; Mix this sine waveform with za variable #1. zawm asin, 1 endin ; Instrument #3 -- generates audio output. instr 3 ; Read za variable #1, containing both waveforms. a1 zar 1 ; Generate the audio output. out a1 ; Clear the za variables, get them ready for ; another pass. zacl 0, 1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 ; Play Instrument #3 for one second. i 3 0 1 e manual_src~dfsg/examples/pvsadsyn.csd0000644000000000000000000000256012262561504017130 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsadsyn.wav -W ;;; for file output any platform sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 opcode FileToPvsBuf, iik, Siiii ;;writes an audio file at the first k-cycle to a fft-buffer (via pvsbuffer) Sfile, ifftsize, ioverlap, iwinsize, iwinshape xin ktimek timeinstk if ktimek == 1 then ilen filelen Sfile kcycles = ilen * kr; number of k-cycles to write the fft-buffer kcount init 0 loop: ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape ibuf, ktim pvsbuffer fftin, ilen + (ifftsize / sr) loop_lt kcount, 1, kcycles, loop xout ibuf, ilen, ktim endif endop instr 1 istretch = p4; time stretching factor ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window ibuffer, ilen, k0 FileToPvsBuf "fox.wav", ifftsize, ioverlap, iwinsize, iwinshape p3 = istretch * ilen; set p3 to the correct value ktmpnt linseg 0, p3, ilen; time pointer fread pvsbufread ktmpnt, ibuffer; read the buffer aout pvsadsyn fread, 10, 1; resynthesis with the first 10 bins out aout endin i 1 0 1 20 e manual_src~dfsg/examples/gen11.csd0000644000000000000000000000130612262561504016171 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen11.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 asig oscil .8, 220, ifn outs asig,asig endin f 1 0 16384 11 1 1 ;number of harmonics = 1 f 2 0 16384 11 10 1 .7 ;number of harmonics = 10 f 3 0 16384 11 10 5 2 ;number of harmonics = 10, 5th harmonic is amplified 2 times i 1 0 2 1 i 1 + 2 2 i 1 + 2 3 e manual_src~dfsg/examples/sfinstr.csd0000644000000000000000000000166712262561504016760 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfinstr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf sfload "sf_GMbank.sf2" sfilist isf instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/5000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfinstr ivel, inum, kamp*ivel, kfreq, 194, gisf ;= Strings 2 tighter outs a1, a2 endin f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e manual_src~dfsg/examples/push.csd0000644000000000000000000000140212262561504016232 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o push.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 stack 100000 instr 1 a1 oscils 0.7, 220, 0 k1 line 0, p3, 1 push "blah", 123.45, a1, k1 push rnd(k1) k_rnd pop S01, i01, a01, k01 pop printf_i "S01 = '%s', i01 = %g\n", 1, S01, i01 ktrig metro 5.0 printf "k01 = %.3f, k_rnd = %.3f\n", ktrig, k01, k_rnd outs a01, a01 endin i 1 0 5 e manual_src~dfsg/examples/tableng.csd0000644000000000000000000000121312262561504016667 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tableng.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifn1 ftgen 1, 0, 0, 1, "flute.aiff", 0, 4, 0 ;deferred-size table instr 1 isize tableng 1 print isize andx phasor 1 / (isize / sr) ;play at correct pitch asig tab andx, 1, 1 outs asig, asig endin i 1 0 2.3 e manual_src~dfsg/examples/cpumeter.csd0000644000000000000000000000134412262561504017104 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpumeter.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 1 0dbfs = 1 instr 1 ;cpu metering; stop when too large k0 cpumeter 0.1 printk2 k0 if k0>70 then event "i", 3, 0.1, 1 endif endin instr 2 event_i "i", 2, 1, 1000 asig oscil 0.2, 440, 1 out asig endin instr 3 exitnow endin f 1 0 32768 10 1 ; sine wave i 1 0 1000 i 2 0 1000 e manual_src~dfsg/examples/minabsaccum.csd0000644000000000000000000000167012262561504017544 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o minabsaccum.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;saw asig vco2 .2, p4 outs asig, asig gasaw = asig endin instr 2 ;sine aout poscil .3, p4, 1 outs aout, aout gasin = aout endin instr 10 accum init 0 minabsaccum accum, gasaw + gasin ;saw and sine accumulated accum dcblock2 accum ;get rid of DC outs accum, accum clear accum endin f 1 0 4096 10 1 i 1 0 7 330 i 2 3 3 440 i 1 10 7 330 ;same notes but without minabsaccum, for comparison i 2 13 3 440 i 10 0 6 ;accumulation note stops after 6 seconds e manual_src~dfsg/examples/fmbell.csd0000644000000000000000000000154112262561504016520 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmbell.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = p4 kfreq = 880 kc1 = p5 kc2 = p6 kvdepth = 0.005 kvrate = 6 asig fmbell kamp, kfreq, kc1, kc2, kvdepth, kvrate outs asig, asig endin instr 2 kamp = p4 kfreq = 880 kc1 = p5 kc2 = p6 kvdepth = 0.005 kvrate = 6 asig fmbell kamp, kfreq, kc1, kc2, kvdepth, kvrate, 1, 1, 1, 1, 1, p7 outs asig, asig endin ; sine wave. f 1 0 32768 10 1 i 1 0 3 .2 5 5 i 1 + 4 .3 .5 1 s i 2 0 12 .2 5 5 16 i 2 + 12 .3 .5 1 12 e manual_src~dfsg/examples/flute.aiff0000644000000000000000000070335012262561504016541 0ustar rootrootFORMAIFFCOMM2@DSSNDl  #%&$&$&.(.--.+1,--53073;88<:?=:A>=A=B@CBGC@DDEBBFD@EGJFGGKLMLJJIDBHGGIOIJJMLJLIOLPKMPJOPRPMRRORKRNRTMURPSRPURXXQYQTVPSTRWYVVQRPRORPOSVYW[STPTVQWSVVTUQYTQRSOOROSPSRQRQSUPUTRVPRQNMLKMJJHNOFMFGJHJHHIBAEAC=A<=>4>=8978?<;>:9;:491:;231428+10,2-.0-21+,')**++)(+((+%*+*$*+&)'(#$$!#!##!%$ %$!  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Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKWurley.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;(FM) Modulator Index One kv3 = p7 asig STKWurley cpspch(p4), 1, 2,kv1, 4, 10, 11, kv3, 1, 30, 128, 75 outs asig, asig endin i 1 0 .5 7.00 75 0 20 i 1 + . 8.00 120 0 20 i 1 + 1 6.00 50 120 20 i 1 + 4 8.00 10 10 127 e manual_src~dfsg/examples/spat3dt.csd0000644000000000000000000000411712262561504016643 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o spat3dt.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 garvb init 0 gaW init 0 gaX init 0 gaY init 0 itmp ftgen 1, 0, 64, -2, 2, 40, -1, -1, -1, 123, \ 1, 13.000, 0.05, 0.85, 20000.0, 0.0, 0.50, 2, \ 1, 2.000, 0.05, 0.85, 20000.0, 0.0, 0.25, 2, \ 1, 16.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 9.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 12.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 8.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2 itmp ftgen 2, 0, 262144, -2, 0 spat3dt 2, -0.2, 1, 0, 1, 1, 2, 0.005 itmp ftgen 3, 0, 262144, -52, 3, 2, 0, 4, 2, 1, 4, 2, 2, 4 instr 1 a1 vco2 1, 440, 10 kfrq port 100, 0.008, 20000 a1 butterlp a1, kfrq a2 linseg 0, 0.003, 1, 0.01, 0.7, 0.005, 0, 1, 0 a1 = a1 * a2 * 2 denorm a1 vincr garvb, a1 aw, ax, ay, az spat3di a1, p4, p5, p6, 1, 1, 2 vincr gaW, aw vincr gaX, ax vincr gaY, ay endin instr 2 denorm garvb ; skip as many samples as possible without truncating the IR arW, arX, arY ftconv garvb, 3, 2048, 2048, (65536 - 2048) aW = gaW + arW aX = gaX + arX aY = gaY + arY garvb = 0 gaW = 0 gaX = 0 gaY = 0 aWre, aWim hilbert aW aXre, aXim hilbert aX aYre, aYim hilbert aY aWXr = 0.0928*aXre + 0.4699*aWre aWXiYr = 0.2550*aXim - 0.1710*aWim + 0.3277*aYre aL = aWXr + aWXiYr aR = aWXr - aWXiYr outs aL, aR endin i 1 0 0.5 0.0 2.0 -0.8 i 1 1 0.5 1.4 1.4 -0.6 i 1 2 0.5 2.0 0.0 -0.4 i 1 3 0.5 1.4 -1.4 -0.2 i 1 4 0.5 0.0 -2.0 0.0 i 1 5 0.5 -1.4 -1.4 0.2 i 1 6 0.5 -2.0 0.0 0.4 i 1 7 0.5 -1.4 1.4 0.6 i 1 8 0.5 0.0 2.0 0.8 i 2 0 10 e manual_src~dfsg/examples/dssiaudio.csd0000644000000000000000000000201712262561504017242 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o DSSIplay_mono.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 gihandle dssiinit "caps.so", 19, 1 ; = mono phaser and gaout init 0 ; verbose about all ports instr 1 ; activate DSSI dssiactivate gihandle, 1 endin instr 2 ain1 diskin2 "beats.wav", 1,0,1 ; loop gaout = gaout+(ain1*.5) endin instr 3 dssictls gihandle, 0, .8, 1 ; range -1 to 1 dssictls gihandle, 1, .05, 1 ; rate 0 to 10 dssictls gihandle, 2, .8, 1 ; depth 0 to 1 dssictls gihandle, 3, 2, 1 ; spread 0 to 3.14 dssictls gihandle, 4, .7, 1 ; feedback 0 to 0.999 endin instr 4 aout1 dssiaudio gihandle, gaout ;get beats.wav, mono out outs aout1,aout1 gaout = 0 endin i 1 0 20 i 2 1 20 i 3 1 20 i 4 0 20 e manual_src~dfsg/examples/sinsyn.csd0000644000000000000000000000142012262561504016576 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sinsyn.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kmxtr init p4 ain diskin2 "fox.wav", 1 fs1,fsi2 pvsifd ain, 2048, 512,1 ; ifd analysis fst partials fs1, fsi2, .03, 1, 3, 500 ; partial tracking aout sinsyn fst, .5, kmxtr, 1 ; scale amplitude down outs aout, aout endin f1 0 8192 10 1 i 1 0 2.7 15 ;filtering effect by using low number of tracks i 1 + 2.7 500 ;maximum number of tracks e manual_src~dfsg/examples/tableimix.csd0000644000000000000000000000135612262561504017241 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tableimix.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisine ftgen 1, 0, 256, 10, 1, 0, 0, .4 ;sinoid gisaw ftgen 2, 0, 1024, 7, 0, 256, 1 ;saw gimix ftgen 100, 0, 256, 7, 0, 256, 1 ;used to mix instr 1 tableimix 100, 0, 256, 1, 0, 1, 2, 0, .5 asig poscil .5, 110, gimix ;mix table 1 & 2 outs asig, asig endin i1 0 5 e manual_src~dfsg/examples/ATSread.csd0000644000000000000000000000114712262561504016544 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSread.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 2 kfreq, kamp ATSread ktime, "beats.ats", 100 aout oscili 0.8, kfreq, 1 outs aout, aout endin ;sine wave. f 1 0 16384 10 1 i 1 0 2 e manual_src~dfsg/examples/pvsdisp.csd0000644000000000000000000000101512262561504016743 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsdisp.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 asig inch 1 ;a1 soundin "input.wav" ;select a soundifle fsig pvsanal asig, 1024,256, 1024, 1 pvsdisp fsig endin i 1 0 30 e manual_src~dfsg/examples/dumpk4.csd0000644000000000000000000000313312262561504016462 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dumpk4.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes four control signals to a file kcf randh 950, 1, 2, 1, 1050 ;generates one random number between 100 and 2000 per second kq randh 10, 1, 2, 1, 11 ;generates another random number between 1 and 21 per second kdb randh 9, 1, 2, 1, -15 ;amplitudes in dB between -24 and -6 kpan randh .5, 1, 2, 1, .5 ;panning between 0 and 1 dumpk4 kcf, kq, kdb, kpan, "dumpk4.txt", 8, 1 ;writes the control signals prints "WRITING:\n" printks "kcf = %f, kq = %f, kdb = %f, kpan = %f\n", 1, kcf, kq, kdb, kpan ;prints them endin instr 2 ;reads the file written by instr 1 kcf,kq,kdb,kp readk4 "dumpk4.txt", 8, 1 prints "READING:\n" printks "kcf = %f, kq = %f, kdb = %f, kpan = %f\n", 1, kcf, kq, kdb, kp ;prints values kdb lineto kdb, .1 ;smoothing amp transition kp lineto kp, .1 ;smoothing pan transition anoise rand ampdb(kdb), 2, 1 kbw = kcf/kq ;bandwidth of resonant filter abp reson anoise, kcf, kbw aout balance abp, anoise aL, aR pan2 aout, kp outs aL, aR endin i 1 0 5 i 2 5 5 e manual_src~dfsg/examples/kickroll.wav0000644000000000000000000044725412262561504017134 0ustar rootrootRIFFNWAVEfmt DdataN.ool:{jGpxaqG$3r9-=?D ,dv  u5@nK.)HI2' ._.u@J7)OI *6.;{t` {El7 R1g5 +   ; 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X 2K ? !  sq(Ms YH :Fsb* Z'\;O# *9N 8 {lT[ D^'iS.vy.[VF8" i B=%KNj M  0(; ( 4!Y% `nT "F2_  pXVji Kp~8ZM  vw A#{ VBDRyvRp ^ f$~>m?/O$w S4Q( %obT2+"Dd N s.N YPs}/-^i] s@ q P Yh8r-4b4_;|"0 /L6.v Q x dw{s* a"1I* s Dz, }i~P # rpm\Own>[ k!w /p9mBrFT nfE $t RHpQK4#`D 1h?LM%~l!#P1`r:)23"~+5ivVup uBNL>{ t]6=rY &81VEf e,CT Z\B@ft)Q7XSpz'T.`! 1'6t:* h!6 a-ruz ^ptUO \N^lfFmk`cX5<rht!3wpL  Xl~T#;.h&.>oDZ oYS7sH't dXX:A0G$ kpI2_]B5O\6 X j+q T% f^!\<f&x16 M 6{ }j$iT6} NGkVC\8 TRs$;&#D _c2H4 FDfF5L{.O0U 0 ,BXls4%|::/_;mijlE(J=sB]4z*jB5fK3H}X=manual_src~dfsg/examples/product.csd0000644000000000000000000000141112262561504016733 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o product.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisine ftgen 0, 0, 2^10, 10, 1 instr 1 a1 oscili 1, 10.0, gisine ;combine 3 sinusses a2 oscili 1, 1.0, gisine ;at different rates a3 oscili 1, 3.0, gisine ares product a1, a2, a3 ares = ares*10000 ;scale result and asig poscil .5, ares+110, gisine ;add to frequency outs asig, asig endin i1 0 5 e manual_src~dfsg/examples/ftsave.csd0000644000000000000000000000224312262561504016547 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ftsave.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Table #1, make a sine wave using the GEN10 routine. gitmp1 ftgen 1, 0, 32768, 10, 1 ; Table #2, create an empty table. gitmp2 ftgen 2, 0, 32768, 7, 0, 32768, 0 ; Instrument #1 - a basic oscillator. instr 1 kamp = 20000 kcps = 440 ; Use Table #1. ifn = 1 a1 oscil kamp, kcps, ifn out a1 endin ; Instrument #2 - Load Table #1 into Table #2. instr 2 ; Save Table #1 to a file called "table1.ftsave". ftsave "table1.ftsave", 0, 1 ; Load the "table1.ftsave" file into Table #2. ftload "table1.ftsave", 0, 2 kamp = 20000 kcps = 440 ; Use Table #2, it should contain Table #1's sine wave now. ifn = 2 a1 oscil kamp, kcps, ifn out a1 endin ; Play Instrument #1 for 1 second. i 1 0 1 ; Play Instrument #2 for 1 second. i 2 2 1 e manual_src~dfsg/examples/readfi.csd0000644000000000000000000000061212262561504016507 0ustar rootroot -n instr 1 Swd pwd printf_i "Working directory is '%s'\n", 1, Swd prints "Reading myself =):\n" read: Sline, iLineNum readfi "readfi.csd" printf_i "Line %d: %s", iLineNum, iLineNum, Sline if iLineNum != -1 igoto read endin i1 0 0.1 e manual_src~dfsg/examples/randomi.csd0000644000000000000000000000160212262561504016706 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o randomi.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 seed 0 ; Instrument #1. instr 1 ; Choose a random frequency between 220 and 440. ; Generate new random numbers at 10 Hz. kmin init 220 kmax init 440 kcps init 10 imode = p4 ifstval = p5 printf_i "\nMode: %d\n", 1, imode k1 randomi kmin, kmax, kcps, imode, ifstval printks "k1 = %f\n", 0.1, k1 endin ; Play Instrument #1 for one second. ; each time with a different mode. i 1 0 1 i 1 1 1 1 i 1 2 1 2 330 i 1 3 1 3 e manual_src~dfsg/examples/q.csd0000644000000000000000000000135212262561504015517 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o q.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .8*aenv, p4, 1 outs asig, asig endin f 1 0 16384 10 1;sine wave s q 1 6 0 ;mute at 6 seconds in this section i 1 0 2 110 i 1 3 2 220 i 1 6 2 440 i 1 9 2 880 s q 1 6 1 ;unmute again at 6 seconds in this section i 1 0 2 110 i 1 3 2 220 i 1 6 2 440 i 1 9 2 880 e manual_src~dfsg/examples/tempest.csd0000644000000000000000000000171712262561504016745 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tempest.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use the "beats.wav" sound file. asig soundin "beats.wav" ; Extract the pitch and the envelope. kcps, krms pitchamdf asig, 150, 500, 200 iprd = 0.01 imindur = 0.1 imemdur = 3 ihp = 1 ithresh = 30 ihtim = 0.005 ixfdbak = 0.05 istartempo = 110 ifn = 1 ; Estimate its tempo. k1 tempest krms, iprd, imindur, imemdur, ihp, ithresh, ihtim, ixfdbak, istartempo, ifn printk2 k1 out asig endin ; Table #1, a declining line. f 1 0 128 16 1 128 1 ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/vbap4.csd0000644000000000000000000000137212262561504016275 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbap4.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 vbaplsinit 2, 4, 0, 90, 180, 270 instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav kaz line 0, p3, p4 ;come from right rear speaker & a1,a2,a3,a4 vbap4 asig, 180, 100, kaz ;change spread of soundsource printks "spread of source = %d\n", 1, kaz ;print spread value outq a1,a2,a3,a4 endin i 1 0 12 100 e manual_src~dfsg/examples/serialRead.csd0000644000000000000000000000171112262561504017331 0ustar rootroot ; Select audio/midi flags here according to platform -n ;;;no output ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o serialRead.wav -W ;;; for file output any platform sr = 44100 ksmps = 500 ; the default krate can be too fast for the arduino to handle nchnls = 2 0dbfs = 1 instr 1 iPort serialBegin "/dev/cu.usbmodemfa131", 9600 ;connect to the arduino with baudrate = 9600 kGain init 16 ;read our knob value kVal serialRead iPort if (kVal != -1) then kGain = kVal/128 endif aSig in ;get our audio input and get its rms kRms rms aSig*kGain kRms = kRms*kRms*255 ;scale the rms to a good value for the LED and send it out serialWrite iPort, (kRms < 255 ? kRms : 255) ;must be in range: 0-255 endin f 1 0 1024 10 1 1 1 1 1 1 i 1 0 200 e manual_src~dfsg/examples/vaset.csd0000644000000000000000000000147312262561504016405 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o avarset.wav -W ;;; for file output any platform sr=44100 ksmps=1 nchnls=2 instr 1 ; Sine Wave ifreq = (p4 > 15 ? p4 : cpspch(p4)) iamp = ampdb(p5) kenv adsr 0.1, 0.05, .9, 0.2 aout init 0 ksampnum init 0 kcount = 0 iperiod = sr / ifreq i2pi = 3.14159 * 2 loopStart: kphase = (ksampnum % iperiod) / iperiod knewval = sin(kphase * i2pi) vaset knewval, kcount,aout ksampnum = ksampnum + 1 loop_lt kcount, 1, ksmps, loopStart aout = aout * iamp * kenv outs aout, aout endin i1 0.0 2 440 80 e manual_src~dfsg/examples/metro.csd0000644000000000000000000000043712262561504016410 0ustar rootroot -odac -B441 -b441 sr = 44100 kr = 100 ksmps = 441 nchnls = 2 instr 1 ktrig metro 0.2 printk2 ktrig endin i 1 0 20 manual_src~dfsg/examples/printk2.csd0000644000000000000000000000136712262561504016656 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o printk2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kval randomh 0, 1.2, 20 ;choose between 0 and 1.2 if kval >0 && kval<=.5 then ;3 possible outcomes kval = 1 elseif kval >.5 && kval<=1 then kval =2 elseif kval >1 then kval =3 endif printk2 kval ;print value when it changes asig poscil .7, 440*kval, 1 outs asig, asig endin f1 0 16384 10 1 i 1 0 5 e manual_src~dfsg/examples/mididefault.csd0000644000000000000000000000171012262561504017544 0ustar rootroot ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mididefault.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 mididefault 60, p3 ;ensures that a MIDI-activated instrument will have a positive p3 field midinoteoncps p4, p5 ;puts MIDI key translated to cycles per second into p4, and MIDI velocity into p5 kvel = p5/127 ;scale midi velocity to 0-1 kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by asig pluck kenv*kvel, p4, p4, 2, 1 ;velocity outs asig, asig endin f 0 30 ;runs for 30 seconds f 2 0 4096 10 1 i 1 0 2 440 100 ; play these notes from score as well i 1 + 2 1440 100 e manual_src~dfsg/examples/times.csd0000644000000000000000000000114112262561504016374 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o times.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the value from times every half-second. k1 times printks "k1 = %f seconds\\n", 0.5, k1 endin ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/fout_ftable.csd0000644000000000000000000000144312262561504017552 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o fout_ftable.wav -W ;;; for file output any platform ; By: Jonathan Murphy 2007 gilen = 131072 gicps = sr/gilen gitab ftgen 1, 0, gilen, 10, 1 instr 1 /******** write file to table ********/ ain diskin2 "beats.wav", 1, 0, 1 aphs phasor gicps andx = aphs * gilen tablew ain, andx, gitab /******** write table to file ********/ aosc table aphs, gitab, 1 out aosc fout "beats_copy.wav", 6, aosc endin i1 0 2 e manual_src~dfsg/examples/event.csd0000644000000000000000000000224712262561504016404 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o event.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - an oscillator with a high note. instr 1 ; Create a trigger and set its initial value to 1. ktrigger init 1 ; If the trigger is equal to 0, continue playing. ; If not, schedule another event. if (ktrigger == 0) goto contin ; kscoreop="i", an i-statement. ; kinsnum=2, play Instrument #2. ; kwhen=1, start at 1 second. ; kdur=0.5, play for a half-second. event "i", 2, 1, 0.5 ; Make sure the event isn't triggered again. ktrigger = 0 contin: a1 oscils 10000, 440, 1 out a1 endin ; Instrument #2 - an oscillator with a low note. instr 2 a1 oscils 10000, 220, 1 out a1 endin ; Make sure the score plays for two seconds. f 0 2 ; Play Instrument #1 for a half-second. i 1 0 0.5 e manual_src~dfsg/examples/sumarray.csd0000644000000000000000000000131412262561504017120 0ustar rootroot ; Select audio/midi flags here according to platform -n ;example by joachim heintz seed 0 instr 1 ;create an array with 10 elements kArr[] init 10 ;fill in random numbers and print them out kIndx = 0 until kIndx == 10 do kNum random 0, 10 kArr[kIndx] = kNum printf "kArr[%d] = %10f\n", kIndx+1, kIndx, kNum kIndx += 1 od ;calculate sum of all values and print it out kSum sumarray kArr printf "Sum of all values in kArr = %f\n", kIndx+1, kSum turnoff endin i1 0 0.1 e manual_src~dfsg/examples/serialWrite.csd0000644000000000000000000000171212262561504017551 0ustar rootroot ; Select audio/midi flags here according to platform -n ;;;no output ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o serialWrite.wav -W ;;; for file output any platform sr = 44100 ksmps = 500 ; the default krate can be too fast for the arduino to handle nchnls = 2 0dbfs = 1 instr 1 iPort serialBegin "/dev/cu.usbmodemfa131", 9600 ;connect to the arduino with baudrate = 9600 kGain init 16 ;read our knob value kVal serialRead iPort if (kVal != -1) then kGain = kVal/128 endif aSig in ;get our audio input and get its rms kRms rms aSig*kGain kRms = kRms*kRms*255 ;scale the rms to a good value for the LED and send it out serialWrite iPort, (kRms < 255 ? kRms : 255) ;must be in range: 0-255 endin f 1 0 1024 10 1 1 1 1 1 1 i 1 0 200 e manual_src~dfsg/examples/pan2.csd0000644000000000000000000000114612262561504016120 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pan2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kline line 0, p3, 1 ; straight line ain oscili .6, 440, giSine ; audio signal.. aL,aR pan2 ain, kline ; sent across image outs aL, aR endin i1 0 5 e manual_src~dfsg/examples/fareyleni.csd0000644000000000000000000000361412262561504017240 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc for RT audio input as well ; For Non-realtime ouput leave only the line below: ; -o fareyleni.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; initialise integer n for Farey Sequence F_8 gifarn init 8 ; calculate length of F_8, it should return 23 for |F_8| gires fareyleni gifarn ; convert to negative number for the GEN routine because ; the table length won't be a power of 2 ; (The length of a Farey Sequence of n > 1 is always an odd number) gilen init gires * -1 ; create F_8 with GENfarey, the negative table number prevents ; unnecessary normalisation (F_8 IS already normalised) ; n mode: gifarey ftgen 100, 0, gilen, "farey", gifarn, 4 ; if mode=4 then 1 is added to each element of F_n. ; Useful for creating just tuning tables that can be read by the cps2pch opcode. instr 1 ; the very last element of F_n is not needed in the case of tuning tables ires = gires - 1 ; read out and print to file kndx init 0 if (kndx < ires) then kelem tab kndx, gifarey fprintks "farey8_tuning.txt", "%2.6f\\n", kelem kndx = kndx+1 endif endin instr 2 ip cps2pch p4, -100 asig poscil .5, ip, 1 aenv linseg 0, 0.1, 1, p3-0.2, 1, 0.1, 0 outs asig * aenv, asig * aenv endin f1 0 8192 10 1 ;sine wave i1 0 .1 i2 1 .5 8.00 i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + 1 8.22 e manual_src~dfsg/examples/wgpluck.csd0000644000000000000000000000136212262561504016734 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o wgpluck.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 icps = 220 iamp = 20000 kpick = 0.5 iplk = 0 idamp = 10 ifilt = 1000 axcite oscil 1, 1, 1 apluck wgpluck icps, iamp, kpick, iplk, idamp, ifilt, axcite out apluck endin ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/trlowest.csd0000644000000000000000000000136112262561504017142 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trlowest.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "beats.wav", 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .003, 1, 3, 500 ; partial tracking flow,kfr,kamp trlowest fst, 1 ; lowest freq-track aout tradsyn flow, 1, 1, 1, 1 ; resynthesis of lowest frequency outs aout*2, aout*2 endin f1 0 8192 10 1 ;sine wave i 1 0 3 e manual_src~dfsg/examples/genexp.csd0000644000000000000000000000155212262561504016547 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o genexp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aswp linseg 0.01, p3*.5, .49, p3*.5, 0.01 ;index sweep function aindex poscil aswp, 110, 1 ;sound to waveshape atable tablei aindex, p4, 1, .5 ;waveshape index aenv linen 0.8, .01, p3, .02 ;amplitude envelope asig = (atable*aenv)*p5 ;impose envelope and scale asig dcblock2 asig ;get rid of DC outs asig, asig endin f 1 0 8192 10 1 ;sine wave f 2 0 8192 "exp" 0 15 0 f 3 0 8192 "exp" 0 3 0 i1 0 3 2 2 i1 + 3 3 3 e manual_src~dfsg/examples/ftgen.csd0000644000000000000000000000145412262561504016365 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftgen.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 gisine ftgen 1, 0, 16384, 10, 1 ;sine wave gisquare ftgen 2, 0, 16384, 10, 1, 0 , .33, 0, .2 , 0, .14, 0 , .11, 0, .09 ;odd harmonics gisaw ftgen 3, 0, 16384, 10, 0, .2, 0, .4, 0, .6, 0, .8, 0, 1, 0, .8, 0, .6, 0, .4, 0,.2 ;even harmonics instr 1 ifn = p4 asig poscil .6, 200, ifn outs asig, asig endin i 1 0 2 1 ;sine wave i 1 3 2 2 ;odd harmonics i 1 6 2 3 ;even harmonics e manual_src~dfsg/examples/nchnls.csd0000644000000000000000000000125012262561504016541 0ustar rootroot ; Select audio/midi flags here according to platform -odac -idac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; nchnls.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 ;two channels out 0dbfs = 1 instr 1 ain1, ainr ins ;grab your mic and sing adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time = 20ms aoutl flanger ain1, adel, .7 aoutr flanger ain1, adel*2, .8 fout "in_s.wav", 14, aoutl, aoutr ;write to stereo file, outs aoutl, aoutr ;16 bits with header endin i 1 0 10 e manual_src~dfsg/examples/vincr-complex.csd0000644000000000000000000000200012262561504020034 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vincr-complex.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gaReverbSend init 0 instr 1 iamp = p4 ifreq = p5 aenv linseg 0.0, 0.1*p3, iamp, 0.6*p3, iamp, 0.3*p3, 0.0 aosc poscil aenv, ifreq, 1 vincr gaReverbSend, aosc endin instr 2 ; global reverb instrument al, ar reverbsc gaReverbSend, gaReverbSend, 0.85, 12000 outs gaReverbSend+al, gaReverbSend+ar clear gaReverbSend endin f1 0 4096 10 1 { 4 CNT { 8 PARTIAL ; start time duration amplitude frequency i1 [0.5 * $CNT.] [1 + ($CNT * 0.2)] [.04 + (~ * .02)] [800 + (200 * $CNT.) + ($PARTIAL. * 20)] } } i2 0 6 e manual_src~dfsg/examples/vadd_i.csd0000644000000000000000000000146512262561504016512 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vadd_i ifn1, ival, ielements, idstoffset endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e manual_src~dfsg/examples/tablexkt.csd0000644000000000000000000000220312262561504017071 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tablexkt.wav -W ;;; for file output any platform ;Example by Jonathan Murphy sr = 44100 ksmps = 10 nchnls = 1 instr 1 ifn = 1 ; query f1 as to number of samples ilen = nsamp(ifn) itrns = 4 ; transpose up 4 octaves ilps = 16 ; allow iwsize/2 samples at start ilpe = ilen - 16 ; and at end imode = 3 ; loop forwards and backwards istrt = 16 ; start 16 samples into loop alphs lphasor itrns, ilps, ilpe, imode, istrt ; use lphasor as index andx = alphs kfn = 1 ; read f1 kwarp = 4 ; anti-aliasing, should be same value as itrns above iwsize = 32 ; iwsize must be at least 8 * kwarp atab tablexkt andx, kfn, kwarp, iwsize atab = atab * 10000 out atab endin f 1 0 262144 1 "beats.wav" 0 4 1 i1 0 60 e manual_src~dfsg/examples/vpowv.csd0000644000000000000000000000163712262561504016446 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ifn2 = p5 ielements = p6 idstoffset = p7 isrcoffset = p8 kval init 25 vpowv ifn1, ifn2, ielements, idstoffset, isrcoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin f 1 0 16 -7 1 16 17 f 2 0 16 -7 1 16 2 i2 0.0 0.2 1 i2 0.2 0.2 2 i1 0.4 0.01 1 2 5 3 8 i2 0.8 0.2 1 i1 1.0 0.01 1 2 5 10 -2 i2 1.2 0.2 1 i1 1.4 0.01 1 2 8 14 0 i2 1.6 0.2 1 i1 1.8 0.01 1 2 8 0 14 i2 2.0 0.2 1 e manual_src~dfsg/examples/fox.ats0000644000000000000000000360434012262561504016102 0ustar rootroot^@@{@@`a@`q@7?mZv@@ @@pmkT@F $>>V@"hX@[ĺX@ӛdUY@%޵cY@Bc\@")q`@urJb@V4d@xe@* f@;h@xg@+1j@-l@!U-bm@Tro@pK6Pq@_0r@zs@4jnt@f%6>u@aZjv@8Da2 x@m,w@!Vcx@z1/`{@i z@H)+|@f{%}@+:s~@A+@e@ !P@h?S߁@]4f@:އ@,(_Q@DB @29 ts@]n" :ć@ݗN"L@cj\@Qj\fԊ@$b{@q@*R@ǯz@wZ)#|@EJ۳a@p#A@J@zؼf@^ck[@Jx@H@ @s%ℴM@c@{^@Xqo@?@~@͛@}ٽ}@E Vy@_+@Jv@!>vޞ@)e@Hr@@ @;5@O@n^+@"5I@&F @Sa@Ωo@c4@#*@J@v@&h;JƦ@|a@o @>[1@x@@ٽ^@\wPE @r63@Ĺ-i@%@?@Mԅ@ ݓ]@T+H @G>pJ@zx@i {@GD@@Z<}uر@!]=<@@Y@@@#@RPc@7K~@XIM!!@rk@tb'@.*@JN a@GT\@ ]6~@8 @[aOw@w@e,@k'@p߶@@win@u @wW @ !k@*TTO@|Rvv`?!7[y>?`l{j?>>ԋ+a?ꨌH>F%>?eÿ>9E>#>W>`>,b`>8>$HA>^kѦ>NCENƺ>>.C>n>GWƓ>dyd >wʹ>r4ų>>^^g>{Gz?pmkT@F $>>V@"hX@[ĺX@ӛdUY@%޵cY@Bc\@")q`@urJb@V4d@xe@* f@;h@xg@+1j@-l@!U-bm@Tro@pK6Pq@_0r@zs@4jnt@f%6>u@aZjv@8Da2 x@m,w@!Vcx@z1/`{@i z@H)+|@f{%}@+:s~@A+@e@ !P@h?S߁@]4f@:އ@,(_Q@DB @29 ts@]n" :ć@ݗN"L@cj\@Qj\fԊ@$b{@q@*R@ǯz@wZ)#|@EJ۳a@p#A@J@zؼf@^ck[@Jx@H@ @s%ℴM@c@{^@Xqo@?@~@͛@}ٽ}@E Vy@_+@Jv@!>vޞ@)e@Hr@@ @;5@O@n^+@"5I@&F @Sa@Ωo@c4@#*@J@v@&h;JƦ@|a@o @>[1@x@@ٽ^@\wPE @r63@Ĺ-i@%@?@Mԅ@ ݓ]@T+H @G>pJ@zx@i {@GD@@Z<}uر@!]=<@@Y@@@#@RPc@7K~@XIM!!@rk@tb'@.*@JN a@GT\@ ]6~@8 @[aOw@w@e,@k'@p߶@@win@u @wW @ !k@*TTO@$c?! p;?t29 ?>RNS> ?z{ ?B>ֈ,>o5>#g.K>vTdgC>Yy6?>V)>OiѴt>"p>bCw>bŽ>BX"g>LA>aF%B>ĺ>59>;_ @>ak>43>{Gz?pmkT@F $>>V@"hX@[ĺX@ӛdUY@%޵cY@Bc\@")q`@urJb@V4d@xe@* f@;h@xg@+1j@-l@!U-bm@Tro@pK6Pq@_0r@zs@4jnt@f%6>u@aZjv@8Da2 x@m,w@!Vcx@z1/`{@i z@H)+|@f{%}@+:s~@A+@e@ !P@h?S߁@]4f@:އ@,(_Q@DB @29 ts@]n" :ć@ݗN"L@cj\@Qj\fԊ@$b{@q@*R@ǯz@wZ)#|@EJ۳a@p#A@J@zؼf@^ck[@Jx@H@ @s%ℴM@c@{^@Xqo@?@~@͛@}ٽ}@E Vy@_+@Jv@!>vޞ@)e@Hr@@ @;5@O@n^+@"5I@&F @Sa@Ωo@c4@#*@J@v@&h;JƦ@|a@o @>[1@x@@ٽ^@\wPE @r63@Ĺ-i@%@?@Mԅ@ ݓ]@T+H @G>pJ@zx@i {@GD@@Z<}uر@!]=<@@Y@@@#@RPc@7K~@XIM!!@rk@tb'@.*@JN a@GT\@ ]6~@8 @[aOw@w@e,@k'@p߶@@win@u @wW @ !k@*TTO@ GKsj?1ߩPF?n?1?2 1?13fGB?312?M),?֯y,?{ ?*?%Z*?%b2? چ1?@".?xQ] /?D?CY,?%Unm4?ʑwz4?#?$?D^'H?bptP?76u$ .?Q?pmkT@F $>>V@"hX@[ĺX@ӛdUY@%޵cY@dU\ Bc\@")q`@urJb@V4d@xe@* f@;h@xg@+1j@-l@!U-bm@Tro@pK6Pq@_0r@zs@P6<>?4jnt@f%6>u@aZjv@8Da2 x@m,w@OUT?!Vcx@cVudjz1/`{@i z@H)+|@f{%}@+:s~@A+@?e@ !P@h?S߁@]4f@:އ@p6tU],(_Q@DB @29 ts@]n" :ć@ݗN"L@r 0@cj\@Qj\fԊ@$b{@q@h*R@ǯz@wZ)#|@EJ۳a@p#A@J@zؼf@^ck[@Jx@H@,׿ @s%ℴM@c@{^@Xe@Xqo@?@~sc~@͛@}ٽ}@E Vy@_+@Jv@!>vޞ@)e@SJRHr@@ @;5@O@n^+@"5I@IGMA @&F @Sa@?Ωo@c4@#*@J@v@ !&h;JƦ@|a@o @>[1@x@@ٽ^@\wPE @Fcۧ?r63@Ĺ-i@o?%@^ d#?@Mԅ@ ݓ]@T+H @G>pJ@zx@i {@GD@@Z<}uر@!]=<@ [?@Y@@@#@RPc@7K~@XIM!!@rk@tb'@.*@JN a@GT\@ ]6~@8 @[aOw@w@e,@k'@p߶@@win@u @wW @ !k@*TTO@Iw^i?:'V?D-k(J?X?a`Y,Q?%\O? 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VMMٿKpT@w9:?p?^W@ x?, V@$6N_#QW@` :mQZ@dU\ ðt>\@OA] w4 g*\]@.bп(5 b@EhTrc@tQ?n*;e@8%ckk;vd@I{:%f@O@m6wh@?\6Aj@~\UKo_l@q!@>n@^Qm @hajq@Ȕy\? p@QHFY?4q@).c̿PKr@P6<>?e:[xt@Fڛm@Q>VHbe@cӾOΉe@R] ?_3˥@ !a1PGȅ@1+@uXç@yy Hv@ٺ|k@]/15@0<1fv@=7-ȭ @(9W@C,+@֟ @g3O|@Fcۧ?Όm@l?۽T t@o?n@^ d#WvXX@A\* @r@|k2@îbC`gcʮ@dsi@1ϡT@B@w@| @( =@w 5E@2"8@C Ek?k쑋k@ [?Ig@*Ya?ڋ"@7e,Bs@ bb@ )?0 $@0l&_&@3hUϰ@O2j@89!Գ@cl1M@qA2=@~?@1S@x2(Kh?o* kO@5,)k'@"{Ͳ0?+T8w@)% @t{Sݭ&@M@3Y@4Bt7ٿ3@“xhr?3|r@:iQ@\edk@;O@; ?]Ț]Ż@, jjwd@:5;?ė@T21@$@ ?3@57*@k?PFl@W?yӥG@؃kL(@(ݫ@A ?v G@+hZ?G~?Vb@S?F @~VG'@Aeut~@=;9?ҼvL@@?KjY?'Y?IN8? :*?Xs?Ne?I!?^/ ?|j ?jۧ>'0G[?hij=?͌8?/u!w?` ?=>W?R>>&>\n>9Cu>^I;>^}z>g/ >Q@-U@KpT@p?^W@, V@6N_#QW@QZ@ðt>\@4 g*\]@(5 b@Trc@n*;e@k;vd@%f@m6wh@\6Aj@Ko_l@>n@hajq@ p@4q@PKr@e:[xt@Q?ة;D1?dF?8]q*>D-h>:5j>:>u6>LR85>l;r?_"?Ž0a(?b%9>W^D>i>YC>{> if>7y~`>XL.>l>Ku2B>'!ˈ>|x_۫>@-U@KpT@p?^W@, V@6N_#QW@QZ@ðt>\@4 g*\]@(5 b@Trc@n*;e@k;vd@%f@m6wh@\6Aj@Ko_l@>n@hajq@ p@4q@PKr@e:[xt@Qf>EfN>(ƙ{>cXI>8~>k:sB&>[%>ɪiU6>*H>KUCq>q">~e0>Ú1>8V>;ɍc>ܸ0ږ>m9ݡ>/z/>i}>Ew>Gz@-U@KpT@p?^W@, V@6N_#QW@QZ@ðt>\@4 g*\]@(5 b@Trc@n*;e@k;vd@%f@m6wh@\6Aj@Ko_l@>n@hajq@ p@4q@PKr@e:[xt@Qx`Bb(C>ɴV}>*چGR>D>x7>>R;:ʦ>zV3>x5X)>ys>M-m>4/*> bf>Wk> \>qnR>éw>UÈ>NyL8>rJ|>-L>R>s\#ҕ> X>)\(@-U@KpT@p?^W@, V@6N_#QW@QZ@ðt>\@4 g*\]@(5 b@Trc@n*;e@k;vd@%f@m6wh@\6Aj@Ko_l@>n@hajq@ p@4q@PKr@e:[xt@Q h>RX>N9+>.><Ʃ>Ml>S!$>m8>;$_u >)>!Ӣ>1j>}!t>X*4z>'{Oj>5,iw>K Sc>*6_>XKi>[Bj>*4Il>Op>;RB>manual_src~dfsg/examples/dbfsamp.csd0000644000000000000000000000110512262561504016667 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dbfsamp.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 iamp = p4 idb = dbfsamp(iamp) print idb asig vco2 iamp, 110 ;sawtooth outs asig, asig endin i 1 0 1 1 i 1 + 1 100 i 1 + 1 1000 i 1 + 1 10000 i 1 + 1 30000 e manual_src~dfsg/examples/sfpreset.csd0000644000000000000000000000253512262561504017116 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out, virtual midi in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfpreset.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf1 sfload "sf_GMbank.sf2" sfplist gisf1 ;list presets of first soundfont gisf2 sfload "07AcousticGuitar.sf2" sfplist gisf2 ;list presets of second soundfont gir sfpreset 50, 0, gisf1, 0 ;assign Synth Strings to index 0 giv sfpreset 0, 0, gisf2, 1 ;assign AcousticGuitar to index 1 print gir print giv instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/5000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay3 ivel, inum, kamp*ivel, kfreq, p6 outs a1, a2 endin f0 60 ; stay active for 1 minute i1 0 1 60 127 0 ;= Synth Strings I from first soundfont i1 + 1 62 < . i1 + 1 65 < . i1 + 1 69 10 . i1 5 1 60 127 1 ;= AcousticGuitar from second soundfont i1 + 1 62 < . i1 + 1 65 < . i1 + 1 69 10 . e manual_src~dfsg/examples/pvadd.csd0000644000000000000000000000127512262561504016361 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvadd.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; analyze "fox.wav" with PVANAL first igatefn = p4 ktime line 0, p3, p3 asig pvadd ktime, 1, "fox.pvx", 1, 300, 2, 2, 0, 0, igatefn outs asig*3, asig*3 endin f 1 0 16384 10 1 ;sine wave f 2 0 512 5 1 256 .001 f 3 0 512 7 0 256 1 256 1 i 1 0 2.8 2 i 1 + 2.8 3 e manual_src~dfsg/examples/vbap4move.csd0000644000000000000000000000123612262561504017163 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbap4move.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 vbaplsinit 2, 4, 0, 90, 180, 270 instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav a1,a2,a3,a4 vbap4move asig, p3, 1, 2, 310, 180 ;change movement of soundsource in outq a1,a2,a3,a4 ;the rear speakers endin i 1 0 5 e manual_src~dfsg/examples/bformenc.csd0000644000000000000000000000230212262561504017046 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages ;-odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -o bformenc.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 8 ;bformenc is deprecated, please use bformenc1 instr 1 ; generate pink noise anoise pinkish 1000 ; two full turns kalpha line 0, p3, 720 kbeta = 0 ; fade ambisonic order from 2nd to 0th during second turn kord0 = 1 kord1 linseg 1, p3 / 2, 1, p3 / 2, 0 kord2 linseg 1, p3 / 2, 1, p3 / 2, 0 ; generate B format aw, ax, ay, az, ar, as, at, au, av bformenc anoise, kalpha, kbeta, kord0, kord1, kord2 ; decode B format for 8 channel circle loudspeaker setup a1, a2, a3, a4, a5, a6, a7, a8 bformdec 4, aw, ax, ay, az, ar, as, at, au, av ; write audio out outo a1, a2, a3, a4, a5, a6, a7, a8 endin ; Play Instrument #1 for 20 seconds. i 1 0 20 e manual_src~dfsg/examples/fouti.csd0000644000000000000000000000106712262561504016410 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fouti.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gihand fiopen "test.txt", 0 instr 1 ires random 0, 10 fouti gihand, 0, 1, ires ficlose gihand endin i 1 0 1 e manual_src~dfsg/examples/midion_simple.csd0000644000000000000000000000137512262561504020114 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1 ;;;RT audio I/O with MIDI in sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 instr 1 ;Triggered by MIDI notes on channel 1 ifund notnum ivel veloc knote1 init ifund knote2 init ifund + 3 knote3 init ifund + 5 ;minor chord on MIDI out channel 1 ;Needs something plugged to csound's MIDI output midion 1, knote1,ivel midion 1, knote2,ivel midion 1, knote3,ivel endin ; Dummy ftable f0 60 manual_src~dfsg/examples/bitwise.csd0000644000000000000000000000136512262561504016731 0ustar rootroot sr = 44100 ksmps = 32 nchnls = 2 instr 1 iresultOr = p4 | p5 iresultAnd = p4 & p5 prints "%i | %i = %i\\n", p4, p5, iresultOr prints "%i & %i = %i\\n", p4, p5, iresultAnd endin instr 2 ; decimal to binary converter Sbinary = "" inumbits = 8 icount init inumbits - 1 pass: ivalue = 2 ^ icount if (p4 & ivalue >= ivalue) then Sdigit = "1" else Sdigit = "0" endif Sbinary strcat Sbinary, Sdigit loop_ge icount, 1, 0, pass Stext sprintf "%i is %s in binary\\n", p4, Sbinary prints Stext endin i 1 0 0.1 1 2 i 1 + . 1 3 i 1 + . 2 4 i 1 + . 3 10 i 2 2 0.1 12 i 2 + . 9 i 2 + . 15 i 2 + . 49 e manual_src~dfsg/examples/gen04.csd0000644000000000000000000000314512262561504016176 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen04.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisine ftgen 0, 0, 16384, 10, 1 ;sine wave instr 1 ihertz = cpspch(p4) ipkamp = p5 iwsfn = p6 ;waveshaping function inmfn = p7 ;normalization function agate linen 1, .01, p3, .1 ;overall amp envelope kctrl linen .9, 2, p3, 2 ;waveshaping index control aindex poscil kctrl/2, ihertz, gisine ;sine wave to be distorted asignal tablei .5+aindex, iwsfn, 1 ;waveshaping knormal tablei 1/kctrl, inmfn , 1 ;amplitude normalization outs asignal*knormal*ipkamp*agate, asignal*knormal*ipkamp*agate endin f1 0 64 21 6 ;Gaussian (random) distribution f2 0 33 4 1 1 ;normalizing function with midpoint bipolar offset s ; st dur pch amp wsfn inmfn i1 0 4 6.00 .7 1 2 i1 4 . 7.00 . i1 8 . 8.00 . ;------------------------------------------------------------------------------------- f3 0 1025 13 1 1 0 5 0 5 0 10 ;Chebyshev algorithm f4 0 513 4 3 1 ;normalizing function with midpoint bipolar offset s ; st dur pch amp wsfn inmfn i1 0 4 6.00 .9 3 4 i1 4 . 7.00 . i1 8 . 8.00 . e manual_src~dfsg/examples/mute.csd0000644000000000000000000000130612262561504016230 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mute.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; Mute Instrument #2. mute 2 ; Mute Instrument three. mute "three" instr 1 a1 oscils 0.2, 440, 0 outs a1, a1 endin instr 2 ; gets muted a1 oscils 0.2, 880, 0 outs a1, a1 endin instr three ; gets muted a1 oscils 0.2, 1000, 0 outs a1, a1 endin i 1 0 1 i 2 0 1 i "three" 0 1 e manual_src~dfsg/examples/buzz.csd0000644000000000000000000000110612262561504016246 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o buzz.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 110 ifn = 1 knh line p4, p3, p5 asig buzz 1, kcps, knh, ifn outs asig, asig endin ;sine wave. f 1 0 16384 10 1 i 1 0 3 20 20 i 1 + 3 3 3 i 1 + 3 10 1 e manual_src~dfsg/examples/cps2pch_ftable.csd0000644000000000000000000000136612262561504020143 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cps2pch_ftable.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ipch = 8.02 ; Use Table #1, a table of frequency multipliers. icps cps2pch ipch, -1 print icps endin ; Table #1: a table of frequency multipliers. ; Creates a 10-note scale of unequal divisions. f 1 0 16 -2 1 1.1 1.2 1.3 1.4 1.6 1.7 1.8 1.9 ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/bformenc1.csd0000644000000000000000000000272612262561504017141 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages ;-odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -o bformenc.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 8 0dbfs = 1 instr 1 ;without arrays ; generate pink noise anoise pinkish 1 ; two full turns kalpha line 0, p3, 720 kbeta = 0 ; generate B format aw, ax, ay, az, ar, as, at, au, av bformenc1 anoise, kalpha, kbeta ; decode B format for 8 channel circle loudspeaker setup a1, a2, a3, a4, a5, a6, a7, a8 bformdec1 4, aw, ax, ay, az, ar, as, at, au, av ; write audio out outo a1, a2, a3, a4, a5, a6, a7, a8 endin instr 2 ;with arrays (csound6) ;set file names for: S_bform = "bform_out.wav" ;b-format (2nd order) output S_sound = "sound_out.wav" ;sound output ; generate pink noise anoise pinkish 1 ; two full turns kalpha line 0, p3, 720 kbeta = 0 ;create array for B-format 2nd order (9 chnls) aBform[] init 9 ; generate B-format aBform bformenc1 anoise, kalpha, kbeta ;write out b-format fout "fout.wav", 18, aBform ;create array for audio output (8 channels) aAudio[] init 8 ;decode B format for 8 channel circle loudspeaker setup aAudio bformdec1 4, aBform ; write audio out fout S_sound, 18, aAudio endin i 1 0 8 i 2 8 8 e manual_src~dfsg/examples/scogen.csd0000644000000000000000000001702412262561504016540 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages ; -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -n ;Don't write audio ouput to disk ;=========================================================== ; scogen.csd by: Matt Ingalls ; ; a "port" of sorts ; of the old "mills" score generator (scogen) ; ; this instrument creates a schottstaedt.sco file ; to be used with the schottstaedt.orc file ; ; as long as you dont save schottstaedt.orc as a .csd ; file, you should be able to keep it open in MacCsound ; and render each newly generated .sco file. ; ;=========================================================== gScoName = "/Users/matt/Desktop/schottstaedt.sco" ; the name of the file to be generated sr = 100 ; this defines our temporal resolution, ; an sr of 100 means we will generate p2 and p3 values ; to the nearest 1/100th of a second ksmps = 1 ; set kr=sr so we can do everything at k-rate ; some print opcodes opcode PrintInteger, 0, k kval xin fprintks gScoName, "%d", kval endop opcode PrintFloat, 0, k kval xin fprintks gScoName, "%f", kval endop opcode PrintTab, 0, 0 fprintks gScoName, "%n" endop opcode PrintReturn, 0, 0 fprintks gScoName, "%r" endop ; recursively calling opcode to handle all the optional parameters opcode ProcessAdditionalPfields, 0, ikio iPtable, kndx, iNumPfields, iPfield xin ; additional pfields start at 5, we use a default 0 to identify the first call iPfield = (iPfield == 0 ? 5 : iPfield) if (iPfield > iNumPfields) goto endloop ; find our tables iMinTable table 2*iPfield-1, iPtable iMaxTable table 2*iPfield, iPtable ; get values from our tables kMin tablei kndx, iMinTable kMax tablei kndx, iMaxTable ; find a random value in the range and write it to the score fprintks gScoName, "%t%f", kMin + rnd(kMax-kMin) ; recursively call for any additional pfields. ProcessAdditionalPfields iPtable, kndx, iNumPfields, iPfield + 1 endloop: endop /* =========================================================== Generate a gesture of i-statements p2 = start of the gesture p3 = duration of the gesture p4 = number of a function that contains a list of all function table numbers used to define the pfield random distribution p5 = scale generated p4 values according to density (0=off, 1=on) [todo] p6 = let durations overlap gesture duration (0=off, 1=on) [todo] p7 = seed for random number generator seed [todo] =========================================================== */ instr Gesture ; initialize iResolution = 1/sr kNextStart init p2 kCurrentTime init p2 iNumPfields table 0, p4 iInstrMinTable table 1, p4 iInstrMaxTable table 2, p4 iDensityMinTable table 3, p4 iDensityMaxTable table 4, p4 iDurMinTable table 5, p4 iDurMaxTable table 6, p4 iAmpMinTable table 7, p4 iAmpMaxTable table 8, p4 ; check to make sure there is enough data print iNumPfields if iNumPfields < 4 then prints "%dError: At least 4 p-fields (8 functions) need to be specified.%n", iNumPfields turnoff endif ; initial comment fprints gScoName, "%!Generated Gesture from %f to %f seconds%n %!%t%twith a p-max of %d%n%n", p2, p3, iNumPfields ; k-rate stuff if (kCurrentTime >= kNextStart) then ; write a new note! kndx = (kCurrentTime-p2)/p3 ; get the required pfield ranges kInstMin tablei kndx, iInstrMinTable kInstMax tablei kndx, iInstrMaxTable kDensMin tablei kndx, iDensityMinTable kDensMax tablei kndx, iDensityMaxTable kDurMin tablei kndx, iDurMinTable kDurMax tablei kndx, iDurMaxTable kAmpMin tablei kndx, iAmpMinTable kAmpMax tablei kndx, iAmpMaxTable ; find random values for all our required parametrs and print the i-statement fprintks gScoName, "i%d%t%f%t%f%t%f", kInstMin + rnd(kInstMax-kInstMin), kNextStart, kDurMin + rnd(kDurMax-kDurMin), kAmpMin + rnd(kAmpMax-kAmpMin) ; now any additional pfields ProcessAdditionalPfields p4, kndx, iNumPfields PrintReturn ; calculate next starttime kDensity = kDensMin + rnd(kDensMax-kDensMin) if (kDensity < iResolution) then kDensity = iResolution endif kNextStart = kNextStart + kDensity endif kCurrentTime = kCurrentTime + iResolution endin /* =========================================================== scogen.sco this csound module generates a score file you specify a gesture of notes by giving the "gesture" instrument a number to a (negative) gen2 table. this table stores numbers to pairs of functions. each function-pair represents a range (min-max) of randomness for every pfield for the notes to be generated. =========================================================== */ ; common tables for pfield ranges f100 0 2 -7 0 2 0 ; static 0 f101 0 2 -7 1 2 1 ; static 1 f102 0 2 -7 0 2 1 ; ramp 0->1 f103 0 2 -7 1 2 0 ; ramp 1->0 f105 0 2 -7 10 2 10 ; static 10 f106 0 2 -7 .1 2 .1 ; static .1 ; specific pfield ranges f10 0 2 -7 .8 2 .01 ; density f11 0 2 -7 8 2 4 ; pitchmin f12 0 2 -7 8 2 12 ; pitchmax ;=== table containing the function numbers used for all the p-field distributions ; ; p1 - table number ; p2 - time table is instantiated ; p3 - size of table (must be >= p5!) ; p4 - gen# (should be = -2) ; p5 - number of pfields of each note to be generated ; p6 - table number of the function representing the minimum possible note number (p1) of a generated note ; p7 - table number of the function representing the maximum possible note number (p1) of a generated note ; p8 - table number of the function representing the minimum possible noteon-to-noteon time (p2 density) of a generated note ; p9 - table number of the function representing the maximum possible noteon-to-noteon time (p2 density) of a generated note ; p10 - table number of the function representing the minimum possible duration (p3) of a generated note ; p11 - table number of the function representing the maximum possible duration (p3) of a generated note ; p12 - table number of the function representing the maximum possible amplitude (p4) of a generated note ; p13 - table number of the function representing the maximum possible amplitude (p5) of a generated note ; p14,p16.. - table number of the function representing the minimum possible value for additional pfields (p5,p6..) of a generated note ; p15,p17.. - table number of the function representing the maximum possible value for additional pfields (p5,p6..) of a generated note ; siz 2 #pds p1min p1max p2min p2max p3min p3max p4min p4max p5min p5max p6min p6max f1 0 32 -2 6 101 101 10 10 101 105 100 106 11 12 100 101 ;gesture definitions ; start dur pTble scale overlap seed i"Gesture" 0 60 1 ;todo-->0 0 123 manual_src~dfsg/examples/follow2.csd0000644000000000000000000000137212262561504016645 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o follow2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig soundin "beats.wav" outs asig, asig endin instr 2 ;using follow2 as soundin "beats.wav" af follow2 as, p4, p5 ar rand 44100 ;noise ; "beats.wav" provides amplitude for noise asig balance ar, af outs asig, asig endin i 1 0 2 i 2 2 2 0.001 0.01 ;quick attack & deacy i 2 5 2 0.1 0.5 ;slow attack & deacy e manual_src~dfsg/examples/cossegr.csd0000644000000000000000000000123112262561504016720 0ustar rootroot ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o linsegr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .3 kenv cossegr 1, .05, 0.5, 1, 0 asig pluck kenv, icps, 200, 1, 1 outs asig, asig endin f 1 0 4096 10 1 ;sine wave f0 30 ;runs 30 seconds e manual_src~dfsg/examples/flanger.csd0000644000000000000000000000126112262561504016674 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o flanger.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfeedback = p4 asnd vco2 .2, 50 adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time =20ms aflg flanger asnd, adel, kfeedback asig clip aflg, 1, 1 outs asig+asnd, asig+asnd ;mix flanger with original endin i 1 0 10 .2 i 1 11 10 .8 ;lot of feedback e manual_src~dfsg/examples/fold.csd0000644000000000000000000000125212262561504016202 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fold.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig poscil3 .8, 400, 1 ;very clean sine kincr line p4, p3, p5 asig fold asig, kincr outs asig, asig endin ;sine wave. f 1 0 16384 10 1 i 1 0 4 2 2 i 1 5 4 5 5 i 1 10 4 10 10 i 1 15 4 1 100 ; Vary the fold-over amount from 1 to 100 e manual_src~dfsg/examples/ahh.aiff0000644000000000000000000000035612262561504016156 0ustar rootrootFORMAIFFCOMM\@ DSSND==Y +9A> 4%\IB  #.k= K'`nCTO ]0 0Ch[$= = ^%P M B k(manual_src~dfsg/examples/transegb.csd0000644000000000000000000000126712262561504017071 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o transeg.wav -W ;;; for file output any platform sr = 44100 ksmps = 128 nchnls = 2 0dbfs = 1 instr 1 ;p4 and p5 determine the type of curve for each ;section of the envelope kenv transegb 0.01, p3*0.25, p4, 1, p3, p5, 0.01 a1 oscil kenv, 440, 1 outs a1, a1 endin ; Table #1, a sine wave. f 1 0 16384 10 1 i 1 0 2 2 2 i 1 + . 5 5 i 1 + . 1 1 i 1 + . 0 0 i 1 + . -2 -2 i 1 + . -2 2 i 1 + . 2 -2 e manual_src~dfsg/examples/outiat.csd0000644000000000000000000000124612262561504016566 0ustar rootroot ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out and midi in and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outiat.wav -W ;;; for file output any platform sr = 44100 ksmps = 10 nchnls = 2 instr 1 ikey notnum ivel veloc ivib = 25 ;low value. outiat 1, ivib, 0, 127 ;assign aftertouch on print ivib ;external synth for example to midion 1, ikey, ivel ;change depth of filter modulation endin f0 30 ;play for 30 seconds e manual_src~dfsg/examples/outipb.csd0000644000000000000000000000112412262561504016556 0ustar rootroot ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out and midi in and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outipb.wav -W ;;; for file output any platform sr = 44100 ksmps = 10 nchnls = 2 instr 1 ikey notnum ivel veloc ipb = 10 ;a little out of tune outipb 1, ipb, 0, 127 ;(= pitchbend) midion 1, ikey, ivel ;of external synth endin f0 30 e manual_src~dfsg/examples/dconv.csd0000644000000000000000000000163512262561504016374 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o dconv.wav -W ;;; for file output any platform sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 #define RANDI(A) #kout randi 1, kfq, $A*.001+iseed, 1 tablew kout, $A, itable# instr 1 itable init 1 iseed init .6 isize init ftlen(itable) kfq line 1, p3, 10 $RANDI(0) $RANDI(1) $RANDI(2) $RANDI(3) $RANDI(4) $RANDI(5) $RANDI(6) $RANDI(7) $RANDI(8) $RANDI(9) $RANDI(10) $RANDI(11) $RANDI(12) $RANDI(13) $RANDI(14) $RANDI(15) asig rand 10000, .5, 1 asig butlp asig, 5000 asig dconv asig, isize, itable out asig *.5 endin f1 0 16 10 1 i1 0 10 e manual_src~dfsg/examples/strcatk.csd0000644000000000000000000000134012262561504016727 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o strcatk.wav -W ;;; for file output any platform sr = 48000 ksmps = 16 nchnls = 2 0dbfs = 1 ; Example by Jonathan Murphy 2007 instr 1 S1 = "1" S2 = " + 1" ktrig init 0 kval init 2 if (ktrig == 1) then S1 strcatk S1, S2 kval = kval + 1 endif String sprintfk "%s = %d", S1, kval puts String, kval ktrig metro 1 endin i1 0 10 e manual_src~dfsg/examples/mirror.csd0000644000000000000000000000204612262561504016572 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mirror.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; Limit / Mirror / Wrap igain = p4 ;gain ilevl1 = p5 ; + level ilevl2 = p6 ; - level imode = p7 ;1 = limit, 2 = mirror, 3 = wrap ain soundin "fox.wav" ain = ain*igain if imode = 1 goto limit if imode = 2 goto mirror asig wrap ain, ilevl2, ilevl1 goto outsignal limit: asig limit ain, ilevl2, ilevl1 goto outsignal mirror: asig mirror ain, ilevl2, ilevl1 outsignal: outs asig*.5, asig*.5 ;mind your speakers endin ; Gain +Levl -Levl Mode i1 0 3 4.00 .25 -1.00 1 ;limit i1 4 3 4.00 .25 -1.00 2 ;mirror i1 8 3 4.00 .25 -1.00 3 ;wrap e manual_src~dfsg/examples/pvscross.csd0000644000000000000000000000174312262561504017145 0ustar rootroot -odac sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 ipermut = p4; 1 = change order of soundfiles ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "fox.wav" Sfile2 = "wave.wav" ain1 soundin Sfile1 ain2 soundin Sfile2 fftin1 pvsanal ain1, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file 1 fftin2 pvsanal ain2, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file 2 ktrans linseg 0, p3, 1; linear transition if ipermut == 1 then fcross pvscross fftin2, fftin1, ktrans, 1-ktrans else fcross pvscross fftin1, fftin2, ktrans, 1-ktrans endif aout pvsynth fcross out aout endin i 1 0 2.757 0; frequencies from fox. wav; amplitudes moving from wave to fox i 1 3 2.757 1; frequencies from wave.wav, amplitudes moving from fox to wave e manual_src~dfsg/examples/meow.sdif0000644000000000000000000114260012262561504016405 0ustar rootrootSDIF1TYP1TYP 1MTD RBEL {Index, Label} 1MTD RBEP {Index, Frequency, Amplitude, Phase, Noise, TimeOffset} 1FTD RBEL { RBEL RABWE_Labels; } 1FTD RBEP { RBEP RABWE_Partials; } RBELRBEL??@@@@@@@@@@@@@@AAAAA A A0A0A@A@APAPA`A`ApApAAAAAAAAAAARBEP?htj~RBEPDR@ ?D@w@DY@c@@E#@ @E$l@v@EM@0 S@Es?@EH@#0AE=>υYAE(W@V0A E? 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Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc -nm0 ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform sr=44100 ksmps=128 nchnls=2 opcode TableDumpSimp, 0, ijo ;prints the content of a table in a simple way ifn, iprec, ippr xin; function table, float precision while printing (default = 3), parameters per row (default = 10, maximum = 32) iprec = (iprec == -1 ? 3 : iprec) ippr = (ippr == 0 ? 10 : ippr) iend = ftlen(ifn) indx = 0 Sformat sprintf "%%.%df\t", iprec Sdump = "" loop: ival tab_i indx, ifn Snew sprintf Sformat, ival Sdump strcat Sdump, Snew indx = indx + 1 imod = indx % ippr if imod == 0 then puts Sdump, 1 Sdump = "" endif if indx < iend igoto loop puts Sdump, 1 endop instr 1 ifn1 = p4 ifn2 = p5 ielements = p6 idstoffset = p7 isrcoffset = p8 kval init 25 vaddv ifn1, ifn2, ielements, idstoffset, isrcoffset, 1 turnoff endin instr 2 TableDumpSimp p4, 3, 16 endin f 1 0 16 -7 1 15 16 f 2 0 16 -7 1 15 2 i2 0.0 0.2 1 i2 0.2 0.2 2 i1 0.4 0.01 1 2 5 3 8 i2 0.8 0.2 1 i1 1.0 0.01 1 2 5 10 -2 i2 1.2 0.2 1 i1 1.4 0.01 1 2 8 14 0 i2 1.6 0.2 1 i1 1.8 0.01 1 2 8 0 14 i2 2.0 0.2 1 i1 2.2 0.002 1 1 8 5 2 i2 2.4 0.2 1 e manual_src~dfsg/examples/flooper.csd0000644000000000000000000000131512262561504016724 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o flooper.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kpitch line 1, p3, .9 ;lower pitch a bit during the note aout flooper .9, kpitch, 1, .53, 0.05, 1 ; loop starts at 1 sec, for .53 secs, 0.05 crossfade outs aout, aout endin ;table size is deferred, ; and format taken from the soundfile header f 1 0 0 1 "fox.wav" 0 0 0 i 1 0 8.2 e manual_src~dfsg/examples/STKWhistle.csd0000644000000000000000000000133712262561504017263 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKWhistle.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;Blowing Frequency Modulation kv3 = p7 ;Fipple Modulation Frequency asig STKWhistle cpspch(p4), 1, 4, 20, 11, kv3, 1, 100, 2, kv1, 128, 127 asig = asig*.7 ;too loud outs asig, asig endin i 1 0 .5 9.00 100 30 30 i 1 1 3 9.00 100 0 20 i 1 4.5 . 9.00 1 0 100 e manual_src~dfsg/examples/randi.csd0000644000000000000000000000144412262561504016356 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o randi.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;same values every time krnd randi 100, 10 printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin instr 2 ;different values every time krnd randi 100, 10, 10 ; seed from system clock printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin f 1 0 16384 10 1 ;sine wave. i 1 0 1 i 2 2 1 e manual_src~dfsg/examples/db.csd0000644000000000000000000000105112262561504015640 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o db.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 instr 1 idec = p4 iamp = db(idec) print iamp asig vco2 iamp, 110 ;sawtooth outs asig, asig endin i 1 0 1 50 i 1 + 1 > i 1 + 1 > i 1 + 1 85 e manual_src~dfsg/examples/pgmassign_ignore.csd0000644000000000000000000000123212262561504020607 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o pgmassign_ignore.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Ignore all program change events. pgmassign 0, -1 ; Instrument #1. instr 1 ; Just an example, no working code in here! endin ; Play Instrument #1 for one second. i 1 0 1 e manual_src~dfsg/examples/FLslider.csd0000644000000000000000000000311412262561504016761 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLslider.wav -W ;;; for file output any platform ; A sine with oscillator with flslider controlled frequency sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Frequency Slider", 900, 400, 50, 50 ; Minimum value output by the slider imin = 200 ; Maximum value output by the slider imax = 5000 ; Logarithmic type slider selected iexp = -1 ; Slider graphic type (5='nice' slider) itype = 5 ; Display handle (-1=not used) idisp = -1 ; Width of the slider in pixels iwidth = 750 ; Height of the slider in pixels iheight = 30 ; Distance of the left edge of the slider ; from the left edge of the panel ix = 125 ; Distance of the top edge of the slider ; from the top edge of the panel iy = 50 gkfreq, ihandle FLslider "Frequency", imin, imax, iexp, itype, idisp, iwidth, iheight, ix, iy ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun ;Set the widget's initial value FLsetVal_i 300, ihandle instr 1 iamp = 15000 ifn = 1 kfreq portk gkfreq, 0.005 ;Smooth gkfreq to avoid zipper noise asig oscili iamp, kfreq, ifn out asig endin ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e manual_src~dfsg/examples/sleighbells.csd0000644000000000000000000000132612262561504017555 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sleighbells.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idamp = p4 asig sleighbells .7, 0.01, 32, idamp outs asig, asig endin i 1 0.00 0.25 0 ;short sound i 1 0.30 0.25 i 1 0.60 0.25 i 1 0.90 0.25 i 1 1.20 0.25 i 1 1.50 1 .3 ;longer sound i 1 1.80 0.25 0 ;short sound again i 1 2.10 0.25 i 1 2.40 0.25 i 1 2.70 0.25 i 1 3.00 0.25 e manual_src~dfsg/examples/nlfilt2.csd0000644000000000000000000000164112262561504016632 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o nlfilt.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;unfiltered noise asig rand .7 outs asig, asig endin instr 2 ;filtered noise ka = p4 kb = p5 kd = p6 kC = p7 kL = p8 asig rand .3 afilt nlfilt2 asig, ka, kb, kd, kC, kL outs afilt, afilt endin i 1 0 2 ; unfiltered ; a b d C L i 2 2 2 0 0 0.8 0.5 20 ; non-linear effect i 2 + 2 .4 0.2 0.7 0.11 200 ; low=pass with non-linear i 2 + 2 0.35 -0.3 0.95 0.1 200 ; high-pass with non-linear i 2 + 2 0.7 -0.2 0.9 0.2 20 ; high-pass with non-linear e manual_src~dfsg/examples/atone.csd0000644000000000000000000000111112262561504016356 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o atone.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;white noise asig rand 1 outs asig, asig endin instr 2 ;filtered noise asig rand 1 khp init 4000 asig atone asig, khp outs asig, asig endin i 1 0 2 i 2 2 2 e manual_src~dfsg/examples/partikkelsync.csd0000644000000000000000000001435512262561504020151 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out -odac ;;;RT audio ; For Non-realtime ouput leave only the line below: ; -o partikkel_softsync.wav -W ;;; for file output any platform sr = 44100 ksmps = 20 nchnls = 2 ; Example by Oeyvind Brandtsegg 2007, revised 2008 giSine ftgen 0, 0, 65537, 10, 1 giCosine ftgen 0, 0, 8193, 9, 1, 1, 90 giSigmoRise ftgen 0, 0, 8193, 19, 0.5, 1, 270, 1 ; rising sigmoid giSigmoFall ftgen 0, 0, 8193, 19, 0.5, 1, 90, 1 ; falling sigmoid ; ************************************************* ; example of soft synchronization of two partikkel instances ; ************************************************* instr 1 /*score parameters*/ igrainrate = p4 ; grain rate igrainsize = p5 ; grain size in ms igrainFreq = p6 ; fundamental frequency of source waveform iosc2Dev = p7 ; partikkel instance 2 grain rate deviation factor iMaxSync = p8 ; max soft sync amount (increasing to this value during length of note) /*overall envelope*/ iattack = 0.001 idecay = 0.2 isustain = 0.7 irelease = 0.2 amp linsegr 0, iattack, 1, idecay, isustain, 1, isustain, irelease, 0 kgrainfreq = igrainrate ; grains per second kdistribution = 0 ; periodic grain distribution idisttab = -1 ; (default) flat distribution used ; for grain distribution async = 0 ; no sync input kenv2amt = 0 ; no secondary enveloping ienv2tab = -1 ; default secondary envelope (flat) ienv_attack = giSigmoRise ; default attack envelope (flat) ienv_decay = giSigmoFall ; default decay envelope (flat) ksustain_amount = 0.3 ; time (in fraction of grain dur) at ; sustain level for each grain ka_d_ratio = 0.2 ; balance between attack and decay time kduration = igrainsize ; set grain duration in ms kamp = 0.2*0dbfs ; amp igainmasks = -1 ; (default) no gain masking kwavfreq = igrainFreq ; fundamental frequency of source waveform ksweepshape = 0 ; shape of frequency sweep (0=no sweep) iwavfreqstarttab = -1 ; default frequency sweep start ; (value in table = 1, which give ; no frequency modification) iwavfreqendtab = -1 ; default frequency sweep end ; (value in table = 1, which give ; no frequency modification) awavfm = 0 ; no FM input ifmamptab = -1 ; default FM scaling (=1) kfmenv = -1 ; default FM envelope (flat) icosine = giCosine ; cosine ftable kTrainCps = kgrainfreq ; set trainlet cps equal to grain ; rate for single-cycle trainlet in ; each grain knumpartials = 3 ; number of partials in trainlet kchroma = 1 ; balance of partials in trainlet ichannelmasks = -1 ; (default) no channel masking, ; all grains to output 1 krandommask = 0 ; no random grain masking kwaveform1 = giSine ; source waveforms kwaveform2 = giSine ; kwaveform3 = giSine ; kwaveform4 = giSine ; iwaveamptab = -1 ; mix of 4 source waveforms and ; trainlets (set to default) asamplepos1 = 0 ; phase offset for reading source waveform asamplepos2 = 0 ; asamplepos3 = 0 ; asamplepos4 = 0 ; kwavekey1 = 1 ; original key for source waveform kwavekey2 = 1 ; kwavekey3 = 1 ; kwavekey4 = 1 ; imax_grains = 100 ; max grains per k period iopcode_id = 1 ; id of opcode, linking partikkel ; to partikkelsync a1 partikkel kgrainfreq, kdistribution, idisttab, async, kenv2amt, \ ienv2tab,ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, \ kduration, kamp, igainmasks, kwavfreq, ksweepshape, \ iwavfreqstarttab, iwavfreqendtab, awavfm, ifmamptab, kfmenv, \ icosine, kTrainCps, knumpartials, kchroma, ichannelmasks, \ krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, asamplepos4, \ kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains, iopcode_id async1 partikkelsync iopcode_id ; clock pulse output of the ; partikkel instance above ksyncGravity line 0, p3, iMaxSync ; strength of synchronization aphase2 init 0 asyncPolarity limit (int(aphase2*2)*2)-1, -1, 1 ; use the phase of partikkelsync instance 2 to find sync ; polarity for partikkel instance 2. ; If the phase of instance 2 is less than 0.5, we want to ; nudge it down when synchronizing, ; and if the phase is > 0.5 we want to nudge it upwards. async1 = async1*ksyncGravity*asyncPolarity ; prepare sync signal ; with polarity and strength kgrainfreq2 = igrainrate * iosc2Dev ; grains per second for second partikkel instance iopcode_id2 = 2 a2 partikkel kgrainfreq2, kdistribution, idisttab, async1, kenv2amt, \ ienv2tab, ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, \ kduration, kamp, igainmasks, kwavfreq, ksweepshape, \ iwavfreqstarttab, iwavfreqendtab, awavfm, ifmamptab, kfmenv, \ icosine, kTrainCps, knumpartials, kchroma, ichannelmasks, \ krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, \ asamplepos4, kwavekey1, kwavekey2, kwavekey3, kwavekey4, \ imax_grains, iopcode_id2 async2, aphase2 partikkelsync iopcode_id2 ; clock pulse and phase ; output of the partikkel instance above, ; we will only use the phase outs a1*amp, a2*amp endin /*score parameters igrainrate = p4 ; grain rate igrainsize = p5 ; grain size in ms igrainFreq = p6 ; frequency of source wave within grain iosc2Dev = p7 ; partikkel instance 2 grain rate deviation factor iMaxSync = p8 ; max soft sync amount (increasing to this value during length of note) */ ; GrRate GrSize GrFund Osc2Dev MaxSync i1 0 10 2 20 880 1.3 0.3 s i1 0 10 5 20 440 0.8 0.3 s i1 0 6 55 15 660 1.8 0.45 s i1 0 6 110 10 440 0.6 0.6 s i1 0 6 220 3 660 2.6 0.45 s i1 0 6 220 3 660 2.1 0.45 s i1 0 6 440 3 660 0.8 0.22 s e e manual_src~dfsg/examples/raises.csd0000644000000000000000000000144212262561504016545 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o ^.wav ; output to audio file sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Lo-Fi sound kpow = 10 ;exponent kbase line 1, p3, 2.2 ;vary the base kQuantize = kbase^kpow kQuantize = kQuantize*0.5 ;half the number of steps for each side of a bipolar signal printk2 kQuantize asig diskin2 "fox.wav", 1, 0, 1 ;loop the fox asig = round(asig * kQuantize) / kQuantize ;quantize and scale audio signal outs asig, asig endin i1 0 19.2 e manual_src~dfsg/examples/wguide2.csd0000644000000000000000000000170312262561504016625 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o wguide2.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aout diskin2 "beats.wav", 1, 0, 1 ;in signal afreq1 line 100, p3, 2000 afreq2 line 1200, p3, p4 ;vary second frequency in the score kcutoff1 = 3000 kcutoff2 = 1500 kfeedback1 = 0.25 ;the sum of the two feedback kfeedback2 = 0.25 ;values should not exceed 0.5 asig wguide2 aout, afreq1, afreq2, kcutoff1, kcutoff2, kfeedback1, kfeedback2 asig dcblock2 asig ;get rid of DC outs asig, asig endin i 1 0 8 1200 ;freqency of afreq2 remains the same i 1 9 8 100 ;freqency of afreq2 gets lower e manual_src~dfsg/examples/bexprnd.csd0000644000000000000000000000140212262561504016715 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o bexprnd.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values kexp bexprnd 100 printk .2, kexp ; look aout oscili 0.8, 440+kexp, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 kexp bexprnd 100 printk .2, kexp ; look aout oscili 0.8, 440+kexp, 1 ; & listen outs aout, aout endin ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e manual_src~dfsg/examples/repluck.csd0000644000000000000000000000151412262561504016724 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o repluck.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 iplk = 0.75 kamp = .8 icps = 110 krefl = p4 kpick = p5 axcite oscil 1, 1, 1 asig repluck iplk, kamp, icps, kpick, krefl, axcite asig dcblock2 asig ;get rid of DC offset outs asig, asig endin f 1 0 16384 10 1 ;sine wave. s i 1 0 1 0.95 0.75 ;sounds heavier (=p5) i 1 + 1 < i 1 + 1 < i 1 + 1 < i 1 + 10 0.6 s i 1 0 1 0.95 0.15 ;sounds softer (=p5) i 1 + 1 < i 1 + 1 < i 1 + 1 < i 1 + 10 0.6 e manual_src~dfsg/examples/pvsmorph2.csd0000644000000000000000000000444012262561504017220 0ustar rootroot -odac sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 ; this example uses the files "flute-C-octave0.wav" and ; "saxophone-alto-C-octave0.wav" from www.archive.org/details/OpenPathMusic44V2 giSine ftgen 0, 0, 4096, 10, 1 instr 1 iampint1 = p4; value for interpolating the amplitudes at the beginning ... iampint2 = p5; ... and at the end ifrqint1 = p6; value for unterpolating the frequencies at the beginning ... ifrqint2 = p7; ... and at the end kampint linseg iampint1, p3, iampint2 kfrqint linseg ifrqint1, p3, ifrqint2 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "flute-C-octave0.wav" Sfile2 = "saxophone-alto-C-octave0.wav" ain1 soundin Sfile1 ain2 soundin Sfile2 fftin1 pvsanal ain1, ifftsize, ioverlap, iwinsize, iwinshape fftin2 pvsanal ain2, ifftsize, ioverlap, iwinsize, iwinshape fmorph pvsmorph fftin1, fftin2, kampint, kfrqint aout pvsynth fmorph out aout * .5 endin instr 2; moving randomly in certain borders between two spectra iampintmin = p4; minimum value for amplitudes iampintmax = p5; maximum value for amplitudes ifrqintmin = p6; minimum value for frequencies ifrqintmax = p7; maximum value for frequencies imovefreq = p8; frequency for generating new random values kampint randomi iampintmin, iampintmax, imovefreq kfrqint randomi ifrqintmin, ifrqintmax, imovefreq ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "flute-C-octave0.wav" Sfile2 = "saxophone-alto-C-octave0.wav" ain1 soundin Sfile1 ain2 soundin Sfile2 fftin1 pvsanal ain1, ifftsize, ioverlap, iwinsize, iwinshape fftin2 pvsanal ain2, ifftsize, ioverlap, iwinsize, iwinshape fmorph pvsmorph fftin1, fftin2, kampint, kfrqint aout pvsynth fmorph out aout * .5 endin i 1 0 3 0 0 1 1; amplitudes from flute, frequencies from saxophone i 1 3 3 1 1 0 0; amplitudes from saxophone, frequencies from flute i 1 6 3 0 1 0 1; amplitudes and frequencies moving from flute to saxophone i 1 9 3 1 0 1 0; amplitudes and frequencies moving from saxophone to flute i 2 12 3 .2 .8 .2 .8 5; amps and freqs moving randomly between the two spectra e manual_src~dfsg/examples/define.csd0000644000000000000000000000147312262561504016515 0ustar rootroot ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o define.wav -W ;;; for file output any platform ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Define the macros. #define VOLUME #5000# #define FREQ #440# #define TABLE #1# ; Instrument #1 instr 1 ; Use the macros. ; This will be expanded to "a1 oscil 5000, 440, 1". a1 oscil $VOLUME, $FREQ, $TABLE ; Send it to the output. out a1 endin ; Define Table #1 with an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e manual_src~dfsg/examples/xadsr.csd0000644000000000000000000000136012262561504016377 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o xadsr.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iatt = p5 idec = p6 islev = p7 irel = p8 kenv xadsr iatt, idec, islev, irel kcps = cpspch(p4) ;frequency asig vco2 kenv * 0.8, kcps outs asig, asig endin i 1 0 1 7.00 .0001 1 .01 .001 ; short attack i 1 2 1 7.02 1 .5 .01 .001 ; long attack i 1 4 2 6.09 .0001 1 .1 .7 ; long release e manual_src~dfsg/examples/prepiano.csd0000644000000000000000000000150012262561504017067 0ustar rootroot ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o prepiano.wav -W ;;; for file output any platform sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;; fund NS detune stiffness decay loss (bndry) (hammer) scan prep aa,ab prepiano 60, 3, 10, p4, 3, 0.002, 2, 2, 1, 5000, -0.01, p5, p6, 0, 0.1, 1, 2 outs aa*.2, ab*.2 endin f1 0 8 2 1 0.6 10 100 0.001 ;; 1 rattle f2 0 8 2 1 0.7 50 500 1000 ;; 1 rubber i1 0.0 1 1 0.09 20 i1 1 . -1 0.09 40 ;; 1 -> skip initialisation i1 2 . -1 0.09 60 i1 3 . -1 0.09 80 i1 4 1.8 -1 0.09 100 e manual_src~dfsg/examples/128-stringcircular0000755000000000000000000010000012262561504020043 0ustar rootroot0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 1 0 1 0 0 0 0 0 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0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 1 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 manual_src~dfsg/examples-xml/0000755000000000000000000000000012262561504015361 5ustar rootrootmanual_src~dfsg/examples-xml/lposcila.csd.xml0000644000000000000000000000455112262561504020466 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lposcila.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 1.3 ;a 3d up kloop = 0 ;loop start time in samples kend = 10000 ;loop end time in samples aenv expsega 0.01, 0.1, 1, 0.1, 0.5, 0.5, 0.01 ;envelope with fast and short segment asig lposcila aenv, kcps, kloop, kend, 1 ;use it for amplitude outs asig, asig endin </CsInstruments> <CsScore> ; Its table size is deferred, ; and format taken from the soundfile header. f 1 0 0 1 "beats.wav" 0 0 0 ; Play Instrument #1 for 6 seconds. ; This will loop the drum pattern several times. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/distort1.csd.xml0000644000000000000000000000503512262561504020427 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o distort1.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gadist init 0 instr 1 iamp = p4 ifqc = cpspch(p5) asig pluck iamp, ifqc, ifqc, 0, 1 gadist = gadist + asig endin instr 50 kpre init p4 kpost init p5 kshap1 init p6 kshap2 init p7 aout distort1 gadist, kpre, kpost, kshap1, kshap2, 1 outs aout, aout gadist = 0 endin </CsInstruments> <CsScore> ; Sta Dur Amp Pitch i1 0.0 3.0 0.5 6.00 i1 0.5 2.5 0.5 7.00 i1 1.0 2.0 0.5 7.07 i1 1.5 1.5 0.5 8.00 ; Sta Dur PreGain PostGain Shape1 Shape2 i50 0 4 2 .5 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hsboscil.csd.xml0000644000000000000000000000436312262561504020467 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o hsboscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; synth waveform giwave ftgen 1, 0, 1024, 10, 1, 1, 1, 1 ; blending window giblend ftgen 2, 0, 1024, -19, 1, 0.5, 270, 0.5 instr 1 ; produces Risset's glissando. kamp = .4 kbrite = 0.3 ibasfreq = 200 ioctcnt = 5 ; Change ktone linearly from 0 to 1, ; over the period defined by p3. ktone line 0, p3, 1 asig hsboscil kamp, ktone, kbrite, ibasfreq, giwave, giblend, ioctcnt outs asig, asig endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/qinf.csd.xml0000644000000000000000000000446012262561504017614 0ustar rootroot <CsoundSynthesizer> <CsOptions> -+rtaudio=alsa -o dac:hw:0 </CsOptions> <CsInstruments> nchnls = 2 ksmps = 400 #define WII_B #3# #define WII_A #4# #define WII_R_A #304# #define WII_PITCH #20# #define WII_ROLL #21# gkcnt init 1 instr 1 i1 wiiconnect 3,1 wiirange $WII_PITCH., -20, 0 kt wiidata $WII_B. ka wiidata $WII_A. kra wiidata $WII_R_A. gka wiidata $WII_PITCH. gkp wiidata $WII_ROLL. ; If the B (trigger) button is pressed then activate a note if (kt==0) goto ee if (qinf(gka)) goto ee if (qinf(gkp)) goto ee event "i", 2, 0, 5 gkcnt = gkcnt + 1 printk2 kb endin instr 2 a1 oscil ampdbfs(gka), 440+gkp, 1 outs a1, a1 endin </CsInstruments> <CsScore> f1 0 4096 10 1 i1 0 300 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vmap.csd.xml0000644000000000000000000000674612262561504017633 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vmap.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ksmps = 256 nchnls = 2 gisize = 64 gitable ftgen 0, 0, gisize, 10, 1 ;Table to be processed gimap1 ftgen 0, 0, gisize, -7, gisize-1, gisize-1, 0 ; Mapping function to reverse table gimap2 ftgen 0, 0, gisize, -5, 1, gisize-1, gisize-1 ; Mapping function for PWM gimap3 ftgen 0, 0, gisize, -7, 1, (gisize/2)-1, gisize-1, 1, 1, (gisize/2)-1, gisize-1 ; Double frequency instr 1 ;Hear an oscillator using gitable asig oscil 10000, 440, gitable outs asig,asig endin instr 2 ;Reverse the table (no sound change, except for a single click vmap gimap1, gitable, gisize vcopy_i gitable, gimap1, gisize turnoff endin instr 3 ;Non-interpolated PWM (or phase waveshaping) vmap gimap2, gitable, gisize vcopy_i gitable, gimap2, gisize turnoff endin instr 4 ;Double frequency vmap gimap3, gitable, gisize vcopy_i gitable, gimap3, gisize turnoff endin </CsInstruments> <CsScore> i 1 0 8 i 2 2 1 i 3 4 1 i 4 6 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zkr.csd.xml0000644000000000000000000000577512262561504017477 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zkr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Linearly vary a k-rate signal from 440 to 880. kline line 440, p3, 880 ; Add the linear signal to zk variable #1. zkw kline, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read zk variable #1. kfreq zkr 1 ; Use the value of zk variable #1 to vary ; the frequency of a sine waveform. a1 oscil 20000, kfreq, 1 ; Generate the audio output. out a1 ; Clear the zk variables, get them ready for ; another pass. zkcl 0, 1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/trcross.csd.xml0000644000000000000000000000502212262561504020351 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trcross.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain1 diskin2 "beats.wav", 1, 0, 1 ain2 diskin2 "fox.wav", 1 imode = p4 fs1,fsi2 pvsifd ain1, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .01, 1, 3, 500 ; partial tracking fs11,fsi12 pvsifd ain2, 2048, 512, 1 ; ifd analysis (second input) fst1 partials fs11, fsi12, .01, 1, 3, 500 ; partial tracking (second input fcr trcross fst, fst1, 1.05, 1, imode ; cross-synthesis (mode 0 and mode 1) aout tradsyn fcr, 1, 1, 500, 1 ; resynthesis of tracks outs aout*3, aout*3 endin </CsInstruments> <CsScore> f1 0 8192 10 1 i 1 0 3 0 i 1 5 3 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hrtfmove.csd.xml0000644000000000000000000000554012262561504020511 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select flags here ; realtime audio out -o dac ; For Non-realtime ouput leave only the line below: ;-o hrtf.wav </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 gasrc init 0 instr 1 ;a plucked string kamp = p4 kcps = cpspch(p5) icps = cpspch(p5) a1 pluck kamp, kcps, icps, 0, 1 gasrc = a1 endin instr 10 ;uses output from instr1 as source kaz linseg 0, p3, 720 ;2 full rotations aleft,aright hrtfmove gasrc, kaz,0, "hrtf-44100-left.dat","hrtf-44100-right.dat" outs aleft, aright endin </CsInstruments> <CsScore> ; Play Instrument 1: a simple arpeggio i1 0 .2 15000 8.00 i1 + .2 15000 8.04 i1 + .2 15000 8.07 i1 + .2 15000 8.11 i1 + .2 15000 9.02 i1 + 1.5 15000 8.11 i1 + 1.5 15000 8.07 i1 + 1.5 15000 8.04 i1 + 1.5 15000 8.00 i1 + 1.5 15000 7.09 i1 + 1.5 15000 8.00 ; Play Instrument 10 for 10 seconds. i10 0 10 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vbaplsinit.csd.xml0000644000000000000000000000473612262561504021040 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbaplsinit.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 8 0dbfs = 1 vbaplsinit 2, 8, 0, 45, 90, 180, 270, 0, 0, 0 ;5 speakers for 5.1 amps instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav kazim line 1, p3, 355 a1,a2,a3,a4,a5,a6,a7,a8 vbap8 asig, kazim, 0, 1 ;change azimuth of soundsource ; Speaker mapping aFL = a1 ; Front Left aMF = a5 ; Mid Front aFR = a2 ; Front Right aBL = a3 ; Back Left aBR = a4 ; Back Right outo aFL,aFR,aBL,aBR,aMF,a6,a7,a8 ;a6, a7 and a8 are dummies endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsftr.csd.xml0000644000000000000000000000575112262561504020207 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsftr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifil ftgen 0, 0, 0, -1, "fox.wav", 0, 0, 1 instr 1 ifftsize = 1024 ;fft size ioverlap = 256 ;overlap knewamp = 0 ;new value for amplitudes ;create fsig stream from function table fsrc pvstanal 1, 1, 1, gifil, 0, 0, 0, ifftsize, ioverlap, 0 ifn ftgen 0, 0, ifftsize/2, 2, 0 ;create empty function table kflag pvsftw fsrc,ifn ;export amps to table ;overwrite the first 10 bins each time the table has been filled new if kflag == 1 then kndx = 0 kmaxbin = 10 loop: tablew knewamp, kndx, ifn loop_le kndx, 1, kmaxbin, loop endif pvsftr fsrc,ifn ;read modified data back to fsrc aout pvsynth fsrc ;and resynth outs aout, aout endin </CsInstruments> <CsScore> i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLbox.csd.xml0000644000000000000000000000535612262561504017676 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLbox.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Text Box", 700, 400, 50, 50 ; Box border type (7=embossed box) itype = 7 ; Font type (10='Times Bold') ifont = 10 ; Font size isize = 20 ; Width of the flbox iwidth = 400 ; Height of the flbox iheight = 30 ; Distance of the left edge of the flbox ; from the left edge of the panel ix = 150 ; Distance of the upper edge of the flbox ; from the upper edge of the panel iy = 100 ih3 FLbox "Use Text Boxes For Labelling", itype, ifont, isize, iwidth, iheight, ix, iy ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 endin </CsInstruments> <CsScore> ; Real-time performance for 1 hour. f 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/imageopcodes.csd.xml0000644000000000000000000000505312262561504021315 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n ;no sound output </CsOptions> <CsInstruments> sr=48000 ksmps=1 nchnls=2 ; this test .csd copies image.png into a new file 'imageout.png' giimage1 imageload "image.png" giimagew, giimageh imagesize giimage1 giimage2 imagecreate giimagew,giimageh instr 1 kndx = 0 kx_ linseg 0, p3, 1 ;print imagewidth and imageheigth of image.png prints "imagewidth = %f pixels, imageheigth = %f pixels\\n", giimagew, giimageh myloop: ky_ = kndx/(giimageh) kr_, kg_, kb_ imagegetpixel giimage1, kx_, ky_ imagesetpixel giimage2, kx_, ky_, kr_, kg_, kb_ loop_lt kndx, 0.5, giimageh, myloop endin instr 2 imagesave giimage2, "imageout.png" endin instr 3 imagefree giimage1 imagefree giimage2 endin </CsInstruments> <CsScore> i1 1 1 i2 2 1 i3 3 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/flooper2.csd.xml0000644000000000000000000000364612262561504020414 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o flooper2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; looping back and forth, 0.05 crossfade kst line .2, p3, 2 ;vary loopstartpoint aout flooper2 .8, 1, 0, kst, 0.05, 1, 0, 2 outs aout, aout endin </CsInstruments> <CsScore> ; Its table size is deferred, ; and format taken from the soundfile header f 1 0 0 1 "fox.wav" 0 0 0 i 1 0 12 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvscent.csd.xml0000644000000000000000000000462112262561504020340 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 giSine ftgen 0, 0, 4096, 10, 1 instr 1 irefrtm = p4; time for generating new values for the spectral centroid ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window ;Sfile = "flute-C-octave0.wav" Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of the audio-signal ktrig metro 1 / irefrtm if ktrig == 1 then kcenter pvscent fftin; spectral center endif aout oscil .2, kcenter, giSine out aout endin </CsInstruments> <CsScore> i 1 0 2.757 .3 i 1 3 2.757 .05 i 1 6 2.757 .005 i 1 9 2.757 .001 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen15.csd.xml0000644000000000000000000001624312262561504017600 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen15.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;example from the Csound Book, page 85 sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = p3 iamp = p4 ifrq = cpspch(p5) ;pitch iswp1 = p6 iswp2 = p7 kswp line iswp1, p3, iswp2 ;amplitude sweep values acosi oscili kswp*.5, ifrq, 2 ;f2=cosine wave asine oscili kswp, ifrq, 1 ;f1=sine wave atab1 tablei acosi, 33, 1, .5 ;tables a1 to GEN13 atab2 tablei acosi, 34, 1, .5 ;tables a1 to GEN14 knrm1 tablei kswp, 35, 1 ;normalizing f35 knrm2 tablei kswp, 36, 1 ;normalizing f36 anrm1 = atab1*knrm1 ;normalize GEN13 signal anrm2 = atab2*knrm2*asine ;normalize GEN14 signal amix = anrm1+anrm2 ;mix GEN13 and GEN14 kenv expseg .001, idur*.1, iamp, idur*.1, iamp*.8, idur*.8, .001 asig = amix*kenv outs asig, asig endin </CsInstruments> <CsScore> f 1 0 8193 10 1 ;sine wave f 2 0 8193 9 1 1 90 ;cosine wave ; Note that all the f33 tables in the following sections are defined with p4=-15, ; which means that tables 33 and 34 will not be normalized. Thus if we display ; tables when running this example, we'll get correct diagrams even if one table ; has very small values instead of 0 values, due to cpu approximations in processing ; sin(180), as in sections 2, 4, and 5. This has no consequence on the audio result, ; because of the use of amp normalization (tables 35 and 36). f 33 0 8193 -15 1 1 1 0 1 180 .8 45 .6 270 .5 90 .4 225 .2 135 .1 315 ;makes function tables 33 and 34 f 35 0 4097 4 33 1 ;amp normalization for f33 f 36 0 4097 4 34 1 ;amp normalization for f34 i 1 0 5 .6 8.00 0 1 i 1 + . .6 8.00 1 0 s ;even harmonics with no phase shift, odd harmonics with phase shift f 33 0 8193 -15 1 1 1 0 1 0 1 180 1 180 1 0 1 0 1 180 1 180 1 0 1 0 1 180 1 180 f 35 0 4097 4 33 1 ;amp normalization for f33 f 36 0 4097 4 34 1 ;amp normalization for f34 i 1 0 5 .6 8.00 0 1 i 1 + . .6 8.00 1 0 s ;different harmonic strenghts and phases f 33 0 8193 -15 1 1 1 0 1 0 .9 180 .5 270 .75 90 .4 45 .2 225 .1 0 f 35 0 4097 4 33 1 ;amp normalization for f33 f 36 0 4097 4 34 1 ;amp normalization for f34 i 1 0 5 .6 8.00 0 1 i 1 + . .6 8.00 1 0 s ;lower harmonics no phase shift, upper harmonics with phase shift f 33 0 8193 -15 1 1 1 0 1 0 .5 0 .9 0 .3 0 .75 0 .2 180 .6 180 .15 180 .5 180 .1 180 f 35 0 4097 4 33 1 ;amp normalization for f33 f 36 0 4097 4 34 1 ;amp normalization for f34 i 1 0 5 .6 8.00 0 1 i 1 + . .6 8.00 1 0 s ;lower harmonics with phase shift, upper harmonics no phase shift f 33 0 8193 -15 1 1 1 180 1 180 .5 180 .9 180 .3 180 .75 180 .2 0 .6 0 .15 0 .5 0 .1 0 f 35 0 4097 4 33 1 ;amp normalization for f33 f 36 0 4097 4 34 1 ;amp normalization for f34 i 1 0 5 .6 8.00 0 1 i 1 + . .6 8.00 1 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/virtual.csd.xml0000644000000000000000000000425612262561504020350 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in Virtual MIDI -M0 is needed anyway -odac -iadc -+rtmidi=virtual -M0 </CsOptions> <CsInstruments> ; By Mark Jamerson 2007 sr=44100 ksmps=10 nchnls=2 massign 1,1 prealloc 1,10 instr 1 ;Midi FM synth inote cpsmidi iveloc ampmidi 10000 idur = 2 xtratim 1 kgate oscil 1,10,2 anoise noise 100*inote,.99 acps samphold anoise,kgate aosc oscili 1000,acps,1 aout = aosc ; Use controller 7 to control volume kvol ctrl7 1, 7, 0.2, 1 outs kvol * aout, kvol * aout endin </CsInstruments> <CsScore> f0 3600 f1 0 1024 10 1 f2 0 16 7 1 8 0 8 f3 0 1024 10 1 .5 .6 .3 .2 .5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hrtfearly.csd.xml0000644000000000000000000000712412262561504020657 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select flags here ; realtime audio out -o dac ; file ouput ; -o hrtf.wav </CsOptions> <CsInstruments> nchnls = 2 gasrc init 0 ;global instr 1 ;a plucked string, distorted and filtered iamp = 15000 icps = cpspch(p4) a1 pluck iamp, icps, icps, 0, 1 adist distort1 a1, 10, .5, 0, 0 afilt moogvcf2 adist, 8000, .5 aout linen afilt, 0, p3, .01 gasrc = gasrc + aout endin instr 10 ;uses output from instr1 as source ;simple path for source kx line 2, p3, 9 ;early reflections, room default 1 aearlyl,aearlyr, irt60low, irt60high, imfp hrtfearly gasrc, kx, 5, 1, 5, 1, 1, "hrtf-44100-left.dat", "hrtf-44100-right.dat", 1 ;later reverb, uses outputs from above arevl, arevr, idel hrtfreverb gasrc, irt60low, irt60high, "hrtf-44100-left.dat", "hrtf-44100-right.dat", 44100, imfp ;delayed and scaled alatel delay arevl * .1, idel alater delay arevr * .1, idel outs aearlyl + alatel, aearlyr + alater gasrc = 0 endin </CsInstruments> <CsScore> ; Play Instrument 1: a simple arpeggio i1 0 .2 8.00 i1 + .2 8.04 i1 + .2 8.07 i1 + .2 8.11 i1 + .2 9.02 i1 + 1.5 8.11 i1 + 1.5 8.07 i1 + 1.5 8.04 i1 + 1.5 8.00 i1 + 1.5 7.09 i1 + 4 8.00 ; Play Instrument 10 for 13 seconds. i10 0 13 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/resyn.csd.xml0000644000000000000000000000414512262561504020017 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o resyn.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktracks = p4 ain diskin2 "fox.wav", 1, 0, 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; pvsifd analysis fst partials fs1, fsi2, .1, 1,3, 500 ; partial tracking aout resyn fst, 1, 1.5, ktracks, 1 ; resynthesis (up a 5th) outs aout, aout endin </CsInstruments> <CsScore> ;sine f1 0 4096 10 1 i 1 0 2.7 500 i 1 3 2.7 10 ;non-linear filtering effect e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/toot8.csd.xml0000644000000000000000000000377012262561504017737 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> instr 8 idur = p3 iamp = p4 iskiptime = p5 iattack = p6 irelease = p7 irvbtime = p8 irvbgain = p9 kamp linen iamp, iattack, idur, irelease asig soundin "fox.wav", iskiptime arampsig = kamp * asig aeffect reverb asig, irvbtime arvbretrn = aeffect * irvbgain out arampsig + arvbretrn endin </CsInstruments> <CsScore> ;ins strt dur amp skip atk rel rvbt rvbgain i8 0 2.28 .3 0 .03 .1 1.5 .3 i8 4 1.6 .3 1.6 .1 .1 1.1 .4 i8 5.5 2.28 .3 0 .5 .1 2.1 .2 i8 6.5 2.28 .4 0 .01 .1 1.1 .1 i8 8 2.28 .5 0.1 .01 .1 0.1 .1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outs2.csd.xml0000644000000000000000000000356212262561504017735 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outs2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig vco2 .01, 110 ; sawtooth waveform at low volume kcut line 300, p3, 60 ; Vary cutoff frequency kresonance = 3 inumlayer = 3 asig lowresx asig, kcut, kresonance, inumlayer outs2 asig ; output stereo channel 2 only endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fprintks-2.csd.xml0000644000000000000000000000445512262561504020662 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages ; -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -n -Fmidichn_advanced.mid ;Don't write audio ouput to disk and use the file midichn_advanced.mid as MIDI input </CsOptions> <CsInstruments> sr = 48000 ksmps = 16 nchnls = 2 ;Example by Jonathan Murphy 2007 ; assign all midi events to instr 1000 massign 0, 1000 pgmassign 0, 1000 instr 1000 ktim timeinsts kst, kch, kd1, kd2 midiin if (kst != 0) then ; p4 = MIDI event type p5 = channel p6= data1 p7= data2 fprintks "MIDI2cs.sco", "i1\\t%f\\t%f\\t%d\\t%d\\t%d\\t%d\\n", ktim, 1/kr, kst, kch, kd1, kd2 endif endin </CsInstruments> <CsScore> i1000 0 10000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/exprand.csd.xml0000644000000000000000000000465212262561504020323 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o exprand.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values klamda exprand 20 printk .2, klamda ; look aout oscili 0.8, 440+klamda, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 klamda exprand 20 printk .2, klamda ; look aout oscili 0.8, 440+klamda, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/syncphasor-CZresonance.csd.xml0000644000000000000000000002027212262561504023257 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o syncphasor-CZresonance.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; by Anthony Kozar. February 2008 ; http://www.anthonykozar.net/ ; Imitation of the Casio CZ-series synthesizer's "Resonance" waveforms ; using a synced phasor to read a sinusoid table. The jumps at the sync ; points are smoothed by multiplying with a windowing function controlled ; by the master phasor. ; Based on information from the Wikipedia article on phase distortion: ; http://en.wikipedia.org/wiki/Phase_distortion_synthesis ; Sawtooth Resonance waveform. Smoothing function is just the inverted ; master phasor. ; The Wikipedia article shows an inverted cosine as the stored waveform, ; which implies that it must be unipolar for the smoothing to work. ; I have substituted a sine wave in the first phrase to keep the output ; bipolar. The second phrase demonstrates the much "rezzier" sound of the ; bipolar cosine due to discontinuities. instr 1 ifreq = cpspch(p4) initReson = p5 itable = p6 imaxamp = 10000 anosync init 0.0 kslavecps line ifreq * initReson, p3, ifreq amaster, async syncphasor ifreq, anosync ; pair of phasors aslave, async2 syncphasor kslavecps, async ; slave synced to master aosc tablei aslave, itable, 1 ; use slave phasor to read a (co)sine table aout = aosc * (1.0 - amaster) ; inverted master smoothes jumps adeclick linseg 0.0, 0.05, 1.0, p3 - 0.1, 1.0, 0.05, 0.0 out aout * adeclick * imaxamp endin ; Triangle or Trapezoidal Resonance waveform. Uses a second table to change ; the shape of the smoothing function. (This is my best guess so far as to ; how these worked). The cosine table works fine with the triangular smoothing ; but we once again need to use a sine table with the trapezoidal smoothing. ; (It might be interesting to be able to vary the "width" of the trapezoid. ; This could be done with the pdhalf opcode). instr 2 ifreq = cpspch(p4) initReson = p5 itable = p6 ismoothtbl = p7 imaxamp = 10000 anosync init 0.0 kslavecps line ifreq * initReson, p3, ifreq amaster, async syncphasor ifreq, anosync ; pair of phasors aslave, async2 syncphasor kslavecps, async ; slave synced to master aosc tablei aslave, itable, 1 ; use slave phasor to read a (co)sine table asmooth tablei amaster, ismoothtbl, 1 ; use master phasor to read smoothing table aout = aosc * asmooth adeclick linseg 0.0, 0.05, 1.0, p3 - 0.1, 1.0, 0.05, 0.0 out aout * adeclick * imaxamp endin </CsInstruments> <CsScore> f1 0 16385 10 1 f3 0 16385 9 1 1 270 ; inverted cosine f5 0 4097 7 0.0 2048 1.0 2049 0.0 ; unipolar triangle f6 0 4097 7 1.0 2048 1.0 2049 0.0 ; "trapezoid" ; Sawtooth resonance with a sine table i1 0 1 7.00 5.0 1 i. + 0.5 7.02 4.0 i. + . 7.05 3.0 i. + . 7.07 2.0 i. + . 7.09 1.0 i. + 2 7.06 12.0 f0 6 s ; Sawtooth resonance with a cosine table i1 0 1 7.00 5.0 3 i. + 0.5 7.02 4.0 i. + . 7.05 3.0 i. + . 7.07 2.0 i. + . 7.09 1.0 i. + 2 7.06 12.0 f0 6 s ; Triangle resonance with a cosine table i2 0 1 7.00 5.0 3 5 i. + 0.5 7.02 4.0 i. + . 7.05 3.0 i. + . 7.07 2.0 i. + . 7.09 1.0 i. + 2 7.06 12.0 f0 6 s ; Trapezoidal resonance with a sine table i2 0 1 7.00 5.0 1 6 i. + 0.5 7.02 4.0 i. + . 7.05 3.0 i. + . 7.07 2.0 i. + . 7.09 1.0 i. + 2 7.06 12.0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/subinstr_named.csd.xml0000644000000000000000000000600412262561504021670 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o subinstr_named.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument "basic_tone" - Creates a basic tone. instr basic_tone ; Print the value of p4, should be equal to ; Instrument #2's iamp field. print p4 ; Print the value of p5, should be equal to ; Instrument #2's ipitch field. print p5 ; Create a tone. asig oscils p4, p5, 0 out asig endin ; Instrument #1 - Demonstrates the subinstr opcode. instr 1 iamp = 20000 ipitch = 440 ; Use the "basic_tone" named instrument to create a ; basic sine-wave tone. ; Its p4 parameter will be set using the iamp variable. ; Its p5 parameter will be set using the ipitch variable. abasic subinstr "basic_tone", iamp, ipitch ; Output the basic tone that we have created. out abasic endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ampdbfs.csd.xml0000644000000000000000000000343612262561504020275 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ampdbfs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 idb = p4 iamp = ampdbfs(idb) asig oscil iamp, 220, 1 print iamp outs asig, asig endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 1 -1 i 1 + 1 -5 i 1 + 1 -6 i 1 + 1 -20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gogobel.csd.xml0000644000000000000000000000405112262561504020271 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gogobel.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 200 ihrd = 0.5 ipos = p4 kvibf = 6.0 kvamp = 0.3 asig gogobel .9, kfreq, ihrd, ipos, 1, 6.0, 0.3, 2 outs asig, asig endin </CsInstruments> <CsScore> ;audio file f 1 0 256 1 "marmstk1.wav" 0 0 0 ;sine wave for the vibrato f 2 0 128 10 1 i 1 0.5 0.5 0.01 i 1 + 0.5 0.561 i 1 + 0.5 0.9 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cngoto.csd.xml0000644000000000000000000000551512262561504020152 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; -o cngoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Change kval linearly from 0 to 2 over ; the period set by the third p-field. kval line 0, p3, 2 ; If kval *is not* greater than or equal to 1 then play ; the high note. Otherwise, play the low note. cngoto (kval >= 1), highnote kgoto lownote highnote: kfreq = 880 goto playit lownote: kfreq = 440 goto playit playit: ; Print the values of kval and kfreq. printks "kval = %f, kfreq = %f\\n", 1, kval, kfreq a1 oscil 10000, kfreq, 1 out a1 endin </CsInstruments> <CsScore> ; Table: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/notequal.csd.xml0000644000000000000000000000426512262561504020512 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o notequal.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = cpspch(p4) iprint = p5 if (iprint != 1) igoto skipPrint print ipch asig vco2 .7, ipch outs asig, asig skipPrint: endin </CsInstruments> <CsScore> f 1 0 65536 10 1 ;sine wave i1 0 .5 8.00 0 i1 0 .5 8.01 1 ; this note will print it's ipch value and only this one will be played i1 0 .5 8.02 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/goto.csd.xml0000644000000000000000000000406012262561504017623 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o goto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 a1 oscil 10000, 440, 1 goto playit ; The goto will go to the playit label. ; It will skip any code in between like this comment. playit: out a1 endin </CsInstruments> <CsScore> ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/randomi.csd.xml0000644000000000000000000000500712262561504020306 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o randomi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 seed 0 ; Instrument #1. instr 1 ; Choose a random frequency between 220 and 440. ; Generate new random numbers at 10 Hz. kmin init 220 kmax init 440 kcps init 10 imode = p4 ifstval = p5 printf_i "\nMode: %d\n", 1, imode k1 randomi kmin, kmax, kcps, imode, ifstval printks "k1 = %f\n", 0.1, k1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. ; each time with a different mode. i 1 0 1 i 1 1 1 1 i 1 2 1 2 330 i 1 3 1 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/clear.csd.xml0000644000000000000000000000422312262561504017742 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o clear.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gaReverb init 0 instr 1 idur = p3 kpitch = p4 a1 diskin2 "fox.wav", kpitch a1 = a1*.5 ;reduce volume vincr gaReverb, a1 endin instr 99 ; global reverb al, ar reverbsc gaReverb, gaReverb, .8, 10000 outs gaReverb+al, gaReverb+ar clear gaReverb endin </CsInstruments> <CsScore> i1 0 3 1 i99 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fout_ftable.csd.xml0000644000000000000000000000605312262561504021151 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o fout_ftable.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; By: Jonathan Murphy 2007 gilen = 131072 gicps = sr/gilen gitab ftgen 1, 0, gilen, 10, 1 instr 1 /******** write file to table ********/ ain diskin2 "beats.wav", 1, 0, 1 aphs phasor gicps andx = aphs * gilen tablew ain, andx, gitab /******** write table to file ********/ aosc table aphs, gitab, 1 out aosc fout "beats_copy.wav", 6, aosc endin </CsInstruments> <CsScore> i1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/soundin.csd.xml0000644000000000000000000000437312262561504020341 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ;-odac ;;;realtime audio out -iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o soundin.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; choose between mono or stereo file ichn filenchnls p4 ;check number of channels print ichn if ichn == 1 then asig soundin p4 ;mono signal outs asig, asig else ;stereo signal aL, aR soundin p4 outs aL, aR endif endin </CsInstruments> <CsScore> i 1 0 3 "fox.wav" ;mono signal i 1 5 2 "kickroll.wav" ;stereo signal e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/crossfm.csd.xml0000644000000000000000000000435512262561504020336 0ustar rootroot <CsoundSynthesizer> <CsOptions> -d -o dac </CsOptions> <CsInstruments> sr = 96000 ksmps = 10 nchnls = 2 0dbfs = 1 FLpanel "crossfmForm", 600, 400, 0, 0 gkfrq1, ihfrq1 FLcount "Freq #1", 0, 20000, 0.001, 1, 1, 200, 30, 20, 50, -1 gkfrq2, ihfrq2 FLcount "Freq #2", 0, 20000, 0.001, 1, 1, 200, 30, 20, 130, -1 gkndx1, gkndx2, ihndx1, ihndx2 FLjoy "Indexes", 0, 10, 0, 10, 0, 0, -1, -1, 200, 200, 300, 50 FLsetVal_i 164.5, ihfrq1 FLsetVal_i 263.712, ihfrq2 FLsetVal_i 1.5, ihndx1 FLsetVal_i 3, ihndx2 FLpanelEnd FLrun maxalloc 1, 2 instr 1 kamp linen 0.5, 0.01, p3, 0.5 a1,a2 crossfm gkfrq1, gkfrq2, gkndx1, gkndx2, 1, 1, 1 outs a1*kamp, a2*kamp endin </CsInstruments> <CsScore> f1 0 16384 10 1 0 i1 0 60 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/polyaft.csd.xml0000644000000000000000000000471412262561504020337 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o polyaft.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 10 nchnls = 1 massign 1, 1 itmp ftgen 1, 0, 1024, 10, 1 ; sine wave instr 1 kcps cpsmidib 2 ; note frequency inote notnum ; note number kaft polyaft inote, 0, 127 ; aftertouch ; interpolate aftertouch to eliminate clicks ktmp phasor 40 ktmp trigger 1 - ktmp, 0.5, 0 kaft tlineto kaft, 0.025, ktmp ; map to sine curve for crossfade kaft = sin(kaft * 3.14159 / 254) * 22000 asnd oscili kaft, kcps, 1 out asnd endin </CsInstruments> <CsScore> t 0 120 f 0 9 2 -2 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLslidBnk2.csd.xml0000644000000000000000000000647212262561504020556 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc -M0 ;;;RT audio I/O with MIDI in </CsOptions> <CsInstruments> sr = 44100 ksmps = 100 nchnls = 2 ;Example by Gabriel Maldonado giElem init 8 giOutTab ftgen 1,0,128, 2, 0 ;min1, max1, exp1, type1, min2, max2, exp2, type2, min3, max3, exp3, type3 etc. giConfigTab ftgen 2,0,128,-2, .1, 1000, -1, 3, .1, 1000, -1, 3, .1, 1000, -1, 3, 30, 2000, -1, 3, \ .1, 5000, -1, 5, .1, 5000, -1, 5, .1, 5000, -1, 5, .1, 5000, -1, 5 giSine ftgen 3,0,256,10, 1 FLpanel "This Panel contains a Slider Bank",600,600 FLslidBnk2 "mod1@mod2@mod3@amp@freq1@freq2@freq3@freqPo", giElem, giOutTab, giConfigTab, 400, 500, 100, 10 FLpanel_end FLrun instr 1 kmodindex1 init 0 kmodindex2 init 0 kmodindex3 init 0 kamp init 0 kfreq1 init 0 kfreq2 init 0 kfreq3 init 0 kfreq4 init 0 vtable1k giOutTab, kmodindex1 , kmodindex2, kmodindex3, kamp, kfreq1, kfreq2 , kfreq3, kfreq4 amod1 oscili kmodindex1, kfreq1, giSine amod2 oscili kmodindex2, kfreq2, giSine amod3 oscili kmodindex3, kfreq3, giSine aout oscili kamp, kfreq4+amod1+amod2+amod3, giSine outs aout, aout endin </CsInstruments> <CsScore> i1 0 3600 f0 3600 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/foscil.csd.xml0000644000000000000000000000412612262561504020135 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o foscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 440 kcar = 1 kmod = p4 kndx line 0, p3, 20 ;intensivy sidebands asig foscil .5, kcps, kcar, kmod, kndx, 1 outs asig, asig endin </CsInstruments> <CsScore> ; sine f 1 0 16384 10 1 i 1 0 9 .01 ;vibrato i 1 10 . 1 i 1 20 . 1.414 ;gong-ish i 1 30 5 2.05 ;with "beat" e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gbuzz.csd.xml0000644000000000000000000000461112262561504020016 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gbuzz.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 instr 1 kcps = 220 knh = p4 ;total no. of harmonics klh = p5 ;lowest harmonic kmul line 0, p3, 1 ;increase amplitude of ;higer partials asig gbuzz .6, kcps, knh, klh, kmul, 1 outs asig, asig endin </CsInstruments> <CsScore> ; a cosine wave f 1 0 16384 11 1 i 1 0 3 3 1 ;3 harmonics, lowest harmonic=1 i 1 + 3 30 1 ;30 harmonics, lowest harmonic=1 i 1 + 3 3 2 ;3 harmonics, lowest harmonic=3 i 1 + 3 30 2 ;30 harmonics, lowest harmonic=3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mclock.csd.xml0000644000000000000000000000227412262561504020130 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -Q0 ;;;midi out </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;let csound synchronize a sequencer mclock 24 endin </CsInstruments> <CsScore> i 1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/flanger.csd.xml0000644000000000000000000000404512262561504020274 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o flanger.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfeedback = p4 asnd vco2 .2, 50 adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time =20ms aflg flanger asnd, adel, kfeedback asig clip aflg, 1, 1 outs asig+asnd, asig+asnd ;mix flanger with original endin </CsInstruments> <CsScore> i 1 0 10 .2 i 1 11 10 .8 ;lot of feedback e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/deltapxw.csd.xml0000644000000000000000000000503712262561504020510 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o deltapxw.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 a1 phasor 300 a1 = a1 - 0.5 a_ delayr 1 adel phasor 4 adel = sin(2 * 3.14159265 * adel) * 0.01 + 0.2 deltapxw a1, adel, 32 adel phasor 2 adel = sin(2 * 3.14159265 * adel) * 0.01 + 0.2 deltapxw a1, adel, 32 adel = 0.3 a2 deltapx adel, 32 a1 = 0 delayw a1 outs a2*.7, a2*.7 endin </CsInstruments> <CsScore> i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vbap8move.csd.xml0000644000000000000000000000557012262561504020571 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vbap4move.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 48000 ksmps = 10 nchnls = 8 ;Example by Hector Centeno 2007 vbaplsinit 2, 8, 15, 65, 115, 165, 195, 245, 295, 345 instr 1 ifldnum = 9 ispread = 30 idur = p3 ;; Generate a sound source kenv loopseg 10, 0, 0, 0, 0.5, 1, 10, 0 a1 pinkish 3000*kenv ;; Move circling around once all the speakers aout1, aout2, aout3, aout4, aout5, aout6, aout7, aout8 vbap8move a1, idur, ispread, ifldnum, 15, 65, 115, 165, 195, 245, 295, 345, 15 ;; Speaker mapping aFL = aout8 ; Front Left aFR = aout1 ; Front Right aMFL = aout7 ; Mid Front Left aMFR = aout2 ; Mid Front Right aMBL = aout6 ; Mid Back Left aMBR = aout3 ; Mid Back Right aBL = aout5 ; Back Left aBR = aout4 ; Back Right outo aFL, aFR, aMFL, aMFR, aMBL, aMBR, aBL, aBR endin </CsInstruments> <CsScore> i1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outleta.csd.xml0000644000000000000000000000477312262561504020343 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outleta.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 connect "1", "Outl", "reverby", "InL" connect "1", "Outr", "reverby", "InR" alwayson "reverby", 1 instr 1 aIn diskin2 "fox.wav", 1 outleta "Outl", aIn outleta "Outr", aIn endin instr reverby aInL inleta "InL" aInR inleta "InR" al, ar reverbsc aInL, aInR, 0.7, 21000 ifxlev = 0.5 al = (aInL*ifxlev)+(al*(1-ifxlev)) ar = (aInR*ifxlev)+(ar*(1-ifxlev)) outs al, ar endin </CsInstruments> <CsScore> i 1 0 3 e4 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vtable1k.csd.xml0000644000000000000000000000757112262561504020376 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O </CsOptions> <CsInstruments> sr = 44100 ksmps = 100 nchnls = 2 giElem init 13 giOutTab ftgen 1,0,128, 2, 0 giFreqTab ftgen 2,0,128,-7, 1,giElem, giElem+1 giSine ftgen 3,0,256,10, 1 FLpanel "This Panel contains a Slider Bank",500,400 FLslidBnk "mod1@mod2@mod3@amp@freq1@freq2@freq3@freqPo", giElem, giOutTab, 360, 600, 100, 10 FLpanel_end FLrun instr 1 kout1 init 0 kout2 init 0 kout3 init 0 kout4 init 0 kout5 init 0 kout6 init 0 kout7 init 0 kout8 init 0 vtable1k giOutTab, kout1 , kout2, kout3, kout4, kout5 , kout6, kout7, kout8 kmodindex1= 2 * db(kout1 * 80 ) kmodindex2= 2 * db(kout2 * 80 ) kmodindex3= 2 * db(kout3 * 80 ) kamp = 50 * db(kout4 * 70 ) kfreq1 = 1.1 * octave(kout5 * 10) kfreq2 = 1.1 * octave(kout6 * 10) kfreq3 = 1.1 * octave(kout7 * 10) kfreq4 = 30 * octave(kout8 * 8) amod1 oscili kmodindex1, kfreq1, giSine amod2 oscili kmodindex2, kfreq2, giSine amod3 oscili kmodindex3, kfreq3, giSine aout oscili kamp, kfreq4+amod1+amod2+amod3, giSine outs aout, aout endin </CsInstruments> <CsScore> i1 0 3600 f0 3600 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sfpreset.csd.xml0000644000000000000000000000712312262561504020511 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out, virtual midi in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfpreset.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf1 sfload "sf_GMbank.sf2" sfplist gisf1 ;list presets of first soundfont gisf2 sfload "07AcousticGuitar.sf2" sfplist gisf2 ;list presets of second soundfont gir sfpreset 50, 0, gisf1, 0 ;assign Synth Strings to index 0 giv sfpreset 0, 0, gisf2, 1 ;assign AcousticGuitar to index 1 print gir print giv instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/5000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay3 ivel, inum, kamp*ivel, kfreq, p6 outs a1, a2 endin </CsInstruments> <CsScore> f0 60 ; stay active for 1 minute i1 0 1 60 127 0 ;= Synth Strings I from first soundfont i1 + 1 62 < . i1 + 1 65 < . i1 + 1 69 10 . i1 5 1 60 127 1 ;= AcousticGuitar from second soundfont i1 + 1 62 < . i1 + 1 65 < . i1 + 1 69 10 . e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/reverbsc.csd.xml0000644000000000000000000000451712262561504020475 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o reverbsc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 48000 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 a1 vco2 0.85, 440, 10 kfrq port 100, 0.004, 20000 a1 butterlp a1, kfrq a2 linseg 0, 0.003, 1, 0.01, 0.7, 0.005, 0, 1, 0 a1 = a1 * a2 a2 = a1 * p5 a1 = a1 * p4 denorm a1, a2 aL, aR reverbsc a1, a2, 0.85, 12000, sr, 0.5, 1 outs a1 + aL, a2 + aR endin </CsInstruments> <CsScore> i 1 0 1 0.71 0.71 i 1 1 1 0 1 i 1 2 1 -0.71 0.71 i 1 3 1 1 0 i 1 4 4 0.71 0.71 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vmult.csd.xml0000644000000000000000000000510712262561504020025 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 kr=4410 ksmps=10 nchnls=2 instr 1 ;table playback ar lposcil 1, 1, 0, 262144, 1 outs ar,ar endin instr 2 vcopy 2, 1, 40000 ;copy vector from sample to empty table vmult 5, 10000, 262144 ;scale noise to make it audible vcopy 1, 5, 40000 ;put noise into sample turnoff endin instr 3 vcopy 1, 2, 40000 ;put original information back in turnoff endin </CsInstruments> <CsScore> f1 0 262144 -1 "beats.wav" 0 4 0 f2 0 262144 2 0 f5 0 262144 21 3 30000 i1 0 4 i2 3 1 s i1 0 4 i3 3 1 s i1 0 4 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/subtracts.csd.xml0000644000000000000000000000532112262561504020666 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; -o -.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = p3 iatt = p4 idec = p5 isus = p3-iatt-idec ;calculate sustain time from subtracting attack and decay printf_i "sutain time= note duration - attack - decay --> %.1f-%.1f-%.1f = %.1f\n", 1, idur, iatt, idec, isus kenv expseg 0.01, iatt, 1, isus, 1, idec, 0.01 ;envelope asig poscil 1*kenv, 200, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 4096 10 1 ;sine wave ; attack decay i 1 0 3 .1 .2 i 1 4 3 .5 1.5 i 1 8 5 4 .5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLxyin-2.csd.xml0000644000000000000000000000432412262561504020226 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O </CsOptions> <CsInstruments> sr=44100 kr=441 ksmps=100 nchnls=2 ; Example by Gabriel Maldonado FLpanel "Move the mouse inside this panel to hear the effect",400,400 FLpanel_end FLrun instr 1 k1, k2, kinside FLxyin 50, 1000, 50, 1000, 100, 300, 50, 250, -2,-3 ;if k1 <= 50 || k1 >=5000 || k2 <=100 || k2 >= 8000 kgoto end ; if cursor is outside bounds, then don't play!!! a1 oscili 3000, k1, 1 a2 oscili 3000, k2, 1 outs a1,a2 printk2 k1 printk2 k2, 10 printk2 kinside, 20 end: endin </CsInstruments> <CsScore> f1 0 1024 10 1 f2 0 17 19 1 1 90 1 f3 0 17 19 2 1 90 1 i1 0 3600 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/betarand.csd.xml0000644000000000000000000000465312262561504020443 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o betarand.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values kbeta betarand 100, 1, 1 printk .2, kbeta ; look aout oscili 0.8, 440+kbeta, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 kbeta betarand 100, 1, 1 printk .2, kbeta ; look aout oscili 0.8, 440+kbeta, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/linseg.csd.xml0000644000000000000000000000607212262561504020141 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o linseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kcps = cpspch(p4) kenv linseg 0, 0.25, 1, 0.75, 0 ; together = 1 sec asig poscil kenv, kcps, giSine outs asig, asig endin instr 2 ; scaling to duration kcps = cpspch(p4) kenv linseg 0, p3*0.25, 1, p3*0.75, 0 asig poscil kenv, kcps, giSine outs asig, asig endin instr 3 ; with negative value kcps = cpspch(p4) aenv linseg 0, 0.1, 1, 0.5, -0.9, 0.4, 0 asig poscil aenv, kcps, giSine outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 7.00 ; = 1 sec, p3 fits exactly i 1 2 2 7.00 ; = 2 sec, p3 truncated at 1 sec i 2 4 1 7.00 ; scales to duration i 2 6 2 7.00 ; of p3 i 3 9 2 7.00 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftsave.csd.xml0000644000000000000000000000577612262561504020162 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ftsave.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Table #1, make a sine wave using the GEN10 routine. gitmp1 ftgen 1, 0, 32768, 10, 1 ; Table #2, create an empty table. gitmp2 ftgen 2, 0, 32768, 7, 0, 32768, 0 ; Instrument #1 - a basic oscillator. instr 1 kamp = 20000 kcps = 440 ; Use Table #1. ifn = 1 a1 oscil kamp, kcps, ifn out a1 endin ; Instrument #2 - Load Table #1 into Table #2. instr 2 ; Save Table #1 to a file called "table1.ftsave". ftsave "table1.ftsave", 0, 1 ; Load the "table1.ftsave" file into Table #2. ftload "table1.ftsave", 0, 2 kamp = 20000 kcps = 440 ; Use Table #2, it should contain Table #1's sine wave now. ifn = 2 a1 oscil kamp, kcps, ifn out a1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for 1 second. i 1 0 1 ; Play Instrument #2 for 1 second. i 2 2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/rtclock.csd.xml0000644000000000000000000000657712262561504020333 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n ;;;no sound ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o rtclock.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;after an example by Iain McCurdy sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 FLcolor 200, 200, 200, 0, 0, 0 ; LABEL | WIDTH | HEIGHT | X | Y FLpanel "rtclock", 500, 130, 0, 0 ; ON,OFF,TYPE,WIDTH, HEIGHT, X, Y, OPCODE, INS,START,IDUR gkOnOff,ihOnOff FLbutton "On/Off", 1, 0, 22, 150, 25, 5, 5, 0, 1, 0, 3600 gkExit,ihExit FLbutton "exitnow",1, 0, 21, 150, 25, 345, 5, 0, 999, 0, 0.001 FLsetColor2 255, 0, 50, ihOnOff ;reddish color ;VALUE DISPLAY BOXES WIDTH,HEIGHT,X, Y gidclock FLvalue "clock", 100, 25, 200, 60 FLsetVal_i 1, ihOnOff FLpanel_end FLrun instr 1 if gkOnOff !=0 kgoto CONTINUE ;sense if FLTK on/off switch is not off (in which case skip the next line) turnoff ;turn this instr. off now CONTINUE: ktime rtclock ;clock continues to run even FLprintk2 ktime, gidclock ;after the on/off button was used to stop endin instr 999 exitnow ;exit Csound as fast as possible endin </CsInstruments> <CsScore> f 0 60 ;runs 60 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vco2init.csd.xml0000644000000000000000000000463212262561504020415 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vco2init.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=1 nchnls=2 ; create waveform with discontinuities, so it has a lot of high freq content gitable ftgen 0, 0, 2^16+1, 7, -1, 2^14, 1, 0, -1, 2^14, 1, 0, -1, 2^15, 1 ; make bandlimited tables of the waveform gi_nextfree vco2init -gitable, gitable+1, 1.05, 128, 2^16, gitable gitable_bl = -gitable instr 1 kfreq expon 14000, p3, 500 kfn vco2ft kfreq, gitable_bl asig oscilikt 5000, kfreq, kfn printk 0.1, kfn ; remove semicolon on next line to hear original waveform, demonstrating the aliasing ;asig oscili 5000, kfreq, gitable outs asig, asig endin </CsInstruments> <CsScore> i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvadsyn.csd.xml0000644000000000000000000000724312262561504020345 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsadsyn.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 opcode FileToPvsBuf, iik, Siiii ;;writes an audio file at the first k-cycle to a fft-buffer (via pvsbuffer) Sfile, ifftsize, ioverlap, iwinsize, iwinshape xin ktimek timeinstk if ktimek == 1 then ilen filelen Sfile kcycles = ilen * kr; number of k-cycles to write the fft-buffer kcount init 0 loop: ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape ibuf, ktim pvsbuffer fftin, ilen + (ifftsize / sr) loop_lt kcount, 1, kcycles, loop xout ibuf, ilen, ktim endif endop instr 1 istretch = p4; time stretching factor ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window ibuffer, ilen, k0 FileToPvsBuf "fox.wav", ifftsize, ioverlap, iwinsize, iwinshape p3 = istretch * ilen; set p3 to the correct value ktmpnt linseg 0, p3, ilen; time pointer fread pvsbufread ktmpnt, ibuffer; read the buffer aout pvsadsyn fread, 10, 1; resynthesis with the first 10 bins out aout endin </CsInstruments> <CsScore> i 1 0 1 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cps2pch_19et.csd.xml0000644000000000000000000000342112262561504021057 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cps2pch_19et.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use 19ET scale. ipch = 8.02 iequal = 19 icps cps2pch ipch, iequal print icps endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/transeg.csd.xml0000644000000000000000000000414212262561504020317 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o transeg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 2 0dbfs = 1 instr 1 ;p4 and p5 determine the type of curve for each ;section of the envelope kenv transeg 0.01, p3*0.25, p4, 1, p3*0.75, p5, 0.01 a1 oscil kenv, 440, 1 outs a1, a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 i 1 0 2 2 2 i 1 + . 5 5 i 1 + . 1 1 i 1 + . 0 0 i 1 + . -2 -2 i 1 + . -2 2 i 1 + . 2 -2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lessthan.csd.xml0000644000000000000000000000424012262561504020474 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o <.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = p4 ipitch = (ipch < 15 ? cpspch(ipch) : ipch) ;if p4 is lower then 15, it assumes p4 to be pitch-class print ipitch ;and not meant to be a frequency in Hertz asig poscil .5, ipitch , 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 8192 10 1 ;sine wave i1 0 3 8.00 ;pitch class i1 4 3 800 ;frequency e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vsubv.csd.xml0000644000000000000000000000567212262561504020032 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ifn2 = p5 ielements = p6 idstoffset = p7 isrcoffset = p8 kval init 25 vsubv ifn1, ifn2, ielements, idstoffset, isrcoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 15 16 f 2 0 16 -7 1 15 2 i2 0.0 0.2 1 i2 0.2 0.2 2 i1 0.4 0.01 1 2 5 3 8 i2 0.8 0.2 1 i1 1.0 0.01 1 2 5 10 -2 i2 1.2 0.2 1 i1 1.4 0.01 1 2 8 14 0 i2 1.6 0.2 1 i1 1.8 0.01 1 2 8 0 14 i2 2.0 0.2 1 i1 2.2 0.002 1 1 8 5 2 i2 2.4 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/timek.csd.xml0000644000000000000000000000343112262561504017765 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o timek.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the value from timek every half-second. k1 timek printks "k1 = %f samples\\n", 0.5, k1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvinterp.csd.xml0000644000000000000000000000404712262561504020527 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvinterp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; analyze "fox.wav" and "flute.aiff" with PVANAL first ktime1 line 0, p3, 2.8 ; used as index in the "fox.pvx" file ktime2 line 0, p3, 3 ; used as index in the "flute.pvx" file kinterp line 1, p3, 0 pvbufread ktime1, "fox.pvx" asig pvinterp ktime2,1,"flute.pvx",.9, 3, .6, 1, kinterp,1-kinterp outs asig, asig endin </CsInstruments> <CsScore> i 1 0 3 i 1 + 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/endop.csd.xml0000644000000000000000000001341212262561504017761 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o endop.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 /* example opcode 1: simple oscillator */ opcode Oscillator, a, kk kamp, kcps xin ; read input parameters a1 vco2 kamp, kcps ; sawtooth oscillator xout a1 ; write output endop /* example opcode 2: lowpass filter with local ksmps */ opcode Lowpass, a, akk setksmps 1 ; need sr=kr ain, ka1, ka2 xin ; read input parameters aout init 0 ; initialize output aout = ain*ka1 + aout*ka2 ; simple tone-like filter xout aout ; write output endop /* example opcode 3: recursive call */ opcode RecursiveLowpass, a, akkpp ain, ka1, ka2, idep, icnt xin ; read input parameters if (icnt >= idep) goto skip1 ; check if max depth reached ain RecursiveLowpass ain, ka1, ka2, idep, icnt + 1 skip1: aout Lowpass ain, ka1, ka2 ; call filter xout aout ; write output endop /* example opcode 4: de-click envelope */ opcode DeClick, a, a ain xin aenv linseg 0, 0.02, 1, p3 - 0.05, 1, 0.02, 0, 0.01, 0 xout ain * aenv ; apply envelope and write output endop /* instr 1 uses the example opcodes */ instr 1 kamp = .6 ; amplitude kcps expon 50, p3, 500 ; pitch a1 Oscillator kamp, kcps ; call oscillator kflt linseg 0.4, 1.5, 0.4, 1, 0.8, 1.5, 0.8 ; filter envelope a1 RecursiveLowpass a1, kflt, 1 - kflt, 10 ; 10th order lowpass a1 DeClick a1 outs a1, a1 endin </CsInstruments> <CsScore> i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hilbert.csd.xml0000644000000000000000000001006312262561504020304 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o hilbert.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = p3 ; Initial amount of frequency shift. ; It can be positive or negative. ibegshift = p4 ; Final amount of frequency shift. ; It can be positive or negative. iendshift = p5 ; A simple envelope for determining the ; amount of frequency shift. kfreq linseg ibegshift, idur, iendshift ; Use the sound of your choice. ain diskin2 "beats.wav", 1, 0, 1 ; Phase quadrature output derived from input signal. areal, aimag hilbert ain ; Quadrature oscillator. asin oscili 1, kfreq, 1 acos oscili 1, kfreq, 1, .25 ; Use a trigonometric identity. ; See the references for further details. amod1 = areal * acos amod2 = aimag * asin ; Both sum and difference frequencies can be ; output at once. ; aupshift corresponds to the sum frequencies. aupshift = (amod1 - amod2) * 0.7 ; adownshift corresponds to the difference frequencies. adownshift = (amod1 + amod2) * 0.7 ; Notice that the adding of the two together is ; identical to the output of ring modulation. outs aupshift, aupshift endin </CsInstruments> <CsScore> ; Sine table for quadrature oscillator. f 1 0 16384 10 1 ; Starting with no shift, ending with all ; frequencies shifted up by 2000 Hz. i 1 0 6 0 2000 ; Starting with no shift, ending with all ; frequencies shifted down by 250 Hz. i 1 7 6 0 -250 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/semitone.csd.xml0000644000000000000000000000443212262561504020501 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o semitone.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iroot = 440 ; root note is A above middle-C (440 Hz) ksem lfo 12, .5, 5 ; generate sawtooth, go from 5 octaves higher to root ksm = int(ksem) ; produce only whole numbers kfactor = semitone(ksm) ; for semitones knew = iroot * kfactor printk2 knew printk2 kfactor asig pluck 1, knew, 1000, 0, 1 asig dcblock asig ;remove DC outs asig, asig endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tempest.csd.xml0000644000000000000000000000511712262561504020340 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tempest.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use the "beats.wav" sound file. asig soundin "beats.wav" ; Extract the pitch and the envelope. kcps, krms pitchamdf asig, 150, 500, 200 iprd = 0.01 imindur = 0.1 imemdur = 3 ihp = 1 ithresh = 30 ihtim = 0.005 ixfdbak = 0.05 istartempo = 110 ifn = 1 ; Estimate its tempo. k1 tempest krms, iprd, imindur, imemdur, ihp, ithresh, ihtim, ixfdbak, istartempo, ifn printk2 k1 out asig endin </CsInstruments> <CsScore> ; Table #1, a declining line. f 1 0 128 16 1 128 1 ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/strrindex.csd.xml0000644000000000000000000000352012262561504020675 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n ;;;no sound output ;-iadc ;;;uncomment -iadc if realtime audio input is needed too </CsOptions> <CsInstruments> ;example by Joachim Heintz opcode FilNam, S, S ;returns the name of a file path Spath xin ipos strrindex Spath, "/" ;look for the rightmost '/' Snam strsub Spath, ipos+1 ;extract the substring xout Snam endop instr name prints "Printing name:\n" Snam FilNam "/my/dir/my/file.WAV" puts Snam, 1 endin </CsInstruments> <CsScore> i "name" 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/harmon.csd.xml0000644000000000000000000000436512262561504020147 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o harmon.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aout diskin2 "fox.wav", 1 kestfrq = p4 ;different estimated frequency kmaxvar = 0.1 imode = 1 iminfrq = 100 iprd = 0.02 asig harmon aout, kestfrq, kmaxvar, kestfrq*.5, kestfrq*4, \ imode, iminfrq, iprd outs (asig + aout)*.6, (asig + aout)*.6 ;mix dry&wet signal endin </CsInstruments> <CsScore> i 1 0 2.7 100 i 1 + . 200 i 1 + . 500 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/chnget.csd.xml0000644000000000000000000001154612262561504020132 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chnget.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;Example by Joachim Heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1; send i-values chnset 1, "sio" chnset -1, "non" endin instr 2; send k-values kfreq randomi 100, 300, 1 chnset kfreq, "cntrfreq" kbw = kfreq/10 chnset kbw, "bandw" endin instr 3; send a-values anois rand .1 chnset anois, "noise" loop: idur random .3, 1.5 timout 0, idur, do reinit loop do: ifreq random 400, 1200 iamp random .1, .3 asig oscils iamp, ifreq, 0 aenv transeg 1, idur, -10, 0 asine = asig * aenv chnset asine, "sine" endin instr 11; receive some chn values and send again ival1 chnget "sio" ival2 chnget "non" print ival1, ival2 kcntfreq chnget "cntrfreq" kbandw chnget "bandw" anoise chnget "noise" afilt reson anoise, kcntfreq, kbandw afilt balance afilt, anoise chnset afilt, "filtered" endin instr 12; mix the two audio signals amix1 chnget "sine" amix2 chnget "filtered" chnmix amix1, "mix" chnmix amix2, "mix" endin instr 20; receive and reverb amix chnget "mix" aL, aR freeverb amix, amix, .8, .5 outs aL, aR endin instr 100; clear chnclear "mix" endin </CsInstruments> <CsScore> i 1 0 20 i 2 0 20 i 3 0 20 i 11 0 20 i 12 0 20 i 20 0 20 i 100 0 20 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midinoteonpch.csd.xml0000644000000000000000000000526212262561504021520 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midinoteonpch.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 midinoteonpch p4, p5 ;gets a MIDI note number value as octave-point-decimal value into p4, and MIDI velocity into p5 print p4 ;display the pitch value when it changes and when key is pressed kvel = kvel/127 ;scale midi velocity to 0-1 ipch = p4 icps = cpspch(ipch) ;convert octave-point-decimal value into Hz kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by asig pluck kenv*kvel, icps, icps, 2, 1 ;velocity value outs asig, asig endin </CsInstruments> <CsScore> f 0 30 ;runs for 30 seconds f 2 0 4096 10 1 i 1 0 2 8.09 100 ; play these notes from score as well i 1 + 2 9.05 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSread-2.csd.xml0000644000000000000000000002235412262561504020303 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac -d -m128 </CsOptions> <CsInstruments> ;example by joachim heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 1024, 10, 1 gSfile = "fox.ats" giNumParts ATSinfo gSfile, 3 ;overall number of partials giDur ATSinfo gSfile, 7 ;duration seed 0 instr ReadOnePartial iPartial = p4 p3 = giDur ktime line 0, giDur, giDur prints "Resynthesizing partial number %d.\n", iPartial kFq,kAmp ATSread ktime, gSfile, iPartial kAmp port kAmp, .1 ;smooth amplitudes - still not satisfactoring aOut poscil kAmp, kFq, giSine aOut linen aOut, 0, p3, .01 ;anti-click outs aOut*10, aOut*10 ;start next instr: normal speed, three loops, pause between loops one second event_i "i", "MasterRand", giDur+3, 1, 1, 3, 2 endin instr MasterRand ;random selections of 10 partials per second, overlapping iSpeed = p4 ;speed of reading / playing iNumLoops = p5 ;number of loops iPause = p6 ;length of pause between loops prints "Resynthesizing random partials.\n" p3 = (giDur/iSpeed+iPause) * iNumLoops ;start next instr: half speed, three loops, three seonds pause between loops event_i "i", "MasterArp", p3+3, 1, .5, 3, 3 ;loop over duration plus pause loop: timout 0, giDur/iSpeed+iPause, play reinit loop play: gkTime line 0, giDur/iSpeed, giDur ;start time from 0 in each loop kTrig metro 10 ;10 new partials per second ;call subinstrument if trigger and no pause if kTrig == 1 && gkTime < giDur then kPart random 1, giNumParts+.999 event "i", "PlayRand", 0, 1, int(kPart) endif endin instr MasterArp ;argeggio-like reading and playing of partials iSpeed = p4 ;speed of reading / playing iNumLoops = p5 ;number of loops iPause = p6 ;length of pause between loops prints "Arpeggiating partials.\n" p3 = (giDur/iSpeed+iPause) * iNumLoops loop: timout 0, giDur/iSpeed+iPause, play reinit loop play: gkTime line 0, giDur/iSpeed, giDur kArp linseg 1, (giDur/iSpeed)/2, giNumParts, (giDur/iSpeed)/2, 1 ;arp up and down kTrig metro 10 ;10 new partials per second if kTrig == 1 && gkTime < giDur then event "i", "PlayArp", 0, 5, int(kArp) endif ;exit csound when finished event_i "i", "End", p3+5, 1 endin instr PlayRand iPartial = p4 kFq,kAmp ATSread gkTime, gSfile, iPartial kamp port kAmp, .15 ;smooth amplitudes aOut poscil kAmp, kFq, giSine aOut linen aOut, .01, p3, .01 outs aOut, aOut endin instr PlayArp kCount init 1 ;k-cycle iPartial = p4 kFq,kAmp ATSread gkTime, gSfile, iPartial if kCount == 1 then ;get freq from first k-cycle kModFq = kFq ;avoid to go with 0 Hz as this blocks the mode filter if kModFq == 0 then turnoff endif endif iVol random -42, -12 ;db iOffset random .01, .1 ;no too regularily ... aImp mpulse ampdb(iVol), p3, iOffset iQ random 500, 5000 aOut mode aImp, kModFq, iQ aOut linen aOut, 0, p3, p3/3 outs aOut, aOut kCount = 2 endin instr End exitnow endin </CsInstruments> <CsScore> i "ReadOnePartial" 0 1 10 e 999 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLcount.csd.xml0000644000000000000000000001011212262561504020220 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLcount.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Demonstration of the flcount opcode ; clicking on the single arrow buttons ; increments the oscillator in semitone steps ; clicking on the double arrow buttons ; increments the oscillator in octave steps sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Counter", 900, 400, 50, 50 ; Minimum value output by counter imin = 6 ; Maximum value output by counter imax = 12 ; Single arrow step size (semitones) istep1 = 1/12 ; Double arrow step size (octave) istep2 = 1 ; Counter type (1=double arrow counter) itype = 1 ; Width of the counter in pixels iwidth = 200 ; Height of the counter in pixels iheight = 30 ; Distance of the left edge of the counter ; from the left edge of the panel ix = 50 ; Distance of the top edge of the counter ; from the top edge of the panel iy = 50 ; Score event type (-1=ignored) iopcode = -1 gkoct, ihandle FLcount "pitch in oct format", imin, imax, istep1, istep2, itype, iwidth, iheight, ix, iy, iopcode, 1, 0, 1 ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 iamp = 15000 ifn = 1 asig oscili iamp, cpsoct(gkoct), ifn out asig endin </CsInstruments> <CsScore> ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/scoreline.csd.xml0000644000000000000000000000416412262561504020643 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o scoreline.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktrig metro 3 ;trigger 3 times a second scoreline {{ ;so it sounds like an echo i 2 0 3 "flute.aiff" i 2 1 3 "beats.wav" }}, ktrig ktrig = 0 endin instr 2 asig soundin p4 outs asig*.3, asig*.3 endin </CsInstruments> <CsScore> i1 0 2 ;play for 2 seconds, so the samples are played 6 times e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hilbert_barberpole.csd.xml0000644000000000000000000001014612262561504022503 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o hilbert_barberpole.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 ; kr must equal sr for the barberpole effect to work. kr = 44100 ksmps = 1 nchnls = 2 ; Instrument #1 instr 1 idur = p3 ibegshift = p4 iendshift = p5 ; sawtooth wave, not bandlimited asaw phasor 100 ; add offset to center phasor amplitude between -.5 and .5 asaw = asaw - .5 ; sawtooth wave, with amplitude of 10000 ain = asaw * 20000 ; The envelope of the frequency shift. kfreq linseg ibegshift, idur, iendshift ; Phase quadrature output derived from input signal. areal, aimag hilbert ain ; The quadrature oscillator. asin oscili 1, kfreq, 1 acos oscili 1, kfreq, 1, .25 ; Based on trignometric identities. amod1 = areal * acos amod2 = aimag * asin ; Calculate the up-shift and down-shift. aupshift = (amod1 + amod2) * 0.7 adownshift = (amod1 - amod2) * 0.7 ; Mix in the original signal to achieve the barberpole effect. amix1 = aupshift + ain amix2 = aupshift + ain ; Make sure the output doesn't get louder than the original signal. aout1 balance amix1, ain aout2 balance amix2, ain outs aout1, aout2 endin </CsInstruments> <CsScore> ; Table 1: A sine wave for the quadrature oscillator. f 1 0 16384 10 1 ; The score. ; p4 = frequency shifter, starting frequency. ; p5 = frequency shifter, ending frequency. i 1 0 6 -10 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zkwm.csd.xml0000644000000000000000000000743412262561504017653 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zkwm.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a basic instrument. instr 1 ; Generate a k-rate signal. ; The signal goes from 30 to 20,000 then back to 30. kramp linseg 30, p3/2, 20000, p3/2, 30 ; Mix the signal into the zk variable #1. zkwm kramp, 1 endin ; Instrument #2 -- another basic instrument. instr 2 ; Generate another k-rate signal. ; This is a low frequency oscillator. klfo lfo 3500, 2 ; Mix this signal into the zk variable #1. zkwm klfo, 1 endin ; Instrument #3 -- generates audio output. instr 3 ; Read zk variable #1, containing a mix of both signals. kamp zkr 1 ; Create a sine waveform. Its amplitude will vary ; according to the values in zk variable #1. a1 oscil kamp, 880, 1 ; Generate the audio output. out a1 ; Clear the zk variable, get it ready for ; another pass. zkcl 0, 1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 5 seconds. i 1 0 5 ; Play Instrument #2 for 5 seconds. i 2 0 5 ; Play Instrument #3 for 5 seconds. i 3 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/xscans.csd.xml0000644000000000000000000001614312262561504020157 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o xscans.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;the matrices can be found in /manual/examples instr 1 ; Plain scanned syntnesis a0 = 0 xscanu 1, .01, 6, 2, "128,8-cylinderX", 4, 5, 2, .1, .1, -.01, .1, .5, 0, 0, a0, 0, 0 a1 xscans .7, cpspch(p4), 7, 0, 1 outs a1, a1 endin instr 2 ; Scan synthesis with audio injection and dual scan paths ; tap the mic or sing to inject audio into the resonators a0,aa ins a0 = a0/.8 xscanu 1, .01, 6, 2, "128,8-gridX", 14, 5, 2, .01, .05, -.05, .1, .5, 0, 0, a0, 0, 0 a1 xscans .5, cpspch(7.00), 7, 0, 1 a2 xscans .5, cpspch(7.001), 77, 0, 1 outs a1+a2,a1+a2 endin instr 3 ; Vibrating structure with audio injection ; Tap the MIC - to inject audio into the resonators a0,aa ins a0 = a0/.8 xscanu 1, .01, 6, 2, "128-stringcircularX", 14, 5, 2, .01, .05, -.05, .25, .75, 0, 0, a0, 0, 1 endin instr 4 ; Modulated scanners i1 bexprnd 5 i2 bexprnd 1 ko oscil i1, i2, 9 ka1 oscili .5, .15*8, p7 ka2 oscili .5, .15*8, p8 kf oscili 1, .15, p4 kf = 2^(kf/12)*p6*440+ko a1 xscans p9*ka1, kf+i1, 777, 1, 1 a2 xscans p9*ka2, (kf+i1)*2.1, 77, 1, 1 outs a1+a2, a1+a2 endin </CsInstruments> <CsScore> ; Initial condition f1 0 128 7 0 64 1 64 0 ; Masses f2 0 128 -7 1 128 1 ; Centering force f4 0 128 -7 0 128 2 f14 0 128 -7 2 64 0 64 2 ; Damping f5 0 128 -7 1 128 1 ; Initial velocity f6 0 128 -7 -.0 128 .0 ; Trajectories f7 0 128 -5 .001 128 128 f777 0 128 -23 "128-stringcircular" f77 0 128 -23 "128-spiral-8,16,128,2,1over2" ; Sine f9 0 16384 10 1 ; Pitch tables f100 0 1024 -7 +3 128 +3 128 -2 128 -2 128 +0 128 +0 128 -4 128 -4 128 +3 f101 0 1024 -7 -2 128 -2 128 -2 128 -2 128 -5 128 -5 128 -4 128 -4 128 -2 f102 0 1024 -7 +3 128 +3 128 +2 128 +2 128 +0 128 +0 128 +0 128 +0 128 +3 f103 0 1024 -7 +7 128 +7 128 +5 128 +5 128 +3 128 +3 128 +3 128 +3 128 +7 ; Amplitude tables f200 0 1024 7 1 128 0 128 0 127 0 1 1 128 0 128 0 127 0 1 1 128 0 127 0 1 1 f201 0 1024 7 0 127 0 1 1 127 0 1 1 128 0 127 0 1 1 127 0 1 1 128 0 127 0 1 1 127 0 1 1 f202 0 1024 7 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 f203 0 1024 7 1 1024 0 ;---------------------------------- ; Note list i1 0 10 6.00 1 s i2 1 10 s i3 1 23 i4 1 23 101 1 .5 200 202 1.5 i4 . . 102 0 .5 200 201 1 i4 . . 103 0 .5 200 201 1 i4 . . 100 0 .25 200 200 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/qnan.csd.xml0000644000000000000000000000446012262561504017614 0ustar rootroot <CsoundSynthesizer> <CsOptions> -+rtaudio=alsa -o dac:hw:0 </CsOptions> <CsInstruments> nchnls = 2 ksmps = 400 #define WII_B #3# #define WII_A #4# #define WII_R_A #304# #define WII_PITCH #20# #define WII_ROLL #21# gkcnt init 1 instr 1 i1 wiiconnect 3,1 wiirange $WII_PITCH., -20, 0 kt wiidata $WII_B. ka wiidata $WII_A. kra wiidata $WII_R_A. gka wiidata $WII_PITCH. gkp wiidata $WII_ROLL. ; If the B (trigger) button is pressed then activate a note if (kt==0) goto ee if (qnan(gka)) goto ee if (qnan(gkp)) goto ee event "i", 2, 0, 5 gkcnt = gkcnt + 1 printk2 kb endin instr 2 a1 oscil ampdbfs(gka), 440+gkp, 1 outs a1, a1 endin </CsInstruments> <CsScore> f1 0 4096 10 1 i1 0 300 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zamod.csd.xml0000644000000000000000000000603012262561504017764 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zamod.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 2 a-rate variables and 2 k-rate variables. zakinit 2, 2 ; Instrument #1 -- a simple waveform. instr 1 ; Vary an a-rate signal linearly from 20,000 to 0. asig line 20000, p3, 0 ; Send the signal to za variable #1. zaw asig, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Generate a simple sine wave. asin oscil 1, 440, 1 ; Modify the sine wave, multiply its amplitude by ; za variable #1. a1 zamod asin, -1 ; Generate the audio output. out a1 ; Clear the za variables, prepare them for ; another pass. zacl 0, 2 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 ; Play Instrument #2 for 2 seconds. i 2 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/connect.csd.xml0000644000000000000000000001563112262561504020312 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o connect.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Michael Gogins */ ; Initialize the global variables. sr = 44100 ksmps = 32 nchnls = 2 ; Connect up the instruments to create a signal flow graph. connect "SimpleSine", "leftout", "Reverberator", "leftin" connect "SimpleSine", "rightout", "Reverberator", "rightin" connect "Moogy", "leftout", "Reverberator", "leftin" connect "Moogy", "rightout", "Reverberator", "rightin" connect "Reverberator", "leftout", "Compressor", "leftin" connect "Reverberator", "rightout", "Compressor", "rightin" connect "Compressor", "leftout", "Soundfile", "leftin" connect "Compressor", "rightout", "Soundfile", "rightin" ; Turn on the "effect" units in the signal flow graph. alwayson "Reverberator", 0.91, 12000 alwayson "Compressor" alwayson "Soundfile" instr SimpleSine ihz = cpsmidinn(p4) iamplitude = ampdb(p5) print ihz, iamplitude ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. isine ftgenonce 0, 0, 4096, 10, 1 a1 oscili iamplitude, ihz, isine aenv madsr 0.05, 0.1, 0.5, 0.2 asignal = a1 * aenv ; Stereo audio outlet to be routed in the orchestra header. outleta "leftout", asignal * 0.25 outleta "rightout", asignal * 0.75 endin instr Moogy ihz = cpsmidinn(p4) iamplitude = ampdb(p5) ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. isine ftgenonce 0, 0, 4096, 10, 1 asignal vco iamplitude, ihz, 1, 0.5, isine kfco line 200, p3, 2000 krez init 0.9 asignal moogvcf asignal, kfco, krez, 100000 ; Stereo audio outlet to be routed in the orchestra header. outleta "leftout", asignal * 0.75 outleta "rightout", asignal * 0.25 endin instr Reverberator ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" idelay = p4 icutoff = p5 aleftout, arightout reverbsc aleftin, arightin, idelay, icutoff ; Stereo output. outleta "leftout", aleftout outleta "rightout", arightout endin instr Compressor ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" kthreshold = 25000 icomp1 = 0.5 icomp2 = 0.763 irtime = 0.1 iftime = 0.1 aleftout dam aleftin, kthreshold, icomp1, icomp2, irtime, iftime arightout dam arightin, kthreshold, icomp1, icomp2, irtime, iftime ; Stereo output. outleta "leftout", aleftout outleta "rightout", arightout endin instr Soundfile ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" outs aleftin, arightin endin </CsInstruments> <CsScore> ; Not necessary to activate "effects" or create f-tables in the score! ; Overlapping notes to create new instances of instruments. i "SimpleSine" 1 5 60 85 i "SimpleSine" 2 5 64 80 i "Moogy" 3 5 67 75 i "Moogy" 4 5 71 70 ;6 extra seconds after the performance e 12 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ceil.csd.xml0000644000000000000000000000323312262561504017570 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 inum = p4 iceil = ceil(inum) print iceil endin </CsInstruments> <CsScore> i 1 0 0 1 i . . . 0.999999 i . . . 0.000001 i . . . 0 i . . . -0.0000001 i . . . -0.9999999 i . . . -1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftconv.csd.xml0000644000000000000000000001237712262561504020164 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ftconv.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 48000 ksmps = 32 nchnls = 2 0dbfs = 1 garvb init 0 gaW init 0 gaX init 0 gaY init 0 itmp ftgen 1, 0, 64, -2, 2, 40, -1, -1, -1, 123, \ 1, 13.000, 0.05, 0.85, 20000.0, 0.0, 0.50, 2, \ 1, 2.000, 0.05, 0.85, 20000.0, 0.0, 0.25, 2, \ 1, 16.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 9.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 12.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 8.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2 itmp ftgen 2, 0, 262144, -2, 0 spat3dt 2, -0.2, 1, 0, 1, 1, 2, 0.005 itmp ftgen 3, 0, 262144, -52, 3, 2, 0, 4, 2, 1, 4, 2, 2, 4 instr 1 a1 vco2 1, 440, 10 kfrq port 100, 0.008, 20000 a1 butterlp a1, kfrq a2 linseg 0, 0.003, 1, 0.01, 0.7, 0.005, 0, 1, 0 a1 = a1 * a2 * 2 denorm a1 vincr garvb, a1 aw, ax, ay, az spat3di a1, p4, p5, p6, 1, 1, 2 vincr gaW, aw vincr gaX, ax vincr gaY, ay endin instr 2 denorm garvb ; skip as many samples as possible without truncating the IR arW, arX, arY ftconv garvb, 3, 2048, 2048, (65536 - 2048) aW = gaW + arW aX = gaX + arX aY = gaY + arY garvb = 0 gaW = 0 gaX = 0 gaY = 0 aWre, aWim hilbert aW aXre, aXim hilbert aX aYre, aYim hilbert aY aWXr = 0.0928*aXre + 0.4699*aWre aWXiYr = 0.2550*aXim - 0.1710*aWim + 0.3277*aYre aL = aWXr + aWXiYr aR = aWXr - aWXiYr outs aL, aR endin </CsInstruments> <CsScore> i 1 0 0.5 0.0 2.0 -0.8 i 1 1 0.5 1.4 1.4 -0.6 i 1 2 0.5 2.0 0.0 -0.4 i 1 3 0.5 1.4 -1.4 -0.2 i 1 4 0.5 0.0 -2.0 0.0 i 1 5 0.5 -1.4 -1.4 0.2 i 1 6 0.5 -2.0 0.0 0.4 i 1 7 0.5 -1.4 1.4 0.6 i 1 8 0.5 0.0 2.0 0.8 i 2 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/unirand.csd.xml0000644000000000000000000000464012262561504020317 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o unirand.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values ktri unirand 100 printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 ktri unirand 100 printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dust.csd.xml0000644000000000000000000000301112262561504017625 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kdens expon 2, p3, 20000 aout dust 0.5, kdens outs aout, aout endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cossegr.csd.xml0000644000000000000000000000351712262561504020326 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o linsegr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .3 kenv cossegr 1, .05, 0.5, 1, 0 asig pluck kenv, icps, 200, 1, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 4096 10 1 ;sine wave f0 30 ;runs 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen20.csd.xml0000644000000000000000000000624112262561504017571 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen20.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 insnd = 10 ;"fox.wav" ibasfrq = 44100 / ftlen(insnd) ;use original sample rate of insnd file kamp expseg .001, p3/2, .7, p3/2, .8 ;envelope kpitch line ibasfrq, p3, ibasfrq * .8 kdens line 600, p3, 10 kaoff line 0, p3, .1 kpoff line 0, p3, ibasfrq * .5 kgdur line .04, p3, .001 ;shorten duration of grain during note imaxgdur = .5 igfn = p4 ;different windows asigL grain kamp, kpitch, kdens, kaoff, kpoff, kgdur, insnd, igfn, imaxgdur, 0.0 asigR grain kamp, kpitch, kdens, kaoff, kpoff, kgdur, insnd, igfn, imaxgdur, 0.0 outs asigL, asigR endin </CsInstruments> <CsScore> f1 0 512 20 2 ;Hanning window f2 0 512 20 6 1 ;Gaussian window f10 0 16384 1 "fox.wav" 0 0 0 i1 0 5 1 ;use Hanning window i1 + 5 2 ;use Gaussian window e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pdhalfy.csd.xml0000644000000000000000000001070312262561504020303 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pdhalfy.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 5 idur = p3 iamp = p4 ifreq = p5 iamtinit = p6 ; initial amount of phase distortion iatt = p7 ; attack time isuslvl = p8 ; sustain amplitude idistdec = p9 ; time for distortion amount to reach zero itable = p10 idec = idistdec - iatt irel = .05 isus = idur - (idistdec + irel) aenv linseg 0, iatt, 1.0, idec, isuslvl, isus, isuslvl, irel, 0, 0, 0 kamount linseg -iamtinit, idistdec, 0.0, idur-idistdec, 0.0 aosc phasor ifreq apd pdhalfy aosc, kamount aout tablei apd, itable, 1 outs aenv*aout*iamp, aenv*aout*iamp endin </CsInstruments> <CsScore> f1 0 16385 10 1 ; sine f3 0 16385 9 1 1 270 ; inverted cosine ; descending "just blues" scale ; pdhalfy with cosine table t 0 100 i5 0 .333 .6 512 1.0 .02 0.5 .12 3 i. + . . 448 < i. + . . 384 < i. + . . 358.4 < i. + . . 341.33 < i. + . . 298.67 < i. + 2 . 256 0.5 s ; pdhalfy with sine table t 0 100 i5 0 .333 .6 512 1.0 .001 0.1 .07 1 i. + . . 448 < i. + . . 384 < i. + . . 358.4 < i. + . . 341.33 < i. + . . 298.67 < i. + 2 . 256 0.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/scans.csd.xml0000644000000000000000000000563712262561504017775 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o scans.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 1 instr 1 a0 = 0 ; scanu init, irate, ifnvel, ifnmass, ifnstif, ifncentr, ifndamp, kmass, kstif, kcentr, kdamp, ileft, iright, kpos, kstrngth, ain, idisp, id scanu 1, .01, 6, 2, 3, 4, 5, 2, .1, .1, -.01, .1, .5, 0, 0, a0, 1, 2 ;ar scans kamp, kfreq, ifntraj, id a1 scans ampdb(p4), cpspch(p5), 7, 2 out a1 endin </CsInstruments> <CsScore> ; Initial condition f1 0 128 7 0 64 1 64 0 ; Masses f2 0 128 -7 1 128 1 ; Spring matrices f3 0 16384 -23 "string-128.matrix" ; Centering force f4 0 128 -7 0 128 2 ; Damping f5 0 128 -7 1 128 1 ; Initial velocity f6 0 128 -7 0 128 0 ; Trajectories f7 0 128 -5 .001 128 128 ; Note list i1 0 10 86 6.00 i1 11 14 86 7.00 i1 15 20 86 5.00 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/grain.csd.xml0000644000000000000000000000523312262561504017756 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o grain.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 insnd = 10 ibasfrq = 44100 / ftlen(insnd) ; Use original sample rate of insnd file kamp expseg .001, p3/2, .1, p3/2, .01 ;a swell in amplitude kpitch line ibasfrq, p3, ibasfrq * .8 kdens line 600, p3, 100 kaoff line 0, p3, .1 kpoff line 0, p3, ibasfrq * .5 kgdur line .4, p3, .01 imaxgdur = .5 asigL grain kamp, kpitch, kdens, kaoff, kpoff, kgdur, insnd, 5, imaxgdur, 0.0 asigR grain kamp, kpitch, kdens, kaoff, kpoff, kgdur, insnd, 5, imaxgdur, 0.0 outs asigL, asigR endin </CsInstruments> <CsScore> f5 0 512 20 2 ; Hanning window f10 0 16384 1 "beats.wav" 0 0 0 i1 0 15 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/reinit.csd.xml0000644000000000000000000000376612262561504020161 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o reinit.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 instr 1 reset: timout 0, p3/10, contin reinit reset contin: kLine expon 440, p3/10, 880 aSig oscil 10000, kLine, 1 out aSig rireturn endin </CsInstruments> <CsScore> f1 0 4096 10 1 i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/inq.csd.xml0000644000000000000000000000530212262561504017442 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -idac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; inq.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 ;2 channels out 0dbfs = 1 nchnls_i = 4 ;4 channels in instr 1 ;4 channels in, two channels out ain1, ain2, ain3, ain4 inq ;grab your mics and sing adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time = 20ms adel2 linseg 0.02, p3*.5, 0, p3*.5, 0.02 ;max delay time = 20ms aoutl flanger ain1, adel, .7 aoutr flanger ain2, adel*2, .8 aoutla flanger ain3, adel2, .9 aoutra flanger ain4, adel2*2, .5 ;write to quad file, 16 bits with header fout "in_4.wav", 14, aoutl, aoutr, aoutla, aoutra outs (aoutl+aoutla)*.5, (aoutr+aoutra)*.5 ;stereo out endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsmorph2.csd.xml0000644000000000000000000001312512262561504020615 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 ; this example uses the files "flute-C-octave0.wav" and ; "saxophone-alto-C-octave0.wav" from www.archive.org/details/OpenPathMusic44V2 giSine ftgen 0, 0, 4096, 10, 1 instr 1 iampint1 = p4; value for interpolating the amplitudes at the beginning ... iampint2 = p5; ... and at the end ifrqint1 = p6; value for unterpolating the frequencies at the beginning ... ifrqint2 = p7; ... and at the end kampint linseg iampint1, p3, iampint2 kfrqint linseg ifrqint1, p3, ifrqint2 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "flute-C-octave0.wav" Sfile2 = "saxophone-alto-C-octave0.wav" ain1 soundin Sfile1 ain2 soundin Sfile2 fftin1 pvsanal ain1, ifftsize, ioverlap, iwinsize, iwinshape fftin2 pvsanal ain2, ifftsize, ioverlap, iwinsize, iwinshape fmorph pvsmorph fftin1, fftin2, kampint, kfrqint aout pvsynth fmorph out aout * .5 endin instr 2; moving randomly in certain borders between two spectra iampintmin = p4; minimum value for amplitudes iampintmax = p5; maximum value for amplitudes ifrqintmin = p6; minimum value for frequencies ifrqintmax = p7; maximum value for frequencies imovefreq = p8; frequency for generating new random values kampint randomi iampintmin, iampintmax, imovefreq kfrqint randomi ifrqintmin, ifrqintmax, imovefreq ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "flute-C-octave0.wav" Sfile2 = "saxophone-alto-C-octave0.wav" ain1 soundin Sfile1 ain2 soundin Sfile2 fftin1 pvsanal ain1, ifftsize, ioverlap, iwinsize, iwinshape fftin2 pvsanal ain2, ifftsize, ioverlap, iwinsize, iwinshape fmorph pvsmorph fftin1, fftin2, kampint, kfrqint aout pvsynth fmorph out aout * .5 endin </CsInstruments> <CsScore> i 1 0 3 0 0 1 1; amplitudes from flute, frequencies from saxophone i 1 3 3 1 1 0 0; amplitudes from saxophone, frequencies from flute i 1 6 3 0 1 0 1; amplitudes and frequencies moving from flute to saxophone i 1 9 3 1 0 1 0; amplitudes and frequencies moving from saxophone to flute i 2 12 3 .2 .8 .2 .8 5; amps and freqs moving randomly between the two spectra e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/foutir.csd.xml0000644000000000000000000000450012262561504020162 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o foutir.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gihand fiopen "foutir.sco", 0 instr 1 ; play virtual midi keyboard inot notnum ;just for priting on screen icps cpsmidi iamp ampmidi 1 foutir gihand, 0, 1, icps, iamp prints "WRITING:\n" prints "note = %f,velocity = %f\n", icps, iamp ;prints them ficlose gihand asig pluck iamp, icps, 1000, 0, 1 outs asig, asig endin </CsInstruments> <CsScore> f 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mintab.csd.xml0000644000000000000000000000132212262561504020123 0ustar rootroot <CsoundSynthesizer> <CsInstruments> instr 1 t1 init 10 t1[3] = 42 k1 mintab t1 printk2 k1 endin </CsInstruments> <CsScore> i1 0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/maxarray.csd.xml0000644000000000000000000000403412262561504020500 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> ;example by joachim heintz seed 0 instr 1 ;create an array with 10 elements kArr[] init 10 ;fill in random numbers and print them out kIndx = 0 until kIndx == 10 do kNum random -100, 100 kArr[kIndx] = kNum printf "kArr[%d] = %10f\n", kIndx+1, kIndx, kNum kIndx += 1 od ;investigate maximum number and print it out kMax, kMaxIndx maxarray kArr printf "Maximum of kArr = %f at index %d\n", kIndx+1, kMax, kMaxIndx turnoff endin </CsInstruments> <CsScore> i1 0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outq.csd.xml0000644000000000000000000000637112262561504017652 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outq.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 asig vco2 .01, 110 ; sawtooth waveform at low volume ;filter the first channel kcut1 line 60, p3, 300 ; Vary cutoff frequency kresonance1 = 3 inumlayer1 = 3 asig1 lowresx asig, kcut1, kresonance1, inumlayer1 ;filter the second channel kcut2 line 300, p3, 60 ; Vary cutoff frequency kresonance2 = 3 inumlayer2 = 3 asig2 lowresx asig, kcut2, kresonance2, inumlayer2 ;filter the third channel kcut3 line 30, p3, 100; Vary cutoff frequency kresonance3 = 6 inumlayer3 = 3 asig3 lowresx asig, kcut3, kresonance3, inumlayer3 asig3 = asig3*.1 ; lower volume ;filter the fourth channel kcut4 line 100, p3, 30; Vary cutoff frequency kresonance4 = 6 inumlayer4 = 3 asig4 lowresx asig, kcut4, kresonance4, inumlayer4 asig4 = asig4*.1 ; lower volume outq asig1, asig2, asig3, asig4; output channels 1, 2, 3 & 4 endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/miditempo.csd.xml0000644000000000000000000000452112262561504020644 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -F Anna.mid ;;;realtime audio out and midi file input ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o miditempo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 massign 0, 1 ; make sure that all channels pgmassign 0, 1 ; and programs are assigned to test instr instr 1 ksig miditempo prints "miditempo = %d\\n", ksig icps cpsmidi ; convert midi note to pitch kenv madsr 0.1, 0, 0.8, 0.3 asig pluck kenv*.15, icps, icps, 1, 1 ;low volume outs asig, asig endin </CsInstruments> <CsScore> f 0 200 ;stay active for 120 seconds f 1 0 4096 10 1 ;sine e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/timout.csd.xml0000644000000000000000000000573512262561504020206 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o timout.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 indx = 0 itim = p4 ;change time for one step clock: timout 0, itim, time reinit clock time: itmp table indx, 2, 0, 0, 1 if itmp == 1 then print itmp event_i "i",2, 0, .1 ;event has duration of .1 second endif indx = indx+1 endin instr 2 ;play it kenv transeg 0.01, p3*0.25, 1, 1, p3*0.75, .5, 0.01 asig oscili kenv*.4, 400, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 1024 10 1 ;sine f 2 0 16 2 1 0 0 1 0 1 0 1 0 1 0 0 0 0 0 0 ;the rythm table i1 0 10 .1 i1 + 10 .05 i1 + 10 .01 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/diff.csd.xml0000644000000000000000000000517412262561504017572 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o diff.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig diskin2 "fox.wav", 1 outs asig, asig endin instr 2 ; with diff asig diskin2 "fox.wav", 1 ares diff asig outs ares, ares endin instr 3 ; with integ asig diskin2 "fox.wav", 1 aint integ asig aint = aint*.05 ;way too loud outs aint, aint endin instr 4 ; with diff and integ asig diskin2 "fox.wav", 1 ares diff asig aint integ ares outs aint, aint endin </CsInstruments> <CsScore> i 1 0 1 i 2 1 1 i 3 2 1 i 4 3 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/strcmp.csd.xml0000644000000000000000000000606712262561504020174 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o strcmp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;modified example from Joachim Heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 opcode Triad, iii, S ;define UDO Sname xin iMaj strcmp "maj", Sname iMin strcmp "min", Sname iPrim = 8.00 ;notes in pitch notattion iQuint = 8.05 if iMaj == 0 then iTer = 8.03 elseif iMin == 0 then iTer = 8.02 endif xout iPrim, iTer, iQuint endop instr 1 Sname strget p4 ia, ib, ic Triad Sname ;apply UDO print ia, ib, ic asig1 pluck 0.7, cpspch(ia), 220, 0, 1 asig2 pluck 0.7, cpspch(ib), 220, 0, 1 asig3 pluck 0.7, cpspch(ic), 220, 0, 1 asig = (asig1+asig2+asig3)*.5 outs asig, asig endin </CsInstruments> <CsScore> i1 0 3 "maj" i1 4 3 "min" </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tabmorpha.csd.xml0000644000000000000000000000537512262561504020642 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tabmorpha.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 giSine ftgen 0, 0, 8193, 10, 1 ; sine wave giSquare ftgen 0, 0, 8193, 7, 1, 4096, 1, 0, -1, 4096, -1 ; square wave giTri ftgen 0, 0, 8193, 7, 0, 2048, 1, 4096, -1, 2048, 0 ; triangle wave giSaw ftgen 0, 0, 8193, 7, 1, 8192, -1 ; sawtooth wave, downward slope instr 1 iamp = .7 aindex phasor 110 ; read table value at this index aweightpoint = 0 ; set weightpoint atabnum1 line 0, p3, 3 ; morph through all tables atabnum2 = 2 ; set to triangle wave asig tabmorpha aindex, aweightpoint, atabnum1,atabnum2, giSine, giSquare, giTri, giSaw asig = asig*iamp outs asig, asig endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fof.csd.xml0000644000000000000000000001152012262561504017424 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fof.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ; Combine five formants together to create ; a transformation from an alto-"a" sound ; to an alto-"i" sound. ; Values common to all of the formants. kfund init 261.659 koct init 0 kris init 0.003 kdur init 0.02 kdec init 0.007 iolaps = 100 ifna = 1 ifnb = 2 itotdur = p3 ; First formant. k1amp = ampdb(0) k1form line 800, p3, 350 k1band line 80, p3, 50 ; Second formant. k2amp line ampdb(-4), p3, ampdb(-20) k2form line 1150, p3, 1700 k2band line 90, p3, 100 ; Third formant. k3amp line ampdb(-20), p3, ampdb(-30) k3form line 2800, p3, 2700 k3band init 120 ; Fourth formant. k4amp init ampdb(-36) k4form line 3500, p3, 3700 k4band line 130, p3, 150 ; Fifth formant. k5amp init ampdb(-60) k5form init 4950 k5band line 140, p3, 200 a1 fof k1amp, kfund, k1form, koct, k1band, kris, \ kdur, kdec, iolaps, ifna, ifnb, itotdur a2 fof k2amp, kfund, k2form, koct, k2band, kris, \ kdur, kdec, iolaps, ifna, ifnb, itotdur a3 fof k3amp, kfund, k3form, koct, k3band, kris, \ kdur, kdec, iolaps, ifna, ifnb, itotdur a4 fof k4amp, kfund, k4form, koct, k4band, kris, \ kdur, kdec, iolaps, ifna, ifnb, itotdur a5 fof k5amp, kfund, k5form, koct, k5band, kris, \ kdur, kdec, iolaps, ifna, ifnb, itotdur ; Combine all of the formants together asig sum (a1+a2+a3+a4+a5) * 13000 outs asig, asig endin </CsInstruments> <CsScore> ; sine wave f 1 0 4096 10 1 ; sigmoid wave f 2 0 1024 19 0.5 0.5 270 0.5 i 1 0 1 i 1 2 5 ; same but slower e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/babo.csd.xml0000644000000000000000000000656412262561504017571 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o babo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Nicola Bernardini */ sr = 44100 ksmps = 32 nchnls = 2 ; minimal babo instrument ; instr 1 ix = p4 ; x position of source iy = p5 ; y position of source iz = p6 ; z position of source ixsize = p7 ; width of the resonator iysize = p8 ; depth of the resonator izsize = p9 ; height of the resonator ainput soundin "beats.wav" al,ar babo ainput*0.7, ix, iy, iz, ixsize, iysize, izsize outs al,ar endin </CsInstruments> <CsScore> /* Written by Nicola Bernardini */ ; simple babo usage: ; ;p4 : x position of source ;p5 : y position of source ;p6 : z position of source ;p7 : width of the resonator ;p8 : depth of the resonator ;p9 : height of the resonator ; i 1 0 20 6 4 3 14.39 11.86 10 ; ^^^^^^^ ^^^^^^^^^^^^^^ ; ||||||| ++++++++++++++: optimal room dims according to ; ||||||| Milner and Bernard JASA 85(2), 1989 ; +++++++++: source position e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpsmidi.csd.xml0000644000000000000000000000322612262561504020306 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M0 ;;;RT audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpsmidi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi asig oscil 0.6, icps, 1 print icps outs asig, asig endin </CsInstruments> <CsScore> f0 20 ;sine wave. f 1 0 16384 10 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dssiplay_stereo.csd.xml0000644000000000000000000000575412262561504022077 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o DSSIplay_stereo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 gihandle dssiinit "g2reverb.so", 0, 1 gaout init 0 instr 1 ; activate DSSI dssiactivate gihandle, 1 endin instr 2 ain1 diskin2 "beats.wav", 1 gaout = gaout+(ain1*.3) endin instr 3 dssictls gihandle, 4, 100, 1 ; room 10 to 150 dssictls gihandle, 5, 10, 1 ; reverb time 1 to 20 dssictls gihandle, 6, .5, 1 ; input bandwith 0 to 1 dssictls gihandle, 7, .25, 1 ; damping 0 to 1 dssictls gihandle, 8, 0, 1 ; dry -80 to 0 dssictls gihandle, 9, -10, 1 ; reflections -80 to 0 dssictls gihandle, 10, -15, 1 ; rev. tail -80 to 0 endin instr 4 aout1, aout2 dssiaudio gihandle, gaout, gaout ;get beats.wav and outs aout1,aout2 ; stereo DSSI plugin gaout = 0 endin </CsInstruments> <CsScore> i 1 0 2 i 2 1 10 i 3 1 10 i 4 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/readk2.csd.xml0000644000000000000000000000633612262561504020033 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o readk2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes two control signals to a file kfreq randh 100, 1, 2, 1, 500 ;generates one random number between 400 and 600 per second kdb randh 12, 1, 2, 1, -12 ;amplitudes in dB between -24 and 0 dumpk2 kfreq, kdb, "dumpk2.txt", 8, 1 ;writes the control signals prints "WRITING:\n" printks "kfreq = %f, kdb = %f\n", 1, kfreq, kdb ;prints them endin instr 2 ;reads the file written by instr 1 kf,kdb readk2 "dumpk2.txt", 8, 1 prints "READING:\n" printks "kfreq = %f, kdb = %f\n", 1, kf, kdb ;prints again kdb lineto kdb, .1 ;smoothing amp transition aout poscil ampdb(kdb), kf, giSine outs aout, aout endin </CsInstruments> <CsScore> i 1 0 5 i 2 5 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen33.csd.xml0000644000000000000000000001000712262561504017570 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen33.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; partials 1, 4, 7, 10, 13, 16, etc. with base frequency of 400 Hz ibsfrq = 400 inumh = int(1.5 + sr * 0.5 / (3 * ibsfrq)) ; estimate number of partials isrcln = int(0.5 + exp(log(2) * int(1.01 + log(inumh * 3) / log(2)))) ; source table length itmp ftgen 1, 0, isrcln, -2, 0 ; create empty source table ifpos = 0 ifrq = ibsfrq inumh = 0 l1: tableiw ibsfrq / ifrq, ifpos, 1 ; amplitude tableiw ifrq, ifpos + 1, 1 ; frequency tableiw 0, ifpos + 2, 1 ; phase ifpos = ifpos + 3 ifrq = ifrq + ibsfrq * 3 inumh = inumh + 1 if (ifrq < (sr * 0.5)) igoto l1 itemp ftgen 2, 0, 262144, -33, 1, inumh, 1, -1 ; store output in ftable 2 (size = 262144) asig poscil .5, ibsfrq, itemp outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/veloc.csd.xml0000644000000000000000000000336412262561504017771 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisine ftgen 0, 0, 1024, 10, 1 instr 1 ivel veloc 0, 1 ;scale 0 - 1 print ivel ;print velocity asig poscil .5*ivel, 220, gisine ;and use it as amplitude outs asig, asig endin </CsInstruments> <CsScore> f 0 30 ;runs 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLvalue.csd.xml0000644000000000000000000000617712262561504020224 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLvalue.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Using the opcode flvalue to display the output of a slider sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Value Display Box", 900, 200, 50, 50 ; Width of the value display box in pixels iwidth = 50 ; Height of the value display box in pixels iheight = 20 ; Distance of the left edge of the value display ; box from the left edge of the panel ix = 65 ; Distance of the top edge of the value display ; box from the top edge of the panel iy = 55 idisp FLvalue "Hertz", iwidth, iheight, ix, iy gkfreq, ihandle FLslider "Frequency", 200, 5000, -1, 5, idisp, 750, 30, 125, 50 FLsetVal_i 500, ihandle ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 iamp = 15000 ifn = 1 asig oscili iamp, gkfreq, ifn out asig endin </CsInstruments> <CsScore> ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fprints.csd.xml0000644000000000000000000000373512262561504020350 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o fprints.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a score generator example. instr 1 ; Print to the file "my.sco". fprints "my.sco", "%!Generated score by ma++\\n \\n" endin </CsInstruments> <CsScore> /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Play Instrument #1. i 1 0 0.001 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tableiw.csd.xml0000644000000000000000000000674512262561504020316 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tableiw.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 ;generate new values every time the instr is played instr 1 ifn = p4 isize = p5 ithresh = 0.5 itemp ftgen ifn, 0, isize, 21, 2 iwrite_value = 0 i_index = 0 loop_start: iread_value tablei i_index, ifn if iread_value > ithresh then iwrite_value = 1 else iwrite_value = -1 endif tableiw iwrite_value, i_index, ifn loop_lt i_index, 1, isize, loop_start turnoff endin instr 2 ifn = p4 isize = ftlen(ifn) prints "Index\tValue\n" i_index = 0 loop_start: ivalue tablei i_index, ifn prints "%d:\t%f\n", i_index, ivalue loop_lt i_index, 1, isize, loop_start ;read table 1 with our index aout oscili .5, 100, ifn ;use table to play the polypulse outs aout, aout endin </CsInstruments> <CsScore> i 1 0 1 100 16 i 2 0 2 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vdelayx.csd.xml0000644000000000000000000000444312262561504020334 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelayx.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ims = .5 ;maximum delay time in seconds iws = 1024 ;window size adl = .5 ;delay time asig diskin2 "fox.wav", 1, 0, 1 ;loop fox.wav a2 poscil .2, .2, 1 ;make an LFO adl = a2 + ims/2 ;offset the LFO so that it is positive aout vdelayx asig, adl, ims, iws ;use the LFO to control delay time outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vdelayxs.csd.xml0000644000000000000000000000456712262561504020526 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelayxs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ims = .5 ;maximum delay time in seconds iws = 1024 ;window size adl = .5 ;delay time asig1, asig2 diskin2 "kickroll.wav", 1, 0, 1 ;loop stereo file kickroll.wav a2 poscil .25, .1, 1 ;make an LFO, 1 cycle per 2 seconds adl = a2 + ims/2 ;offset the LFO so that it is positive aoutL, aoutR vdelayxs asig1, asig2, adl, ims, iws ;use the LFO to control delay time outs aoutL, aoutR endin </CsInstruments> <CsScore> f1 0 8192 10 1 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ficlose.csd.xml0000644000000000000000000000314312262561504020300 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ficlose.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gihand fiopen "test1.txt", 0 instr 1 ires random 0, 100 fouti gihand, 0, 1, ires ficlose gihand endin </CsInstruments> <CsScore> i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/strget.csd.xml0000644000000000000000000000324512262561504020167 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o strget.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 strset 1, "fox.wav" strset 2, "beats.wav" instr 1 Sfile strget p4 asig soundin Sfile outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2.7 1 i 1 + 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/0dbfs.csd.xml0000644000000000000000000000575112262561504017661 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o 0dbfs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 ksmps = 10 nchnls = 2 ; Set the 0dbfs to 1. 0dbfs = 1 instr 1 ; from linear amplitude (0-1 range) print p4 a1 oscil p4, 440, 1 outs a1, a1 endin instr 2 ; from linear amplitude (0-32767 range) iamp = p4 / 32767 print iamp a1 oscil iamp, 440, 1 outs a1, a1 endin instr 3 ; from dB FS iamp = ampdbfs(p4) print iamp a1 oscil iamp, 440, 1 outs a1, a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 i 1 0 1 1 i 1 + 1 0.5 i 1 + 1 0.1 s i 2 0 1 32767 i 2 + 1 [32767/2] i 2 + 1 [3276.7] s i 3 0 1 0 i 3 + 1 -6 i 3 + 1 -20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/exprandi.csd.xml0000644000000000000000000000345112262561504020470 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o exprand.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 klambda exprandi 20, 1, 3 printk2 klambda ; look aout oscili 0.8, 440+klambda, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pow.csd.xml0000644000000000000000000000445012262561504017463 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o pow.wav ; output to audio file </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Lo-Fi sound kpow = 10 ;exponent kbase line 1, p3, 1.4 ;vary the base kQuantize pow kbase, kpow kQuantize = kQuantize*0.5 ;half the number of steps for each side of a bipolar signal printk2 kQuantize asig diskin2 "fox.wav", 1, 0, 1 ;loop the fox asig = round(asig * kQuantize) / kQuantize ;quantize and scale audio signal outs asig, asig endin </CsInstruments> <CsScore> i1 0 19.2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/table1.inc.xml0000644000000000000000000000033112262561504020020 0ustar rootroot /* table1.inc */ ; Table #1, a sine wave. f 1 0 16384 10 1 /* table1.inc */ manual_src~dfsg/examples-xml/syncphasor.csd.xml0000644000000000000000000000551712262561504021054 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o abs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> instr 1 ; Use two syncphasors - one is the "master", ; the other the "slave" ; master's frequency determines pitch imastercps = cpspch(p4) imaxamp = 10000 ; the slave's frequency affects the timbre kslavecps line imastercps, p3, imastercps * 3 ; the master "oscillator" ; the master has no sync input anosync init 0.0 am, async syncphasor imastercps, anosync ; the slave "oscillator" aout, as syncphasor kslavecps, async adeclick linseg 0.0, 0.05, 1.0, p3 - 0.1, 1.0, 0.05, 0.0 ; Output the slave's phase value which is a rising ; sawtooth wave. This produces aliasing, but hey, this ; this is just an example ;) out aout * adeclick * imaxamp endin </CsInstruments> <CsScore> i1 0 1 7.00 i1 + 0.5 7.02 i1 + . 7.05 i1 + . 7.07 i1 + . 7.09 i1 + 2 7.06 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsarp.csd.xml0000644000000000000000000000452312262561504020172 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc ;;;-d RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsarp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 ksmps = 100 nchnls = 1 0dbfs = 1 instr 1 asig in ; get the signal in idepth = p4 fsig pvsanal asig, 1024, 256, 1024, 1 ; analyse it kbin oscili 0.1, 0.5, 1 ; ftable 1 in the 0-1 range ftps pvsarp fsig, kbin+0.01, idepth, 2 ; arpeggiate it (range 220.5 - 2425.5) atps pvsynth ftps ; synthesise it out atps endin </CsInstruments> <CsScore> f 1 0 4096 10 1 ;sine wave i 1 0 10 0.9 i 1 + 10 0.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/phasor.csd.xml0000644000000000000000000000436112262561504020153 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o phasor.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = 1 ;read table 1 with our index ixmode = 1 kndx phasor p4 kfrq table kndx, ifn, ixmode asig poscil .6, kfrq, 2 ;re-synthesize with sine outs asig, asig endin </CsInstruments> <CsScore> f 1 0 1025 -7 200 1024 2000 ;a line from 200 to 2,000 f 2 0 16384 10 1;sine wave i 1 0 1 1 ;once per second i 1 2 2 .5 ;once per 2 seconds i 1 5 1 2 ;twice per second e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/taninv2.csd.xml0000644000000000000000000000326512262561504020242 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o taninv2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Returns the arctangent for 1/2. i1 taninv2 1, 2 print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lposcilsa.csd.xml0000644000000000000000000000457012262561504020652 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lposcilsa.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 1.3 ;a 3th up kloop = 0 ;loop start time in samples kend = 45000 ;loop end time in samples aenv expsega 0.01, 0.1, 1, 0.1, 0.5, 0.5, 0.01 ;envelope with fast and short segment aL, aR lposcilsa aenv, kcps, kloop, kend, 1 ;use it for amplitude outs aL, aR endin </CsInstruments> <CsScore> ; table size of stereo file is deferred, ; and format taken from the soundfile header. f 1 0 0 1 "kickroll.wav" 0 0 0 ; Play Instrument #1 for 6 seconds. ; This will loop the drum pattern several times. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen34.csd.xml0000644000000000000000000001000612262561504017570 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen34.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; partials 1, 4, 7, 10, 13, 16, etc. with base frequency of 400 Hz ibsfrq = 400 inumh = int(1.5 + sr * 0.5 / (3 * ibsfrq)) ; estimate number of partials isrcln = int(0.5 + exp(log(2) * int(1.01 + log(inumh * 3) / log(2)))) ; source table length itmp ftgen 1, 0, isrcln, -2, 0 ; create empty source table ifpos = 0 ifrq = ibsfrq inumh = 0 l1: tableiw ibsfrq / ifrq, ifpos, 1 ; amplitude tableiw ifrq, ifpos + 1, 1 ; frequency tableiw 0, ifpos + 2, 1 ; phase ifpos = ifpos + 3 ifrq = ifrq + ibsfrq * 3 inumh = inumh + 1 if (ifrq < (sr * 0.5)) igoto l1 itmp ftgen 2, 0, 262144, -34, 1, inumh, 1, -1 ; store output in ftable 2 (size = 262144) asig poscil .5, ibsfrq, itmp outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/rnd31_time.csd.xml0000644000000000000000000000405312262561504020622 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd31_time.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; i-rate random numbers with linear distribution, ; seeding from the current time. (Note that the seed ; was used only in the first call.) i1 rnd31 1, 0.5, 0 i2 rnd31 1, 0.5 i3 rnd31 1, 0.5 print i1 print i2 print i3 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluidCCk.csd.xml0000644000000000000000000000537312262561504020347 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidCCk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "22Bassoon.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 70 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 kpan line 0, p3, 127 ;panning from left to right fluidCCk giengine, 1, 10, kpan ;CC 10 = pan fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 7 asigl, asigr fluidOut giengine outs asigl*imvol, asigr*imvol endin </CsInstruments> <CsScore> i 1 0 4 48 100 i 1 4 2 50 120 i 1 6 1 53 80 i 1 7 1 45 70 i 1 8 1.5 48 80 i 99 0 10 ;keep instr 99 active e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/granule.csd.xml0000644000000000000000000000434612262561504020317 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o granule.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 ; k1 linseg 0,0.5,1,(p3-p2-1),1,0.5,0 a1 granule p4*k1,p5,p6,p7,p8,p9,p10,p11,p12,p13,p14,p15,\ p16,p17,p18,p19,p20,p21,p22,p23,p24 a2 granule p4*k1,p5,p6,p7,p8,p9,p10,p11,p12,p13,p14,p15,\ p16,p17,p18,p19, p20+0.17,p21,p22,p23,p24 outs a1,a2 endin </CsInstruments> <CsScore> ; f statement read sound file mary.wav in the SFDIR ; directory into f-table 1 f1 0 262144 1 "mary.wav" 0 0 0 i1 0 10 2000 64 0.5 0 0 1 4 0 0.005 5 0.01 50 0.02 50 30 30 0.39 \ 1 1.42 0.29 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/out.csd.xml0000644000000000000000000000370112262561504017463 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o out.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 1 0dbfs = 1 instr 1 kamp = .6 kcps = 440 ifn = p4 asig oscil kamp, kcps, ifn out asig ;one channel endin </CsInstruments> <CsScore> f1 0 16384 10 1 ; Sine f2 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth i 1 0 2 1 i 1 3 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/noise-2.csd.xml0000644000000000000000000000422412262561504020131 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac ; -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o noise.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 FLpanel "noise", 200, 50, -1 , -1 gkbeta, gislider1 FLslider "kbeta", -1, 1, 0, 5, -1, 180, 20, 10, 10 FLpanelEnd FLrun instr 1 iamp = 0dbfs / 4 ; Peaks 12 dB below 0dbfs print iamp a1 noise iamp, gkbeta printk2 gkbeta outs a1,a1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one minute. i 1 0 60 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/strcpyk.csd.xml0000644000000000000000000001272612262561504020362 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o strcpyk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 instr 1 ;get one element of the input string whenever the metro ;triggers, and call a subinstrument to play the file Smember strget p4 istrlen strlen Smember kprint init 0 ktrig metro .6 ;whenever the trigger gives signal if ktrig == 1 then ;choose a random element (0, 1 or 2) kel random 0, 3.9999 kel = int(kel) ;make a copy for leaving Smember intact Scopy strcpyk Smember ;set the initial index for reading substrings kndx = 0 ;set counter for searching the element kcount = 0 ;start looping over the elements in Smember loop: kdelim strindexk Scopy, ":" ;as long as ":" occurs in Scopy, do: if kdelim > 0 then ;if this is the element to get if kel == kcount then ;read it as substring Sfile strsubk Scopy, kndx, kdelim kprint = kprint+1 ;and jump out kgoto call ;if not else ;cut off this element from Scopy Scopy strsubk Scopy, kdelim+1, istrlen endif ;if no element has been found,go back to loop ;and look for the next element kcount = kcount+1 kgoto loop ;if there is no delimiter left, the rest is the searched element else Sfile strcpyk Scopy endif call: ;print the result, call the subinstrument and play the file printf "kel = %d, file = '%s'\n", ktrig+kprint, kel, Sfile S_call sprintfk {{i 2 0 1 "%s"}}, Sfile scoreline S_call, ktrig endif endin instr 2 ;play Sfile strget p4 ilen filelen Sfile p3 = ilen asig soundin Sfile outs asig, asig endin </CsInstruments> <CsScore> i 1 0 30 "mary.wav:fox.wav:beats.wav:flute.aiff" e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/phaser1.csd.xml0000644000000000000000000000715012262561504020221 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o phaser1.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; demonstration of phase shifting abilities of phaser1. instr 1 ; Input mixed with output of phaser1 to generate notches. ; Shows the effects of different iorder values on the sound idur = p3 iamp = p4 * .05 iorder = p5 ; number of 1st-order stages in phaser1 network. ; Divide iorder by 2 to get the number of notches. ifreq = p6 ; frequency of modulation of phaser1 ifeed = p7 ; amount of feedback for phaser1 kamp linseg 0, .2, iamp, idur - .2, iamp, .2, 0 iharms = (sr*.4) / 100 asig gbuzz 1, 100, iharms, 1, .95, 2 ; "Sawtooth" waveform modulation oscillator for phaser1 ugen. kfreq oscili 5500, ifreq, 1 kmod = kfreq + 5600 aphs phaser1 asig, kmod, iorder, ifeed out (asig + aphs) * iamp endin </CsInstruments> <CsScore> ; inverted half-sine, used for modulating phaser1 frequency f1 0 16384 9 .5 -1 0 ; cosine wave for gbuzz f2 0 8192 9 1 1 .25 ; phaser1 i1 0 5 7000 4 .2 .9 i1 6 5 7000 6 .2 .9 i1 12 5 7000 8 .2 .9 i1 18 5 7000 16 .2 .9 i1 24 5 7000 32 .2 .9 i1 30 5 7000 64 .2 .9 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/schedule.csd.xml0000644000000000000000000000460012262561504020447 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o schedule.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - oscillator with a high note. instr 1 ; Play Instrument #2 at the same time. schedule 2, 0, p3 ; Play a high note. a1 oscils 10000, 880, 1 out a1 endin ; Instrument #2 - oscillator with a low note. instr 2 ; Play a low note. a1 oscils 10000, 220, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for half a second. i 1 0 0.5 ; Play Instrument #1 for half a second. i 1 1 0.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vincr-complex.csd.xml0000644000000000000000000000506612262561504021450 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vincr-complex.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gaReverbSend init 0 instr 1 iamp = p4 ifreq = p5 aenv linseg 0.0, 0.1*p3, iamp, 0.6*p3, iamp, 0.3*p3, 0.0 aosc poscil aenv, ifreq, 1 vincr gaReverbSend, aosc endin instr 2 ; global reverb instrument al, ar reverbsc gaReverbSend, gaReverbSend, 0.85, 12000 outs gaReverbSend+al, gaReverbSend+ar clear gaReverbSend endin </CsInstruments> <CsScore> f1 0 4096 10 1 { 4 CNT { 8 PARTIAL ; start time duration amplitude frequency i1 [0.5 * $CNT.] [1 + ($CNT * 0.2)] [.04 + (~ * .02)] [800 + (200 * $CNT.) + ($PARTIAL. * 20)] } } i2 0 6 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tabsum.csd.xml0000644000000000000000000000470012262561504020147 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ; For Non-realtime ouput leave only the line below: ; -o tab.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 2205 nchnls = 1 0dbfs = 1 instr 1 ;;; Give a value to the increment kmax = 256 knorm tabsum 1, 0, kmax gkinc = knorm/10 endin instr 2 kmax = 256 kx = rnd(kmax) krnd tabsum 1, 0, kx knorm tabsum 1, 0, kmax kvar = krnd / knorm ;;; now n [0,1] range asig oscil kvar, p4, 2 out asig ;;; Make randomness give 1 more often kc tab 0, 1 tablew kc+gkinc, 0, 1 endin </CsInstruments> <CsScore> f1 0 256 21 1 f2 0 4096 10 1 i1 0 0.1 i2 0.1 3 440 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLknob.csd.xml0000644000000000000000000000705712262561504020037 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLknob.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; A sine with oscillator with flknob controlled frequency sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Frequency Knob", 900, 400, 50, 50 ; Minimum value output by the knob imin = 200 ; Maximum value output by the knob imax = 5000 ; Logarithmic type knob selected iexp = -1 ; Knob graphic type (1=3D knob) itype = 1 ; Display handle (-1=not used) idisp = -1 ; Width of the knob in pixels iwidth = 70 ; Distance of the left edge of the knob ; from the left edge of the panel ix = 70 ; Distance of the top edge of the knob ; from the top of the panel iy = 125 gkfreq, ihandle FLknob "Frequency", imin, imax, iexp, itype, idisp, iwidth, ix, iy ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun ; Set the widget's initial value FLsetVal_i 300, ihandle instr 1 iamp = 15000 ifn = 1 asig oscili iamp, gkfreq, ifn out asig endin </CsInstruments> <CsScore> ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vrandh.csd.xml0000644000000000000000000000564412262561504020146 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vranh.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;Example by Andres Cabrera sr=44100 ksmps=128 nchnls=2 gitab ftgen 0, 0, 16, -7, 0, 128, 0 instr 1 krange init p4 kcps init p5 ioffset init p6 kav1 init 0 kav2 init 0 kcount init 0 ; table krange kcps ielements idstoffset iseed isize ioffset vrandh gitab, krange, kcps, 3, 3, 2, 0, ioffset kfreq1 table 3, gitab kfreq2 table 4, gitab kfreq3 table 5, gitab ;Change the frequency of three oscillators according to the random values aosc1 oscili 4000, kfreq1, 1 aosc2 oscili 2000, kfreq2, 1 aosc3 oscili 4000, kfreq3, 1 outs aosc1+aosc2, aosc3+aosc2 endin </CsInstruments> <CsScore> f 1 0 1024 10 1 ; krange kcps ioffset i 1 0 5 100 1 300 i 1 5 5 300 1 400 i 1 10 5 100 2 1000 i 1 15 5 400 4 1000 i 1 20 5 1000 8 2000 i 1 25 5 250 16 300 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/centroid.csd.xml0000644000000000000000000000413212262561504020462 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o centroid.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 16384, 10, 1 instr 1 ktrig init 1 ;calculate centroid a1 oscil3 0.5, p4, giSine ;of the sine wave k1 centroid a1, ktrig, 16384 asig oscil3 0.5, k1, giSine printk2 k1 ;print & compare: outs a1, asig ;left = original, right = centroid signal endin </CsInstruments> <CsScore> i1 0 2 20 i1 + 2 200 i1 + 2 2000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/define.csd.xml0000644000000000000000000000432112262561504020105 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o define.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Define the macros. #define VOLUME #5000# #define FREQ #440# #define TABLE #1# ; Instrument #1 instr 1 ; Use the macros. ; This will be expanded to "a1 oscil 5000, 440, 1". a1 oscil $VOLUME, $FREQ, $TABLE ; Send it to the output. out a1 endin </CsInstruments> <CsScore> ; Define Table #1 with an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/trirand.csd.xml0000644000000000000000000000464012262561504020322 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trirand.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values ktri trirand 100 printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 ktri trirand 100 printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/strchar.csd.xml0000644000000000000000000000655712262561504020336 0ustar rootroot <CsoundSynthesizer> <CsOptions> -ndm0 </CsOptions> <CsInstruments> ;example by joachim heintz 2013 opcode ToAscii, S, S ;returns the ASCII numbers of the input string as string Sin xin ;input string ilen strlen Sin ;its length ipos = 0 ;set counter to zero Sres = "" ;initialize output string loop: ;for all characters in input string: ichr strchar Sin, ipos ;get its ascii code number Snew sprintf "%d ", ichr ;put this number into a new string Sres strcat Sres, Snew ;append this to the output string loop_lt ipos, 1, ilen, loop ;see comment for 'loop:' xout Sres ;return output string endop instr Characters printf_i "\nCharacters:\n given as single strings: %s%s%s%s%s%s\n", 1, "c", "s", "o", "u", "n", "d" printf_i " but can also be given as numbers: %c%c%c%c%c%c\n", 1, 99, 115, 111, 117, 110, 100 Scsound ToAscii "csound" printf_i " in csound, the ASCII code of a character can be accessed with the opcode strchar.%s", 1, "\n" printf_i " the name 'csound' returns the numbers %s\n\n", 1, Scsound endin </CsInstruments> <CsScore> i "Characters" 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tempoval.csd.xml0000644000000000000000000000361512262561504020507 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc -t60 ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tempoval.wav -W -t60 ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Adjust the tempo to 120 beats per minute. tempo 120, 60 ; Get the tempo value. kval tempoval printks "kval = %f\\n", 0.1, kval endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/oscil3.csd.xml0000644000000000000000000000555412262561504020060 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = .6 kcps = 440 ifn = p4 asig oscil kamp, kcps, ifn outs asig,asig endin instr 2 kamp = .6 kcps = 440 ifn = p4 asig oscil3 kamp, kcps, ifn outs asig,asig endin </CsInstruments> <CsScore> f1 0 128 10 1 ; Sine with a small amount of data f2 0 128 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth with a small amount of data f3 0 128 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square with a small amount of data f4 0 128 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse with a small amount of data i 1 0 2 1 i 2 3 2 1 i 1 6 2 2 i 2 9 2 2 i 1 12 2 3 i 2 15 2 3 i 1 18 2 4 i 2 21 2 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/maxalloc.csd.xml0000644000000000000000000000412612262561504020456 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o maxalloc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 maxalloc 1, 3 ; Limit to three instances. instr 1 asig oscil .3, p4, 1 outs asig, asig endin </CsInstruments> <CsScore> ; sine f 1 0 32768 10 1 i 1 0 5 220 ;1 i 1 1 4 440 ;2 i 1 2 3 880 ;3, limit is reached i 1 3 2 1320 ;is not played i 1 4 1 1760 ;is not played e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/xadsr.csd.xml0000644000000000000000000000425212262561504017777 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o xadsr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iatt = p5 idec = p6 islev = p7 irel = p8 kenv xadsr iatt, idec, islev, irel kcps = cpspch(p4) ;frequency asig vco2 kenv * 0.8, kcps outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 7.00 .0001 1 .01 .001 ; short attack i 1 2 1 7.02 1 .5 .01 .001 ; long attack i 1 4 2 6.09 .0001 1 .1 .7 ; long release e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lpsholdp.csd.xml0000644000000000000000000000471512262561504020507 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpsholdp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipan = p4 ktrig = 0 ; (no retriggering) ; kphase - looping pointer - is generated using a random spline kphase rspline 0,1,0.5,5 ; a loop sequence of midi note numbers and durations ; val dur val dur etc... knote lpsholdp kphase, 60, 1, 59, 0.1, 62, 1, 64, 1, 67, 0.1, 65, 1 asig gbuzz 0.2, cpsmidinn(knote), 30, 1, 0.5, 1 outs asig*ipan, asig*(1-ipan) endin </CsInstruments> <CsScore> ; cosine wave. f 1 0 16384 11 1 ; 2 layers of the loop are played each with a different pan position (p4) i 1 0 60 0.25 i 1 0 60 0.75 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tonek.csd.xml0000644000000000000000000000414312262561504017775 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tonek.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 0, 0, 2^10, 10, 1 instr 1 ksig randomh 400, 1800, 150 aout poscil .2, 100+ksig, gisin outs aout, aout endin instr 2 ksig randomh 400, 1800, 150 khp line 1, p3, 100 ;vary high-pass ksig tonek ksig, khp aout poscil .2, 100+ksig, gisin outs aout, aout endin </CsInstruments> <CsScore> i 1 0 5 i 2 5.5 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tableimix.csd.xml0000644000000000000000000000371612262561504020640 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tableimix.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisine ftgen 1, 0, 256, 10, 1, 0, 0, .4 ;sinoid gisaw ftgen 2, 0, 1024, 7, 0, 256, 1 ;saw gimix ftgen 100, 0, 256, 7, 0, 256, 1 ;used to mix instr 1 tableimix 100, 0, 256, 1, 0, 1, 2, 0, .5 asig poscil .5, 110, gimix ;mix table 1 & 2 outs asig, asig endin </CsInstruments> <CsScore> i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/nchnls.csd.xml0000644000000000000000000000400012262561504020132 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -idac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; nchnls.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 ;two channels out 0dbfs = 1 instr 1 ain1, ainr ins ;grab your mic and sing adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time = 20ms aoutl flanger ain1, adel, .7 aoutr flanger ain1, adel*2, .8 fout "in_s.wav", 14, aoutl, aoutr ;write to stereo file, outs aoutl, aoutr ;16 bits with header endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/loop_le.csd.xml0000644000000000000000000000546112262561504020312 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loop_le.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 gisine ftgen 0, 0, 2^10, 10, 1 instr 1 ;master instrument ininstr = 5 ;number of called instances indx = 0 loop: prints "play instance %d\\n", indx ipan random 0, 1 ifreq random 100, 1000 iamp = 1/ininstr event_i "i", 10, 0, p3, iamp, ifreq, ipan loop_le indx, 1, ininstr, loop endin instr 10 ipeak random 0, 1 ;where is the envelope peak asig poscil3 p4, p5, gisine aenv transeg 0, p3*ipeak, 6, 1, p3-p3*ipeak, -6, 0 aL,aR pan2 asig*aenv, p6 outs aL, aR endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/adsr.csd.xml0000644000000000000000000000424312262561504017607 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o adsr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iatt = p5 idec = p6 islev = p7 irel = p8 kenv adsr iatt, idec, islev, irel kcps = cpspch(p4) ;frequency asig vco2 kenv * 0.8, kcps outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 7.00 .0001 1 .01 .001 ; short attack i 1 2 1 7.02 1 .5 .01 .001 ; long attack i 1 4 2 6.09 .0001 1 .1 .7 ; long release e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/maparray.csd.xml0000644000000000000000000001250112262561504020466 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;create an array and fill with numbers kArrSrc[] fillarray 1.01, 2.02, 3.03, 4.05, 5.08, 6.13, 7.21 ;print source array printf "%s", 1, "\nSource array:\n" kndx = 0 until kndx == lenarray(kArrSrc) do printf "kArrSrc[%d] = %f\n", kndx+1, kndx, kArrSrc[kndx] kndx += 1 od ;create an empty array for the results kArrRes[] init 7 ;apply the sqrt() function to each element kArrRes maparray kArrSrc, "sqrt" ;print the result printf "%s", 1, "\nResult after applying sqrt() to source array\n" kndx = 0 until kndx == lenarray(kArrRes) do printf "kArrRes[%d] = %f\n", kndx+1, kndx, kArrRes[kndx] kndx += 1 od ;apply the log() function to each element kArrRes maparray kArrSrc, "log" ;print the result printf "%s", 1, "\nResult after applying log() to source array\n" kndx = 0 until kndx == lenarray(kArrRes) do printf "kArrRes[%d] = %f\n", kndx+1, kndx, kArrRes[kndx] kndx += 1 od ;apply the int() function to each element kArrRes maparray kArrSrc, "int" ;print the result printf "%s", 1, "\nResult after applying int() to source array\n" kndx = 0 until kndx == lenarray(kArrRes) do printf "kArrRes[%d] = %f\n", kndx+1, kndx, kArrRes[kndx] kndx += 1 od ;apply the frac() function to each element kArrRes maparray kArrSrc, "frac" ;print the result printf "%s", 1, "\nResult after applying frac() to source array\n" kndx = 0 until kndx == lenarray(kArrRes) do printf "kArrRes[%d] = %f\n", kndx+1, kndx, kArrRes[kndx] kndx += 1 od ;turn instrument instance off turnoff endin </CsInstruments> <CsScore> i 1 0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dssiinit.csd.xml0000644000000000000000000000560412262561504020506 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dssiinit.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 gihandle dssiinit "g2reverb.so", 0, 1 gaout init 0 instr 1 ; activate DSSI dssiactivate gihandle, 1 endin instr 2 ain1 diskin2 "beats.wav", 1 gaout = gaout+(ain1*.3) endin instr 3 dssictls gihandle, 4, 100, 1 ; room 10 to 150 dssictls gihandle, 5, 10, 1 ; reverb time 1 to 20 dssictls gihandle, 6, .5, 1 ; input bandwith 0 to 1 dssictls gihandle, 7, .25, 1 ; damping 0 to 1 dssictls gihandle, 8, 0, 1 ; dry -80 to 0 dssictls gihandle, 9, -10, 1 ; reflections -80 to 0 dssictls gihandle, 10, -15, 1 ; rev. tail -80 to 0 endin instr 4 aout1, aout2 dssiaudio gihandle, gaout, gaout ;get beats.wav and outs aout1,aout2 ; stereo DSSI plugin gaout = 0 endin </CsInstruments> <CsScore> i 1 0 2 i 2 1 10 i 3 1 10 i 4 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sfload.csd.xml0000644000000000000000000001025612262561504020127 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out, virtual midi in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfload.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;load two soundfonts isf sfload "07AcousticGuitar.sf2" ir sfload "01hpschd.sf2" sfplist isf sfplist ir sfpassign 0, isf sfpassign 1, ir instr 1 ; play guitar from score and midi keyboard - preset index = 0 mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/3000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay3 ivel, inum, kamp*ivel, kfreq, 0 ;preset index = 0 outs a1, a2 endin instr 2 ; play harpsichord from score and midi keyboard - preset index = 1 mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/1000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay3 ivel, inum, kamp*ivel, kfreq, 1 ;preset index = 1 outs a1, a2 endin </CsInstruments> <CsScore> f0 60 ; stay active for 1 minute i1 0 1 60 100 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 i2 5 1 60 100 i2 + 1 62 < i2 7 1 65 < i2 7 1 69 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/deltapn.csd.xml0000644000000000000000000000636212262561504020311 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o deltap3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasig init 0 gidel = 1 ;delay time in seconds instr 1 ain pluck .7, 440, 1000, 0, 1 outs ain, ain vincr gasig, ain ;send to global delay endin instr 2 ifeedback = p4 abuf2 delayr gidel adelL deltapn 4000 ;first tap (on left channel) adelM deltapn 44100 ;second tap (on middle channel) delayw gasig + (adelL * ifeedback) abuf3 delayr gidel kdel line 100, p3, 1 ;vary delay time adelR deltapn 100 * kdel ;one pitch changing tap (on the right chn.) delayw gasig + (adelR * ifeedback) ;make a mix of all deayed signals outs adelL + adelM, adelR + adelM clear gasig endin </CsInstruments> <CsScore> i 1 0 1 i 1 3 1 i 2 0 3 0 ;no feedback i 2 3 8 .8 ;lots of feedback e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/delayr.csd.xml0000644000000000000000000000635012262561504020137 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o delayr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasig init 0 gidel = 1 ;delay time in seconds instr 1 ain pluck .7, 440, 1000, 0, 1 outs ain, ain vincr gasig, ain ;send to global delay endin instr 2 ifeedback = p4 abuf2 delayr gidel adelL deltap .4 ;first tap (on left channel) adelM deltap 1 ;second tap (on middle channel) delayw gasig + (adelL * ifeedback) abuf3 delayr gidel kdel line 1, p3, .01 ;vary delay time adelR deltap .65 * kdel ;one pitch changing tap (on the right chn.) delayw gasig + (adelR * ifeedback) ;make a mix of all deayed signals outs adelL + adelM, adelR + adelM clear gasig endin </CsInstruments> <CsScore> i 1 0 1 i 1 3 1 i 2 0 3 0 ;no feedback i 2 3 8 .8 ;lots of feedback e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/chuap.csd.xml0000644000000000000000000001044412262561504017756 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chuas_oscillator.wav.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gibuzztable ftgen 1, 0, 16384, 10, 1 instr 1 ; sys_variables = system_vars(5:12); % L,R0,C2,G,Ga,Gb,E,C1 or p8:p15 ; integ_variables = [system_vars(14:16),system_vars(1:2)]; % x0,y0,z0,dataset_size,step_size or p17:p19, p4:p5 istep_size = p5 iL = p8 iR0 = p9 iC2 = p10 iG = p11 iGa = p12 iGb = p13 iE = p14 iC1 = p15 iI3 = p17 iV2 = p18 iV1 = p19 iattack = 0.02 isustain = p3 irelease = 0.02 p3 = iattack + isustain + irelease iscale = 1.0 adamping linseg 0.0, iattack, iscale, isustain, iscale, irelease, 0.0 aguide buzz 0.5, 440, sr/440, gibuzztable aI3, aV2, aV1 chuap iL, iR0, iC2, iG, iGa, iGb, iE, iC1, iI3, iV2, iV1, istep_size asignal balance aV2, aguide outs adamping * asignal, adamping * asignal endin </CsInstruments> <CsScore> ; Adapted from ABC++ MATLAB example data. i 1 0 20 1500 .1 -1 -1 -0.00707925 0.00001647 100 1 -.99955324 -1.00028375 1 -.00222159 204.8 -2.36201596260071 3.08917625807226e-03 3.87075614929199 7 .4 .004 1 86 30; torus attractor ( gallery of attractors ) i 1 + 20 1500 .425 0 -1 1.3506168 0 -4.50746268737 -1 2.4924 .93 1 1 0 -22.28662665 .009506608 -22.2861576 32 10 2 20 86 30 ; heteroclinic orbit i 1 + 20 1024 .05 -1 -1 0.00667 0.000651 10 -1 .856 1.1 1 .06 51.2 -20.200590133667 .172539323568344 -4.07686233520508 2.5 10 .2 1 66 81 ; periodic attractor (torus breakdown route) i 1 + 20 1024 0.05 -1 -1 0.00667 0.000651 10 -1 0.856 1.1 1 0.1 153.6 21.12496758 0.03001749 0.515828669 2.5 10 0.2 1 66 81 ; torus attractor (torus breakdown route)' </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/i_statement.csd.xml0000644000000000000000000001204112262561504021165 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o i_statement.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 10 icps init cpspch(p4) ; Get target pitch from score event iportime init abs(p3)/7 ; Portamento time dep on note length iamp0 init p5 ; Set default amps iamp1 init p5 iamp2 init p5 itie tival ; Check if this note is tied, if itie == 1 igoto nofadein ; if not fade in iamp0 init 0 nofadein: if p3 < 0 igoto nofadeout ; Check if this note is held, if not fade out iamp2 init 0 nofadeout: ; Now do amp from the set values: kamp linseg iamp0, .03, iamp1, abs(p3)-.03, iamp2 ; Skip rest of initialization on tied note: tigoto tieskip kcps init icps ; Init pitch for untied note kcps port icps, iportime, icps ; Drift towards target pitch kpw oscil .4, rnd(1), 1, rnd(.7) ; A simple triangle-saw oscil ar vco kamp, kcps, 3, kpw+.5, 1, 1/icps ; (Used in testing - one may set ipch to cpspch(p4+2) ; and view output spectrum) ; ar oscil kamp, kcps, 1 outs ar, ar tieskip: ; Skip some initialization on tied note endin </CsInstruments> <CsScore> f1 0 8192 10 1 ; Sine i10.1 0 -1 7.00 .15 i10.2 0 -1 7.04 i10.3 0 -1 7.07 i10.1 1 -1 8.00 i10.2 1 -1 8.04 i10.3 1 -1 8.07 i10.1 2 1 7.11 i10.2 2 1 8.04 i10.3 2 1 8.07 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outic.csd.xml0000644000000000000000000000343112262561504017777 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out -+rtmidi=virtual ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outic.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ikey notnum ivel veloc kbrt = 40 ;set controller 74 (=brightness) outic 1, 74, kbrt, 0, 127 ;so filter closes a bit midion 1, ikey, ivel ;play external synth endin </CsInstruments> <CsScore> f0 30 ;runs 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vdelayxws.csd.xml0000644000000000000000000000457112262561504020710 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelayxws.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ims = .5 ;maximum delay time in seconds iws = 1024 ;window size adl = .5 ;delay time asig1, asig2 diskin2 "kickroll.wav", 1, 0, 1 ;loop stereo file kickroll.wav a2 poscil .2, .1, 1 ;make an LFO, 1 cycle per 10 seconds adl = a2 + ims/2 ;offset the LFO so that it is positive aoutL, aoutR vdelayxws asig1, asig2, adl, ims, iws ;use the LFO to control delay time outs aoutL, aoutR endin </CsInstruments> <CsScore> f1 0 8192 10 1 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/port.csd.xml0000644000000000000000000000412312262561504017637 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o port.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 aout diskin2 "fox.wav",1, 0, 1 kf,ka ptrack aout, 512 ; pitch track with winsize=1024 kcps port kf, 0.01 ; smooth freq kamp port ka, 0.01 ; smooth amp ; drive an oscillator asig poscil ampdb(kamp)*0dbfs, kcps, 1 outs asig, asig endin </CsInstruments> <CsScore> ; simple sine wave f 1 0 4096 10 1 i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midinoteonkey.csd.xml0000644000000000000000000000442112262561504021532 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M1 -Q1 ;;;realtime audio out and midi in and midi out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midinoteonkey.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;uses external midi device kkey init 0 ;initialize key number kvel init 0 ;initialize velocity midinoteonkey kkey, kvel ;MIDI note number value printk2 kkey ;display the key value when it changes and when key is pressed midion 1, kkey, kvel ;sent note to external device endin </CsInstruments> <CsScore> f 0 30 ;runs for 30 seconds i 1 0 2 62 ; play these notes from score as well i 1 + 2 65 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/resonxk.csd.xml0000644000000000000000000000513412262561504020347 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o resonxk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 0, 0, 2^10, 10, 1 instr 1 ksig randomh 400, 1800, 150 aout poscil .2, 1000+ksig, gisin outs aout, aout endin instr 2 ksig randomh 400, 1800, 150 kcf line 1, p3, 1000 ;vary high-pass ilay = p4 ksig resonxk ksig, kcf, 100, ilay aout poscil .2, 1000+ksig, gisin asig interp ksig ;convert k-rate to a-rate aout balance asig, aout ;avoid getting asig out of range outs aout, aout endin </CsInstruments> <CsScore> i 1 0 5 i 2 6 5 1 ;number of filter stack = 1 i 2 12 5 5 ;number of filter stack = 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/printk2.csd.xml0000644000000000000000000000464312262561504020253 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o printk2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kval randomh 0, 1.2, 20 ;choose between 0 and 1.2 if kval >0 && kval<=.5 then ;3 possible outcomes kval = 1 elseif kval >.5 && kval<=1 then kval =2 elseif kval >1 then kval =3 endif printk2 kval ;print value when it changes asig poscil .7, 440*kval, 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 16384 10 1 i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/seqtime2.csd.xml0000644000000000000000000000661312262561504020412 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o seqtime2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gitabMap2 ftgen 57,0,512,-2, 1,1/4,1/4,1/8,1/8,1/8,1/8,1/4,1/4,.5,1/4,1/4,1/16,1/16,1/16,1/16,1/16,1/16,1/16,1/16 gisine ftgen 1,0,512,10, 1 instr 1 ktrigin metro .333333333333 ktrig2 metro 1 schedkwhen ktrig2, 0,0, 2, 0, .1 ; just to set the metronome! kspeed init 1 ; ktime_unit, kstart, kloop, initndx, kfn_times ktrig seqtime2 ktrigin, kspeed, 0, 20, 2, gitabMap2 ;ktrig seqtime kspeed, 0, 20, 0, gitabMap2 ; try with seqtime too... schedkwhen ktrig, 0, 0, 3, 0, ktrig ; the duration is got from seqtime2 output! endin instr 2 a1 line 1,p3,0 aout oscili 0.7*a1,500,gisine outs1 aout endin instr 3 a1 line 1,p3,0 aout oscili 0.7*a1,1000,gisine outs2 aout endin </CsInstruments> <CsScore> i1 0 20 ;f0 3600 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tablemix.csd.xml0000644000000000000000000000414012262561504020457 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tablemix.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisinoid ftgen 1, 0, 256, 10, 1, 0, 0, .4 ;sinoid gisaw ftgen 2, 0, 1024, 7, 0, 256, 1 ;saw gimix ftgen 100, 0, 256, 7, 0, 256, 1 ;destination table instr 1 kgain linseg 0, p3*.5, .5, p3*.5, 0 tablemix 100, 0, 256, 1, 0, 1, 2, 0, kgain asig poscil .5, 110, gimix ;mix table 1 & 2 outs asig, asig endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mode.csd.xml0000644000000000000000000001041512262561504017600 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o moogvcf.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1; 2 modes excitator idur init p3 ifreq11 init p4 ifreq12 init p5 iQ11 init p6 iQ12 init p7 iamp init ampdb(p8) ifreq21 init p9 ifreq22 init p10 iQ21 init p11 iQ22 init p12 ; to simulate the shock between the excitator and the resonator ashock mpulse 3,0 aexc1 mode ashock,ifreq11,iQ11 aexc1 = aexc1*iamp aexc2 mode ashock,ifreq12,iQ12 aexc2 = aexc2*iamp aexc = (aexc1+aexc2)/2 ;"Contact" condition : when aexc reaches 0, the excitator looses ;contact with the resonator, and stops "pushing it" aexc limit aexc,0,3*iamp ; 2modes resonator ares1 mode aexc,ifreq21,iQ21 ares2 mode aexc,ifreq22,iQ22 ares = (ares1+ares2)/2 display aexc+ares,p3 outs aexc+ares,aexc+ares endin </CsInstruments> <CsScore> ;wooden excitator against glass resonator i1 0 8 1000 3000 12 8 70 440 888 500 420 ;felt against glass i1 4 8 80 188 8 3 70 440 888 500 420 ;wood against wood i1 8 8 1000 3000 12 8 70 440 630 60 53 ;felt against wood i1 12 8 80 180 8 3 70 440 630 60 53 i1 16 8 1000 3000 12 8 70 440 888 2000 1630 i1 23 8 80 180 8 3 70 440 888 2000 1630 ;With a metallic excitator i1 33 8 1000 1800 1000 720 70 440 882 500 500 i1 37 8 1000 1800 1000 850 70 440 630 60 53 i1 42 8 1000 1800 2000 1720 70 440 442 500 500 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSinfo.csd.xml0000644000000000000000000000230212262561504020153 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ; "fox.ats" is created by atsa inum_partials ATSinfo "fox.ats", 3 print inum_partials endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen42.csd.xml0000644000000000000000000000616212262561504017577 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o GEN42.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktab = 1 ;ftable 1 kurd = urd(ktab) ktrig metro 5 ;triggers 5 times per second kres samphold kurd, ktrig ;sample and hold value of kurd printk2 kres ;print it asig poscil .5, 220+kres, 2 outs asig, asig endin instr 2 seed 0 ;every run new values ktab = 1 ;ftable 1 kurd = urd(ktab) ktrig metro 5 ;triggers 5 times per second kres samphold kurd, ktrig ;sample and hold value of kurd printk2 kres ;print it asig poscil .5, 220+kres, 2 outs asig, asig endin </CsInstruments> <CsScore> f1 0 -20 -42 10 20 .3 100 200 .7 ;30% choose between 10 and 20 and 70% between 100 and 200 f2 0 8192 10 1 ;sine wave i 1 0 5 i 2 6 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/toot5.csd.xml0000644000000000000000000000607212262561504017732 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> instr 5 irel = 0.01 ; set vibrato release time idel1 = p3 * p10 ; calculate initial delay (% of dur) isus = p3 - (idel1 + irel) ; calculate remaining duration iamp = ampdb(p4) iscale = iamp * .333 ; p4=amp inote = cpspch(p5) ; p5=freq k3 linseg 0, idel1, p9, isus, p9, irel, 0 ; p6=attack time k2 oscil k3, p8, 1 ; p7=release time k1 linen iscale, p6, p3, p7 ; p8=vib rate a3 oscil k1, inote*.995+k2, 1 ; p9=vib depth a2 oscil k1, inote*1.005+k2, 1 ; p10=vib delay (0-1) a1 oscil k1, inote+k2, 1 out a1+a2+a3 endin </CsInstruments> <CsScore> f1 0 4096 10 1 ; sine wave ;ins strt dur amp freq atk rel vibrt vbdpt vbdel i5 0 3 86 10.00 0.1 0.7 7 6 .4 i5 4 3 86 10.02 1 0.2 6 6 .4 i5 8 4 86 10.04 2 1 5 6 .4 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/adsynt.csd.xml0000644000000000000000000001414212262561504020157 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o adsynt.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; Generate a sinewave table. giwave ftgen 1, 0, 1024, 10, 1 ; Generate two empty tables for adsynt. gifrqs ftgen 2, 0, 32, 7, 0, 32, 0 ; A table for freqency and amp parameters. giamps ftgen 3, 0, 32, 7, 0, 32, 0 ; Generates parameters at init time instr 1 ; Generate 10 voices. icnt = 10 ; Init loop index. index = 0 ; Loop only executed at init time. loop: ; Define non-harmonic partials. ifreq pow index + 1, 1.5 ; Define amplitudes. iamp = 1 / (index+1) ; Write to tables. tableiw ifreq, index, gifrqs ; Used by adsynt. tableiw iamp, index, giamps index = index + 1 ; Do loop/ if (index < icnt) igoto loop asig adsynt 0.3, 150, giwave, gifrqs, giamps, icnt outs asig, asig endin ; Generates parameters every k-cycle. instr 2 ; Generate 10 voices. icnt = 10 ; Reset loop index. kindex = 0 ; Loop executed every k-cycle. loop: ; Generate lfo for frequencies. kspeed pow kindex + 1, 1.6 ; Individual phase for each voice. kphas phasorbnk kspeed * 0.7, kindex, icnt klfo table kphas, giwave, 1 ; Arbitrary parameter twiddling... kdepth pow 1.4, kindex kfreq pow kindex + 1, 1.5 kfreq = kfreq + klfo*0.006*kdepth ; Write freqs to table for adsynt. tablew kfreq, kindex, gifrqs ; Generate lfo for amplitudes. kspeed pow kindex + 1, 0.8 ; Individual phase for each voice. kphas phasorbnk kspeed*0.13, kindex, icnt, 2 klfo table kphas, giwave, 1 ; Arbitrary parameter twiddling... kamp pow 1 / (kindex + 1), 0.4 kamp = kamp * (0.3+0.35*(klfo+1)) ; Write amps to table for adsynt. tablew kamp, kindex, giamps kindex = kindex + 1 ; Do loop. if (kindex < icnt) kgoto loop asig adsynt 0.25, 150, giwave, gifrqs, giamps, icnt outs asig, asig endin </CsInstruments> <CsScore> ; Play Instrument #1 for 2.5 seconds. i 1 0 2.5 ; Play Instrument #2 for 2.5 seconds. i 2 3 2.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/filevalid.csd.xml0000644000000000000000000000407012262561504020613 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o filevalid.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 Sfile strget p4 ivld filevalid Sfile if ivld>0 then asig diskin2 Sfile, 1 outs asig, asig else printf_i "Audiofile '%s' does not exist!\n", 1, Sfile endif endin </CsInstruments> <CsScore> i 1 0 3 "frox.wav";file does not exist!!! i 1 + 3 "fox.wav" ;but this one certainly does... e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tigoto.csd.xml0000644000000000000000000000634112262561504020164 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = abs(p3) ;make p3 positive even if p3 is negative in score itiv tival i1 = -1 ;assume this is tied note, so keep fase of oscili tigoto slur ;no reinitialisation on tied notes i1 = 0 ;first note, so reset phase aatt line p4, idur, 0 ;primary envelope slur: if itiv==0 kgoto note ;no expression on first and second note aslur linseg 0, idur*.3, p4, idur*.7, 0 ;envelope for slurred note aatt = aatt + aslur note: asig oscili aatt, p5, 1, i1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 4096 10 1 ;sine wave i1 0 -5 .8 451 ;p3 = 5 seconds i1 1.5 -1.5 .1 512 i1 3 2 .7 440 ;3 notes together--> duration = 5 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vpvoc.csd.xml0000644000000000000000000000377412262561504020023 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vpvoc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; analyze "fox.wav" with PVANAL first iend = p4 ktime line 0, p3, iend tablexseg p5, p3, p6 ;morph from table 1 asig vpvoc ktime, 1, "fox.pvx" ;to table 2 outs asig*3, asig*3 endin </CsInstruments> <CsScore> f 1 0 512 9 .5 1 0 f 2 0 512 5 1 60 0.01 390 0.01 62 1 i 1 0 5 2.7 1 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftgenonce.csd.xml0000644000000000000000000000361012262561504020623 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftgenonce.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Use ftgenonce instead of ftgen, ftgentmp, or f statement iHz = p4 isine ftgenonce 0, 0, 1024, 10, 1 aoscili pluck .7, iHz, 100, isine, 1 aadsr adsr 0.015, 0.07, 0.6, 0.3 asig = aoscili * aadsr outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 220 i 1 2 1 261 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsifd.csd.xml0000644000000000000000000000367712262561504020163 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsifd.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "beats.wav", 1, 0, 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; pvsifd analysis fst partials fs1, fsi2, .1, 1,3, 500 ; partial tracking aout resyn fst, 1, 1.5, 500, 1 ; resynthesis (up a 5th) outs aout, aout endin </CsInstruments> <CsScore> ;sine f1 0 4096 10 1 i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/stack.csd.xml0000644000000000000000000000444212262561504017764 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o stack.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 stack 100000 instr 1 a1 oscils 0.7, 220, 0 k1 line 0, p3, 1 push "blah", 123.45, a1, k1 push rnd(k1) k_rnd pop S01, i01, a01, k01 pop printf_i "S01 = '%s', i01 = %g\n", 1, S01, i01 ktrig metro 5.0 printf "k01 = %.3f, k_rnd = %.3f\n", ktrig, k01, k_rnd outs a01, a01 endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/readk.csd.xml0000644000000000000000000000425412262561504017746 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o readk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 0dbfs = 1 ; By Andres Cabrera 2008 instr 1 ; Read a number from the file every 0.5 seconds kfibo readk "fibonacci.txt", 7, 0.5 kpitchclass = 8 + ((kfibo % 12)/100) printk2 kpitchclass kcps = cpspch( kpitchclass ) printk2 kcps a1 oscil 0.5, kcps, 1 out a1 endin </CsInstruments> <CsScore> f 1 0 1024 10 1 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen12.csd.xml0000644000000000000000000001316212262561504017572 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen12.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;example from the Csound Book, page 87 sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = p3 iamp = p4 icarfrq = p5 imodfrq = p6 aenv expseg .01, idur*.1, iamp, idur*.8, iamp*.75, idur*.1, .01 i1 = p7*imodfrq ;p7=modulation index start i2 = p8*imodfrq ;p8=modulation index end adev line i1, idur, i2 ;modulation frequency aindex line p7, idur, p8 ;modulation index ar linseg 1, .1, p9, p3-.2, p10, .1, 1 ; r value envelope: p9-p10 =exp. partial strength parameter start and end amp1 = (aindex*(ar+(1/ar)))/2 afmod oscili amp1, imodfrq, 1 ;FM modulator (sine) atab = (aindex*(ar-(1/ar)))/2 ;index to table alook tablei atab, 37 ;table lookup to GEN12 aamod oscili atab, adev, 2 ;am modulator (cosine) aamod = (exp(alook+aamod))*aenv acar oscili aamod, afmod+icarfrq, 1 ;AFM (carrier) asig balance acar, aenv outs asig, asig endin </CsInstruments> <CsScore> f 1 0 8192 10 1 f 2 0 8192 9 1 1 90 f37 0 1024 -12 40 ;Bessel function-defined from 0 to 40 i 1 0 2 .2 800 800 1 6 .1 2 i 1 + . . 1900 147 8 1 4 .2 i 1 . . . 1100 380 2 9 .5 2 i 1 . 10 . 100 100 11 3 .2 5 s i 1 0 1 .1 200 100 1 6 .1 2 i 1 + . < < < < < < < i 1 + . . < < < < < < i 1 + . . < < < < < < i 1 + . . < < < < < < i 1 + . . < < < < < < i 1 + . . < < < < < < i 1 + 10 .2 800 800 9 1 .9 6 s i 1 0 11 .25 50 51 1 6 .1 2 i 1 1 9 .05 700 401 1 6 .1 2 i 1 2 8 . 900 147 8 1 4 .2 i 1 3 7 . 1100 381 2 9 .5 2 i 1 4 6 . 200 102 11 3 .2 5 i 1 5 6 . 800 803 9 1 .9 6 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/trscale.csd.xml0000644000000000000000000000425712262561504020320 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trscale.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kpitch = p4 ain diskin2 "fox.wav", 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .003, 1, 3, 500 ; partial tracking fscl trscale fst, kpitch ; frequency scale aout tradsyn fscl, 1, 1, 500, 1 ; resynthesis outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 i 1 0 3 1.5 ;up a 5th i 1 3 3 3 ;two octaves higher e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tabmorphi.csd.xml0000644000000000000000000000576312262561504020653 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tabmorphi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 8193, 10, 1 ;sine wave giSquare ftgen 0, 0, 8193, 7, 1, 4096, 1, 0, -1, 4096, -1 ;square wave giTri ftgen 0, 0, 8193, 7, 0, 2048, 1, 4096, -1, 2048, 0 ;triangle wave giSaw ftgen 0, 0, 8193, 7, 1, 8192, -1 ;sawtooth wave, downward slope instr 1 iamp = .7 kindex phasor 440 ;read table value at this index kindex = kindex*8192 ;for all 8192 index points kweightpoint = 0.5 ;set weightpoint ktabnum1 line 0, p3, 3 ;morph through all tables ktabnum2 = 2 ;set to triangle wave ksig tabmorphi kindex, kweightpoint, ktabnum1, ktabnum2, giSine, giSquare, giTri, giSaw ksig = ksig*iamp asig interp ksig ;convert to audio signal outs asig, asig endin </CsInstruments> <CsScore> i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen10.csd.xml0000644000000000000000000000440312262561504017566 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen10.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = .6 kcps = 440 ifn = p4 asig oscil kamp, kcps, ifn outs asig,asig endin </CsInstruments> <CsScore> f1 0 16384 10 1 ; Sine f2 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth f3 0 16384 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square f4 0 16384 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse i 1 0 2 1 i 1 3 2 2 i 1 6 2 3 i 1 9 2 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen41.csd.xml0000644000000000000000000000341412262561504017573 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o GEN41.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 k1 duserrnd 1 printk 0, k1 asig poscil .5, 220*k1, 2 outs asig, asig endin </CsInstruments> <CsScore> f1 0 -20 -41 2 .1 8 .9 ;choose 2 at 10% probability, 8 at 90% f2 0 8192 10 1 i1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/octpch.csd.xml0000644000000000000000000000351712262561504020141 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o octpch.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Convert a pitch-class value into an ; octave-point-decimal value. ipch = 8.09 ioct = octpch(ipch) print ioct endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/reson.csd.xml0000644000000000000000000000414412262561504020004 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o reson.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 asaw vco2 .2, 220 ;sawtooth kcf line 220, p3, 1760 ;vary cut-off frequency from 220 to 1280 Hz kbw = p4 ;vary bandwidth of filter too ares reson asaw, kcf, kbw asig balance ares, asaw outs asig, asig endin </CsInstruments> <CsScore> i 1 0 4 10 ;bandwidth of filter = 10 Hz i 1 + 4 50 ;50 Hz and i 1 + 4 200 ;200 Hz e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsblur.csd.xml0000644000000000000000000000360712262561504020356 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of the audio-signal fftblur pvsblur fftin, p4, 1; blur aout pvsynth fftblur; resynthesis out aout endin </CsInstruments> <CsScore> i 1 0 3 0 i 1 3 3 .1 i 1 6 3 .5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pchoct.csd.xml0000644000000000000000000000351712262561504020141 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pchoct.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Convert an octave-point-decimal value into a ; pitch-class value. ioct = 8.75 ipch = pchoct(ioct) print ipch endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/maxabsaccum.csd.xml0000644000000000000000000000544112262561504021143 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o maxabsaccum.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;saw asig vco2 .2, p4 outs asig, asig gasaw = asig endin instr 2 ;sine aout poscil .3, p4, 1 outs aout, aout gasin = aout endin instr 10 accum init 0 maxabsaccum accum, gasaw + gasin ;saw and sine accumulated accum dcblock2 accum ;get rid of DC outs accum, accum clear accum endin </CsInstruments> <CsScore> f 1 0 4096 10 1 ;sine wave i 1 0 7 330 i 2 3 3 440 i 1 10 7 330 ;same notes but without maxabsaccum, for comparison i 2 13 3 440 i 10 0 6 ;accumulation note stops after 6 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/printks.csd.xml0000644000000000000000000000432412262561504020350 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o printks.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Change a value linearly from 0 to 100, ; over the period defined by p3. kup line 0, p3, 100 ; Change a value linearly from 30 to 10, ; over the period defined by p3. kdown line 30, p3, 10 ; Print the value of kup and kdown, once per second. printks "kup = %f, kdown = %f\\n", 1, kup, kdown endin </CsInstruments> <CsScore> ; Play Instrument #1 for 5 seconds. i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vaset.csd.xml0000644000000000000000000000533412262561504020002 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o avarset.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=1 nchnls=2 instr 1 ; Sine Wave ifreq = (p4 > 15 ? p4 : cpspch(p4)) iamp = ampdb(p5) kenv adsr 0.1, 0.05, .9, 0.2 aout init 0 ksampnum init 0 kcount = 0 iperiod = sr / ifreq i2pi = 3.14159 * 2 loopStart: kphase = (ksampnum % iperiod) / iperiod knewval = sin(kphase * i2pi) vaset knewval, kcount,aout ksampnum = ksampnum + 1 loop_lt kcount, 1, ksmps, loopStart aout = aout * iamp * kenv outs aout, aout endin </CsInstruments> <CsScore> i1 0.0 2 440 80 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wgpluck_brighter.csd.xml0000644000000000000000000000416012262561504022216 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o wgpluck_brighter.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 icps = 220 iamp = 20000 kpick = 0.5 iplk = 0 idamp = 30 ifilt = 10 axcite oscil 1, 1, 1 apluck wgpluck icps, iamp, kpick, iplk, idamp, ifilt, axcite out apluck endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lpreson.csd.xml0000644000000000000000000000542212262561504020340 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpreson.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; works with or without -a option when analyzing "fox.wav" from the manual ;both options sound a little different instr 1 ilen filelen "fox.wav" ; length of soundfile prints "fox.wav = %f seconds\\n",ilen ktime line 0, p3, p4 krmsr,krmso,kerr,kcps lpread ktime,"fox_poles.lpc" krmso = krmso*.00001 ; low volume asig buzz krmso, kcps, int(sr/2/kcps), 1 ; max harmonics without aliasing aout lpreson asig asig clip aout, 0, 1 ; prevents distortion outs asig, asig endin </CsInstruments> <CsScore> ; sine f1 0 4096 10 1 i 1 0 2.8 1 ; first words only i 1 4 2.8 2.8 ; whole sentence e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midichannelaftertouch.csd.xml0000644000000000000000000000563212262561504023221 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midichannelaftertouch.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 midinoteoncps p4, p5 ;puts MIDI key translated to cycles per second into p4, and MIDI velocity into p5 kafter init 127 ;full open midichannelaftertouch kafter printk2 kafter ;display the key value when it changes and when key is pressed kvel = p5/127 ;scale midi velocity to 0-1 kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by ain pluck kenv*kvel, p4, p4, 2, 1 ;velocity value asig moogvcf2 ain, 200+(kafter*40), .5 ;scale value of aftertouch and control filter frequency outs asig, asig ;base freq of filter = 200 Hz endin </CsInstruments> <CsScore> f 0 30 ;runs 30 seconds f 2 0 4096 10 1 i 1 0 2 440 100 ; play these notes from score as well i 1 + 2 1440 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftsr.csd.xml0000644000000000000000000000444612262561504017641 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftsr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 itab = p4 isr = ftsr(itab) prints "sampling-rate of table number %d = %d\n", itab, isr endin </CsInstruments> <CsScore> f 1 0 0 1 "kickroll.wav" 0 0 0 ;stereo file f 2 0 0 1 "ahh.aiff" 0 0 0 ;& different formats f 3 0 0 1 "beats.mp3" 0 0 0 f 4 0 0 1 "beats.ogg" 0 0 0 i 1 0 1 1 i 1 + 1 2 i 1 + 1 3 i 1 + 1 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vpow-2.csd.xml0000644000000000000000000000611612262561504020011 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vpow-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "fox.wav", 1 ;soundfile fsrc pvsanal ain, 1024, 256, 1024, 1 ifn ftgen 0, 0, 1024/2, 2, 0 ;create empty function table for the 513 bins kflag pvsftw fsrc,ifn ;export only amplitudes to table kval line .001, p3, 1 ;start with big distortion, cahnge over note duration to clean sound kbin line p4, p3, p5 ;vary the bins vpow ifn, kval, kbin, 0 ;note that this operation is applied each k-cycle! ;vpow ifn, kval, kbin, 10 ;if you set kdstoffset to 10 it will affect bins 10+(kbin line p4, p3, p5) pvsftr fsrc,ifn ;read modified data back to fsrc aout pvsynth fsrc ;and resynth outs aout*p6, aout*p6 ;adjust volume to compensate endin </CsInstruments> <CsScore> i 1 0 4 100 100 .02 ;first 100 bins are affected i 1 + 4 10 10 .1 ;first 10 bins i 1 + 4 1 400 .05 ;sweep from bin 1 to 400 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mandel.csd.xml0000644000000000000000000001474212262561504020123 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mandel.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;example by Brian Evans sr = 44100 ksmps = 32 nchnls = 2 instr 1 ipitchtable = 1 ; pitch table in score ipitchndx = p5 ; p5=pitch index from table ipitch table ipitchndx, ipitchtable kenv expseg 1.0, 1.0, 1.0, 11.5, .0001 asig pluck ampdb(p4)*kenv, cpspch(ipitch), cpspch(ipitch), 0, 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 32 -2 6.00 6.02 6.04 6.05 6.07 6.09 6.11 ; f1 is a pitch table defining a four octave C major scale starting 7.00 7.02 7.04 7.05 7.07 7.09 7.11 ; on C two octaves below middle C 8.00 8.02 8.04 8.05 8.07 8.09 8.11 9.00 9.02 9.04 9.05 9.07 9.09 9.11 ;ins start dur ampdb(p4) pitchndx(p5) i1 0 12.0 75 3 i1 1.5999 12.0 75 4 i1 3.4000 12.0 75 5 i1 4.2000 12.0 75 6 i1 4.4000 12.0 75 7 i1 4.6000 12.0 75 9 i1 4.8000 12.0 75 10 i1 5.0000 12.0 75 5 i1 5.2000 12.0 75 27 i1 5.4000 12.0 75 5 i1 5.6000 12.0 75 20 i1 6.0000 12.0 75 24 i1 6.2000 12.0 75 2 i1 6.4000 12.0 75 27 i1 6.6000 12.0 75 20 i1 6.8000 12.0 75 15 i1 7.0000 12.0 75 3 i1 7.2000 12.0 75 3 i1 7.4000 12.0 75 23 i1 7.6000 12.0 75 9 i1 7.8000 12.0 75 17 i1 8.0000 12.0 75 18 i1 8.2000 12.0 75 3 i1 8.4000 12.0 75 26 i1 8.6000 12.0 75 15 i1 8.8 12.0 75 2 i1 9 12.0 75 26 i1 9.2 12.0 75 8 i1 9.3999 12.0 75 22 i1 9.5999 12.0 75 22 i1 9.7999 12.0 75 20 i1 9.9999 12.0 75 19 i1 10.399 12.0 75 20 i1 10.799 12.0 75 22 i1 10.999 12.0 75 27 i1 11.199 12.0 75 25 i1 11.399 12.0 75 20 i1 11.599 12.0 75 21 i1 11.799 12.0 75 24 i1 11.999 12.0 75 24 i1 12.199 12.0 75 4 i1 12.399 12.0 75 13 i1 12.599 12.0 75 15 i1 12.799 12.0 75 14 i1 12.999 12.0 75 3 i1 13.199 12.0 75 21 i1 13.399 12.0 75 6 i1 13.599 12.0 75 3 i1 13.799 12.0 75 10 i1 13.999 12.0 75 25 i1 14.199 12.0 75 21 i1 14.399 12.0 75 20 i1 14.599 12.0 75 19 i1 14.799 12.0 75 18 i1 15.199 12.0 75 17 i1 15.599 12.0 75 16 i1 15.999 12.0 75 15 i1 16.599 12.0 75 14 i1 17.199 12.0 75 13 i1 18.399 12.0 75 12 i1 18.599 12.0 75 11 i1 19.199 12.0 75 10 i1 19.799 12.0 75 9 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outipb.csd.xml0000644000000000000000000000331112262561504020153 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out and midi in and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outipb.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 10 nchnls = 2 instr 1 ikey notnum ivel veloc ipb = 10 ;a little out of tune outipb 1, ipb, 0, 127 ;(= pitchbend) midion 1, ikey, ivel ;of external synth endin </CsInstruments> <CsScore> f0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ckgoto.csd.xml0000644000000000000000000000613312262561504020144 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ckgoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Change kval linearly from 0 to 2 over ; the period set by the third p-field. kval line 0, p3, 2 ; If kval is greater than or equal to 1 then play the high note. ; If not then play the low note. ckgoto (kval >= 1), highnote kgoto lownote highnote: kfreq = 880 goto playit lownote: kfreq = 440 goto playit playit: ; Print the values of kval and kfreq. printks "kval = %f, kfreq = %f\\n", 1, kval, kfreq a1 oscil 10000, kfreq, 1 out a1 endin </CsInstruments> <CsScore> ; Table: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLpanel-sensekey2.csd.xml0000644000000000000000000000647412262561504022115 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac ; -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLpanel-sensekey2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 48000 ksmps = 32 nchnls = 1 ; Example by Istvan Varga ; if the FLTK opcodes are commented out, sensekey will read keyboard ; events from stdin FLpanel "", 150, 50, 100, 100, 0, 1 FLlabel 18, 10, 1, 0, 0, 0 FLgroup "Keyboard Input", 150, 50, 0, 0, 0 FLgroupEnd FLpanelEnd FLrun instr 1 ktrig1 init 1 ktrig2 init 1 nxtKey1: k1, k2 sensekey if (k1 != -1 || k2 != 0) then printf "Key code = %02X, state = %d\n", ktrig1, k1, k2 ktrig1 = 3 - ktrig1 kgoto nxtKey1 endif nxtKey2: k3 sensekey if (k3 != -1) then printf "Character = '%c'\n", ktrig2, k3 ktrig2 = 3 - ktrig2 kgoto nxtKey2 endif endin </CsInstruments> <CsScore> i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKWurley.csd.xml0000644000000000000000000000364612262561504020535 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKWurley.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;(FM) Modulator Index One kv3 = p7 asig STKWurley cpspch(p4), 1, 2,kv1, 4, 10, 11, kv3, 1, 30, 128, 75 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 .5 7.00 75 0 20 i 1 + . 8.00 120 0 20 i 1 + 1 6.00 50 120 20 i 1 + 4 8.00 10 10 127 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSreadnz.csd.xml0000644000000000000000000000360512262561504020512 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSreadnz.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 2 kenergy ATSreadnz ktime, "beats.ats", 2 anoise randi kenergy, 500 aout oscili 0.005, 455, 1 aout = aout * anoise outs aout, aout endin </CsInstruments> <CsScore> ; cosine wave f 1 0 16384 11 1 1 i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/osciliktp.csd.xml0000644000000000000000000000404212262561504020654 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o osciliktp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1: osciliktp example instr 1 kphs line 0, p3, 4 a1x osciliktp 220.5, 1, 0 a1y osciliktp 220.5, 1, -kphs a1 = a1x - a1y out a1 * 14000 endin </CsInstruments> <CsScore> ; Table #1: Sawtooth wave f 1 0 3 -2 1 0 -1 ; Play Instrument #1 for four seconds. i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/scantable.csd.xml0000644000000000000000000000506212262561504020612 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o scantable.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gi1 ftgen 1, 0, 128, 7, 0, 64, 1, 64, 0 ; initial position gi2 ftgen 2, 0, 128, -7, 1, 128, 1 ; masses gi3 ftgen 3, 0, 128, -7, 0, 64, 100, 64, 0 ; stiffness gi4 ftgen 4, 0, 128, -7, 1, 128, 1 ; damping gi5 ftgen 5, 0, 128, -7, 0, 128, 0.5 ; initial velocity instr 1 iamp ampmidi .5 ipch cpsmidi kenv madsr .1, .1, .8, .3 asig scantable iamp, ipch, 1, 2, 3, 4, 5 asig dcblock asig outs asig*kenv, asig*kenv endin </CsInstruments> <CsScore> f0 60 ; play for 60 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/valpass.csd.xml0000644000000000000000000000341612262561504020330 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o valpass.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 krvt = 1.5 klpt line p4, p3, p5 imaxlpt = .1 a1 diskin2 "fox.wav", 1 a1 valpass a1, krvt, klpt, imaxlpt a2 valpass a1, krvt, klpt*.5, imaxlpt outs a1, a2 endin </CsInstruments> <CsScore> i 1 0 5 .01 .2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lpfreson.csd.xml0000644000000000000000000000542212262561504020506 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpfreson.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;do not use -a option when analyzing "fox.wav" with lpfreson, ;it needs a filter coefficient type of file instr 1 ilen filelen "fox.wav" ; length of soundfile prints "fox.wav = %f seconds\\n",ilen ktime line 0, p3, p4 krmsr,krmso,kerr,kcps lpread ktime,"fox_nopoles.lpc" krmso = krmso*.00001 ; low volume asig buzz krmso, kcps, int(sr/2/kcps), 1 ; max harmonics without aliasing aout lpfreson asig, 1.2 asig clip aout, 0, 1 ; prevents distortion outs asig, asig endin </CsInstruments> <CsScore> ; sine f1 0 4096 10 1 i 1 0 2.8 1 ; first words only i 1 4 2.8 2.8 ; whole sentence e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSbufread.csd.xml0000644000000000000000000000421112262561504020631 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc for RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSbufread.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" and "fox.ats" are created by atsa ktime line 0, p3, 4 ktime2 line 0, p3, 4 kline expseg 0.001, .3, 1, p3-.3, 1 kline2 expseg 0.001, p3, 3 ATSbufread ktime2, 1, "fox.ats", 20 aout ATScross ktime, 2, "beats.ats", 1, kline, 0.001 * (4 - kline2), 180 outs aout*2, aout*2 endin </CsInstruments> <CsScore> ; sine wave. f 1 0 16384 10 1 i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vpowv.csd.xml0000644000000000000000000000551512262561504020042 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ifn2 = p5 ielements = p6 idstoffset = p7 isrcoffset = p8 kval init 25 vpowv ifn1, ifn2, ielements, idstoffset, isrcoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 16 17 f 2 0 16 -7 1 16 2 i2 0.0 0.2 1 i2 0.2 0.2 2 i1 0.4 0.01 1 2 5 3 8 i2 0.8 0.2 1 i1 1.0 0.01 1 2 5 10 -2 i2 1.2 0.2 1 i1 1.4 0.01 1 2 8 14 0 i2 1.6 0.2 1 i1 1.8 0.01 1 2 8 0 14 i2 2.0 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/metro-2.csd.xml0000644000000000000000000000373212262561504020145 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o metro-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kpch random 1,20 ;produce values at k-rate ktrig metro 10 ;trigger 10 times per second kval samphold kpch, ktrig ;change value whenever ktrig = 1 asig buzz 1, 220, kval, 1;harmonics outs asig, asig endin </CsInstruments> <CsScore> f 1 0 4096 10 1 ; sine i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wgclar.csd.xml0000644000000000000000000000413212262561504020132 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o wgclar.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 330 kstiff = -0.3 iatt = 0.1 idetk = 0.1 kngain init p4 ;vary breath kvibf = 5.735 kvamp = 0.1 asig wgclar .9, kfreq, kstiff, iatt, idetk, kngain, kvibf, kvamp, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave i 1 0 2 0.2 i 1 + 2 0.5 ;more breath e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/jitter2.csd.xml0000644000000000000000000000432412262561504020241 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o jitter2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktotamp init p4 kamp1 init .5 kcps1 init 10 kamp2 init .5 kcps2 init 2 kamp3 init .5 kcps3 init 3 kj2 jitter2 ktotamp, kamp1, kcps1, kamp2, kcps2, kamp3, kcps3 aout pluck 1, 200+kj2, 1000, 0, 1 aout dcblock aout outs aout, aout endin </CsInstruments> <CsScore> i 1 0 15 2 ;a bit jitter i 1 8 15 10 ;some more i 1 16 15 20 ;lots more e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tabmorphak.csd.xml0000644000000000000000000000572712262561504021016 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tabmorphak.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 giSine ftgen 0, 0, 8193, 10, 1 ; sine wave giSquare ftgen 0, 0, 8193, 7, 1, 4096, 1, 0, -1, 4096, -1 ; square wave giTri ftgen 0, 0, 8193, 7, 0, 2048, 1, 4096, -1, 2048, 0 ; triangle wave giSaw ftgen 0, 0, 8193, 7, 1, 8192, -1 ; sawtooth wave, downward slope instr 1 iamp = .7 aindex phasor 110 ; read table value at this index kweightpoint expon 0.001, p3, 1 ; using the weightpoint to morph between two tables exponentially ktabnum1 = p4 ; first wave, it morphs to ktabnum2 = p5 ; the second wave asig tabmorphak aindex, kweightpoint, ktabnum1,ktabnum2, giSine, giSquare, giTri, giSaw asig = asig*iamp outs asig, asig endin </CsInstruments> <CsScore> i1 0 5 0 1 ;from sine to square wave i1 6 5 2 3 ;from triangle to saw e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/active_scale.csd.xml0000644000000000000000000000372312262561504021302 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o atone.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr= 44100 ksmps = 64 nchnls = 1 0dbfs = 1 ;by Victor Lazzarini 2008 instr 1 kscal active 1 kamp port 1/kscal, 0.01 asig oscili kamp, p4, 1 kenv linseg 0, 0.1,1,p3-0.2,1,0.1, 0 out asig*kenv endin </CsInstruments> <CsScore> f1 0 16384 10 1 i1 0 10 440 i1 1 3 220 i1 2 5 350 i1 4 3 700 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gendy-2.csd.xml0000644000000000000000000000454212262561504020125 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kenv expseg 0.01, p3*0.1, 1, p3*0.8, 1, p3*0.1, 0.001 kosc oscil 0.1, 3/p3, 1 seed 20120123 kdis bexprnd kosc knum linseg 3, p3*0.75, 10, p3*0.20, 12, p3*0.05, 5 asig gendy 0.2, kosc*60, 6, 0.7, kdis, 500*kenv, 4800, 0.23, 0.3, 12, knum aflt resonz asig, 1400, 400 aout comb kenv*aflt*0.1, 0.9, 0.1 outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 0 .8 0 0 .3 0 0 0 .1 i1 0 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/genexp.csd.xml0000644000000000000000000000476612262561504020156 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o genexp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aswp linseg 0.01, p3*.5, .49, p3*.5, 0.01 ;index sweep function aindex poscil aswp, 110, 1 ;sound to waveshape atable tablei aindex, p4, 1, .5 ;waveshape index aenv linen 0.8, .01, p3, .02 ;amplitude envelope asig = (atable*aenv)*p5 ;impose envelope and scale asig dcblock2 asig ;get rid of DC outs asig, asig endin </CsInstruments> <CsScore> f 1 0 8192 10 1 ;sine wave f 2 0 8192 "exp" 0 15 0 f 3 0 8192 "exp" 0 3 0 i1 0 3 2 2 i1 + 3 3 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/turnoff.csd.xml0000644000000000000000000000452312262561504020342 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o turnoff.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 k1 expon 440, p3/10,880 ; begin gliss and continue if k1 < sr/2 kgoto contin ; until Nyquist detected turnoff ; then quit contin: a1 oscil 10000, k1, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1: an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for 4 seconds. i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midiprogramchange.csd.xml0000644000000000000000000000462512262561504022342 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M1 -Q1 ;;;realtime audio out and midi in and midi out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midiprogramchange.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;uses external midi device kkey init 0 ;initialize key number kvel init 0 ;initialize velocity midiprogramchange 52 ;choose GM program number 52 = string pad midinoteonkey kkey, kvel ;MIDI note number value printk2 kkey ;display the key value when it changes and when key is pressed midion 1, kkey, kvel ;sent note to external device endin </CsInstruments> <CsScore> f 0 30 ;runs for 30 seconds i 1 0 2 62 ; play these notes from score as well i 1 + 2 65 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvscale.csd.xml0000644000000000000000000000430712262561504020314 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of the audio-signal fftblur pvscale fftin, p4, p5, p6; scale aout pvsynth fftblur; resynthesis out aout endin </CsInstruments> <CsScore> i 1 0 3 1 0 1; original sound i 1 3 3 1.5 0 2; fifth higher without ... i 1 6 3 1.5 1 2; ... and with different ... i 1 9 3 1.5 2 5; ... kinds of formant preservation e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/noteondur2.csd.xml0000644000000000000000000000352612262561504020760 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1;;;RT audio I/O with MIDI in </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 instr 1 ifund notnum ivel veloc idur = 1 ;chord with single key noteondur2 1, ifund, ivel, idur noteondur2 1, ifund+3, ivel, idur noteondur2 1, ifund+7, ivel, idur noteondur2 1, ifund+9, ivel, idur endin </CsInstruments> <CsScore> ; Dummy ftable f 0 60 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsvoc.csd.xml0000644000000000000000000000436312262561504020201 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsvoc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisaw ftgen 1, 0, 2048, 10, 1, 0.5, 0.3, 0.25, 0.2 ;sawtooth-like instr 1 asig in ;get the signal in asyn poscil .6, 150, gisaw ;excitation signal of 150 Hz famp pvsanal asig, 1024, 256, 1024, 1 ;analyse in signal fexc pvsanal asyn, 1024, 256, 1024, 1 ;analyse excitation signal ftps pvsvoc famp, fexc, 1, 1 ;cross it atps pvsynth ftps ;synthesise it outs atps, atps endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLvslidBnk.csd.xml0000644000000000000000000000617512262561504020662 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O </CsOptions> <CsInstruments> sr = 44100 kr = 441 ksmps = 100 nchnls = 1 gitypetable ftgen 0, 0, 8, -2, 1, 1, 3, 3, 5, 5, 7, 7 giouttable ftgen 0, 0, 8, -2, 0, 0.2, 0.3, 0.4, 0.5, 0.6, 0.8, 1 FLpanel "Slider Bank", 400, 400, 50, 50 ;Number of sliders inum = 8 ; Table to store output iouttable = giouttable ; Width of the slider bank in pixels iwidth = 350 ; Height of the slider in pixels iheight = 160 ; Distance of the left edge of the slider ; from the left edge of the panel ix = 30 ; Distance of the top edge of the slider ; from the top edge of the panel iy = 10 ; Table containing fader types itypetable = gitypetable FLvslidBnk "1@2@3@4@5@6@7@8@9@10@11@12@13@14@15@16", 16 , iouttable , iwidth , iheight , ix \ , iy , itypetable FLvslidBnk " ", inum , iouttable , iwidth , iheight , ix \ , iy + 200 , -23 ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 ;Dummy instrument endin </CsInstruments> <CsScore> ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLsetcolor.csd.xml0000644000000000000000000000600512262561504020730 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLsetcolor.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Using the opcode flsetcolor to change from the ; default colours for widgets sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Coloured Sliders", 900, 360, 50, 50 gkfreq, ihandle FLslider "A Red Slider", 200, 5000, -1, 5, -1, 750, 30, 85, 50 ired1 = 255 igreen1 = 0 iblue1 = 0 FLsetColor ired1, igreen1, iblue1, ihandle gkfreq, ihandle FLslider "A Green Slider", 200, 5000, -1, 5, -1, 750, 30, 85, 150 ired1 = 0 igreen1 = 255 iblue1 = 0 FLsetColor ired1, igreen1, iblue1, ihandle gkfreq, ihandle FLslider "A Blue Slider", 200, 5000, -1, 5, -1, 750, 30, 85, 250 ired1 = 0 igreen1 = 0 iblue1 = 255 FLsetColor ired1, igreen1, iblue1, ihandle ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 endin </CsInstruments> <CsScore> ; 'Dummy' score event for 1 hour. f 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mididefault.csd.xml0000644000000000000000000000472412262561504021151 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mididefault.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 mididefault 60, p3 ;ensures that a MIDI-activated instrument will have a positive p3 field midinoteoncps p4, p5 ;puts MIDI key translated to cycles per second into p4, and MIDI velocity into p5 kvel = p5/127 ;scale midi velocity to 0-1 kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by asig pluck kenv*kvel, p4, p4, 2, 1 ;velocity outs asig, asig endin </CsInstruments> <CsScore> f 0 30 ;runs for 30 seconds f 2 0 4096 10 1 i 1 0 2 440 100 ; play these notes from score as well i 1 + 2 1440 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftgen-2.csd.xml0000644000000000000000000000505312262561504020120 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ftgen-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 48000 ksmps = 16 nchnls = 2 ;Example by Jonathan Murphy 2007 0dbfs = 1 instr 1 Sfile = "beats.wav" ilen filelen Sfile ; Find length isr filesr Sfile ; Find sample rate isamps = ilen * isr ; Total number of samples isize init 1 loop: isize = isize * 2 ; Loop until isize is greater than number of samples if (isize < isamps) igoto loop itab ftgen 0, 0, isize, 1, Sfile, 0, 0, 0 print isize print isamps turnoff endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tablefilter.csd.xml0000644000000000000000000001113312262561504021147 0ustar rootroot <CsoundSynthesizer> <CsOptions> </CsOptions> <CsInstruments> sr=44100 ksmps=10 nchnls=1 gifarn init 8 ; initialise integer for Farey Sequence F_8 gires fareyleni gifarn ; calculate length of F_8, returns 23 ; the table length won't be a power of 2 ; (The length of a Farey Sequence with n > 1 is always odd) gilen init gires * -1 gifarey ftgen 200, 0, gilen, "farey", gifarn, 0 ; initialize destiniation table with 0s gifiltered ftgen 0, 0, gilen, 21, 1, 0 ; initialize second destiniation table with 0s gifiltered2 ftgen 0, 0, gilen, 21, 1, 0 ; table filtering opcode: dest. source, mode, threshold ginumpassed tablefilteri gifiltered, gifarey, 1, 6 ; the threshold parameter indicates that denominators whose weights are heavier ; than 6 are not passing through the filter. The weight is calculated using ; Clarence Barlow's function of indigestibility of a number. According to this function, ; higher prime numbers contribute to an increased weight of any natural integer they divide. ; ginumpassed is the number of elements from the source table 'gifarey' ; that have passed the test and which have been copied to the destination table 'gifiltered' ; apply a different filter: ginumpassed2 tablefilteri gifiltered2, gifarey, 2, 5 ; In mode=2 we again test the digestibility of the denominators of the ; fractions in the source table. ; The difference to mode=1 is that we now let pass only vaules from the ; source that are as heavy as the threshold or greater. instr 4 kndx init 0 ; read out elements of now filtered F_8 sequentially and print to file if (kndx < ginumpassed) then kelem tab kndx, gifiltered fprintks "fareyfilter_lp.txt", "%2.6f\\n", kelem kndx = kndx+1 endif endin instr 5 kndx init 0 ; read out elements and print to file if (kndx < ginumpassed2) then kelem tab kndx, gifiltered2 fprintks "fareyfilter_hp.txt", "%2.6f\\n", kelem kndx = kndx+1 endif endin </CsInstruments> <CsScore> i4 0 1 i5 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen09.csd.xml0000644000000000000000000000567412262561504017611 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen09.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gi1 ftgen 1,0,2^10,9, 1,3,0, 3,1,0, 9,0.333,180 ;an approximation of a square wave gi2 ftgen 2,0,2^10,9, 1,3,180, 3,1,0, 9,0.333,0 ;same values as gi1, except some phase values gi3 ftgen 3,0,2^10,9, 1, .4, 0, 2.2, .5, 0, 3.8, 1, 0 ;inharmonic, but does not sound well --> wave does not end at same point as where it begins --> artefacts gi4 ftgen 4,0,2^10,9, 10, .4, 0, 22, .5, 0, 38, 1, 0 ;the same proportions, but value of partial number is multiplied 10 times -->the sound is much clearer, ;because the sudden "jump" like the one in gi3 will pop up only once in 10 repetitions instr 1 kamp = .6 kcps = 220 ifn = p4 asig poscil kamp, kcps*p5, ifn outs asig,asig endin </CsInstruments> <CsScore> i 1 0 2 1 1 ;subtle difference between table 1 and 2 i 1 3 2 2 1 i 1 7 2 3 1 ;big difference between table 3 and 4 i 1 10 2 4 .1 ;p5 has to compensate for the 10 repetitions of gi4 as opposed to gi3 to get the same pitch e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvread.csd.xml0000644000000000000000000000423412262561504020137 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvread.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; analyze "fox.wav" with PVANAL first ibin = p4 ktime line 0, p3, 2.8 kfreq, kamp pvread ktime, "fox.pvx", ibin ;read data from 7th analysis bin. asig poscil kamp, kfreq, 1 ;function 1 is a stored sine outs asig*5, asig*5 ;compensate loss of volume endin </CsInstruments> <CsScore> ;sine wave f1 0 4096 10 1 i 1 0 6 7 i 1 + 6 15 i 1 + 2 25 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen05.csd.xml0000644000000000000000000000452212262561504017574 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen05.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ;choose different tables for GEN05 kcps init 1/p3 ;index over the length of entire note kndx phasor kcps ixmode = 1 ;normalize index data kamp tablei kndx, ifn, ixmode asig poscil kamp, 440, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 8192 10 1 ;sine wave f 2 0 129 5 1 100 0.0001 29 ;short attack f 3 0 129 5 0.00001 87 1 22 .5 20 0.0001 ;long attack i 1 0 2 2 i 1 3 2 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/schedkwhennamed.csd.xml0000644000000000000000000000377512262561504022017 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ; For Non-realtime ouput leave only the line below: ; -o schedkwhennamed.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 48000 ksmps = 16 nchnls = 2 0dbfs = 1 ; Example by Jonathan Murphy 2007 gSinstr2 = "printer" instr 1 ktrig metro 1 if (ktrig == 1) then ;Call instrument "printer" once per second schedkwhennamed ktrig, 0, 1, gSinstr2, 0, 1 endif endin instr printer ktime timeinsts printk2 ktime endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sfinstr.csd.xml0000644000000000000000000000524712262561504020353 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfinstr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf sfload "sf_GMbank.sf2" sfilist isf instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/5000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfinstr ivel, inum, kamp*ivel, kfreq, 194, gisf ;= Strings 2 tighter outs a1, a2 endin </CsInstruments> <CsScore> f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpsxpch_pierce.csd.xml0000644000000000000000000000363412262561504021660 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cpsxpch_pierce.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a Pierce scale centered on middle A. ipch = 2.02 iequal = 12 irepeat = 3 ibase = 261.62561 icps cpsxpch ipch, iequal, irepeat, ibase print icps endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvshift.csd.xml0000644000000000000000000000603712262561504020344 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 ishift = p4; shift amount in Hz ilowest = p5; lowest frequency to be shifted ikeepform = p6; 0=no formant keeping, 1=keep by amps, 2=keep by spectral envelope ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file fshift pvshift fftin, ishift, ilowest, ikeepform; shift frequencies aout pvsynth fshift; resynthesize out aout endin </CsInstruments> <CsScore> i 1 0 2.757 0 0 0; no shift at all i 1 3 2.757 100 0 0; shift all frequencies by 100 Hz i 1 6 2.757 200 0 0; by 200 Hz i 1 9 2.757 200 0 1; keep formants by method 1 i 1 12 2.757 200 0 2; by method 2 i 1 15 2.757 200 1000 0; shift by 200 Hz but just above 1000 Hz i 1 18 2.757 1000 500 0; shift by 1000 Hz above 500 Hz i 1 21 2.757 1000 300 0; above 300 Hz e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vrandi.csd.xml0000644000000000000000000000543312262561504020143 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vrandi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 ;Example by Andres Cabrera gitab ftgen 0, 0, 16, -7, 0, 128, 0 instr 1 krange init p4 kcps init p5 ioffset init p6 ; table krange kcps ielements idstoffset iseed isize ioffset vrandi gitab, krange, kcps, 3, 3, 2, 1, ioffset kfreq1 table 3, gitab kfreq2 table 4, gitab kfreq3 table 5, gitab ;Change the frequency of three oscillators according to the random values aosc1 oscili 4000, kfreq1, 1 aosc2 oscili 2000, kfreq2, 1 aosc3 oscili 4000, kfreq3, 1 outs aosc1+aosc2, aosc3+aosc2 endin </CsInstruments> <CsScore> f 1 0 2048 10 1 ; krange kcps ioffset i 1 0 5 100 1 300 i 1 5 5 5 1 400 i 1 10 5 100 2 1000 i 1 15 5 400 4 1000 i 1 20 5 1000 8 2000 i 1 20 5 300 32 350 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsdisp.csd.xml0000644000000000000000000000307312262561504020346 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsdisp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 asig inch 1 ;a1 soundin "input.wav" ;select a soundifle fsig pvsanal asig, 1024,256, 1024, 1 pvsdisp fsig endin </CsInstruments> <CsScore> i 1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ziw.csd.xml0000644000000000000000000000454412262561504017473 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ziw.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple instrument. instr 1 ; Set zk variable #1 to 64.182. ziw 64.182, 1 endin ; Instrument #2 -- prints out zk variable #1. instr 2 ; Read zk variable #1 at i-rate. i1 zir 1 ; Print out the value of zk variable #1. print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/notnum.csd.xml0000644000000000000000000000365112262561504020200 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M0 -+rtmidi=virtual ;;;realtime audio out with virtual MIDI in </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 massign 1, 1 ; set MIDI channel 1 to play instr 1 instr 1 iNum notnum print iNum ; Convert MIDI note number to Hz iHz = (440.0*exp(log(2.0)*((iNum)-69.0)/12.0)) aosc oscil 0.6, iHz, 1 outs aosc, aosc endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave f 0 60 ;play 60 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSaddnz-2.csd.xml0000644000000000000000000000532712262561504020471 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac -d -m1 </CsOptions> <CsInstruments> ;example by joachim heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr AllTheNoise Sfile = "fox.ats" prints "Resynthesizing with all the noise.\n" iDur ATSinfo Sfile, 7 p3 = iDur ktime line 0, iDur, iDur asig ATSaddnz ktime, Sfile, 25 outs asig, asig ;start next instr event_i "i", "NoiseInBandsOfFive", iDur+1, 1, 0 endin instr NoiseInBandsOfFive Sfile = "fox.ats" prints "Resynthesizing with noise bands %d - %d.\n", p4, p4+5 iDur ATSinfo Sfile, 7 p3 = iDur ktime line 0, iDur, iDur asig ATSaddnz ktime, Sfile, 5, p4 outs asig, asig ;start next instr if p4 < 20 then event_i "i", "NoiseInBandsOfFive", iDur+1, 1, p4+5 endif endin </CsInstruments> <CsScore> i "AllTheNoise" 0 1 e 25 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sumarray.csd.xml0000644000000000000000000000375312262561504020526 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> ;example by joachim heintz seed 0 instr 1 ;create an array with 10 elements kArr[] init 10 ;fill in random numbers and print them out kIndx = 0 until kIndx == 10 do kNum random 0, 10 kArr[kIndx] = kNum printf "kArr[%d] = %10f\n", kIndx+1, kIndx, kNum kIndx += 1 od ;calculate sum of all values and print it out kSum sumarray kArr printf "Sum of all values in kArr = %f\n", kIndx+1, kSum turnoff endin </CsInstruments> <CsScore> i1 0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen01computed.csd.xml0000644000000000000000000000415212262561504021330 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o gen01computed.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 kcps = 1 ifn = 1 ibas = 1 ; Play the audio sample stored in Table #1. a1 loscil kamp, kcps, ifn, ibas out a1 endin </CsInstruments> <CsScore> ; Table #1: an audio file (using GEN01). ; Since our table size is 0, Csound will compute it. f 1 0 0 1 "beats.wav" 0 0 0 ; Play Instrument #1 for 2 seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/changed.csd.xml0000644000000000000000000000315712262561504020252 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o changed.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ksig oscil 2, 0.5, 1 kint = int(ksig) ktrig changed kint printk 0.2, kint printk2 ktrig endin </CsInstruments> <CsScore> f 1 0 1024 10 1 i 1 0 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tabmorph.csd.xml0000644000000000000000000000576112262561504020500 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tabmorph.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 8193, 10, 1 ;sine wave giSquare ftgen 0, 0, 8193, 7, 1, 4096, 1, 0, -1, 4096, -1 ;square wave giTri ftgen 0, 0, 8193, 7, 0, 2048, 1, 4096, -1, 2048, 0 ;triangle wave giSaw ftgen 0, 0, 8193, 7, 1, 8192, -1 ;sawtooth wave, downward slope instr 1 iamp = .7 kindex phasor 440 ;read table value at this index kindex = kindex*8192 ;for all 8192 index points kweightpoint = 0.5 ;set weightpoint ktabnum1 line 0, p3, 3 ;morph through all tables ktabnum2 = 2 ;set to triangle wave ksig tabmorph kindex, kweightpoint, ktabnum1, ktabnum2, giSine, giSquare, giTri, giSaw ksig = ksig*iamp asig interp ksig ;convert to audio signal outs asig, asig endin </CsInstruments> <CsScore> i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hvs2-2.csd.xml0000644000000000000000000001515312262561504017701 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o hvs2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=48000 ksmps=100 nchnls=2 ; Example by James Hearon 2008 ; Edited by Andres Cabrera ginumPointsX init 16 ginumPointsY init 16 ginumParms init 3 ;Generate 9 tables with arbitrary points gitmp ftgen 100, 0, 16, -2, 70, 260, 390, 180, 200, 300, 980, 126, \ 330, 860, 580, 467, 220, 399, 1026, 1500 gitmp ftgen 200, 0, 16, -2, 100, 200, 300, 140, 600, 700, 880, 126, \ 330, 560, 780, 167, 220, 999, 1026, 1500 gitmp ftgen 300, 0, 16, -2, 400, 200, 300, 540, 600, 700, 880, 126, \ 330, 160, 780, 167, 820, 999, 1026, 1500 gitmp ftgen 400, 0, 16, -2, 100, 200, 800, 640, 600, 300, 880, 126, \ 330, 660, 780, 167, 220, 999, 1026, 1500 gitmp ftgen 500, 0, 16, -2, 200, 200, 360, 440, 600, 700, 880, 126, \ 330, 560, 380, 167, 220, 499, 1026, 1500 gitmp ftgen 600, 0, 16, -2, 100, 600, 300, 840, 600, 700, 880, 126, \ 330, 260, 980, 367, 120, 399, 1026, 1500 gitmp ftgen 700, 0, 16, -2, 100, 200, 300, 340, 200, 500, 380, 126, \ 330, 860, 780, 867, 120, 999, 1026, 1500 gitmp ftgen 800, 0, 16, -2, 100, 600, 300, 240, 200, 700, 880, 126, \ 130, 560, 980, 167, 220, 499, 1026, 1500 gitmp ftgen 900, 0, 16, -2, 100, 800, 200, 140, 600, 700, 680, 126, \ 330, 560, 780, 167, 120, 299, 1026, 1500 giOutTab ftgen 5,0,8, -2, 0 giPosTab ftgen 6,0,32, -2, 0,1,2,3,4,5,6,7,8,9,10, 11, 15, 14, 13, 12 giSnapTab ftgen 8,0,64, -2, 1,1,1, 2,0,0, 3,2,0, 2,2,2, \ 5,2,1, 2,3,4, 6,1,7, 0,0,0, 1,3,5, 3,4,4, 1,5,8, 1,1,5, \ 4,3,2, 3,4,5, 7,6,5, 7,8,9 tb0_init giOutTab FLpanel "hsv2",440,100,10,10,0 gk1,ih1 FLslider "X", 0,1, 0, 5, -1, 400,20, 20,10 gk2, ih2 FLslider "Y", 0, 1, 0, 5, -1, 400, 20, 20, 50 FLpanel_end FLpanel "hvsBox",280,280,500,1000,0 ;ihandle FLhvsBox inumlinesX, inumlinesY, iwidth, iheight, ix, iy [, image] gih1 FLhvsBox 16, 16, 250, 250, 10, 1 FLpanel_end FLrun instr 1 FLhvsBoxSetValue gk1, gk2, gih1 hvs2 gk1,gk2, ginumParms, ginumPointsX, ginumPointsY, giOutTab, giPosTab, giSnapTab ;, iConfigTab k0 init 0 k1 init 1 k2 init 2 kspeed init 0 kspeed = int((tb0(k2)) + 1)*.10 kenv oscil 25000, kspeed*16, 10 k1 phasor kspeed ;slow phasor: 200 sec. kpch tableikt k1 * 16, int((tb0(k1)) +1)*100 ;scale phasor * length a1 oscilikt kenv, kpch, int(tb0(k0)) +1000;scale pitch slightly ahp butterlp a1, 2500 outs ahp, ahp endin </CsInstruments> <CsScore> f 10 0 1024 20 5 ;use of windowing function f1000 0 1024 10 .33 .25 .5 f1001 0 1024 10 1 f1002 0 1024 10 .5 .25 .05 f1003 0 1024 10 .05 .10 .3 .5 1 f1004 0 1024 10 1 .5 .25 .125 .625 f1005 0 1024 10 .33 .44 .55 .66 f1006 0 1024 10 1 1 1 1 1 f1007 0 1024 10 .05 .25 .05 .25 .05 1 f0 3600 i1 0 3600 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/trmix.csd.xml0000644000000000000000000000444612262561504020026 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trmix.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "fox.wav", 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .003, 1, 3, 500 ; partial tracking fslo,fshi trsplit fst, 1000 ; split partial tracks at 1000 Hz fscl trscale fshi, 1.3 ; shift the upper tracks fmix trmix fslo,fscl ; mix the shifted and unshifted tracks aout tradsyn fmix, 1, 1, 500, 1 ; resynthesis of tracks outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 ;sine wave i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lineto.csd.xml0000644000000000000000000000403412262561504020146 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lineto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kfreq randh 1000, 20, 2, 1, 2000 ;generates ten random number between 100 and 300 per second kpan randh .5, 1, 2, 1, .5 ;panning between 0 and 1 kp lineto kpan, .5 ;smoothing pan transition aout poscil .4, kfreq, giSine aL, aR pan2 aout, kp outs aL, aR endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/syncloop.csd.xml0000644000000000000000000000451412262561504020525 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o syncloop.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 iolaps = 2 igrsize = 0.01 ifreq = iolaps/igrsize ips = 1/iolaps istr = p4 /* timescale */ ipitch = 1 /* pitchscale */ asig syncloop 1, ifreq, ipitch, igrsize, ips*istr, .3, .75, 1, 2, iolaps outs asig, asig endin </CsInstruments> <CsScore> f1 0 0 1 "beats.wav" 0 0 0 f2 0 8192 20 2 1 i1 0 6 .5 i1 7 6 .15 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/shaker.csd.xml0000644000000000000000000000327312262561504020135 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o shaker.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 instr 1 kfreq line p4, p3, 440 a1 shaker 10000, kfreq, 8, 0.999, 100, 0 out a1 endin </CsInstruments> <CsScore> i 1 0 1 440 i 1 + 1 4000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftlptim.csd.xml0000644000000000000000000000333012262561504020331 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftlptim.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 instr 1 itim = ftlptim(1) print itim aout loscil3 .8, 40, 1 outs aout, aout endin </CsInstruments> <CsScore> f 1 0 0 1 "Church.wav" 0 0 0 ;Csound computes tablesize i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tempo.csd.xml0000644000000000000000000000622112262561504020000 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc -t60 ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tempo.wav -W -t60 ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kval tempoval printk 0.1, kval ; If the fourth p-field is 1, increase the tempo. if (p4 == 1) kgoto speedup kgoto playit speedup: ; Increase the tempo to 150 beats per minute. tempo 150, 60 playit: a1 oscil 10000, 440, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; p4 = plays at a faster tempo (when p4=1). ; Play Instrument #1 at the normal tempo, repeat 3 times. r3 i 1 00.00 00.25 0 i 1 00.25 00.25 0 i 1 00.50 00.25 0 i 1 00.75 00.25 0 s ; Play Instrument #1 at a faster tempo, repeat 3 times. r3 i 1 00.00 00.25 1 i 1 00.25 00.25 0 i 1 00.50 00.25 0 i 1 00.75 00.25 0 s e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/in.csd.xml0000644000000000000000000000362612262561504017270 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -idac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; in.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;1 channel in, two channels out ain1 in ;grab your mic and sing adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time = 20ms aout flanger ain1, adel, .7 fout "in_1.wav", 14, aout, aout ;write to stereo file, outs aout, aout ;16 bits with header endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen28.csd.xml0000644000000000000000000000650112262561504017600 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen28.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 4 ga1 init 0 ga2 init 0 ga3 init 0 ga4 init 0 instr 1 ;uses GEN28 file "move", as found in /manual/examples kx init 0 ky init 0 ktime line 0, 5, 5 ;same time as in table 1 (="move") asig diskin2 "beats.wav", 1, 0, 1 ;sound source is looped a1, a2, a3, a4 space asig, 1, ktime, .1, kx, ky ;use table 1 = GEN28 ar1, ar2, ar3, ar4 spsend ;send to reverb ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 a3 reverb2 ga3, 2.5, .5 a4 reverb2 ga4, 2.5, .5 outq a1, a2, a3, a4 ga1=0 ga2=0 ga3=0 ga4=0 endin </CsInstruments> <CsScore> f1 0 0 28 "move" i1 0 5 ;same time as ktime i 99 0 10 ;keep reverb active e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/serialRead.csd.xml0000644000000000000000000000540712262561504020734 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n ;;;no output ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o serialRead.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 500 ; the default krate can be too fast for the arduino to handle nchnls = 2 0dbfs = 1 instr 1 iPort serialBegin "/dev/cu.usbmodemfa131", 9600 ;connect to the arduino with baudrate = 9600 kGain init 16 ;read our knob value kVal serialRead iPort if (kVal != -1) then kGain = kVal/128 endif aSig in ;get our audio input and get its rms kRms rms aSig*kGain kRms = kRms*kRms*255 ;scale the rms to a good value for the LED and send it out serialWrite iPort, (kRms < 255 ? kRms : 255) ;must be in range: 0-255 endin </CsInstruments> <CsScore> f 1 0 1024 10 1 1 1 1 1 1 i 1 0 200 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/looptseg.csd.xml0000644000000000000000000000513012262561504020506 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o looptseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 1 ; frequency of loop repetition ktrig init 0 ; loop restart trigger (not used) iphase = 0 ; initial phase ktyp line 6,p3,-6 ; explore the useful range of curve types ; loop of filter cutoff values (oct format) ; value curve dur. kcfoct looptseg kfreq, ktrig, iphase,13, ktyp, 1, \ 4, ktyp, 0, \ 11, ktyp, 1, \ 4 asig vco2 0.2,cpsmidinn(48),0 ; a sawtooth asig moogladder asig,cpsoct(kcfoct),rnd(0.6) ; filter sawtooth outs asig, asig endin </CsInstruments> <CsScore> i 1 0 12 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pchmidi.csd.xml0000644000000000000000000000341212262561504020270 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o pchmidi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; This example expects MIDI note inputs on channel 1 i1 pchmidi print i1 endin </CsInstruments> <CsScore> ;Dummy f-table to give time for real-time MIDI events f 0 8000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/array_array_math.csd.xml0000644000000000000000000001251612262561504022205 0ustar rootroot <CsoundSynthesizer> <CsOptions> -n -m128 </CsOptions> <CsInstruments> instr 1 ;create array and fill with numbers 1..10 resp .1..1 kArr1[] fillarray 1, 2, 3, 4, 5, 6, 7, 8, 9, 10 kArr2[] fillarray 1, 2, 3, 5, 8, 13, 21, 34, 55, 89 ;print contents printf "%s", 1, "\nkArr1:\n" kndx = 0 until kndx == lenarray(kArr1) do printf "kArr1[%d] = %f\n", kndx+1, kndx, kArr1[kndx] kndx += 1 od printf "%s", 1, "\nkArr2:\n" kndx = 0 until kndx == lenarray(kArr2) do printf "kArr2[%d] = %f\n", kndx+1, kndx, kArr2[kndx] kndx += 1 od ;add arrays kArr3[] = kArr1 + kArr2 ;print content printf "%s", 1, "\nkArr1 + kArr2:\n" kndx = 0 until kndx == lenarray(kArr3) do printf "kArr3[%d] = %f\n", kndx+1, kndx, kArr3[kndx] kndx += 1 od ;subtract arrays kArr4[] = kArr1 - kArr2 ;print content printf "%s", 1, "\nkArr1 - kArr2:\n" kndx = 0 until kndx == lenarray(kArr4) do printf "kArr4[%d] = %f\n", kndx+1, kndx, kArr4[kndx] kndx += 1 od ;multiply arrays kArr5[] = kArr1 * kArr2 ;print content printf "%s", 1, "\nkArr1 * kArr2:\n" kndx = 0 until kndx == lenarray(kArr5) do printf "kArr5[%d] = %f\n", kndx+1, kndx, kArr5[kndx] kndx += 1 od ;divide arrays kArr6[] = kArr1 / kArr2 ;print content printf "%s", 1, "\nkArr1 / kArr2:\n" kndx = 0 until kndx == lenarray(kArr6) do printf "kArr5[%d] = %f\n", kndx+1, kndx, kArr6[kndx] kndx += 1 od ;turnoff turnoff endin </CsInstruments> <CsScore> i 1 0 .1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vdelayxw.csd.xml0000644000000000000000000000452212262561504020521 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelayxw.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ims = .5 ;maximum delay time in seconds iws = 1024 ;best quality adl = .5 ;delay time asig diskin2 "flute.aiff", .5, 0, 1 ;loop flute.aiff at half speed a2 poscil3 .2, .1, 1 ;make an LFO, 1 cycle per 2 seconds adl = a2 + ims/2 ;offset the LFO so that it is positive aout vdelayxw asig, adl, ims, iws ;use the LFO to control delay time outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsmaska.csd.xml0000644000000000000000000000423012262561504020477 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 ; function table for defining amplitude peaks (from the example of Richard Dobson) giTab ftgen 0, 0, 513, 8, 0, 2, 1, 3, 0, 4, 1, 6, 0, 10, 1, 12, 0, 16, 1, 32, 0, 1, 0, 436, 0 instr 1 imod = p4; degree of midification (0-1) ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file fmask pvsmaska fftin, giTab, imod aout pvsynth fmask; resynthesize out aout endin </CsInstruments> <CsScore> i 1 0 2.757 0 i 1 3 2.757 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vlowres.csd.xml0000644000000000000000000000527112262561504020361 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vlowres.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp init p4 asig vco2 kamp, 110 ;saw wave kfco line 30, p3, 300 ;vary the cutoff frequency from 30 to 300 Hz. kres = 20 ksep = p5 ;different resonance values iord = p6 ;and different number of filters aout vlowres asig, kfco, kres, iord, ksep aclp clip aout, 1, 1 ;avoid distortion outs aclp, aclp endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine s i 1 0 10 .1 5 2 ;compensate volume and i 1 + 10 .1 25 2 ;number of filters = 2 s i 1 0 10 .01 5 6 ;compensate volume and i 1 + 10 .04 15 6 ;number of filters = 6 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/trshift.csd.xml0000644000000000000000000000444412262561504020344 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trshift.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kpsft = p4 ain diskin2 "fox.wav", 1 fs1, fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, 0.003, 1, 3, 500 ; partial tracking fscl trshift fst, kpsft ; frequency shift aout tradsyn fscl, 1, 1, 500, 1 ; resynthesis outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 ;sine i 1 0 3 150 ;adds 150Hz to all tracks i 1 + 3 500 ;adds 500Hz to all tracks e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/chanctrl.csd.xml0000644000000000000000000000445512262561504020461 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out No messages MIDI in -odac -d -M0 ;;;RT audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chanctrl.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; press your midi keyboard and move your midi controller to see result ichnl = 1 ;MIDI note inputs on channel 1 ictlno = 7 ;use midi volume controller kch chanctrl ichnl, 7 ;to control amplitude of oscil printk2 kch asig oscil kch*(1/127), 220, 1 outs asig, asig endin </CsInstruments> <CsScore> ;Dummy f-table to give time for real-time MIDI events f 0 30 ;sine wave. f 1 0 16384 10 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vmult_i.csd.xml0000644000000000000000000000507112262561504020335 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vmult_i ifn1, ival, ielements, idstoffset endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sum.csd.xml0000644000000000000000000000420412262561504017457 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sum.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisine ftgen 0, 0, 2^10, 10, 1 instr 1 a1 oscili 1, 10.0, gisine ;combine 3 sinusses a2 oscili 1, 1.0, gisine ;at different rates a3 oscili 1, 3.0, gisine ares sum a1, a2, a3 ;sum them ares = ares*100 ;scale result and asig poscil .5, ares+110, gisine ;add to frequency outs asig, asig endin </CsInstruments> <CsScore> i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/filenchnls.csd.xml0000644000000000000000000000465712262561504021014 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ;-odac ;;;realtime audio out -iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o filechnls.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; choose between mono or stereo file ilen filelen p4 ;calculate length of soundfile ;print ilen ichn filenchnls p4 ;check number of channels print ichn if (ichn == 1) then ;mono signal asig diskin2 p4, 1 outs asig, asig else ;stereo signal aL, aR diskin2 p4, .5, 0, 1 outs aL, aR endif endin </CsInstruments> <CsScore> i 1 0 3 "fox.wav" ;mono signal i 1 5 2 "kickroll.wav" ;stereo signal e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/butterlp.csd.xml0000644000000000000000000000363212262561504020520 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o butterlp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; White noise signal asig rand 0.5 outs asig, asig endin instr 2 ; filtered noise asig rand 0.7 alp butterlp asig, 1000 ;cutting frequencies above 1 KHz outs alp, alp endin </CsInstruments> <CsScore> i 1 0 2 i 2 2.5 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/chnclear.csd.xml0000644000000000000000000001155012262561504020434 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chnclear.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;Example by Joachim Heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1; send i-values chnset 1, "sio" chnset -1, "non" endin instr 2; send k-values kfreq randomi 100, 300, 1 chnset kfreq, "cntrfreq" kbw = kfreq/10 chnset kbw, "bandw" endin instr 3; send a-values anois rand .1 chnset anois, "noise" loop: idur random .3, 1.5 timout 0, idur, do reinit loop do: ifreq random 400, 1200 iamp random .1, .3 asig oscils iamp, ifreq, 0 aenv transeg 1, idur, -10, 0 asine = asig * aenv chnset asine, "sine" endin instr 11; receive some chn values and send again ival1 chnget "sio" ival2 chnget "non" print ival1, ival2 kcntfreq chnget "cntrfreq" kbandw chnget "bandw" anoise chnget "noise" afilt reson anoise, kcntfreq, kbandw afilt balance afilt, anoise chnset afilt, "filtered" endin instr 12; mix the two audio signals amix1 chnget "sine" amix2 chnget "filtered" chnmix amix1, "mix" chnmix amix2, "mix" endin instr 20; receive and reverb amix chnget "mix" aL, aR freeverb amix, amix, .8, .5 outs aL, aR endin instr 100; clear chnclear "mix" endin </CsInstruments> <CsScore> i 1 0 20 i 2 0 20 i 3 0 20 i 11 0 20 i 12 0 20 i 20 0 20 i 100 0 20 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gendyc.csd.xml0000644000000000000000000000275112262561504020131 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aout gendyc 0.5, 1, 1, 1, 1, 220, 440, 0.5, 0.5 outs aout, aout endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/endin.csd.xml0000644000000000000000000000341612262561504017754 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o endin.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 iamp = 10000 icps = 440 iphs = 0 a1 oscils iamp, icps, iphs out a1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for 2 seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/int.csd.xml0000644000000000000000000000365412262561504017455 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o int.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icount init 0 loop: inum = icount / 3 inm = int(inum) prints "integer (%f/3) = %f\\n", icount, inm loop_lt icount, 1, 10, loop endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/endif.csd.xml0000644000000000000000000000564312262561504017750 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o endif.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Get the note value from the fourth p-field. knote = p4 ; Does the user want a low note? if (knote == 0) then kcps = 220 ; Does the user want a middle note? elseif (knote == 1) then kcps = 440 ; Does the user want a high note? elseif (knote == 2) then kcps = 880 endif ; Create the note. kamp init .8 ifn = 1 a1 oscili kamp, kcps, ifn outs a1, a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; p4: 0=low note, 1=middle note, 2=high note. ; Play Instrument #1 for one second, low note. i 1 0 1 0 ; Play Instrument #1 for one second, middle note. i 1 1 1 1 ; Play Instrument #1 for one second, high note. i 1 2 1 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/signalflowgraph.csd.xml0000644000000000000000000001551412262561504022050 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o madsr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Michael Gogins */ ; Initialize the global variables. sr = 44100 ksmps = 100 nchnls = 2 ; Connect up the instruments to create a signal flow graph. connect "SimpleSine", "leftout", "Reverberator", "leftin" connect "SimpleSine", "rightout", "Reverberator", "rightin" connect "Moogy", "leftout", "Reverberator", "leftin" connect "Moogy", "rightout", "Reverberator", "rightin" connect "Reverberator", "leftout", "Compressor", "leftin" connect "Reverberator", "rightout", "Compressor", "rightin" connect "Compressor", "leftout", "Soundfile", "leftin" connect "Compressor", "rightout", "Soundfile", "rightin" ; Turn on the "effect" units in the signal flow graph. alwayson "Reverberator", 0.91, 12000 alwayson "Compressor" alwayson "Soundfile" instr SimpleSine ihz = cpsmidinn(p4) iamplitude = ampdb(p5) print ihz, iamplitude ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. isine ftgenonce 0, 0, 4096, 10, 1 a1 oscili iamplitude, ihz, isine aenv madsr 0.05, 0.1, 0.5, 0.2 asignal = a1 * aenv ; Stereo audio outlet to be routed in the orchestra header. outleta "leftout", asignal * 0.25 outleta "rightout", asignal * 0.75 endin instr Moogy ihz = cpsmidinn(p4) iamplitude = ampdb(p5) ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. isine ftgenonce 0, 0, 4096, 10, 1 asignal vco iamplitude, ihz, 1, 0.5, isine kfco line 200, p3, 2000 krez init 0.9 asignal moogvcf asignal, kfco, krez, 100000 ; Stereo audio outlet to be routed in the orchestra header. outleta "leftout", asignal * 0.75 outleta "rightout", asignal * 0.25 endin instr Reverberator ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" idelay = p4 icutoff = p5 aleftout, arightout reverbsc aleftin, arightin, idelay, icutoff ; Stereo output. outleta "leftout", aleftout outleta "rightout", arightout endin instr Compressor ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" kthreshold = 25000 icomp1 = 0.5 icomp2 = 0.763 irtime = 0.1 iftime = 0.1 aleftout dam aleftin, kthreshold, icomp1, icomp2, irtime, iftime arightout dam arightin, kthreshold, icomp1, icomp2, irtime, iftime ; Stereo output. outleta "leftout", aleftout outleta "rightout", arightout endin instr Soundfile ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" outs aleftin, arightin endin </CsInstruments> <CsScore> ; Not necessary to activate "effects" or create f-tables in the score! ; Overlapping notes to create new instances of instruments. i "SimpleSine" 1 5 60 85 i "SimpleSine" 2 5 64 80 i "Moogy" 3 5 67 75 i "Moogy" 4 5 71 70 e 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/trandom.csd.xml0000644000000000000000000000437512262561504020330 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trandom.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 ; every run time different values instr 1 kmin init 0 ;random number between 0 and 220 kmax init 220 ktrig = p4 k1 trandom ktrig, kmin, kmax printk2 k1 ;print when k1 changes asig poscil .4, 220+k1, 1 ;if triggered, add random values to frequency outs asig, asig endin </CsInstruments> <CsScore> f1 0 4096 10 1 i 1 0 2 0 ;not triggered i 1 + 2 1 ;triggered e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLbutBank.csd.xml0000644000000000000000000000455212262561504020471 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLbutton.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 nchnls = 1 FLpanel "Button Bank", 520, 140, 100, 100 ;itype = 2 ;Light Buttons itype = 22 ;Plastic Light Buttons inumx = 10 inumy = 4 iwidth = 500 iheight = 120 ix = 10 iy = 10 iopcode = 0 istarttim = 0 idur = 1 gkbutton, ihbb FLbutBank itype, inumx, inumy, iwidth, iheight, ix, iy, iopcode, 1, istarttim, idur FLpanelEnd FLrun instr 1 ibutton = i(gkbutton) prints "Button %i pushed!\\n", ibutton endin </CsInstruments> <CsScore> ; Real-time performance for 1 hour. f 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/strcat.csd.xml0000644000000000000000000000310112262561504020146 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o strcat.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 Sname = "beats" Sname strcat Sname, ".wav" asig soundin Sname outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluidEngine.csd.xml0000644000000000000000000001320712262561504021107 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -F midichn_advanced.mid ;;;reatime audio out and midifile in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fluidEngine.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; LOAD SOUNDFONTS gienginenum1 fluidEngine gienginenum2 fluidEngine isfnum1 fluidLoad "sf_GMbank.sf2", gienginenum1, 1 ; Piano 2, program 1, channel 1 fluidProgramSelect gienginenum1, 1, isfnum1, 0, 1 ; Piano 3, program 2, channel 2 fluidProgramSelect gienginenum1, 2, isfnum1, 0, 2 isfnum2 fluidLoad "19Trumpet.sf2", gienginenum2, 1 ; Trumpet, program 56, channel 3 fluidProgramSelect gienginenum2, 3, isfnum2, 0, 56 ;Look for midifile in folder manual/examples ;"midichn_advanced.mid" sends notes to the soundfonts instr 1 ; GM soundfont ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 1 ikey = p4 ivelocity = p5 fluidNote gienginenum1, ichannel, ikey, ivelocity endin instr 2 ; GM soundfont ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 2 ikey = p4 ivelocity = p5 fluidNote gienginenum1, ichannel, ikey, ivelocity endin instr 3 ; Trumpet ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 3 ikey = p4 ivelocity = p5 fluidNote gienginenum2, ichannel, ikey, ivelocity endin ; COLLECT AUDIO FROM ALL SOUNDFONTS instr 100 ; Fluidsynth output iamplitude1 = 7 iamplitude2 = 7 ; AUDIO aleft1, aright1 fluidOut gienginenum1 aleft2, aright2 fluidOut gienginenum2 outs (aleft1 * iamplitude1) + (aleft2 * iamplitude2), \ (aright1 * iamplitude1) + (aright2 * iamplitude2) endin </CsInstruments> <CsScore> i 1 0 3 60 100 i 2 1 3 60 100 i 3 3 3 63 100 i 100 0 10 ;run for 10 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wterrain.csd.xml0000644000000000000000000000474012262561504020513 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o wterrain.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 kdclk linseg 0, 0.01, 1, p3-0.02, 1, 0.01, 0 kcx line 0.1, p3, 1.9 krx linseg 0.1, p3/2, 0.5, p3/2, 0.1 kpch line cpspch(p4), p3, p5 * cpspch(p4) a1 wterrain 10000, kpch, kcx, kcx, -krx, krx, p6, p7 a1 dcblock a1 out a1*kdclk endin </CsInstruments> <CsScore> f1 0 8192 10 1 0 0.33 0 0.2 0 0.14 0 0.11 f2 0 4096 10 1 i1 0 4 7.00 1 1 1 i1 4 4 6.07 1 1 2 i1 8 8 6.00 1 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fmpercfl.csd.xml0000644000000000000000000000355412262561504020460 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmpercfl.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 220 kc1 = 5 kvdepth = .01 kvrate = 6 kc2 line 5, p3, p4 asig fmpercfl .5, kfreq, kc1, kc2, kvdepth, kvrate outs asig, asig endin </CsInstruments> <CsScore> ; sine wave. f 1 0 32768 10 1 i 1 0 4 5 i 1 5 8 .1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/expsegb.csd.xml0000644000000000000000000000505612262561504020316 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o expseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; p4 = frequency in pitch-class notation. kcps = cpspch(p4) ; Create an amplitude envelope. kenv expsegb 0.01, p3*0.25, 1, p3, 0.01 kamp = kenv * 30000 a1 oscil kamp, kcps, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for a half-second, p4=8.00 i 1 0 0.5 8.00 ; Play Instrument #1 for a half-second, p4=8.01 i 1 1 0.5 8.01 ; Play Instrument #1 for a half-second, p4=8.02 i 1 2 0.5 8.02 ; Play Instrument #1 for a half-second, p4=8.03 i 1 3 0.5 8.03 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/urandom_krate.csd.xml0000644000000000000000000000351012262561504021505 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd31_krate.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create random numbers at k-rate in the range -1 to 1 ; with a uniform distribution. k1 urandom printks "k1=%f\\n", 0.1, k1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/randomh.csd.xml0000644000000000000000000000461512262561504020311 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o randomh.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 seed 0 ; Instrument #1. instr 1 ; Choose a random frequency between 220 and 440 Hz. ; Generate new random numbers at 10 Hz. kmin init 220 kmax init 440 kcps init 10 imode = p4 ifstval = p5 printf_i "\nMode: %d\n", 1, imode k1 randomh kmin, kmax, kcps, imode, ifstval printk2 k1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second, ; each time with a different mode. i 1 0 1 i 1 1 1 2 330 i 1 2 1 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/compilestr.csd.xml0000644000000000000000000000356012262561504021040 0ustar rootroot <CsoundSynthesizer> <CsOptions> -o dac -d </CsOptions> <CsInstruments> sr = 44100 nchnls = 1 ksmps = 32 0dbfs = 1 instr 1 ;will fail because of wrong code ires compilestr {{ instr 2 a1 oscilb p4, p5, 0 out a1 endin }} print ires ; returns -1 because not successfull ;will compile ... ires compilestr {{ instr 2 a1 oscils p4, p5, 0 out a1 endin }} print ires ; ... and returns 0 ;call the new instrument ;(note that the overall performance is extended) scoreline_i "i 2 0 3 .2 415" endin </CsInstruments> <CsScore> i1 0 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tonex.csd.xml0000644000000000000000000000365612262561504020022 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tonex.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 asig diskin2 "beats.wav", 1 outs asig, asig endin instr 2 kton line 10000, p3, 0 ;all the way down to 0 Hz asig diskin2 "beats.wav", 1 asig tonex asig, kton, 8 ;8 filters outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/readfi.csd.xml0000644000000000000000000000267612262561504020120 0ustar rootroot <CsoundSynthesizer> <CsOptions> -n </CsOptions> <CsInstruments> instr 1 Swd pwd printf_i "Working directory is '%s'\n", 1, Swd prints "Reading myself =):\n" read: Sline, iLineNum readfi "readfi.csd" printf_i "Line %d: %s", iLineNum, iLineNum, Sline if iLineNum != -1 igoto read endin </CsInstruments> <CsScore> i1 0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fold.csd.xml0000644000000000000000000000371412262561504017604 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fold.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig poscil3 .8, 400, 1 ;very clean sine kincr line p4, p3, p5 asig fold asig, kincr outs asig, asig endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 4 2 2 i 1 5 4 5 5 i 1 10 4 10 10 i 1 15 4 1 100 ; Vary the fold-over amount from 1 to 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/jacktransport.csd.xml0000644000000000000000000000263712262561504021550 0ustar rootroot <CsoundSynthesizer> <CsOptions> -+rtaudio=JACK -b 64 --sched -o dac:system:playback_ </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 2 instr 1 jacktransport p4, p5 endin instr 2 jacktransport p4 endin </CsInstruments> <CsScore> i2 0 5 1; play i2 5 1 0; stop i1 6 5 1 2 ; move at 2 seconds and start playing back i1 11 1 0 0 ; stop and rewind e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/combinv.csd.xml0000644000000000000000000000332312262561504020311 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o comb.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps expon p5, p3, p4 asig oscil3 0.3, kcps, 1 krvt = 3.5 ilpt = 0.1 aleft combinv asig, krvt, ilpt outs aleft, asig endin </CsInstruments> <CsScore> f1 0 4096 10 1 i 1 0 3 20 2000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLsavesnap.csd.xml0000644000000000000000000001651512262561504020725 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O </CsOptions> <CsInstruments> sr=48000 ksmps=128 nchnls=2 ; Example by Hector Centeno and Andres Cabrera 2007 ; giSWMtab4 ftgen 0, 0, 513, 21, 10, 1, .3 ; giSWMtab4M ftgen 0, 0, 64, 7, 1, 50, 1 FLpanel "Snapshots", 530, 350, 40, 410, 3 FLcolor 100, 118 ,140 FLsetSnapGroup 0 ivalSM1 FLvalue "", 70, 20, 270, 20 ivalSM2 FLvalue "", 70, 20, 270, 60 ivalSM3 FLvalue "", 70, 20, 270, 100 ivalSM4 FLvalue "", 70, 20, 270, 140 gksliderA, gislidSM1 FLslider "Slider A", -4, 4, 0, 3, ivalSM1, 250, 20, 20, 20 gksliderB, gislidSM2 FLslider "Slider B", 1, 10, 0, 3, ivalSM2, 250, 20, 20, 60 gksliderC, gislidSM3 FLslider "Slider C", 0, 1, 0, 3, ivalSM3, 250, 20, 20, 100 gksliderD, gislidSM4 FLslider "Slider D", 0, 1, 0, 3, ivalSM4, 250, 20, 20, 140 itext1 FLbox "store", 1, 1, 14, 50, 25, 355, 15 itext2 FLbox "load", 1, 1, 14, 50, 25, 415, 15 itext3 FLbox "Group 1", 1, 1, 14, 30, 145, 485, 15 gksnap, ibuttn1 FLbutton "1", 1, 0, 11, 25, 25, 364, 45, 0, 3, 0, 3, 1 gksnap, ibuttn2 FLbutton "2", 1, 0, 11, 25, 25, 364, 75, 0, 3, 0, 3, 2 gksnap, ibuttn3 FLbutton "3", 1, 0, 11, 25, 25, 364, 105, 0, 3, 0, 3, 3 gksnap, ibuttn4 FLbutton "4", 1, 0, 11, 25, 25, 364, 135, 0, 3, 0, 3, 4 gkload, ibuttn1 FLbutton "1", 1, 0, 11, 25, 25, 424, 45, 0, 4, 0, 3, 1 gkload, ibuttn2 FLbutton "2", 1, 0, 11, 25, 25, 424, 75, 0, 4, 0, 3, 2 gkload, ibuttn3 FLbutton "3", 1, 0, 11, 25, 25, 424, 105, 0, 4, 0, 3, 3 gkload, ibuttn4 FLbutton "4", 1, 0, 11, 25, 25, 424, 135, 0, 4, 0, 3, 4 FLcolor 100, 140 ,118 FLsetSnapGroup 1 ivalSM5 FLvalue "", 70, 20, 270, 190 ivalSM6 FLvalue "", 70, 20, 270, 230 ivalSM7 FLvalue "", 70, 20, 270, 270 ivalSM8 FLvalue "", 70, 20, 270, 310 gkknobA, gislidSM5 FLknob "Knob A", -4, 4, 0, 3, ivalSM5, 45, 10, 230 gkknobB, gislidSM6 FLknob "Knob B", 1, 10, 0, 3, ivalSM6, 45, 75, 230 gkknobC, gislidSM7 FLknob "Knob C", 0, 1, 0, 3, ivalSM7, 45, 140, 230 gkknobD, gislidSM8 FLknob "Knob D", 0, 1, 0, 3, ivalSM8, 45, 205, 230 itext4 FLbox "store", 1, 1, 14, 50, 25, 355, 185 itext5 FLbox "load", 1, 1, 14, 50, 25, 415, 185 itext6 FLbox "Group 2", 1, 1, 14, 30, 145, 485, 185 gksnap, ibuttn1 FLbutton "5", 1, 0, 11, 25, 25, 364, 215, 0, 3, 0, 3, 5 gksnap, ibuttn2 FLbutton "6", 1, 0, 11, 25, 25, 364, 245, 0, 3, 0, 3, 6 gksnap, ibuttn3 FLbutton "7", 1, 0, 11, 25, 25, 364, 275, 0, 3, 0, 3, 7 gksnap, ibuttn4 FLbutton "8", 1, 0, 11, 25, 25, 364, 305, 0, 3, 0, 3, 8 gkload, ibuttn1 FLbutton "5", 1, 0, 11, 25, 25, 424, 215, 0, 4, 0, 3, 5 gkload, ibuttn2 FLbutton "6", 1, 0, 11, 25, 25, 424, 245, 0, 4, 0, 3, 6 gkload, ibuttn3 FLbutton "7", 1, 0, 11, 25, 25, 424, 275, 0, 4, 0, 3, 7 gkload, ibuttn4 FLbutton "8", 1, 0, 11, 25, 25, 424, 305, 0, 4, 0, 3, 8 FLpanelEnd FLsetVal_i 1, gislidSM1 FLsetVal_i 1, gislidSM2 FLsetVal_i 0, gislidSM3 FLsetVal_i 0, gislidSM4 FLsetVal_i 1, gislidSM5 FLsetVal_i 1, gislidSM6 FLsetVal_i 0, gislidSM7 FLsetVal_i 0, gislidSM8 FLrun instr 1 endin instr 3 ; Save snapshot index init 0 ipstno = p4 igroup = 0 Sfile sprintf "PVCsynth.%d.snap", ipstno if ipstno > 4 then igroup = 1 endif inumsnap, inumval FLsetsnap index , -1, igroup FLsavesnap Sfile endin instr 4 ;Load snapshot index init 0 ipstno = p4 igroup = 0 Sfile sprintf "PVCsynth.%d.snap", ipstno if ipstno > 4 then igroup = 1 endif FLloadsnap Sfile inumload FLgetsnap index , igroup endin </CsInstruments> <CsScore> ;Dummy table for FLgetsnap ; f 1 0 1024 10 1 f 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/locsig_quad.csd.xml0000644000000000000000000000624012262561504021147 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o locsig_quad.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 ga1 init 0 ga2 init 0 ga3 init 0 ga4 init 0 instr 1 krevsend = p4 aout diskin2 "beats.wav", 1, 0, 1 kdegree line 0, p3, 360 ;full circle kdistance line 1, p3 , 1 a1, a2, a3, a4 locsig aout, kdegree, kdistance, krevsend ar1, ar2, ar3, ar4 locsend ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 3.5, .5 a2 reverb2 ga2, 3.5, .5 a3 reverb2 ga3, 3.5, .5 a4 reverb2 ga4, 3.5, .5 outq a1, a2, a3, a4 ga1 = 0 ga2 = 0 ga3 = 0 ga4 = 0 endin </CsInstruments> <CsScore> ; sine wave. f 1 0 16384 10 1 i 1 0 14 .1 ;with reverb i 1 14 14 0 ;no reverb i99 0 36 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/polynomial.csd.xml0000644000000000000000000001611712262561504021044 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o polynomial.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; The polynomial y=x^n where n is odd always produces a curve ; that traverses the range [-1, 1] when the input is within ; the same range. Therefore, we can use one of these curves ; to make a nonlinear phasor that repeatedly reads a table ; from beginning to end like a linear phasor (maintaining ; continuity) but that distorts the waveform in the table. instr 4 ; This instrument demonstrates phase distortion with x^3 idur = p3 iamp = p4 ifreq = p5 itable = p6 aenv linseg 0, .001, 1.0, idur - .051, 1.0, .05, 0 ; declicking envelope aosc phasor ifreq ; create a linear phasor apd polynomial aosc, 0, 0, 0, 1 ; distort the phasor with x^3 aout tablei apd, itable, 1 ; read a sine table with the nonlinear phasor outs aenv*aout*iamp, aenv*aout*iamp endin instr 5 ; This instrument demonstrates phase distortion with x^11 idur = p3 iamp = p4 ifreq = p5 itable = p6 aenv linseg 0, .001, 1.0, idur - .051, 1.0, .05, 0 ; declicking envelope aosc phasor ifreq ; create a linear phasor apd polynomial aosc, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1 ; distort the phasor with x^11 aout tablei apd, itable, 1 ; read a sine table with the nonlinear phasor outs aenv*aout*iamp, aenv*aout*iamp endin instr 6 ; This instrument crossfades between a pure sine and one distorted with x^11 idur = p3 iamp = p4 ifreq = p5 itable = p6 aenv linseg 0, .001, 1.0, idur - .051, 1.0, .05, 0 ; declicking envelope aosc phasor ifreq ; create a linear phasor aout3 tablei aosc, itable, 1 ; read a sine table without the linear phasor apd11 polynomial aosc, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1 ; distort the phasor with x^11 aout11 tablei apd11, itable, 1 ; read a sine table with the nonlinear phasor kamount linseg 1.0, 0.05, 0.9, 1.0, 0.0 ; crossfade between two outputs aout = aout3*kamount + aout11*(1.0 - kamount) outs aenv*aout*iamp, aenv*aout*iamp endin </CsInstruments> <CsScore> f1 0 16385 10 1 ; sine wave ; descending "just blues" scale t 0 100 i4 0 .333 .7 512 1 i. + . . 448 i. + . . 384 i. + . . 360 i. + . . 341.33 i. + . . 298.67 i. + 2 . 256 s t 0 100 i5 0 .333 .7 512 1 i. + . . 448 i. + . . 384 i. + . . 360 i. + . . 341.33 i. + . . 298.67 i. + 2 . 256 s t 0 100 i6 0 .333 .7 512 1 i. + . . 448 i. + . . 384 i. + . . 360 i. + . . 341.33 i. + . . 298.67 i. + 2 . 256 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/butterbp.csd.xml0000644000000000000000000000362512262561504020510 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o butterbp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; White noise signal asig rand 0.6 outs asig, asig endin instr 2 ;filtered noise asig rand 1 abp butterbp asig, 2000, 100 ;passing only 1950 to 2050 Hz outs abp, abp endin </CsInstruments> <CsScore> i 1 0 2 i 2 2.5 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/subinstr.csd.xml0000644000000000000000000000564312262561504020534 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o subinstr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - Creates a basic tone. instr 1 ; Print the value of p4, should be equal to ; Instrument #2's iamp field. print p4 ; Print the value of p5, should be equal to ; Instrument #2's ipitch field. print p5 ; Create a tone. asig oscils p4, p5, 0 out asig endin ; Instrument #2 - Demonstrates the subinstr opcode. instr 2 iamp = 20000 ipitch = 440 ; Use Instrument #1 to create a basic sine-wave tone. ; Its p4 parameter will be set using the iamp variable. ; Its p5 parameter will be set using the ipitch variable. abasic subinstr 1, iamp, ipitch ; Output the basic tone that we have created. out abasic endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #2 for one second. i 2 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/median.csd.xml0000644000000000000000000000202612262561504020110 0ustar rootroot <CsoundSynthesizer> <CsInstruments> sr = 44100 kr = 441 ksmps = 100 nchnls = 1 instr 1 a1 oscil 30000, 10, 1 a2 median a1, 5, 8 out a2 endin </CsInstruments> <CsScore> f1 0 4096 10 1 i 1 0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dumpk.csd.xml0000644000000000000000000000421512262561504017775 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o dumpk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 20 nchnls = 1 ; By Andres Cabrera 2008 instr 1 ; Write fibonacci numbers to file "fibonacci.txt" ; as ascii long integers (mode 7), using the orchestra's ; control rate (iprd = 0) knumber init 0 koldnumber init 1 ktrans init 1 ktrans = knumber knumber = knumber + koldnumber koldnumber = ktrans dumpk knumber, "fibonacci.txt", 7, 0 printk2 knumber endin </CsInstruments> <CsScore> ;Write to the file for 1 second. Since control rate is 20, 20 values will be written i 1 0 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/rezzy.csd.xml0000644000000000000000000000376612262561504020052 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o rezzy.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 asaw vco2 .3, 110 ;sawtooth kcf line 1760, p3, 220 ;vary cut-off frequency from 220 to 1280 Hz kres = p4 ;vary resonance too ares rezzy asaw, kcf, kres asig balance ares, asaw outs asig, asig endin </CsInstruments> <CsScore> i 1 0 4 10 i 1 + 4 30 i 1 + 4 120 ;lots of resonance e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sandpaper.csd.xml0000644000000000000000000000360612262561504020635 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sandpaper.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idmp = p4 a1 line 2, p3, 2 ;preset amplitude increase a2 sandpaper 1, 0.01, 128, idmp ;sandpaper needs a little amp help at these settings asig product a1, a2 ;increase amplitude outs asig, asig endin </CsInstruments> <CsScore> i1 0 1 0.5 i1 + 1 0.95 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tabmap.csd.xml0000644000000000000000000000267612262561504020132 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kS[] init 10 kT[] init 10 kS tabgen 0,10,0.01 kT tabmap kS,"sin" printk2 kS[0] printk2 kS[4] printk2 kS[9] printk2 kT[0] printk2 kT[4] printk2 kT[9] endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/log10.csd.xml0000644000000000000000000000320712262561504017577 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o log10.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = log10(8) print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen07.csd.xml0000644000000000000000000000541312262561504017576 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen07.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;use GEN07 to alter frequency ifn = p4 ;use different GEN07 tables kcps init 10/p3 ;index ftable 10 times over the duration of entire note kndx phasor kcps ixmode = 1 ;normalize index data kfrq tablei kndx, ifn, ixmode kfrq = kfrq*1000 ;scale asig poscil .8, 1220+kfrq, 1 ;add to frequency outs asig, asig endin </CsInstruments> <CsScore> f 1 0 8192 10 1 ;sine wave f 2 0 1024 7 0 512 1 0 -1 512 0 ;sawtooth up and down f 3 0 1024 7 1 512 1 0 -1 512 -1 ;square f 4 0 1024 7 1 1024 -1 ;saw down i 1 0 2 2 i 1 + 2 3 i 1 + 1 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLslider.csd.xml0000644000000000000000000000753012262561504020364 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLslider.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; A sine with oscillator with flslider controlled frequency sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Frequency Slider", 900, 400, 50, 50 ; Minimum value output by the slider imin = 200 ; Maximum value output by the slider imax = 5000 ; Logarithmic type slider selected iexp = -1 ; Slider graphic type (5='nice' slider) itype = 5 ; Display handle (-1=not used) idisp = -1 ; Width of the slider in pixels iwidth = 750 ; Height of the slider in pixels iheight = 30 ; Distance of the left edge of the slider ; from the left edge of the panel ix = 125 ; Distance of the top edge of the slider ; from the top edge of the panel iy = 50 gkfreq, ihandle FLslider "Frequency", imin, imax, iexp, itype, idisp, iwidth, iheight, ix, iy ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun ;Set the widget's initial value FLsetVal_i 300, ihandle instr 1 iamp = 15000 ifn = 1 kfreq portk gkfreq, 0.005 ;Smooth gkfreq to avoid zipper noise asig oscili iamp, kfreq, ifn out asig endin </CsInstruments> <CsScore> ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tab.csd.xml0000644000000000000000000000373512262561504017431 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tab.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifn1 ftgen 1, 0, 0, 1, "flute.aiff", 0, 4, 0 ;deferred-size table instr 1 atab init 0 isize tableng 1 ;length of table? print isize andx phasor 1 / (isize / sr) asig tab andx, 1, 1 ;has a 0 to 1 range outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2.3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKFlute.csd.xml0000644000000000000000000000353312262561504020320 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKFlute.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;jet delay kv4 line 0, p3, 100 ;vibrato depth asig STKFlute cpspch(ifrq), 1, 2, kv1, 4, 100, 11, 100, 1, kv4, 128, 100 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 5 8.00 0 0 i 1 7 3 9.00 20 120 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sfplay.csd.xml0000644000000000000000000000532312262561504020154 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfplay.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 isf sfload "07AcousticGuitar.sf2" sfplist isf sfpassign 0, isf instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/5000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay ivel, inum, kamp*ivel, kfreq, 0 ;preset index = 0 outs a1, a2 endin </CsInstruments> <CsScore> f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpspch.csd.xml0000644000000000000000000000366712262561504020147 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpspch.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Convert pitch-class value into Hz ipch = p4 icps = cpspch(ipch) print icps asig oscil 0.7, icps, 1 outs asig, asig endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 1 8.01 i 1 + 1 8.02 i 1 + 1 8.03 i 1 + .5 5.09 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/envlpx.csd.xml0000644000000000000000000000503512262561504020172 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o envlpx.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 irise = 0.2 idec = 0.5 idur = p3 - idec ifn = 1 iatss = p5 iatdec = 0.01 kenv envlpx .6, irise, idur, idec, ifn, iatss, iatdec kcps = cpspch(p4) asig vco2 kenv, kcps ;apply envlpx to the filter cut-off frequency asig moogvcf asig, kcps + (kenv * 8 * kcps) , .5 ;the higher the pitch, the higher the filter cut-off frequency outs asig, asig endin </CsInstruments> <CsScore> ; a linear rising envelope f 1 0 129 -7 0 128 1 i 1 0 2 7.00 .1 i 1 + 2 7.02 1 i 1 + 2 7.03 2 i 1 + 2 7.05 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATScross.csd.xml0000644000000000000000000000420712262561504020357 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc for RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATScross.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" and "fox.ats" are created by atsa ktime line 0, p3, 4 ktime2 line 0, p3, 4 kline expseg 0.001, .3, 1, p3-.3, 1 kline2 expseg 0.001, p3, 3 ATSbufread ktime2, 1, "fox.ats", 20 aout ATScross ktime, 2, "beats.ats", 1, kline, 0.001 * (4 - kline2), 180 outs aout*2, aout*2 endin </CsInstruments> <CsScore> ; sine wave. f 1 0 16384 10 1 i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cell.csd.xml0000644000000000000000000001236512262561504017601 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ;-odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cell.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; cell.csd by Gleb Rogozinsky sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 0dbfs = 1 ; Cellular automaton-driven synthesis in spectral domain instr 1 iatt = p4 ; envelope attack time isus = p5 ; envelope sustain time irel = p6 ; envelope release time ivol = p7 ; overall volume ; create some white noise asig rand 0.8 ; spectral analysis of asig fsig pvsanal asig, 2048, 1024, 2048, 0 ; get a vector of magnitudes ; calculate cellular automaton state kfreq line 50, 5, 1 ; variable CA triggering frequency ktrig metro kfreq ; trigger the CA to update cells cell ktrig, 0, 3, 1, 2, 2048 ; cells are written into ftable 3 ; use current row of cells in spectral domain fmas pvstencil fsig, 0, 1, 3 ; apply spectral mask aout pvsynth fmas ; reconstruct time signal ; apply envelope and out signal kenv expseg .001, iatt, 1, isus, 1, irel, .001 out aout*kenv*ivol endin </CsInstruments> <CsScore> ; This example uses one-dimensional cellular automaton ; to produce structures in spectral domain ; We have to prepare initial row of cells. ; One alive cell is enough to produce a simple fractal, ; so two alivee cells will make structure more sophisticated f1 0 2048 7 0 150 0 0 1 1 1 0 0 45 0 0 1 1 1 0 0 ; The CA rule is used as follows: ; the states (values) of each cell are summed with their neighboring cells. ; Each sum is used as an index to read a next state of cell ; from the rule table. ; Let us try rule # 129 (LSB binary 1 0 0 0 0 0 0 1). ; This rule will produce a fractal structure for single active cell ; For more rules see http://mathworld.wolfram.com/ElementaryCellularAutomaton.html f2 0 8 -2 1 0 0 0 0 0 0 1 ; Try some different rules i.E. 254 (0 1 1 1 1 1 1 1) or 169 (1 0 0 1 0 1 0 1) ; Prepare the output table of ielements size f3 0 2048 10 0 ; Time to make it sound! i1 0 13 0.3 7 3 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSadd.csd.xml0000644000000000000000000000346012262561504017756 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc for RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSadd.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 2 asig ATSadd ktime, 1, "beats.ats", 1, 20, 0, 2 outs asig*3, asig*3 ;amplify endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outipc.csd.xml0000644000000000000000000000324212262561504020157 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out and midi in and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outipc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 outipc 1, 80, 0, 127 ;program change --> 80 ikey notnum ivel veloc midion 1, ikey, ivel ;play external synth endin </CsInstruments> <CsScore> f0 30 ;runs 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/b.csd.xml0000644000000000000000000000632512262561504017102 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o b.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .4*aenv, 220, 1 outs asig, asig endin instr 2 asig pluck 0.7, p4, 220, 0, 1 outs asig, asig endin instr 3 asig loscil .8, 1, 2, 1 outs asig, asig endin instr 4 asig bamboo .8, 0.01 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave f 2 0 0 1 "fox.wav" 0 0 0 ;sample i1 0 2 i1 10 2 b 5 ; set the clock "forward" i2 1 2 220 ; start time = 6 i2 2 2 110 ; start time = 7 b -1 i3 3 2 ; start time = 2 i3 5.5 1 ; start time = 4.5 b 0 ; reset clock to normal i4 10 2 ; start time = 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fmrhode.csd.xml0000644000000000000000000000453012262561504020301 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmrhode.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 220 kc1 = p4 kc2 = p5 kvdepth = 0.01 kvrate = 3 ifn1 = 1 ifn2 = 1 ifn3 = 1 ifn4 = 2 ivfn = 1 asig fmrhode .5, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn outs asig, asig endin </CsInstruments> <CsScore> ; sine wave. f 1 0 32768 10 1 ; audio file. f 2 0 256 1 "fwavblnk.aiff" 0 0 0 i 1 0 3 6 0 i 1 + . 6 3 i 1 + . 20 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/scogen.csd.xml0000644000000000000000000004461012262561504020136 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages ; -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -n ;Don't write audio ouput to disk </CsOptions> <CsInstruments> ;=========================================================== ; scogen.csd by: Matt Ingalls ; ; a "port" of sorts ; of the old "mills" score generator (scogen) ; ; this instrument creates a schottstaedt.sco file ; to be used with the schottstaedt.orc file ; ; as long as you dont save schottstaedt.orc as a .csd ; file, you should be able to keep it open in MacCsound ; and render each newly generated .sco file. ; ;=========================================================== gScoName = "/Users/matt/Desktop/schottstaedt.sco" ; the name of the file to be generated sr = 100 ; this defines our temporal resolution, ; an sr of 100 means we will generate p2 and p3 values ; to the nearest 1/100th of a second ksmps = 1 ; set kr=sr so we can do everything at k-rate ; some print opcodes opcode PrintInteger, 0, k kval xin fprintks gScoName, "%d", kval endop opcode PrintFloat, 0, k kval xin fprintks gScoName, "%f", kval endop opcode PrintTab, 0, 0 fprintks gScoName, "%n" endop opcode PrintReturn, 0, 0 fprintks gScoName, "%r" endop ; recursively calling opcode to handle all the optional parameters opcode ProcessAdditionalPfields, 0, ikio iPtable, kndx, iNumPfields, iPfield xin ; additional pfields start at 5, we use a default 0 to identify the first call iPfield = (iPfield == 0 ? 5 : iPfield) if (iPfield > iNumPfields) goto endloop ; find our tables iMinTable table 2*iPfield-1, iPtable iMaxTable table 2*iPfield, iPtable ; get values from our tables kMin tablei kndx, iMinTable kMax tablei kndx, iMaxTable ; find a random value in the range and write it to the score fprintks gScoName, "%t%f", kMin + rnd(kMax-kMin) ; recursively call for any additional pfields. ProcessAdditionalPfields iPtable, kndx, iNumPfields, iPfield + 1 endloop: endop /* =========================================================== Generate a gesture of i-statements p2 = start of the gesture p3 = duration of the gesture p4 = number of a function that contains a list of all function table numbers used to define the pfield random distribution p5 = scale generated p4 values according to density (0=off, 1=on) [todo] p6 = let durations overlap gesture duration (0=off, 1=on) [todo] p7 = seed for random number generator seed [todo] =========================================================== */ instr Gesture ; initialize iResolution = 1/sr kNextStart init p2 kCurrentTime init p2 iNumPfields table 0, p4 iInstrMinTable table 1, p4 iInstrMaxTable table 2, p4 iDensityMinTable table 3, p4 iDensityMaxTable table 4, p4 iDurMinTable table 5, p4 iDurMaxTable table 6, p4 iAmpMinTable table 7, p4 iAmpMaxTable table 8, p4 ; check to make sure there is enough data print iNumPfields if iNumPfields < 4 then prints "%dError: At least 4 p-fields (8 functions) need to be specified.%n", iNumPfields turnoff endif ; initial comment fprints gScoName, "%!Generated Gesture from %f to %f seconds%n %!%t%twith a p-max of %d%n%n", p2, p3, iNumPfields ; k-rate stuff if (kCurrentTime >= kNextStart) then ; write a new note! kndx = (kCurrentTime-p2)/p3 ; get the required pfield ranges kInstMin tablei kndx, iInstrMinTable kInstMax tablei kndx, iInstrMaxTable kDensMin tablei kndx, iDensityMinTable kDensMax tablei kndx, iDensityMaxTable kDurMin tablei kndx, iDurMinTable kDurMax tablei kndx, iDurMaxTable kAmpMin tablei kndx, iAmpMinTable kAmpMax tablei kndx, iAmpMaxTable ; find random values for all our required parametrs and print the i-statement fprintks gScoName, "i%d%t%f%t%f%t%f", kInstMin + rnd(kInstMax-kInstMin), kNextStart, kDurMin + rnd(kDurMax-kDurMin), kAmpMin + rnd(kAmpMax-kAmpMin) ; now any additional pfields ProcessAdditionalPfields p4, kndx, iNumPfields PrintReturn ; calculate next starttime kDensity = kDensMin + rnd(kDensMax-kDensMin) if (kDensity < iResolution) then kDensity = iResolution endif kNextStart = kNextStart + kDensity endif kCurrentTime = kCurrentTime + iResolution endin </CsInstruments> <CsScore> /* =========================================================== scogen.sco this csound module generates a score file you specify a gesture of notes by giving the "gesture" instrument a number to a (negative) gen2 table. this table stores numbers to pairs of functions. each function-pair represents a range (min-max) of randomness for every pfield for the notes to be generated. =========================================================== */ ; common tables for pfield ranges f100 0 2 -7 0 2 0 ; static 0 f101 0 2 -7 1 2 1 ; static 1 f102 0 2 -7 0 2 1 ; ramp 0->1 f103 0 2 -7 1 2 0 ; ramp 1->0 f105 0 2 -7 10 2 10 ; static 10 f106 0 2 -7 .1 2 .1 ; static .1 ; specific pfield ranges f10 0 2 -7 .8 2 .01 ; density f11 0 2 -7 8 2 4 ; pitchmin f12 0 2 -7 8 2 12 ; pitchmax ;=== table containing the function numbers used for all the p-field distributions ; ; p1 - table number ; p2 - time table is instantiated ; p3 - size of table (must be >= p5!) ; p4 - gen# (should be = -2) ; p5 - number of pfields of each note to be generated ; p6 - table number of the function representing the minimum possible note number (p1) of a generated note ; p7 - table number of the function representing the maximum possible note number (p1) of a generated note ; p8 - table number of the function representing the minimum possible noteon-to-noteon time (p2 density) of a generated note ; p9 - table number of the function representing the maximum possible noteon-to-noteon time (p2 density) of a generated note ; p10 - table number of the function representing the minimum possible duration (p3) of a generated note ; p11 - table number of the function representing the maximum possible duration (p3) of a generated note ; p12 - table number of the function representing the maximum possible amplitude (p4) of a generated note ; p13 - table number of the function representing the maximum possible amplitude (p5) of a generated note ; p14,p16.. - table number of the function representing the minimum possible value for additional pfields (p5,p6..) of a generated note ; p15,p17.. - table number of the function representing the maximum possible value for additional pfields (p5,p6..) of a generated note ; siz 2 #pds p1min p1max p2min p2max p3min p3max p4min p4max p5min p5max p6min p6max f1 0 32 -2 6 101 101 10 10 101 105 100 106 11 12 100 101 ;gesture definitions ; start dur pTble scale overlap seed i"Gesture" 0 60 1 ;todo-->0 0 123 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vdelay3.csd.xml0000644000000000000000000000422412262561504020224 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelay3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ims = 100 ;maximum delay time in msec aout poscil .8, 220, 1 ;make a signal a2 poscil ims/2, 1/p3, 1 ;make an LFO a2 = a2 + ims/2 ;offset the LFO so that it is positive asig vdelay3 aout, a2, ims ;use the LFO to control delay time outs asig, asig endin </CsInstruments> <CsScore> f1 0 8192 10 1 i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hvs1.csd.xml0000644000000000000000000001072012262561504017534 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O </CsOptions> <CsInstruments> sr=44100 ksmps=100 nchnls=2 0dbfs = 1 ; Example by Gabriel Maldonado and Andres Cabrera ginumLinesX init 16 ginumParms init 3 giOutTab ftgen 5,0,8, -2, 0 giPosTab ftgen 6,0,32, -2, 3,2,1,0,4,5,6,7,8,9,10, 11, 15, 14, 13, 12 giSnapTab ftgen 8,0,64, -2, 1,1,1, 2,0,0, 3,2,0, 2,2,2, 5,2,1, 2,3,4, 6,1,7, 0,0,0, \ 1,3,5, 3,4,4, 1,5,8, 1,1,5, 4,3,2, 3,4,5, 7,6,5, 7,8,9 tb0_init giOutTab FLpanel "hsv1",500,100,10,10,0 gk1,ih1 FLslider "X", 0,1, 0,5, -1, 400,30, 50,20 FLpanel_end FLrun instr 1 ; kx, inumParms, inumPointsX, iOutTab, iPosTab, iSnapTab [, iConfigTab] hvs1 gk1, ginumParms, ginumLinesX, giOutTab, giPosTab, giSnapTab ;, iConfigTab k0 init 0 k1 init 1 k2 init 2 printk2 tb0(k0) printk2 tb0(k1), 10 printk2 tb0(k2), 20 aosc1 oscil tb0(k0)/20, tb0(k1)*100 + 200, 1 aosc2 oscil tb0(k1)/20, tb0(k2)*100 + 200, 1 aosc3 oscil tb0(k2)/20, tb0(k0)*100 + 200, 1 aosc4 oscil tb0(k1)/20, tb0(k0)*100 + 200, 1 aosc5 oscil tb0(k2)/20, tb0(k1)*100 + 200, 1 aosc6 oscil tb0(k0)/20, tb0(k2)*100 + 200, 1 outs aosc1 + aosc2 + aosc3, aosc4 + aosc5 + aosc6 endin </CsInstruments> <CsScore> f1 0 1024 10 1 f0 3600 i1 0 3600 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsinfo.csd.xml0000644000000000000000000000417012262561504020341 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsinfo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; create a PVOC-EX (*.pvx) file with PVANAL first idur filelen "fox.pvx" ;find duration of analysis file kpos line 0,p3,idur ;to ensure we process whole file fsrc pvsfread kpos, "fox.pvx" ;create fsig from (mono) file iovl,inb,iws,ifmt pvsinfo fsrc ;get info print iovl,inb,iws,ifmt ;print info aout pvsynth fsrc outs aout, aout endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/schedkwhen.csd.xml0000644000000000000000000000563312262561504021005 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o schedkwhen.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1 - oscillator with a high note. instr 1 ; Use the fourth p-field as the trigger. ktrigger = p4 kmintim = 0 kmaxnum = 2 kinsnum = 2 kwhen = 0 kdur = 0.5 ; Play Instrument #2 at the same time, if the trigger is set. schedkwhen ktrigger, kmintim, kmaxnum, kinsnum, kwhen, kdur ; Play a high note. a1 oscils 10000, 880, 1 out a1 endin ; Instrument #2 - oscillator with a low note. instr 2 ; Play a low note. a1 oscils 10000, 220, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; p4 = trigger for Instrument #2 (when p4 > 0). ; Play Instrument #1 for half a second, no trigger. i 1 0 0.5 0 ; Play Instrument #1 for half a second, trigger Instrument #2. i 1 1 0.5 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vco2ft.csd.xml0000644000000000000000000001075312262561504020064 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vco2ft.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 10 nchnls = 1 ; user defined waveform -1: trapezoid wave with default parameters (can be ; accessed at ftables starting from 10000) itmp ftgen 1, 0, 16384, 7, 0, 2048, 1, 4096, 1, 4096, -1, 4096, -1, 2048, 0 ift vco2init -1, 10000, 0, 0, 0, 1 ; user defined waveform -2: fixed table size (4096), number of partials ; multiplier is 1.02 (~238 tables) itmp ftgen 2, 0, 16384, 7, 1, 4095, 1, 1, -1, 4095, -1, 1, 0, 8192, 0 ift vco2init -2, ift, 1.02, 4096, 4096, 2 instr 1 kcps expon p4, p3, p5 ; instr 1: basic vco2 example a1 vco2 12000, kcps ; (sawtooth wave with default out a1 ; parameters) endin instr 2 kcps expon p4, p3, p5 ; instr 2: kpw linseg 0.1, p3/2, 0.9, p3/2, 0.1 ; PWM example a1 vco2 10000, kcps, 2, kpw out a1 endin instr 3 kcps expon p4, p3, p5 ; instr 3: vco2 with user a1 vco2 14000, kcps, 14 ; defined waveform (-1) aenv linseg 1, p3 - 0.1, 1, 0.1, 0 ; de-click envelope out a1 * aenv endin instr 4 kcps expon p4, p3, p5 ; instr 4: vco2ft example, kfn vco2ft kcps, -2, 0.25 ; with user defined waveform a1 oscilikt 12000, kcps, kfn ; (-2), and sr/4 bandwidth out a1 endin </CsInstruments> <CsScore> i 1 0 3 20 2000 i 2 4 2 200 400 i 3 7 3 400 20 i 4 11 2 100 200 f 0 14 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/signum.csd.xml0000644000000000000000000000477412262561504020171 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 1 gaArr[] init 2 instr 1 kEnv transeg 1, p3, -3, 0 a_pi = 4 * taninv(1.0); a1 phasor 440; a2 = sin(2 * a_pi * 1/ksmps * a1); a3 dcblock2 a2 asig = signum(a3) gaArr[0] = a2 * 0.6 * kEnv gaArr[1] = asig * 0.6 * kEnv outs gaArr[0], gaArr[1] endin </CsInstruments> <CsScore> i 1 0 3 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/butterhp.csd.xml0000644000000000000000000000361312262561504020513 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o butterhp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; White noise asig rand 0.5 outs asig, asig endin instr 2 ; filtered noise asig rand 0.6 ahp butterhp asig, 500 ;pass frequencies above 500 Hz outs ahp, ahp endin </CsInstruments> <CsScore> i 1 0 2 i 2 2.5 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/metro.csd.xml0000644000000000000000000000206212262561504020001 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac -B441 -b441 </CsOptions> <CsInstruments> sr = 44100 kr = 100 ksmps = 441 nchnls = 2 instr 1 ktrig metro 0.2 printk2 ktrig endin </CsInstruments> <CsScore> i 1 0 20 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/trigger.csd.xml0000644000000000000000000000461212262561504020321 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trigger.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kmtr lfo 1, 1, 1 ;triangle wave kmode = p4 ktr trigger kmtr, .5, kmode printk2 ktr schedkwhen ktr, 0, 3, 2, 0, .3 endin instr 2 aenv linseg 0,p3*.1,1,p3*.3,1,p3*.6,0 ;envelope a1 poscil .3*aenv, 1000, 1 outs a1, a1 endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine i 1 0 3 0 ;down-up i 1 4 3 2 ;down-up & up=down e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midremot.csd.xml0000644000000000000000000001273412262561504020502 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o midremot.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 441 ksmps = 100 nchnls = 2 massign 1,1 massign 2,2 massign 3,3 massign 4,4 massign 5,5 ga1 init 0 ga2 init 0 gi1 sfload "19Trumpet.sf2" gi2 sfload "01hpschd.sf2" gi3 sfload "07AcousticGuitar.sf2" gi4 sfload "22Bassoon.sf2" gitab ftgen 1,0,1024,10,1 midremot "192.168.1.100", "192.168.1.101", 1 midremot "192.168.1.100", "192.168.1.102", 2 midremot "192.168.1.100", "192.168.1.103", 3 midglobal "192.168.1.100", 5 instr 1 sfpassign 0, gi1 ifreq cpsmidi iamp ampmidi 10 inum notnum ivel veloc kamp linsegr 1,1,1,.1,0 kfreq init 1 a1,a2 sfplay ivel,inum,kamp*iamp,kfreq,0,0 outs a1,a2 vincr ga1, a1*.5 vincr ga2, a2*.5 endin instr 2 sfpassign 0, gi2 ifreq cpsmidi iamp ampmidi 15 inum notnum ivel veloc kamp linsegr 1,1,1,.1,0 kfreq init 1 a1,a2 sfplay ivel,inum,kamp*iamp,kfreq,0,0 outs a1,a2 vincr ga1, a1*.4 vincr ga2, a2*.4 endin instr 3 sfpassign 0, gi3 ifreq cpsmidi iamp ampmidi 10 inum notnum ivel veloc kamp linsegr 1,1,1,.1,0 kfreq init 1 a1,a2 sfplay ivel,inum,kamp*iamp,kfreq,0,0 outs a1,a2 vincr ga1, a1*.5 vincr ga2, a2*.5 endin instr 4 sfpassign 0, gi4 ifreq cpsmidi iamp ampmidi 15 inum notnum ivel veloc kamp linsegr 1,1,1,.1,0 kfreq init 1 a1,a2 sfplay ivel,inum,kamp*iamp,kfreq,0,0 outs a1,a2 vincr ga1, a1*.5 vincr ga2, a2*.5 endin instr 5 kamp midic7 1,0,1 denorm ga1 denorm ga2 aL, aR reverbsc ga1, ga2, .9, 16000, sr, 0.5 outs aL, aR ga1 = 0 ga2 = 0 endin </CsInstruments> <CsScore> ; Score f0 160 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cross2.csd.xml0000644000000000000000000000703512262561504020073 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cross2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; after example from Kevin Conder sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;play audio file aout soundin "fox.wav" outs aout, aout endin instr 2 ;cross-synthesize icps = p4 ifn = p5 ; Use the "ahh.aiff" sound and "eee.aiff" ain1 oscil 0.6, p4, ifn ain2 soundin "fox.wav" ; Use the "fox.wav" as modulator isize = 4096 ioverlap = 2 iwin = 3 kbias init 1 aout cross2 ain1, ain2, isize, ioverlap, iwin, kbias outs aout, aout endin </CsInstruments> <CsScore> ;audio files f 1 0 128 1 "ahh.aiff" 0 4 0 f 2 0 128 1 "eee.aiff" 0 4 0 f 3 0 2048 20 2 ;windowing function i 1 0 3 i 2 3 3 50 1 ;"eee.aiff" i 2 + 3 50 2 ;"ahh.aiff" i 2 + 3 100 1 ;"eee.aiff" i 2 + 3 100 2 ;"ahh.aiff" i 2 + 3 250 1 ;"eee.aiff" i 2 + 3 250 2 ;"ahh.aiff" i 2 + 3 20 1 ;"eee.aiff" i 2 + 3 20 2 ;"ahh.aiff" e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsfilter.csd.xml0000644000000000000000000000603112262561504020671 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 giSine ftgen 0, 0, 4096, 10, 1 giBell ftgen 0, 0, 4096, 9, .56, 1, 0, .57, .67, 0, .92, 1.8, 0, .93, 1.8, 0, 1.19, 2.67, 0, 1.7, 1.67, 0, 2, 1.46, 0, 2.74, 1.33, 0, 3, 1.33, 0, 3.76, 1, 0, 4.07, 1.33, 0; bell-like (after Risset) instr 1 ipermut = p4; 1 = change order of soundfiles ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "fox.wav" ain1 soundin Sfile1 kfreq randomi 200, 300, 3 ain2 oscili .2, kfreq, giBell ;ain2 oscili .2, kfreq, giSine; try also this fftin1 pvsanal ain1, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file 1 fftin2 pvsanal ain2, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file 2 if ipermut == 1 then fcross pvsfilter fftin2, fftin1, 1 else fcross pvsfilter fftin1, fftin2, 1 endif aout pvsynth fcross out aout * 20 endin </CsInstruments> <CsScore> i 1 0 2.757 0; frequencies from fox.wav, amplitudes multiplied by amplitudes of giBell i 1 3 2.757 1; frequencies from giBell, amplitudes multiplied by amplitudes of fox.wav e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vexp.csd.xml0000644000000000000000000000507112262561504017640 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vexp ifn1, ival, ielements, idstoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/filebit.csd.xml0000644000000000000000000000340612262561504020274 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o filebit.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the number of channels in the ; audio file "mary.wav". ibits filebit "mary.wav" print ibits endin </CsInstruments> <CsScore> ; Play Instrument #1 for 1 second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midion2.csd.xml0000644000000000000000000000267712262561504020230 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -M0 -Q1 ;;;midi in and midi out </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps line 3, p3, .1 klf lfo 1, kcps, 3 ;use a unipolar square to trigger ktr trigger klf, 1, 1 ;from 3 times to .1 time per sec. midion2 1, 60, 100, ktr endin </CsInstruments> <CsScore> i 1 0 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/rand.csd.xml0000644000000000000000000000466612262561504017613 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o rand.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;same values every time krnd rand 100 printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin instr 2 ;different values every time krnd rand 100, 10 ; seed from system clock printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave. i 1 0 1 i 2 2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/seed.csd.xml0000644000000000000000000000510112262561504017570 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o seed.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;same values every time seed 10 krnd randomh 100, 200, 5 printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin instr 2 ;different values every time - value is derived from system clock seed 0 ; seed from system clock krnd randomh 100, 200, 5 printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave. i 1 0 1 i 2 2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/readclock.csd.xml0000644000000000000000000000502112262561504020600 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o readclock.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Start clock #1. clockon 1 ; Do something that keeps Csound busy. a1 oscili 10000, 440, 1 out a1 ; Stop clock #1. clockoff 1 ; Print the time accumulated in clock #1. i1 readclock 1 print i1 endin </CsInstruments> <CsScore> ; Initialize the function tables. ; Table 1: an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for one second starting at 0:00. i 1 0 1 ; Play Instrument #1 for one second starting at 0:01. i 1 1 1 ; Play Instrument #1 for one second starting at 0:02. i 1 2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/include.csd.xml0000644000000000000000000000344012262561504020277 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o include.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a basic oscillator. instr 1 kamp = 10000 kcps = 440 ifn = 1 a1 oscil kamp, kcps, ifn out a1 endin </CsInstruments> <CsScore> ; Include the file for Table #1. #include "table1.inc" ; Play Instrument #1 for 2 seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/waveset.csd.xml0000644000000000000000000000347112262561504020336 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o waveset.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 instr 1 krep init p4 asig soundin "flute.aiff" aout waveset asig, krep outs aout, aout endin </CsInstruments> <CsScore> i 1 0 3 1 ;no repetitions i 1 + 10 3 ;stretching 3 times i 1 + 14 6 ;6 times e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLknob-2.csd.xml0000644000000000000000000001055212262561504020170 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLknob.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 441 ksmps = 100 nchnls = 1 ;By Andres Cabrera 2007 FLpanel "Knob Types", 330, 230, 50, 50 ; Distance of the left edge of the knob ; from the left edge of the panel ix = 20 ; Distance of the top edge of the knob ; from the top of the panel iy = 20 ;Create boxes that display a widget's value ihandleA FLvalue "A", 60, 20, ix + 130, iy + 110 ihandleB FLvalue "B", 60, 20, ix + 220, iy + 110 ihandleC FLvalue "C", 60, 20, ix + 130, iy + 160 ihandleD FLvalue "D", 60, 20, ix + 220, iy + 160 ; The foru types of FLknobs gkdummy1, ihandle1 FLknob "Type 1", 200, 5000, -1, 1, ihandleA, 70, ix, iy, 90 gkdummy2, ihandle2 FLknob "Type 2", 200, 5000, -1, 2, ihandleB, 70, ix + 100, iy gkdummy3, ihandle3 FLknob "Type 3", 200, 5000, -1, 3, ihandleC, 70, ix + 200, iy gkdummy4, ihandle4 FLknob "Type 4", 200, 5000, -1, 4, ihandleD, 70, ix , iy + 100 ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun ; Set the color of widgets FLsetColor 20, 23, 100, ihandle1 FLsetColor 0, 123, 100, ihandle2 FLsetColor 180, 23, 12, ihandle3 FLsetColor 10, 230, 0, ihandle4 FLsetColor2 200, 230, 0, ihandle1 FLsetColor2 200,0 ,123 , ihandle2 FLsetColor2 180, 180, 100, ihandle3 FLsetColor2 180, 23, 12, ihandle4 ; Set the initial value of the widget FLsetVal_i 300, ihandle1 FLsetVal_i 1000, ihandle2 instr 1 ; Nothing here for now endin </CsInstruments> <CsScore> f 0 3600 ;Dumy table to make csound wait for realtime events e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKPercFlut.csd.xml0000644000000000000000000000406012262561504020761 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKPercFlut.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;Total Modulator Index kv2 line p7, p3, 0 ;Modulator Crossfade asig STKPercFlut cpspch(ifrq), 1, 2, kv1, 4, kv2, 11, 0, 1, 100, 128, 40 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 7 8.05 100 0 100 i 1 3 7 9.03 20 120 0 i 1 3 .5 8.05 20 120 0 i 1 4 .5 9.09 20 120 0 i 1 5 3 9.00 20 120 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluid.csd.xml0000644000000000000000000000366512262561504017770 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Audio out Audio in -odac -iadc -M0 ;;;RT audio I/O and RT midi </CsOptions> <CsInstruments> sr = 44100 ksmps = 100 nchnls = 2 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "07AcousticGuitar.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 0 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 120000 asigl, asigr fluidOut giengine outs asigl * imvol, asigr * imvol endin </CsInstruments> <CsScore> i 1 0 3600 60 100 i 99 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fiopen.csd.xml0000644000000000000000000000314212262561504020133 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fiopen.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gihand fiopen "test1.txt", 0 instr 1 ires random 0, 100 fouti gihand, 0, 1, ires ficlose gihand endin </CsInstruments> <CsScore> i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/0dbfs-1.csd.xml0000644000000000000000000000470312262561504020013 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o 0dbfs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Set the 0dbfs to 1. 0dbfs = 1 ; Instrument #1. instr 1 ; Linearly increase the amplitude value "kamp" from ; -90 to p4 (in dBfs) over the duration defined by p3. kamp line -90, p3, p4 print ampdbfs(p4) ; Generate a basic tone using our amplitude value. a1 oscil ampdbfs(kamp), 440, 1 ; Since 0dbfs = 1 we don't need to multiply the output out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 -6 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/space_stereo.csd.xml0000644000000000000000000000560012262561504021330 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o space_stereo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 ga1 init 0 ga2 init 0 instr 1 kx = p4 ky = p5 asig diskin2 "beats.wav", 1 a1, a2, a3, a4 space asig, 0, 0, .1, kx, ky ;take position values from p4, p5 ar1, ar2, ar3, ar4 spsend ;send to reverb ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a2 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 outs a1, a2 ga1=0 ga2=0 endin </CsInstruments> <CsScore> ;place the sound in the left speaker and near i1 0 1 -1 1 ;place the sound in the right speaker and far i1 1 1 45 45 ;place the sound equally between left and right and in the middle ground distance i1 2 1 0 12 i 99 0 7 ;keep reverb active e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/trsplit.csd.xml0000644000000000000000000000410212262561504020351 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trsplit.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "beats.wav", 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .003, 1, 3, 500 ; partial tracking fslo,fshi trsplit fst, 1500 ; split partial tracks at 1500 Hz aout tradsyn fshi, 1, 1, 500, 1 ; resynthesis of tracks above 1500Hz outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 ;sine i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pset-midi.csd.xml0000644000000000000000000000475412262561504020560 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 --midi-key-oct=4 --midi-velocity=5 ;;;realtime audio out and virtual midi keyboard ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pset-midi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 pset 0, 0, 3600, 0, 0, 0 iinstrument = p1 istarttime = p2 iattack = 0.005 isustain = p3 irelease = 0.06 p3 = isustain + iattack + irelease kdamping linsegr 0.0, iattack, 1.0, isustain, 1.0, irelease, 0.0 ioctave = p4 ifrequency = cpsoct(ioctave) iamplitude = p5*.15 ;lower volume print p1, p2, p3, p4, p5 asig STKBandedWG ifrequency, iamplitude outs asig, asig endin </CsInstruments> <CsScore> f 0 60 ; runs 69 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluidAllOut.csd.xml0000644000000000000000000000630112262561504021077 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidAllOut.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine1 fluidEngine isfnum1 fluidLoad "sf_GMbank.sf2", giengine1, 1 fluidProgramSelect giengine1, 1, isfnum1, 0, 0 giengine2 fluidEngine ; soundfont path to manual/examples isfnum2 fluidLoad "22Bassoon.sf2", giengine2, 1 fluidProgramSelect giengine2, 1, isfnum2, 0, 70 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine1, 1, ikey, ivel endin instr 2 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine2, 1, ikey, ivel endin instr 100 imvol init 7 ;amplify a bit asigl, asigr fluidAllOut outs asigl*imvol, asigr*imvol endin </CsInstruments> <CsScore> i 1 0 2 60 127 ;play one note on instr 1 i 2 2 2 60 127 ;play another note on instr 2 and... i 100 0 60 ;play virtual midi keyboard e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/JackoInfo.csd.xml0000644000000000000000000000174712262561504020527 0ustar rootroot <CsoundSynthesizer> <CsOptions> -n </CsOptions> <CsInstruments> sr = 48000 ksmps = 128 nchnls = 2 0dbfs = 1 instr 1 JackoInit "default", "csound" JackoInfo endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen14.csd.xml0000644000000000000000000001014312262561504017570 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen14.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;after the example from The Csound Book, page 83 instr 1 ;compare results from GEN13 & GEN14 iwshpfn = p6 inrmfn = p7 aswp linseg 0.01, p3*.5, .49, p3*.5, 0.01 ;index sweep function aindex poscil aswp, p5, 2 ;sound to waveshape atable tablei aindex, iwshpfn, 1, .5 ;waveshape index anrm tablei aswp*2, inrmfn, 1 ;normalization aenv linen p4, .01, p3, .02 ;amplitude envelope asig = (atable*anrm)*aenv ;normalize and impose envelope asig dcblock2 asig ;get rid of DC outs asig, asig endin </CsInstruments> <CsScore> f 2 0 8192 10 1 ;sine wave f 28 0 4097 13 1 1 1 0 .8 0 .5 0 .2 ;waveshaping function: GEN13 - odd harmonics f 280 0 2049 4 28 1 ;normalization function for f28 f 29 0 4097 14 1 1 1 0 .8 0 .5 0 .2 ;waveshaping function: GEN14 - same harmonics f 290 0 2049 4 29 1 ;normalization function for f29 f 30 0 4097 13 1 1 0 1 0 .6 0 .4 0 .1 ;waveshaping function: GEN13 - even harmonics f 301 0 2049 4 30 1 ;normalization function for f30 f 31 0 4097 14 1 1 0 1 0 .6 0 .4 0 .1 ;waveshaping function: GEN13 - even harmonics f 310 0 2049 4 31 1 ;normalization function for 31 s i1 0 3 .7 440 28 280 i1 4 . .7 . 29 290 i1 8 . .7 . 30 301 i1 12 3 .7 . 31 310 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vbapg.csd.xml0000644000000000000000000000436112262561504017756 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbap4.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 vbaplsinit 2.01, 4, 0, 90, 180, 270 instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav kaz line 0, p3, p4 ;come from right rear speaker & k1,k2,k3,k4 vbapg 180, 100, kaz, 1 ;change spread of soundsource printks "spread of source = %d\n", 1, kaz ;print spread value outq asig*k1,asig*k2,asig*k3,asig*k4 endin </CsInstruments> <CsScore> i 1 0 12 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wgflute.csd.xml0000644000000000000000000000407212262561504020333 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o wgflute.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 440 kjet init p4 ;vary air jet iatt = 0.1 idetk = 0.1 kngain = 0.15 kvibf = 5.925 kvamp = 0.05 asig wgflute .8, kfreq, kjet, iatt, idetk, kngain, kvibf, kvamp, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave i 1 0 2 0.02 ;more air jet i 1 + 2 0.32 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/nchnls_i.csd.xml0000644000000000000000000000530712262561504020455 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -idac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; nchnls_i.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 ;2 channels out 0dbfs = 1 nchnls_i = 4 ;4 channels in instr 1 ;4 channels in, two channels out ain1, ain2, ain3, ain4 inq ;grab your mics and sing adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time = 20ms adel2 linseg 0.02, p3*.5, 0, p3*.5, 0.02 ;max delay time = 20ms aoutl flanger ain1, adel, .7 aoutr flanger ain2, adel*2, .8 aoutla flanger ain3, adel2, .9 aoutra flanger ain4, adel2*2, .5 ;write to quad file, 16 bits with header fout "in_4.wav", 14, aoutl, aoutr, aoutla, aoutra outs (aoutl+aoutla)*.5, (aoutr+aoutra)*.5 ;stereo out endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen09square.csd.xml0000644000000000000000000000376412262561504021030 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o gen09square.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 kcps = 440 ifn = 1 ; Play the waveform stored in Table #1. a1 oscil kamp, kcps, ifn out a1 endin </CsInstruments> <CsScore> ; Table #1: an approximation of a square wave (using GEN09). f 1 0 16384 9 1 3 0 3 1 0 9 0.3333 180 ; Play Instrument #1 for 2 seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen30.csd.xml0000644000000000000000000001262412262561504017574 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac ;;;realtime audio out ; For Non-realtime ouput leave only the line below: ; -o gen30.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;a simplified example of Istvan Varga sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 isaw ftgen 1, 0, 16384, 7, 1, 16384, -1 ;sawtooth wave iFM ftgen 3, 0, 4096, 7, 0, 512, 0.25, 512, 1, 512, 0.25, 512, \ 0, 512, -0.25, 512, -1, 512, -0.25, 512, 0 ;FM waveform iAM ftgen 4, 0, 4096, 5, 1, 4096, 0.01 ;AM waveform iEQ ftgen 5, 0, 1024, 5, 1, 512, 32, 512, 1 ;FM to EQ isine ftgen 6, 0, 1024, 10, 1 ;sine wave /* generate bandlimited sawtooth waves */ i0 = 0 loop1: imaxh = sr / (2 * 440.0 * exp(log(2.0) * (i0 - 69) / 12)) i_ ftgen i0 + 10, 0, 4096, -30, 1, 1, imaxh ;use gen 30 i0 = i0 + 1 if (i0 < 127.5) igoto loop1 instr 1 kcps = 440.0 * exp(log(2.0) * (p4 - 69) / 12) ;note frequency klpmaxf limit p5 * kcps, 1000.0, 12000.0 ;lowpass max. frequency kfmd1 = 0.03 * kcps ;FM depth in Hz kamfr = kcps * 0.02 ;AM frequency kamfr2 = kcps * 0.1 kfnum = (10 + 69 + 0.5 + 12 * log(kcps / 440.0) / log(2.0)) ;table number aenv linseg 0, p3*0.25, 1, p3*0.75, 0 ;amp. envelope asig oscbnk kcps, 0.0, kfmd1, 0.0, 40, 200, 0.1, 0.2, 0, 0, 144, \ 0.0, klpmaxf, 0.0, 0.0, 1.5, 1.5, 2, kfnum, 3, 0, 5, 5, 5 asig = asig * aenv*.03 outs asig, asig endin </CsInstruments> <CsScore> s i 1 0 6 41 10 i 1 0 6 60 i 1 0 6 65 i 1 0 6 69 s i 1 0 6 41 64 i 1 0 6 60 i 1 0 6 65 i 1 0 6 69 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/minarray.csd.xml0000644000000000000000000000403212262561504020474 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> ;example by joachim heintz seed 0 instr 1 ;create an array with 10 elements kArr[] init 10 ;fill in random numbers and print them out kIndx = 0 until kIndx == 10 do kNum random -100, 100 kArr[kIndx] = kNum printf "kArr[%d] = %10f\n", kIndx+1, kIndx, kNum kIndx += 1 od ;investigate minimum number and print it out kMin, kMinIndx minarray kArr printf "Minimum of kArr = %f at index %d\n", kIndx+1, kMin, kMinIndx turnoff endin </CsInstruments> <CsScore> i1 0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/harmon3.csd.xml0000644000000000000000000000411512262561504020223 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o harmon3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ilow = p4 ;lowest value to harmonize aout diskin2 "fox.wav", 1, 0, 1 koct, kamp pitch aout, .01, 6, 10, 10 ;track pitch asig harmon3 aout, koct, .9, 1.5, 0.7, 0, ilow outs (asig + aout)*.4, (asig + aout)*.4 ;mix dry&wet signal endin </CsInstruments> <CsScore> i1 0 2.7 8.8 i1 3 2.7 8.2 i1 6 2.7 7.0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen11.csd.xml0000644000000000000000000000375212262561504017575 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen11.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 asig oscil .8, 220, ifn outs asig,asig endin </CsInstruments> <CsScore> f 1 0 16384 11 1 1 ;number of harmonics = 1 f 2 0 16384 11 10 1 .7 ;number of harmonics = 10 f 3 0 16384 11 10 5 2 ;number of harmonics = 10, 5th harmonic is amplified 2 times i 1 0 2 1 i 1 + 2 2 i 1 + 2 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/phasorbnk.csd.xml0000644000000000000000000000646312262561504020653 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o phasorbnk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Generate a sinewave table. giwave ftgen 1, 0, 1024, 10, 1 ; Instrument #1 instr 1 ; Generate 10 voices. icnt = 10 ; Empty the output buffer. asum = 0 ; Reset the loop index. kindex = 0 ; This loop is executed every k-cycle. loop: ; Generate non-harmonic partials. kcps = (kindex+1)*100+30 ; Get the phase for each voice. aphas phasorbnk kcps, kindex, icnt ; Read the wave from the table. asig table aphas, giwave, 1 ; Accumulate the audio output. asum = asum + asig ; Increment the index. kindex = kindex + 1 ; Perform the loop until the index (kindex) reaches ; the counter value (icnt). if (kindex < icnt) kgoto loop out asum*3000 endin </CsInstruments> <CsScore> ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/q.csd.xml0000644000000000000000000000454012262561504017116 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o q.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .8*aenv, p4, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1;sine wave s q 1 6 0 ;mute at 6 seconds in this section i 1 0 2 110 i 1 3 2 220 i 1 6 2 440 i 1 9 2 880 s q 1 6 1 ;unmute again at 6 seconds in this section i 1 0 2 110 i 1 3 2 220 i 1 6 2 440 i 1 9 2 880 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dcblock.csd.xml0000644000000000000000000000410612262561504020255 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o dcblock.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ;add DC to "beats.wav" asig soundin "beats.wav" asig = asig+5000 ;adds DC of 5000 outs asig, asig endin instr 2 ;dcblock audio asig soundin "beats.wav" asig = asig+5000 ;adds DC adc dcblock asig ;remove DC again outs adc, adc endin </CsInstruments> <CsScore> i 1 0 2 i 2 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pitch.csd.xml0000644000000000000000000000564712262561504017776 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pitch.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;clean audio asig soundin p4 outs asig, asig endin instr 2 ;use pitch iupdte = 0.001 ;high definition ilo = 6 ihi = 10 idbthresh = 10 ifrqs = 12 iconf = 10 istrt = 8 asig soundin p4 koct, kamp pitch asig, iupdte, ilo, ihi, idbthresh, ifrqs, iconf, istrt kamp = kamp*.00005 ;lower volume kcps = cpsoct(koct) asig poscil kamp, kcps, 1 ;re-synthesize with sawtooth outs asig, asig endin </CsInstruments> <CsScore> f1 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ;sawtooth i 1 0 3 "fox.wav" i 2 3 3 "fox.wav" i 1 6 3 "mary.wav" i 2 9 3 "mary.wav" i 1 12 3 "beats.wav" i 2 15 3 "beats.wav" e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen06.csd.xml0000644000000000000000000000500412262561504017571 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen06.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ;choose between tables kcps init 1/p3 ;create index over duration of note. kndx phasor kcps ixmode = 1 kval table kndx, ifn, ixmode ;normalize mode kfreq = kval * 30 ;scale frequency to emphasixe effect asig poscil .7, 220 + kfreq, 1 ;add to frequency outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave. f 2 0 513 6 1 128 -1 128 1 64 -.5 64 .5 16 -.5 8 1 16 -.5 8 1 16 -.5 84 1 16 -.5 8 .1 16 -.1 17 0 f 3 0 513 6 0 128 0.5 128 1 128 0 129 -1 i 1 0 3 2 i 1 4 3 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen08.csd.xml0000644000000000000000000000473112262561504017601 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen08.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ;choose between tables kcps init 1/p3 ;create index over duration of note. kndx phasor kcps ixmode = 1 kval table kndx, 2, ixmode ;normalize index data ibasefreq = 440 kfreq = kval * 100 ;scale asig poscil .7, ibasefreq + kfreq, 1 ;and add to frequency outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave. f 2 0 65 8 0 16 1 16 1 16 0 17 0 f 3 0 65 8 -1 32 1 2 0 14 0 17 0 i 1 0 2 1 i 1 3 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hsboscil_midi.csd.xml0000644000000000000000000000452012262561504021464 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out -odac -M0 ;;;realtime audio out and realtime MIDI in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; -o hsboscil_midi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; synth waveform giwave ftgen 1, 0, 1024, 10, 1, 1, 1, 1 ; blending window giblend ftgen 2, 0, 1024, -19, 1, 0.5, 270, 0.5 instr 1 ibase = cpsoct(6) ioctcnt = 5 ; all octaves sound alike. itona octmidi ; velocity is mapped to brightness ibrite ampmidi 4 ; Map an exponential envelope for the amplitude. kenv expon .8, 1, .01 asig hsboscil kenv, itona, ibrite, ibase, giwave, giblend, ioctcnt outs asig, asig endin </CsInstruments> <CsScore> i 1 0 30 ; play for 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpstun.csd.xml0000644000000000000000000000516412262561504020175 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cpstun.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Table #1, a normal 12-tone equal temperament scale. ; numgrades = 12 (twelve tones) ; interval = 2 (one octave) ; basefreq = 261.659 (Middle C) ; basekeymidi = 60 (Middle C) gitemp ftgen 1, 0, 64, -2, 12, 2, 261.659, 60, 1.00, \ 1.059, 1.122, 1.189, 1.260, 1.335, 1.414, \ 1.498, 1.588, 1.682, 1.782, 1.888, 2.000 ; Instrument #1. instr 1 ; Set the trigger. ktrig init 1 ; Use Table #1. kfn init 1 ; If the base key (note #60) is C, then 9 notes ; above it (note #60 + 9 = note #69) should be A. kindex init 69 k1 cpstun ktrig, kindex, kfn printk2 k1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/minabsaccum.csd.xml0000644000000000000000000000536212262561504021143 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o minabsaccum.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;saw asig vco2 .2, p4 outs asig, asig gasaw = asig endin instr 2 ;sine aout poscil .3, p4, 1 outs aout, aout gasin = aout endin instr 10 accum init 0 minabsaccum accum, gasaw + gasin ;saw and sine accumulated accum dcblock2 accum ;get rid of DC outs accum, accum clear accum endin </CsInstruments> <CsScore> f 1 0 4096 10 1 i 1 0 7 330 i 2 3 3 440 i 1 10 7 330 ;same notes but without minabsaccum, for comparison i 2 13 3 440 i 10 0 6 ;accumulation note stops after 6 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midiin.csd.xml0000644000000000000000000001314712262561504020132 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -+rtmidi=virtual ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o midiin.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 10 nchnls = 1 ; Example by schwaahed 2006 massign 0, 130 ; make sure that all channels pgmassign 0, 130 ; and programs are assigned to test instr instr 130 knotelength init 0 knoteontime init 0 kstatus, kchan, kdata1, kdata2 midiin if (kstatus == 128) then knoteofftime times knotelength = knoteofftime - knoteontime printks "kstatus= %d, kchan = %d, \\tnote# = %d, velocity = %d \\tNote OFF\\t%f %f\\n", 0, kstatus, kchan, kdata1,kdata2, knoteofftime, knotelength elseif (kstatus == 144) then knoteontime times printks "kstatus= %d, kchan = %d, \\tnote# = %d, velocity = %d \\tNote ON\\t%f\\n", 0, kstatus, kchan, kdata1, kdata2, knoteontime elseif (kstatus == 160) then printks "kstatus= %d, kchan = %d, \\tkdata1 = %d, kdata2 = %d \\tPolyphonic Aftertouch\\n", 0, kstatus, kchan, kdata1, kdata2 elseif (kstatus == 176) then printks "kstatus= %d, kchan = %d, \\t CC = %d, value = %d \\tControl Change\\n", 0, kstatus, kchan, kdata1, kdata2 elseif (kstatus == 192) then printks "kstatus= %d, kchan = %d, \\tkdata1 = %d, kdata2 = %d \\tProgram Change\\n", 0, kstatus, kchan, kdata1, kdata2 elseif (kstatus == 208) then printks "kstatus= %d, kchan = %d, \\tkdata1 = %d, kdata2 = %d \\tChannel Aftertouch\\n", 0, kstatus, kchan, kdata1, kdata2 elseif (kstatus == 224) then printks "kstatus= %d, kchan = %d, \\t ( data1 , kdata2 ) = ( %d, %d )\\tPitch Bend\\n", 0, kstatus, kchan, kdata1, kdata2 endif endin </CsInstruments> <CsScore> i130 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/m.csd.xml0000644000000000000000000000442612262561504017115 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o m.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .8*aenv, p4, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave m foo ;mark section i 1 0 1 110 i 1 1.5 1 220 i 1 3 1 440 i 1 4.5 1 880 s ;second section i 1 0 2 110 i 1 2 2 220 s n foo ;repeat marked section e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tlineto.csd.xml0000644000000000000000000000432412262561504020334 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tlineto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kmtr lfo 1, .5, 1 ;produce trigger signal ktr trigger kmtr, .5, 0 ;with triangle wave ktime = p4 kfreq randh 1000, 3, .2, 0, 500 ;generate random values kfreq tlineto kfreq, ktime, ktr ;different glissando times aout poscil .4, kfreq, giSine outs aout, aout endin </CsInstruments> <CsScore> i 1 0 10 .2 ;short glissando i 1 11 10 .8 ;longer glissande e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/elseif.csd.xml0000644000000000000000000000525712262561504020133 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o elseif.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = cpspch(p4) ienv = p5 if (ienv == 0) then ;ADSR kenv adsr 0.05, 0.05, .95, .05 elseif (ienv == 1) then ;Linear Triangular Envelope kenv linseg 0, p3 * .5, 1, p3 * .5, 0 elseif (ienv == 2) then ;Ramp Up kenv linseg 0, p3 - .01, 1, .01, 0 endif aout vco2 .8, ipch, 10 aout moogvcf aout, ipch + (kenv * 5 * ipch) , .5 aout = aout * kenv outs aout, aout endin </CsInstruments> <CsScore> i 1 0 2 8.00 0 i 1 3 2 8.00 1 i 1 6 2 8.00 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outq3.csd.xml0000644000000000000000000000354712262561504017737 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outq3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 asig vco2 .05, 30 ; sawtooth waveform at low volume kcut line 30, p3, 100 ; Vary cutoff frequency kresonance = 7 inumlayer = 2 asig lowresx asig, kcut, kresonance, inumlayer outq3 asig ; output channel 3 endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/maxabs.csd.xml0000644000000000000000000000442112262561504020127 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o maxabs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 k1 oscili 1, 10.0, 1 ;combine 3 sinusses k2 oscili 1, 1.0, 1 ;at different rates k3 oscili 1, 3.0, 1 kmax maxabs k1, k2, k3 kmax = kmax*250 ;scale kmax printk2 kmax ;check the values aout vco2 .5, 220, 6 ;sawtooth asig moogvcf2 aout, 600+kmax, .5 ;change filter above 600 Hz outs asig, asig endin </CsInstruments> <CsScore> f1 0 32768 10 1 i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sfinstr3.csd.xml0000644000000000000000000000654212262561504020435 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfinstr3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gi24 ftgen 1, 0, 32, -2, 24, 2, 261.626, 60, 1, 1.0293022, 1.059463, 1.0905076, 1.1224619, 1.1553525, 1.1892069, \ 1.2240532, 1.2599207, 1.2968391, 1.33483924, 1.3739531, 1.414213, 1.4556525, 1.4983063, 1.54221, 1.5874001, \ 1.6339145, 1.6817917, 1.73107, 1.7817962, 1.8340067, 1.8877471, 1.9430623, 2 ;table for microtuning, a 24 tone equal temperament giSF sfload "sf_GMbank.sf2" sfilist giSF instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey = p4 ivel = p5 aenv linsegr 1, 1, 1, 1, 0 ;envelope icps cpstuni ikey, 1 ;24 tones per octave iamp = 0.0002 ;scale amplitude iamp = iamp * ivel * 1/128 ;make velocity-dependent aL, aR sfinstr3 ivel, ikey, iamp, icps, 180, giSF, 1 ;= Slap Bass 3 aL = aL * aenv aR = aR * aenv outs aL, aR endin </CsInstruments> <CsScore> f0 60 ;play for 60 seconds i1 0 1 60 100 1 ;using ftable 1 i1 + 1 62 < . i1 + 1 65 < . i1 + 1 69 40 . e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cuserrnd.csd.xml0000644000000000000000000000477612262561504020516 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cuserrnd.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values kuser cuserrnd 0, 100, 1 printk .2, kuser asig poscil .5, 220+kuser, 3 outs asig, asig endin instr 2 ; every run time different values seed 0 kuser cuserrnd 0, 100, 1 printk .2, kuser asig poscil .5, 220+kuser, 3 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16 -7 1 4 0 8 0 4 1 ;distrubution using GEN07 f 2 0 16384 40 1 ;GEN40 is to be used with cuserrnd f 3 0 8192 10 1 ;sine i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/timeinstk.csd.xml0000644000000000000000000000344512262561504020670 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o timeinstk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the value from timeinstk every half-second. k1 timeinstk printks "k1 = %f samples\\n", 0.5, k1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvadd.csd.xml0000644000000000000000000000377012262561504017760 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvadd.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; analyze "fox.wav" with PVANAL first igatefn = p4 ktime line 0, p3, p3 asig pvadd ktime, 1, "fox.pvx", 1, 300, 2, 2, 0, 0, igatefn outs asig*3, asig*3 endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave f 2 0 512 5 1 256 .001 f 3 0 512 7 0 256 1 256 1 i 1 0 2.8 2 i 1 + 2.8 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outletk.csd.xml0000644000000000000000000000426712262561504020353 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o inletk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 connect "bend", "bendout", "guitar", "bendin" instr bend kbend line p4, p3, p5 outletk "bendout", kbend endin instr guitar kbend inletk "bendin" kpch pow 2, kbend/12 printk2 kpch asig oscili .4, 440*kpch, 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 1024 10 1 i"guitar" 0 5 8.00 i"bend" 3 .2 -12 12 i"bend" 4 .1 -17 40 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen32.csd.xml0000644000000000000000000001203212262561504017567 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen32.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 itmp ftgen 1, 0, 16384, 7, 1, 16384, -1 ; sawtooth itmp ftgen 2, 0, 8192, 10, 1 ; sine itmp ftgen 5, 0, 4096, -32, -2, 1.5, 1.0, 0.25, 1, 2, 0.5, 0, 1, 3, -0.25, 0.5 ; mix tables itmp ftgen 6, 0, 16384, 20, 3, 1 ; window ; generate band-limited waveforms inote = 0 loop0: icps = 440 * exp(log(2) * (inote - 69) / 12) ; one table for inumh = sr / (2 * icps) ; each MIDI note number ift = int(inote + 256.5) itmp ftgen ift, 0, 4096, -30, 5, 1, inumh inote = inote + 1 if (inote < 127.5) igoto loop0 instr 1 kcps expon 20, p3, 16000 kft = int(256.5 + 69 + 12 * log(kcps / 440) / log(2)) kft = (kft > 383 ? 383 : kft) a1 phasor kcps a1 tableikt a1, kft, 1, 0, 1 outs a1*.5, a1*.5 endin instr 2 kcps expon 20, p3, 16000 kft = int(256.5 + 69 + 12 * log(kcps / 440) / log(2)) kft = (kft > 383 ? 383 : kft) kgdur limit 10 / kcps, 0.1, 1 a1 grain2 kcps, 0.02, kgdur, 30, kft, 6, -0.5 outs a1*.08, a1*.08 endin </CsInstruments> <CsScore> t 0 60 i 1 0 10 i 2 12 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lorisread.csd.xml0000644000000000000000000001022212262561504020634 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lorisread.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; Play the partials in clarinet.sdif from 0 to 3 sec with 1 ms fadetime ; and no frequency , amplitude, or bandwidth modification. instr 1 ktime linseg 0, p3, 3 ; linear time function from 0 to 3 seconds lorisread ktime, "clarinet.sdif", 1, 1, 1, 1, .001 asig lorisplay 1, 1, 1, 1 outs asig, asig endin ; Play the partials in clarinet.sdif from 0 to 3 sec with 1 ms fadetime ; adding tuning and vibrato, increasing the "breathiness" (noisiness) and overall ; amplitude, and adding a highpass filter. instr 2 ktime linseg 0, p3, 3 ; linear time function from 0 to 3 seconds ; compute frequency scale for tuning ifscale = cpspch(p4)/cpspch(8.08) ; (original pitch was G#4) ; make a vibrato envelope kvenv linseg 0, p3/6, 0, p3/6, .02, p3/3, .02, p3/6, 0, p3/6, 0 kvib oscil kvenv, 4, 1 ; table 1, sinusoid kbwenv linseg 1, p3/6, 1, p3/6, 2, 2*p3/3, 2 ;lots of noise lorisread ktime, "clarinet.sdif", 1, 1, 1, 1, .001 a1 lorisplay 1, ifscale+kvib, 2, kbwenv asig atone a1, 1000 ; highpass filter, cutoff 1000 Hz outs asig, asig endin </CsInstruments> <CsScore> ; a sinus f 1 0 4096 10 1 i 1 0 3 i 1 + 1 i 1 + 6 s ; pitch i 2 1 3 8.08 i 2 3.5 1 8.04 i 2 4 6 8.00 i 2 4 6 8.07 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKDrummer.csd.xml0000644000000000000000000000332312262561504020651 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M0 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKDrummer.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;STK Drummer - has no controllers but plays samples (11) icps cpsmidi iamp ampmidi 1 asig STKDrummer icps, iamp outs asig, asig endin </CsInstruments> <CsScore> ; play 5 minutes f0 300 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKModalBar.csd.xml0000644000000000000000000000355312262561504020724 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKModalBar.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 = p5 ;stick hardness ; asig STKModalBar cpspch(ifrq), 1, 2, kv1, 4, 120, 11, 0, 1, 0, 8, 10, 16, 1 asig = asig * 3 ;amplify outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 8.00 0 i 1 + 2 8.05 120 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vbap.csd.xml0000644000000000000000000000413012262561504017601 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbap.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 vbaplsinit 2, 4, 0, 90, 180, 270 instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav kaz line 0, p3, p4 ;come from right rear speaker & a1,a2,a3,a4 vbap asig, 180, 100, kaz ;change spread of soundsource printks "spread of source = %d\n", 1, kaz ;print spread value outq a1,a2,a3,a4 endin </CsInstruments> <CsScore> i 1 0 12 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tablexkt.csd.xml0000644000000000000000000000571412262561504020500 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tablexkt.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;Example by Jonathan Murphy sr = 44100 ksmps = 10 nchnls = 1 instr 1 ifn = 1 ; query f1 as to number of samples ilen = nsamp(ifn) itrns = 4 ; transpose up 4 octaves ilps = 16 ; allow iwsize/2 samples at start ilpe = ilen - 16 ; and at end imode = 3 ; loop forwards and backwards istrt = 16 ; start 16 samples into loop alphs lphasor itrns, ilps, ilpe, imode, istrt ; use lphasor as index andx = alphs kfn = 1 ; read f1 kwarp = 4 ; anti-aliasing, should be same value as itrns above iwsize = 32 ; iwsize must be at least 8 * kwarp atab tablexkt andx, kfn, kwarp, iwsize atab = atab * 10000 out atab endin </CsInstruments> <CsScore> f 1 0 262144 1 "beats.wav" 0 4 1 i1 0 60 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluidProgramSelect.csd.xml0000644000000000000000000000472512262561504022456 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidProgramSelect.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "19Trumpet.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 56 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 7 asigl, asigr fluidOut giengine outs asigl*imvol, asigr*imvol endin </CsInstruments> <CsScore> i 1 0 2 60 100 ;play one note from score and... i 99 0 60 ;play virtual keyboard for 60 sec. e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/init.csd.xml0000644000000000000000000000343612262561504017624 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n ;no sound output </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;shows what init does kinit init 0 kinit = kinit + 1 printk .1, kinit endin instr 2 ;shows what an assignment does knoinit = 0 knoinit = knoinit + 1 printk .1, knoinit endin </CsInstruments> <CsScore> ;play one second each i1 0 1 i2 2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wgbrass.csd.xml0000644000000000000000000000451112262561504020324 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o wgbrass.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 ksmps = 10 nchnls = 1 0dbfs = 1 ; Instrument #1. instr 1 kamp = 0.7 kfreq = p4 ktens = p5 iatt = p6 kvibf = p7 ifn = 1 ; Create an amplitude envelope for the vibrato. kvamp line 0, p3, 0.5 a1 wgbrass kamp, kfreq, ktens, iatt, kvibf, kvamp, ifn out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 1024 10 1 ; freq tens att vibf i 1 0 4 440 0.4 0.1 6.137 i 1 4 4 440 0.4 0.01 0.137 i 1 8 4 880 0.4 0.1 6.137 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsbandp.csd.xml0000644000000000000000000000412112262561504020466 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsbandp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 Sfile = "fox.wav" klowcut = 100 klowfull = 200 khighfull = 1900 khighcut = 2000 ain soundin Sfile fftin pvsanal ain, 1024, 256, 1024, 1; fft-analysis of the audio-signal fftbp pvsbandp fftin, klowcut, klowfull, khighfull, khighcut ; band pass abp pvsynth fftbp; resynthesis out abp endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/oscil.csd.xml0000644000000000000000000000440312262561504017765 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = .6 kcps = 440 ifn = p4 asig oscil kamp, kcps, ifn outs asig,asig endin </CsInstruments> <CsScore> f1 0 16384 10 1 ; Sine f2 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth f3 0 16384 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square f4 0 16384 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse i 1 0 2 1 i 1 3 2 2 i 1 6 2 3 i 1 9 2 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/printf.csd.xml0000644000000000000000000000406612262561504020163 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o printf.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 Sfile strget p4 ivld filevalid Sfile if ivld=0 then printf_i "Audiofile '%s' does not exist!\n", 1, Sfile else asig diskin2 Sfile, 1 outs asig, asig endif endin </CsInstruments> <CsScore> i 1 0 3 "frox.wav";file does not exist!!! i 1 + 3 "fox.wav";but this one certainly does... e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/partikkel_softsync.csd.xml0000644000000000000000000003323512262561504022577 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out -odac ;;;RT audio ; For Non-realtime ouput leave only the line below: ; -o partikkel_softsync.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 20 nchnls = 2 ; Example by Oeyvind Brandtsegg 2007, revised 2008 giSine ftgen 0, 0, 65537, 10, 1 giCosine ftgen 0, 0, 8193, 9, 1, 1, 90 giSigmoRise ftgen 0, 0, 8193, 19, 0.5, 1, 270, 1 ; rising sigmoid giSigmoFall ftgen 0, 0, 8193, 19, 0.5, 1, 90, 1 ; falling sigmoid ; ************************************************* ; example of soft synchronization of two partikkel instances ; ************************************************* instr 1 /*score parameters*/ igrainrate = p4 ; grain rate igrainsize = p5 ; grain size in ms igrainFreq = p6 ; fundamental frequency of source waveform iosc2Dev = p7 ; partikkel instance 2 grain rate deviation factor iMaxSync = p8 ; max soft sync amount (increasing to this value during length of note) /*overall envelope*/ iattack = 0.001 idecay = 0.2 isustain = 0.7 irelease = 0.2 amp linsegr 0, iattack, 1, idecay, isustain, 1, isustain, irelease, 0 kgrainfreq = igrainrate ; grains per second kdistribution = 0 ; periodic grain distribution idisttab = -1 ; (default) flat distribution used ; for grain distribution async = 0 ; no sync input kenv2amt = 0 ; no secondary enveloping ienv2tab = -1 ; default secondary envelope (flat) ienv_attack = giSigmoRise ; default attack envelope (flat) ienv_decay = giSigmoFall ; default decay envelope (flat) ksustain_amount = 0.3 ; time (in fraction of grain dur) at ; sustain level for each grain ka_d_ratio = 0.2 ; balance between attack and decay time kduration = igrainsize ; set grain duration in ms kamp = 0.2*0dbfs ; amp igainmasks = -1 ; (default) no gain masking kwavfreq = igrainFreq ; fundamental frequency of source waveform ksweepshape = 0 ; shape of frequency sweep (0=no sweep) iwavfreqstarttab = -1 ; default frequency sweep start ; (value in table = 1, which give ; no frequency modification) iwavfreqendtab = -1 ; default frequency sweep end ; (value in table = 1, which give ; no frequency modification) awavfm = 0 ; no FM input ifmamptab = -1 ; default FM scaling (=1) kfmenv = -1 ; default FM envelope (flat) icosine = giCosine ; cosine ftable kTrainCps = kgrainfreq ; set trainlet cps equal to grain ; rate for single-cycle trainlet in ; each grain knumpartials = 3 ; number of partials in trainlet kchroma = 1 ; balance of partials in trainlet ichannelmasks = -1 ; (default) no channel masking, ; all grains to output 1 krandommask = 0 ; no random grain masking kwaveform1 = giSine ; source waveforms kwaveform2 = giSine ; kwaveform3 = giSine ; kwaveform4 = giSine ; iwaveamptab = -1 ; mix of 4 source waveforms and ; trainlets (set to default) asamplepos1 = 0 ; phase offset for reading source waveform asamplepos2 = 0 ; asamplepos3 = 0 ; asamplepos4 = 0 ; kwavekey1 = 1 ; original key for source waveform kwavekey2 = 1 ; kwavekey3 = 1 ; kwavekey4 = 1 ; imax_grains = 100 ; max grains per k period iopcode_id = 1 ; id of opcode, linking partikkel ; to partikkelsync a1 partikkel kgrainfreq, kdistribution, idisttab, async, kenv2amt, \ ienv2tab,ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, \ kduration, kamp, igainmasks, kwavfreq, ksweepshape, \ iwavfreqstarttab, iwavfreqendtab, awavfm, ifmamptab, kfmenv, \ icosine, kTrainCps, knumpartials, kchroma, ichannelmasks, \ krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, asamplepos4, \ kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains, iopcode_id async1 partikkelsync iopcode_id ; clock pulse output of the ; partikkel instance above ksyncGravity line 0, p3, iMaxSync ; strength of synchronization aphase2 init 0 asyncPolarity limit (int(aphase2*2)*2)-1, -1, 1 ; use the phase of partikkelsync instance 2 to find sync ; polarity for partikkel instance 2. ; If the phase of instance 2 is less than 0.5, we want to ; nudge it down when synchronizing, ; and if the phase is > 0.5 we want to nudge it upwards. async1 = async1*ksyncGravity*asyncPolarity ; prepare sync signal ; with polarity and strength kgrainfreq2 = igrainrate * iosc2Dev ; grains per second for second partikkel instance iopcode_id2 = 2 a2 partikkel kgrainfreq2, kdistribution, idisttab, async1, kenv2amt, \ ienv2tab, ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, \ kduration, kamp, igainmasks, kwavfreq, ksweepshape, \ iwavfreqstarttab, iwavfreqendtab, awavfm, ifmamptab, kfmenv, \ icosine, kTrainCps, knumpartials, kchroma, ichannelmasks, \ krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, \ asamplepos4, kwavekey1, kwavekey2, kwavekey3, kwavekey4, \ imax_grains, iopcode_id2 async2, aphase2 partikkelsync iopcode_id2 ; clock pulse and phase ; output of the partikkel instance above, ; we will only use the phase outs a1*amp, a2*amp endin </CsInstruments> <CsScore> /*score parameters igrainrate = p4 ; grain rate igrainsize = p5 ; grain size in ms igrainFreq = p6 ; frequency of source wave within grain iosc2Dev = p7 ; partikkel instance 2 grain rate deviation factor iMaxSync = p8 ; max soft sync amount (increasing to this value during length of note) */ ; GrRate GrSize GrFund Osc2Dev MaxSync i1 0 10 2 20 880 1.3 0.3 s i1 0 10 5 20 440 0.8 0.3 s i1 0 6 55 15 660 1.8 0.45 s i1 0 6 110 10 440 0.6 0.6 s i1 0 6 220 3 660 2.6 0.45 s i1 0 6 220 3 660 2.1 0.45 s i1 0 6 440 3 660 0.8 0.22 s e e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/bformenc1.csd.xml0000644000000000000000000000721412262561504020533 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages ;-odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -o bformenc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 8 0dbfs = 1 instr 1 ;without arrays ; generate pink noise anoise pinkish 1 ; two full turns kalpha line 0, p3, 720 kbeta = 0 ; generate B format aw, ax, ay, az, ar, as, at, au, av bformenc1 anoise, kalpha, kbeta ; decode B format for 8 channel circle loudspeaker setup a1, a2, a3, a4, a5, a6, a7, a8 bformdec1 4, aw, ax, ay, az, ar, as, at, au, av ; write audio out outo a1, a2, a3, a4, a5, a6, a7, a8 endin instr 2 ;with arrays (csound6) ;set file names for: S_bform = "bform_out.wav" ;b-format (2nd order) output S_sound = "sound_out.wav" ;sound output ; generate pink noise anoise pinkish 1 ; two full turns kalpha line 0, p3, 720 kbeta = 0 ;create array for B-format 2nd order (9 chnls) aBform[] init 9 ; generate B-format aBform bformenc1 anoise, kalpha, kbeta ;write out b-format fout "fout.wav", 18, aBform ;create array for audio output (8 channels) aAudio[] init 8 ;decode B format for 8 channel circle loudspeaker setup aAudio bformdec1 4, aBform ; write audio out fout S_sound, 18, aAudio endin </CsInstruments> <CsScore> i 1 0 8 i 2 8 8 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/barmodel.csd.xml0000644000000000000000000000363212262561504020444 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o barmodel.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 aq barmodel 1, 1, p4, 0.001, 0.23, 5, p5, p6, p7 out aq endin </CsInstruments> <CsScore> i1 0.0 0.5 3 0.2 500 0.05 i1 0.5 0.5 -3 0.3 1000 0.05 i1 1.0 0.5 -3 0.4 1000 0.1 i1 1.5 4.0 -3 0.5 800 0.05 e /* barmodel */ </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/octmidib.csd.xml0000644000000000000000000000341412262561504020447 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o octmidib.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; This example expects MIDI note inputs on channel 1 i1 octmidib print i1 endin </CsInstruments> <CsScore> ;Dummy f-table to give time for real-time MIDI events f 0 8000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/noteondur.csd.xml0000644000000000000000000000366112262561504020676 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1;;;RT audio I/O with MIDI in </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 instr 1 ;Turned on by MIDI notes on channel 1 ifund notnum ivel veloc idur = 1 ;chord with single key noteondur 1, ifund, ivel, idur noteondur 1, ifund+3, ivel, idur noteondur 1, ifund+7, ivel, idur noteondur 1, ifund+9, ivel, idur endin </CsInstruments> <CsScore> ; Play Instrument #1 for 60 seconds. i1 0 60 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/rnd31.csd.xml0000644000000000000000000000430712262561504017606 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd31.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create random numbers at a-rate in the range -2 to 2 with ; a triangular distribution, seed from the current time. a31 rnd31 2, -0.5 ; Use the random numbers to choose a frequency. afreq = a31 * 500 + 100 a1 oscil 30000, afreq, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSinterpread.csd.xml0000644000000000000000000000357112262561504021366 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc for RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSinterpread.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 1.8 ATSbufread ktime, 1, "beats.ats", 42 kamp ATSinterpread p4 aosc oscili kamp, p4, 1 outs aosc * 25, aosc *25 endin </CsInstruments> <CsScore> ; sine wave. f 1 0 16384 10 1 i 1 0 2 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/invalue.csd.xml0000644000000000000000000000314012262561504020314 0ustar rootroot <CsoundSynthesizer> <CsOptions> </CsOptions> <CsInstruments> ;run this example in CsoundQt, a Csound editor that provides widgets ;make the Widgets-panel visible, by clicking the Widgets symbol in the menu or pressing (Alt+1). sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; written by Andres Cabrera instr 1 kfreq invalue "freq" ; Quotes are needed here asig oscil 0.1, kfreq, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 1024 10 1 ;sine i 1 0 300 ;play for 300 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vadd_i.csd.xml0000644000000000000000000000507012262561504020103 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vadd_i ifn1, ival, ielements, idstoffset endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/randh.csd.xml0000644000000000000000000000467212262561504017760 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o randh.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;same values every time krnd randh 100, 10 printk2 krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin instr 2 ;different values every time krnd randh 100, 10, 10 ; seed from system clock printk2 krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave. i 1 0 1 i 2 2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/array_scalar_math.csd.xml0000644000000000000000000001140312262561504022326 0ustar rootroot <CsoundSynthesizer> <CsOptions> -n -m128 </CsOptions> <CsInstruments> instr 1 ;create array and fill with numbers 1..10 kArr1[] fillarray 1, 2, 3, 4, 5, 6, 7, 8, 9, 10 ;print content printf "%s", 1, "\nInitial content:\n" kndx = 0 until kndx == lenarray(kArr1) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr1[kndx] kndx += 1 od ;add 10 kArr2[] = kArr1 + 10 ;print content printf "%s", 1, "\nAfter adding 10:\n" kndx = 0 until kndx == lenarray(kArr2) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr2[kndx] kndx += 1 od ;subtract 5 kArr3[] = kArr2 - 5 ;print content printf "%s", 1, "\nAfter subtracting 5:\n" kndx = 0 until kndx == lenarray(kArr3) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr3[kndx] kndx += 1 od ;multiply by -1.5 kArr4[] = kArr3 * -1.5 ;print content printf "%s", 1, "\nAfter multiplying by -1.5:\n" kndx = 0 until kndx == lenarray(kArr4) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr4[kndx] kndx += 1 od ;divide by -3/2 kArr5[] = kArr4 / -(3/2) ;print content printf "%s", 1, "\nAfter dividing by -3/2:\n" kndx = 0 until kndx == lenarray(kArr5) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr5[kndx] kndx += 1 od ;turnoff turnoff endin </CsInstruments> <CsScore> i 1 0 .1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/minabs.csd.xml0000644000000000000000000000442112262561504020125 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o minabs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 k1 oscili 1, 10.0, 1 ;combine 3 sinusses k2 oscili 1, 1.0, 1 ;at different rates k3 oscili 1, 3.0, 1 kmin minabs k1, k2, k3 kmin = kmin*250 ;scale kmin printk2 kmin ;check the values aout vco2 .5, 220, 6 ;sawtooth asig moogvcf2 aout, 600+kmin, .5 ;change filter above 600 Hz outs asig, asig endin </CsInstruments> <CsScore> f1 0 32768 10 1 i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLsavesnap_simple.csd.xml0000644000000000000000000001006712262561504022272 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O </CsOptions> <CsInstruments> sr=48000 ksmps=128 nchnls=2 ; Example by Hector Centeno and Andres Cabrera 2007 ; giSWMtab4 ftgen 0, 0, 513, 21, 10, 1, .3 ; giSWMtab4M ftgen 0, 0, 64, 7, 1, 50, 1 FLpanel "Snapshots", 530, 190, 40, 410, 3 FLcolor 100, 118 ,140 ivalSM1 FLvalue "", 70, 20, 270, 20 gksliderA, gislidSM1 FLslider "Slider", -4, 4, 0, 3, ivalSM1, 250, 20, 20, 20 itext1 FLbox "store", 1, 1, 14, 50, 25, 355, 15 itext2 FLbox "load", 1, 1, 14, 50, 25, 415, 15 gksnap, ibuttn1 FLbutton "1", 1, 0, 11, 25, 25, 364, 45, 0, 3, 0, 3, 1 gksnap, ibuttn2 FLbutton "2", 1, 0, 11, 25, 25, 364, 75, 0, 3, 0, 3, 2 gksnap, ibuttn3 FLbutton "3", 1, 0, 11, 25, 25, 364, 105, 0, 3, 0, 3, 3 gksnap, ibuttn4 FLbutton "4", 1, 0, 11, 25, 25, 364, 135, 0, 3, 0, 3, 4 gkload, ibuttn1 FLbutton "1", 1, 0, 11, 25, 25, 424, 45, 0, 4, 0, 3, 1 gkload, ibuttn2 FLbutton "2", 1, 0, 11, 25, 25, 424, 75, 0, 4, 0, 3, 2 gkload, ibuttn3 FLbutton "3", 1, 0, 11, 25, 25, 424, 105, 0, 4, 0, 3, 3 gkload, ibuttn4 FLbutton "4", 1, 0, 11, 25, 25, 424, 135, 0, 4, 0, 3, 4 ivalSM2 FLvalue "", 70, 20, 270, 80 gkknobA, gislidSM2 FLknob "Knob", -4, 4, 0, 3, ivalSM2, 60, 120, 60 FLpanelEnd FLsetVal_i 1, gislidSM1 FLsetVal_i 1, gislidSM2 FLrun instr 1 endin instr 3 ; Save snapshot index init 0 ipstno = p4 Sfile sprintf "snapshot_simple.%d.snap", ipstno inumsnap, inumval FLsetsnap index ;, -1, igroup FLsavesnap Sfile endin instr 4 ;Load snapshot index init 0 ipstno = p4 Sfile sprintf "snapshot_simple.%d.snap", ipstno FLloadsnap Sfile inumload FLgetsnap index ;, igroup endin </CsInstruments> <CsScore> f 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zkcl.csd.xml0000644000000000000000000000600612262561504017620 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zkcl.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Linearly vary a k-rate signal from 220 to 1760. kline line 220, p3, 1760 ; Add the linear signal to zk variable #1. zkw kline, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read zk variable #1. kfreq zkr 1 ; Use the value of zk variable #1 to vary ; the frequency of a sine waveform. a1 oscil 20000, kfreq, 1 ; Generate the audio output. out a1 ; Clear the zk variables, get them ready for ; another pass. zkcl 0, 1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 ; Play Instrument #2 for three seconds. i 2 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pcount.csd.xml0000644000000000000000000001201312262561504020160 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pcount.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; This UDO returns a pfield value but returns 0 if it does not exist. opcode mypvalue, i, i index xin inum pcount if (index > inum) then iout = 0.0 else iout pindex index endif xout iout endop ; Envelope UDO that reads parameters from a flexible number of pfields ; Syntax: kenv flexlinseg ipstart ; ipstart is the first pfield of the envelope ; parameters. Reads remaining pfields (up to 21 of them). ; kenv is the output envelope. opcode flexlinseg, k, i ipstart xin iep1 mypvalue ipstart iep2 mypvalue ipstart + 1 iep3 mypvalue ipstart + 2 iep4 mypvalue ipstart + 3 iep5 mypvalue ipstart + 4 iep6 mypvalue ipstart + 5 iep7 mypvalue ipstart + 6 iep8 mypvalue ipstart + 7 iep9 mypvalue ipstart + 8 iepa mypvalue ipstart + 9 iepb mypvalue ipstart + 10 iepc mypvalue ipstart + 11 iepd mypvalue ipstart + 12 iepe mypvalue ipstart + 13 iepf mypvalue ipstart + 14 iepg mypvalue ipstart + 15 ieph mypvalue ipstart + 16 iepi mypvalue ipstart + 17 iepj mypvalue ipstart + 18 iepk mypvalue ipstart + 19 iepl mypvalue ipstart + 20 kenv linseg iep1, iep2, iep3, iep4, iep5, iep6, iep7, iep8, \ iep9, iepa, iepb, iepc, iepd, iepe, iepf, iepg, \ ieph, iepi, iepj, iepk, iepl xout kenv endop instr 1 ; This instrument only requires 3 pfields but can accept up to 24. ; (You will still get warnings about more than 3). kenv flexlinseg 4 ; envelope params start at p4 aout oscili kenv*.5, 256, 1 outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 i1 0 2 0.0 1.0 1.0 1.0 0.0 i1 2 2 0.0 0.1 1.0 1.0 1.0 0.1 0.0 i1 4 2 0.0 0.5 0.0 ; one problem is that "missing" pfields carry i1 6 2 0.0 0.5 0.0 ! ; now we can fix this problem with ! i1 8 10 0.0 3.0 1.0 0.3 0.1 0.3 1.0 0.3 0.1 0.3 1.0 0.3 0.1 0.8 0.9 5.0 0.0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/print.csd.xml0000644000000000000000000000365212262561504020015 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o print.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 irand = rnd(3) ;generate a random number from 0 to 3 print irand ;print it asig poscil .7, 440*irand, 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 16384 10 1 ;sine wave i 1 0 1 i 1 2 1 i 1 4 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/a.csd.xml0000644000000000000000000000540412262561504017076 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o a.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .8*aenv, p4, 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 16384 10 1 ;sine wave ;two sections s a 0 0 6 ;advance score 6 seconds i 1 0 2 110 ;these first 2 notes i 1 3 2 220 ;will not sound i 1 6 2 440 i 1 9 2 880 s a 0 3 6 ;advance score 6 seconds, but do this after 3 seconds i 1 0 2 110 ;this will sound, because action time (p2) from a statement = 3 i 1 3 2 220 ;so these 2 notes i 1 6 2 440 ;will not sound i 1 9 2 880 ;and this one will e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvscross.csd.xml0000644000000000000000000000536612262561504020547 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 ipermut = p4; 1 = change order of soundfiles ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "fox.wav" Sfile2 = "wave.wav" ain1 soundin Sfile1 ain2 soundin Sfile2 fftin1 pvsanal ain1, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file 1 fftin2 pvsanal ain2, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file 2 ktrans linseg 0, p3, 1; linear transition if ipermut == 1 then fcross pvscross fftin2, fftin1, ktrans, 1-ktrans else fcross pvscross fftin1, fftin2, ktrans, 1-ktrans endif aout pvsynth fcross out aout endin </CsInstruments> <CsScore> i 1 0 2.757 0; frequencies from fox. wav; amplitudes moving from wave to fox i 1 3 2.757 1; frequencies from wave.wav, amplitudes moving from fox to wave e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gendyx.csd.xml0000644000000000000000000000276212262561504020160 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aout gendyx 0.7, 1, 1, 1, 1, 20, 1000, 0.5, 0.5, 4, 0.13 outs aout, aout endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/planet.csd.xml0000644000000000000000000000700012262561504020133 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o planet.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 2 ; Instrument #1 - a planet oribiting in 3D space. instr 1 ; Create a basic tone. kamp init 5000 kcps init 440 ifn = 1 asnd oscil kamp, kcps, ifn ; Figure out its X, Y, Z coordinates. km1 init 0.5 km2 init 0.35 ksep init 2.2 ix = 0 iy = 0.1 iz = 0 ivx = 0.5 ivy = 0 ivz = 0 ih = 0.0003 ifric = -0.1 ax1, ay1, az1 planet km1, km2, ksep, ix, iy, iz, \ ivx, ivy, ivz, ih, ifric ; Place the basic tone within 3D space. kx downsamp ax1 ky downsamp ay1 kz downsamp az1 idist = 1 ift = 0 imode = 1 imdel = 1.018853416 iovr = 2 aw2, ax2, ay2, az2 spat3d asnd, kx, ky, kz, idist, \ ift, imode, imdel, iovr ; Convert the 3D sound to stereo. aleft = aw2 + ay2 aright = aw2 - ay2 outs aleft, aright endin </CsInstruments> <CsScore> ; Table #1 a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 10 seconds. i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/urd.csd.xml0000644000000000000000000000616012262561504017450 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o urd.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktab = 1 ;ftable 1 kurd = urd(ktab) ktrig metro 5 ;triggers 5 times per second kres samphold kurd, ktrig ;sample and hold value of kurd printk2 kres ;print it asig poscil .5, 220+kres, 2 outs asig, asig endin instr 2 seed 0 ;every run new values ktab = 1 ;ftable 1 kurd = urd(ktab) ktrig metro 5 ;triggers 5 times per second kres samphold kurd, ktrig ;sample and hold value of kurd printk2 kres ;print it asig poscil .5, 220+kres, 2 outs asig, asig endin </CsInstruments> <CsScore> f1 0 -20 -42 10 20 .3 100 200 .7 ;30% choose between 10 and 20 and 70% between 100 and 200 f2 0 8192 10 1 ;sine wave i 1 0 5 i 2 6 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gausstrig.csd.xml0000644000000000000000000000347212262561504020671 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kdev line 0, p3, 0.9 seed 20120125 aimp gausstrig 0.5, 10, kdev aenv filter2 aimp, 1, 1, 0.993, 0.993 anoi fractalnoise 0.2, 1.7 al = anoi*aenv ar delay al, 0.02 outs al, ar endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/trhighest.csd.xml0000644000000000000000000000426412262561504020662 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trhighest.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "fox.wav", 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .1, 1, 3, 500 ; partial tracking fhi,kfr,kamp trhighest fst, 1 ; highest freq-track aout tradsyn fhi, 1, 1, 1, 1 ; resynthesis of highest frequency outs aout*40, aout*40 ; compensate energy loss endin </CsInstruments> <CsScore> f1 0 8192 10 1 ;sine wave i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tablei.csd.xml0000644000000000000000000000674412262561504020126 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tablei.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 ;generate new values every time the instr is played instr 1 ifn = p4 isize = p5 ithresh = 0.5 itemp ftgen ifn, 0, isize, 21, 2 iwrite_value = 0 i_index = 0 loop_start: iread_value tablei i_index, ifn if iread_value > ithresh then iwrite_value = 1 else iwrite_value = -1 endif tableiw iwrite_value, i_index, ifn loop_lt i_index, 1, isize, loop_start turnoff endin instr 2 ifn = p4 isize = ftlen(ifn) prints "Index\tValue\n" i_index = 0 loop_start: ivalue tablei i_index, ifn prints "%d:\t%f\n", i_index, ivalue loop_lt i_index, 1, isize, loop_start ;read table 1 with our index aout oscili .5, 100, ifn ;use table to play the polypulse outs aout, aout endin </CsInstruments> <CsScore> i 1 0 1 100 16 i 2 0 2 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/floor.csd.xml0000644000000000000000000000404512262561504017777 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o floor.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idiv init 1 loop: inumber = 9 i1 = inumber / idiv ifl = floor(i1) print inumber, idiv, ifl ;print number / idiv = result using floor idiv = idiv + 1 if (idiv <= 10) igoto loop endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/space_quad.csd.xml0000644000000000000000000000650612262561504020767 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o space_quad.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 4 ga1 init 0 ga2 init 0 ga3 init 0 ga4 init 0 instr 1 ;uses GEN28 file "move", as found in /manual/examples kx init 0 ky init 0 ktime line 0, 5, 5 ;same time as in table 1 (="move") asig diskin2 "beats.wav", 1, 0, 1 ;sound source is looped a1, a2, a3, a4 space asig, 1, ktime, .1, kx, ky ;use table 1 = GEN28 ar1, ar2, ar3, ar4 spsend ;send to reverb ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 a3 reverb2 ga3, 2.5, .5 a4 reverb2 ga4, 2.5, .5 outq a1, a2, a3, a4 ga1=0 ga2=0 ga3=0 ga4=0 endin </CsInstruments> <CsScore> f1 0 0 28 "move" i1 0 5 ;same time as ktime i 99 0 10 ;keep reverb active e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsmooth.csd.xml0000644000000000000000000000476712262561504020550 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsmooth.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kacf = p4 kfcf = p5 asig soundin "fox.wav" fsig pvsanal asig, 1024, 256, 1024, 1 ; analyse it ftps pvsmooth fsig, kacf, kfcf atps pvsynth ftps ; synthesise it outs atps*3, atps*3 endin </CsInstruments> <CsScore> ; amp freq i 1 0 3 0.01 0.01 ;smooth amplitude and frequency with cutoff frequency of filter at 1% of 1/2 frame-rate (ca 0.86 Hz) i 1 + 3 1 0.01 ;no smoothing on amplitude, but frequency with cf at 1% of 1/2 frame-rate (ca 0.86 Hz) i 1 + 10 .001 1 ;smooth amplitude with cf at 0.1% of 1/2 frame-rate (ca 0.086 Hz) ;and no smoothing of frequency e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dssiaudio.csd.xml0000644000000000000000000000537012262561504020644 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o DSSIplay_mono.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 gihandle dssiinit "caps.so", 19, 1 ; = mono phaser and gaout init 0 ; verbose about all ports instr 1 ; activate DSSI dssiactivate gihandle, 1 endin instr 2 ain1 diskin2 "beats.wav", 1,0,1 ; loop gaout = gaout+(ain1*.5) endin instr 3 dssictls gihandle, 0, .8, 1 ; range -1 to 1 dssictls gihandle, 1, .05, 1 ; rate 0 to 10 dssictls gihandle, 2, .8, 1 ; depth 0 to 1 dssictls gihandle, 3, 2, 1 ; spread 0 to 3.14 dssictls gihandle, 4, .7, 1 ; feedback 0 to 0.999 endin instr 4 aout1 dssiaudio gihandle, gaout ;get beats.wav, mono out outs aout1,aout1 gaout = 0 endin </CsInstruments> <CsScore> i 1 0 20 i 2 1 20 i 3 1 20 i 4 0 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outkpc.csd.xml0000644000000000000000000000346712262561504020172 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1 ;;;RT audio I/O with MIDI in </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 kprogram init 0 instr 1 ;Triggered by MIDI notes on channel 1 ifund notnum ivel veloc idur = 1 ; Sends a MIDI program change message according to ; the triggering note's velocity outkpc 1 ,ivel ,0 ,127 noteondur 1 ,ifund ,ivel ,idur endin </CsInstruments> <CsScore> ; Dummy ftable f 0 60 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/duserrnd.csd.xml0000644000000000000000000000353512262561504020507 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o duserrnd.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 k1 duserrnd 1 printk 0, k1 asig poscil .5, 220*k1, 2 ;multiply frequency with random value outs asig, asig endin </CsInstruments> <CsScore> f1 0 -20 -41 2 .1 8 .9 ;choose 2 at 10% probability, and 8 at 90% f2 0 8192 10 1 i1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ins.csd.xml0000644000000000000000000000377512262561504017460 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -idac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; ins.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 ;two channels out 0dbfs = 1 instr 1 ain1, ainr ins ;grab your mic and sing adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time = 20ms aoutl flanger ain1, adel, .7 aoutr flanger ain1, adel*2, .8 fout "in_s.wav", 14, aoutl, aoutr ;write to stereo file, outs aoutl, aoutr ;16 bits with header endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midicontrolchange.csd.xml0000644000000000000000000000453712262561504022355 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midicontrolchange.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; use slider of contr. 7 of virtual keyboard kcont init 1 ; max. volume midicontrolchange 7, kcont, 0, 1; use controller 7, scaled between 0 and 1 printk2 kcont ; Display the key value when it changes and key is pressed kenv madsr 0.5, 0.8, 0.8, 0.5 ; envelope multiplied by asig pluck kenv*kcont, 220, 220, 2, 1 ; value of controller 7 outs asig, asig endin </CsInstruments> <CsScore> f 0 30 f 2 0 4096 10 1 i 1 10 2 ; play a note from score as well e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/luamoog.csd.xml0000644000000000000000000005435412262561504020331 0ustar rootroot <CsoundSynthesizer> <CsInstruments> sr = 48000 ksmps = 100 nchnls = 1 gibegan rtclock lua_opdef "moogladder", {{ local ffi = require("ffi") local math = require("math") local string = require("string") local csoundApi = ffi.load('csound64.dll.5.2') ffi.cdef[[ int csoundGetKsmps(void *); double csoundGetSr(void *); struct moogladder_t { double *out; double *inp; double *freq; double *res; double *istor; double sr; double ksmps; double thermal; double f; double fc; double fc2; double fc3; double fcr; double acr; double tune; double res4; double input; double i; double j; double k; double kk; double stg[6]; double delay[6]; double tanhstg[6]; }; ]] local moogladder_ct = ffi.typeof('struct moogladder_t *') function moogladder_init(csound, opcode, carguments) local p = ffi.cast(moogladder_ct, carguments) p.sr = csoundApi.csoundGetSr(csound) p.ksmps = csoundApi.csoundGetKsmps(csound) if p.istor[0] == 0 then for i = 0, 5 do p.delay[i] = 0.0 end for i = 0, 3 do p.tanhstg[i] = 0.0 end end return 0 end function moogladder_kontrol(csound, opcode, carguments) local p = ffi.cast(moogladder_ct, carguments) -- transistor thermal voltage p.thermal = 1.0 / 40000.0 if p.res[0] < 0.0 then p.res[0] = 0.0 end -- sr is half the actual filter sampling rate p.fc = p.freq[0] / p.sr p.f = p.fc / 2.0 p.fc2 = p.fc * p.fc p.fc3 = p.fc2 * p.fc -- frequency & amplitude correction p.fcr = 1.873 * p.fc3 + 0.4955 * p.fc2 - 0.6490 * p.fc + 0.9988 p.acr = -3.9364 * p.fc2 + 1.8409 * p.fc + 0.9968 -- filter tuning p.tune = (1.0 - math.exp(-(2.0 * math.pi * p.f * p.fcr))) / p.thermal p.res4 = 4.0 * p.res[0] * p.acr -- Nested 'for' loops crash, not sure why. -- Local loop variables also are problematic. -- Lower-level loop constructs don't crash. p.i = 0 while p.i < p.ksmps do p.j = 0 while p.j < 2 do p.k = 0 while p.k < 4 do if p.k == 0 then p.input = p.inp[p.i] - p.res4 * p.delay[5] p.stg[p.k] = p.delay[p.k] + p.tune * (math.tanh(p.input * p.thermal) - p.tanhstg[p.k]) else p.input = p.stg[p.k - 1] p.tanhstg[p.k - 1] = math.tanh(p.input * p.thermal) if p.k < 3 then p.kk = p.tanhstg[p.k] else p.kk = math.tanh(p.delay[p.k] * p.thermal) end p.stg[p.k] = p.delay[p.k] + p.tune * (p.tanhstg[p.k - 1] - p.kk) end p.delay[p.k] = p.stg[p.k] p.k = p.k + 1 end -- 1/2-sample delay for phase compensation p.delay[5] = (p.stg[3] + p.delay[4]) * 0.5 p.delay[4] = p.stg[3] p.j = p.j + 1 end p.out[p.i] = p.delay[5] p.i = p.i + 1 end return 0 end }} /* Moogladder - An improved implementation of the Moog ladder filter DESCRIPTION This is an new digital implementation of the Moog ladder filter based on the work of Antti Huovilainen, described in the paper \"Non-Linear Digital Implementation of the Moog Ladder Filter\" (Proceedings of DaFX04, Univ of Napoli). This implementation is probably a more accurate digital representation of the original analogue filter. This is version 2 (revised 14/DEC/04), with improved amplitude/resonance scaling and frequency correction using a couple of polynomials,as suggested by Antti. SYNTAX ar Moogladder asig, kcf, kres PERFORMANCE asig - input signal kcf - cutoff frequency (Hz) kres - resonance (0 - 1). CREDITS Victor Lazzarini */ opcode moogladderu, a, akk asig, kcf, kres xin setksmps 1 ipi = 4 * taninv(1) /* filter delays */ az1 init 0 az2 init 0 az3 init 0 az4 init 0 az5 init 0 ay4 init 0 amf init 0 if kres > 1 then kres = 1 elseif kres < 0 then kres = 0 endif /* twice the \'thermal voltage of a transistor\' */ i2v = 40000 /* sr is half the actual filter sampling rate */ kfc = kcf/sr kf = kcf/(sr*2) /* frequency & amplitude correction */ kfcr = 1.8730 * (kfc^3) + 0.4955 * (kfc^2) - 0.6490 * kfc + 0.9988 kacr = -3.9364 * (kfc^2) + 1.8409 * kfc + 0.9968; /* filter tuning */ k2vg = i2v * (1 - exp(-2 * ipi * kfcr * kf)) /* cascade of 4 1st order sections */ ay1 = az1 + k2vg * (tanh((asig - 4 * kres * amf * kacr) / i2v) - tanh(az1 / i2v)) az1 = ay1 ay2 = az2 + k2vg * (tanh(ay1 / i2v) - tanh(az2 / i2v )) az2 = ay2 ay3 = az3 + k2vg * (tanh(ay2 / i2v) - tanh(az3 / i2v)) az3 = ay3 ay4 = az4 + k2vg * (tanh(ay3 / i2v) - tanh(az4 / i2v)) az4 = ay4 /* 1/2-sample delay for phase compensation */ amf = (ay4 + az5) *0.5 az5 = ay4 /* oversampling */ ay1 = az1 + k2vg * (tanh((asig - 4 * kres * amf * kacr) / i2v) - tanh(az1 / i2v)) az1 = ay1 ay2 = az2 + k2vg * (tanh(ay1 / i2v) - tanh(az2 / i2v )) az2 = ay2 ay3 = az3 + k2vg * (tanh(ay2 / i2v) - tanh(az3 / i2v)) az3 = ay3 ay4 = az4 + k2vg * (tanh(ay3 / i2v) - tanh(az4 / i2v)) az4 = ay4 amf = (ay4 + az5) * 0.5 az5 = ay4 xout amf endop instr 1 prints "No filter.\n" kfe expseg 500, p3*0.9, 1800, p3*0.1, 3000 kenv linen 10000, 0.05, p3, 0.05 asig buzz kenv, 100, sr/(200), 1 ; afil moogladder asig, kfe, 1 out asig endin instr 2 prints "Native moogladder.\n" kfe expseg 500, p3*0.9, 1800, p3*0.1, 3000 kenv linen 10000, 0.05, p3, 0.05 asig buzz kenv, 100, sr/(200), 1 afil moogladder asig, kfe, 1 out afil endin instr 3 prints "UDO moogladder.\n" kfe expseg 500, p3*0.9, 1800, p3*0.1, 3000 kenv linen 10000, 0.05, p3, 0.05 asig buzz kenv, 100, sr/(200), 1 afil moogladderu asig, kfe, 1 out afil endin instr 4 prints "Lua moogladder.\n" kres init 1 istor init 0 kfe expseg 500, p3*0.9, 1800, p3*0.1, 3000 kenv linen 10000, 0.05, p3, 0.05 asig buzz kenv, 100, sr/(200), 1 afil init 0 lua_ikopcall "moogladder", afil, asig, kfe, kres, istor out afil endin instr 5 giended rtclock ielapsed = giended - gibegan print ielapsed gibegan rtclock endin </CsInstruments> <CsScore> f 1 0 65536 10 1 i 5.1 0 1 i 4 1 20 i 5.2 21 1 i 4 22 20 i 5.3 42 1 i 2 43 20 i 5.4 63 1 i 2 64 20 i 5.5 84 1 i 3 85 20 i 5.6 105 1 i 3 106 20 i 5.7 126 1 i 1 127 20 i 5.8 147 1 i 1 148 20 i 5.9 168 1 i 4 169 20 i 4 170 20 i 4 171 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSpartialtap.csd.xml0000644000000000000000000000440012262561504021362 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc for RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSpartialtap.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 2 ATSbufread ktime, 1, "beats.ats", 30 kfreq1, kam1 ATSpartialtap 5 kfreq2, kam2 ATSpartialtap 20 kfreq3, kam3 ATSpartialtap 30 aout1 oscil kam1, kfreq1, 1 aout2 oscil kam2, kfreq2, 1 aout3 oscil kam3, kfreq3, 1 aout = (aout1+aout2+aout3)*10 ; amplify some more outs aout, aout endin </CsInstruments> <CsScore> ; sine wave. f 1 0 16384 10 1 i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/oscils.csd.xml0000644000000000000000000000315212262561504020150 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscils.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iflg = p4 asig oscils .7, 220, 0, iflg outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 0 i 1 3 2 2 ;double precision e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/setctrl.csd.xml0000644000000000000000000000450512262561504020337 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o setctrl.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Display the label "Volume" on Slider #1. setctrl 1, "Volume", 4 ; Set Slider #1's initial value to 20. setctrl 1, 20, 1 ; Capture and display the values for Slider #1. k1 control 1 printk2 k1 ; Play a simple oscillator. ; Use the values from Slider #1 for amplitude. kamp = k1 * 128 a1 oscil kamp, 440, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for thirty seconds. i 1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pcauchy.csd.xml0000644000000000000000000000465312262561504020317 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pcauchy.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values kalpha pcauchy 1000 printk .2, kalpha ; look aout oscili 0.8, 440+kalpha, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 kalpha pcauchy 1000 printk .2, kalpha ; look aout oscili 0.8, 440+kalpha, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sfilist.csd.xml0000644000000000000000000000535612262561504020341 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out, virtual midi in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfilist.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf sfload "sf_GMbank.sf2" sfilist gisf ;list all instruments instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/3000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1, a2 sfinstr3 ivel, inum, kamp*ivel, kfreq, 100, gisf ;choose Halo Pad outs a1, a2 endin </CsInstruments> <CsScore> f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/maxaccum.csd.xml0000644000000000000000000000517612262561504020462 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o maxaccum.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;saw asig vco2 .2, p4 outs asig, asig gasaw = asig endin instr 2 ;sine aout poscil .3, p4, 1 outs aout, aout gasin = aout endin instr 10 accum init 0 maxaccum accum, gasaw + gasin ;saw and sine accumulated outs accum, accum clear accum endin </CsInstruments> <CsScore> f 1 0 4096 10 1 i 1 0 7 330 i 2 3 3 440 i 1 10 7 330 ;same notes but without maxaccum, for comparison i 2 13 3 440 i 10 0 6 ;accumulation note stops after 6 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/scale.csd.xml0000644000000000000000000000401012262561504017735 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in Silent -odac -iadc -d ;;;realtime output </CsOptions> <CsInstruments> sr = 22050 ksmps = 10 nchnls = 2 /*--- ---*/ instr 1 ; scale test kmod ctrl7 1, 1, 0, 1 printk2 kmod kout scale kmod, 0, -127 printk2 kout endin /*--- ---*/ </CsInstruments> <CsScore> i1 0 8888 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/table.csd.xml0000644000000000000000000000457012262561504017750 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o table.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Vary our index linearly from 0 to 1. kndx line 0, p3, 1 ; Read Table #1 with our index. ifn = 1 ixmode = 1 kfreq table kndx, ifn, ixmode ; Generate a sine waveform, use our table values ; to vary its frequency. a1 oscil 20000, kfreq, 2 out a1 endin </CsInstruments> <CsScore> ; Table #1, a line from 200 to 2,000. f 1 0 1025 -7 200 1024 2000 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/genarray_i.csd.xml0000644000000000000000000000541112262561504020774 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;create and fill two arrays kArr1[] genarray_i 1, 6 kArr2[] genarray_i 1, 6, 2/3 ;print the content of kArr1 printf "%s", 1, "kArr1: start=1, end=6, step=default\n" kndx = 0 until kndx == lenarray(kArr1) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr1[kndx] kndx += 1 od ;print the content of kArr2 printf "%s", 1, "kArr2: start=1, end=6, step=2/3\n" kndx = 0 until kndx == lenarray(kArr2) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr2[kndx] kndx += 1 od turnoff endin </CsInstruments> <CsScore> i 1 0 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ifthen.csd.xml0000644000000000000000000000566412262561504020143 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ifthen.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Get the note value from the fourth p-field. knote = p4 ; Does the user want a low note? if (knote == 0) then kcps = 220 ; Does the user want a middle note? elseif (knote == 1) then kcps = 440 ; Does the user want a high note? elseif (knote == 2) then kcps = 880 endif ; Create the note. kamp init 25000 ifn = 1 a1 oscili kamp, kcps, ifn out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; p4: 0=low note, 1=middle note, 2=high note. ; Play Instrument #1 for one second, low note. i 1 0 1 0 ; Play Instrument #1 for one second, middle note. i 1 1 1 1 ; Play Instrument #1 for one second, high note. i 1 2 1 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKFMVoices.csd.xml0000644000000000000000000000374712262561504020723 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKFMVoices.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;vowel kv2 line p7, p3, p8 ;specral tilt asig STKFMVoices cpspch(ifrq), 1, 2, kv1, 4, kv2, 11, 10, 1, 10, 128, 50 asig = asig * 4 ;amplify outs asig, asig endin </CsInstruments> <CsScore> i 1 0 5 5.00 10 120 0 0 i 1 + 2 8.00 80 82 127 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/jacko.csd.xml0000644000000000000000000001276012262561504017750 0ustar rootroot <CsoundSynthesizer> <CsOptions> csound -m255 -M0 -+rtmidi=null -RWf --midi-key=4 --midi-velocity=5 -o jacko_test.wav </CsOptions> <CsInstruments> ;;;; ;;;; NOTE: this csd must be run after starting "aeolus -t". ;;;; sr = 48000 ; The control rate must be BOTH a power of 2 (for Jack) ; AND go evenly into sr. This is about the only one that works! ksmps = 128 nchnls = 2 0dbfs = 1 JackoInit "default", "csound" ; To use ALSA midi ports, use "jackd -Xseq" ; and use "jack_lsp -A -c" or aliases from JackInfo, ; probably together with information from the sequencer, ; to figure out the damn port names. ; JackoMidiInConnect "alsa_pcm:in-131-0-Master", "midiin" JackoAudioInConnect "aeolus:out.L", "leftin" JackoAudioInConnect "aeolus:out.R", "rightin" JackoMidiOutConnect "midiout", "aeolus:Midi/in" ; Note that Jack enables audio to be output to a regular ; Csound soundfile and, at the same time, to a sound ; card in real time to the system client via Jack. JackoAudioOutConnect "leftout", "system:playback_1" JackoAudioOutConnect "rightout", "system:playback_2" JackoInfo ; Turning freewheeling on seems automatically ; to turn system playback off. This is good! JackoFreewheel 1 JackoOn alwayson "jackin" instr 1 ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ichannel = p1 - 1 itime = p2 iduration = p3 ikey = p4 ivelocity = p5 JackoNoteOut "midiout", ichannel, ikey, ivelocity print itime, iduration, ichannel, ikey, ivelocity endin instr jackin ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; JackoTransport 3, 1.0 aleft JackoAudioIn "leftin" aright JackoAudioIn "rightin" ; Aeolus uses MIDI controller 98 to control stops. ; Only 1 data value byte is used, not the 2 data ; bytes often used with NRPNs. ; The format for control mode is 01mm0ggg: ; mm 10 to set stops, 0, ggg group (or Division, 0 based). ; The format for stop selection is 000bbbbb: ; bbbbb for button number (0 based). ; Mode to enable stops for Divison I: b1100010 (98 ; [this controller VALUE is a pure coincidence]). JackoMidiOut "midiout", 176, 0, 98, 98 ; Stops: Principal 8 (0), Principal 4 (1) , Flote 8 (8) , Flote 2 (10) JackoMidiOut "midiout", 176, 0, 98, 0 JackoMidiOut "midiout", 176, 0, 98, 1 JackoMidiOut "midiout", 176, 0, 98, 8 JackoMidiOut "midiout", 176, 0, 98, 10 ; Sends audio coming in from Aeolus out ; not only to the Jack system out (sound card), ; but also to the output soundfile. ; Note that in freewheeling mode, "leftout" ; and "rightout" simply go silent. JackoAudioOut "leftout", aleft JackoAudioOut "rightout", aright outs aright, aleft endin </CsInstruments> <CsScore> f 0 30 i 1 1 30 60 60 i 1 2 30 64 60 i 1 3 30 71 60 e 2 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/greaterequal.csd.xml0000644000000000000000000000460012262561504021334 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o greaterequal.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Get the 4th p-field from the score. k1 = p4 ; Is it greater than or equal to 3? (1 = true, 0 = false) k2 = (p4 >= 3 ? 1 : 0) ; Print the values of k1 and k2. printks "k1 = %f, k2 = %f\\n", 1, k1, k2 endin </CsInstruments> <CsScore> ; Call Instrument #1 with a p4 = 2. i 1 0 0.5 2 ; Call Instrument #1 with a p4 = 3. i 1 1 0.5 3 ; Call Instrument #1 with a p4 = 4. i 1 2 0.5 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/portk.csd.xml0000644000000000000000000000452112262561504020014 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ; -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o portk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 1 ;Example by Andres Cabrera 2007 FLpanel "Slider", 650, 140, 50, 50 gkval1, gislider1 FLslider "Watch me", 0, 127, 0, 5, -1, 580, 30, 25, 20 gkval2, gislider2 FLslider "Move me", 0, 127, 0, 5, -1, 580, 30, 25, 80 gkhtim, gislider3 FLslider "khtim", 0.1, 1, 0, 6, -1, 30, 100, 610, 10 FLpanelEnd FLrun FLsetVal_i 0.1, gislider3 ;set initial time to 0.1 instr 1 kval portk gkval2, gkhtim ; take the value of slider 2 and apply portamento FLsetVal 1, kval, gislider1 ;set the value of slider 1 to kval endin </CsInstruments> <CsScore> ; Play Instrument #1 for one minute. i 1 0 60 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lowres.csd.xml0000644000000000000000000000411612262561504020170 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lowres.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kres = p4 asig vco .2, 220, 1 ;sawtooth kcut line 1000, p3, 10 ;note: kcut is not in Hz as lowres asig, kcut, kres ;note: kres is not in dB aout balance as, asig ;avoid very loud sounds outs aout, aout endin </CsInstruments> <CsScore> ; a sine f 1 0 16384 10 1 i 1 0 4 3 i 1 + 4 30 i 1 + 4 60 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pan2.csd.xml0000644000000000000000000000343512262561504017520 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pan2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kline line 0, p3, 1 ; straight line ain oscili .6, 440, giSine ; audio signal.. aL,aR pan2 ain, kline ; sent across image outs aL, aR endin </CsInstruments> <CsScore> i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dam.csd.xml0000644000000000000000000000416412262561504017421 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dam.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;normal audio asig diskin2 "beats.wav", 1, 0, 1 outs asig, asig endin instr 2 ; compressed audio kthreshold = 0.2 icomp1 = 0.8 icomp2 = 0.2 irtime = 0.01 iftime = 0.5 asig diskin2 "beats.wav", 1, 0, 1 asig dam asig, kthreshold, icomp1, icomp2, irtime, iftime outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 i 2 2.5 8.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vmultv.csd.xml0000644000000000000000000000567112262561504020221 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ifn2 = p5 ielements = p6 idstoffset = p7 isrcoffset = p8 kval init 25 vmultv ifn1, ifn2, ielements, idstoffset, isrcoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 16 17 f 2 0 16 -7 1 16 2 i2 0.0 0.2 1 i2 0.2 0.2 2 i1 0.4 0.01 1 2 5 3 8 i2 0.8 0.2 1 i1 1.0 0.01 1 2 5 10 -2 i2 1.2 0.2 1 i1 1.4 0.01 1 2 8 14 0 i2 1.6 0.2 1 i1 1.8 0.01 1 2 8 0 14 i2 2.0 0.2 1 i1 2.2 0.002 1 1 8 5 2 i2 2.4 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cossegb.csd.xml0000644000000000000000000000475712262561504020315 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o linseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kcps = cpspch(p4) kenv cossegb 0, 0.25, 1, 1, 0 asig poscil kenv, kcps, giSine outs asig, asig endin instr 2 ; scaling to duration kcps = cpspch(p4) kenv cossegb 0, p3*0.25, 1, p3, 0 asig poscil kenv, kcps, giSine outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 7.00 ; = 1 sec, p3 fits exactly i 1 2 2 7.00 ; = 2 sec, p3 truncated at 1 sec i 2 4 1 7.00 ; scales to duration i 2 6 2 7.00 ; of p3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zkmod.csd.xml0000644000000000000000000000720512262561504020003 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zkmod.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Initialize the ZAK space. ; Create 2 a-rate variables and 2 k-rate variables. zakinit 2, 2 ; Instrument #1 -- a signal with jitter. instr 1 ; Generate a k-rate signal goes from 30 to 2,000. kline line 30, p3, 2000 ; Add the signal into zk variable #1. zkw kline, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Create a k-rate signal modulated the jitter opcode. kamp init 20 kcpsmin init 40 kcpsmax init 60 kjtr jitter kamp, kcpsmin, kcpsmax ; Get the frequency values from zk variable #1. kfreq zkr 1 ; Add the the frequency values in zk variable #1 to ; the jitter signal. kjfreq zkmod kjtr, 1 ; Use a simple sine waveform for the left speaker. aleft oscil 20000, kfreq, 1 ; Use a sine waveform with jitter for the right speaker. aright oscil 20000, kjfreq, 1 ; Generate the audio output. outs aleft, aright ; Clear the zk variables, prepare them for ; another pass. zkcl 0, 2 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 ; Play Instrument #2 for 2 seconds. i 2 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/x.csd.xml0000644000000000000000000000522112262561504017122 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o x.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .8*aenv, p4, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave s ;first section i 1 0 2 110 i 1 3 2 220 i 1 6 2 440 i 1 9 2 880 s ;second section x ;skip the rest i 1 0 2 110 ;of this section i 1 3 2 220 i 1 6 2 440 i 1 9 2 880 s ;but continue with this one i 1 0 2 880 i 1 3 2 440 i 1 6 2 220 i 1 9 2 110 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midic7.csd.xml0000644000000000000000000000436312262561504020035 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midic7.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; This example expects MIDI controller input on channel 1 ; run, play a note and move your midi controller 7 to see result imax = 1 imin = 0 ichan = 1 ictlno= 7 ; = midi volume kamp midic7 ictlno, imin, imax printk2 kamp asig oscili kamp, 220, 1 outs asig, asig endin </CsInstruments> <CsScore> ; no score events allowed f 0 20 ;20 sec. for real-time MIDI events f 1 0 4096 10 1 ;sine wave e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/oscili.csd.xml0000644000000000000000000000555412262561504020146 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscili.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = .6 kcps = 440 ifn = p4 asig oscil kamp, kcps, ifn outs asig,asig endin instr 2 kamp = .6 kcps = 440 ifn = p4 asig oscili kamp, kcps, ifn outs asig,asig endin </CsInstruments> <CsScore> f1 0 128 10 1 ; Sine with a small amount of data f2 0 128 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth with a small amount of data f3 0 128 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square with a small amount of data f4 0 128 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse with a small amount of data i 1 0 2 1 i 2 3 2 1 i 1 6 2 2 i 2 9 2 2 i 1 12 2 3 i 2 15 2 3 i 1 18 2 4 i 2 21 2 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/osciln.csd.xml0000644000000000000000000000560712262561504020152 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o osciln.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gione ftgen 1, 0, 1024, 7, 0, 1, 1, 1024, 0 gitwo ftgen 2, 0, 1024, 7, 0, 512, 1, 512, 0 instr 1 ;very simple waveguide system ifn = p4 ipitch = p5 itimes = p6 iperiod = 1000/ipitch afeed init 0 aimpl osciln 1, ipitch, ifn, itimes ;use as excitation signal arefl tone aimpl + afeed, 4000 aout atone arefl, 5000 afeed vdelay arefl, iperiod, 10 outs aout*3, aout*3 endin </CsInstruments> <CsScore> i 1 0 4 1 110 1 ;use different tables, i 1 5 4 2 110 1 ;& different pitch i 1 10 4 1 110 10 ;& different number of times the table is read i 1 15 4 2 110 10 i 1 20 6 1 880 1 i 1 25 3 2 880 1 i 1 30 3 1 880 10 i 1 35 3 2 880 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpsmidib.csd.xml0000644000000000000000000000334512262561504020452 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M0 ;;;RT audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpsmidi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; move pitch bend wheel while you play kcps cpsmidib asig oscil 0.6, kcps, 1 printk2 kcps outs asig, asig endin </CsInstruments> <CsScore> f0 20 ;sine wave. f 1 0 16384 10 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/product.csd.xml0000644000000000000000000000413512262561504020336 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o product.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisine ftgen 0, 0, 2^10, 10, 1 instr 1 a1 oscili 1, 10.0, gisine ;combine 3 sinusses a2 oscili 1, 1.0, gisine ;at different rates a3 oscili 1, 3.0, gisine ares product a1, a2, a3 ares = ares*10000 ;scale result and asig poscil .5, ares+110, gisine ;add to frequency outs asig, asig endin </CsInstruments> <CsScore> i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outc.csd.xml0000644000000000000000000000361012262561504017625 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 5 0dbfs = 1 instr 1 asig vco2 .05, 30 ; sawtooth waveform at low volume kcut line 100, p3, 30 ; Vary cutoff frequency kresonance = 7 inumlayer = 2 asig lowresx asig, kcut, kresonance, inumlayer ; output same sound to 5 channels outc asig,asig,asig,asig,asig endin </CsInstruments> <CsScore> i 1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vdivv.csd.xml0000644000000000000000000000567212262561504020023 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ifn2 = p5 ielements = p6 idstoffset = p7 isrcoffset = p8 kval init 25 vdivv ifn1, ifn2, ielements, idstoffset, isrcoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 15 16 f 2 0 16 -7 1 15 2 i2 0.0 0.2 1 i2 0.2 0.2 2 i1 0.4 0.01 1 2 5 3 8 i2 0.8 0.2 1 i1 1.0 0.01 1 2 5 10 -2 i2 1.2 0.2 1 i1 1.4 0.01 1 2 8 14 0 i2 1.6 0.2 1 i1 1.8 0.01 1 2 8 0 14 i2 2.0 0.2 1 i1 2.2 0.002 1 1 8 5 2 i2 2.4 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/trigseq.csd.xml0000644000000000000000000001524612262561504020341 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trigseq.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giTimes ftgen 91, 0, 128, -2, 1, 1/2, 1/2, 1/8, 1/8, 1/2,1/2, 1/16, 1/16, 1/16, 1/16, 1/16, 1/16, 1/16, 1/16; times giSeq ftgen 90, 0, 128, -2, 1, 2, .5, 3, .25, 4, .10, 5, .05, 6 ;** sequence amplitude and freq-ratio bins instr 1 icps init p4 iamp init .3 kloop init p5 initndx init p6 kloop2 init p7 initndx2 init p8 kdur init p9 iminTime init p10 imaxTime init p11 kampratio init 1 kfreqratio init 1 ktime_unit expseg iminTime,p3/8,iminTime,p3* 3/4,imaxTime,p3/8,imaxTime ;**ktrig seqtime ktime_unit, kstart, kloop, initndx, kfn_times ;ktrig seqtime 1/ktime_unit, 0, 15, 0, giTimes ktrig metro ktime_unit ;**** trigseq ktrig_in, kstart, kloop, initndx, kfn_values, kout1 [, kout2, kout3, ...., koutN] trigseq ktrig, 0, kloop2,initndx2, giSeq, kampratio, kfreqratio ;atrig = ktrig*10000 schedkwhen ktrig, -1, -1, 3, 0, kdur, kampratio*iamp, kfreqratio*icps ; schedkwhen ktrig, -1, -1, 2, 0, ktrig, kampratio*iamp, kfreqratio*icps endin instr 2 icps init p4 iamp init 20000 kloop init p5 initndx init p6 kloop2 init p7 initndx2 init p8 kdur init p9 iminTime init p10 imaxTime init p11 kampratio init 1 kfreqratio init 1 ktime_unit expseg iminTime,p3/8,iminTime,p3* 3/4,imaxTime,p3/8,imaxTime ;**ktrig seqtime ktime_unit, kstart, kloop, initndx, kfn_times ktrig seqtime 1/ktime_unit, 0, 15, 0, giTimes ;ktrig metro ktime_unit ;**** trigseq ktrig_in, kstart, kloop, initndx, kfn_values, kout1 [, kout2, kout3, ...., koutN] trigseq ktrig, 0, kloop2, initndx2, giSeq, kampratio, kfreqratio printk2 ktrig ;atrig = ktrig*10000 ; schedkwhen ktrig, -1, -1, 2, 0, kdur, kampratio*iamp, kfreqratio*icps schedkwhen ktrig, -1, -1, 3, 0, ktrig, kampratio*iamp, kfreqratio*icps endin instr 3 print p3 kenv expseg 1.04, p3,.04 a1 foscili p4*a(kenv-0.04), p5,1,1,kenv*5, 2 outs a1, a1 endin </CsInstruments> <CsScore> f2 0 8192 10 1 ; icps unused unused kloop2 initndx2 kdur iminTime imaxTime s i1 0 6 100 0 0 5 0 .2 3 15 i1 8 6 150 0 0 4 1 .1 4 30 i1 16 6 200 0 0 5 3 .25 8 50 i1 24 6 300 0 0 3 0 .1 1 30 i2 32 6 100 0 0 5 0 .2 1 1 i2 40 6 150 0 0 4 1 .1 .5 .5 i2 48 6 200 0 0 5 3 .25 3 .5 i2 56 6 300 0 0 5 0 .1 1 8 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vlinseg.csd.xml0000644000000000000000000000512312262561504020323 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac -B441 -b441 </CsOptions> <CsInstruments> sr=44100 ksmps=10 nchnls=2 gilen init 32 gitable1 ftgen 0, 0, gilen, 10, 1 gitable2 ftgen 0, 0, gilen, 10, 1 gitable3 ftgen 0, 0, gilen, -7, 30, gilen, 35 gitable4 ftgen 0, 0, gilen, -7, 400, gilen, 450 gitable5 ftgen 0, 0, gilen, -7, 5000, gilen, 5500 instr 1 vcopy gitable2, gitable1, gilen turnoff endin instr 2 vlinseg gitable2, 16, gitable3, 2, gitable4, 2, gitable5 endin instr 3 kcount init 0 if kcount < 16 then kval table kcount, gitable2 printk 0,kval kcount = kcount +1 else turnoff endif endin </CsInstruments> <CsScore> i1 0 1 s i2 0 10 i3 0 1 i3 1 1 i3 1.5 1 i3 2 1 i3 2.5 1 i3 3 1 i3 3.5 1 i3 4 1 i3 4.5 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLbutton.csd.xml0000644000000000000000000001372112262561504020414 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLbutton.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Using FLbuttons to create on screen controls for play, ; stop, fast forward and fast rewind of a sound file ; This example also makes use of a preset graphic for buttons. sr = 44100 kr = 44100 ksmps = 1 nchnls = 2 FLpanel "Buttons", 240, 400, 100, 100 ion = 0 ioff = 0 itype = 1 iwidth = 50 iheight = 50 ix = 10 iy = 10 iopcode = 0 istarttim = 0 idur = -1 ;Turn instruments on idefinitely ; Normal speed forwards gkplay, ihb1 FLbutton "@>", ion, ioff, itype, iwidth, iheight, ix, iy, iopcode, 1, istarttim, idur, 1 ; Stationary gkstop, ihb2 FLbutton "@square", ion,ioff, itype, iwidth, iheight, ix+55, iy, iopcode, 2, istarttim, idur ; Double speed backwards gkrew, ihb3 FLbutton "@<<", ion, ioff, itype, iwidth, iheight, ix + 110, iy, iopcode, 1, istarttim, idur, -2 ; Double speed forward gkff, ihb4 FLbutton "@>>", ion, ioff, itype, iwidth, iheight, ix+165, iy, iopcode, 1, istarttim, idur, 2 ; Type 1 gkt1, iht1 FLbutton "1-Normal Button", ion, ioff, 1, 200, 40, ix, iy + 65, -1 ; Type 2 gkt2, iht2 FLbutton "2-Light Button", ion, ioff, 2, 200, 40, ix, iy + 110, -1 ; Type 3 gkt3, iht3 FLbutton "3-Check Button", ion, ioff, 3, 200, 40, ix, iy + 155, -1 ; Type 4 gkt4, iht4 FLbutton "4-Round Button", ion, ioff, 4, 200, 40, ix, iy + 200, -1 ; Type 21 gkt5, iht5 FLbutton "21-Plastic Button", ion, ioff, 21, 200, 40, ix, iy + 245, -1 ; Type 22 gkt6, iht6 FLbutton "22-Plastic Light Button", ion, ioff, 22, 200, 40, ix, iy + 290, -1 ; Type 23 gkt7, iht7 FLbutton "23-Plastic Check Button", ion, ioff, 23, 200, 40, ix, iy + 335, -1 FLpanelEnd FLrun ; Ensure that only 1 instance of instr 1 ; plays even if the play button is clicked repeatedly insnum = 1 icount = 1 maxalloc insnum, icount instr 1 asig diskin2 "beats.wav", p4, 0, 1 outs asig, asig endin instr 2 turnoff2 1, 0, 0 ;Turn off instr 1 turnoff ;Turn off this instrument endin </CsInstruments> <CsScore> ; Real-time performance for 1 hour. f 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen27.csd.xml0000644000000000000000000000431612262561504017601 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen27.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 1, 0, 32768, 10, 1 gienv ftgen 2, 0, 1025, 27, 0, 0,200, 1, 400, -1, 513, 0 instr 1 kcps init 3/p3 ;play 3x over duration of note kndx phasor kcps ixmode = 1 ;normalize to 0-1 kval table kndx, gienv, ixmode kval = kval*100 ;scale 0-100 asig poscil 1, 220+kval, 1 ;add to 220 Hz outs asig, asig endin </CsInstruments> <CsScore> i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fmvoice.csd.xml0000644000000000000000000000525712262561504020314 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmvoice.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 110 kvowel = p4 ; p4 = vowel (0 - 64) ktilt = p5 kvibamt = 0.005 kvibrate = 6 asig fmvoice .5, kfreq, kvowel, ktilt, kvibamt, kvibrate outs asig, asig endin </CsInstruments> <CsScore> ; sine wave. f 1 0 16384 10 1 i 1 0 1 1 0 ; tilt=0 i 1 1 1 > . i 1 2 1 > . i 1 3 1 > . i 1 4 1 > . i 1 5 1 > . i 1 6 1 > . i 1 7 1 12 . i 1 10 1 1 90 ; tilt=90 i 1 11 1 > . i 1 12 1 > . i 1 13 1 > . i 1 14 1 > . i 1 15 1 > . i 1 16 1 > . i 1 17 1 12 . e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ampdb.csd.xml0000644000000000000000000000343112262561504017737 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ampdb.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 idb = p4 iamp = ampdb(idb) asig oscil iamp, 220, 1 print iamp outs asig, asig endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 1 50 i 1 + 1 90 i 1 + 1 68 i 1 + 1 80 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/leftbrace.csd.xml0000644000000000000000000000472612262561504020613 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o abs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> nchnls = 2 gaReverbSend init 0 ; a simple sine wave partial instr 1 idur = p3 iamp = p4 ifreq = p5 aenv linseg 0.0, 0.1*idur, iamp, 0.6*idur, iamp, 0.3*idur, 0.0 aosc oscili aenv, ifreq, 1 vincr gaReverbSend, aosc endin ; global reverb instrument instr 2 al, ar reverbsc gaReverbSend, gaReverbSend, 0.85, 12000 outs gaReverbSend+al, gaReverbSend+ar clear gaReverbSend endin </CsInstruments> <CsScore> f1 0 4096 10 1 { 4 CNT { 8 PARTIAL ; start time duration amplitude frequency i1 [0.5 * $CNT.] [1 + ($CNT * 0.2)] [500 + (~ * 200)] [800 + (200 * $CNT.) + ($PARTIAL. * 20)] } } i2 0 6 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftgen.csd.xml0000644000000000000000000000426612262561504017766 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftgen.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 gisine ftgen 1, 0, 16384, 10, 1 ;sine wave gisquare ftgen 2, 0, 16384, 10, 1, 0 , .33, 0, .2 , 0, .14, 0 , .11, 0, .09 ;odd harmonics gisaw ftgen 3, 0, 16384, 10, 0, .2, 0, .4, 0, .6, 0, .8, 0, 1, 0, .8, 0, .6, 0, .4, 0,.2 ;even harmonics instr 1 ifn = p4 asig poscil .6, 200, ifn outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 1 ;sine wave i 1 3 2 2 ;odd harmonics i 1 6 2 3 ;even harmonics e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsmix.csd.xml0000644000000000000000000000403512262561504020203 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsmix.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 1, 0, 2048, 10, 1 instr 1 asig1 diskin2 "fox.wav", 1 ;signal in 1 asig2 oscil .3, 100, gisin ;signal in 2 fsig1 pvsanal asig1,1024,256,1024,0 ;pvoc analysis fsig2 pvsanal asig2,1024,256,1024,0 ;of both signals fsall pvsmix fsig1, fsig2 asig pvsynth fsall outs asig, asig endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/equal.csd.xml0000644000000000000000000000333512262561504017766 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o equal.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 instr 1 ; Assign a value to the variable i1. i1 = 1234 ; Print the value of the i1 variable. print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/t.csd.xml0000644000000000000000000000505612262561504017124 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o t.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .8*aenv, p4, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave t 0 240 12 30 15 240 ;start tempo = 240 i 1 0 2 110 ;tempo = 240 i 1 3 2 220 ;slow down & i 1 6 2 440 ;slow down & i 1 9 2 880 ;slow down & i 1 12 2 110 ;slow down to 30 at 12 seconds i 1 15 2 220 ;speed up to 240 again i 1 18 2 440 ;stay at tempo 240 i 1 21 2 880 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lfo.csd.xml0000644000000000000000000000410512262561504017433 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lfo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 5 itype = p4 ;lfo type klfo line 0, p3, 20 al lfo klfo, kcps, itype asig poscil .5, 220+al, 1 outs asig, asig endin </CsInstruments> <CsScore> ; sine wave. f 1 0 32768 10 1 i 1 0 3 0 ;lfo = sine i 1 + 3 2 ;lfo = square i 1 + 3 5 ;lfo = saw-tooth down e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outs1.csd.xml0000644000000000000000000000356212262561504017734 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outs1.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig vco2 .01, 110 ; sawtooth waveform at low volume kcut line 60, p3, 300 ; Vary cutoff frequency kresonance = 3 inumlayer = 3 asig lowresx asig, kcut, kresonance, inumlayer outs1 asig ; output stereo channel 1 only endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pindex.csd.xml0000644000000000000000000000407212262561504020145 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc ; -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ;-o pindex.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;Example by Anthony Kozar Dec 2006 instr 1 inum pcount index init 1 loop1: ivalue pindex index printf_i "p%d = %f\n", 1, index, ivalue index = index + 1 if (index <= inum) igoto loop1 print inum endin </CsInstruments> <CsScore> i1 0 3 40 50 ; has 5 pfields i1 1 2 80 ; has 5 due to carry i1 2 1 40 50 60 70 ; has 7 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lorismorph.csd.xml0000644000000000000000000001263212262561504021055 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lorismorph.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; clarinet.sdif and meow.sdif can be found in /manual/examples ; Morph the partials in meow.sdif into the partials in clarinet.sdif over the duration of ; the sustained portion of the two tones (from .2 to 2.0 seconds in the meow, and from ; .5 to 2.1 seconds in the clarinet). The onset and decay portions in the morphed sound are ; specified by parameters p4 and p5, respectively. The morphing time is the time between the ; onset and the decay. The meow partials are shfited in pitch to match the pitch of the clarinet ; tone (D above middle C). instr 1 ionset = p4 idecay = p5 itmorph = p3 - (ionset + idecay) ipshift = cpspch(8.02)/cpspch(8.08) ktme linseg 0, ionset, .2, itmorph, 2.0, idecay, 2.1 ; meow time function, morph from .2 to 2.0 seconds ktcl linseg 0, ionset, .5, itmorph, 2.1, idecay, 2.3 ; clarinet time function, morph from .5 to 2.1 seconds kmurph linseg 0, ionset, 0, itmorph, 1, idecay, 1 lorisread ktme, "meow.sdif", 1, ipshift, 2, 1, .001 lorisread ktcl, "clarinet.sdif", 2, 1, 1, 1, .001 lorismorph 1, 2, 3, kmurph, kmurph, kmurph asig lorisplay 3, 1, 1, 1 outs asig, asig endin ; Morph the partials in clarinet.sdif into the partials in meow.sdif. The start and end times ; for the morph are specified by parameters p4 and p5, respectively. The morph occurs over the ; second of four pitches in each of the sounds, from .75 to 1.2 seconds in the flutter-tongued ; clarinet tone, and from 1.7 to 2.2 seconds in the cat's meow. Different morphing functions are ; used for the frequency and amplitude envelopes, so that the partial amplitudes make a faster ; transition from clarinet to cat than the frequencies. (The bandwidth envelopes use the same ; morphing function as the amplitudes.) instr 2 ionset = p4 imorph = p5 - p4 irelease = p3 - p5 ktclar linseg 0, ionset, .75, imorph, 1.2, irelease, 2.4 ktmeow linseg 0, ionset, 1.7, imorph, 2.2, irelease, 3.4 kmfreq linseg 0, ionset, 0, .75*imorph, .25, .25*imorph, 1, irelease, 1 kmamp linseg 0, ionset, 0, .75*imorph, .9, .25*imorph, 1, irelease, 1 lorisread ktclar, "clarinet.sdif", 1, 1, 1, 1, .001 lorisread ktmeow, "meow.sdif", 2, 1, 1, 1, .001 lorismorph 1, 2, 3, kmfreq, kmamp, kmamp asig lorisplay 3, 1, 1, 1 outs asig, asig endin </CsInstruments> <CsScore> ; strt dur onset decay i 1 0 3 .25 .15 i 1 + 1 .10 .10 i 1 + 6 1. 1. ; strt dur morph_start morph_end i 2 9 4 .75 2.75 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/toot10.csd.xml0000644000000000000000000000706712262561504020013 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> nchnls = 2 garvbsig init 0 instr 10 iattack = .01 irelease = .2 iwhite = 22050 idur = p3 iamp = p4 iswpstart = p5 isweepend = p6 ibndwidth = p7 ibalance = p8 ; 1 = left, .5 = center, 0 = right irvbgain = p9 kamp linen iamp, iattack, idur, irelease ksweep line iswpstart, idur, isweepend asig rand iwhite afilt reson asig, ksweep, ibndwidth arampsig = kamp * afilt outs arampsig * ibalance, arampsig * (1 - ibalance) garvbsig = garvbsig + arampsig * p9 endin instr 100 irvbtime = p4 asig reverb garvbsig, irvbtime outs asig, asig garvbsig = 0 endin </CsInstruments> <CsScore> ;ins strt dur rvbtime i100 0 15 1.1 i100 15 10 5 ;ins strt dur amp stsw ndsw bdw bal(0-1) rvsnd i10 0 2 .01 5000 500 20 .15 .1 i10 3 1 .01 1500 5000 30 .95 .1 i10 5 2 .01 850 1100 40 .45 .1 i10 8 2 .01 1100 8000 50 .05 .1 i10 8 .5 .01 5000 1000 30 .35 .2 i10 9 .5 .01 1000 8000 40 .75 .1 i10 11 .5 .01 500 2100 50 .14 .2 i10 12 .5 .01 2100 1220 75 .96 .1 i10 13 .5 .01 1700 3500 100 .45 .2 i10 15 5 .005 8000 800 60 .85 .1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sin.csd.xml0000644000000000000000000000712012262561504017444 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sin.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 isin1 = sin(0) ;sine of 0 is 0 isin2 = sin($M_PI_2) ;sine of pi/2 (1.5707...) is 1 isin3 = sin($M_PI) ;sine of pi (3.1415...) is 0 isin4 = sin($M_PI_2 * 3) ;sine of 3/2pi (4.7123...) is -1 isin5 = sin($M_PI * 2) ;sine of 2pi (6.2831...) is 0 isin6 = sin($M_PI * 4) ;sine of 4pi is also 0 print isin1, isin2, isin3, isin4, isin5, isin6 endin instr 2 ;sin used in panning, after an example from Hans Mikelson aout vco2 0.8, 220 ; sawtooth kpan linseg p4, p3, p5 ;0 = left, 1 = right kpan = kpan*$M_PI_2 ;range 0-1 becomes 0-pi/2 kpanl = cos(kpan) kpanr = sin(kpan) outs aout*kpanl, aout*kpanr endin </CsInstruments> <CsScore> i 1 0 0 i 2 0 5 0 1 ;move left to right i 2 5 5 1 0 ;move right to left e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mp3in.csd.xml0000644000000000000000000000311612262561504017702 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mp3in.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iskptim = .3 ibufsize = 64 ar1, ar2 mp3in "beats.mp3", iskptim, 0, 0, ibufsize outs ar1, ar2 endin </CsInstruments> <CsScore> i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKBowed.csd.xml0000644000000000000000000000432212262561504020276 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKBowed.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = p4 kv2 = p7 ;position on bow kv1 line p5, p3, p6 ;bow pressure kv4 line 0, p3, 7 ;depth of low-frequency oscillator asig STKBowed cpspch(ipch), 1, 2, kv1, 4, kv2, 11, 40, 1, kv4, 128, 100 asig = asig*2 ;amplify outs asig, asig endin </CsInstruments> <CsScore> i 1 0 15 6.00 20 100 115 i 1 17 3 7.00 120 0 0 i 1 21 3 7.09 120 0 30 i 1 21 4 7.03 50 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vdelayxq.csd.xml0000644000000000000000000000531012262561504020507 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelayxq.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 ims = .5 ;maximum delay time in seconds iws = 1024 ;window size adl = .5 aout1 diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav aout2 diskin2 "fox.wav", 1, 0, 1 ;loop fox.wav aout3 diskin2 "Church.wav", 1, 0, 1 ;loop Church.wav aout4 diskin2 "flute.aiff", 1, 0, 1 ;loop flute.aiff a2 poscil .1, .5, 1 ;make an LFO, 1 cycle per 2 seconds adl = a2 + ims/2 ;offset the LFO so that it is positive aout1, aout2, aout3, aout4 vdelayxq aout1, aout2, aout3, aout4, adl, ims, iws; Use the LFO to control delay time outq aout1, aout2, aout3, aout4 endin </CsInstruments> <CsScore> f1 0 8192 10 1 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpsxpch.csd.xml0000644000000000000000000000362012262561504020324 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cpsxpch.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a normal twelve-tone scale. ipch = 8.02 iequal = 12 irepeat = 2 ibase = 1.02197503906 icps cpsxpch ipch, iequal, irepeat, ibase print icps endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/imageopcodesdemo2.csd.xml0000644000000000000000000000717512262561504022253 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o imageopcodesdemo2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 48000 ksmps = 100 nchnls = 2 ;By Cesare Marilungo 2008 zakinit 10,1 ;Load the image - should be 512x512 pixels giimage imageload "test1.png" ;giimage imageload "test2.png" ;--try this too giimagew, giimageh imagesize giimage giwave ftgen 1, 0, 1024, 10, 1 gifrqs ftgen 2,0,512,-5, 1,512,10 giamps ftgen 3, 0, 512, 10, 1 instr 100 kindex = 0 icnt = giimageh kx_ linseg 0, p3, 1 kenv linseg 0, .2, 500, p3 - .4, 500, .2, 0 ; Read a column of pixels and store the red values ; inside the table 'giamps' loop: ky_ = kindex/giimageh ;Get the pixel color values at kx_, ky_ kred, kgreen, kblue imagegetpixel giimage, kx_, ky_ ;Write the red values inside the table 'giamps' tablew kred, kindex, giamps kindex = kindex+1 if (kindex < icnt) kgoto loop ; Use an oscillator bank (additive synthesis) to generate sound ; setting amplitudes for each partial according to the image asig adsynt kenv, 220, giwave, gifrqs, giamps, icnt, 2 outs asig, asig endin instr 101 ; Free memory used by the image imagefree giimage endin </CsInstruments> <CsScore> t 0 60 i100 1 20 i101 21 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sfplaym.csd.xml0000644000000000000000000000533412262561504020333 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfplaym.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf sfload "07AcousticGuitar.sf2" sfplist gisf sfpassign 100, gisf instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/7000 ;scale amplitude kfreq init 1 ;do not change freq from sf aout sfplaym ivel, inum, kamp*ivel, kfreq, 100 ;preset index = 100 outs aout, aout endin </CsInstruments> <CsScore> f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/multiplies.csd.xml0000644000000000000000000000375512262561504021054 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o multiplies.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kenv expon 0.01, p3, 1 aout poscil 0.8*kenv, 440, 1 ;multiply amplitude from 0 to 1 * 0.8 printks "base amplitude * rising expon output = 0.8 * %f\n", .1, kenv outs aout, aout endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ; sine wave i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/toot6.csd.xml0000644000000000000000000000770312262561504017735 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> instr 6 ifunc = p11 ; select the basic waveform irel = 0.01 ; set vibrato release time idel1 = p3 * p10 ; calculate initial delay (% of dur) isus = p3 - (idel1 + irel) ; calculate remaining duration iamp = ampdb(p4) iscale = iamp * .333 ; p4=amp inote = cpspch(p5) ; p5=freq k3 linseg 0, idel1, p9, isus, p9, irel, 0 ; p6=attack time k2 oscil k3, p8, 1 ; p7=release time k1 linen iscale, p6, p3, p7 ; p8=vib rate a3 oscil k1, inote*.999+k2, ifunc ; p9=vib depth a2 oscil k1, inote*1.001+k2, ifunc ; p10=vib delay (0-1) a1 oscil k1, inote+k2, ifunc out a1+a2+a3 endin </CsInstruments> <CsScore> f1 0 2048 10 1 ; Sine f2 0 2048 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth f3 0 2048 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square f4 0 2048 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse ;ins strt dur amp frq atk rel vbrt vbdpt vibdl waveform i6 0 2 86 8.00 .03 .7 6 9 .8 1 i6 3 2 86 8.02 .03 .7 6 9 .8 2 i6 6 2 86 8.04 .03 .7 6 9 .8 3 i6 9 3 86 8.05 .03 .7 6 9 .8 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/expon.csd.xml0000644000000000000000000000465712262561504020020 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o expon.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kpitch = p6 ;choose between expon or line if (kpitch == 0) then kpitch expon p4, p3, p5 elseif (kpitch == 1) then kpitch line p4, p3, p5 endif asig vco2 .6, kpitch outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 300 600 0 ;if p6=0 then expon is used i 1 3 2 300 600 1 ;if p6=1 then line is used i 1 6 2 600 1200 0 i 1 9 2 600 1200 1 i 1 12 2 1200 2400 0 i 1 15 2 1200 2400 1 i 1 18 2 2400 30 0 i 1 21 2 2400 30 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/nlfilt2.csd.xml0000644000000000000000000000517012262561504020230 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o nlfilt.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;unfiltered noise asig rand .7 outs asig, asig endin instr 2 ;filtered noise ka = p4 kb = p5 kd = p6 kC = p7 kL = p8 asig rand .3 afilt nlfilt2 asig, ka, kb, kd, kC, kL outs afilt, afilt endin </CsInstruments> <CsScore> i 1 0 2 ; unfiltered ; a b d C L i 2 2 2 0 0 0.8 0.5 20 ; non-linear effect i 2 + 2 .4 0.2 0.7 0.11 200 ; low=pass with non-linear i 2 + 2 0.35 -0.3 0.95 0.1 200 ; high-pass with non-linear i 2 + 2 0.7 -0.2 0.9 0.2 20 ; high-pass with non-linear e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lpreson-2.csd.xml0000644000000000000000000000422312262561504020475 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpreson-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; works with or without -a option when analyzing "fox.wav" from the manual ;both options sound a little different instr 1 ilen filelen "fox.wav" ;length of soundfile 1 prints "fox.wav = %f seconds\\n",ilen ktime line 0, p3, ilen krmsr,krmso,kerr,kcps lpread ktime,"fox_nopoles.lpc" asig diskin2 "flute.aiff", 1 aout lpreson asig ares balance aout, asig outs ares, ares endin </CsInstruments> <CsScore> i 1 0 2.8 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluidcomplex.csd.xml0000644000000000000000000001137612262561504021356 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ;Load external midi file from manual/examples -odac -T -F Anna.mid;;;realtime audio I/O and midifile in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fluidcomplex.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; Example by Istvan Varga ; disable triggering of instruments by MIDI events ichn = 1 lp1: massign ichn, 0 loop_le ichn, 1, 16, lp1 pgmassign 0, 0 ; initialize FluidSynth gifld fluidEngine gisf2 fluidLoad "sf_GMbank.sf2", gifld, 1 ; k-rate version of fluidProgramSelect opcode fluidProgramSelect_k, 0, kkkkk keng, kchn, ksf2, kbnk, kpre xin igoto skipInit doInit: fluidProgramSelect i(keng), i(kchn), i(ksf2), i(kbnk), i(kpre) reinit doInit rireturn skipInit: endop instr 1 ; initialize channels kchn init 1 if (kchn == 1) then lp2: fluidControl gifld, 192, kchn - 1, 0, 0 fluidControl gifld, 176, kchn - 1, 7, 100 fluidControl gifld, 176, kchn - 1, 10, 64 loop_le kchn, 1, 16, lp2 endif ; send any MIDI events received to FluidSynth nxt: kst, kch, kd1, kd2 midiin if (kst != 0) then if (kst != 192) then fluidControl gifld, kst, kch - 1, kd1, kd2 else fluidProgramSelect_k gifld, kch - 1, gisf2, 0, kd1 endif kgoto nxt endif ; get audio output from FluidSynth ivol init 3 ;a bit louder aL, aR fluidOut gifld outs aL*ivol, aR*ivol endin </CsInstruments> <CsScore> i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zar.csd.xml0000644000000000000000000000542212262561504017452 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zar.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Generate a simple sine waveform. asin oscil 20000, 440, 1 ; Send the sine waveform to za variable #1. zaw asin, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read za variable #1. a1 zar 1 ; Generate audio output. out a1 ; Clear the za variables, get them ready for ; another pass. zacl 0, 1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vcomb.csd.xml0000644000000000000000000001127312262561504017765 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc -M0 ;;;RT audio I/O with MIDI in </CsOptions> <CsInstruments> ; Example by Jonathan Murphy and Charles Gran 2007 sr = 44100 ksmps = 10 nchnls = 2 ; new, and important. Make sure that midi note events are only ; received by instruments that actually need them. ; turn default midi routing off massign 0, 0 ; route note events on channel 1 to instr 1 massign 1, 1 ; Define your midi controllers #define C1 #21# #define C2 #22# #define C3 #23# ; Initialize MIDI controllers initc7 1, $C1, 0.5 ;delay send initc7 1, $C2, 0.5 ;delay: time to zero initc7 1, $C3, 0.5 ;delay: rate gaosc init 0 ; Define an opcode to "smooth" the MIDI controller signal opcode smooth, k, k kin xin kport linseg 0, 0.0001, 0.01, 1, 0.01 kin portk kin, kport xout kin endop instr 1 ; Generate a sine wave at the frequency of the MIDI note that triggered the intrument ifqc cpsmidi iamp ampmidi 10000 aenv linenr iamp, .01, .1, .01 ;envelope a1 oscil aenv, ifqc, 1 ; All sound goes to the global variable gaosc gaosc = gaosc + a1 endin instr 198 ; ECHO kcmbsnd ctrl7 1, $C1, 0, 1 ;delay send ktime ctrl7 1, $C2, 0.01, 6 ;time loop fades out kloop ctrl7 1, $C3, 0.01, 1 ;loop speed ; Receive MIDI controller values and then smooth them kcmbsnd smooth kcmbsnd ktime smooth ktime kloop smooth kloop imaxlpt = 1 ;max loop time ; Create a variable reverberation (delay) of the gaosc signal acomb vcomb gaosc, ktime, kloop, imaxlpt, 1 aout = (acomb * kcmbsnd) + gaosc * (1 - kcmbsnd) outs aout, aout gaosc = 0 endin </CsInstruments> <CsScore> f1 0 16384 10 1 i198 0 10000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsdiskin.csd.xml0000644000000000000000000000351012262561504020664 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsdiskin.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; create a PVOC-EX (*.pvx) file with PVANAL first ktscale line 1, p3, .05 ;change speed fsigr pvsdiskin "fox.pvx", ktscale, 1 ;read PVOCEX file aout pvsynth fsigr ;resynthesise it outs aout, aout endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/stix.csd.xml0000644000000000000000000000332012262561504017640 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o stix.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 idamp = p4 ;vary damping amount asig stix .5, 0.01, 30, idamp outs asig, asig endin </CsInstruments> <CsScore> i1 0 1 .3 i1 + 1 > i1 + 1 > i1 + 1 .95 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pindex-2.csd.xml0000644000000000000000000001201512262561504020300 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pindex-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; This UDO returns a pfield value but returns 0 if it does not exist. opcode mypvalue, i, i index xin inum pcount if (index > inum) then iout = 0.0 else iout pindex index endif xout iout endop ; Envelope UDO that reads parameters from a flexible number of pfields ; Syntax: kenv flexlinseg ipstart ; ipstart is the first pfield of the envelope ; parameters. Reads remaining pfields (up to 21 of them). ; kenv is the output envelope. opcode flexlinseg, k, i ipstart xin iep1 mypvalue ipstart iep2 mypvalue ipstart + 1 iep3 mypvalue ipstart + 2 iep4 mypvalue ipstart + 3 iep5 mypvalue ipstart + 4 iep6 mypvalue ipstart + 5 iep7 mypvalue ipstart + 6 iep8 mypvalue ipstart + 7 iep9 mypvalue ipstart + 8 iepa mypvalue ipstart + 9 iepb mypvalue ipstart + 10 iepc mypvalue ipstart + 11 iepd mypvalue ipstart + 12 iepe mypvalue ipstart + 13 iepf mypvalue ipstart + 14 iepg mypvalue ipstart + 15 ieph mypvalue ipstart + 16 iepi mypvalue ipstart + 17 iepj mypvalue ipstart + 18 iepk mypvalue ipstart + 19 iepl mypvalue ipstart + 20 kenv linseg iep1, iep2, iep3, iep4, iep5, iep6, iep7, iep8, \ iep9, iepa, iepb, iepc, iepd, iepe, iepf, iepg, \ ieph, iepi, iepj, iepk, iepl xout kenv endop instr 1 ; This instrument only requires 3 pfields but can accept up to 24. ; (You will still get warnings about more than 3). kenv flexlinseg 4 ; envelope params start at p4 aout oscili kenv*.5, 256, 1 outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 i1 0 2 0.0 1.0 1.0 1.0 0.0 i1 2 2 0.0 0.1 1.0 1.0 1.0 0.1 0.0 i1 4 2 0.0 0.5 0.0 ; one problem is that "missing" pfields carry i1 6 2 0.0 0.5 0.0 ! ; now we can fix this problem with ! i1 8 10 0.0 3.0 1.0 0.3 0.1 0.3 1.0 0.3 0.1 0.3 1.0 0.3 0.1 0.8 0.9 5.0 0.0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lorisplay.csd.xml0000644000000000000000000001064012262561504020672 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lorisplay.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; clarinet.sdif can be found in /manual/examples ; Play the partials in clarinet.sdif from 0 to 3 sec with 1 ms fadetime ; and no frequency , amplitude, or bandwidth modification. instr 1 ktime linseg 0, p3, 3.0 ; linear time function from 0 to 3 seconds lorisread ktime, "clarinet.sdif", 1, 1, 1, 1, .001 kfrq = p4 ; pitch shifting asig lorisplay 1, kfrq, 1, 1 outs asig, asig endin ; Play the partials in clarinet.sdif from 0 to 3 sec with 1 ms fadetime ; adding tuning and vibrato, increasing the "breathiness" (noisiness) and overall ; amplitude, and adding a highpass filter. instr 2 ktime linseg 0, p3, 3.0 ; linear time function from 0 to 3 seconds ; compute frequency scale for tuning ifscale = cpspch(p4)/cpspch(8.08) ; (original pitch was G#4) ; make a vibrato envelope kvenv linseg 0, p3/6, 0, p3/6, .02, p3/3, .02, p3/6, 0, p3/6, 0 kvib oscil kvenv, 4, 1 ; table 1, sinusoid kbwenv linseg 1, p3/6, 1, p3/6, 100, 2*p3/3, 100 ;lots of noise lorisread ktime, "clarinet.sdif", 1, 1, 1, 1, .001 a1 lorisplay 1, ifscale+kvib, 2, kbwenv asig atone a1, 1000 ; highpass filter, cutoff 1000 Hz outs asig, asig endin </CsInstruments> <CsScore> ; a sinusoid f 1 0 4096 10 1 i 1 0 3 1.2 ; shifted up i 1 + 1 1.5 i 1 + 6 .5 ; shifted down s ; strt dur ptch i 2 1 3 8.08 i 2 3.5 1 8.04 i 2 4 6 8.00 i 2 4 6 8.07 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLtabs.csd.xml0000644000000000000000000001016412262561504020030 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLtabs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; A single oscillator with frequency, amplitude and ; panning controls on separate file tab cards sr = 44100 kr = 441 ksmps = 100 nchnls = 2 FLpanel "Tabs", 300, 350, 100, 100 itabswidth = 280 itabsheight = 330 ix = 5 iy = 5 FLtabs itabswidth,itabsheight, ix,iy itab1width = 280 itab1height = 300 itab1x = 10 itab1y = 40 FLgroup "Tab 1", itab1width, itab1height, itab1x, itab1y gkfreq, i1 FLknob "Frequency", 200, 5000, -1, 1, -1, 70, 70, 130 FLsetVal_i 400, i1 FLgroupEnd itab2width = 280 itab2height = 300 itab2x = 10 itab2y = 40 FLgroup "Tab 2", itab2width, itab2height, itab2x, itab2y gkamp, i2 FLknob "Amplitude", 0, 15000, 0, 1, -1, 70, 70, 130 FLsetVal_i 15000, i2 FLgroupEnd itab3width = 280 itab3height = 300 itab3x = 10 itab3y = 40 FLgroup "Tab 3", itab3width, itab3height, itab3x, itab3y gkpan, i3 FLknob "Pan position", 0, 1, 0, 1, -1, 70, 70, 130 FLsetVal_i 0.5, i3 FLgroupEnd FLtabsEnd FLpanelEnd ; Run the widget thread! FLrun instr 1 ifn = 1 asig oscili gkamp, gkfreq, ifn outs asig*(1-gkpan), asig*gkpan endin </CsInstruments> <CsScore> ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sndwarp.csd.xml0000644000000000000000000000520112262561504020327 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sndwarp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ktimewarp line 0, p3, 2.7 ;length of "fox.wav" kresample init 1 ;do not change pitch ibeg = 0 ;start at beginning iwsize = 4410 ;window size in samples with irandw = 882 ;bandwidth of a random number generator itimemode = 1 ;ktimewarp is "time" pointer ioverlap = p4 asig sndwarp .5, ktimewarp, kresample, 1, ibeg, iwsize, irandw, ioverlap, 2, itimemode outs asig, asig endin </CsInstruments> <CsScore> f 1 0 131072 1 "fox.wav" 0 0 0 ; audio file f 2 0 1024 9 0.5 1 0 ; half of a sine wave i 1 0 7 2 ;different overlaps i 1 + 7 5 i 1 + 7 15 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/round.csd.xml0000644000000000000000000000404612262561504020006 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o round.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idiv init 1 loop: inumber = 9 i1 = inumber / idiv iro = round(i1) print inumber, idiv, iro ;print number / idiv = result using round idiv = idiv + 1 if (idiv <= 10) igoto loop endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tableng.csd.xml0000644000000000000000000000357312262561504020277 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tableng.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifn1 ftgen 1, 0, 0, 1, "flute.aiff", 0, 4, 0 ;deferred-size table instr 1 isize tableng 1 print isize andx phasor 1 / (isize / sr) ;play at correct pitch asig tab andx, 1, 1 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2.3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wgbow.csd.xml0000644000000000000000000000430712262561504020004 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o wgbow.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kpres = p4 ;pressure value krat = p5 ;position along string kvibf = 6.12723 kvib linseg 0, 0.5, 0, 1, 1, p3-0.5, 1 ; amplitude envelope for the vibrato. kvamp = kvib * 0.01 asig wgbow .7, 55, kpres, krat, kvibf, kvamp, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 2048 10 1 ;sine wave i 1 0 3 3 0.127236 i 1 + 3 5 0.127236 i 1 + 3 5 0.23 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/streson.csd.xml0000644000000000000000000000370512262561504020355 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o streson.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 asig diskin2 "fox.wav", 1, 0, 1 kfr = p4 ifdbgain = 0.90 astr streson asig, kfr, ifdbgain asig clip astr, 0, 1 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 20 i 1 + . > i 1 + . > i 1 + . > i 1 + . > i 1 + . > i 1 + . 1000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/moog.csd.xml0000644000000000000000000000444012262561504017616 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o moog.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = cpspch(p4) kfiltq = p5 kfiltrate = 0.0002 kvibf = 5 kvamp = .01 ;low volume is needed asig moog .15, kfreq, kfiltq, kfiltrate, kvibf, kvamp, 1, 2, 3 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 8192 1 "mandpluk.aiff" 0 0 0 f 2 0 256 1 "impuls20.aiff" 0 0 0 f 3 0 256 10 1 ; sine i 1 0 3 6.00 .1 i 1 + 3 6.05 .89 i 1 + 3 6.09 .50 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/spsend_quad.csd.xml0000644000000000000000000000616012262561504021164 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o space_quad.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 4 ga1 init 0 ga2 init 0 ga3 init 0 ga4 init 0 instr 1 asig diskin2 "beats.wav", 1 a1, a2, a3, a4 space asig, 0, 0, .1, p4, p5 ; take position values from p4, p5 ar1, ar2, ar3, ar4 spsend ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 a3 reverb2 ga3, 2.5, .5 a4 reverb2 ga4, 2.5, .5 outq a1, a2, a3, a4 ga1=0 ga2=0 ga3=0 ga4=0 endin </CsInstruments> <CsScore> ;place the sound in the left speaker and near i1 0 1 -1 1 ;place the sound in the right speaker and far i1 1 1 45 45 ;place the sound equally between left and right and in the middle ground distance i1 2 1 0 12 i 99 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fillarray.csd.xml0000644000000000000000000000235012262561504020640 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kS[] fillarray 1,7,5 printk2 kS[0] printk2 kS[1] printk2 kS[2] endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftchnls.csd.xml0000644000000000000000000000461312262561504020320 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ;-odac ;;;realtime audio out -iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftchnls.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ichnls = ftchnls(p4) print ichnls if (ichnls == 1) then asigL loscil3 .8, 4, p4 asigR = asigL elseif (ichnls == 2) then asigL, asigR loscil3 .8, 4, p4 ;safety precaution if not mono or stereo else asigL = 0 asigR = 0 endif outs asigL, asigR endin </CsInstruments> <CsScore> f 1 0 0 1 "mary.wav" 0 0 0 f 2 0 0 1 "kickroll.wav" 0 0 0 i 1 0 3 1 ;mono file i 1 + 2 2 ;stereo file e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tableikt.csd.xml0000644000000000000000000001000612262561504020447 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tableikt.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ilowfn = p4 ;lowest ftable wave ihighfn = p5 ;highest ftable wave kswpenv line 1, p3, 0 ;sweep envelope, calculate current table pair and interpolation amount inumtables = ihighfn - ilowfn ;1 less than number of tables kfn1 = int(kswpenv*inumtables) + ilowfn printks "play table no: %d\n", 1, kfn1 kfn2 = kfn1 + 1 kinterp = frac(kswpenv*inumtables) ixmode = 1 ;read tables with phasor aphase phasor 40 asig tableikt aphase, kfn1, ixmode ;normalized index if kswpenv == 1.0 kgoto skipfn2 ;if kfn1 is last table, there is no kfn2 asig2 tableikt aphase, kfn2, ixmode skipfn2: amix ntrpol asig, asig2, kinterp ;interpolate between tables and output outs amix*.5, amix*.5 endin </CsInstruments> <CsScore> f 1 0 16384 10 1 f 2 0 16384 10 1 .5 f 3 0 16384 1 "fox.wav" 0 0 0 ;a sample f 4 0 16384 10 1 .5 .3 .25 .2 .16 .14 .125 .111 ;sawtooth f 5 0 16384 10 1 .4 .3 .25 .2 f 6 0 16384 10 1 .3 .3 .25 .2 .16 f 7 0 16384 10 1 1 1 1 .7 .5 .3 .1 ;pulse f 8 0 16384 1 "beats.wav" 0 0 0 ;a sample i 1 0 10 1 8 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/strfromurl.csd.xml0000644000000000000000000000363612262561504021102 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o strget.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 #include "http://codemist.co.uk/jpff/test.in" instr 2 Sfile strget "http://codemist.co.uk/jpff/test.in" prints Sfile endin </CsInstruments> <CsScore> i 1 0 1 i 2 + . e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/min.csd.xml0000644000000000000000000000441512262561504017442 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o min.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 k1 oscili 1, 10.0, 1 ;combine 3 sinusses k2 oscili 1, 1.0, 1 ;at different rates k3 oscili 1, 3.0, 1 kmin min k1, k2, k3 kmin = kmin*250 ;scale kmin printk2 kmin ;check the values aout vco2 .5, 220, 6 ;sawtooth asig moogvcf2 aout, 600+kmin, .5 ;change filter around 600 Hz outs asig, asig endin </CsInstruments> <CsScore> f1 0 32768 10 1 i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/filelen.csd.xml0000644000000000000000000000465512262561504020303 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ;-odac ;;;realtime audio out -iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o filelen.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; choose between mono or stereo file ilen filelen p4 ;calculate length of soundfile print ilen ichn filenchnls p4 ;check number of channels ;print ichn if (ichn == 1) then ;mono signal asig diskin2 p4, 1 outs asig, asig else ;stereo signal aL, aR diskin2 p4, .5, 0, 1 outs aL, aR endif endin </CsInstruments> <CsScore> i 1 0 3 "fox.wav" ;mono signal i 1 5 2 "kickroll.wav" ;stereo signal e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/loop_lt.csd.xml0000644000000000000000000000546112262561504020331 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loop_lt.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 gisine ftgen 0, 0, 2^10, 10, 1 instr 1 ;master instrument ininstr = 5 ;number of called instances indx = 0 loop: prints "play instance %d\\n", indx ipan random 0, 1 ifreq random 100, 1000 iamp = 1/ininstr event_i "i", 10, 0, p3, iamp, ifreq, ipan loop_lt indx, 1, ininstr, loop endin instr 10 ipeak random 0, 1 ;where is the envelope peak asig poscil3 p4, p5, gisine aenv transeg 0, p3*ipeak, 6, 1, p3-p3*ipeak, -6, 0 aL,aR pan2 asig*aenv, p6 outs aL, aR endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/r.csd.xml0000644000000000000000000000567712262561504017133 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o r.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; The score's p4 parameter has the number of repeats. kreps = p4 ; The score's p5 parameter has our note's frequency. kcps = p5 ; Print the number of repeats. printks "Repeated %i time(s).\\n", 1, kreps ; Generate a nice beep. a1 oscil 20000, kcps, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; We'll repeat this section 6 times. Each time it ; is repeated, its macro REPS_MACRO is incremented. r6 REPS_MACRO ; Play Instrument #1. ; p4 = the r statement's macro, REPS_MACRO. ; p5 = the frequency in cycles per second. i 1 00.10 00.10 $REPS_MACRO 1760 i 1 00.30 00.10 $REPS_MACRO 880 i 1 00.50 00.10 $REPS_MACRO 440 i 1 00.70 00.10 $REPS_MACRO 220 ; Marks the end of the section. s e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vexpv.csd.xml0000644000000000000000000000567112262561504020034 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ifn2 = p5 ielements = p6 idstoffset = p7 isrcoffset = p8 kval init 25 vexpv ifn1, ifn2, ielements, idstoffset, isrcoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 16 17 f 2 0 16 -7 0 16 1 i2 0.0 0.2 1 i2 0.2 0.2 2 i1 0.4 0.01 1 2 5 3 8 i2 0.8 0.2 1 i1 1.0 0.01 1 2 5 10 -2 i2 1.2 0.2 1 i1 1.4 0.01 1 2 8 14 0 i2 1.6 0.2 1 i1 1.8 0.002 1 2 8 0 14 i2 2.0 0.2 1 i1 2.2 0.002 1 1 8 5 2 i2 2.4 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLjoy.csd.xml0000644000000000000000000001046012262561504017677 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLjoy.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Demonstration of the flpanel opcode ; Horizontal click-dragging controls the frequency of the oscillator ; Vertical click-dragging controls the amplitude of the oscillator sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "X Y Panel", 900, 400, 50, 50 ; Minimum value output by x movement (frequency) iminx = 200 ; Maximum value output by x movement (frequency) imaxx = 5000 ; Minimum value output by y movement (amplitude) iminy = 0 ; Maximum value output by y movement (amplitude) imaxy = 15000 ; Logarithmic change in x direction iexpx = -1 ; Linear change in y direction iexpy = 0 ; Display handle x direction (-1=not used) idispx = -1 ; Display handle y direction (-1=not used) idispy = -1 ; Width of the x y panel in pixels iwidth = 800 ; Height of the x y panel in pixels iheight = 300 ; Distance of the left edge of the x y panel from ; the left edge of the panel ix = 50 ; Distance of the top edge of the x y ; panel from the top edge of the panel iy = 50 gkfreqx, gkampy, ihandlex, ihandley FLjoy "X - Frequency Y - Amplitude", iminx, imaxx, iminy, imaxy, iexpx, iexpy, idispx, idispy, iwidth, iheight, ix, iy ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 ifn = 1 asig oscili gkampy, gkfreqx, ifn out asig endin </CsInstruments> <CsScore> ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpstmid.csd.xml0000644000000000000000000000453012262561504020320 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M0 ;;;RT audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpstmid.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; after an example from Kevin Conder sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; Table #1, a normal 12-tone equal temperament scale. ; numgrades = 12 (twelve tones) ; interval = 2 (one octave) ; basefreq = 261.659 (Middle C) ; basekeymidi = 60 (Middle C) gitemp ftgen 1, 0, 64, -2, 12, 2, 261.659, 60, 1.00, \ 1.059, 1.122, 1.189, 1.260, 1.335, 1.414, \ 1.498, 1.588, 1.682, 1.782, 1.888, 2.000 instr 1 ifn = 1 icps cpstmid ifn print icps asig oscil 0.6, icps, 2 outs asig, asig endin </CsInstruments> <CsScore> f 0 20 ;sine wave. f 2 0 16384 10 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/alwayson.csd.xml0000644000000000000000000002357212262561504020521 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-Wfo alwayson.wav </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 ksmps = 32 nchnls = 2 ; Connect up instruments and effects to create the signal flow graph. connect "SimpleSine", "leftout", "Reverberator", "leftin" connect "SimpleSine", "rightout", "Reverberator", "rightin" connect "Moogy", "leftout", "Reverberator", "leftin" connect "Moogy", "rightout", "Reverberator", "rightin" connect "Reverberator", "leftout", "Compressor", "leftin" connect "Reverberator", "rightout", "Compressor", "rightin" connect "Compressor", "leftout", "Soundfile", "leftin" connect "Compressor", "rightout", "Soundfile", "rightin" ; Turn on the "effect" units in the signal flow graph. alwayson "Reverberator", 0.91, 12000 alwayson "Compressor" alwayson "Soundfile" ; Define instruments and effects in order of signal flow. instr SimpleSine ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Default values: p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 pset 0, 0, 10, 0, 0, 0, 0.5 iattack = 0.015 idecay = 0.07 isustain = p3 irelease = 0.3 p3 = iattack + idecay + isustain + irelease adamping linsegr 0.0, iattack, 1.0, idecay + isustain, 1.0, irelease, 0.0 iHz = cpsmidinn(p4) ; Rescale MIDI velocity range to a musically usable range of dB. iamplitude = ampdb(p5 / 127 * 15.0 + 60.0) ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. icosine ftgenonce 0, 0, 65537, 11, 1 aoscili oscili iamplitude, iHz, icosine aadsr madsr iattack, idecay, 0.6, irelease asignal = aoscili * aadsr aleft, aright pan2 asignal, p7 ; Stereo audio output to be routed in the orchestra header. outleta "leftout", aleft outleta "rightout", aright endin instr Moogy ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Default values: p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 pset 0, 0, 10, 0, 0, 0, 0.5 iattack = 0.003 isustain = p3 irelease = 0.05 p3 = iattack + isustain + irelease adamping linsegr 0.0, iattack, 1.0, isustain, 1.0, irelease, 0.0 iHz = cpsmidinn(p4) ; Rescale MIDI velocity range to a musically usable range of dB. iamplitude = ampdb(p5 / 127 * 20.0 + 60.0) print iHz, iamplitude ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. isine ftgenonce 0, 0, 65537, 10, 1 asignal vco iamplitude, iHz, 1, 0.5, isine kfco line 2000, p3, 200 krez = 0.8 asignal moogvcf asignal, kfco, krez, 100000 asignal = asignal * adamping aleft, aright pan2 asignal, p7 ; Stereo audio output to be routed in the orchestra header. outleta "leftout", aleft outleta "rightout", aright endin instr Reverberator ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" idelay = p4 icutoff = p5 aleft, aright reverbsc aleftin, arightin, idelay, icutoff ; Stereo output. outleta "leftout", aleft outleta "rightout", aright endin instr Compressor ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" kthreshold = 25000 icomp1 = 0.5 icomp2 = 0.763 irtime = 0.1 iftime = 0.1 aleftout dam aleftin, kthreshold, icomp1, icomp2, irtime, iftime arightout dam arightin, kthreshold, icomp1, icomp2, irtime, iftime ; Stereo output. outleta "leftout", aleftout outleta "rightout", arightout endin instr Soundfile ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" outs aleftin, arightin endin </CsInstruments> <CsScore> ; It is not necessary to activate "effects" or create f-tables in the score! ; Overlapping notes create new instances of instruments with proper connections. i "SimpleSine" 1 5 60 85 i "SimpleSine" 2 5 64 80 i "Moogy" 3 5 67 75 i "Moogy" 4 5 71 70 ; 1 extra second after the performance e 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sfplist.csd.xml0000644000000000000000000000537412262561504020350 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out, virtual midi in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfplist.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf sfload "sf_GMbank.sf2" sfplist gisf ;list all presets gir sfpreset 125, 3, gisf, 0 ;preset = Car Pass instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/6000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay3 ivel, inum, kamp, kfreq, gir outs a1, a2 endin </CsInstruments> <CsScore> f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/repluck.csd.xml0000644000000000000000000000512712262561504020325 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o repluck.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 iplk = 0.75 kamp = .8 icps = 110 krefl = p4 kpick = p5 axcite oscil 1, 1, 1 asig repluck iplk, kamp, icps, kpick, krefl, axcite asig dcblock2 asig ;get rid of DC offset outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave. s i 1 0 1 0.95 0.75 ;sounds heavier (=p5) i 1 + 1 < i 1 + 1 < i 1 + 1 < i 1 + 10 0.6 s i 1 0 1 0.95 0.15 ;sounds softer (=p5) i 1 + 1 < i 1 + 1 < i 1 + 1 < i 1 + 10 0.6 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/powershape.csd.xml0000644000000000000000000000373112262561504021034 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o abs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> instr 1 imaxamp = 10000 kshapeamt line p5, p3, p6 aosc oscili 1.0, cpspch(p4), 1 aout powershape aosc, kshapeamt adeclick linseg 0.0, 0.01, 1.0, p3 - 0.06, 1.0, 0.05, 0.0 out aout * adeclick * imaxamp endin </CsInstruments> <CsScore> f1 0 32768 10 1 i1 0 1 7.00 0.000001 0.8 i1 + 0.5 7.02 0.01 1.0 i1 + . 7.05 0.5 1.0 i1 + . 7.07 4.0 1.0 i1 + . 7.09 1.0 10.0 i1 + 2 7.06 1.0 25.0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/oscilikt.csd.xml0000644000000000000000000000461412262561504020501 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o oscilikt.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a uni-polar (0-1) square wave. kamp1 init 1 kcps1 init 2 itype = 3 ksquare lfo kamp1, kcps1, itype ; Use the square wave to switch between Tables #1 and #2. kamp2 init 20000 kcps2 init 220 kfn = ksquare + 1 a1 oscilikt kamp2, kcps2, kfn out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine waveform. f 1 0 4096 10 0 1 ; Table #2: a sawtooth wave f 2 0 3 -2 1 0 -1 ; Play Instrument #1 for two seconds. i 1 0 2 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pchmidib.csd.xml0000644000000000000000000000341412262561504020434 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o pchmidib.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; This example expects MIDI note inputs on channel 1 i1 pchmidib print i1 endin </CsInstruments> <CsScore> ;Dummy f-table to give time for real-time MIDI events f 0 8000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/xtratim.csd.xml0000644000000000000000000000574412262561504020355 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in Silent MIDI in -odac -iadc -d -M0 ;;;realtime I/O </CsOptions> <CsInstruments> ;Simple usage of the xtratim opcode sr = 44100 ksmps = 10 nchnls = 2 ; sine wave for oscillators gisin ftgen 1, 0, 4096, 10, 1 instr 1 inum notnum icps cpsmidi iamp ampmidi 4000 ; ;------- complex envelope block ------ xtratim 1 ;extra-time, i.e. release dur krel init 0 krel release ;outputs release-stage flag (0 or 1 values) if (krel == 1) kgoto rel ;if in release-stage goto release section ; ;************ attack and sustain section *********** kmp1 linseg 0, .03, 1, .05, 1, .07, 0, .08, .5, 4, 1, 50, 1 kmp = kmp1*iamp kgoto done ; ;--------- release section -------- rel: kmp2 linseg 1, .3, .2, .7, 0 kmp = kmp1*kmp2*iamp done: ;------ a1 oscili kmp, icps, gisin outs a1, a1 endin </CsInstruments> <CsScore> f 0 3600 ;dummy table to wait for realtime MIDI events e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/igoto.csd.xml0000644000000000000000000000617412262561504020004 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o igoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Get the value of the 4th p-field from the score. iparam = p4 ; If iparam is 1 then play the high note. ; If not then play the low note. if (iparam == 1) igoto highnote igoto lownote highnote: ifreq = 880 goto playit lownote: ifreq = 440 goto playit playit: ; Print the values of iparam and ifreq. print iparam print ifreq a1 oscil 10000, ifreq, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; p4: 1 = high note, anything else = low note ; Play Instrument #1 for one second, a low note. i 1 0 1 0 ; Play a Instrument #1 for one second, a high note. i 1 1 1 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midictrl.csd.xml0000644000000000000000000000405312262561504020464 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midictrl.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .5 ips midictrl 9, 10, 500 ;controller 9 kenv madsr 0.5, 0, 1, 0.5 asig pluck kenv, icps, ips, 2, 1 ;change tone color outs asig, asig endin </CsInstruments> <CsScore> f 2 0 4096 10 1 ;sine wave ; no score events allowed f0 30 ;runs 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen17.csd.xml0000644000000000000000000000505212262561504017576 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M0 -+rtmidi=virtual ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen17.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 inote cpsmidi iveloc ampmidi .5 ictl midictrl 5 ;move slider of controller 5 to change ftable itab table ictl, 2 aout poscil iveloc, inote, itab outs aout, aout endin </CsInstruments> <CsScore> f 1 0 8193 10 1 f 2 0 128 -17 0 10 32 20 64 30 96 40 ;inhibit rescaling f 10 0 16384 10 1 ; Sine f 20 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111; Sawtooth f 30 0 16384 10 1 0 0.3 0 0.2 0 0.14 0 .111; Square f 40 0 16384 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse f 0 30 ;run for 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mpulse.csd.xml0000644000000000000000000000430312262561504020160 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o mpulse.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 gkfreq init 0.1 instr 1 kamp = 10000 a1 mpulse kamp, gkfreq out a1 endin instr 2 ; Assign the value of p4 to gkfreq gkfreq init p4 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 11 i 2 2 1 0.05 i 2 4 1 0.01 i 2 6 1 0.005 ; only last notes are audible i 2 8 1 0.003 i 2 10 1 0.002 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/chebyshevpoly.csd.xml0000644000000000000000000000613512262561504021544 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chebyshevpoly.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 ; time-varying mixture of first six harmonics instr 1 ; According to the GEN13 manual entry, ; the pattern + - - + + - - for the signs of ; the chebyshev coefficients has nice properties. ; these six lines control the relative powers of the harmonics k1 line 1.0, p3, 0.0 k2 line -0.5, p3, 0.0 k3 line -0.333, p3, -1.0 k4 line 0.0, p3, 0.5 k5 line 0.0, p3, 0.7 k6 line 0.0, p3, -1.0 ; play the sine wave at a frequency of 256 Hz with amplitude = 1.0 ax oscili 1, 256, 1 ; waveshape it ay chebyshevpoly ax, 0, k1, k2, k3, k4, k5, k6 ; avoid clicks, scale final amplitude, and output adeclick linseg 0.0, 0.05, 1.0, p3 - 0.1, 1.0, 0.05, 0.0 outs ay * adeclick * 10000, ay * adeclick * 10000 endin </CsInstruments> <CsScore> f1 0 32768 10 1 ; a sine wave i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sensekey.csd.xml0000644000000000000000000000316012262561504020501 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o sensekey.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 k1 sensekey printk2 k1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for thirty seconds. i 1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/initc7.csd.xml0000644000000000000000000000370212262561504020052 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M0 ;;;realtime audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if RT audio input is needed too </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; expects MIDI controller input on channel 1 ; run and move your midi controller to see result imax = 1 imin = 0 ichan = 1 ictlno = 7 initc7 1, 7, 1 ; start at max. volume kamp ctrl7 ichan, ictlno, imin, imax ; controller 7 printk2 kamp asig oscil kamp, 220, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 4096 10 1 i1 0 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mdelay.csd.xml0000644000000000000000000000443412262561504020133 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1;;;RT audio I/O with MIDI in </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 instr 1 ;Triggered by MIDI notes on channel 1 kstatus init 0 ifund notnum ivel veloc noteondur 1, ifund, ivel, 1 kstatus = kstatus + 1 idel1 = .2 idel2 = .4 idel3 = .6 idel4 = .8 ;make four delay lines mdelay kstatus,1,ifund+2, ivel,idel1 mdelay kstatus,1,ifund+4, ivel,idel2 mdelay kstatus,1,ifund+6, ivel,idel3 mdelay kstatus,1,ifund+8, ivel,idel4 endin </CsInstruments> <CsScore> ; Dummy ftable f 0 60 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sfpassign.csd.xml0000644000000000000000000001026712262561504020656 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out, virtual midi in ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfpassign.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;load two soundfonts gisf sfload "07AcousticGuitar.sf2" gir sfload "01hpschd.sf2" sfplist gisf sfplist gir sfpassign 0, gisf sfpassign 1, gir instr 1 ; play guitar from score and midi keyboard - preset index = 0 mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/5000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay3 ivel, inum, kamp*ivel, kfreq, 0 ;preset index = 0 outs a1, a2 endin instr 2 ; play harpsichord from score and midi keyboard - preset index = 1 mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/1000 ;scale amplitude kfreq init 1 ;do not change freq from sf a1,a2 sfplay3 ivel, inum, kamp*ivel, kfreq, 1 ;preset index = 1 outs a1, a2 endin </CsInstruments> <CsScore> f0 60 ; stay active for 1 minute i1 0 1 60 100 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 i2 5 1 60 100 i2 + 1 62 < i2 7 1 65 < i2 7 1 69 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midion_scale.csd.xml0000644000000000000000000000416512262561504021307 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1 ;;;RT audio I/O with MIDI in </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 instr 1 ; Triggered by MIDI notes on channel 1 ivel veloc krate = 8 iscale = 100 ;f ; Random sequence from table f100 krnd randh int(14),krate,-1 knote table abs(krnd),iscale ; Generates random notes from the scale on ftable 100 ; on channel 1 of csound's MIDI output midion 1,knote,ivel endin </CsInstruments> <CsScore> f100 0 32 -2 40 50 60 70 80 44 54 65 74 84 39 49 69 69 ; Dummy ftable f0 60 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/notnum_complex.csd.xml0000644000000000000000000000473312262561504021731 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in </CsOptions> <CsInstruments> sr = 44100 ksmps = 10 nchnls = 2 ; Set MIDI channel 1 to play instr 1. massign 1, 1 instr 1 ; Returns MIDI note number - an integer in range (0-127) iNum notnum ; Convert MIDI note number to Hz iHz = (440.0*exp(log(2.0)*((iNum)-69.0)/12.0)) ; Generate audio by indexing a table; fixed amplitude. aosc oscil 10000, iHz, 1 ; Since there is no enveloping, there will be clicks. outs aosc, aosc endin </CsInstruments> <CsScore> ; Generate a Sine-wave to be indexed at audio rate ; by the oscil opcode. f1 0 16384 10 1 ; Keep the score "open" for 1 hour so that MIDI ; notes can allocate new note events, arbitrarily. f0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/exp.csd.xml0000644000000000000000000000666712262561504017466 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o exp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 gisine ftgen 0, 0, 2^10, 10, 1 ;table for a sine wave instr 1 ;master instrument koct linseg 6, p3, 12 ; octave register straight rising from 6 to 12 kexp linseg 0, p3/3, 5, p3/3, 5, p3/3, 0 ;exponent goes from 0 to 5 and back kdens = exp(kexp) ;density is e to the power of kexp printks "Generated events per second: %d\n", 1, kdens ktrig metro kdens ;trigger single notes in kdens frequency if ktrig == 1 then ;call instr 10 for 1/kdens duration, .5 amplitude and koct register event "i", 10, 0, 1/kdens, .5, koct endif endin instr 10 ;performs one tone ioct rnd31 1, 0 ;random deviation maximum one octave plus/minus aenv transeg p4, p3, -6, 0 ;fast decaying envelope for p4 amplitude asin poscil aenv, cpsoct(p5+ioct), gisine ;sine for p5 octave register plus random deviation outs asin, asin endin </CsInstruments> <CsScore> i 1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/plltrack.csd.xml0000644000000000000000000000375612262561504020502 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o plltrack.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kd = p4 a1 diskin2 "fox.wav", 1,0,1 apitch, aloc plltrack a1, kd krms rms a1 krms port krms, 0.01 asig buzz krms, apitch, 10, 1 outs asig, asig ;mix in some dry signal as well endin </CsInstruments> <CsScore> f1 0 65536 10 1 ;sine wave i 1 0 6 0.1 i 1 7 6 0.3 ;more feedback e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/spat3d_stereo.csd.xml0000644000000000000000000001700512262561504021435 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o spat3d_stereo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Istvan Varga */ sr = 48000 kr = 1000 ksmps = 48 nchnls = 2 /* room parameters */ idep = 3 /* early reflection depth */ itmp ftgen 1, 0, 64, -2, \ /* depth1, depth2, max delay, IR length, idist, seed */ \ idep, 48, -1, 0.01, 0.25, 123, \ 1, 21.982, 0.05, 0.87, 4000.0, 0.6, 0.7, 2, /* ceil */ \ 1, 1.753, 0.05, 0.87, 3500.0, 0.5, 0.7, 2, /* floor */ \ 1, 15.220, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* front */ \ 1, 9.317, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* back */ \ 1, 17.545, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* right */ \ 1, 12.156, 0.05, 0.87, 5000.0, 0.8, 0.7, 2 /* left */ instr 1 /* some source signal */ a1 phasor 150 ; oscillator a1 butterbp a1, 500, 200 ; filter a1 = taninv(a1 * 100) a2 phasor 3 ; envelope a2 mirror 40*a2, -100, 5 a2 limit a2, 0, 1 a1 = a1 * a2 * 9000 kazim line 0, 2.5, 360 ; move sound source around kdist line 1, 10, 4 ; distance ; convert polar coordinates kX = sin(kazim * 3.14159 / 180) * kdist kY = cos(kazim * 3.14159 / 180) * kdist kZ = 0 a1 = a1 + 0.000001 * 0.000001 ; avoid underflows imode = 1 ; change this to 3 for 8 spk in a cube, ; or 1 for simple stereo aW, aX, aY, aZ spat3d a1, kX, kY, kZ, 1.0, 1, imode, 2, 2 aW = aW * 1.4142 ; stereo ; aL = aW + aY /* left */ aR = aW - aY /* right */ ; quad (square) ; ;aFL = aW + aX + aY /* front left */ ;aFR = aW + aX - aY /* front right */ ;aRL = aW - aX + aY /* rear left */ ;aRR = aW - aX - aY /* rear right */ ; eight channels (cube) ; ;aUFL = aW + aX + aY + aZ /* upper front left */ ;aUFR = aW + aX - aY + aZ /* upper front right */ ;aURL = aW - aX + aY + aZ /* upper rear left */ ;aURR = aW - aX - aY + aZ /* upper rear right */ ;aLFL = aW + aX + aY - aZ /* lower front left */ ;aLFR = aW + aX - aY - aZ /* lower front right */ ;aLRL = aW - aX + aY - aZ /* lower rear left */ ;aLRR = aW - aX - aY - aZ /* lower rear right */ outs aL, aR endin </CsInstruments> <CsScore> /* Written by Istvan Varga */ i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/randi.csd.xml0000644000000000000000000000470012262561504017751 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o randi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;same values every time krnd randi 100, 10 printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin instr 2 ;different values every time krnd randi 100, 10, 10 ; seed from system clock printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave. i 1 0 1 i 2 2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/urandom.csd.xml0000644000000000000000000000410312262561504020316 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd31.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create random numbers at a-rate in the range -2 to 2 aur urandom -2, 2 ; Use the random numbers to choose a frequency. afreq = aur * 500 + 100 a1 oscil 30000, afreq, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ampmidi.csd.xml0000644000000000000000000000353212262561504020276 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o ampmidi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;expects MIDI note inputs on channel 1 iamp ampmidi 1 ; scale amplitude between 0 and 1 asig oscil iamp, 220, 1 print iamp outs asig, asig endin </CsInstruments> <CsScore> ;Dummy f-table for 1 minute f 0 60 ;sine wave. f 1 0 16384 10 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cggoto.csd.xml0000644000000000000000000000456512262561504020147 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; -o cggoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = p4 ; If i1 is equal to one, play a high note. ; Otherwise play a low note. cggoto (i1 == 1), highnote lownote: a1 oscil 10000, 220, 1 goto playit highnote: a1 oscil 10000, 440, 1 goto playit playit: out a1 endin </CsInstruments> <CsScore> ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; Play lownote for one second. i 1 0 1 1 ; Play highnote for one second. i 1 1 1 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/divz.csd.xml0000644000000000000000000000455612262561504017641 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o divz.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Define the numbers to be divided. ka init 200 ; Linearly change the value of kb from 200 to 0. kb line 0, p3, 200 ; If a "divide by zero" error occurs, substitute -1. ksubst init -1 ; Safely divide the numbers. kresults divz ka, kb, ksubst ; Print out the results. printks "%f / %f = %f\\n", 0.1, ka, kb, kresults endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen40.csd.xml0000644000000000000000000000477312262561504017603 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o GEN40.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values kuser cuserrnd 0, 100, 1 printk .2, kuser asig poscil .5, 220+kuser, 3 outs asig, asig endin instr 2 ; every run time different values seed 0 kuser cuserrnd 0, 100, 1 printk .2, kuser asig poscil .5, 220+kuser, 3 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16 -7 1 4 0 8 0 4 1 ;distrubution using GEN07 f 2 0 16384 40 1 ;GEN40 is to be used with cuserrnd f 3 0 8192 10 1 ;sine i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lorenz.csd.xml0000644000000000000000000000644412262561504020174 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o lorenz.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 2 ; Instrument #1 - a lorenz system in 3D space. instr 1 ; Create a basic tone. kamp init 25000 kcps init 1000 ifn = 1 asnd oscil kamp, kcps, ifn ; Figure out its X, Y, Z coordinates. ksv init 10 krv init 28 kbv init 2.667 kh init 0.0003 ix = 0.6 iy = 0.6 iz = 0.6 iskip = 1 ax1, ay1, az1 lorenz ksv, krv, kbv, kh, ix, iy, iz, iskip ; Place the basic tone within 3D space. kx downsamp ax1 ky downsamp ay1 kz downsamp az1 idist = 1 ift = 0 imode = 1 imdel = 1.018853416 iovr = 2 aw2, ax2, ay2, az2 spat3d asnd, kx, ky, kz, idist, \ ift, imode, imdel, iovr ; Convert the 3D sound to stereo. aleft = aw2 + ay2 aright = aw2 - ay2 outs aleft, aright endin </CsInstruments> <CsScore> ; Table #1 a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 5 seconds. i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/deltap.csd.xml0000644000000000000000000000635012262561504020130 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o deltap.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasig init 0 gidel = 1 ;delay time in seconds instr 1 ain pluck .7, 440, 1000, 0, 1 outs ain, ain vincr gasig, ain ;send to global delay endin instr 2 ifeedback = p4 abuf2 delayr gidel adelL deltap .4 ;first tap (on left channel) adelM deltap 1 ;second tap (on middle channel) delayw gasig + (adelL * ifeedback) abuf3 delayr gidel kdel line 1, p3, .01 ;vary delay time adelR deltap .65 * kdel ;one pitch changing tap (on the right chn.) delayw gasig + (adelR * ifeedback) ;make a mix of all deayed signals outs adelL + adelM, adelR + adelM clear gasig endin </CsInstruments> <CsScore> i 1 0 1 i 1 3 1 i 2 0 3 0 ;no feedback i 2 3 8 .8 ;lots of feedback e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/eqfil.csd.xml0000644000000000000000000000442112262561504017754 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o eqfil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcf = p4 kfe expseg 10, p3*0.9, 1800, p3*0.1, 175 kenv linen .03, 0.05, p3, 0.05 ;low amplitude is needed to avoid clipping asig buzz kenv, kfe, sr/(2*kfe), 1 afil eqfil asig, kcf, 200, 10 outs afil*20, afil*20 endin </CsInstruments> <CsScore> ; a sine wave. f 1 0 16384 10 1 i 1 0 10 200 ;filter centre freq=200 i 1 + 10 1500 ;filter centre freq=1500 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKBrass.csd.xml0000644000000000000000000000377712262561504020325 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKBrass.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;lip tension kv3 line p7, p3, p8 ;speed of low-frequency oscillator asig STKBrass cpspch(ifrq), 1, 2, kv1, 4, 100, 11, kv3, 1, 10, 128, 40 asig = asig *3 ;amplify outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 8.05 100 120 50 0 i 1 + 3 9.00 80 82 10 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/readscratch.csd.xml0000644000000000000000000000273012262561504021140 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ilast readscratch writescratch p2 ilen readscratch 1 writescratch p3, 1 printf_i "last run at %f for %f\n", ilen, ilast, ilen endin </CsInstruments> <CsScore> i 1 0 1 i 1 2 3 i 1 6 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/readk-2.csd.xml0000644000000000000000000000454312262561504020106 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o readk-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes a control signal to a file kfreq randh 100, 1, 2, 1, 500 ;generates one random number between 400 and 600 per second dumpk kfreq, "dumpk.txt", 8, 1 ;writes the control signal printk 1, kfreq ;prints it endin instr 2 ;reads the file written by instr 1 kfreq readk "dumpk.txt", 8, 1 printk 1, kfreq ;prints it aout poscil .2, kfreq, giSine outs aout, aout endin </CsInstruments> <CsScore> i 1 0 5 i 2 5 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsftw.csd.xml0000644000000000000000000000517112262561504020210 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsftw.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 inbins = 512 ifn ftgen 0,0,inbins,10,1 ; make ftable fsrc pvsdiskin "fox.pvx", 1, 1 ; read PVOCEX file kflag pvsftw fsrc,ifn ; export amps to table, kamp init 0 if kflag==0 kgoto contin ; only proc when frame is ready tablew kamp,1,ifn ; kill lowest bins, for obvious effect tablew kamp,2,ifn tablew kamp,3,ifn tablew kamp,4,ifn ; read modified data back to fsrc pvsftr fsrc,ifn contin: ; and resynth aout pvsynth fsrc outs aout, aout endin </CsInstruments> <CsScore> i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/joystick.csd.xml0000644000000000000000000001147012262561504020515 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o joystick.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 ;0dbfs = 1 instr 1 ; gives information about your joystick in real time kmask joystick 0, 1 kidx = 2 kaxes tab 0, 1 ; number of axes has been stored in position 0 kbuttons tab 1, 1 ; number of buttons has been stored in position 1 printf "this joystick has %d axes, %d buttons\n", kidx, kaxes, kbuttons kuniq init 0 reportaxis: ; first we see if we have any x/y input kcheck = kmask & (1<<kidx) if kcheck == 0 kgoto nexta kres tab kidx, 1 kuniq = kuniq + 1 ; to be sure to make the printf print printf "axis %d, value %6d\n", kuniq, kidx-2, kres nexta: kidx = kidx+1 if kidx < (kaxes+2) kgoto reportaxis reportbutton: ; now we check for any buttons pressed kcheck = kmask & 1<<kidx if kcheck == 0 kgoto nextb kres tab kidx, 1 ; a button has been pressed, get from table printf "button %d, pushed\n", kidx*kres, (kidx-(kaxes+2)) printf "button %d, released\n", kidx*(1-kres), (kidx-(kaxes+2)) nextb: kidx = kidx+1 if kidx < (kaxes+kbuttons+2) kgoto reportbutton endin </CsInstruments> <CsScore> f1 0 32 7 0 7 0 ; will hold the joystick data i1 0 60000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/bamboo.csd.xml0000644000000000000000000000260612262561504020116 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o bamboo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 asig bamboo p4, 0.01 outs asig, asig endin </CsInstruments> <CsScore> i1 0 1 20000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluid-2.orc.xml0000644000000000000000000001163012262561504020130 0ustar rootroot sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS gienginenum1 fluidEngine gienginenum2 fluidEngine isfnum1 fluidLoad "Piano Steinway Grand Model C (21,738KB).sf2", gienginenum1, 1 ; Bright Steinway, program 1, channel 1 fluidProgramSelect gienginenum1, 1, isfnum1, 0, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidProgramSelect gienginenum1, 3, isfnum1, 0, 2 isfnum2 fluidLoad "63.3mg The Sound Site Album Bank V1.0.SF2", gienginenum2, 1 ; General MIDI, program 50, channel 2 fluidProgramSelect gienginenum2, 2, isfnum2, 0, 50 ; SEND NOTES TO STEINWAY SOUNDFONT instr 1 ; FluidSynth Steinway Rev ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 1 ikey = p4 ivelocity = p5 fluidNote gienginenum1, ichannel, ikey, ivelocity endin instr 2 ; GM soundfont ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 2 ikey = p4 ivelocity = p5 fluidNote gienginenum2, ichannel, ikey, ivelocity endin instr 3 ; FluidSynth Steinway Rev ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 3 ikey = p4 ivelocity = p5 fluidNote gienginenum1, ichannel, ikey, ivelocity endin ; COLLECT AUDIO FROM ALL SOUNDFONTS instr 100 ; Fluidsynth output ; INITIALIZATION ; Normalize so iamplitude for p5 of 80 == ampdb(80). iamplitude1 = ampdb(p5) * (10000.0 / 0.1) iamplitude2 = ampdb(p6) * (10000.0 / 0.1) ; AUDIO aleft1, aright1 fluidOut gienginenum1 aleft2, aright2 fluidOut gienginenum2 outs (aleft1 * iamplitude1) + (aleft2 * iamplitude2), \ (aright1 * iamplitude1) + (aright2 * iamplitude2) endin manual_src~dfsg/examples-xml/nreverb_ftable.csd.xml0000644000000000000000000000400612262561504021633 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o nreverb_ftable.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 a1 soundin "beats.wav" a2 nreverb a1, 1.5, .75, 0, 8, 71, 4, 72 out a1 + a2 * .4 endin </CsInstruments> <CsScore> ; freeverb time constants, as direct (negative) sample, with arbitrary gains f71 0 16 -2 -1116 -1188 -1277 -1356 -1422 -1491 -1557 -1617 0.8 0.79 0.78 0.77 0.76 0.75 0.74 0.73 f72 0 16 -2 -556 -441 -341 -225 0.7 0.72 0.74 0.76 i1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lpread.csd.xml0000644000000000000000000000531512262561504020126 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpread.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; works with or without -a option when analyzing "fox.wav" from the manual ilen filelen "fox.wav" ; length of soundfile prints "fox.wav = %f seconds\\n",ilen ktime line 0, p3, p4 krmsr,krmso,kerr,kcps lpread ktime,"fox_poles.lpc" krmso = krmso*.00007 ; low volume aout buzz krmso, kcps, 15, 1 krmsr = krmsr*.0001 ; low volume asig rand krmsr outs (aout*2)+asig, (aout*2)+asig ; mix buzz and rand endin </CsInstruments> <CsScore> ; sine f1 0 4096 10 1 i 1 0 2.8 1 ; first words only i 1 4 2.8 2.8 ; whole sentence e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/voice.csd.xml0000644000000000000000000000534112262561504017763 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o voice.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = p4 kphon = p5 asig voice kamp, 200, kphon, 0.488, 0, 1, 1, 2 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 256 1 "impuls20.aiff" 0 0 0 ;audio file for the carrier waveform f 2 0 256 10 1 ;sine wave for the vibrato waveform ; ampl phoneme i 1 0 2 0.8 1 i 1 + . 0.6 2 i 1 + . 1.8 3 i 1 + . 15.0 4 i 1 + . 0.05 5 i 1 + . 0.06 6 i 1 + . 0.03 7 i 1 + . 0.0002 8 i 1 + . 0.1 9 i 1 + . 0.5 10 i 1 + . 100 11 i 1 + . 0.03 12 i 1 + . 0.04 13 i 1 + . 0.04 14 i 1 + . 0.04 15 i 1 + . 0.05 16 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/strindexk.csd.xml0000644000000000000000000001265212262561504020674 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o strindexk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 instr 1 ;get one element of the input string whenever the metro ;triggers, and call a subinstrument to play the file Smember strget p4 istrlen strlen Smember kprint init 0 ktrig metro .5 ;whenever the trigger gives signal if ktrig == 1 then ;choose a random element (0, 1 or 2) kel random 0, 2.9999 kel = int(kel) ;make a copy for leaving Smember intact Scopy strcpyk Smember ;set the initial index for reading substrings kndx = 0 ;set counter for searching the element kcount = 0 ;start looping over the elements in Smember loop: kdelim strindexk Scopy, ":" ;as long as ":" occurs in Scopy, do: if kdelim > 0 then ;if this is the element to get if kel == kcount then ;read it as substring Sfile strsubk Scopy, kndx, kdelim kprint = kprint+1 ;and jump out kgoto call ;if not else ;cut off this element from Scopy Scopy strsubk Scopy, kdelim+1, istrlen endif ;if no element has been found,go back to loop ;and look for the next element kcount = kcount+1 kgoto loop ;if there is no delimiter left, the rest is the searched element else Sfile strcpyk Scopy endif call: ;print the result, call the subinstrument and play the file printf "kel = %d, file = '%s'\n", ktrig+kprint, kel, Sfile S_call sprintfk {{i 2 0 1 "%s"}}, Sfile scoreline S_call, ktrig endif endin instr 2 ;play Sfile strget p4 ilen filelen Sfile p3 = ilen asig soundin Sfile outs asig, asig endin </CsInstruments> <CsScore> i 1 0 30 "mary.wav:fox.wav:beats.wav" e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dssiactivate.csd.xml0000644000000000000000000000606512262561504021345 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dssiactivate.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 gihandle dssiinit "caps.so", 19, 1 ; = mono phaser and gaout init 0 ; verbose about all ports instr 1 ; activate DSSI ktoggle = p4 dssiactivate gihandle, ktoggle endin instr 2 ain1 diskin2 "beats.wav", 1,0,1 ; loop ain1 = ain1*.5 outs ain1, ain1 gaout = gaout+ain1 endin instr 3 dssictls gihandle, 0, 1, 1 ; range -1 to 1 dssictls gihandle, 1, 2, 1 ; rate 0 to 10 dssictls gihandle, 2, .8, 1 ; depth 0 to 1 dssictls gihandle, 3, 3, 1 ; spread 0 to 3.14 dssictls gihandle, 4, .9, 1 ; feedback 0 to 0.999 endin instr 4 aout1 dssiaudio gihandle, gaout ;get beats.wav, mono out outs aout1,aout1 gaout = 0 endin </CsInstruments> <CsScore> i 1 0 4 1 i 1 + . 0 i 1 + . 1 i 1 + . 0 i 1 + . 1 i 2 1 20 i 3 1 20 i 4 0 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outkc.csd.xml0000644000000000000000000000366212262561504020007 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out and midi in and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outkc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ikey notnum ivel veloc kcut linseg 100, .5, 20 ;vary controller in .5 second kbrt = int(kcut) ;whole numbers only outkc 1, 74, kbrt, 0, 127 ;controller 74 (= brightness) midion 1, ikey, ivel ;of external synth endin </CsInstruments> <CsScore> f0 30 ;runs 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sininv.csd.xml0000644000000000000000000000327112262561504020164 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o sininv.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 0.5 i1 = sininv(irad) print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zkw.csd.xml0000644000000000000000000000600212262561504017464 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zkw.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Linearly vary a k-rate signal from 100 to 1,000. kline line 100, p3, 1000 ; Add the linear signal to zk variable #1. zkw kline, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read zk variable #1. kfreq zkr 1 ; Use the value of zk variable #1 to vary ; the frequency of a sine waveform. a1 oscil 20000, kfreq, 1 ; Generate the audio output. out a1 ; Clear the zk variables, get them ready for ; another pass. zkcl 0, 1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fog.csd.xml0000644000000000000000000000476712262561504017444 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fog.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;p4 = transposition factor ;p5 = speed factor ;p6 = function table for grain data i1 = sr/ftlen(1) ;scaling to reflect sample rate and table length a1 phasor i1*p5 ;index for speed asigl fog .5, 15, p4, a1, 1, 0, .01, .5, .01, 30, 1, 2, p3 ;left channel asigr fog .4, 25, p4+.2, a1, 1, 0, .01, .5, .01, 30, 1, 2, p3, .5 ;right channel outs asigl, asigr endin </CsInstruments> <CsScore> f 1 0 131072 1 "fox.wav" 0 0 0 f 2 0 1024 19 .5 .5 270 .5 i 1 0 10 .7 .1 i 1 + 4 1.2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sr.csd.xml0000644000000000000000000000443512262561504017305 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ; -iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;use sr to find maximum harmonics ihar = int(sr/2/p4) ; maximum possible number of harmonics w/o aliasing prints "maximum number of harmonics = %d \\n", ihar kenv linen .5, 1, p3, .2 ; envelope asig buzz kenv, p4, ihar, 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 4096 10 1 ;sine wave i 1 0 3 100 ;different frequencies i 1 + 3 1000 i 1 + 3 10000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLslidBnk.csd.xml0000644000000000000000000000644712262561504020476 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLslidBnk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 441 ksmps = 100 nchnls = 1 gitypetable ftgen 0, 0, 8, -2, 1, 1, 3, 3, 5, 5, 7, 7 giouttable ftgen 0, 0, 8, -2, 0, 0.2, 0.3, 0.4, 0.5, 0.6, 0.8, 1 FLpanel "Slider Bank", 400, 380, 50, 50 ;Number of sliders inum = 8 ; Table to store output iouttable = giouttable ; Width of the slider bank in pixels iwidth = 350 ; Height of the slider in pixels iheight = 160 ; Distance of the left edge of the slider ; from the left edge of the panel ix = 30 ; Distance of the top edge of the slider ; from the top edge of the panel iy = 10 ; Table containing fader types itypetable = gitypetable FLslidBnk "1@2@3@4@5@6@7@8", inum , iouttable , iwidth , iheight , ix \ , iy , itypetable FLslidBnk "1@2@3@4@5@6@7@8", inum , iouttable , iwidth , iheight , ix \ , iy + 200 , -23 ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 ;Dummy instrument endin </CsInstruments> <CsScore> ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen16.csd.xml0000644000000000000000000000467212262561504017604 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o gen16.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 1 instr 1 kcps init 1/p3 kndx phasor kcps ifn = p4 ixmode = 1 kval table kndx, ifn, ixmode ibasefreq = 440 kfreq = kval * ibasefreq a1 oscil 20000, ibasefreq + kfreq, 1 out a1 endin </CsInstruments> <CsScore> f 1 0 16384 10 1 f 2 0 1024 16 1 1024 1 0 f 3 0 1024 16 1 1024 2 0 f 4 0 1024 16 1 1024 10 0 f 5 0 1024 16 1 1024 -1 0 f 6 0 1024 16 1 1024 -2 0 f 7 0 1024 16 1 1024 -10 0 i 1 0 2 2 i 1 + . 3 i 1 + . 4 i 1 + . 5 i 1 + . 6 i 1 + . 7 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsosc.csd.xml0000644000000000000000000000623312262561504020174 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsosc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 ; a band-limited sawtooth-wave oscillator fsig pvsosc 10000, 440, 1, 1024 ; generate wave spectral signal asig pvsynth fsig ; resynthesise it out asig endin instr 2 ; a band-limited square-wave oscillator fsig pvsosc 10000, 440, 2, 1024 ; generate wave spectral signal asig pvsynth fsig ; resynthesise it out asig endin instr 3 ; a pulse oscillator fsig pvsosc 10000, 440, 3, 1024 ; generate wave spectral signal asig pvsynth fsig ; resynthesise it out asig endin instr 4 ; a cosine-wave oscillator fsig pvsosc 10000, 440, 4, 1024 ; generate wave spectral signal asig pvsynth fsig ; resynthesise it out asig endin </CsInstruments> <CsScore> i 1 0 1 i 2 2 1 i 3 4 1 i 4 6 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wguide1.csd.xml0000644000000000000000000000471112262561504020223 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o wguide1.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a simple noise waveform. instr 1 ; Generate some noise. asig noise 20000, 0.5 out asig endin ; Instrument #2 - a waveguide example. instr 2 ; Generate some noise. asig noise 20000, 0.5 ; Run it through a wave-guide model. kfreq init 200 kcutoff init 3000 kfeedback init 0.8 awg1 wguide1 asig, kfreq, kcutoff, kfeedback out awg1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for 2 seconds. i 1 0 2 ; Play Instrument #2 for 2 seconds. i 2 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/expcurve.csd.xml0000644000000000000000000000374112262561504020521 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in Silent -odac -iadc -d ;;;realtime output </CsOptions> <CsInstruments> sr = 48000 ksmps = 1000 nchnls = 2 instr 1 ; logcurve test kmod phasor 1/p3 kout expcurve kmod, p4 printks "mod = %f out= %f\\n", 0.5, kmod, kout endin /*--- ---*/ </CsInstruments> <CsScore> i1 0 5 2 i1 5 5 5 i1 10 5 30 i1 15 5 0.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluidNote.csd.xml0000644000000000000000000000471412262561504020612 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidNote.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "19Trumpet.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 56 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 7 asigl, asigr fluidOut giengine outs asigl*imvol, asigr*imvol endin </CsInstruments> <CsScore> i 1 0 2 60 100 ;play one note from score and... i 99 0 60 ;play virtual keyboard for 60 sec. e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvcross.csd.xml0000644000000000000000000000523612262561504020360 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvcross.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; analyze "beats.wav", "flute.aiff" and "mary.wav" with PVANAL first ktime1 line 0, p3, 2 ; used as index in the "beats.pvx" file ktime2 line 0, p3, 2.6 ; used as index in the "flute.pvx" or "mary.pvx" pvbufread ktime1, "beats.pvx" ;take only amplitude from "beats.pvx" if p4 = 0 then asig pvcross ktime2, 1, "flute.pvx", 1, 0 ;and keep freqs of "flute.aiff" asig = asig*.8 ;scale output else asig pvcross ktime2, 1, "mary.pvx", 1, 0 ;and keep freqs of "mary.wav" asig = asig*.4 ;scale output endif outs asig, asig endin </CsInstruments> <CsScore> i 1 0 3 0 i 1 + 3 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/expsegr.csd.xml0000644000000000000000000000352012262561504020330 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o expsegr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .3 kenv expsegr 1, .05, 0.5, 1, .01 asig pluck kenv, icps, 200, 1, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 4096 10 1 ;sine wave f0 30 ;runs 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/line.csd.xml0000644000000000000000000000464112262561504017607 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o line.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kp = p6 ;choose between expon or line if (kp == 0) then kpitch expon p4, p3, p5 elseif (kp == 1) then kpitch line p4, p3, p5 endif asig vco2 .6, kpitch outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 300 600 0 ;if p6=0 then expon is used i 1 3 2 300 600 1 ;if p6=1 then line is used i 1 6 2 600 1200 0 i 1 9 2 600 1200 1 i 1 12 2 1200 2400 0 i 1 15 2 1200 2400 1 i 1 18 2 2400 30 0 i 1 21 2 2400 30 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/comb.csd.xml0000644000000000000000000000432012262561504017572 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o comb.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gamix init 0 instr 1 kcps expon p5, p3, p4 asig vco2 0.3, kcps outs asig, asig gamix = gamix + asig endin instr 99 krvt = 3.5 ilpt = 0.1 aleft comb gamix, krvt, ilpt aright comb gamix, krvt, ilpt*.2 outs aleft, aright clear gamix ; clear mixer endin </CsInstruments> <CsScore> i 1 0 3 20 2000 i 1 5 .01 440 440 i 99 0 8 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outq2.csd.xml0000644000000000000000000000354712262561504017736 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outq2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 asig vco2 .05, 30 ; sawtooth waveform at low volume kcut line 300, p3, 60 ; Vary cutoff frequency kresonance = 7 inumlayer = 2 asig lowresx asig, kcut, kresonance, inumlayer outq2 asig ; output channel 2 endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/xyin.csd.xml0000644000000000000000000000606212262561504017646 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc --displays ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o xyin.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 ksmps = 10 nchnls = 2 ; Instrument #1. instr 1 ; Print and capture values every 0.1 seconds. iprd = 0.1 ; The x values are from 1 to 30. ixmin = 1 ixmax = 30 ; The y values are from 1 to 30. iymin = 1 iymax = 30 ; The initial values for X and Y are both 15. ixinit = 15 iyinit = 15 ; Get the values kx and ky using the xyin opcode. kx, ky xyin iprd, ixmin, ixmax, iymin, iymax, ixinit, iyinit ; Print out the values of kx and ky. printks "kx=%f, ky=%f\\n", iprd, kx, ky ; Play an oscillator, use the x values for amplitude and ; the y values for frequency. kamp = kx * 1000 kcps = ky * 220 a1 poscil kamp, kcps, 1 outs a1, a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 30 seconds. i 1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLhvsBox.csd.xml0000644000000000000000000000355212262561504020353 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 ;Example by Andres Cabrera 2007 inumlinesX = 3 inumlinesY = 3 iwidth = 200 iheight = 200 ix = 20 iy = 20 FLpanel "FLhvsBox", 400, 240, -1, -1, 5, 1, 1 ihandle FLhvsBox inumlinesX, inumlinesY, iwidth, iheight, ix, iy ; [, image] FLpanelEnd FLrun 0dbfs = 1 instr 1 endin </CsInstruments> <CsScore> i 1 0 120 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/partikkel.csd.xml0000644000000000000000000001610112262561504020640 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o partikkel.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 20 nchnls = 2 giSine ftgen 0, 0, 65537, 10, 1 giCosine ftgen 0, 0, 8193, 9, 1, 1, 90 instr 1 kgrainfreq = 200 ; 4 grains per second kdistribution = 0 ; periodic grain distribution idisttab = -1 ; (default) flat distribution used for grain distribution async = 0 ; no sync input kenv2amt = 0 ; no secondary enveloping ienv2tab = -1 ; default secondary envelope (flat) ienv_attack = -1 ; ; default attack envelope (flat) ienv_decay = -1 ; ; default decay envelope (flat) ksustain_amount = 0.5 ; time (in fraction of grain dur) at sustain level for each grain ka_d_ratio = 0.5 ; balance between attack and decay time kduration = (0.5/kgrainfreq)*1000 ; set grain duration relative to grain rate kamp = 5000 ; amp igainmasks = -1 ; (default) no gain masking kwavfreq = 440 ; fundamental frequency of source waveform ksweepshape = 0 ; shape of frequency sweep (0=no sweep) iwavfreqstarttab = -1 ; default frequency sweep start (value in table = 1, which give no frequency modification) iwavfreqendtab = -1 ; default frequency sweep end (value in table = 1, which give no frequency modification) awavfm = 0 ; no FM input ifmamptab = -1 ; default FM scaling (=1) kfmenv = -1 ; default FM envelope (flat) icosine = giCosine ; cosine ftable kTrainCps = kgrainfreq ; set trainlet cps equal to grain rate for single-cycle trainlet in each grain knumpartials = 3 ; number of partials in trainlet kchroma = 1 ; balance of partials in trainlet ichannelmasks = -1 ; (default) no channel masking, all grains to output 1 krandommask = 0 ; no random grain masking kwaveform1 = giSine ; source waveforms kwaveform2 = giSine ; kwaveform3 = giSine ; kwaveform4 = giSine ; iwaveamptab = -1 ; (default) equal mix of all 4 sourcve waveforms and no amp for trainlets asamplepos1 = 0 ; phase offset for reading source waveform asamplepos2 = 0 ; asamplepos3 = 0 ; asamplepos4 = 0 ; kwavekey1 = 1 ; original key for source waveform kwavekey2 = 1 ; kwavekey3 = 1 ; kwavekey4 = 1 ; imax_grains = 100 ; max grains per k period asig partikkel kgrainfreq, kdistribution, idisttab, async, kenv2amt, ienv2tab, \ ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, kduration, kamp, igainmasks, \ kwavfreq, ksweepshape, iwavfreqstarttab, iwavfreqendtab, awavfm, \ ifmamptab, kfmenv, icosine, kTrainCps, knumpartials, \ kchroma, ichannelmasks, krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, asamplepos4, \ kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains outs asig, asig endin </CsInstruments> <CsScore> i1 0 5 ; partikkel e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sndwarpst.csd.xml0000644000000000000000000000517312262561504020706 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sndwarpst.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktimewarp line 0, p3, 1 ;length of stereo file "kickroll.wav" kresample init 1 ;playback at the normal speed ibeg = 0 iwsize = 4410 irandw = 441 ioverlap = p4 itimemode = 1 ; Use the ktimewarp parameter as a "time" pointer aL, aR sndwarpst .3, ktimewarp, kresample, 1, ibeg, iwsize, irandw, ioverlap, 2, itimemode aL dcblock aL ;get rid of DC offsets for left channel & aR dcblock aR ;right channel outs aL, aR endin </CsInstruments> <CsScore> f 1 0 65536 1 "kickroll.wav" 0 0 0 f 2 0 16384 9 0.5 1 0 ;half of a sine wave i 1 0 7 2 ;different overlaps i 1 + 7 5 i 1 + 7 15 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tablecopy.csd.xml0000644000000000000000000001004512262561504020635 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tablecopy.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;after an example from Jonathan Murphy gilen = 7 gilist ftgen 1, 0, gilen + 1, -2, 2, 3, 4, 5, 6, 7 ;only 6 elements, so 1 is doubled gitmp ftgen 2, 0, gilen + 1, -2, 0 ;empty table gkmax init gilen seed 0 ;each time different instr 1 ktrig metro 3 ;trigger values krnd random 0, gkmax if (ktrig == 1) then kval table krnd, gilist tablew 0, krnd, gilist kread = 0 kwrite = 0 start: knew table kread, gilist if (knew != 0) then tablew knew, kwrite, gitmp kwrite = kwrite + 1 endif kread = kread + 1 if (kread <= gilen) kgoto start tablecopy gilist, gitmp ;fill with zeroes gkmax = gkmax - 1 endif printk2 kval if (gkmax < 0) then event "i", 2, 0, 1/kr ;when ready, then stop endif asig vco2 .5, 40*kval ;sound generation outs asig, asig endin instr 2 exitnow endin </CsInstruments> <CsScore> i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pluck.csd.xml0000644000000000000000000000347412262561504020001 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pluck.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 220 icps = 220 ifn = 0 imeth = p4 asig pluck 0.7, 220, 220, ifn, imeth, .1, 10 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 5 1 i 1 5 5 4 ;needs 2 extra parameters (iparm1, iparm2) i 1 10 5 6 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/filter2.csd.xml0000644000000000000000000000526312262561504020230 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o filter2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; 2 saw waves of which one is slightly detuned ib1 = p5 ivol = p6 ;volume to compensate kcps init cpspch(p4) asig1 vco2 .05, kcps ;saw 1 asaw1 filter2 asig1, 1, 1, 1, ib1 ;filter 1 asig2 vco2 .05, kcps+1 ;saw 2 asaw2 filter2 asig2, 1, 1, 1, ib1 ;filter 2 aout = (asaw1+asaw2)*ivol ;mix outs aout, aout endin </CsInstruments> <CsScore> i 1 0 4 6.00 -.001 5 ;different filter values i 1 + 4 6.00 -.6 2 ;and different volumes i 1 + 4 6.00 -.95 .3 ;to compensate e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/noise.csd.xml0000644000000000000000000000332512262561504017773 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o noise.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kbeta line -1, p3, 1 ;change beta value from -1 to 1 asig noise .3, kbeta asig clip asig, 2, .9 ;clip signal outs asig, asig endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fof2-2.csd.xml0000644000000000000000000003671512262561504017662 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o fof2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 2 ; Example by Chuckk Hubbard 2007 instr 1 ;table-lookup vocal synthesis iolaps = 120 ifna = 1 ;f1 - sine wave ifnb = 2 ;f2 - linear rise shape itotdur = p3 iamp = p4 * 0dbfs ifreq1 = p5 ;starting frequency ifreq2 = p6 ;ending frequency kamp linseg 0, .003, iamp, itotdur-.007, iamp, .003, 0, .001, 0 kfund expseg ifreq1, itotdur, ifreq2 koct init 0 kris init .003 kdur init .02 kdec init .007 kphs init 0 kgliss init 0 iforma = p7 ;starting spectrum iformb = p8 ;ending spectrum iform1a tab_i 0, iforma ;read values of 5 formants for 1st spectrum iform2a tab_i 1, iforma iform3a tab_i 2, iforma iform4a tab_i 3, iforma iform5a tab_i 4, iforma idb1a tab_i 5, iforma ;read decibel levels for same 5 formants idb2a tab_i 6, iforma idb3a tab_i 7, iforma idb4a tab_i 8, iforma idb5a tab_i 9, iforma iband1a tab_i 10, iforma ;read bandwidths for same 5 formants iband2a tab_i 11, iforma iband3a tab_i 12, iforma iband4a tab_i 13, iforma iband5a tab_i 14, iforma iamp1a = ampdb(idb1a) ;convert db to linear multipliers iamp2a = ampdb(idb2a) iamp3a = ampdb(idb3a) iamp4a = ampdb(idb4a) iamp5a = ampdb(idb5a) iform1b tab_i 0, iformb ;values of 5 formants for 2nd spectrum iform2b tab_i 1, iformb iform3b tab_i 2, iformb iform4b tab_i 3, iformb iform5b tab_i 4, iformb idb1b tab_i 5, iformb ;decibel levels for 2nd set of formants idb2b tab_i 6, iformb idb3b tab_i 7, iformb idb4b tab_i 8, iformb idb5b tab_i 9, iformb iband1b tab_i 10, iformb ;bandwidths for 2nd set of formants iband2b tab_i 11, iformb iband3b tab_i 12, iformb iband4b tab_i 13, iformb iband5b tab_i 14, iformb iamp1b = ampdb(idb1b) ;convert db to linear multipliers iamp2b = ampdb(idb2b) iamp3b = ampdb(idb3b) iamp4b = ampdb(idb4b) iamp5b = ampdb(idb5b) kform1 line iform1a, itotdur, iform1b ;transition between formants kform2 line iform2a, itotdur, iform2b kform3 line iform3a, itotdur, iform3b kform4 line iform4a, itotdur, iform4b kform5 line iform5a, itotdur, iform5b kband1 line iband1a, itotdur, iband1b ;transition of bandwidths kband2 line iband2a, itotdur, iband2b kband3 line iband3a, itotdur, iband3b kband4 line iband4a, itotdur, iband4b kband5 line iband5a, itotdur, iband5b kamp1 line iamp1a, itotdur, iamp1b ;transition of amplitudes of formants kamp2 line iamp2a, itotdur, iamp2b kamp3 line iamp3a, itotdur, iamp3b kamp4 line iamp4a, itotdur, iamp4b kamp5 line iamp5a, itotdur, iamp5b ;5 formants for each spectrum a1 fof2 kamp1, kfund, kform1, koct, kband1, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss a2 fof2 kamp2, kfund, kform2, koct, kband2, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss a3 fof2 kamp3, kfund, kform3, koct, kband3, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss a4 fof2 kamp4, kfund, kform4, koct, kband4, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss a5 fof2 kamp5, kfund, kform5, koct, kband5, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss aout = (a1+a2+a3+a4+a5) * kamp/5 ;sum and scale aenv linen 1, 0.05, p3, 0.05 ;to avoid clicking outs aout*aenv, aout*aenv endin </CsInstruments> <CsScore> f1 0 8192 10 1 f2 0 4096 7 0 4096 1 ;**************************************************************** ; tables of formant values adapted from MiscFormants.html ; 100's: soprano 200's: alto 300's: countertenor 400's: tenor 500's: bass ; -01: "a" sound -02: "e" sound -03: "i" sound -04: "o" sound -05: "u" sound ; p-5 through p-9: frequencies of 5 formants ; p-10 through p-14: decibel levels of 5 formants ; p-15 through p-19: bandwidths of 5 formants ; formant frequencies decibel levels bandwidths ;soprano f101 0 16 -2 800 1150 2900 3900 4950 0.001 -6 -32 -20 -50 80 90 120 130 140 f102 0 16 -2 350 2000 2800 3600 4950 0.001 -20 -15 -40 -56 60 100 120 150 200 f103 0 16 -2 270 2140 2950 3900 4950 0.001 -12 -26 -26 -44 60 90 100 120 120 f104 0 16 -2 450 800 2830 3800 4950 0.001 -11 -22 -22 -50 40 80 100 120 120 f105 0 16 -2 325 700 2700 3800 4950 0.001 -16 -35 -40 -60 50 60 170 180 200 ;alto f201 0 16 -2 800 1150 2800 3500 4950 0.001 -4 -20 -36 -60 80 90 120 130 140 f202 0 16 -2 400 1600 2700 3300 4950 0.001 -24 -30 -35 -60 60 80 120 150 200 f203 0 16 -2 350 1700 2700 3700 4950 0.001 -20 -30 -36 -60 50 100 120 150 200 f204 0 16 -2 450 800 2830 3500 4950 0.001 -9 -16 -28 -55 70 80 100 130 135 f205 0 16 -2 325 700 2530 3500 4950 0.001 -12 -30 -40 -64 50 60 170 180 200 ;countertenor f301 0 16 -2 660 1120 2750 3000 3350 0.001 -6 -23 -24 -38 80 90 120 130 140 f302 0 16 -2 440 1800 2700 3000 3300 0.001 -14 -18 -20 -20 70 80 100 120 120 f303 0 16 -2 270 1850 2900 3350 3590 0.001 -24 -24 -36 -36 40 90 100 120 120 f304 0 16 -2 430 820 2700 3000 3300 0.001 -10 -26 -22 -34 40 80 100 120 120 f305 0 16 -2 370 630 2750 3000 3400 0.001 -20 -23 -30 -34 40 60 100 120 120 ;tenor f401 0 16 -2 650 1080 2650 2900 3250 0.001 -6 -7 -8 -22 80 90 120 130 140 f402 0 16 -2 400 1700 2600 3200 3580 0.001 -14 -12 -14 -20 70 80 100 120 120 f403 0 16 -2 290 1870 2800 3250 3540 0.001 -15 -18 -20 -30 40 90 100 120 120 f404 0 16 -2 400 800 2600 2800 3000 0.001 -10 -12 -12 -26 70 80 100 130 135 f405 0 16 -2 350 600 2700 2900 3300 0.001 -20 -17 -14 -26 40 60 100 120 120 ;bass f501 0 16 -2 600 1040 2250 2450 2750 0.001 -7 -9 -9 -20 60 70 110 120 130 f502 0 16 -2 400 1620 2400 2800 3100 0.001 -12 -9 -12 -18 40 80 100 120 120 f503 0 16 -2 250 1750 2600 3050 3340 0.001 -30 -16 -22 -28 60 90 100 120 120 f504 0 16 -2 400 750 2400 2600 2900 0.001 -11 -21 -20 -40 40 80 100 120 120 f505 0 16 -2 350 600 2400 2675 2950 0.001 -20 -32 -28 -36 40 80 100 120 120 ;**************************************************************** ; start dur amp start freq end freq start formant end formant i1 0 1 .8 440 412.5 201 203 i1 + . .8 412.5 550 201 204 i1 + . .8 495 330 202 205 i1 + . .8 110 103.125 501 503 i1 + . .8 103.125 137.5 501 504 i1 + . .8 123.75 82.5 502 505 i1 7 . .4 440 412.5 201 203 i1 8 . .4 412.5 550 201 204 i1 9 . .4 495 330 202 205 i1 7 . .4 110 103.125 501 503 i1 8 . .4 103.125 137.5 501 504 i1 9 . .4 123.75 82.5 502 505 i1 + . .4 440 412.5 101 103 i1 + . .4 412.5 550 101 104 i1 + . .4 495 330 102 105 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sprintf.csd.xml0000644000000000000000000000360412262561504020343 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sprintf.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;the file "impuls20.aiff" can be found in /manual/examples instr 1 ifn = 20 Sname sprintf "impuls%02d.aiff", ifn Smsg sprintf "The file name is: '%s'", Sname puts Smsg, 1 asig soundin Sname outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fill.csd.xml0000644000000000000000000000230312262561504017577 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kS[] fill 1,7,5 printk2 kS[0] printk2 kS[1] printk2 kS[2] endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/instr.csd.xml0000644000000000000000000000336712262561504020023 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o instr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 iamp = 10000 icps = 440 iphs = 0 a1 oscils iamp, icps, iphs out a1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for 2 seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKMandolin.csd.xml0000644000000000000000000000371512262561504021004 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKMandolin.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;body size kv3 = p7 ;sustain asig STKMandolin cpspch(ifrq), 1, 2, kv1, 4, 10, 11, kv3, 1, 100, 128, 100 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 .3 7.00 100 0 20 i 1 + . 8.00 10 100 20 i 1 + . 8.00 100 0 120 i 1 + 4 8.00 10 10 127 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/linsegr.csd.xml0000644000000000000000000000351712262561504020324 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o linsegr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .3 kenv linsegr 1, .05, 0.5, 1, 0 asig pluck kenv, icps, 200, 1, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 4096 10 1 ;sine wave f0 30 ;runs 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvstanal.csd.xml0000644000000000000000000000510612262561504020505 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvstanal.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifil ftgen 0, 0, 0, 1, "fox.wav", 0, 0, 1 instr 1 fsig pvstanal p4, 1, p5, gifil, p6, p7 aout pvsynth fsig outs aout, aout endin instr 2 kspeed randi 2, 2, 2 ;speed randomly between -2 and 2 kpitch randi 2, 2, 2 ;pitch between 2 octaves lower or higher fsig pvstanal kspeed, 1, octave(kpitch), gifil aout pvsynth fsig outs aout, aout endin </CsInstruments> <CsScore> ; speed pch det wrap i 1 0 2.757 1 1 0 0 i 1 3 . 2 1 0 0 i 1 6 . 2 1 0 1 i 1 9 . 1 .75 i 2 12 10 ;random scratching e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/opand.csd.xml0000644000000000000000000000464112262561504017761 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o opand.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kval randomh 0, 1.2, 20 ;choose between 0 and 1.2 if kval >0 && kval<=.5 then ;3 possible outcomes kval = 1 elseif kval >.5 && kval<=1 then kval =2 elseif kval >1 then kval =3 endif printk2 kval ;print value when it changes asig poscil .7, 440*kval, 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 16384 10 1 i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/weibull.csd.xml0000644000000000000000000000604612262561504020324 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o weibull.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values ktri weibull 100, 1 printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 ktri weibull 100, 1 printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin instr 3 ; every run time different values seed 0 ktri weibull 100, 10 ; closer to ksigma.. printk .2, ktri ; look aout oscili 0.8, 440+ktri, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 i 3 6 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ntrpol.csd.xml0000644000000000000000000000370312262561504020174 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ntrpol.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSin ftgen 1, 0, 1024, 10, 1 instr 1 avco vco2 .5, 110 ;sawtootyh wave asin poscil .5, 220, giSin ;sine wave but octave higher kx linseg 0, p3*.4, 1, p3*.6, 1 ;crossfade between saw and sine asig ntrpol avco, asin, kx outs asig, asig endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKBeeThree.csd.xml0000644000000000000000000000364112262561504020724 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKBeeThree.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kv1 = p6 ;feedback of operator 4 kv2 line p4, p3, p5 ;gain of operator 3 kv5 line 0, p3, 100 ipch = p7 asig STKBeeThree cpspch(ipch), 1, 2, kv1, 4, kv2, 11, 50, 1, 0, 128, kv5 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 20 100 127 8.00 i 1 + 8 120 0 0 6.09 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dconv.csd.xml0000644000000000000000000000451212262561504017766 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o dconv.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 #define RANDI(A) #kout randi 1, kfq, $A*.001+iseed, 1 tablew kout, $A, itable# instr 1 itable init 1 iseed init .6 isize init ftlen(itable) kfq line 1, p3, 10 $RANDI(0) $RANDI(1) $RANDI(2) $RANDI(3) $RANDI(4) $RANDI(5) $RANDI(6) $RANDI(7) $RANDI(8) $RANDI(9) $RANDI(10) $RANDI(11) $RANDI(12) $RANDI(13) $RANDI(14) $RANDI(15) asig rand 10000, .5, 1 asig butlp asig, 5000 asig dconv asig, isize, itable out asig *.5 endin </CsInstruments> <CsScore> f1 0 16 10 1 i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/scalearray.csd.xml0000644000000000000000000000470212262561504021004 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> ;example by joachim heintz seed 0 instr 1 ;create an array with 10 elements kArr[] init 10 ;fill in random numbers and print them out printks "kArr in maximum range 0..100:\n", 0 kIndx = 0 until kIndx == 10 do kNum random 0, 100 kArr[kIndx] = kNum printf "kArr[%d] = %10f\n", kIndx+1, kIndx, kNum kIndx += 1 od ;scale numbers 0...1 and print them out again scalearray kArr, 0, 1 kIndx = 0 printks "kArr in range 0..1\n", 0 until kIndx == 10 do printf "kArr[%d] = %10f\n", kIndx+1, kIndx, kArr[kIndx] kIndx += 1 od turnoff endin </CsInstruments> <CsScore> i1 0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fout.csd.xml0000644000000000000000000000527312262561504017637 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fout.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 asig poscil3 .5, 880, giSine ;write a raw file: 32 bits with header fout "fout_880.wav", 15, asig outs asig, asig endin instr 2 klfo lfo 1, 2, 0 asig poscil3 .5*klfo, 220, giSine ;write an aiff file: 32 bits with header fout "fout_aif.aiff", 25, asig ; fout "fout_all3.wav", 14, asig outs asig, asig endin instr 99 ;read the stereo csound output buffer allL, allR monitor ;write the output of csound to an audio file ;to a wav file: 16 bits with header fout "fout_all.wav", 14, allL, allR endin </CsInstruments> <CsScore> i 1 0 2 i 2 0 3 i 99 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsfread.csd.xml0000644000000000000000000000405012262561504020464 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsfread.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; create a PVOC-EX (*.pvx) file with PVANAL first idur filelen "kickroll.pvx" ;find duration of (stereo) analysis file kpos line 0,p3,idur ;to ensure we process whole file fsigr pvsfread kpos,"kickroll.pvx", 1 ;create fsig from right channel aout pvsynth fsigr ;resynthesise it outs aout, aout endin </CsInstruments> <CsScore> i 1 0 10 i 1 11 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cosh.csd.xml0000644000000000000000000000326312262561504017613 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cosh.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 1 i1 = cosh(irad) print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluidAllOut.orc.xml0000644000000000000000000001120212262561504021105 0ustar rootroot sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS gienginenum1 fluidEngine gienginenum2 fluidEngine isfnum1 fluidLoad "Piano Steinway Grand Model C (21,738KB).sf2", gienginenum1, 1 ; Bright Steinway, program 1, channel 1 fluidProgramSelect gienginenum1, 1, isfnum1, 0, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidProgramSelect gienginenum1, 3, isfnum1, 0, 2 isfnum2 fluidLoad "63.3mg The Sound Site Album Bank V1.0.SF2", gienginenum2, 1 ; General MIDI, program 50, channel 2 fluidProgramSelect gienginenum2, 2, isfnum2, 0, 50 ; SEND NOTES TO STEINWAY SOUNDFONT instr 1 ; FluidSynth Steinway Rev ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 1 ikey = p4 ivelocity = p5 istatus = 144 fluidControl gienginenum1, istatus, ichannel, ikey, ivelocity endin instr 2 ; GM soundfont ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 2 ikey = p4 ivelocity = p5 istatus = 144 fluidNote gienginenum2, ichannel, ikey, ivelocity endin instr 3 ; FluidSynth Steinway Rev ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 3 ikey = p4 ivelocity = p5 istatus = 144 fluidNote gienginenum1, ichannel, ikey, ivelocity endin ; COLLECT AUDIO FROM ALL SOUNDFONTS instr 100 ; Fluidsynth output ; INITIALIZATION ; Normalize so iamplitude for p5 of 80 == ampdb(80). iamplitude = ampdb(p5) * (10000.0 / 0.1) ; AUDIO aleft, aright fluidAllOut outs aleft * iamplitude, aright * iamplitude endin manual_src~dfsg/examples-xml/linrand.csd.xml0000644000000000000000000000464012262561504020306 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o linrand.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values klin linrand 100 printk .2, klin ; look aout oscili 0.8, 440+klin, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 klin linrand 100 printk .2, klin ; look aout oscili 0.8, 440+klin, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/prealloc.csd.xml0000644000000000000000000000446312262561504020463 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o prealloc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Pre-allocate memory for five instances of Instrument #1. prealloc 1, 5 ; Instrument #1 instr 1 ; Generate a waveform, get the cycles per second from the 4th p-field. a1 oscil 6500, p4, 1 out a1 endin </CsInstruments> <CsScore> ; Just generate a nice, ordinary sine wave. f 1 0 32768 10 1 ; Play five instances of Instrument #1 for one second. ; Note that 4th p-field contains cycles per second. i 1 0 1 220 i 1 0 1 440 i 1 0 1 880 i 1 0 1 1320 i 1 0 1 1760 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zacl.csd.xml0000644000000000000000000000542712262561504017614 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zacl.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Generate a simple sine waveform. asin oscil 20000, 440, 1 ; Send the sine waveform to za variable #1. zaw asin, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read za variable #1. a1 zar 1 ; Generate the audio output. out a1 ; Clear the za variables, get them ready for ; another pass. zacl 0, 1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/grain3.csd.xml0000644000000000000000000002061312262561504020040 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o grain3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 48000 kr = 1000 ksmps = 48 nchnls = 1 /* Bartlett window */ itmp ftgen 1, 0, 16384, 20, 3, 1 /* sawtooth wave */ itmp ftgen 2, 0, 16384, 7, 1, 16384, -1 /* sine */ itmp ftgen 4, 0, 1024, 10, 1 /* window for "soft sync" with 1/32 overlap */ itmp ftgen 5, 0, 16384, 7, 0, 256, 1, 7936, 1, 256, 0, 7936, 0 /* generate bandlimited sawtooth waves */ itmp ftgen 3, 0, 4096, -30, 2, 1, 2048 icnt = 0 loop01: ; 100 tables for 8 octaves from 30 Hz ifrq = 30 * exp(log(2) * 8 * icnt / 100) itmp ftgen icnt + 100, 0, 4096, -30, 3, 1, sr / (2 * ifrq) icnt = icnt + 1 if (icnt < 99.5) igoto loop01 /* convert frequency to table number */ #define FRQ2FNUM(xout'xcps'xbsfn) # $xout = int(($xbsfn) + 0.5 + (100 / 8) * log(($xcps) / 30) / log(2)) $xout limit $xout, $xbsfn, $xbsfn + 99 # /* instr 1: pulse width modulated grains */ instr 1 kfrq = 523.25 ; frequency $FRQ2FNUM(kfnum'kfrq'100) ; table number kfmd = kfrq * 0.02 ; random variation in frequency kgdur = 0.2 ; grain duration kdens = 200 ; density iseed = 1 ; random seed kphs oscili 0.45, 1, 4 ; phase a1 grain3 kfrq, 0, kfmd, 0.5, kgdur, kdens, 100, \ kfnum, 1, -0.5, 0, iseed, 2 a2 grain3 kfrq, 0.5 + kphs, kfmd, 0.5, kgdur, kdens, 100, \ kfnum, 1, -0.5, 0, iseed, 2 ; de-click aenv linseg 0, 0.01, 1, p3 - 0.05, 1, 0.04, 0, 1, 0 out aenv * 2250 * (a1 - a2) endin /* instr 2: phase variation */ instr 2 kfrq = 220 ; frequency $FRQ2FNUM(kfnum'kfrq'100) ; table number kgdur = 0.2 ; grain duration kdens = 200 ; density iseed = 2 ; random seed kprdst expon 0.5, p3, 0.02 ; distribution a1 grain3 kfrq, 0.5, 0, 0.5, kgdur, kdens, 100, \ kfnum, 1, 0, -kprdst, iseed, 64 ; de-click aenv linseg 0, 0.01, 1, p3 - 0.05, 1, 0.04, 0, 1, 0 out aenv * 1500 * a1 endin /* instr 3: "soft sync" */ instr 3 kdens = 130.8 ; base frequency kgdur = 2 / kdens ; grain duration kfrq expon 880, p3, 220 ; oscillator frequency $FRQ2FNUM(kfnum'kfrq'100) ; table number a1 grain3 kfrq, 0, 0, 0, kgdur, kdens, 3, kfnum, 5, 0, 0, 0, 2 a2 grain3 kfrq, 0.667, 0, 0, kgdur, kdens, 3, kfnum, 5, 0, 0, 0, 2 ; de-click aenv linseg 0, 0.01, 1, p3 - 0.05, 1, 0.04, 0, 1, 0 out aenv * 10000 * (a1 - a2) endin </CsInstruments> <CsScore> t 0 60 i 1 0 3 i 2 4 3 i 3 8 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/times.csd.xml0000644000000000000000000000343112262561504017775 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o times.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the value from times every half-second. k1 times printks "k1 = %f seconds\\n", 0.5, k1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vbapgmove.csd.xml0000644000000000000000000000373712262561504020653 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbapgmove.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 vbaplsinit 2, 4, 0, 90, 180, 270 instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav k0,k1,k2,k3 vbapgmove p3, 1, 2, 310, 180 ;change movement of soundsource in outq k0*asig,k1*asig,k2*asig,k3*asig ;the rear speakers endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/doppler.csd.xml0000644000000000000000000000527612262561504020332 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o doppler.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 48000 ksmps = 128 nchnls = 2 0dbfs = 1 instr 1 iattack init 0.05 irelease init 0.05 isustain init p3 p3 init iattack + isustain + irelease kdamping linseg 0.0, iattack, 1.0, isustain, 1.0, irelease, 0.0 kmic init 4 ; Position envelope, with a changing rate of change of position. ; transeg a dur ty b dur ty c dur ty d kposition transeg 4, p3*.4, 0, 120, p3*.3, -3, 50, p3*.3, 2, 4 ismoothinghz init 6 ispeedofsound init 340.29 asignal vco2 0.5, 110 aoutput doppler asignal, kposition, kmic, ispeedofsound, ismoothinghz outs aoutput*kdamping, aoutput * kdamping endin </CsInstruments> <CsScore> i1 0.0 20 e1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mute.csd.xml0000644000000000000000000000432212262561504017626 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mute.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; Mute Instrument #2. mute 2 ; Mute Instrument three. mute "three" instr 1 a1 oscils 0.2, 440, 0 outs a1, a1 endin instr 2 ; gets muted a1 oscils 0.2, 880, 0 outs a1, a1 endin instr three ; gets muted a1 oscils 0.2, 1000, 0 outs a1, a1 endin </CsInstruments> <CsScore> i 1 0 1 i 2 0 1 i "three" 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/crunch.csd.xml0000644000000000000000000000300412262561504020132 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o crunch.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig crunch 0.8, 0.1, 7, p4 outs asig, asig endin </CsInstruments> <CsScore> i1 0 1 .9 i1 1 1 .1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/spat3dt.csd.xml0000644000000000000000000001233412262561504020240 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o spat3dt.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 garvb init 0 gaW init 0 gaX init 0 gaY init 0 itmp ftgen 1, 0, 64, -2, 2, 40, -1, -1, -1, 123, \ 1, 13.000, 0.05, 0.85, 20000.0, 0.0, 0.50, 2, \ 1, 2.000, 0.05, 0.85, 20000.0, 0.0, 0.25, 2, \ 1, 16.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 9.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 12.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 8.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2 itmp ftgen 2, 0, 262144, -2, 0 spat3dt 2, -0.2, 1, 0, 1, 1, 2, 0.005 itmp ftgen 3, 0, 262144, -52, 3, 2, 0, 4, 2, 1, 4, 2, 2, 4 instr 1 a1 vco2 1, 440, 10 kfrq port 100, 0.008, 20000 a1 butterlp a1, kfrq a2 linseg 0, 0.003, 1, 0.01, 0.7, 0.005, 0, 1, 0 a1 = a1 * a2 * 2 denorm a1 vincr garvb, a1 aw, ax, ay, az spat3di a1, p4, p5, p6, 1, 1, 2 vincr gaW, aw vincr gaX, ax vincr gaY, ay endin instr 2 denorm garvb ; skip as many samples as possible without truncating the IR arW, arX, arY ftconv garvb, 3, 2048, 2048, (65536 - 2048) aW = gaW + arW aX = gaX + arX aY = gaY + arY garvb = 0 gaW = 0 gaX = 0 gaY = 0 aWre, aWim hilbert aW aXre, aXim hilbert aX aYre, aYim hilbert aY aWXr = 0.0928*aXre + 0.4699*aWre aWXiYr = 0.2550*aXim - 0.1710*aWim + 0.3277*aYre aL = aWXr + aWXiYr aR = aWXr - aWXiYr outs aL, aR endin </CsInstruments> <CsScore> i 1 0 0.5 0.0 2.0 -0.8 i 1 1 0.5 1.4 1.4 -0.6 i 1 2 0.5 2.0 0.0 -0.4 i 1 3 0.5 1.4 -1.4 -0.2 i 1 4 0.5 0.0 -2.0 0.0 i 1 5 0.5 -1.4 -1.4 0.2 i 1 6 0.5 -2.0 0.0 0.4 i 1 7 0.5 -1.4 1.4 0.6 i 1 8 0.5 0.0 2.0 0.8 i 2 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKVoicForm.csd.xml0000644000000000000000000000360312262561504020763 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKVoicForm.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv2 line p5, p3, p6 ;Vowel/Phoneme Selection asig STKVoicForm cpspch(p4), 1, 2, 1, 4, kv2, 128, 100, 1, 10, 11, 100 asig = asig * .5 ;too loud outs asig, asig endin </CsInstruments> <CsScore> i 1 0 5 7.00 100 0 i 1 + 10 7.00 1 50 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/partials.csd.xml0000644000000000000000000000357012262561504020477 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o partials.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "fox.wav", 1 fs1,fsi2 pvsifd ain,2048,512,1 ; ifd analysis fst partials fs1,fsi2,.003,1,3,500 ; partial tracking aout resyn fst, 1, 1.5, 500, 1 ; resynthesis (up a 5th) outs aout, aout endin </CsInstruments> <CsScore> f 1 0 4096 10 1 i 1 0 2.8 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zir.csd.xml0000644000000000000000000000455412262561504017467 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zir.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple instrument. instr 1 ; Set the zk variable #1 to 32.594. ziw 32.594, 1 endin ; Instrument #2 -- prints out zk variable #1. instr 2 ; Read the zk variable #1 at i-rate. i1 zir 1 ; Print out the value of zk variable #1. print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cos.csd.xml0000644000000000000000000000711012262561504017436 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cos.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icos1 = cos(0) ;cosine of 0 is 1 icos2 = cos($M_PI_2) ;cosine of pi/2 (1.5707...) is 0 icos3 = cos($M_PI) ;cosine of pi (3.1415...) is -1 icos4 = cos($M_PI_2 * 3) ;cosine of 3/2pi (4.7123...) is 0 icos5 = cos($M_PI * 2) ;cosine of 2pi (6.2831...) is 1 icos6 = cos($M_PI * 4) ;cosine of 4pi is also 1 as it is periodically to 2pi print icos1, icos2, icos3, icos4, icos5, icos6 endin instr 2 ;cos used in panning, after an example from Hans Mikelson aout vco2 0.8, 220 ; sawtooth kpan linseg p4, p3, p5 ;0 = left, 1 = right kpan = kpan*$M_PI_2 ;range 0-1 becomes 0-pi/2 kpanl = cos(kpan) kpanr = sin(kpan) outs aout*kpanl, aout*kpanr endin </CsInstruments> <CsScore> i 1 0 0 i 2 0 5 0 1 ;move left to right i 2 6 5 1 0 ;move right to left e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/delayw.csd.xml0000644000000000000000000000635012262561504020144 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o delayw.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasig init 0 gidel = 1 ;delay time in seconds instr 1 ain pluck .7, 440, 1000, 0, 1 outs ain, ain vincr gasig, ain ;send to global delay endin instr 2 ifeedback = p4 abuf2 delayr gidel adelL deltap .4 ;first tap (on left channel) adelM deltap 1 ;second tap (on middle channel) delayw gasig + (adelL * ifeedback) abuf3 delayr gidel kdel line 1, p3, .01 ;vary delay time adelR deltap .65 * kdel ;one pitch changing tap (on the right chn.) delayw gasig + (adelR * ifeedback) ;make a mix of all deayed signals outs adelL + adelM, adelR + adelM clear gasig endin </CsInstruments> <CsScore> i 1 0 1 i 1 3 1 i 2 0 3 0 ;no feedback i 2 3 8 .8 ;lots of feedback e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsbin.csd.xml0000644000000000000000000000420412262561504020154 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsbin.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 ifftsize = 1024 iwtype = 1 /* cleaner with hanning window */ ;a1 soundin "input.wav" ;select a soundifle a1 inch 1 ;Use realtime input fsig pvsanal a1, ifftsize, ifftsize/4, ifftsize, iwtype kamp, kfr pvsbin fsig, 10 adm oscil kamp, kfr, 1 out adm endin </CsInstruments> <CsScore> ; sine wave f 1 0 4096 10 1 i 1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKShakers.csd.xml0000644000000000000000000000334212262561504020637 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKShakers.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 asig STKShakers cpspch(p4), 1, 2, 10, 4, 10, 11, 10, 1, 112, 128, 80, 1071, 5 asig = asig *2 ;amplify outs asig, asig endin </CsInstruments> <CsScore> i 1 0.2 .5 7.00 75 0 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSadd-2.csd.xml0000644000000000000000000000712412262561504020116 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac -d -m1 </CsOptions> <CsInstruments> ;example by joachim heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 1024, 10, 1 instr AllTheTones Sfile = "fox.ats" prints "Resynthesizing with all the tones.\n" iDur ATSinfo Sfile, 7 p3 = iDur iNumParts ATSinfo Sfile, 3 prints "Overall number of partials = %d\n", iNumParts ktime line 0, iDur, iDur asig ATSadd ktime, 1, Sfile, giSine, iNumParts outs asig, asig ;start next instr event_i "i", "TonesInBandsOfTen", iDur+1, iDur, 0, iNumParts endin instr TonesInBandsOfTen Sfile = "fox.ats" iOffset = p4 ;start at this partial iNumParts = p5 ;overall number of partials prints "Resynthesizing with partials %d - %d.\n", iOffset+1, iOffset+10 ktime line 0, p3, p3 asig ATSadd ktime, 1, Sfile, giSine, 10, iOffset outs asig, asig ;start next instance until there are enough partials left if iOffset+20 < iNumParts then event_i "i", "TonesInBandsOfTen", p3+1, p3, iOffset+10, iNumParts else event_i "i", "End", p3, 1 endif endin instr End exitnow endin </CsInstruments> <CsScore> i "AllTheTones" 0 1 e 999 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/OSCmidisend.csd.xml0000644000000000000000000000354512262561504021023 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 1 ; Example by David Akbari 2007 ; Modified by Jonathan Murphy ; Use this file to generate OSC events for OSCmidircv.csd #define IPADDRESS # "localhost" # #define PORT # 47120 # turnon 1000 instr 1000 kst, kch, kd1, kd2 midiin OSCsend kst+kch+kd1+kd2, $IPADDRESS, $PORT, "/midi", "iiii", kst, kch, kd1, kd2 endin </CsInstruments> <CsScore> f 0 3600 ;Dummy f-table e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lposcil.csd.xml0000644000000000000000000000421312262561504020320 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lposcil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 1.5 ; a fifth up kloop = 0 ; loop start time in samples kend = 45000 ; loop end time in samples asig lposcil 1, kcps, kloop, kend, 1 outs asig, asig endin </CsInstruments> <CsScore> ; Its table size is deferred, ; and format taken from the soundfile header. f 1 0 0 1 "beats.wav" 0 0 0 ; Play Instrument #1 for 6 seconds. ; This will loop the drum pattern several times. i 1 0 6 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dbfsamp.csd.xml0000644000000000000000000000343512262561504020274 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dbfsamp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 iamp = p4 idb = dbfsamp(iamp) print idb asig vco2 iamp, 110 ;sawtooth outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 1 i 1 + 1 100 i 1 + 1 1000 i 1 + 1 10000 i 1 + 1 30000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/trfilter.csd.xml0000644000000000000000000000457212262561504020516 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trfilter.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifn ftgen 2, 0, -22050, 5, 1, 1000, 1, 4000, 0.000001, 17050, 0.000001 ; low-pass filter curve of 22050 points instr 1 kam line 1, p3, p4 ain diskin2 "beats.wav", 1, 0, 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .003, 1, 3, 500 ; partial tracking fscl trfilter fst, kam, gifn ; filtering using function table 2 aout tradsyn fscl, 1, 1, 500, 1 ; resynthesis outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 i 1 0 4 1 i 1 5 4 0 ;reduce filter effect e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/active_k.csd.xml0000644000000000000000000000563612262561504020452 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o active_k.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a noisy waveform. instr 1 ; Generate a really noisy waveform. anoisy rand 44100 ; Turn down its amplitude. aoutput gain anoisy, 2500 ; Send it to the output. out aoutput endin ; Instrument #2 - counts active instruments at k-rate. instr 2 ; Count the active instances of Instrument #1. kcount active 1 ; Print the number of active instances. printk2 kcount endin </CsInstruments> <CsScore> ; Start the first instance of Instrument #1 at 0:00 seconds. i 1 0.0 3.0 ; Start the second instance of Instrument #1 at 0:015 seconds. i 1 1.5 1.5 ; Play Instrument #2 at 0:01 seconds, when we have only ; one active instance of Instrument #1. i 2 1.0 0.1 ; Play Instrument #2 at 0:02 seconds, when we have ; two active instances of Instrument #1. i 2 2.0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outkpb.csd.xml0000644000000000000000000000353212262561504020162 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out and midi in and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outkpb.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 10 nchnls = 2 instr 1 ikey notnum ivel veloc kpch linseg 100, 1, 0 ;vary in 1 second kpb = int(kpch) ;whole numbers only outkpb 1, kpb, 0, 127 ;(= pitchbend) midion 1, ikey, ivel ;of external synth endin </CsInstruments> <CsScore> f0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/transegb.csd.xml0000644000000000000000000000407712262561504020470 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o transeg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 2 0dbfs = 1 instr 1 ;p4 and p5 determine the type of curve for each ;section of the envelope kenv transegb 0.01, p3*0.25, p4, 1, p3, p5, 0.01 a1 oscil kenv, 440, 1 outs a1, a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 i 1 0 2 2 2 i 1 + . 5 5 i 1 + . 1 1 i 1 + . 0 0 i 1 + . -2 -2 i 1 + . -2 2 i 1 + . 2 -2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lphasor.csd.xml0000644000000000000000000000525512262561504020332 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o lphashor.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Example by Jonathan Murphy Dec 2006 sr = 44100 ksmps = 10 nchnls = 1 instr 1 ifn = 1 ; table number ilen = nsamp(ifn) ; return actual number of samples in table itrns = 1 ; no transposition ilps = 0 ; loop starts at index 0 ilpe = ilen ; ends at value returned by nsamp above imode = 3 ; loop forwards & backwards istrt = 10000 ; commence playback at index 10000 samples ; lphasor provides index into f1 alphs lphasor itrns, ilps, ilpe, imode, istrt atab tablei alphs, ifn ; amplify signal atab = atab * 10000 out atab endin </CsInstruments> <CsScore> f 1 0 262144 1 "beats.wav" 0 4 1 i1 0 60 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vst4cs.csd.xml0000644000000000000000000001074212262561504020105 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Credits: Adapted by Michael Gogins ; from code by David Horowitz and Lian Cheung. ; The "--displays" option is required in order for ; the Pianoteq GUI to dispatch events and display properly. -m3 --displays -odac </CsOptions> <CsInstruments> sr = 44100 ksmps = 20 nchnls = 2 ; Load the Pianoteq into memory. gipianoteq vstinit "C:\\Program Files\\Steinberg\\VstPlugins\\Pianoteq 3.0 Trial\\Pianoteq30 Trial.dll", 1 ; Print information about the Pianoteq, such as parameter names and numbers. vstinfo gipianoteq ; Open the Pianoteq's GUI. vstedit gipianoteq ; Send notes from the score to the Pianoteq. instr 1 ; MIDI channels are numbered starting at 0. ; p3 always contains the duration of the note. ; p4 contains the MIDI key number (pitch), ; p5 contains the MIDI velocity number (loudness), imidichannel init 0 vstnote gipianoteq, imidichannel, p4, p5, p3 endin ; Send parameter changes to the Pianoteq. instr 2 ; p4 is the parameter number. ; p5 is the parameter value. vstparamset gipianoteq, p4, p5 endin ; Send audio from the Pianoteq to the output. instr 3 ablankinput init 0 aleft, aright vstaudio gipianoteq, ablankinput, ablankinput outs aleft, aright endin </CsInstruments> <CsScore> ; Turn on the instrument that receives audio from the Pianoteq indefinitely. i 3 0 -1 ; Send parameter changes to Pianoteq before sending any notes. ; NOTE: All parameters must be between 0.0 and 1.0. ; Length of piano strings: i 2 0 1 33 0.5 ; Hammer noise: i 2 0 1 25 0.1 ; Send a C major 7th arpeggio to the Pianoteq. i 1 1 10 60 76 i 1 2 10 64 73 i 1 3 10 67 70 i 1 4 10 71 67 ; End the performance, leaving some time ; for the Pianoteq to finish sending out its audio, ; or for the user to play with the Pianoteq virtual keyboard. e 20 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vbapmove.csd.xml0000644000000000000000000000352512262561504020477 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbapmove.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 vbaplsinit 2, 4, 0, 90, 180, 270 instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav a1,a2,a3,a4 vbapmove asig, p3, 1, 2, 310, 180 ;change movement of soundsource in outq a1,a2,a3,a4 ;the rear speakers endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mirror.csd.xml0000644000000000000000000000657012262561504020175 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mirror.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; Limit / Mirror / Wrap igain = p4 ;gain ilevl1 = p5 ; + level ilevl2 = p6 ; - level imode = p7 ;1 = limit, 2 = mirror, 3 = wrap ain soundin "fox.wav" ain = ain*igain if imode = 1 goto limit if imode = 2 goto mirror asig wrap ain, ilevl2, ilevl1 goto outsignal limit: asig limit ain, ilevl2, ilevl1 goto outsignal mirror: asig mirror ain, ilevl2, ilevl1 outsignal: outs asig*.5, asig*.5 ;mind your speakers endin </CsInstruments> <CsScore> ; Gain +Levl -Levl Mode i1 0 3 4.00 .25 -1.00 1 ;limit i1 4 3 4.00 .25 -1.00 2 ;mirror i1 8 3 4.00 .25 -1.00 3 ;wrap e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fof2.csd.xml0000644000000000000000000000763312262561504017520 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o fof2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 2 ;By Andres Cabrera 2007 instr 1 ;table-lookup vocal synthesis kris init p12 kdur init p13 kdec init p14 iolaps init p15 ifna init 1 ; Sine wave ifnb init 2 ; Straight line rise shape itotdur init p3 kphs init 0 ; No phase modulation (constant kphs) kfund line p4, p3, p5 kform line p6, p3, p7 koct line p8, p3, p9 kband line p10, p3, p11 kgliss line p16, p3, p17 kenv linen 5000, 0.03, p3, 0.03 ;to avoid clicking aout fof2 kenv, kfund, kform, koct, kband, kris, kdur, kdec, iolaps, \ ifna, ifnb, itotdur, kphs, kgliss outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 f2 0 4096 7 0 4096 1 ; kfund1 kfund2 kform1 kform2 koct1 koct2 kband1 kband2 kris kdur kdec iolaps kgliss1 kgliss2 i1 0 4 220 220 510 510 0 0 30 30 0.01 0.03 0.01 20 1 1 i1 + . 220 220 510 910 0 0 30 30 0.01 0.03 0.01 20 1 1 i1 + . 220 220 510 510 0 0 100 30 0.01 0.03 0.01 20 1 1 i1 + . 220 220 510 510 0 1 30 30 0.01 0.03 0.01 20 1 1 i1 + . 220 220 510 510 0 0 30 30 0.01 0.03 0.01 20 1 2 i1 + . 220 220 510 510 0 0 30 30 0.01 0.03 0.01 20 0.5 1 i1 + . 220 220 510 510 0 0 30 30 0.01 0.05 0.01 100 1 1 i1 + . 220 440 510 510 0 0 30 30 0.01 0.05 0.01 100 1 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vtablewk.csd.xml0000644000000000000000000000324212262561504020473 0ustar rootroot <CsoundSynthesizer> <CsOptions> ;-ovtablewa.wav -W -b441 -B441 -odac -b441 -B441 </CsOptions> <CsInstruments> sr=44100 kr=441 ksmps=100 nchnls=2 instr 1 ilen = ftlen(1) knew1 oscil 10000, 440, 3 knew2 oscil 15000, 440, 3, 0.5 kindex phasor 0.3 asig oscil 1, sr/ilen , 1 vtablewk kindex*ilen, 1, 0, knew1, knew2 out asig,asig endin </CsInstruments> <CsScore> f1 0 262144 -1 "beats.wav" 0 4 0 f2 0 262144 2 0 f3 0 1024 10 1 i1 0 10 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/push.csd.xml0000644000000000000000000000444112262561504017635 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o push.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 stack 100000 instr 1 a1 oscils 0.7, 220, 0 k1 line 0, p3, 1 push "blah", 123.45, a1, k1 push rnd(k1) k_rnd pop S01, i01, a01, k01 pop printf_i "S01 = '%s', i01 = %g\n", 1, S01, i01 ktrig metro 5.0 printf "k01 = %.3f, k_rnd = %.3f\n", ktrig, k01, k_rnd outs a01, a01 endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsgain.csd.xml0000644000000000000000000000361712262561504020331 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsgain.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kgain = p4 asig diskin2 "beats.wav", 1 fsig pvsanal asig, 1024, 256, 1024, 1; analyse it ftps pvsgain fsig, kgain ; amplify it atps pvsynth ftps ; synthesise it outs atps, atps endin </CsInstruments> <CsScore> i1 0 2 .5 i1 + 2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pchbend.csd.xml0000644000000000000000000000371312262561504020262 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -Q1 -Ma ;;;realtime audio out and midi in (on all inputs) and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pchbend.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;use external midi keyboard icps cpsmidi kbnd pchbend 0, 100 ;one octave lower and higher kenv linsegr 0,.001, 1, .1, 0 ;amplitude envelope asig pluck .8 * kenv, icps+kbnd, 440, 0, 1 outs asig, asig endin </CsInstruments> <CsScore> f 0 30 ;runs 30 seconds </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/adsyn.csd.xml0000644000000000000000000000342012262561504017770 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o adsyn.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; analyze the file "kickroll.wav" first kamod = 1 kfmod = p4 ksmod = p5 asig adsyn kamod, kfmod, ksmod, "kickroll.het" outs asig, asig endin </CsInstruments> <CsScore> i 1 0 4 1 .2 i 1 + 1 2 1 i 1 + 1 .3 1.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/resonk.csd.xml0000644000000000000000000000415112262561504020155 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o resonk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 0, 0, 2^10, 10, 1 instr 1 ksig randomh 400, 1800, 150 aout poscil .2, 1000+ksig, gisin outs aout, aout endin instr 2 ksig randomh 400, 1800, 150 khp line 1, p3, 400 ;vary high-pass ksig resonk ksig, khp, 50 aout poscil .2, 1000+ksig, gisin outs aout, aout endin </CsInstruments> <CsScore> i 1 0 5 i 2 5.5 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gensone.csd.xml0000644000000000000000000000624712262561504020322 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gensone.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; simple oscillator with loudness correction. kcps = cpspch(p4) kenv linseg 0, p3*0.25, 1, p3*0.75, 0 ;amplitude envelope kamp tablei 16384 *kenv, 2 asig oscil kamp, kcps, 1 outs asig, asig endin instr 2 ;neutral oscillator to compare with kcps = cpspch(p4) kenv linseg 0, p3*0.25, 1, p3*0.75, 0 ;amplitude envelope asig oscil kenv, kcps, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave f 2 0 16385 "sone" 0 32000 32000 0 s f 0 1 ;1 second of silence before we start... s i 1 0 2 7.00 i 1 + . 7.01 i 1 + . 8.02 i 1 + . 8.03 s i 2 0 2 7.00 i 2 + . 7.01 i 2 + . 8.02 i 2 + . 8.03 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/e.csd.xml0000644000000000000000000000447012262561504017104 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o e.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ga1 init 0 instr 1 aenv expseg .01, p3*0.1, 1, p3*0.9, 0.01 ga1 poscil3 .5*aenv, cpspch(p4), 1 outs ga1,ga1 endin instr 99 aL, aR reverbsc ga1, ga1, 0.85, 12000, sr, 0.5, 1 outs aL,aR ga1 = 0 endin </CsInstruments> <CsScore> f 1 0 128 10 1 ;sine wave i 1 1 0.1 8.00 i 1 2 0.1 8.02 i 1 3 0.1 8.04 i 1 4 0.1 9.06 i 99 0 6 ;remains active for 6 seconds e10 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/nsamp.csd.xml0000644000000000000000000000562612262561504020002 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o nsamp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; slightly adapted example from Jonathan Murphy Dec 2006 sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ; table number ilen = nsamp(ifn) prints "actual numbers of samples = %d\n", ilen itrns = 1 ; no transposition ilps = 0 ; loop starts at index 0 ilpe = ilen ; ends at value returned by nsamp above imode = 1 ; loops forward istrt = 0 ; commence playback at index 0 samples ; lphasor provides index into f1 alphs lphasor itrns, ilps, ilpe, imode, istrt atab tablei alphs, ifn outs atab, atab endin </CsInstruments> <CsScore> f 1 0 262144 1 "kickroll.wav" 0 4 1 ;stereo file in table, with lots of zeroes f 2 0 262144 1 "fox.wav" 0 4 1 ;mono file in table, with lots of zeroes i1 0 10 1 i1 + 10 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/chnmix.csd.xml0000644000000000000000000001154612262561504020150 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chnmix.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;Example by Joachim Heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1; send i-values chnset 1, "sio" chnset -1, "non" endin instr 2; send k-values kfreq randomi 100, 300, 1 chnset kfreq, "cntrfreq" kbw = kfreq/10 chnset kbw, "bandw" endin instr 3; send a-values anois rand .1 chnset anois, "noise" loop: idur random .3, 1.5 timout 0, idur, do reinit loop do: ifreq random 400, 1200 iamp random .1, .3 asig oscils iamp, ifreq, 0 aenv transeg 1, idur, -10, 0 asine = asig * aenv chnset asine, "sine" endin instr 11; receive some chn values and send again ival1 chnget "sio" ival2 chnget "non" print ival1, ival2 kcntfreq chnget "cntrfreq" kbandw chnget "bandw" anoise chnget "noise" afilt reson anoise, kcntfreq, kbandw afilt balance afilt, anoise chnset afilt, "filtered" endin instr 12; mix the two audio signals amix1 chnget "sine" amix2 chnget "filtered" chnmix amix1, "mix" chnmix amix2, "mix" endin instr 20; receive and reverb amix chnget "mix" aL, aR freeverb amix, amix, .8, .5 outs aL, aR endin instr 100; clear chnclear "mix" endin </CsInstruments> <CsScore> i 1 0 20 i 2 0 20 i 3 0 20 i 11 0 20 i 12 0 20 i 20 0 20 i 100 0 20 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKResonate.csd.xml0000644000000000000000000000421212262561504021014 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKResonate.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; frequency and amplitude of STKResonate have no effect kv2 = p4 ;pole radii kv1 line 100, p3, 0 ;resonance freq + notch freq kv3 = kv1 asig STKResonate 1, 1, 2, kv1, 4, kv2, 1, 10, 11, kv3, 128, 100 asig = asig * .3 ;too loud outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 0 i 1 + . > i 1 + . > i 1 + . > i 1 + . 120 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/array.csd.xml0000644000000000000000000000240612262561504017773 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kS[] = array(1,7,5) printk2 kS[0] printk2 kS[1] printk2 kS[2] endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sinsyn.csd.xml0000644000000000000000000000416412262561504020203 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sinsyn.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kmxtr init p4 ain diskin2 "fox.wav", 1 fs1,fsi2 pvsifd ain, 2048, 512,1 ; ifd analysis fst partials fs1, fsi2, .03, 1, 3, 500 ; partial tracking aout sinsyn fst, .5, kmxtr, 1 ; scale amplitude down outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 i 1 0 2.7 15 ;filtering effect by using low number of tracks i 1 + 2.7 500 ;maximum number of tracks e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/freeverb.csd.xml0000644000000000000000000000371712262561504020463 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o freeverb.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 a1 vco2 0.75, 440, 10 kfrq port 100, 0.008, 20000 a1 butterlp a1, kfrq a2 linseg 0, 0.003, 1, 0.01, 0.7, 0.005, 0, 1, 0 a1 = a1 * a2 denorm a1 aL, aR freeverb a1, a1, 0.9, 0.35, sr, 0 outs a1 + aL, a1 + aR endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/deltapx.csd.xml0000644000000000000000000000503612262561504020320 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o deltapx.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 a1 phasor 300 a1 = a1 - 0.5 a_ delayr 1 adel phasor 4 adel = sin(2 * 3.14159265 * adel) * 0.01 + 0.2 deltapxw a1, adel, 32 adel phasor 2 adel = sin(2 * 3.14159265 * adel) * 0.01 + 0.2 deltapxw a1, adel, 32 adel = 0.3 a2 deltapx adel, 32 a1 = 0 delayw a1 outs a2*.7, a2*.7 endin </CsInstruments> <CsScore> i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSsinnoi.csd.xml0000644000000000000000000000343212262561504020524 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSsinnoi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 2 knzfade expon 0.001, p3, 2 aout ATSsinnoi ktime, 1, knzfade, 1, "beats.ats", 150 outs aout*2, aout*2 ;amplify some more endin </CsInstruments> <CsScore> i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fofilter.csd.xml0000644000000000000000000000367012262561504020473 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fofilter.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfe expseg 10, p3*0.9, 180, p3*0.1, 175 kenv linen .1, 0.05, p3, 0.05 asig buzz kenv, kfe, sr/(2*kfe), 1 afil fofilter asig, 900, 0.007, 0.04 outs afil, afil endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/xtratim-2.csd.xml0000644000000000000000000001175612262561504020514 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in Silent MIDI in -odac -iadc -d -M0 ;;;realtime I/O </CsOptions> <CsInstruments> ;xtratim example by Jonathan Murphy Dec. 2006 sr = 44100 ksmps = 10 nchnls = 2 ; sine wave for oscillators gisin ftgen 1, 0, 4096, 10, 1 ; set volume initially to midpoint ctrlinit 1, 7,64 ;;; simple two oscil, two envelope synth instr 1 ; frequency kcps cpsmidib ; initial velocity (noteon) ivel veloc ; master volume kamp ctrl7 1, 7, 0, 127 kamp = kamp * ivel ; parameters for aenv1 iatt1 = 0.03 idec1 = 1 isus1 = 0.25 irel1 = 1 ; parameters for aenv2 iatt2 = 0.06 idec2 = 2 isus2 = 0.5 irel2 = 2 ; extra (release) time allocated xtratim (irel1>irel2 ? irel1 : irel2) ; krel is used to trigger envelope release krel init 0 krel release ; if noteoff received, krel == 1, otherwise krel == 0 if (krel == 1) kgoto rel ; attack, decay, sustain segments atmp1 linseg 0, iatt1, 1, idec1, isus1 , 1, isus1 atmp2 linseg 0, iatt2, 1, idec2, isus2 , 1, isus2 aenv1 = atmp1 aenv2 = atmp2 kgoto done ; release segment rel: atmp3 linseg 1, irel1, 0, 1, 0 atmp4 linseg 1, irel2, 0, 1, 0 aenv1 = atmp1 * atmp3 ;to go from the current value (in case aenv2 = atmp2 * atmp4 ;the attack hasn't finished) to the release. ; control oscillator amplitude using envelopes done: aosc1 oscil aenv1, kcps, gisin aosc2 oscil aenv2, kcps * 1.5, gisin aosc1 = aosc1 * kamp aosc2 = aosc2 * kamp ; send aosc1 to left channel, aosc2 to right, ; release times are noticably different outs aosc1, aosc2 endin </CsInstruments> <CsScore> f 0 3600 ;dummy table to wait for realtime MIDI events </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKSimple.csd.xml0000644000000000000000000000373112262561504020472 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKSimple.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;Filter Pole Position kv2 line 20, p3, 90 ;Noise/Pitched Cross-Fade asig STKSimple cpspch(p4), 1, 2, kv1, 4, kv2, 11, 10, 128, 120 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 5 7.00 100 0 i 1 + . 7.05 10 127 i 1 + . 8.03 100 0 i 1 + . 5.00 10 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vdelay.csd.xml0000644000000000000000000000431012262561504020135 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelay.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ims = 100 ;maximum delay time in msec aout poscil .8, 220, 1 ;make a signal a2 poscil3 ims/2, 1/p3, 1 ;make an LFO a2 = a2 + ims/2 ;offset the LFO so that it is positive asig vdelay aout, a2, ims ;use the LFO to control delay time outs asig, asig endin </CsInstruments> <CsScore> f1 0 8192 10 1 ;sine wave i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/poscil3-file.csd.xml0000644000000000000000000000520712262561504021150 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o poscil3-file.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Example written by Joachim Heintz 07/2008 sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; non-normalized function table with a sample 1 giFile ftgen 1, 0, 0, -1, "fox.wav", 0, 0, 0 ; Instrument #1 - uses poscil3 for playing samples from a function table instr 1 kamp = p4 kspeed = p5 ifn = 1 iskip = p6 kcps = kspeed / (ftlen(ifn) / ftsr(ifn)); frequency of the oscillator iphs = iskip / (ftlen(ifn) / ftsr(ifn)); calculates skiptime to phase values (0-1) a1 poscil3 kamp, kcps, ifn, iphs out a1 endin </CsInstruments> <CsScore> i1 0 2.756 1 1 0 i1 3 2.756 1 -1 0 i1 6 1.378 1 .5 2.067 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLxyin.csd.xml0000644000000000000000000000324112262561504020064 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O </CsOptions> <CsInstruments> sr=48000 ksmps=128 nchnls=2 ; Example by Andres Cabrera 2007 FLpanel "FLxyin", 200, 100, -1, -1, 3 FLpanelEnd FLrun instr 1 koutx, kouty, kinside FLxyin 0, 10, 100, 1000, 10, 190, 10, 90 aout buzz 10000, kouty, koutx, 1 printk2 koutx outs aout, aout endin </CsInstruments> <CsScore> f 1 0 1024 10 1 i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lposcilsa2.csd.xml0000644000000000000000000000457612262561504020742 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lposcilsa2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 2 ;only integers are allowed kloop = 0 ;loop start time in samples kend = 45000 ;loop end time in samples aenv expsega 0.01, 0.1, 1, 0.1, 0.5, 0.5, 0.01 ;envelope with fast and short segment aL, aR lposcilsa2 aenv, kcps, kloop, kend, 1 ;use it for amplitude outs aL, aR endin </CsInstruments> <CsScore> ; Its table size is deferred, ; and format taken from the soundfile header. f 1 0 0 1 "kickroll.wav" 0 0 0 ; Play Instrument #1 for 6 seconds. ; This will loop the drum pattern several times. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/modmatrix.csd.xml0000644000000000000000000002250412262561504020662 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio flags here according to platform ; Audio out Audio in ;-odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -o modmatrix.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 441 ksmps = 100 nchnls = 2 0dbfs = 1 ; basic waveforms giSine ftgen 0, 0, 65537, 10, 1 ; sine wave giSaw ftgen 0, 0, 4097, 7, 1, 4096, -1 ; saw (linear) giSoftSaw ftgen 0, 0, 65537, 30, giSaw, 1, 10 ; soft saw (only 10 first harmonics) ; modmatrix tables giMaxNumParam = 128 giMaxNumMod = 32 giParam_In ftgen 0, 0, giMaxNumParam, 2, 0 ; input parameters table ; output parameters table (parameter values with added modulators) giParam_Out ftgen 0, 0, giMaxNumParam, 2, 0 giModulators ftgen 0, 0, giMaxNumMod, 2, 0 ; modulators table ; modulation scaling and routing (mod matrix) table, start with empty table giModScale ftgen 0, 0, giMaxNumParam*giMaxNumMod, -2, 0 ;******************************************** ; generate the modulator signals ;******************************************** instr 1 ; LFO1, 1.5 Hz, normalized range (0.0 to 1.0) kLFO1 oscil 0.5, 1.5, giSine ; generate LFO signal kLFO1 = kLFO1+0.5 ; offset ; LFO2, 0.4 Hz, normalized range (0.0 to 1.0) kLFO2 oscil 0.5, 0.4, giSine ; generate LFO signal kLFO2 = kLFO2+0.5 ; offset ; write modulators to table tablew kLFO1, 0, giModulators tablew kLFO2, 1, giModulators endin ;******************************************** ; set parameter values ;******************************************** instr 2 ; Here we can set the parameter values icps1 = p4 icps2 = p5 icutoff = p6 ; write parameters to table tableiw icps1, 0, giParam_In tableiw icps2, 1, giParam_In tableiw icutoff, 2, giParam_In endin ;******************************************** ; mod matrix edit ;******************************************** instr 3 ; Here we can write to the modmatrix table by using tablew or tableiw iLfo1ToCps1 = p4 iLfo1ToCps2 = p5 iLfo1ToCutoff = p6 iLfo2ToCps1 = p7 iLfo2ToCps2 = p8 iLfo2ToCutoff = p9 tableiw iLfo1ToCps1, 0, giModScale tableiw iLfo1ToCps2, 1, giModScale tableiw iLfo1ToCutoff, 2, giModScale tableiw iLfo2ToCps1, 3, giModScale tableiw iLfo2ToCps2, 4, giModScale tableiw iLfo2ToCutoff, 5, giModScale ; and set the update flag for modulator matrix ; ***(must update to enable changes) ktrig init 1 chnset ktrig, "modulatorUpdateFlag" ktrig = 0 endin ;******************************************** ; mod matrix ;******************************************** instr 4 ; get the update flag kupdate chnget "modulatorUpdateFlag" ; run the mod matrix inum_mod = 2 inum_parm = 3 modmatrix giParam_Out, giModulators, giParam_In, \\ giModScale, inum_mod, inum_parm, kupdate ; and reset the update flag chnset 0, "modulatorUpdateFlag" ; reset the update flag endin ;******************************************** ; audio generator to test values ;******************************************** instr 5 ; basic parameters iamp = ampdbfs(-5) ; read modulated parameters from table kcps1 table 0, giParam_Out kcps2 table 1, giParam_Out kcutoff table 2, giParam_Out ; set filter parameters kCF_freq1 = kcps1*kcutoff kCF_freq2 = kcps2*kcutoff kReso = 0.7 kDist = 0.3 ; oscillators and filters a1 oscili iamp, kcps1, giSoftSaw a1 lpf18 a1, kCF_freq1, kReso, kDist a2 oscili iamp, kcps2, giSoftSaw a2 lpf18 a2, kCF_freq2, kReso, kDist outs a1, a2 endin </CsInstruments> <CsScore> ;******************************************** ; set initial parameters ; cps1 cps2 cutoff i2 0 1 400 800 3 ;******************************************** ; set modmatrix values ; lfo1ToCps1 lfo1ToCps2 lfo1ToCut lfo2ToCps1 lfo2ToCps2 lfo2ToCut i3 0 1 40 0 -2 -50 100 3 ;******************************************** ; start "always on" instruments #define SCORELEN # 20 # ; set length of score i1 0 $SCORELEN ; start modulators i4 0 $SCORELEN ; start mod matrix i5 0 $SCORELEN ; start audio oscillator e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/biquad.csd.xml0000644000000000000000000001161412262561504020123 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o biquad.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Instrument #1. instr 1 ; Get the values from the score. idur = p3 iamp = p4 icps = cpspch(p5) kfco = p6 krez = p7 ; Calculate the biquadratic filter's coefficients kfcon = 2*3.14159265*kfco/sr kalpha = 1-2*krez*cos(kfcon)*cos(kfcon)+krez*krez*cos(2*kfcon) kbeta = krez*krez*sin(2*kfcon)-2*krez*cos(kfcon)*sin(kfcon) kgama = 1+cos(kfcon) km1 = kalpha*kgama+kbeta*sin(kfcon) km2 = kalpha*kgama-kbeta*sin(kfcon) kden = sqrt(km1*km1+km2*km2) kb0 = 1.5*(kalpha*kalpha+kbeta*kbeta)/kden kb1 = kb0 kb2 = 0 ka0 = 1 ka1 = -2*krez*cos(kfcon) ka2 = krez*krez ; Generate an input signal. axn vco 1, icps, 1 ; Filter the input signal. ayn biquad axn, kb0, kb1, kb2, ka0, ka1, ka2 outs ayn*iamp/2, ayn*iamp/2 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Sta Dur Amp Pitch Fco Rez i 1 0.0 1.0 20000 6.00 1000 .8 i 1 1.0 1.0 20000 6.03 2000 .95 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/rnd31_seed7.csd.xml0000644000000000000000000000376512262561504020704 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd31_seed7.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; i-rate random numbers with linear distribution, seed=7. ; (Note that the seed was used only in the first call.) i1 rnd31 1, 0.5, 7 i2 rnd31 1, 0.5 i3 rnd31 1, 0.5 print i1 print i2 print i3 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/strset.csd.xml0000644000000000000000000000302012262561504020172 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O </CsOptions> <CsInstruments> sr = 44100 ksmps = 1 nchnls = 1 ;Example by Andres Cabrera 2008 ; \\n is used to denote "new line" strset 1, "String 1\\n" strset 2, "String 2\\n" instr 1 Str strget p4 prints Str endin </CsInstruments> <CsScore> ; p4 is used to select string i 1 0 1 1 i 1 3 1 2 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pgmassign_ignore.csd.xml0000644000000000000000000000347012262561504022212 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o pgmassign_ignore.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Ignore all program change events. pgmassign 0, -1 ; Instrument #1. instr 1 ; Just an example, no working code in here! endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/toot1.csd.xml0000644000000000000000000000175312262561504017727 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> instr 1 a1 oscil 10000, 440, 1 out a1 endin </CsInstruments> <CsScore> f1 0 4096 10 1 ; use GEN10 to compute a sine wave ;ins strt dur i1 0 4 e ; indicates the end of the score </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/spdist.csd.xml0000644000000000000000000000753612262561504020174 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o spdist.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 4 ga1 init 0 ga2 init 0 ga3 init 0 ga4 init 0 instr 1 ;uses GEN28 file "move2", as found in /manual/examples ifreq = 1 kx init 0 ky init 0 ktime line 0, 5.6, 5.6 ;same time as in table 1 (="move2") kdist spdist 1, ktime, kx, ky kfreq = (ifreq*340) / (340 + kdist) ;calculate doppler shift printk2 kdist ;print distance values asig diskin2 "flute.aiff", kfreq, 0, 1 ;sound source is looped a1, a2, a3, a4 space asig, 1, ktime, .1, kx, ky ;use table 1 = GEN28 ar1, ar2, ar3, ar4 spsend ;send to reverb ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 a3 reverb2 ga3, 2.5, .5 a4 reverb2 ga4, 2.5, .5 outq a1, a2, a3, a4 ga1=0 ga2=0 ga3=0 ga4=0 endin </CsInstruments> <CsScore> f1 0 0 28 "move2" ;from left front and left rear to the middle in front i 1 0 5.6 ;same time as ktime i 99 0 10 ;keep reverb active e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midichn_advanced.csd.xml0000644000000000000000000001007312262561504022114 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -F midichn_advanced.mid ;;;realtime audio out with MIDI file input ; For Non-realtime ouput leave only the line below: ; -o midichn_advanced.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 massign 1, 1 ; all channels use instr 1 massign 2, 1 massign 3, 1 massign 4, 1 massign 5, 1 massign 6, 1 massign 7, 1 massign 8, 1 massign 9, 1 massign 10, 1 massign 11, 1 massign 12, 1 massign 13, 1 massign 14, 1 massign 15, 1 massign 16, 1 gicnt = 0 ; note counter instr 1 gicnt = gicnt + 1 ; update note counter kcnt init gicnt ; copy to local variable ichn midichn ; get channel number istime times ; note-on time if (ichn > 0.5) goto l2 ; MIDI note printks "note %.0f (time = %.2f) was activated from the score\\n", \ 3600, kcnt, istime goto l1 l2: printks "note %.0f (time = %.2f) was activated from channel %.0f\\n", \ 3600, kcnt, istime, ichn l1: icps cpsmidi ; convert midi note to pitch kenv madsr 0.1, 0, 0.8, 0.9 asig pluck kenv, icps, icps, 1, 1 outs asig, asig endin </CsInstruments> <CsScore> t 0 60 ;beats per minute f 0 8 ;stay active for 8 seconds f 1 0 4096 10 1 ;sine e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/event_i.csd.xml0000644000000000000000000000546212262561504020313 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o event_i.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 gisine ftgen 0, 0, 2^10, 10, 1 instr 1 ;master instrument ininstr = 10 ;number of called instances indx = 0 loop: ipan random 0, 1 ifreq random 100, 1000 iamp = 1/ininstr event_i "i", 10, 0, p3, iamp, ifreq, ipan loop_lt indx, 1, ininstr, loop endin instr 10 print p4, p5, p6 ipeak random 0, 1 ;where is the envelope peak asig poscil3 p4, p5, gisine aenv transeg 0, p3*ipeak, 6, 1, p3-p3*ipeak, -6, 0 aL,aR pan2 asig*aenv, p6 outs aL, aR endin </CsInstruments> <CsScore> i1 0 10 i1 8 10 i1 16 15 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen02.csd.xml0000644000000000000000000000455312262561504017575 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen02.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ;choose different tables of GEN02 kcps init 1/p3 ;index over the length of entire note kndx phasor kcps ixmode = 1 ;normalize index data kamp tablei kndx, ifn, ixmode asig poscil kamp, 440, 1 ;use GEN02 as envelope for amplitude outs asig, asig endin </CsInstruments> <CsScore> f 1 0 8192 10 1 ;sine wave f 2 0 5 2 0 2 0 f 3 0 5 2 0 2 10 0 f 4 0 9 2 0 2 10 100 0 i 1 0 2 2 i 1 3 2 3 i 1 6 2 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/filepeak.csd.xml0000644000000000000000000000610012262561504020430 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o filepeak.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iscaldb = p4 ;set peak amplitude in dB ipeak filepeak "Church.wav" iscal = ampdb(iscaldb)/ipeak ;calculate amp multiplier printf_i "\nPeak value in file '%s' is %f (%.3f dB).\n\n", 1, "Church.wav", ipeak, dbamp(ipeak) asnd soundin "Church.wav" outs asnd, asnd ; scale & write file to disk asig = asnd*iscal ;scale to p4 fout "Church_norm.wav", 14, asig endin instr 2 ; play scaled file aout soundin "Church_norm.wav" ipknew filepeak "Church_norm.wav" printf_i "\nPeak value in file '%s' is %f (%.3f dB).\n\n", 1, "Church_norm.wav", ipknew, dbamp(ipknew) outs aout, aout endin </CsInstruments> <CsScore> i 1 0 2 -6 ; normalize audio to -6 dB i 2 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/p.csd.xml0000644000000000000000000000344512262561504017120 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o p.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Get the value in the fourth p-field, p4. i1 = p(4) print i1 endin </CsInstruments> <CsScore> ; p4 = value to be printed. ; Play Instrument #1 for one second, p4 = 50.375. i 1 0 1 50.375 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLexecButton.csd.xml0000644000000000000000000001720212262561504021217 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No display -odac -iadc -d ;;;RT audio I/O </CsOptions> <CsInstruments> sr = 44100 ksmps = 10 nchnls = 1 ; Example by Jonathan Murphy 2007 ;;; reset amplitude range 0dbfs = 1 ;;; set the base colour for the panel FLcolor 100, 0, 200 ;;; define the panel FLpanel "FLexecButton", 250, 100, 0, 0 ;;; sliders to control time stretch and pitch gkstr, gistretch FLslider "Time", 0.5, 1.5, 0, 6, -1, 10, 60, 150, 20 gkpch, gipitch FLslider "Pitch", 0.5, 1.5, 0, 6, -1, 10, 60, 200, 20 ;;; set FLexecButton colour FLcolor 255, 255, 0 ;;; when this button is pressed, fourier analysis is performed on the file ;;; "beats.wav", producing the analysis file "beats.pvx" gipvoc FLexecButton "csound -U pvanal beats.wav beats.pvx", 60, 20, 20, 20 ;;; set FLexecButton text FLsetText "PVOC", gipvoc ;;; when this button is pressed, instr 10000 is called, exiting ;;; Csound immediately ;;; cancel previous colour FLcolor -1 ;;; set colour for kill button FLcolor 255, 0, 0 gkkill, gikill FLbutton "X", 1, 1, 1, 20, 20, 100, 20, 0, 10000, 0, 0.1 ;;; cancel previous colour FLcolor -1 ;;; set colour for play/stop and pause buttons FLcolor 0, 200, 0 ;;; pause and play/stop buttons gkpause, gipause FLbutton "@||", 1, 0, 2, 40, 20, 20, 60, -1 gkplay, giplay FLbutton "@|>", 1, 0, 2, 40, 20, 80, 60, -1 ;;; end the panel FLpanelEnd ;;; set initial values for time stretch and pitch FLsetVal_i 1, gistretch FLsetVal_i 1, gipitch ;;; run the panel FLrun instr 1 ; trigger play/stop ;;; is the play/stop button on or off? ;;; either way we need to trigger something, ;;; so we can't just use the value of gkplay kon trigger gkplay, 0, 0 koff trigger gkplay, 1, 1 ;;; if on, start instr 2 schedkwhen kon, -1, -1, 2, 0, -1 ;;; if off, stop instr 2 schedkwhen koff, -1, -1, -2, 0, -1 endin instr 2 ;;; paused or playing? if (gkpause == 1) kgoto pause kgoto start pause: ;;; if the pause button is on, skip sound production kgoto end start: ;;; get the length of the analysis file in seconds ilen filelen "beats.pvx" ;;; determine base frequency of playback icps = 1/ilen ;;; create a table over the length of the file itpt ftgen 0, 0, 513, -7, 0, 512, ilen ;;; phasor for time control kphs phasor icps * gkstr ;;; use phasor as index into table kndx = kphs * 512 ;;; read table ktpt tablei kndx, itpt ;;; use value from table as time pointer into file fsig1 pvsfread ktpt, "beats.pvx" ;;; change playback pitch fsig2 pvscale fsig1, gkpch ;;; resynthesize aout pvsynth fsig2 ;;; envelope to avoid clicks and clipping aenv linsegr 0, 0.3, 0.75, 0.1, 0 aout = aout * aenv out aout end: endin instr 10000 ; kill exitnow endin </CsInstruments> <CsScore> i1 0 10000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sleighbells.csd.xml0000644000000000000000000000425112262561504021152 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sleighbells.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idamp = p4 asig sleighbells .7, 0.01, 32, idamp outs asig, asig endin </CsInstruments> <CsScore> i 1 0.00 0.25 0 ;short sound i 1 0.30 0.25 i 1 0.60 0.25 i 1 0.90 0.25 i 1 1.20 0.25 i 1 1.50 1 .3 ;longer sound i 1 1.80 0.25 0 ;short sound again i 1 2.10 0.25 i 1 2.40 0.25 i 1 2.70 0.25 i 1 3.00 0.25 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLslider-2.csd.xml0000644000000000000000000002456612262561504020533 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLslider-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 441 ksmps = 100 nchnls = 1 ;By Andres Cabrera 2007 FLpanel "Slider Types", 410, 260, 50, 50 ; Distance of the left edge of the slider ; from the left edge of the panel ix = 10 ; Distance of the top edge of the slider ; from the top edge of the panel iy = 10 ; Create boxes to display widget values givalue1 FLvalue "1", 60, 20, ix + 330, iy givalue3 FLvalue "3", 60, 20, ix + 330, iy + 40 givalue5 FLvalue "5", 60, 20, ix + 330, iy + 80 givalue2 FLvalue "2", 60, 20, ix + 60, iy + 140 givalue4 FLvalue "4", 60, 20, ix + 195, iy + 140 givalue6 FLvalue "6", 60, 20, ix + 320, iy + 140 ;Horizontal sliders gkdummy1, gihandle1 FLslider "Type 1", 200, 5000, -1, 1, givalue1, 320, 20, ix, iy gkdummy3, gihandle3 FLslider "Type 3", 0, 15000, 0, 3, givalue3, 320, 20, ix, iy + 40 ; Reversed slider gkdummy5, gihandle5 FLslider "Type 5", 1, 0, 0, 5, givalue5, 320, 20, ix, iy + 80 ;Vertical sliders gkdummy2, gihandle2 FLslider "Type 2", 0, 1, 0, 2, givalue2, 20, 100, ix+ 30 , iy + 120 ; Reversed slider gkdummy4, gihandle4 FLslider "Type 4", 1, 0, 0, 4, givalue4, 20, 100, ix + 165 , iy + 120 gkdummy6, gihandle6 FLslider "Type 6", 0, 1, 0, 6, givalue6, 20, 100, ix + 290 , iy + 120 FLpanelEnd FLpanel "Plastic Slider Types", 410, 300, 150, 150 ; Distance of the left edge of the slider ; from the left edge of the panel ix = 10 ; Distance of the top edge of the slider ; from the top edge of the panel iy = 10 ; Create boxes to display widget values givalue21 FLvalue "21", 60, 20, ix + 330, iy givalue23 FLvalue "23", 60, 20, ix + 330, iy + 40 givalue25 FLvalue "25", 60, 20, ix + 330, iy + 80 givalue22 FLvalue "22", 60, 20, ix + 60, iy + 140 givalue24 FLvalue "24", 60, 20, ix + 195, iy + 140 givalue26 FLvalue "26", 60, 20, ix + 320, iy + 140 ;Horizontal sliders gkdummy21, gihandle21 FLslider "Type 21", 200, 5000, -1, 21, givalue21, 320, 20, ix, iy gkdummy23, gihandle23 FLslider "Type 23", 0, 15000, 0, 23, givalue23, 320, 20, ix, iy + 40 ; Reversed slider gkdummy25, gihandle25 FLslider "Type 25", 1, 0, 0, 25, givalue25, 320, 20, ix, iy + 80 ;Vertical sliders gkdummy22, gihandle22 FLslider "Type 22", 0, 1, 0, 22, givalue22, 20, 100, ix+ 30 , iy + 120 ; Reversed slider gkdummy24, gihandle24 FLslider "Type 24", 1, 0, 0, 24, givalue24, 20, 100, ix + 165 , iy + 120 gkdummy26, gihandle26 FLslider "Type 26", 0, 1, 0, 26, givalue26, 20, 100, ix + 290 , iy + 120 ;Button to add color to the sliders gkcolors, ihdummy FLbutton "Color", 1, 0, 21, 150, 30, 30, 260, 0, 10, 0, 1 FLpanelEnd FLrun ;Set some widget's initial value FLsetVal_i 500, gihandle1 FLsetVal_i 1000, gihandle3 instr 10 ; Set the color of widgets FLsetColor 200, 230, 0, gihandle1 FLsetColor 0, 123, 100, gihandle2 FLsetColor 180, 23, 12, gihandle3 FLsetColor 10, 230, 0, gihandle4 FLsetColor 0, 0, 0, gihandle5 FLsetColor 0, 0, 0, gihandle6 FLsetColor 200, 230, 0, givalue1 FLsetColor 0, 123, 100, givalue2 FLsetColor 180, 23, 12, givalue3 FLsetColor 10, 230, 0, givalue4 FLsetColor 255, 255, 255, givalue5 FLsetColor 255, 255, 255, givalue6 FLsetColor2 20, 23, 100, gihandle1 FLsetColor2 200,0 ,123 , gihandle2 FLsetColor2 180, 180, 100, gihandle3 FLsetColor2 180, 23, 12, gihandle4 FLsetColor2 180, 180, 100, gihandle5 FLsetColor2 180, 23, 12, gihandle6 FLsetColor 200, 230, 0, gihandle21 FLsetColor 0, 123, 100, gihandle22 FLsetColor 180, 23, 12, gihandle23 FLsetColor 10, 230, 0, gihandle24 FLsetColor 0, 0, 0, gihandle25 FLsetColor 0, 0, 0, gihandle26 FLsetColor 200, 230, 0, givalue21 FLsetColor 0, 123, 100, givalue22 FLsetColor 180, 23, 12, givalue23 FLsetColor 10, 230, 0, givalue24 FLsetColor 255, 255, 255, givalue25 FLsetColor 255, 255, 255, givalue26 FLsetColor2 20, 23, 100, gihandle21 FLsetColor2 200,0 ,123 , gihandle22 FLsetColor2 180, 180, 100, gihandle23 FLsetColor2 180, 23, 12, gihandle24 FLsetColor2 180, 180, 100, gihandle25 FLsetColor2 180, 23, 12, gihandle26 ; Slider values must be updated for colors to change FLsetVal_i 250, gihandle1 FLsetVal_i 0.5, gihandle2 FLsetVal_i 0, gihandle3 FLsetVal_i 0, gihandle4 FLsetVal_i 0, gihandle5 FLsetVal_i 0.5, gihandle6 FLsetVal_i 250, gihandle21 FLsetVal_i 0.5, gihandle22 FLsetVal_i 500, gihandle23 FLsetVal_i 0, gihandle24 FLsetVal_i 0, gihandle25 FLsetVal_i 0.5, gihandle26 endin </CsInstruments> <CsScore> f 0 3600 ;Dumy table to make csound wait for realtime events e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/luaopcode.csd.xml0000644000000000000000000000753312262561504020636 0ustar rootroot <CsoundSynthesizer> <CsInstruments> lua_opdef "luatest", {{ local ffi = require("ffi") local string = require("string") local csoundLibrary = ffi.load('csound64.dll.5.2') ffi.cdef[[ int csoundGetKsmps(void *); double csoundGetSr(void *); struct luatest_arguments {double *out; double *stringout; char *stringin; double *in1; double *in2; double sr; int ksmps; }; ]] function luatest_init(csound, opcode, carguments) local arguments = ffi.cast("struct luatest_arguments *", carguments) arguments.sr = csoundLibrary.csoundGetSr(csound) print(string.format('stringin: %s', ffi.string(arguments.stringin))) print(string.format('sr: %f', arguments.sr)) arguments.ksmps = csoundLibrary.csoundGetKsmps(csound) print(string.format('ksmps: %d', arguments.ksmps)) arguments.out[0] = arguments.in1[0] * arguments.in2[0] ffi.copy(arguments.stringout, 'Hello, world!') return 0 end function luatest_kontrol(csound, opcode, carguments) local arguments = ffi.cast("struct luatest_arguments *", carguments) arguments.out[0] = arguments.in1[0] * arguments.in2[0] return 0 end function luatest_noteoff(csound, opcode, carguments) local arguments = ffi.cast("struct luatest_arguments *", carguments) arguments.out[0] = arguments.in1[0] * arguments.in2[0] print('off') return 0 end }} instr 1 iresult = 0 Stringin = "stringin" Stringout = "stringout" lua_iopcall "luatest", iresult, Stringout, Stringin, p2, p3 prints Stringout endin instr 2 iresult = 0 Stringin = "stringin" Stringout = "initial value" lua_iopcall_off "luatest", iresult, Stringout, Stringin, p2, p3 print iresult prints Stringout endin </CsInstruments> <CsScore> i 1 1 2 i 2 2 2 i 1 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/strcatk.csd.xml0000644000000000000000000000431112262561504020325 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o strcatk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 48000 ksmps = 16 nchnls = 2 0dbfs = 1 ; Example by Jonathan Murphy 2007 instr 1 S1 = "1" S2 = " + 1" ktrig init 0 kval init 2 if (ktrig == 1) then S1 strcatk S1, S2 kval = kval + 1 endif String sprintfk "%s = %d", S1, kval puts String, kval ktrig metro 1 endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/spsend.csd.xml0000644000000000000000000000676112262561504020161 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o spsend.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 ;stereo output ga1 init 0 ga2 init 0 instr 1 ;sends different amounts to reverb irev = p6 asig diskin2 "fox.wav", 1 a1, a2, a3, a4 space asig, 0, 0, irev, p4, p5 ;take position values from p4, p5 ar1, ar2, ar3, ar4 spsend ;send to reverb ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a2 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 outs a1, a2 ga1=0 ga2=0 endin </CsInstruments> <CsScore> ;WITH REVERB ;place the sound in the left speaker and near i1 0 1 -1 1 .1 ;place the sound in the right speaker and far i1 1 1 45 45 .1 ;place the sound equally between left and right and in the middle ground distance i1 2 1 0 12 .1 ;NO REVERB ;place the sound in the left speaker and near i1 6 1 -1 1 0 ;place the sound in the right speaker and far i1 7 1 45 45 0 ;place the sound equally between left and right and in the middle ground distance i1 8 1 0 12 0 i 99 0 12 ;keep reverb active all the time e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/event_named.csd.xml0000644000000000000000000000614512262561504021146 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o event_named.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - an oscillator with a high note. instr 1 ; Create a trigger and set its initial value to 1. ktrigger init 1 ; If the trigger is equal to 0, continue playing. ; If not, schedule another event. if (ktrigger == 0) goto contin ; kscoreop="i", an i-statement. ; kinsnum="low_note", instrument named "low_note". ; kwhen=1, start at 1 second. ; kdur=0.5, play for a half-second. event "i", "low_note", 1, 0.5 ; Make sure the event isn't triggered again. ktrigger = 0 contin: a1 oscils 10000, 440, 1 out a1 endin ; Instrument "low_note" - an oscillator with a low note. instr low_note a1 oscils 10000, 220, 1 out a1 endin </CsInstruments> <CsScore> ; Make sure the score plays for two seconds. f 0 2 ; Play Instrument #1 for a half-second. i 1 0 0.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/bbcutm.csd.xml0000644000000000000000000000460212262561504020131 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o bbcutm.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - Play an audio file normally. instr 1 asource soundin "beats.wav" out asource endin ; Instrument #2 - Cut-up an audio file. instr 2 asource soundin "beats.wav" ibps = 4 isubdiv = 8 ibarlength = 4 iphrasebars = 1 inumrepeats = 2 a1 bbcutm asource, ibps, isubdiv, ibarlength, iphrasebars, inumrepeats out a1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mandol.csd.xml0000644000000000000000000000443512262561504020133 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mandol.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = p4 ksize = p5 kdetune = p6 asig mandol kamp, 880, .4, kdetune, 0.99, ksize, 1, 220 outs asig, asig endin </CsInstruments> <CsScore> ; "mandpluk.aiff" audio file f 1 0 8192 1 "mandpluk.aiff" 0 0 0 i 1 .5 1 1 2 .99 i 1 + 1 .5 1 .99 ;lower volume to compensate i 1 + 3 .3 .3 .99 ;lower volume to compensate i 1 4 1 1 2 .39 ;change detune value i 1 + 1 .5 1 .39 i 1 + 3 .3 .3 .39 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dates.csd.xml0000644000000000000000000000601212262561504017752 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dates.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 ;each time different seed instr 1 ;;generating a different filename each time csound renders itim date Stim dates itim Syear strsub Stim, 20, 24 Smonth strsub Stim, 4, 7 Sday strsub Stim, 8, 10 iday strtod Sday Shor strsub Stim, 11, 13 Smin strsub Stim, 14, 16 Ssec strsub Stim, 17, 19 Sfilnam sprintf "%s_%s_%02d_%s_%s_%s.wav", Syear, Smonth, iday, Shor,Smin, Ssec ;;rendering with random frequency, amp and pan, and writing to disk ifreq random 400, 1000 iamp random .1, 1 ipan random 0, 1 asin oscils iamp, ifreq, 0 aL, aR pan2 asin, ipan fout Sfilnam, 14, aL, aR outs aL, aR printf_i "File '%s' written to the same directory as this CSD file is!\n", 1, Sfilnam endin </CsInstruments> <CsScore> i 1 0 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/OSCmidircv.csd.xml0000644000000000000000000001312112262561504020653 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 1 ; Example by Jonathan Murphy and Andres Cabrera 2007 ; Use file OSCmidisend.csd to generate OSC events for this file 0dbfs = 1 gilisten OSCinit 47120 gisin ftgen 1, 0, 16384, 10, 1 givel ftgen 2, 0, 128, -2, 0 gicc ftgen 3, 0, 128, -7, 100, 128, 100 ;Default all controllers to 100 ;Define scale tuning giji_12 ftgen 202, 0, 32, -2, 12, 2, 256, 60, 1, 16/15, 9/8, 6/5, 5/4, 4/3, 7/5, \ 3/2, 8/5, 5/3, 9/5, 15/8, 2 #define DEST #"/midi"# ; Use controller number 7 for volume #define VOL #7# turnon 1000 instr 1000 kst init 0 kch init 0 kd1 init 0 kd2 init 0 next: kk OSClisten gilisten, $DEST, "iiii", kst, kch, kd1, kd2 if (kk == 0) goto done printks "kst = %i, kch = %i, kd1 = %i, kd2 = %i\\n", \ 0, kst, kch, kd1, kd2 if (kst == 176) then ;Store controller information in a table tablew kd2, kd1, gicc endif if (kst == 144) then ;Process noteon and noteoff messages. kkey = kd1 kvel = kd2 kcps cpstun kvel, kkey, giji_12 kamp = kvel/127 if (kvel == 0) then turnoff2 1001, 4, 1 elseif (kvel > 0) then event "i", 1001, 0, -1, kcps, kamp endif endif kgoto next ;Process all events in queue done: endin instr 1001 ;Simple instrument icps init p4 kvol table $VOL, gicc ;Read MIDI volume from controller table kvol = kvol/127 aenv linsegr 0, .003, p5, 0.03, p5 * 0.5, 0.3, 0 aosc oscil aenv, icps, gisin out aosc * kvol endin </CsInstruments> <CsScore> f 0 3600 ;Dummy f-table e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/spat3d_quad.csd.xml0000644000000000000000000001773712262561504021102 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o spat3d_quad.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Istvan Varga */ sr = 48000 kr = 1000 ksmps = 48 nchnls = 4 /* room parameters */ idep = 3 /* early reflection depth */ itmp ftgen 1, 0, 64, -2, \ /* depth1, depth2, max delay, IR length, idist, seed */ \ idep, 48, -1, 0.01, 0.25, 123, \ 1, 21.982, 0.05, 0.87, 4000.0, 0.6, 0.7, 2, /* ceil */ \ 1, 1.753, 0.05, 0.87, 3500.0, 0.5, 0.7, 2, /* floor */ \ 1, 15.220, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* front */ \ 1, 9.317, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* back */ \ 1, 17.545, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* right */ \ 1, 12.156, 0.05, 0.87, 5000.0, 0.8, 0.7, 2 /* left */ instr 1 /* some source signal */ a1 phasor 150 ; oscillator a1 butterbp a1, 500, 200 ; filter a1 = taninv(a1 * 100) a2 phasor 3 ; envelope a2 mirror 40*a2, -100, 5 a2 limit a2, 0, 1 a1 = a1 * a2 * 9000 kazim line 0, 2.5, 360 ; move sound source around kdist line 1, 10, 4 ; distance ; convert polar coordinates kX = sin(kazim * 3.14159 / 180) * kdist kY = cos(kazim * 3.14159 / 180) * kdist kZ = 0 a1 = a1 + 0.000001 * 0.000001 ; avoid underflows imode = 2 ; change this to 3 for 8 spk in a cube, ; or 1 for simple stereo aW, aX, aY, aZ spat3d a1, kX, kY, kZ, 1.0, 1, imode, 2, 2 aW = aW * 1.4142 ; stereo ; ;aL = aW + aY /* left */ ;aR = aW - aY /* right */ ; quad (square) ; aFL = aW + aX + aY /* front left */ aFR = aW + aX - aY /* front right */ aRL = aW - aX + aY /* rear left */ aRR = aW - aX - aY /* rear right */ ; eight channels (cube) ; ;aUFL = aW + aX + aY + aZ /* upper front left */ ;aUFR = aW + aX - aY + aZ /* upper front right */ ;aURL = aW - aX + aY + aZ /* upper rear left */ ;aURR = aW - aX - aY + aZ /* upper rear right */ ;aLFL = aW + aX + aY - aZ /* lower front left */ ;aLFR = aW + aX - aY - aZ /* lower front right */ ;aLRL = aW - aX + aY - aZ /* lower rear left */ ;aLRR = aW - aX - aY - aZ /* lower rear right */ outq aFL, aFR, aRL, aRR endin </CsInstruments> <CsScore> /* Written by Istvan Varga */ t 0 60 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zarg.csd.xml0000644000000000000000000000560512262561504017624 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zarg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Generate a simple sine waveform, with an amplitude ; between 0 and 1. asin oscil 1, 440, 1 ; Send the sine waveform to za variable #1. zaw asin, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read za variable #1, multiply its amplitude by 20,000. a1 zarg 1, 20000 ; Generate audio output. out a1 ; Clear the za variables, get them ready for ; another pass. zacl 0, 1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tan.csd.xml0000644000000000000000000000326212262561504017440 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tan.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 25 i1 = tan(irad) print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midinoteoff.csd.xml0000644000000000000000000000432512262561504021162 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M1 -Q1 ;;;realtime audio out and midi in and midi out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midinoteoff.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;uses external midi device kkey init 0 ;initialize key number kvel init 0 ;initialize velocity midinoteoff kkey,kvel ;MIDI noteoff value printk2 kvel ;display noteoff value midion 1, kkey, kvel ;sent note to external device endin </CsInstruments> <CsScore> f 0 30 ;runs for 30 seconds i 1 0 2 62 ; play these notes from score as well i 1 + 2 65 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/delayk.csd.xml0000644000000000000000000000540712262561504020132 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o delayk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;example shows "delayk" for fm index and ;a second "delayk" for panning sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 0, 0, 2^10, 10, 1 instr 1 kenv1 transeg 0, .02, 0, 1, 3.98, -6, 0 ;envelope kenv2 delayk kenv1, 2 ;delayed by two seconds kindx expon 5, p3, 1 ;fm index decreasing over p3 asig foscili .6, 400, 1, 11/4, kindx, gisin kpan1 linseg 0, 4, 1 ;panning for first sound kpan2 linseg 1, 4, 0 ;panning for second sound ... kpan2 delayk kpan2, 2 ;delayed by two seconds a1 = asig * kenv1 a2 = asig * kenv2 aL1,aR1 pan2 a1, kpan1 aL2,aR2 pan2 a2, kpan2 outs aL1+aL2, aR1+aR2 endin </CsInstruments> <CsScore> i 1 0 6 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/convolve.csd.xml0000644000000000000000000001153712262561504020515 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ;-odac ;;;RT audio out -iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o convolve.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; NB: 'Small' reverbs often require a much higher ; percentage of wet signal to sound interesting. 'Large' ; reverbs seem require less. Experiment! The wet/dry mix is ; very important - a small change can make a large difference. sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;The analysis file is not system independent! ; create "rv_mono.wav" and "rv_stereo.wav" with cvanal first! instr 1 imix = 0.25 ;wet/dry mix. Vary as desired. ivol = 1 ;Overall volume level of reverb. May need to adjust ;when wet/dry mix is changed, to avoid clipping. idel filelen p4 ;calculate length and number of channels of soundfile print idel ichnls filenchnls p4 print ichnls if (ichnls == 1) then adry soundin "fox.wav" ; input (dry) audio awet convolve adry,"rv_mono.cva" ; mono convolved (wet) audio awet diff awet ; brighten adrydel delay (1-imix)*adry, idel ; Delay dry signal to align it with convolved signal ; Apply level adjustment here too. outs ivol*(adrydel+imix*awet),ivol*(adrydel+imix*awet) ; Mix wet & dry else adry soundin "fox.wav" ; input (dry) audio awet1, awet2 convolve adry,"rv_stereo.cva" ; stereo convolved (wet) audio awet1 diff awet1 ; brighten left awet2 diff awet2 ; and brighten right adrydel delay (1-imix)*adry, idel ; Delay dry signal to align it with convolved signal ; Apply level adjustment here too. outs ivol*(adrydel+imix*awet1),ivol*(adrydel+imix*awet2) ; Mix wet & dry signals endif endin </CsInstruments> <CsScore> i 1 0 4 "rv_mono.wav" i 1 5 4 "rv_stereo.wav" e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hrtfstat.csd.xml0000644000000000000000000000510012262561504020506 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o hrtfstat.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasrc init 0 instr 1 ;a plucked string kamp = p4 kcps = cpspch(p5) icps = cpspch(p5) a1 pluck kamp, kcps, icps, 0, 1 gasrc = a1 endin instr 10;uses output from instr1 as source aleft,aright hrtfstat gasrc, 90,0, "hrtf-44100-left.dat","hrtf-44100-right.dat" outs aleft, aright clear gasrc endin </CsInstruments> <CsScore> i1 0 2 .7 8.00 ; Play Instrument 1: a plucked string i1 .5 2 .7 8.00 i1 1 2 .7 8.00 i1 2 2 .7 7.00 i10 0 12 ; Play Instrument 10 for 2 seconds. </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/JackoInit.csd.xml0000644000000000000000000000174712262561504020537 0ustar rootroot <CsoundSynthesizer> <CsOptions> -n </CsOptions> <CsInstruments> sr = 48000 ksmps = 128 nchnls = 2 0dbfs = 1 instr 1 JackoInit "default", "csound" JackoInfo endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/copyf2array.csd.xml0000644000000000000000000000370412262561504021120 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;8 points sine wave function table giSine ftgen 0, 0, 8, 10, 1 instr 1 ;create array kArr[] init 8 ;copy table values in it copyf2array kArr, giSine ;print values kndx = 0 until kndx == lenarray(kArr) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr[kndx] kndx += 1 od ;turn instrument off turnoff endin </CsInstruments> <CsScore> i 1 0 0.1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vbap4move.csd.xml0000644000000000000000000000352712262561504020565 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbap4move.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 vbaplsinit 2, 4, 0, 90, 180, 270 instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav a1,a2,a3,a4 vbap4move asig, p3, 1, 2, 310, 180 ;change movement of soundsource in outq a1,a2,a3,a4 ;the rear speakers endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKSitar.csd.xml0000644000000000000000000000350512262561504020322 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKSitar.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;STKSitar - has no controllers ifrq = p4 asig STKSitar cpspch(p4), 1 asig = asig * 2 ;amplify outs asig, asig endin </CsInstruments> <CsScore> i 1 0 4 6.00 i 1 + 2 7.05 i 1 + 7 5.05 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftlen.csd.xml0000644000000000000000000000400512262561504017762 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftlen.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 instr 1 ift = ftlen(p4) print ift aout loscil3 .8, 4, p4 outs aout, aout endin </CsInstruments> <CsScore> f 1 0 0 1 "fox.wav" 0 0 0 ;Csound computes tablesize f 2 0 0 1 "beats.wav" 0 0 0 ;Csound computes tablesize i 1 0 3 1 ;"fox.wav" i 1 3 3 2 ;"beats.wav" e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsadsyn.csd.xml0000644000000000000000000000724312262561504020530 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsadsyn.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 opcode FileToPvsBuf, iik, Siiii ;;writes an audio file at the first k-cycle to a fft-buffer (via pvsbuffer) Sfile, ifftsize, ioverlap, iwinsize, iwinshape xin ktimek timeinstk if ktimek == 1 then ilen filelen Sfile kcycles = ilen * kr; number of k-cycles to write the fft-buffer kcount init 0 loop: ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape ibuf, ktim pvsbuffer fftin, ilen + (ifftsize / sr) loop_lt kcount, 1, kcycles, loop xout ibuf, ilen, ktim endif endop instr 1 istretch = p4; time stretching factor ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window ibuffer, ilen, k0 FileToPvsBuf "fox.wav", ifftsize, ioverlap, iwinsize, iwinshape p3 = istretch * ilen; set p3 to the correct value ktmpnt linseg 0, p3, ilen; time pointer fread pvsbufread ktmpnt, ibuffer; read the buffer aout pvsadsyn fread, 10, 1; resynthesis with the first 10 bins out aout endin </CsInstruments> <CsScore> i 1 0 1 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/moogladder.csd.xml0000644000000000000000000000350712262561504020775 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o moogladder.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfe expseg 500, p3*0.9, 1800, p3*0.1, 3000 asig buzz 1, 100, 20, 1 kres line .1, p3, .99 ;increase resonance afil moogladder asig, kfe, kres outs afil, afil endin </CsInstruments> <CsScore> f 1 0 4096 10 1 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wgpluck.csd.xml0000644000000000000000000000415112262561504020330 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o wgpluck.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 icps = 220 iamp = 20000 kpick = 0.5 iplk = 0 idamp = 10 ifilt = 1000 axcite oscil 1, 1, 1 apluck wgpluck icps, iamp, kpick, iplk, idamp, ifilt, axcite out apluck endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLkeyIn.csd.xml0000644000000000000000000000400512262561504020153 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 ;Example by Andres Cabrera 2007 FLpanel "FLkeyIn", 400, 300, -1, -1, 5, 1, 1 FLpanelEnd FLrun 0dbfs = 1 instr 1 kascii FLkeyIn ktrig changed kascii if (kascii > 0) then printf "Key Down: %i\n", ktrig, kascii else printf "Key Up: %i\n", ktrig, -kascii endif endin </CsInstruments> <CsScore> i 1 0 120 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vcella.csd.xml0000644000000000000000000001607112262561504020126 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vcella.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; vcella.csd ; by Anthony Kozar ; This file demonstrates some of the new opcodes available in ; Csound 5 that come from Gabriel Maldonado's CsoundAV. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Cellular automata-driven oscillator bank using vcella and adsynt instr 1 idur = p3 iCArate = p4 ; number of times per second the CA calculates new values ; f-tables for CA parameters iCAinit = p5 ; CA initial states iCArule = p6 ; CA rule values ; The rule is used as follows: ; the states (values) of each cell are summed with their neighboring cells within ; the specied radius (+/- 1 or 2 cells). Each sum is used as an index to read a ; value from the rule table which becomes the new state value for its cell. ; All new states are calculated first, then the new values are all applied ; simultaneously. ielements = ftlen(iCAinit) inumrules = ftlen(iCArule) iradius = 1 ; create some needed tables iCAstate ftgen 0, 0, ielements, -2, 0 ; will hold the current CA states ifreqs ftgen 0, 0, ielements, -2, 0 ; will hold the oscillator frequency for each cell iamps ftgen 0, 0, ielements, -2, 0 ; will hold the amplitude for each cell ; calculate cellular automata state ktrig metro iCArate ; trigger the CA to update iCArate times per second vcella ktrig, 0, iCAstate, iCAinit, iCArule, ielements, inumrules, iradius ; scale CA state for use as amplitudes of the oscillator bank vcopy iamps, iCAstate, ielements vmult iamps, (1/3), ielements ; divide by 3 since state values are 0-3 vport iamps, .01, ielements ; need to smooth the amplitude changes for adsynt ; we could use adsynt2 instead of adsynt, but it does not seem to be working ; i-time loop for calculating frequencies index = 0 inew = 1 iratio = 1.125 ; just major second (creating a whole tone scale) loop1: tableiw inew, index, ifreqs, 0 ; 0 indicates integer indices inew = inew * iratio index = index + 1 if (index < ielements) igoto loop1 ; create sound with additive oscillator bank ifreqbase = 64 iwavefn = 1 iphs = 2 ; random oscillator phases kenv linseg 0.0, 0.5, 1.0, idur - 1.0, 1.0, 0.5, 0.0 aosc adsynt kenv, ifreqbase, iwavefn, ifreqs, iamps, ielements, iphs out aosc * ampdb(68) endin </CsInstruments> <CsScore> f1 0 16384 10 1 ; This example uses a 4-state cellular automata ; Possible state values are 0, 1, 2, and 3 ; CA initial state ; We have 16 cells in our CA, so the initial state table is size 16 f10 0 16 -2 0 1 0 0 1 0 0 2 2 0 0 1 0 0 1 0 ; CA rule ; The maximum sum with radius 1 (3 cells) is 9, so we need 10 values in the rule (0-9) f11 0 16 -2 1 0 3 2 1 0 0 2 1 0 ; Here is our one and only note! i1 0 20 4 10 11 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSaddnz.csd.xml0000644000000000000000000000331112262561504020321 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc for RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSaddnzwav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 2 asig ATSaddnz ktime, "beats.ats", 1, 24 outs asig*10, asig*10 ;amplify endin </CsInstruments> <CsScore> i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen19.csd.xml0000644000000000000000000000500512262561504017576 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen19.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;p4 = transposition factor ;p5 = speed factor ;p6 = function table for grain data i1 = sr/ftlen(1) ;scaling to reflect sample rate and table length a1 phasor i1*p5 ;index for speed asigl fog .5, 15, p4, a1, 1, 0, .01, .5, .01, 30, 1, 2, p3 ;left channel asigr fog .4, 25, p4+.2, a1, 1, 0, .01, .5, .01, 30, 1, 2, p3, .5 ;right channel outs asigl, asigr endin </CsInstruments> <CsScore> f 1 0 131072 1 "fox.wav" 0 0 0 f 2 0 1024 19 .5 .5 270 .5 i 1 0 10 .7 .1 i 1 + 4 1.2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/envlpxr.csd.xml0000644000000000000000000000353412262561504020356 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o envlpxr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .5 kenv envlpxr iamp, 0.2, 1, 1, 1, .01 asig pluck kenv, icps, 200, 2, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 129 -7 0 128 1 f 2 0 4096 10 1 f0 30 ;runs 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outs.csd.xml0000644000000000000000000000443112262561504017647 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig vco2 .01, 110 ; sawtooth waveform at low volume ;filter a channel kcut1 line 60, p3, 300 ; Vary cutoff frequency kresonance1 = 3 inumlayer1 = 3 asig1 lowresx asig, kcut1, kresonance1, inumlayer1 ;filter the other channel kcut2 line 300, p3, 60 ; Vary cutoff frequency kresonance2 = 3 inumlayer2 = 3 asig2 lowresx asig, kcut2, kresonance2, inumlayer2 outs asig1, asig2 ; output both channels 1 & 2 endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/frac.csd.xml0000644000000000000000000000332312262561504017567 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o frac.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = 16 / 5 i2 = frac(i1) print i2 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cps2pch.csd.xml0000644000000000000000000000343412262561504020221 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cps2pch.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a normal twelve-tone scale. ipch = 8.02 iequal = 12 icps cps2pch ipch, iequal print icps endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vibes.csd.xml0000644000000000000000000000456412262561504017774 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vibes.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 ksmps = 10 nchnls = 2 ; Instrument #1. instr 1 ; kamp = 20000 ; kfreq = 440 ; ihrd = 0.5 ; ipos = p4 ; imp = 1 ; kvibf = 6.0 ; kvamp = 0.05 ; ivibfn = 2 ; idec = 0.1 asig vibes 20000, 440, .5, p4 , 1, 6.0, 0.05, 2, .1 outs asig, asig endin </CsInstruments> <CsScore> ; Table #1, the "marmstk1.wav" audio file. f 1 0 256 1 "marmstk1.wav" 0 0 0 ; Table #2, a sine wave for the vibrato. f 2 0 128 10 1 ; Play Instrument #1 for four seconds. i 1 0 4 0.561 i 1 + 4 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/copya2ftab.csd.xml0000644000000000000000000000725412262561504020715 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;an 'empty' function table with 10 points giTable ftgen 0, 0, -10, 2, 0 instr 1 ;print inital values of giTable puts "\nInitial table content:", 1 indx = 0 until indx == ftlen(giTable) do iVal table indx, giTable printf_i "Table index %d = %f\n", 1, indx, iVal indx += 1 od ;create array kArr[] init 10 ;fill in values kArr genarray 1, 10 ;print array values printf "%s", 1, "\nArray content:\n" kndx = 0 until kndx == lenarray(kArr) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr[kndx] kndx += 1 od ;copy array values to table copya2ftab kArr, giTable ;print modified values of giTable printf "%s", 1, "\nModified table content after copya2ftab:\n" kndx = 0 until kndx == ftlen(giTable) do kVal table kndx, giTable printf "Table index %d = %f\n", kndx+1, kndx, kVal kndx += 1 od ;turn instrument off turnoff endin </CsInstruments> <CsScore> i 1 0 0.1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/xscanu.csd.xml0000644000000000000000000001360212262561504020156 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o xscanu.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;the matrices can be found in /manual/examples instr 1 ; Plain scanned syntnesis ; Note Also that I am using quadratic interpolation on these. a0 = 0 xscanu 1, .01, 6, 2, "128,8-gridX", 4, 5, 2, .1, .1, -.01, .1, .5, 0, 0, a0, 0, 0 a1 xscans .5, cpspch(p4), 333, 0, p6 ; NOTE LEFT RIGHT TRAJECTORY (f333) IS CLEAN! a1 dcblock a1 outs a1, a1 endin instr 2 ; Scan synthesis with audio injection and dual scan paths a0 diskin2 "fox.wav",1,0,1 ; a0,aa ins a0 = a0/.8 xscanu 1, .01, 6, 2, "128,8-torusX", 14, 5, 2, .01, .05, -.05, .1, .5, 0, 0, a0, 0, 0 a1 xscans .3, cpspch(7.00), 333, 0, 2 ; NOTE LEFT RIGHT TRAJECTORY (f333) IS CLEAN! a2 xscans .3, cpspch(6.00), 77, 0, 2 a1 dcblock a1 a2 dcblock a2 outs a1*.5,a2*.1 endin </CsInstruments> <CsScore> ; Initial condition ;f1 0 16 7 0 8 1 8 0 f1 0 128 7 0 64 1 64 0 ; Masses f2 0 128 -7 1 128 1 ; Centering force f4 0 128 -7 0 128 2 f14 0 128 -7 2 64 0 64 2 ; Damping f5 0 128 -7 1 128 1 ; Initial velocity f6 0 128 -7 -.0 128 .0 ; Trajectories f7 0 128 -5 .001 128 128 f77 0 128 -23 "128-spiral-8,16,128,2,1over2" f777 0 128 -23 "128,8-torusX" ; Spring matrices f3 0 128 -23 "128-stringX" f33 0 128 -23 "128-stringcircularX" f333 0 128 -23 "128-left_rightX" f3333 0 128 -23 "128,8-torusX" f33333 0 128 -23 "128,8-cylinderX" f333333 0 128 -23 "128,8-gridX" ; Sine f9 0 1024 10 1 ; Pitch tables f100 0 1024 -7 +3 128 +3 128 -2 128 -2 128 +0 128 +0 128 -4 128 -4 128 +3 f101 0 1024 -7 -2 128 -2 128 -2 128 -2 128 -5 128 -5 128 -4 128 -4 128 -2 f102 0 1024 -7 +3 128 +3 128 +2 128 +2 128 +0 128 +0 128 +0 128 +0 128 +3 f103 0 1024 -7 +7 128 +7 128 +5 128 +5 128 +3 128 +3 128 +3 128 +3 128 +7 ; Amplitude tables f200 0 1024 7 1 128 0 128 0 127 0 1 1 128 0 128 0 127 0 1 1 128 0 127 0 1 1 f201 0 1024 7 0 127 0 1 1 127 0 1 1 128 0 127 0 1 1 127 0 1 1 128 0 127 0 1 1 127 0 1 1 f202 0 1024 7 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 1 127 0 1 f203 0 1024 7 1 1024 0 ;;f204 0 1024 7 1 512 0 511 0 1 1 ;----------------------- ; Note list i1 0 10 6.00 1 2 s i2 0 15 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/log2.csd.xml0000644000000000000000000000320612262561504017517 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o log10.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = log2(8) print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvoc.csd.xml0000644000000000000000000000347012262561504017626 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvoc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; analyze "fox.wav" with PVANAL first ispec = p4 ktime line 0, p3, 1.55 kfrq line .8, p3, 2 asig pvoc ktime, kfrq, "fox.pvx", ispec outs asig, asig endin </CsInstruments> <CsScore> i 1 0 6 0 i 1 + 6 1 ;preserve spectral envelope e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluidLoad.csd.xml0000644000000000000000000000472212262561504020563 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidLoad.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "07AcousticGuitar.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 0 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 7 asigl, asigr fluidOut giengine outs asigl*imvol, asigr*imvol endin </CsInstruments> <CsScore> i 1 0 2 60 100 ;play one note from score and... i 99 0 60 ;play virtual keyboard for 60 sec. e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/atonex.csd.xml0000644000000000000000000000413612262561504020155 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o atonex.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; unfiltered noise asig rand 0.7 ; white noise outs asig, asig endin instr 2 ; filtered noise asig rand 0.7 khp line 100, p3, 3000 afilt atonex asig, khp, 32 ; Clip the filtered signal's amplitude to 85 dB. a1 clip afilt, 2, ampdb(85) outs a1, a1 endin </CsInstruments> <CsScore> i 1 0 2 i 2 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/nestedap.csd.xml0000644000000000000000000000736412262561504020470 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o nestedap.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 5 insnd = p4 gasig diskin2 insnd, 1 endin instr 10 imax = 1 idel1 = p4/1000 igain1 = p5 idel2 = p6/1000 igain2 = p7 idel3 = p8/1000 igain3 = p9 idel4 = p10/1000 igain4 = p11 idel5 = p12/1000 igain5 = p13 idel6 = p14/1000 igain6 = p15 afdbk init 0 aout1 nestedap gasig+afdbk*.4, 3, imax, idel1, igain1, idel2, igain2, idel3, igain3 aout2 nestedap aout1, 2, imax, idel4, igain4, idel5, igain5 aout nestedap aout2, 1, imax, idel6, igain6 afdbk butterlp aout, 1000 outs gasig+(aout+aout1)/2, gasig-(aout+aout1)/2 gasig = 0 endin </CsInstruments> <CsScore> f1 0 8192 10 1 ; Diskin ; Sta Dur Soundin i5 0 3 "beats.wav" ; Reverb ; St Dur Del1 Gn1 Del2 Gn2 Del3 Gn3 Del4 Gn4 Del5 Gn5 Del6 Gn6 i10 0 4 97 .11 23 .07 43 .09 72 .2 53 .2 119 .3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tablexseg.csd.xml0000644000000000000000000000400012262561504020623 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tablexseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; analyze "fox.wav" with PVANAL first iend = p4 ktime line 0, p3, iend tablexseg p5, p3, p6 ;morph from table 1 asig vpvoc ktime, 1, "fox.pvx" ;to table 2 outs asig*3, asig*3 endin </CsInstruments> <CsScore> f 1 0 512 9 .5 1 0 f 2 0 512 5 1 60 0.01 390 0.01 62 1 i 1 0 5 2.7 1 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ihold.csd.xml0000644000000000000000000000531412262561504017755 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ihold.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; A simple oscillator with its note held indefinitely. a1 oscil 10000, 440, 1 ihold ; If p4 equals 0, turn the note off. if (p4 == 0) kgoto offnow kgoto playit offnow: ; Turn the note off now. turnoff playit: ; Play the note. out a1 endin </CsInstruments> <CsScore> ; Table #1: an ordinary sine wave. f 1 0 32768 10 1 ; p4 = turn the note off (if it is equal to 0). ; Start playing Instrument #1. i 1 0 1 1 ; Turn Instrument #1 off after 3 seconds. i 1 3 1 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zaw.csd.xml0000644000000000000000000000542612262561504017463 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zaw.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple waveform. instr 1 ; Generate a simple sine waveform. asin oscil 20000, 440, 1 ; Send the sine waveform to za variable #1. zaw asin, 1 endin ; Instrument #2 -- generates audio output. instr 2 ; Read za variable #1. a1 zar 1 ; Generate the audio output. out a1 ; Clear the za variables, get them ready for ; another pass. zacl 0, 1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lpf18.csd.xml0000644000000000000000000000516112262561504017610 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o lpf18.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a sine waveform. ; Note that its amplitude (kamp) ranges from 0 to 1. kamp init 1 kcps init 440 knh init 3 ifn = 1 asine buzz kamp, kcps, knh, ifn ; Filter the sine waveform. ; Vary the cutoff frequency (kfco) from 300 to 3,000 Hz. kfco line 300, p3, 3000 kres init 0.8 kdist = p4 ivol = p5 aout lpf18 asine, kfco, kres, kdist out aout * ivol endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; different distortion and volumes to compensate i 1 0 4 0.2 30000 i 1 4.5 4 0.9 27000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midinoteonoct.csd.xml0000644000000000000000000000525312262561504021533 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midinoteonoct.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 midinoteonoct p4, p5 ;gets a MIDI note number value as octave-point-decimal value into p4, and MIDI velocity into p5 print p4 ;display the key value when it changes and when key is pressed kvel = p5/127 ;scale midi velocity to 0-1 ioct = p4 icps = cpsoct(ioct) ;convert octave-point-decimal value into Hz kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by asig pluck kenv*kvel, icps, icps, 2, 1 ;velocity value outs asig, asig endin </CsInstruments> <CsScore> f 0 30 ;runs for 30 seconds f 2 0 4096 10 1 i 1 0 2 8.000 100 ; play these notes from score as well i 1 + 2 8.917 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/resonz.csd.xml0000644000000000000000000001322712262561504020200 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o resonz.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Sean Costello */ ; Orchestra file for resonant filter sweep of a sawtooth-like waveform. ; The outputs of reson, resonr, and resonz are scaled by coefficients ; specified in the score, so that each filter can be heard on its own ; from the same instrument. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 idur = p3 ibegfreq = p4 ; beginning of sweep frequency iendfreq = p5 ; ending of sweep frequency ibw = p6 ; bandwidth of filters in Hz ifreq = p7 ; frequency of gbuzz that is to be filtered iamp = p8 ; amplitude to scale output by ires = p9 ; coefficient to scale amount of reson in output iresr = p10 ; coefficient to scale amount of resonr in output iresz = p11 ; coefficient to scale amount of resonz in output ; Frequency envelope for reson cutoff kfreq linseg ibegfreq, idur * .5, iendfreq, idur * .5, ibegfreq ; Amplitude envelope to prevent clicking kenv linseg 0, .1, iamp, idur - .2, iamp, .1, 0 ; Number of harmonics for gbuzz scaled to avoid aliasing iharms = (sr*.4)/ifreq asig gbuzz 1, ifreq, iharms, 1, .9, 1 ; "Sawtooth" waveform ain = kenv * asig ; output scaled by amp envelope ares reson ain, kfreq, ibw, 1 aresr resonr ain, kfreq, ibw, 1 aresz resonz ain, kfreq, ibw, 1 out ares * ires + aresr * iresr + aresz * iresz endin </CsInstruments> <CsScore> /* Written by Sean Costello */ f1 0 8192 9 1 1 .25 ; cosine table for gbuzz generator i1 0 10 1 3000 200 100 4000 1 0 0 ; reson output with bw = 200 i1 10 10 1 3000 200 100 4000 0 1 0 ; resonr output with bw = 200 i1 20 10 1 3000 200 100 4000 0 0 1 ; resonz output with bw = 200 i1 30 10 1 3000 50 200 8000 1 0 0 ; reson output with bw = 50 i1 40 10 1 3000 50 200 8000 0 1 0 ; resonr output with bw = 50 i1 50 10 1 3000 50 200 8000 0 0 1 ; resonz output with bw = 50 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/scoreline_i.csd.xml0000644000000000000000000000350512262561504021151 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac --old-parser ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o scoreline.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 scoreline_i {{ i 2 0 3 "flute.aiff" i 2 1 3 "beats.wav" }} endin instr 2 asig soundin p4 outs asig*.8, asig*.8 endin </CsInstruments> <CsScore> i1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/joystick-2.csd.xml0000644000000000000000000001231612262561504020654 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o joystick-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 ;0dbfs = 1 instr 1 kmask joystick 0, 1 kaxes init 0 kbuttons init 0 kx0 init 0 ; first two entries are # of axes and # of buttons, ky0 init 0 ; then axes, then buttons vtabk 0, 1, kaxes, kbuttons, kx0, ky0 kidx = 2+kaxes buttons: kcheck = kmask & 1<<kidx ; if the button was just now pressed and... kres tab kidx, 1 ; if button value is one, start a note schedkwhen kres*kcheck, 1, 20, 2, 0, 60000, kidx, kx0, ky0 kidx = kidx+1 if kidx < (kaxes+kbuttons+2) kgoto buttons endin instr 2 ; play a tone until the button is released kstop tab p4, 1 ; when this button is released, we fade out ihz init cpsoct(((p5+32767)/9362)+5) ; ~ 30 hz to 4khz print ihz ito init ampdb(((p6+32767)/2184)+60) ; ~ 60 - 90 db kenv init 0 kdelta init ito/(kr*10) if kstop == 1 kgoto output if kdelta < 0 kgoto output kdelta = kdelta*-1 output: kenv = kenv+kdelta kenv limit kenv, 0, ito aout oscils 1, ihz, 0 aout = kenv*aout outs aout, aout if kenv != 0 kgoto noexit if kdelta > 0 kgoto noexit turnoff noexit: endin </CsInstruments> <CsScore> f1 0 32 7 0 7 0 ; will hold the joystick data i1 0 60000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vosim.csd.xml0000644000000000000000000002001712262561504020010 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vosim.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 100 nchnls = 1 ;################################################# ; By Rasmus Ekman 2008 ; Square each point in table #p4. This should only be run once in the performance. instr 10 index tableng p4 index = index - 1 ; start from last point loop: ival table index, p4 ival = ival * ival tableiw ival, index, p4 index = index - 1 if index < 0 igoto endloop igoto loop endloop: endin ;################################################# ; Main vosim instrument. Sweeps from a fund1/form1 to fund2/form2, ; trying for narrowest formant bandwidth (still quite wide by the looks of it) ; p4: amp ; p5, p6: fund beg-end ; p7, p8: form beg-end ; p9: amp decay (ignored) ; p10: pulse count (ignored - calc internally) ; p11: pulse length mod ; p12: skip (for tied events) ; p13: don't fade out (if followed by tied note) instr 1 kamp init p4 ; freq start, end kfund line p5, p3, p6 ; formant start, end kform line p7, p3, p8 ; Try for constant ratio burst/silence, and narrowest formant bandwidth kPulseCount = (kform / kfund) ;init p10 ; Attempt to smooth steps between format bandwidths, ; increasing decay before we are forced to a lower pulse count kDecay = kPulseCount/(kform % kfund) ; init p9 if (kDecay * kPulseCount) > kamp then kDecay = kamp / kPulseCount endif kDecay = 0.3 * kDecay kPulseFactor init p11 ; ar vosim kamp, kFund, kForm, kDecay, kPulseCount, kPulseFactor, ifn [, iskip] ar1 vosim kamp, kfund, kform, kDecay, kPulseCount, kPulseFactor, 17, p12 ; scale amplitude for 16-bit files, with quick fade out amp init 20000 if (p13 != 0) goto nofade amp linseg 20000, p3-.02, 20000, .02, 0 nofade: out ar1 * amp endin </CsInstruments> <CsScore> f1 0 32768 9 1 1 0 ; sine wave f17 0 32768 9 0.5 1 0 ; half sine wave i10 0 0 17 ; init run only, square table 17 ; Vosim score ; Picking some formants from the table in Csound manual ; p4=amp fund form decay pulses pulsemod [skip] nofade ; tenor a -> e i1 0 .5 .5 280 240 650 400 .03 5 1 i1 . . .3 . . 1080 1700 .03 5 . i1 . . .2 . . 2650 2600 .03 5 . i1 . . .15 . . 2900 3200 .03 5 . ; tenor a -> o i1 0.6 .2 .5 300 210 650 400 .03 5 1 0 1 i1 . . .3 . . 1080 800 .03 5 . . . i1 . . .2 . . 2650 2600 .03 5 . . . i1 . . .15 . . 2900 2800 .03 5 . . . ; tenor o -> aah i1 .8 .3 .5 210 180 400 650 .03 5 1 1 1 i1 . . .3 . . 800 1080 .03 5 . . . i1 . . .2 . . 2600 2650 .03 5 . . . i1 . . .15 . . 2800 2900 .03 5 . . . ; tenor aa -> i i1 1.1 .2 .5 180 250 650 290 .03 5 1 1 1 i1 . . .3 . . 1080 1870 .03 5 . . . i1 . . .2 . . 2650 2800 .03 5 . . . i1 . . .15 . . 2900 3250 .03 5 . . . ; tenor i -> u i1 1.3 .3 .5 250 270 290 350 .03 5 1 1 0 i1 . . .3 . . 1870 600 .03 5 . . . i1 . . .2 . . 2800 2700 .03 5 . . . i1 . . .15 . . 3250 2900 .03 5 . . . e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dssictls.csd.xml0000644000000000000000000000723712262561504020514 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dssictls.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 gihandle dssiinit "caps.so", 1, 1 ; = equaliser and gaoutl init 0 ; verbose about all ports gaoutr init 0 instr 1 ; activate DSSI dssiactivate gihandle, 1 endin instr 2 ain1 diskin2 "beats.wav", 1,0,1 ; loop gaoutl = gaoutl+(ain1*.1) ; temper input gaoutr = gaoutr+(ain1*.1) endin instr 3 dssictls gihandle, 2, -48, 1 ; 31 Hz range -48 to 24 dssictls gihandle, 3, -48, 1 ; 63 Hz range -48 to 24 dssictls gihandle, 4, -48, 1 ; 125 Hz range -48 to 24 dssictls gihandle, 5, 20, 1 ; 250 Hz range -48 to 24 dssictls gihandle, 6, -48, 1 ; 500 Hz range -48 to 24 dssictls gihandle, 7, -48, 1 ; 1 kHz Hz range -48 to 24 dssictls gihandle, 8, -48, 1 ; 2 kHz range -48 to 24 dssictls gihandle, 9, 24, 1 ; 4 kHz range -48 to 24 dssictls gihandle, 10, 24, 1 ; 8 kHz range -48 to 24 dssictls gihandle, 11, 24, 1 ; 16 kHz range -48 to 24 endin instr 4 aout1, aout2 dssiaudio gihandle, gaoutl, gaoutr ;get beats.wav, mono out outs aout1,aout2 gaoutl = 0 gaoutr = 0 endin </CsInstruments> <CsScore> i 1 0 20 i 2 1 20 i 3 1 20 i 4 0 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lenarray.csd.xml0000644000000000000000000001030712262561504020471 0ustar rootroot <CsoundSynthesizer> <CsOptions> -n -m128 ;no sound output, reduced messages </CsOptions> <CsInstruments> ;example by joachim heintz sr = 44100 ksmps = 32 nchnls = 1 0dbfs = 1 instr 1 ;simple example kArr[] fillarray 1, 2, 3, 4 ;fill array manually printks "Length of kArr = %d\n\n", 0, lenarray(kArr) ;print out its length turnoff ;only do this in the first k-cycle endin instr 2 ;random array length iNumEls random 1, 11 ;create random number between 1 and 10 kArr[] init int(iNumEls) ;create array of this length printks "Random length of kArr = %d\n", 0, lenarray(kArr) ;print out turnoff endin instr 3 ;fill random array length with random elements iNumEls random 1, 11 ;create random number between 1 and 10 kArr[] init int(iNumEls) ;create array of this length printks "Random length of kArr = %d\n", 0, lenarray(kArr) ;print out ;fill kIndx = 0 ;initialize index until kIndx == lenarray(kArr) do kArr[kIndx] rnd31 10, 0 ;set element to random value -10...10 kIndx += 1 ;increase index od ;print kIndx = 0 ;initialize index until kIndx == lenarray(kArr) do printf("kArr[%d] = %f\n", kIndx+1, kIndx, kArr[kIndx]) kIndx += 1 ;increase index od turnoff endin </CsInstruments> <CsScore> i 1 0 .1 i 2 0 .1 i 2 0 .1 i 3 0 .1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/linen.csd.xml0000644000000000000000000000437412262561504017770 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o linen.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; p4=amp ; p5=freq ; p6=attack time ; p7=release time ares linen p4, p6, p3, p7 asig poscil ares, p5, 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 4096 10 1 ; sine wave ;ins strt dur amp freq attack release i1 0 1 .5 440 0.5 0.7 i1 1.5 1 .2 440 0.9 0.1 i1 3 1 .2 880 0.02 0.99 i1 4.5 1 .2 880 0.7 0.01 i1 6 3 .7 220 0.5 0.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/poscil.csd.xml0000644000000000000000000000416212262561504020147 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o poscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 gisine ftgen 0, 0, 2^10, 10, 1 instr 1 ipeak random 0, 1 ;where is the envelope peak asig poscil .8, 220, gisine aenv transeg 0, p3*ipeak, 6, 1, p3-p3*ipeak, -6, 0 aL,aR pan2 asig*aenv, ipeak ;pan according to random value outs aL, aR endin </CsInstruments> <CsScore> i1 0 5 i1 4 5 i1 8 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/expseg.csd.xml0000644000000000000000000000512112262561504020145 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o expseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; p4 = frequency in pitch-class notation. kcps = cpspch(p4) ; Create an amplitude envelope. kenv expseg 0.01, p3*0.25, 1, p3*0.75, 0.01 kamp = kenv * 30000 a1 oscil kamp, kcps, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for a half-second, p4=8.00 i 1 0 0.5 8.00 ; Play Instrument #1 for a half-second, p4=8.01 i 1 1 0.5 8.01 ; Play Instrument #1 for a half-second, p4=8.02 i 1 2 0.5 8.02 ; Play Instrument #1 for a half-second, p4=8.03 i 1 3 0.5 8.03 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/raises.csd.xml0000644000000000000000000000443412262561504020146 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o ^.wav ; output to audio file </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Lo-Fi sound kpow = 10 ;exponent kbase line 1, p3, 2.2 ;vary the base kQuantize = kbase^kpow kQuantize = kQuantize*0.5 ;half the number of steps for each side of a bipolar signal printk2 kQuantize asig diskin2 "fox.wav", 1, 0, 1 ;loop the fox asig = round(asig * kQuantize) / kQuantize ;quantize and scale audio signal outs asig, asig endin </CsInstruments> <CsScore> i1 0 19.2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/loopseg.csd.xml0000644000000000000000000000743412262561504020333 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loopseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq init p4 ; frequency of loop repetition ifrac = p5 ; frequency ratio of restart triggers ktrig metro kfreq * ifrac ; triggers to restart loop iphase = 0 ; initial phase ; loop of note values (some glissandi) ; val dur val dur etc... knote loopseg kfreq, ktrig, iphase, 40, 1, 40, 0, 43,1,43,0, 49,2,48,0, \ 47,1,47,0, 46,1,46,0, 46,1,47,0, 49,1,49,0, 43,1,43,0, 46,1,46,0, 40,1,39,0 ; loop of filter cutoff values (oct format). This loop, half speed of note loop. kcfoct loopseg kfreq*0.5, ktrig, iphase, 11,2,4,0, 12,1,4,0, 13,1,4,0, \ 11.5,3,4,0, 12.5,1,4,0, 13,2,4,0, 12.5,1,4,0 kenv linseg 0,0.01,1,p3-5.01,1,5,0 ioct = int((rnd(0.999)*4)-2) ; random value either -1, 0 or 1 asig vco2 0.2*kenv,cpsmidinn(knote)*octave(ioct),0 ; sawtooth asig moogladder asig,cpsoct(kcfoct),rnd(0.6) ; filter sawtooth aL,aR pan2 asig,rnd(1) ; random static pan location outs aL, aR endin </CsInstruments> <CsScore> ; 4 layers, each with a different frequency of loop repetition (p4), ; frequency ratio of restart triggers (p5) and pan position (p6). i 1 0 30 0.5 [11/19] i 1 6 30 0.25 [11/13] i 1 12 30 0.125 [11/16] i 1 18 30 1 [11/12] e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/compileorc.csd.xml0000644000000000000000000000253112262561504021010 0ustar rootroot <CsoundSynthesizer> <CsOptions> -o dac -d </CsOptions> <CsInstruments> sr = 44100 nchnls = 1 ksmps = 32 0dbfs = 1 instr 1 ires compileorc "does_not_exist.orc" print ires ; -1 as could not compile ires compileorc "my.orc" print ires ; 0 as compiled successfully event_i "i", 2, 0, 3, .2, 465 ;send event endin </CsInstruments> <CsScore> i1 0 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/logicOR.csd.xml0000644000000000000000000000440012262561504020207 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and virtual midi keyboard ;-iadc ;;;uncomment -iadc if realtime audio input is needed too </CsOptions> <CsInstruments> ;after a UDO from Rory Walsh sr = 44100 ksmps = 32 nchnls = 2 instr 1 ;displays notes, midi channel and control number information kstatus, kchan, kdata1, kdata2 midiin k1 changed kstatus k2 changed kchan k3 changed kdata1 k4 changed kdata2 if((k1==1)||(k2==1)||(k3==1)||(k4==1)) then printks "Value:%d ChanNo:%d CtrlNo:%d\n" , 0, kdata2, kchan, kdata1 endif endin </CsInstruments> <CsScore> i1 0 60 ;print values for 60 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/linsegb.csd.xml0000644000000000000000000000475612262561504020312 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o linseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kcps = cpspch(p4) kenv linsegb 0, 0.25, 1, 1, 0 asig poscil kenv, kcps, giSine outs asig, asig endin instr 2 ; scaling to duration kcps = cpspch(p4) kenv linseg 0, p3*0.25, 1, p3, 0 asig poscil kenv, kcps, giSine outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 7.00 ; = 1 sec, p3 fits exactly i 1 2 2 7.00 ; = 2 sec, p3 truncated at 1 sec i 2 4 1 7.00 ; scales to duration i 2 6 2 7.00 ; of p3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sfplay3.csd.xml0000644000000000000000000001163112262561504020236 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfplay3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gitwelve ftgen 1, 0, 16, -2, 12, 2, 440, 69, 1, 16/15, 9/8, 6/5, 5/4, 4/3, 7/5, 3/2, 8/5, 5/3, 9/5, 15/8, 2 givife ftgen 2, 0, 16, -2, 5, 2, 261.659, 60, 1, 1.1486, 1.3195, 1.5157, 1.7411, 2.00 giSF sfload "01hpschd.sf2" sfplist giSF gipre sfpreset 0, 0, giSF, 0 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey = p4 ivel = p5 aenv linsegr 1, 1, 1, 1, 0 ;envelope icps cpstuni ikey, gitwelve ;12 tones per octave iamp = 0.0004 ;scale amplitude iamp = iamp * ivel * 1/128 ;make velocity-dependent aL, aR sfplay3 ivel, ikey, iamp, icps, gipre, 1 aL = aL * aenv aR = aR * aenv outs aL, aR endin instr 2 mididefault 60, p3 midinoteonkey p4, p5 ikey = p4 ivel = p5 aenv linsegr 1, 1, 1, 1, 0 ;envelope icps cpstuni ikey, givife ;5 tones per octave iamp = 0.0004 ;scale amplitude iamp = iamp * ivel * 1/128 ;make velocity-dependent aL, aR sfplay3 ivel, ikey, iamp, icps, gipre, 1 aL = aL * aenv aR = aR * aenv outs aL, aR endin </CsInstruments> <CsScore> f0 60 ;play for 60 seconds ;instr.1 using ftable 1 i1 0 1 60 100 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 40 ;instr.2 using ftable 2 i2 5 1 60 100 i2 + 1 62 < i2 + 1 65 < i2 + 1 69 40 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/nlfilt.csd.xml0000644000000000000000000000525312262561504020150 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o nlfilt.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;unfiltered noise asig rand .7 outs asig, asig endin instr 2 ;filtered noise ka = p4 kb = p5 kd = p6 kC = p7 kL = p8 asig rand .3 afilt nlfilt asig, ka, kb, kd, kC, kL asig clip afilt, 2, .9 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 ; unfiltersd ; a b d C L i 2 2 2 0 0 0.8 0.5 20 ; non-linear effect i 2 + 2 .4 0.2 0.7 0.11 200 ; low=paas with non-linear i 2 + 2 0.35 -0.3 0.95 0.1 200 ; high-pass with non-linear i 2 + 2 0.7 -0.2 0.9 0.2 20 ; high-pass with non-linear e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/grain2.csd.xml0000644000000000000000000001772312262561504020047 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o grain2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 48000 kr = 750 ksmps = 64 nchnls = 2 /* square wave */ i_ ftgen 1, 0, 4096, 7, 1, 2048, 1, 0, -1, 2048, -1 /* window */ i_ ftgen 2, 0, 16384, 7, 0, 4096, 1, 4096, 0.3333, 8192, 0 /* sine wave */ i_ ftgen 3, 0, 1024, 10, 1 /* room parameters */ i_ ftgen 7, 0, 64, -2, 4, 50, -1, -1, -1, 11, \ 1, 26.833, 0.05, 0.85, 10000, 0.8, 0.5, 2, \ 1, 1.753, 0.05, 0.85, 5000, 0.8, 0.5, 2, \ 1, 39.451, 0.05, 0.85, 7000, 0.8, 0.5, 2, \ 1, 33.503, 0.05, 0.85, 7000, 0.8, 0.5, 2, \ 1, 36.151, 0.05, 0.85, 7000, 0.8, 0.5, 2, \ 1, 29.633, 0.05, 0.85, 7000, 0.8, 0.5, 2 ga01 init 0 /* generate bandlimited square waves */ i0 = 0 loop1: imaxh = sr / (2 * 440.0 * exp(log(2.0) * (i0 - 69) / 12)) i_ ftgen i0 + 256, 0, 4096, -30, 1, 1, imaxh i0 = i0 + 1 if (i0 < 127.5) igoto loop1 instr 1 p3 = p3 + 0.2 /* note velocity */ iamp = 0.0039 + p5 * p5 / 16192 /* vibrato */ kcps oscili 1, 8, 3 kenv linseg 0, 0.05, 0, 0.1, 1, 1, 1 /* frequency */ kcps = (kcps * kenv * 0.01 + 1) * 440 * exp(log(2) * (p4 - 69) / 12) /* grain ftable */ kfn = int(256 + 69 + 0.5 + 12 * log(kcps / 440) / log(2)) /* grain duration */ kgdur port 100, 0.1, 20 kgdur = kgdur / kcps a1 grain2 kcps, kcps * 0.02, kgdur, 50, kfn, 2, -0.5, 22, 2 a1 butterlp a1, 3000 a2 grain2 kcps, kcps * 0.02, 4 / kcps, 50, kfn, 2, -0.5, 23, 2 a2 butterbp a2, 12000, 8000 a2 butterbp a2, 12000, 8000 aenv1 linseg 0, 0.01, 1, 1, 1 aenv2 linseg 3, 0.05, 1, 1, 1 aenv3 linseg 1, p3 - 0.2, 1, 0.07, 0, 1, 0 a1 = aenv1 * aenv3 * (a1 + a2 * 0.7 * aenv2) ga01 = ga01 + a1 * 10000 * iamp endin /* output instr */ instr 81 i1 = 0.000001 aLl, aLh, aRl, aRh spat3di ga01 + i1*i1*i1*i1, 3.0, 4.0, 0.0, 0.5, 7, 4 ga01 = 0 aLl butterlp aLl, 800.0 aRl butterlp aRl, 800.0 outs aLl + aLh, aRl + aRh endin </CsInstruments> <CsScore> t 0 60 i 1 0.0 1.3 60 127 i 1 2.0 1.3 67 127 i 1 4.0 1.3 64 112 i 1 4.0 1.3 72 112 i 81 0 6.4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cigoto.csd.xml0000644000000000000000000000613212262561504020141 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Get the value of the 4th p-field from the score. iparam = p4 ; If iparam is 1 then play the high note. ; If not then play the low note. cigoto (iparam ==1), highnote igoto lownote highnote: ifreq = 880 goto playit lownote: ifreq = 440 goto playit playit: ; Print the values of iparam and ifreq. print iparam print ifreq a1 oscil 10000, ifreq, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; p4: 1 = high note, anything else = low note ; Play Instrument #1 for one second, a low note. i 1 0 1 0 ; Play a Instrument #1 for one second, a high note. i 1 1 1 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/max_k.csd.xml0000644000000000000000000000517312262561504017760 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -Ma ;;;realtime audio out and midi in (on all inputs) ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o max_k.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 FLpanel "This Panel contains VU-meter",300,100 gk1,gih1 FLslider "VU-meter", 0,1,0,1, -1, 250,30, 30,30 FLsetColor2 50, 50, 255, gih1 FLpanel_end FLrun ga1 init 0 instr 1 kenv linsegr 0,.5,.7,.5,.5,.2,0 ifreq cpsmidi a1 poscil 0dbfs*kenv, ifreq, 1 ga1 = ga1+a1 endin instr 2 outs ga1, ga1 ktrig metro 25 ;refresh 25 times per second kval max_k ga1, ktrig, 1 FLsetVal ktrig, kval, gih1 ga1 = 0 endin </CsInstruments> <CsScore> f1 0 1024 10 1 i2 0 3600 f0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midipitchbend.csd.xml0000644000000000000000000000556312262561504021467 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midipitchbend.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 midinoteonoct p4, p5 ;gets a MIDI note number value as octave-point-decimal value into p4, and MIDI velocity into p5 kvel = p5/127 ;scale midi velocity to 0-1 kpb init 0 midipitchbend kpb printk2 kpb ;display the pitch-bend value when it changes koct = p4+kpb ;add pitchbend values to octave-point-decimal value kcps = cpsoct(koct) ;convert octave-point-decimal value into Hz kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by asig vco2 kenv*kvel, kcps ;velocity value outs asig, asig endin </CsInstruments> <CsScore> f 0 30 ;runs for 30 seconds f 2 0 4096 10 1 i 1 0 2 8.000 100 ; play these notes from score as well i 1 + 2 8.917 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vmult-2.csd.xml0000644000000000000000000000507212262561504020165 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vmult ifn1, ival, ielements, idstoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSread.csd.xml0000644000000000000000000000337412262561504020145 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ATSread.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "beats.ats" is created by atsa ktime line 0, p3, 2 kfreq, kamp ATSread ktime, "beats.ats", 100 aout oscili 0.8, kfreq, 1 outs aout, aout endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/powoftwo.csd.xml0000644000000000000000000000510312262561504020536 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o poweroftwo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; bit reduction for Lo-Fi sound iBits = p5 ;bit depth iQuantize = powoftwo(iBits) ;find number of discrete steps for this bit depth iQuantize = iQuantize*0.5 ;half the number of steps for each side of a bipolar signal print iQuantize asig soundin "fox.wav" asig = round(asig * iQuantize) / iQuantize ;quantize audio signal (bit reduce) outs asig, asig endin </CsInstruments> <CsScore> ; bits i1 0 3 16 i1 ^+3 . 12 i1 ^+3 . 8 i1 ^+3 . 4 i1 ^+3 . 2 i1 ^+3 . 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/seqtime.csd.xml0000644000000000000000000000655112262561504020331 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o seqtime.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 64 nchnls = 1 ; By Tim Mortimer and Andres Cabrera 2007 0dbfs = 1 gisine ftgen 0, 0, 8192, 10, 1 ;;; table defining an integer pitch set gipset ftgen 0, 0, 4, -2, 8.00, 8.04, 8.07, 8.10 ;;;DELTA times for seqtime gidelta ftgen 0, 0, 4, -2, .5, 1, .25, 1.25 instr 1 kndx init 0 ktrigger init 0 ktime_unit init 1 kstart init p4 kloop init p5 kinitndx init 0 kfn_times init gidelta ktrigger seqtime ktime_unit, kstart, kloop, kinitndx, kfn_times printk2 ktrigger if (ktrigger > 0) then kpitch table kndx, gipset event "i", 2, 0, 1, kpitch kndx = kndx + 1 kndx = kndx % kloop endif endin instr 2 icps = cpspch (p4) a1 buzz 1, icps, 7, gisine aamp expseg 0.00003,.02,1,p3-.02,0.00003 a1 = a1 * aamp * 0.5 out a1 endin </CsInstruments> <CsScore> ; start dur kstart kloop i 1 0 7 0 4 i 1 8 10 0 3 i 1 19 10 4 4 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSinfo-2.csd.xml0000644000000000000000000000512012262561504020313 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n ;;;no audio out </CsOptions> <CsInstruments> ;example by joachim heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; "fox.ats" has been created by ATSanal Sfile = "fox.ats" isr ATSinfo Sfile, 0 ifs ATSinfo Sfile, 1 iws ATSinfo Sfile, 2 inp ATSinfo Sfile, 3 inf ATSinfo Sfile, 4 ima ATSinfo Sfile, 5 imf ATSinfo Sfile, 6 id ATSinfo Sfile, 7 ift ATSinfo Sfile, 8 prints {{ Sample rate = %d Hz Frame Size = %d samples Window Size = %d samples Number of Partials = %d Number of Frames = %d Maximum Amplitude = %f Maximum Frequency = %f Hz Duration = %f seconds ATS file Type = %d }}, isr, ifs, iws, inp, inf, ima, imf, id, ift endin </CsInstruments> <CsScore> i 1 0 0 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vco2ift.csd.xml0000644000000000000000000000432012262561504020226 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vco2ift.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; user defined waveform -2: fixed table size (64), number of partials ; multiplier is 1.4 itmp ftgen 2, 0, 64, 5, 1, 2, 120, 60, 1, 1, 0.001, 1 ift vco2init -2, 2, 1.4, 4096, 4096, 2 instr 1 icps = p4 ifn vco2ift icps, -2, 0.5 ;with user defined waveform print ifn asig oscili 1, 220, ifn ; (-2), and sr/2 bandwidth outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 20 i 1 3 2 2000 i 1 6 2 20000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/denorm.csd.xml0000644000000000000000000000470612262561504020146 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o denorm.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; slightly simplified example from Istvan Varga 2006 sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 garvb init 0 instr 1 a1 oscils 0.6, 440, 0 a2 linsegr 0, 0.005, 1, 3600, 1, 0.08, 0 a1 = a1 * a2 vincr garvb, a1 outs a1, a1 endin instr 99 ;"Always on" denorm garvb aL, aR reverbsc garvb * 0.5, garvb * 0.5, 0.92, 10000 clear garvb outs aL, aR endin </CsInstruments> <CsScore> i 99 0 -1 ;held by a negative p3, means "always on" i 1 0 0.5 i 1 4 0.5 e 8 ;8 extra seconds after the performance </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pgmassign.csd.xml0000644000000000000000000000344212262561504020646 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o pgmassign.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Program 55 (synth vox) uses Instrument #10. pgmassign 55, 10 ; Instrument #10. instr 10 ; Just an example, no working code in here! endin </CsInstruments> <CsScore> ; Play Instrument #10 for one second. i 10 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gauss.csd.xml0000644000000000000000000000617312262561504020004 0ustar rootroot <CsoundSynthesizer> <CsOptions> -d -o dac </CsOptions> <CsInstruments> instr 1 irange = p4 imu = p5 isamples = p6 indx = 0 icount = 1 ix = 0.0 ix2 = 0.0 loop: i1 gauss irange i1 = i1 + imu ix = ix + i1 ix2 = ix2 + i1*i1 if i1 >= -(irange+imu) && i1 <= (irange+imu) then icount = icount+1 endif loop_lt indx, 1, isamples, loop imean = ix / isamples ;mean value istd = sqrt(ix2/isamples - imean*imean) ;standard deviation prints "mean = %3.3f, std = %3.3f, ", imean, istd prints "samples inside the given range: %3.3f\%\n", icount*100.0/isamples endin </CsInstruments> <CsScore> i 1 0 0.1 1.0 0 100000 ; range = 1, mu = 0.0, sigma = 1/3.83 = 0.261 i 1 0.1 0.1 3.83 0 100000 ; range = 3.83, mu = 0.0, sigma = 1 i 1 0.2 0.1 5.745 2.7 100000 ; range = 5.745, mu = 2.7, sigma = 5.745/3.83 = 1.5 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cosinv.csd.xml0000644000000000000000000000327112262561504020157 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cosinv.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 0.5 i1 = cosinv(irad) print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pset.csd.xml0000644000000000000000000000345312262561504017633 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pset.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ;this shows an example with non-midi use pset 1, 0, 1, 220, 0.5 asig poscil p5, p4, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 1024 10 1 ;sine wave i 1 0 1 i 1 1 1 440 i 1 2 1 440 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKBandedWG.csd.xml0000644000000000000000000000401212262561504020645 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKBandedWG.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;pressure of bow kenv line 1, p3, 0 asig STKBandedWG cpspch(ifrq), 1, 2, kv1, 4, 100, 11, 0, 1, 0, 64, 100, 128, 120, 16, 2 asig = asig * kenv ;simple envelope outs asig, asig endin </CsInstruments> <CsScore> i 1 0 10 5.00 100 0 i 1 10 8 6.03 10 . i 1 20 5 7.05 50 127 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLpanel.csd.xml0000644000000000000000000000436112262561504020200 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLpanel.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Creates an empty window panel sr = 44100 kr = 441 ksmps = 100 nchnls = 1 ; Panel height in pixels ipanelheight = 900 ; Panel width in pixels ipanelwidth = 400 ; Horizontal position of the panel on screen in pixels ix = 50 ; Vertical position of the panel on screen in pixels iy = 50 FLpanel "A Window Panel", ipanelheight, ipanelwidth, ix, iy ; End of panel contents FLpanelEnd ;Run the widget thread! FLrun instr 1 endin </CsInstruments> <CsScore> ; 'Dummy' score event of 1 hour. f 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ctrl7.csd.xml0000644000000000000000000000367412262561504017720 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M0 ;;;RT audio I/O with MIDI in ;-iadc ;;;uncomment -iadc if RT audio input is needed too </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; expects MIDI controller input on channel 1 ; run and move your midi controller to see result imax = 1 imin = 0 ichan = 1 ictlno = 7 initc7 1, 7, 1 ; start at max. volume kamp ctrl7 ichan, ictlno, imin, imax ; controller 7 printk2 kamp asig oscil kamp, 220, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 4096 10 1 i1 0 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outch-2.csd.xml0000644000000000000000000000453012262561504020136 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outch-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 seed 0 instr 1 ;random movements between 4 speakers with outch ichn1 random 1, 4.999 ;channel to start ichn2 random 1, 4.999 ;channel to end prints "Moving from speaker %d to speaker %d%n", int(ichn1), int(ichn2) asamp soundin "fox.wav" kmov linseg 0, p3, 1 a1, a2 pan2 asamp, kmov outch int(ichn1), a1, int(ichn2), a2 endin </CsInstruments> <CsScore> r 5 i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cabasa.csd.xml0000644000000000000000000000316012262561504020065 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cabasa.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 inum = p4 idamp = p5 asig cabasa 0.9, 0.01, inum, idamp outs asig, asig endin </CsInstruments> <CsScore> i1 1 1 48 .95 i1 + 1 1000 .5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluid-2.csd.xml0000644000000000000000000001332112262561504020115 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out No messages Look for midifile in folder manual/examples -odac -d -F midichn_advanced.mid ;;;reatime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fluid-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; LOAD SOUNDFONTS gienginenum1 fluidEngine gienginenum2 fluidEngine isfnum1 fluidLoad "Piano Steinway Grand Model C (21,738KB).sf2", gienginenum1, 1 ; Bright Steinway, program 1, channel 1 fluidProgramSelect gienginenum1, 1, isfnum1, 0, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidProgramSelect gienginenum1, 3, isfnum1, 0, 2 isfnum2 fluidLoad "63.3mg The Sound Site Album Bank V1.0.SF2", gienginenum2, 1 ; General MIDI, program 50, channel 2 fluidProgramSelect gienginenum2, 2, isfnum2, 0, 50 ; SEND NOTES TO STEINWAY SOUNDFONT instr 1 ; FluidSynth Steinway Rev ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 1 ikey = p4 ivelocity = p5 fluidNote gienginenum1, ichannel, ikey, ivelocity endin instr 2 ; GM soundfont ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 2 ikey = p4 ivelocity = p5 fluidNote gienginenum2, ichannel, ikey, ivelocity endin instr 3 ; FluidSynth Steinway Rev ; INITIALIZATION mididefault 60, p3 ; Default duration of 60 -- overridden by score. midinoteonkey p4, p5 ; Channels MIDI input to pfields. ; Use channel assigned in fluidload. ichannel = 3 ikey = p4 ivelocity = p5 fluidNote gienginenum1, ichannel, ikey, ivelocity endin ; COLLECT AUDIO FROM ALL SOUNDFONTS instr 100 ; Fluidsynth output iamplitude1 = 7 iamplitude2 = 7 ; AUDIO aleft1, aright1 fluidOut gienginenum1 aleft2, aright2 fluidOut gienginenum2 outs (aleft1 * iamplitude1) + (aleft2 * iamplitude2), \ (aright1 * iamplitude1) + (aright2 * iamplitude2) endin </CsInstruments> <CsScore> i 1 0 3 60 100 i 2 1 3 60 100 i 3 3 3 63 100 i 100 0 10 ;run for 10seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wgpluck2.csd.xml0000644000000000000000000000361212262561504020413 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o wgpluck2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 iplk = 0.75 kamp = 30000 icps = 220 kpick = 0.75 krefl = 0.5 apluck wgpluck2 iplk, kamp, icps, kpick, krefl out apluck endin </CsInstruments> <CsScore> ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/biquad-2.csd.xml0000644000000000000000000002351512262561504020265 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o biquad-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 /* modal synthesis using biquad filters as oscillators Example by Scott Lindroth 2007 */ instr 1 ipi = 3.1415926 idenom = sr*0.5 ipulseSpd = p4 icps = p5 ipan = p6 iamp = p7 iModes = p8 apulse mpulse iamp, 0 icps = cpspch( icps ) ; filter gain iamp1 = 600 iamp2 = 1000 iamp3 = 1000 iamp4 = 1000 iamp5 = 1000 iamp6 = 1000 ; resonance irpole1 = 0.99999 irpole2 = irpole1 irpole3 = irpole1 irpole4 = irpole1 irpole5 = irpole1 irpole6 = irpole1 ; modal frequencies if (iModes == 1) goto modes1 if (iModes == 2) goto modes2 modes1: if1 = icps * 1 ;pot lid if2 = icps * 6.27 if3 = icps * 3.2 if4 = icps * 9.92 if5 = icps * 14.15 if6 = icps * 6.23 goto nextPart modes2: if1 = icps * 1 ;uniform wood bar if2 = icps * 2.572 if3 = icps * 4.644 if4 = icps * 6.984 if5 = icps * 9.723 if6 = icps * 12.0 goto nextPart nextPart: ; convert frequency to radian frequency itheta1 = (if1/idenom) * ipi itheta2 = (if2/idenom) * ipi itheta3 = (if3/idenom) * ipi itheta4 = (if4/idenom) * ipi itheta5 = (if5/idenom) * ipi itheta6 = (if6/idenom) * ipi ; calculate coefficients ib11 = -2 * irpole1 * cos(itheta1) ib21 = irpole1 * irpole1 ib12 = -2 * irpole2 * cos(itheta2) ib22 = irpole2 * irpole2 ib13 = -2 * irpole3 * cos(itheta3) ib23 = irpole3 * irpole3 ib14 = -2 * irpole4 * cos(itheta4) ib24 = irpole4 * irpole4 ib15 = -2 * irpole5 * cos(itheta5) ib25 = irpole5 * irpole5 ib16 = -2 * irpole6 * cos(itheta6) ib26 = irpole6 * irpole6 ;printk 1, ib 11 ;printk 1, ib 21 ; also try setting the -1 coeff. to 0, but be sure to scale down the amplitude! asin1 biquad apulse * iamp1, 1, 0, -1, 1, ib11, ib21 asin2 biquad apulse * iamp2, 1, 0, -1, 1, ib12, ib22 asin3 biquad apulse * iamp3, 1, 0, -1, 1, ib13, ib23 asin4 biquad apulse * iamp4, 1, 0, -1, 1, ib14, ib24 asin5 biquad apulse * iamp5, 1, 0, -1, 1, ib15, ib25 asin6 biquad apulse * iamp6, 1, 0, -1, 1, ib16, ib26 afin = (asin1 + asin2 + asin3 + asin4 + asin5 + asin6) outs afin * sqrt(p6), afin*sqrt(1-p6) endin </CsInstruments> <CsScore> ;ins st dur pulseSpd pch pan amp Modes i1 0 12 0 7.089 0 0.7 2 i1 . . . 7.09 1 . . i1 . . . 7.091 0.5 . . i1 0 12 0 8.039 0 0.7 2 i1 0 12 0 8.04 1 0.7 2 i1 0 12 0 8.041 0.5 0.7 2 i1 9 . . 7.089 0 . 2 i1 . . . 7.09 1 . . i1 . . . 7.091 0.5 . . i1 9 12 0 8.019 0 0.7 2 i1 9 12 0 8.02 1 0.7 2 i1 9 12 0 8.021 0.5 0.7 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/birnd.csd.xml0000644000000000000000000000315412262561504017754 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 ; Generate a random number from -1 to 1. kbin = birnd(1) printk .2, kbin endin </CsInstruments> <CsScore> i 1 0 1 i 1 + . i 1 + . i 1 + . i 1 + . i 1 + . i 1 + . e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wrap.csd.xml0000644000000000000000000000656612262561504017641 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o wrap.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; Limit / Mirror / Wrap igain = p4 ;gain ilevl1 = p5 ; + level ilevl2 = p6 ; - level imode = p7 ;1 = limit, 2 = mirror, 3 = wrap ain soundin "fox.wav" ain = ain*igain if imode = 1 goto limit if imode = 2 goto mirror asig wrap ain, ilevl2, ilevl1 goto outsignal limit: asig limit ain, ilevl2, ilevl1 goto outsignal mirror: asig mirror ain, ilevl2, ilevl1 outsignal: outs asig*.5, asig*.5 ;mind your speakers endin </CsInstruments> <CsScore> ; Gain +Levl -Levl Mode i1 0 3 4.00 .25 -1.00 1 ;limit i1 4 3 4.00 .25 -1.00 2 ;mirror i1 8 3 4.00 .25 -1.00 3 ;wrap e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsanal.csd.xml0000644000000000000000000000602212262561504020317 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvsanal.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;pvsanal has no influence when there is no transformation of original sound ifftsize = p4 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1 ;von-Hann window Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape ;fft-analysis of the audio-signal fftblur pvscale fftin, p5 ;scale aout pvsynth fftblur ;resynthesis outs aout, aout endin </CsInstruments> <CsScore> s i 1 0 3 512 1 ;original sound - ifftsize of pvsanal does not have any influence i 1 3 3 1024 1 ;even with different i 1 6 3 2048 1 ;settings s i 1 0 3 512 1.5 ;but transformation - here a fifth higher i 1 3 3 1024 1.5 ;but with different settings i 1 6 3 2048 1.5 ;for ifftsize of pvsanal e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsbandr.csd.xml0000644000000000000000000000412312262561504020472 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsbandr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 Sfile = "fox.wav" klowcut = 100 klowfull = 200 khighfull = 1900 khighcut = 2000 ain soundin Sfile fftin pvsanal ain, 1024, 256, 1024, 1; fft-analysis of the audio-signal fftbp pvsbandr fftin, klowcut, klowfull, khighfull, khighcut ; band reject abp pvsynth fftbp; resynthesis out abp endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen23.csd.xml0000644000000000000000000000603012262561504017570 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen23.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;"spectrum.txt" is created by the spectrum plotter of Audacity (set at size 128), using "fox.wav". instr 1 ;performs additive synthesis based on spectrum.txt indx =0 ;start reading at first value loop: ifreq tab_i indx, 2 ;take odd values of list (= frequency) iamp tab_i indx+1, 2 ;take even values of list (= amplitude) event_i "i", 10, 0, p3, iamp, ifreq ;use "event_i" to trigger instr. 10 loop_lt indx, 2, 128, loop ;use all 128 frequency and amplitude values endin instr 10 ;generate sound iamp = p4 ifreq = p5 asig poscil ampdb(iamp), ifreq, 1 asig linen asig, .01, p3, p2 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave f 2 0 128 -23 "spectrum.txt" ;"spectrum.txt" can be found in /manual/examples i1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gaussi.csd.xml0000644000000000000000000000345012262561504020150 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o exprand.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 klambda gaussi 100, 1, 3 printk2 klambda ; look aout oscili 0.8, 440+klambda, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/logbtwo.csd.xml0000644000000000000000000000321312262561504020327 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o logbtwo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = logbtwo(3) print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/toot9.csd.xml0000644000000000000000000000625412262561504017740 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> nchnls = 2 ; stereo output garvbsig init 0 ; global "a" variable initialized to 0 instr 9 idur = p3 iamp = p4 iskiptime = p5 iattack = p6 irelease = p7 ibalance = p8 ; 1 = left, .5 = center, 0 = right irvbgain = p9 kamp linen iamp, iattack, idur, irelease asig soundin "fox.wav", iskiptime arampsig = kamp * asig outs arampsig * ibalance, arampsig * (1 - ibalance) garvbsig = garvbsig + arampsig * irvbgain endin instr 99 irvbtime = p4 asig reverb garvbsig, irvbtime ; put global sig into reverb outs asig, asig garvbsig = 0 ; then clear it endin </CsInstruments> <CsScore> ;ins strt dur rvbtime i99 0 15 2.2 ;ins strt dur amp skip atk rel blnce(0-1) rvbsend i9 0 1.2 .5 0 .02 .1 1 .2 i9 2 1.4 .5 0 .03 .1 0 .3 i9 3.5 2.28 .5 0 .9 .1 .5 .1 i9 4.5 2.28 .5 0 1.2 .1 0 .2 i9 5 2.28 .5 0 .2 .1 1 .3 i9 9 2.28 .7 0 .1 .1 .5 .03 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/valpass-2.csd.xml0000644000000000000000000000452512262561504020471 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o valpass-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 65536, 10, 1 ;sine wave instr 1 asig diskin2 "beats.wav", 1, 0, 1 krvt line 0.01, p3, p3 ;reverb time adepth = p4 ;sine depth krate = 0.3 ;sine rate (speed) adel oscil 0.5, krate, giSine ;delay time oscillator (LFO) adel = ((adel+0.5)*adepth) ;scale and offset LFO aout valpass asig, krvt, adel*0.01, 0.5 outs aout, aout endin </CsInstruments> <CsScore> i1 0 10 1 i1 11 10 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/oscbnk.csd.xml0000644000000000000000000002620212262561504020134 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o oscbnk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Istvan Varga */ sr = 48000 kr = 750 ksmps = 64 nchnls = 2 ga01 init 0 ga02 init 0 /* sawtooth wave */ i_ ftgen 1, 0, 16384, 7, 1, 16384, -1 /* FM waveform */ i_ ftgen 3, 0, 4096, 7, 0, 512, 0.25, 512, 1, 512, 0.25, 512, \ 0, 512, -0.25, 512, -1, 512, -0.25, 512, 0 /* AM waveform */ i_ ftgen 4, 0, 4096, 5, 1, 4096, 0.01 /* FM to EQ */ i_ ftgen 5, 0, 1024, 5, 1, 512, 32, 512, 1 /* sine wave */ i_ ftgen 6, 0, 1024, 10, 1 /* room parameters */ i_ ftgen 7, 0, 64, -2, 4, 50, -1, -1, -1, 11, \ 1, 26.833, 0.05, 0.85, 10000, 0.8, 0.5, 2, \ 1, 1.753, 0.05, 0.85, 5000, 0.8, 0.5, 2, \ 1, 39.451, 0.05, 0.85, 7000, 0.8, 0.5, 2, \ 1, 33.503, 0.05, 0.85, 7000, 0.8, 0.5, 2, \ 1, 36.151, 0.05, 0.85, 7000, 0.8, 0.5, 2, \ 1, 29.633, 0.05, 0.85, 7000, 0.8, 0.5, 2 /* generate bandlimited sawtooth waves */ i0 = 0 loop1: imaxh = sr / (2 * 440.0 * exp(log(2.0) * (i0 - 69) / 12)) i_ ftgen i0 + 256, 0, 4096, -30, 1, 1, imaxh i0 = i0 + 1 if (i0 < 127.5) igoto loop1 instr 1 p3 = p3 + 0.4 ; note frequency kcps = 440.0 * exp(log(2.0) * (p4 - 69) / 12) ; lowpass max. frequency klpmaxf limit 64 * kcps, 1000.0, 12000.0 ; FM depth in Hz kfmd1 = 0.02 * kcps ; AM frequency kamfr = kcps * 0.02 kamfr2 = kcps * 0.1 ; table number kfnum = (256 + 69 + 0.5 + 12 * log(kcps / 440.0) / log(2.0)) ; amp. envelope aenv linseg 0, 0.1, 1.0, p3 - 0.5, 1.0, 0.1, 0.5, 0.2, 0, 1.0, 0 /* oscillator / left */ a1 oscbnk kcps, 0.0, kfmd1, 0.0, 40, 200, 0.1, 0.2, 0, 0, 144, \ 0.0, klpmaxf, 0.0, 0.0, 1.5, 1.5, 2, \ kfnum, 3, 0, 5, 5, 5 a2 oscbnk kcps, 1.0, kfmd1, 0.0, 40, 201, 0.1, 0.2, kamfr, kamfr2, 148, \ 0, 0, 0, 0, 0, 0, -1, \ kfnum, 3, 4 a2 pareq a2, kcps * 8, 0.0, 0.7071, 2 a0 = a1 + a2 * 0.12 /* delay */ adel = 0.001 a01 vdelayx a0, adel, 0.01, 16 a_ oscili 1.0, 0.25, 6, 0.0 adel = adel + 1.0 / (exp(log(2.0) * a_) * 8000) a02 vdelayx a0, adel, 0.01, 16 a0 = a01 + a02 ga01 = ga01 + a0 * aenv * 2500 /* oscillator / right */ ; lowpass max. frequency a1 oscbnk kcps, 0.0, kfmd1, 0.0, 40, 202, 0.1, 0.2, 0, 0, 144, \ 0.0, klpmaxf, 0.0, 0.0, 1.0, 1.0, 2, \ kfnum, 3, 0, 5, 5, 5 a2 oscbnk kcps, 1.0, kfmd1, 0.0, 40, 203, 0.1, 0.2, kamfr, kamfr2, 148, \ 0, 0, 0, 0, 0, 0, -1, \ kfnum, 3, 4 a2 pareq a2, kcps * 8, 0.0, 0.7071, 2 a0 = a1 + a2 * 0.12 /* delay */ adel = 0.001 a01 vdelayx a0, adel, 0.01, 16 a_ oscili 1.0, 0.25, 6, 0.25 adel = adel + 1.0 / (exp(log(2.0) * a_) * 8000) a02 vdelayx a0, adel, 0.01, 16 a0 = a01 + a02 ga02 = ga02 + a0 * aenv * 2500 endin /* output / left */ instr 81 i1 = 0.000001 aLl, aLh, aRl, aRh spat3di ga01 + i1*i1*i1*i1, -8.0, 4.0, 0.0, 0.3, 7, 4 ga01 = 0 aLl butterlp aLl, 800.0 aRl butterlp aRl, 800.0 outs aLl + aLh, aRl + aRh endin /* output / right */ instr 82 i1 = 0.000001 aLl, aLh, aRl, aRh spat3di ga02 + i1*i1*i1*i1, 8.0, 4.0, 0.0, 0.3, 7, 4 ga02 = 0 aLl butterlp aLl, 800.0 aRl butterlp aRl, 800.0 outs aLl + aLh, aRl + aRh endin </CsInstruments> <CsScore> /* Written by Istvan Varga */ t 0 60 i 1 0 4 41 i 1 0 4 60 i 1 0 4 65 i 1 0 4 69 i 81 0 5.5 i 82 0 5.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/log.csd.xml0000644000000000000000000000320312262561504017432 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o log.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = log(8) print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/active.csd.xml0000644000000000000000000000562012262561504020131 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o active.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a noisy waveform. instr 1 ; Generate a really noisy waveform. anoisy rand 44100 ; Turn down its amplitude. aoutput gain anoisy, 2500 ; Send it to the output. out aoutput endin ; Instrument #2 - counts active instruments. instr 2 ; Count the active instances of Instrument #1. icount active 1 ; Print the number of active instances. print icount endin </CsInstruments> <CsScore> ; Start the first instance of Instrument #1 at 0:00 seconds. i 1 0.0 3.0 ; Start the second instance of Instrument #1 at 0:015 seconds. i 1 1.5 1.5 ; Play Instrument #2 at 0:01 seconds, when we have only ; one active instance of Instrument #1. i 2 1.0 0.1 ; Play Instrument #2 at 0:02 seconds, when we have ; two active instances of Instrument #1. i 2 2.0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dam_expanded.csd.xml0000644000000000000000000000431612262561504021270 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dam_expanded.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig diskin2 "beats.wav", 1, 0, 1 outs asig, asig endin instr 2 ;expanded audio kthreshold = .5 icomp1 = 2 icomp2 = 3 irtime = 0.01 iftime = 0.1 asig diskin2 "beats.wav", 1, 0, 1 asig dam asig, kthreshold, icomp1, icomp2, irtime, iftime outs asig*.2, asig*.2 ;adjust volume of expanded beat endin </CsInstruments> <CsScore> i 1 0 2 i 2 2.5 6.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/locsend_quad.csd.xml0000644000000000000000000000613412262561504021320 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o locsend_quad.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 4 ga1 init 0 ga2 init 0 ga3 init 0 ga4 init 0 ; Instrument #1 instr 1 ; Use the "beats.wav" audio file. aout diskin2 "beats.wav", 1, 0, 1 kdegree line 0, p3, 360 kdistance line 1, p3 , 10 a1, a2, a3, a4 locsig aout, kdegree, kdistance, .1 ar1, ar2, ar3, ar4 locsend ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 3.5, .5 a2 reverb2 ga2, 3.5, .5 a3 reverb2 ga3, 3.5, .5 a4 reverb2 ga4, 3.5, .5 outq a1, a2, a3, a4 ga1 = 0 ga2 = 0 ga3 = 0 ga4 = 0 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 i 1 0 2 i99 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/s.csd.xml0000644000000000000000000000433412262561504017121 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o s.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .8*aenv, p4, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1;sine wave s ;first section i 1 0 2 110 i 1 3 2 220 i 1 6 2 440 i 1 9 2 880 s ;second section i 1 0 2 880 i 1 3 2 440 i 1 6 2 220 i 1 9 2 110 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKPlucked.csd.xml0000644000000000000000000000332112262561504020623 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKPlucked.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;STKPlucked - has no controllers ifrq = p4 asig STKPlucked cpspch(ifrq), 1 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 6.00 i 1 + 8 5.00 i 1 + .5 8.00 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/adds.csd.xml0000644000000000000000000000421312262561504017566 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o adds.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; add unipolar square to oscil kamp = p4 kcps = 1 itype = 3 klfo lfo kamp, kcps, itype printk2 klfo asig oscil 0.7, 440+klfo, 1 outs asig, asig endin </CsInstruments> <CsScore> ;sine wave. f 1 0 32768 10 1 i 1 0 2 1 ;adds 1 Hz to frequency i 1 + 2 10 ;adds 10 Hz to frequency i 1 + 2 220 ;adds 220 Hz to frequency e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dumpk3.csd.xml0000644000000000000000000000726512262561504020070 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dumpk3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes three control signals to a file kfreq randh 100, 1, 2, 1, 500 ;generates one random number between 400 and 600 per second kdb randh 12, 1, 2, 1, -12 ;amplitudes in dB between -24 and 0 kpan randh .5, 1, 2, 1, .5 ;panning between 0 and 1 dumpk3 kfreq, kdb, kpan, "dumpk3.txt", 8, 1 ;writes the control signals prints "WRITING:\n" printks "kfreq = %f, kdb = %f, kpan = %f\n", 1, kfreq, kdb, kpan ;prints them endin instr 2 ;reads the file written by instr 1 kf,kdb,kp readk3 "dumpk3.txt", 8, 1 prints "READING:\n" printks "kfreq = %f, kdb = %f, kpan = %f\n", 1, kf, kdb, kp ;prints again kdb lineto kdb, .1 ;smoothing amp transition kp lineto kp, .1 ;smoothing pan transition aout poscil ampdb(kdb), kf, giSine aL, aR pan2 aout, kp outs aL, aR endin </CsInstruments> <CsScore> i 1 0 5 i 2 5 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sinh.csd.xml0000644000000000000000000000326312262561504017620 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o sinh.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 1 i1 = sinh(irad) print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/poscil3.csd.xml0000644000000000000000000000356112262561504020234 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o poscil3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 krnd randomh 40, 440, 1 ; produce random values ain poscil3 .6, krnd, giSine kline line 1, p3, 0 ; straight line aL,aR pan2 ain, kline ; sent across image outs aL, aR endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/moogvcf2.csd.xml0000644000000000000000000000343412262561504020401 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o moogvcf2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aout diskin2 "beats.wav", 1, 0, 1 kfco line 100, p3, 10000 ;filter-cutoff krez init p4 asig moogvcf2 aout, kfco, krez outs asig, asig endin </CsInstruments> <CsScore> i 1 0 4 .1 i 1 + 4 .6 i 1 + 4 .9 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/babo_expert.csd.xml0000644000000000000000000001741412262561504021154 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o babo_expert.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Nicola Bernardini */ sr = 44100 ksmps = 32 nchnls = 2 ; full blown babo instrument with movement ; instr 2 ixstart = p4 ; start x position of source (left-right) ixend = p7 ; end x position of source iystart = p5 ; start y position of source (front-back) iyend = p8 ; end y position of source izstart = p6 ; start z position of source (up-down) izend = p9 ; end z position of source ixsize = p10 ; width of the resonator iysize = p11 ; depth of the resonator izsize = p12 ; height of the resonator idiff = p13 ; diffusion coefficient iexpert = p14 ; power user values stored in this function ainput soundin "beats.wav" ksource_x line ixstart, p3, ixend ksource_y line iystart, p3, iyend ksource_z line izstart, p3, izend al,ar babo ainput*0.7, ksource_x, ksource_y, ksource_z, ixsize, iysize, izsize, idiff, iexpert outs al,ar endin </CsInstruments> <CsScore> /* Written by Nicola Bernardini */ ; full blown instrument ;p4 : start x position of source (left-right) ;p5 : end x position of source ;p6 : start y position of source (front-back) ;p7 : end y position of source ;p8 : start z position of source (up-down) ;p9 : end z position of source ;p10 : width of the resonator ;p11 : depth of the resonator ;p12 : height of the resonator ;p13 : diffusion coefficient ;p14 : power user values stored in this function ; decay hidecay rx ry rz rdistance direct early_diff f1 0 8 -2 0.95 0.95 0 0 0 0.3 0.5 0.8 ; brighter f2 0 8 -2 0.95 0.5 0 0 0 0.3 0.5 0.8 ; default (to be set as) f3 0 8 -2 0.95 0.01 0 0 0 0.3 0.5 0.8 ; darker f4 0 8 -2 0.95 0.7 0 0 0 0.3 0.1 0.4 ; to hear the effect of diffusion f5 0 8 -2 0.9 0.5 0 0 0 0.3 2.0 0.98 ; to hear the movement f6 0 8 -2 0.99 0.1 0 0 0 0.3 0.5 0.8 ; default vals ; ^ ; ----- gen. number: negative to avoid rescaling i2 0 10 6 4 3 6 4 3 14.39 11.86 10 1 6 ; defaults i2 + 4 6 4 3 6 4 3 14.39 11.86 10 1 1 ; hear brightness 1 i2 + 4 6 4 3 -6 -4 3 14.39 11.86 10 1 2 ; hear brightness 2 i2 + 4 6 4 3 -6 -4 3 14.39 11.86 10 1 3 ; hear brightness 3 i2 + 3 .6 .4 .3 -.6 -.4 .3 1.439 1.186 1.0 0.0 4 ; hear diffusion 1 i2 + 3 .6 .4 .3 -.6 -.4 .3 1.439 1.186 1.0 1.0 4 ; hear diffusion 2 i2 + 4 12 4 3 -12 -4 -3 24.39 21.86 20 1 5 ; hear movement ; i2 + 4 6 4 3 6 4 3 14.39 11.86 10 1 1 ; hear brightness 1 i2 + 4 6 4 3 -6 -4 3 14.39 11.86 10 1 2 ; hear brightness 2 i2 + 4 6 4 3 -6 -4 3 14.39 11.86 10 1 3 ; hear brightness 3 i2 + 3 .6 .4 .3 -.6 -.4 .3 1.439 1.186 1.0 0.0 4 ; hear diffusion 1 i2 + 3 .6 .4 .3 -.6 -.4 .3 1.439 1.186 1.0 1.0 4 ; hear diffusion 2 i2 + 4 12 4 3 -12 -4 -3 24.39 21.86 20 1 5 ; hear movement ; ^^^^^^^^^^^^^^^^^^^ ^^^^^^^^^^^^^^^^^ ^ ^ ; ||||||||||||||||||| ||||||||||||||||| | --: expert values function ; ||||||||||||||||||| ||||||||||||||||| +--: diffusion ; ||||||||||||||||||| ----------------: optimal room dims according to Milner and Bernard JASA 85(2), 1989 ; ||||||||||||||||||| ; --------------------: source position start and end e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/adsynt2.csd.xml0000644000000000000000000001415112262561504020241 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o adsynt2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; Generate a sinewave table. giwave ftgen 1, 0, 1024, 10, 1 ; Generate two empty tables for adsynt2. gifrqs ftgen 2, 0, 32, 7, 0, 32, 0 ; A table for freqency and amp parameters. giamps ftgen 3, 0, 32, 7, 0, 32, 0 ; Generates parameters at init time instr 1 ; Generate 10 voices. icnt = 10 ; Init loop index. index = 0 ; Loop only executed at init time. loop: ; Define non-harmonic partials. ifreq pow index + 1, 1.5 ; Define amplitudes. iamp = 1 / (index+1) ; Write to tables. tableiw ifreq, index, gifrqs ; Used by adsynt2. tableiw iamp, index, giamps index = index + 1 ; Do loop/ if (index < icnt) igoto loop asig adsynt2 0.4, 150, giwave, gifrqs, giamps, icnt outs asig, asig endin ; Generates parameters every k-cycle. instr 2 ; Generate 10 voices. icnt = 10 ; Reset loop index. kindex = 0 ; Loop executed every k-cycle. loop: ; Generate lfo for frequencies. kspeed pow kindex + 1, 1.6 ; Individual phase for each voice. kphas phasorbnk kspeed * 0.7, kindex, icnt klfo table kphas, giwave, 1 ; Arbitrary parameter twiddling... kdepth pow 1.4, kindex kfreq pow kindex + 1, 1.5 kfreq = kfreq + klfo*0.006*kdepth ; Write freqs to table for adsynt2. tablew kfreq, kindex, gifrqs ; Generate lfo for amplitudes. kspeed pow kindex + 1, 0.8 ; Individual phase for each voice. kphas phasorbnk kspeed*0.13, kindex, icnt, 2 klfo table kphas, giwave, 1 ; Arbitrary parameter twiddling... kamp pow 1 / (kindex + 1), 0.4 kamp = kamp * (0.3+0.35*(klfo+1)) ; Write amps to table for adsynt2. tablew kamp, kindex, giamps kindex = kindex + 1 ; Do loop. if (kindex < icnt) kgoto loop asig adsynt2 0.25, 150, giwave, gifrqs, giamps, icnt outs asig, asig endin </CsInstruments> <CsScore> ; Play Instrument #1 for 2.5 seconds. i 1 0 2.5 ; Play Instrument #2 for 2.5 seconds. i 2 3 2.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vcopy.csd.xml0000644000000000000000000000514112262561504020014 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vcopy.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 kr=4410 ksmps=10 nchnls=2 instr 1 ;table playback ar lposcil 1, 1, 0, 262144, 1 outs ar,ar endin instr 2 vcopy 2, 1, 20000 ;copy vector from sample to empty table vmult 5, 20000, 262144 ;scale noise to make it audible vcopy 1, 5, 20000 ;put noise into sample turnoff endin instr 3 vcopy 1, 2, 20000 ;put original information back in turnoff endin </CsInstruments> <CsScore> f1 0 262144 -1 "beats.wav" 0 4 0 f2 0 262144 2 0 f5 0 262144 21 3 30000 i1 0 4 i2 3 1 s i1 0 4 i3 3 1 s i1 0 4 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/rbjeq.csd.xml0000644000000000000000000000413512262561504017761 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o rbjeq.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 imode = p4 a1 vco2 .3, 155.6 ; sawtooth wave kfco expon 8000, p3, 200 ; filter frequency asig rbjeq a1, kfco, 1, kfco * 0.005, 1, imode outs asig, asig endin </CsInstruments> <CsScore> i 1 0 5 0 ;lowpass i 1 6 5 2 ;highpass i 1 12 5 4 ;bandpass i 1 18 5 8 ;equalizer e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/genarray.csd.xml0000644000000000000000000000540512262561504020467 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;create and fill two arrays iArr1[] genarray 1, 6 iArr2[] genarray 1, 6, 2/3 ;print the content of iArr1 printf "%s", 1, "iArr1: start=1, end=6, step=default\n" kndx = 0 until kndx == lenarray(iArr1) do printf "iArr[%d] = %f\n", kndx+1, kndx, iArr1[kndx] kndx += 1 od ;print the content of iArr2 printf "%s", 1, "iArr2: start=1, end=6, step=2/3\n" kndx = 0 until kndx == lenarray(iArr2) do printf "iArr[%d] = %f\n", kndx+1, kndx, iArr2[kndx] kndx += 1 od turnoff endin </CsInstruments> <CsScore> i 1 0 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsbufread.csd.xml0000644000000000000000000000647112262561504021024 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsbufread.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 opcode FileToPvsBuf, iik, Siiii ;;writes an audio file at the first k-cycle to a fft-buffer (via pvsbuffer) Sfile, ifftsize, ioverlap, iwinsize, iwinshape xin ktimek timeinstk if ktimek == 1 then ilen filelen Sfile kcycles = ilen * kr; number of k-cycles to write the fft-buffer kcount init 0 loop: ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape ibuf, ktim pvsbuffer fftin, ilen + (ifftsize / sr) loop_lt kcount, 1, kcycles, loop xout ibuf, ilen, ktim endif endop instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window ibuffer, ilen, k0 FileToPvsBuf "fox.wav", ifftsize, ioverlap, iwinsize, iwinshape ktmpnt linseg ilen, p3, 0; reads the buffer backwards in p3 seconds fread pvsbufread ktmpnt, ibuffer aout pvsynth fread out aout endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mediank.csd.xml0000644000000000000000000000220712262561504020264 0ustar rootroot <CsoundSynthesizer> <CsOptions> -n </CsOptions> <CsInstruments> sr = 44100 kr = 147 ksmps = 300 nchnls = 1 instr 1 k1 oscil 100, 10, 1 k2 mediank k1, 5, 8 printk 0, k2 endin </CsInstruments> <CsScore> f1 0 4096 10 1 i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/PartikkelExample1.csd.xml0000644000000000000000000003137512262561504022207 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out -odac ;;;RT audio ; For Non-realtime ouput leave only the line below: ; -o partikkel.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 20 nchnls = 2 ; Example by Joachim Heintz and Oeyvind Brandtsegg 2008 giCosine ftgen 0, 0, 8193, 9, 1, 1, 90 ; cosine giDisttab ftgen 0, 0, 32768, 7, 0, 32768, 1 ; for kdistribution giFile ftgen 0, 0, 0, 1, "fox.wav", 0, 0, 0 ; soundfile for source waveform giWin ftgen 0, 0, 4096, 20, 9, 1 ; grain envelope giPan ftgen 0, 0, 32768, -21, 1 ; for panning (random values between 0 and 1) ; ************************************************* ; partikkel example, processing of soundfile ; uses the file "fox.wav" ; ************************************************* instr 1 /*score parameters*/ ispeed = p4 ; 1 = original speed igrainrate = p5 ; grain rate igrainsize = p6 ; grain size in ms icent = p7 ; transposition in cent iposrand = p8 ; time position randomness (offset) of the pointer in ms icentrand = p9 ; transposition randomness in cents ipan = p10 ; panning narrow (0) to wide (1) idist = p11 ; grain distribution (0=periodic, 1=scattered) /*get length of source wave file, needed for both transposition and time pointer*/ ifilen tableng giFile ifildur = ifilen / sr /*sync input (disabled)*/ async = 0 /*grain envelope*/ kenv2amt = 1 ; use only secondary envelope ienv2tab = giWin ; grain (secondary) envelope ienv_attack = -1 ; default attack envelope (flat) ienv_decay = -1 ; default decay envelope (flat) ksustain_amount = 0.5 ; no meaning in this case (use only secondary envelope, ienv2tab) ka_d_ratio = 0.5 ; no meaning in this case (use only secondary envelope, ienv2tab) /*amplitude*/ kamp = 0.4*0dbfs ; grain amplitude igainmasks = -1 ; (default) no gain masking /*transposition*/ kcentrand rand icentrand ; random transposition iorig = 1 / ifildur ; original pitch kwavfreq = iorig * cent(icent + kcentrand) /*other pitch related (disabled)*/ ksweepshape = 0 ; no frequency sweep iwavfreqstarttab = -1 ; default frequency sweep start iwavfreqendtab = -1 ; default frequency sweep end awavfm = 0 ; no FM input ifmamptab = -1 ; default FM scaling (=1) kfmenv = -1 ; default FM envelope (flat) /*trainlet related (disabled)*/ icosine = giCosine ; cosine ftable kTrainCps = igrainrate ; set trainlet cps equal to grain rate for single-cycle trainlet in each grain knumpartials = 1 ; number of partials in trainlet kchroma = 1 ; balance of partials in trainlet /*panning, using channel masks*/ imid = .5; center ileftmost = imid - ipan/2 irightmost = imid + ipan/2 giPanthis ftgen 0, 0, 32768, -24, giPan, ileftmost, irightmost ; rescales giPan according to ipan tableiw 0, 0, giPanthis ; change index 0 ... tableiw 32766, 1, giPanthis ; ... and 1 for ichannelmasks ichannelmasks = giPanthis ; ftable for panning /*random gain masking (disabled)*/ krandommask = 0 /*source waveforms*/ kwaveform1 = giFile ; source waveform kwaveform2 = giFile ; all 4 sources are the same kwaveform3 = giFile kwaveform4 = giFile iwaveamptab = -1 ; (default) equal mix of source waveforms and no amplitude for trainlets /*time pointer*/ afilposphas phasor ispeed / ifildur /*generate random deviation of the time pointer*/ iposrandsec = iposrand / 1000 ; ms -> sec iposrand = iposrandsec / ifildur ; phase values (0-1) krndpos linrand iposrand ; random offset in phase values /*add random deviation to the time pointer*/ asamplepos1 = afilposphas + krndpos; resulting phase values (0-1) asamplepos2 = asamplepos1 asamplepos3 = asamplepos1 asamplepos4 = asamplepos1 /*original key for each source waveform*/ kwavekey1 = 1 kwavekey2 = kwavekey1 kwavekey3 = kwavekey1 kwavekey4 = kwavekey1 /* maximum number of grains per k-period*/ imax_grains = 100 aL, aR partikkel igrainrate, idist, giDisttab, async, kenv2amt, ienv2tab, \ ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, igrainsize, kamp, igainmasks, \ kwavfreq, ksweepshape, iwavfreqstarttab, iwavfreqendtab, awavfm, \ ifmamptab, kfmenv, icosine, kTrainCps, knumpartials, \ kchroma, ichannelmasks, krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, asamplepos4, \ kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains outs aL, aR endin </CsInstruments> <CsScore> ;i1 st dur speed grate gsize cent posrnd cntrnd pan dist i1 0 2.757 1 200 15 0 0 0 0 0 s i1 0 2.757 1 200 15 400 0 0 0 0 s i1 0 2.757 1 15 450 400 0 0 0 0 s i1 0 2.757 1 15 450 400 0 0 0 0.4 s i1 0 2.757 1 200 15 0 400 0 0 1 s i1 0 5.514 .5 200 20 0 0 600 .5 1 s i1 0 11.028 .25 200 15 0 1000 400 1 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vbap8.csd.xml0000644000000000000000000000531312262561504017675 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in ;-odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -o vbap8.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 441 ksmps = 100 nchnls = 4 vbaplsinit 2, 8, 0, 45, 90, 135, 200, 245, 290, 315 instr 1 asig oscil 20000, 440, 1 a1,a2,a3,a4,a5,a6,a7,a8 vbap8 asig, p4, 0, 20 ;p4 = azimuth ;render twice with alternate outq statements ; to obtain two 4 channel .wav files: outq a1,a2,a3,a4 ; outq a5,a6,a7,a8 ; or use an 8-channel output for realtime output (set nchnls to 8): ; outo a1,a2,a3,a4,a5,a6,a7,a8 endin </CsInstruments> <CsScore> f 1 0 8192 10 1 ; Play Instrument #1 for one second. ; azimuth i 1 0 1 20 i 1 + . 40 i 1 + . 60 i 1 + . 80 i 1 + . 100 i 1 + . 120 i 1 + . 140 i 1 + . 160 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKHevyMetl.csd.xml0000644000000000000000000000406012262561504020772 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKHevyMetl.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;Total Modulator Index kv2 line p7, p3, 0 ;Modulator Crossfade asig STKHevyMetl cpspch(ifrq), 1, 2, kv1, 4, kv2, 11, 0, 1, 100, 128, 40 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 7 8.05 100 0 100 i 1 3 7 9.03 20 120 0 i 1 3 .5 8.05 20 120 0 i 1 4 .5 9.09 20 120 0 i 1 5 3 9.00 20 120 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/flooper.csd.xml0000644000000000000000000000365512262561504020332 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o flooper.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kpitch line 1, p3, .9 ;lower pitch a bit during the note aout flooper .9, kpitch, 1, .53, 0.05, 1 ; loop starts at 1 sec, for .53 secs, 0.05 crossfade outs aout, aout endin </CsInstruments> <CsScore> ;table size is deferred, ; and format taken from the soundfile header f 1 0 0 1 "fox.wav" 0 0 0 i 1 0 8.2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsfwrite.csd.xml0000644000000000000000000000415612262561504020712 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsfwrite.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ;By Victor Lazzarini 2008 instr 1 asig oscili 10000, 440, 1 fss pvsanal asig, 1024,256,1024,0 pvsfwrite fss, "mypvs.pvx" ase pvsynth fss out ase endin instr 2 ; must be called after instr 1 finishes ktim timeinsts fss pvsfread ktim, "mypvs.pvx" asig pvsynth fss out asig endin </CsInstruments> <CsScore> f1 0 16384 10 1 i1 0 1 i2 1 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vaget.csd.xml0000644000000000000000000000652312262561504017767 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o avarget.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=16 nchnls=2 instr 1 ; Sqrt Signal ifreq = (p4 > 15 ? p4 : cpspch(p4)) iamp = ampdb(p5) aout init 0 ksampnum init 0 kenv linseg 0, p3 * .5, 1, p3 * .5, 0 aout1 vco2 1, ifreq aout2 vco2 .5, ifreq * 2 aout3 vco2 .2, ifreq * 4 aout sum aout1, aout2, aout3 ;Take Sqrt of signal, checking for negatives kcount = 0 loopStart: kval vaget kcount,aout if (kval > .0) then kval = sqrt(kval) elseif (kval < 0) then kval = sqrt(-kval) * -1 else kval = 0 endif vaset kval, kcount,aout loop_lt kcount, 1, ksmps, loopStart aout = aout * kenv aout moogladder aout, 8000, .1 aout = aout * iamp outs aout, aout endin </CsInstruments> <CsScore> i1 0.0 2 440 80 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpsoct.csd.xml0000644000000000000000000000367512262561504020161 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpsoct.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Convert octave-point-decimal value into Hz ioct = p4 icps = cpsoct(ioct) print icps asig oscil 0.7, icps, 1 outs asig, asig endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 1 8.75 i 1 + 1 8.77 i 1 + 1 8.79 i 1 + .5 6.30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/p5g.csd.xml0000644000000000000000000000736112262561504017355 0ustar rootroot <CsoundSynthesizer> <CsOptions> -+rtaudio=alsa -o dac:hw:0 </CsOptions> <CsInstruments> nchnls = 1 ksmps = 1000 #define P5G_BUTTONS #0# #define P5G_BUTTON_A #1# #define P5G_BUTTON_B #2# #define P5G_BUTTON_C #4# #define P5G_JUSTPUSH #8# #define P5G_JUSTPU_A #9# #define P5G_JUSTPU_B #10# #define P5G_JUSTPU_C #12# #define P5G_RELEASED #16# #define P5G_RELSED_A #17# #define P5G_RELSED_B #18# #define P5G_RELSED_C #20# #define P5G_FINGER_INDEX #32# #define P5G_FINGER_MIDDLE #33# #define P5G_FINGER_RING #34# #define P5G_FINGER_PINKY #35# #define P5G_FINGER_THUMB #36# #define P5G_DELTA_X #37# #define P5G_DELTA_Y #38# #define P5G_DELTA_Z #39# #define P5G_DELTA_XR #40# #define P5G_DELTA_YR #41# #define P5G_DELTA_ZR #42# #define P5G_ANGLES #43# gka init 0 gkp init 0 instr 1 p5gconnect ka p5gdata $P5G_JUSTPU_A. kc p5gdata $P5G_BUTTON_C. ; If the A button is just pressed then activate a note if (ka==0) goto ee event "i", 2, 0, 2 ee: gka p5gdata $P5G_DELTA_X. gkp p5gdata $P5G_DELTA_Y. printk2 gka printk2 gkp if (kc==0) goto ff printks "turning off (%d)\n", 0, kc turnoff ff: endin instr 2 a1 oscil ampdbfs(gkp), 440+100*gka, 1 ;; a1 oscil 10000, 440, 1 out a1 endin </CsInstruments> <CsScore> f1 0 4096 10 1 i1 0 300 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsfreeze.csd.xml0000644000000000000000000000422112262561504020663 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 seed 0 instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "fox.wav" ain soundin Sfile1 kfreq randomh .7, 1.1, 3; probability of freezing freqs: 1/4 kamp randomh .7, 1.1, 3; idem for amplitudes fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of file freeze pvsfreeze fftin, kamp, kfreq; freeze amps or freqs independently aout pvsynth freeze; resynthesize out aout endin </CsInstruments> <CsScore> r 10 i 1 0 2.757 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpsmidinn.csd.xml0000644000000000000000000000455712262561504020652 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform. ; This example produces no audio, so we render in ; non-realtime and turn off sound to disk: -n </CsOptions> <CsInstruments> instr 1 ; i-time loop to print conversion table imidiNN = 0 loop1: icps = cpsmidinn(imidiNN) ioct = octmidinn(imidiNN) ipch = pchmidinn(imidiNN) print imidiNN, icps, ioct, ipch imidiNN = imidiNN + 1 if (imidiNN < 128) igoto loop1 endin instr 2 ; test k-rate converters kMiddleC = 60 kcps = cpsmidinn(kMiddleC) koct = octmidinn(kMiddleC) kpch = pchmidinn(kMiddleC) printks "%d %f %f %f\n", 1.0, kMiddleC, kcps, koct, kpch endin </CsInstruments> <CsScore> i1 0 0 i2 0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/spsend_stereo.csd.xml0000644000000000000000000000537612262561504021543 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o spsend_stereo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ga1 init 0 ga2 init 0 instr 1 asig diskin2 "beats.wav", 1 a1, a2, a3, a4 space asig, 0, 0, .1, p4, p5 ; take position values from p4, p5 ar1, ar2, ar3, ar4 spsend ; a3, a4, ar3, ar4 are dummies ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a2 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 outs a1, a2 ga1=0 ga2=0 endin </CsInstruments> <CsScore> ;place the sound in the left speaker and near i1 0 1 -1 1 ;place the sound in the right speaker and far i1 1 1 45 45 ;place the sound equally between left and right and in the middle ground distance i1 2 1 0 12 i 99 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/syncgrain.csd.xml0000644000000000000000000000471212262561504020654 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o syncgrain.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 iolaps = 2 igrsize = 0.04 ifreq = iolaps/igrsize ips = 1/iolaps istr = .3 /* timescale */ ipitch = p4 /* pitchscale */ asig syncgrain 1, ifreq, ipitch, igrsize, ips*istr, 1, 2, iolaps outs asig, asig endin </CsInstruments> <CsScore> f1 0 0 1 "fox.wav" 0 0 0 ;deferred table f2 0 8192 20 2 1 i1 0 5 1 i1 + 5 4 i1 + 5 .8 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/statevar.csd.xml0000644000000000000000000000412512262561504020506 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o statevar.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 kenv linseg 0,0.1,1, p3-0.2,1, 0.1, 0 ;declick envelope asig buzz .6*kenv, 100, 100, 1 kf expseg 100, p3/2, 5000, p3/2, 1000 ;envelope for filter cutoff ahp,alp,abp,abr statevar asig, kf, 4 outs alp,ahp ; lowpass left, highpass right endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/madsr.csd.xml0000644000000000000000000000505512262561504017766 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o madsr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Iain McCurdy */ ; Initialize the global variables. sr = 44100 kr = 441 ksmps = 100 nchnls = 1 ; Instrument #1. instr 1 ; Attack time. iattack = 0.5 ; Decay time. idecay = 0 ; Sustain level. isustain = 1 ; Release time. irelease = 0.5 aenv madsr iattack, idecay, isustain, irelease a1 oscili 10000, 440, 1 out a1*aenv endin </CsInstruments> <CsScore> /* Written by Iain McCurdy */ ; Table #1, a sine wave. f 1 0 1024 10 1 ; Leave the score running for 6 seconds. f 0 6 ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lowpass2.csd.xml0000644000000000000000000000547412262561504020437 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o lowpass2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Sean Costello */ ; Orchestra file for resonant filter sweep of a sawtooth-like waveform. sr = 44100 kr = 2205 ksmps = 20 nchnls = 1 instr 1 idur = p3 ifreq = p4 iamp = p5 * .5 iharms = (sr*.4) / ifreq ; Sawtooth-like waveform asig gbuzz 1, ifreq, iharms, 1, .9, 1 ; Envelope to control filter cutoff kfreq linseg 1, idur * 0.5, 5000, idur * 0.5, 1 afilt lowpass2 asig,kfreq, 30 ; Simple amplitude envelope kenv linseg 0, .1, iamp, idur -.2, iamp, .1, 0 out afilt * kenv endin </CsInstruments> <CsScore> /* Written by Sean Costello */ f1 0 8192 9 1 1 .25 i1 0 5 100 1000 i1 5 5 200 1000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dumpk2.csd.xml0000644000000000000000000000633612262561504020065 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dumpk2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes two control signals to a file kfreq randh 100, 1, 2, 1, 500 ;generates one random number between 400 and 600 per second kdb randh 12, 1, 2, 1, -12 ;amplitudes in dB between -24 and 0 dumpk2 kfreq, kdb, "dumpk2.txt", 8, 1 ;writes the control signals prints "WRITING:\n" printks "kfreq = %f, kdb = %f\n", 1, kfreq, kdb ;prints them endin instr 2 ;reads the file written by instr 1 kf,kdb readk2 "dumpk2.txt", 8, 1 prints "READING:\n" printks "kfreq = %f, kdb = %f\n", 1, kf, kdb ;prints again kdb lineto kdb, .1 ;smoothing amp transition aout poscil ampdb(kdb), kf, giSine outs aout, aout endin </CsInstruments> <CsScore> i 1 0 5 i 2 5 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/serialBegin.csd.xml0000644000000000000000000000541012262561504021077 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n ;;;no output ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o serialBegin.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 500 ; the default krate can be too fast for the arduino to handle nchnls = 2 0dbfs = 1 instr 1 iPort serialBegin "/dev/cu.usbmodemfa131", 9600 ;connect to the arduino with baudrate = 9600 kGain init 16 ;read our knob value kVal serialRead iPort if (kVal != -1) then kGain = kVal/128 endif aSig in ;get our audio input and get its rms kRms rms aSig*kGain kRms = kRms*kRms*255 ;scale the rms to a good value for the LED and send it out serialWrite iPort, (kRms < 255 ? kRms : 255) ;must be in range: 0-255 endin </CsInstruments> <CsScore> f 1 0 1024 10 1 1 1 1 1 1 i 1 0 200 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wii.csd.xml0000644000000000000000000000565312262561504017454 0ustar rootroot <CsoundSynthesizer> <CsOptions> -+rtaudio=alsa -o dac:hw:0 </CsOptions> <CsInstruments> nchnls = 2 ksmps = 400 #define WII_B #3# #define WII_A #4# #define WII_R_A #304# #define WII_PITCH #20# #define WII_ROLL #21# #define WII_BATTERY #27# #define WII_RUMBLE #3# #define WII_SET_LEDS #4# gkcnt init 1 instr 1 i1 wiiconnect 3,1 wiirange $WII_PITCH., -20, 0 kb wiidata $WII_BATTERY. kt wiidata $WII_B. ka wiidata $WII_A. kra wiidata $WII_R_A. gka wiidata $WII_PITCH. gkp wiidata $WII_ROLL. ; If the B (trigger) button is pressed then activate a note if (kt==0) goto ee event "i", 2, 0, 5 gkcnt = gkcnt + 1 wiisend $WII_SET_LEDS., gkcnt ee: if (ka==0) goto ff wiisend $WII_RUMBLE., 1 ff: if (kra==0) goto gg wiisend $WII_RUMBLE., 0 gg: printk2 kb endin instr 2 a1 oscil ampdbfs(gka), 440+gkp, 1 outs a1, a1 endin </CsInstruments> <CsScore> f1 0 4096 10 1 i1 0 300 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/bitwise.csd.xml0000644000000000000000000000621312262561504020323 0ustar rootroot <CsoundSynthesizer> <CsOptions> </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 iresultOr = p4 | p5 iresultAnd = p4 & p5 prints "%i | %i = %i\\n", p4, p5, iresultOr prints "%i & %i = %i\\n", p4, p5, iresultAnd endin instr 2 ; decimal to binary converter Sbinary = "" inumbits = 8 icount init inumbits - 1 pass: ivalue = 2 ^ icount if (p4 & ivalue >= ivalue) then Sdigit = "1" else Sdigit = "0" endif Sbinary strcat Sbinary, Sdigit loop_ge icount, 1, 0, pass Stext sprintf "%i is %s in binary\\n", p4, Sbinary prints Stext endin </CsInstruments> <CsScore> i 1 0 0.1 1 2 i 1 + . 1 3 i 1 + . 2 4 i 1 + . 3 10 i 2 2 0.1 12 i 2 + . 9 i 2 + . 15 i 2 + . 49 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/foscili.csd.xml0000644000000000000000000000413012262561504020301 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o foscili.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 440 kcar = 1 kmod = p4 kndx line 0, p3, 20 ;intensivy sidebands asig foscili .5, kcps, kcar, kmod, kndx, 1 outs asig, asig endin </CsInstruments> <CsScore> ; sine f 1 0 16384 10 1 i 1 0 9 .01 ;vibrato i 1 10 . 1 i 1 20 . 1.414 ;gong-ish i 1 30 5 2.05 ;with "beat" e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLscroll.csd.xml0000644000000000000000000000456112262561504020401 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLscroll.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Demonstration of the flscroll opcode which enables ; the use of widget sizes and placings beyond the ; dimensions of the containing panel sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Text Box", 420, 200, 50, 50 iwidth = 420 iheight = 200 ix = 0 iy = 0 FLscroll iwidth, iheight, ix, iy ih3 FLbox "DRAG THE SCROLL BAR TO THE RIGHT IN ORDER TO READ THE REST OF THIS TEXT!", 1, 10, 20, 870, 30, 10, 100 FLscrollEnd ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 endin </CsInstruments> <CsScore> ; 'Dummy' score event of 1 hour. f 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/distort.csd.xml0000644000000000000000000000376412262561504020355 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o distort.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifn ftgen 0,0, 257, 9, .5,1,270 ; define a sigmoid, or better ;gifn ftgen 0,0, 257, 9, .5,1,270,1.5,.33,90,2.5,.2,270,3.5,.143,90,4.5,.111,270 instr 1 kdist line 0, p3, 2 ; and over 10 seconds asig poscil 0.3, 440, 1 aout distort asig, kdist, gifn ; gradually increase the distortion outs aout, aout endin </CsInstruments> <CsScore> f 1 0 16384 10 1 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKMoog.csd.xml0000644000000000000000000000373412262561504020145 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKMoog.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;filter Q asig STKMoog cpspch(ifrq), 1, 2,kv1, 4, 120, 11, 40, 1, 1, 128, 120 asig = asig * .3 ;too loud outs asig, asig endin </CsInstruments> <CsScore> i 1 0 .5 6.00 100 0 i 1 + . 5.05 10 127 i 1 + . 7.06 100 0 i 1 + 3 7.00 10 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/rnd31_krate.csd.xml0000644000000000000000000000357012262561504020775 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd31_krate.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create random numbers at k-rate in the range -1 to 1 ; with a uniform distribution, seed=10. k1 rnd31 1, 0, 10 printks "k1=%f\\n", 0.1, k1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/readk4.csd.xml0000644000000000000000000001041212262561504020023 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o readk4.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes four control signals to a file kcf randh 950, 1, 2, 1, 1050 ;generates one random number between 100 and 2000 per second kq randh 10, 1, 2, 1, 11 ;generates another random number between 1 and 21 per second kdb randh 9, 1, 2, 1, -15 ;amplitudes in dB between -24 and -6 kpan randh .5, 1, 2, 1, .5 ;panning between 0 and 1 dumpk4 kcf, kq, kdb, kpan, "dumpk4.txt", 8, 1 ;writes the control signals prints "WRITING:\n" printks "kcf = %f, kq = %f, kdb = %f, kpan = %f\n", 1, kcf, kq, kdb, kpan ;prints them endin instr 2 ;reads the file written by instr 1 kcf,kq,kdb,kp readk4 "dumpk4.txt", 8, 1 prints "READING:\n" printks "kcf = %f, kq = %f, kdb = %f, kpan = %f\n", 1, kcf, kq, kdb, kp ;prints values kdb lineto kdb, .1 ;smoothing amp transition kp lineto kp, .1 ;smoothing pan transition anoise rand ampdb(kdb), 2, 1 kbw = kcf/kq ;bandwidth of resonant filter abp reson anoise, kcf, kbw aout balance abp, anoise aL, aR pan2 aout, kp outs aL, aR endin </CsInstruments> <CsScore> i 1 0 5 i 2 5 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/moogvcf.csd.xml0000644000000000000000000000406312262561504020316 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o moogvcf.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; iscale does not need to be set here because already 0dbfs = 1 aout vco .3, 220, 1 ; Use a nice sawtooth waveform. kfco line 200, p3, 2000 ; filter-cutoff frequency from .2 to 2 KHz krez init p4 asig moogvcf aout, kfco, krez outs asig, asig endin </CsInstruments> <CsScore> ;a sine wave f 1 0 16384 10 1 i 1 0 3 .1 i 1 + 3 .7 i 1 + 3 .95 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/printk.csd.xml0000644000000000000000000000353612262561504020171 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o printk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Change a value linearly from 0 to 100, ; over the period defined by p3. kval line 0, p3, 100 ; Print the value of kval, once per second. printk 1, kval endin </CsInstruments> <CsScore> ; Play Instrument #1 for 5 seconds. i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tabslice.csd.xml0000644000000000000000000000263112262561504020443 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kS[] init 10 kT[] init 5 kS tabgen 0,10,0.01 kT[] tabslice kS,3,7 printk2 kT[0] printk2 kT[1] printk2 kT[2] printk2 kT[3] printk2 kT[4] endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluidOut.csd.xml0000644000000000000000000001016412262561504020450 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidOut.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine1 fluidEngine ; soundfont path to manual/examples isfnum1 fluidLoad "01hpschd.sf2", giengine1, 1 fluidProgramSelect giengine1, 1, isfnum1, 0, 0 giengine2 fluidEngine ; soundfont path to manual/examples isfnum2 fluidLoad "22Bassoon.sf2", giengine2, 1 fluidProgramSelect giengine2, 1, isfnum2, 0, 70 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine1, 1, ikey, ivel endin instr 2 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine2, 1, ikey, ivel endin instr 98 imvol init 7 asigl, asigr fluidOut giengine1 outs asigl*imvol, asigr*imvol endin instr 99 imvol init 4 asigl, asigr fluidOut giengine2 ;add a stereo flanger adelL linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time =20ms adelR linseg 0.02, p3*.5, 0, p3*.5, 0.02 ;max delay time =20ms asigL flanger asigl, adelL, .6 asigR flanger asigr, adelR, .6 outs asigL*imvol, asigR*imvol endin </CsInstruments> <CsScore> i 1 0 2 60 100 ;play one note of instr 1 i 2 2 2 60 100 ;play another note of instr 2 and... i 98 0 60 ;play virtual keyboard for 60 sec. i 99 0 60 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluid.orc.xml0000644000000000000000000000237412262561504017776 0ustar rootroot sr = 44100 ksmps = 100 nchnls = 2 giengine fluidEngine isfnum fluidLoad "07AcousticGuitar.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 0 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 70000 asigl, asigr fluidOut giengine outs asigl * imvol, asigr * imvol endinmanual_src~dfsg/examples-xml/array_udo.csd.xml0000644000000000000000000000217312262561504020643 0ustar rootroot <CsoundSynthesizer> <CsOptions> -nm128 </CsOptions> <CsInstruments> opcode FirstEl, k, k[] ;returns the first element of vector kArr kArr[] xin xout kArr[0] endop instr 1 kArr[] fillarray 6, 3, 9, 5, 1 kFirst FirstEl kArr printf "kFirst = %d\n", 1, kFirst turnoff endin </CsInstruments> <CsScore> i 1 0 .1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKBlowBotl.csd.xml0000644000000000000000000000373012262561504020764 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKBlowBotl.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = p4 kv1 line p5, p3, p6 ;gain of noise kv4 line 100, p3, 70 ;volume asig STKBlowBotl cpspch(ipch), 1, 4, kv1, 11, 10, 1, 50, 128, kv4 asig = asig * .5 ;too loud outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 9.00 20 100 i 1 + 5 8.03 120 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tbvcf.csd.xml0000644000000000000000000001404512262561504017763 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tbvcf.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;--------------------------------------------------------- ; TBVCF Test ; Coded by Hans Mikelson December, 2000 ;--------------------------------------------------------- sr = 44100 ; Sample rate ksmps = 10 ; Samples/Kontrol period nchnls = 2 ; Normal stereo 0dbfs = 1 instr 10 idur = p3 ; Duration iamp = p4 ; Amplitude ifqc = cpspch(p5) ; Pitch to frequency ipanl = sqrt(p6) ; Pan left ipanr = sqrt(1-p6) ; Pan right iq = p7 idist = p8 iasym = p9 kdclck linseg 0, .002, 1, idur-.004, 1, .002, 0 ; Declick envelope kfco expseg 10000, idur, 1000 ; Frequency envelope ax vco 1, ifqc, 2, 0.5 ; Square wave ay tbvcf ax, kfco, iq, idist, iasym ; TB-VCF ay buthp ay/1, 100 ; Hi-pass outs ay*iamp*ipanl*kdclck, ay*iamp*ipanr*kdclck endin </CsInstruments> <CsScore> f1 0 65536 10 1 ; TeeBee Test ; Sta Dur Amp Pitch Pan Q Dist1 Asym i10 0 0.2 0.5 7.00 .5 0.0 2.0 0.0 i10 0.3 0.2 0.5 7.00 .5 0.8 2.0 0.0 i10 0.6 0.2 0.5 7.00 .5 1.6 2.0 0.0 i10 0.9 0.2 0.5 7.00 .5 2.4 2.0 0.0 i10 1.2 0.2 0.5 7.00 .5 3.2 2.0 0.0 ;i10 1.5 0.2 0.5 7.00 .5 4.0 2.0 0.0 i10 1.8 0.2 0.5 7.00 .5 0.0 2.0 0.25 i10 2.1 0.2 0.5 7.00 .5 0.8 2.0 0.25 i10 2.4 0.2 0.5 7.00 .5 1.6 2.0 0.25 i10 2.7 0.2 0.5 7.00 .5 2.4 2.0 0.25 i10 3.0 0.2 0.5 7.00 .5 3.2 2.0 0.25 i10 3.3 0.2 0.5 7.00 .5 4.0 2.0 0.25 i10 3.6 0.2 0.5 7.00 .5 0.0 2.0 0.5 i10 3.9 0.2 0.5 7.00 .5 0.8 2.0 0.5 i10 4.2 0.2 0.5 7.00 .5 1.6 2.0 0.5 i10 4.5 0.2 0.5 7.00 .5 2.4 2.0 0.5 i10 4.8 0.2 0.5 7.00 .5 3.2 2.0 0.5 i10 5.1 0.2 0.5 7.00 .5 4.0 2.0 0.5 i10 5.4 0.2 0.5 7.00 .5 0.0 2.0 0.75 i10 5.7 0.2 0.5 7.00 .5 0.8 2.0 0.75 i10 6.0 0.2 0.5 7.00 .5 1.6 2.0 0.75 i10 6.3 0.2 0.5 7.00 .5 2.4 2.0 0.75 i10 6.6 0.2 0.5 7.00 .5 3.2 2.0 0.75 i10 6.9 0.2 0.5 7.00 .5 4.0 2.0 0.75 i10 7.2 0.2 0.5 7.00 .5 0.0 2.0 1.0 i10 7.5 0.2 0.5 7.00 .5 0.8 2.0 1.0 i10 7.8 0.2 0.5 7.00 .5 1.6 2.0 1.0 i10 8.1 0.2 0.5 7.00 .5 2.4 2.0 1.0 i10 8.4 0.2 0.5 7.00 .5 3.2 2.0 1.0 i10 8.7 0.2 0.5 7.00 .5 4.0 2.0 1.0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/assign.csd.xml0000644000000000000000000000311412262561504020136 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o assign.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ; Assign a value to the variable i1. i1 = 1234 ; Print the value of the i1 variable. print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gentanh.csd.xml0000644000000000000000000000543012262561504020301 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gentanh.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aswp linseg 0.01, p3*.5, .49, p3*.5, 0.01 ;index sweep function aindex poscil aswp, 110, 1 ;sound to waveshape atable tablei aindex, p4, 1, .5 ;waveshape index aenv linen 0.8, .01, p3, .02 ;amplitude envelope asig = (atable*aenv)*p5 ;impose envelope and scale asig dcblock2 asig ;get rid of DC outs asig, asig endin </CsInstruments> <CsScore> f 1 0 8192 10 1 ;sine wave f 2 0 8192 "tanh" -100 100 0 ;symmetrical transfer fuction f 3 0 8192 "tanh" -10 10 0 ;symmetrical f 4 0 8192 "tanh" 0 10 0 ;not symmetrical i1 0 3 2 1 i1 + 3 3 1 i1 + 3 4 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/rms.csd.xml0000644000000000000000000000661112262561504017460 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -m0 ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rms.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 ksmps = 128 nchnls = 1 ;Example by Andres Cabrera 2007 0dbfs = 1 FLpanel "rms", 400, 100, 50, 50 gkrmstext, gihrmstext FLtext "Rms", -100, 0, 0.1, 3, 110, 30, 60, 50 gkihp, gihandle FLtext "ihp", 0, 10, 0.05, 1, 100, 30, 220, 50 gkrmsslider, gihrmsslider FLslider "", -60, -0.5, -1, 5, -1, 380, 20, 10, 10 FLpanelEnd FLrun FLsetVal_i 5, gihandle ; Instrument #1. instr 1 a1 inch 1 label: kval rms a1, i(gkihp) ;measures rms of input channel 1 rireturn kval = dbamp(kval) ; convert to db full scale printk 0.5, kval FLsetVal 1, kval, gihrmsslider ;update the slider and text values FLsetVal 1, kval, gihrmstext knewihp changed gkihp ; reinit when ihp text has changed if (knewihp == 1) then reinit label ;needed because ihp is an i-rate parameter endif endin </CsInstruments> <CsScore> ; Play Instrument #1 for one minute i 1 0 60 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKRhodey.csd.xml0000644000000000000000000000375312262561504020477 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKRhodey.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;(FM) Modulator Index One kv5 = p7 ;ADSR 2 and 4 target asig STKRhodey cpspch(p4), 1, 2, kv1, 4, 10, 11, 100, 1, 3, 128, kv5 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 .5 7.00 75 0 0 i 1 + . 8.00 120 0 120 i 1 + 1 6.00 50 120 50 i 1 + 4 8.00 10 120 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dssiplay_mono.csd.xml0000644000000000000000000000550612262561504021541 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o DSSIplay_mono.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 gihandle dssiinit "caps.so", 19, 1 ; = mono phaser and gaout init 0 ; verbose about all ports instr 1 ; activate DSSI dssiactivate gihandle, 1 endin instr 2 ain1 diskin2 "beats.wav", 1,0,1 ; loop gaout = gaout+(ain1*.5) endin instr 3 dssictls gihandle, 0, .8, 1 ; range -1 to 1 dssictls gihandle, 1, .05, 1 ; rate 0 to 10 dssictls gihandle, 2, .8, 1 ; depth 0 to 1 dssictls gihandle, 3, 2, 1 ; spread 0 to 3.14 dssictls gihandle, 4, .7, 1 ; feedback 0 to 0.999 endin instr 4 aout1 dssiaudio gihandle, gaout ;get beats.wav, mono out outs aout1,aout1 gaout = 0 endin </CsInstruments> <CsScore> i 1 0 20 i 2 1 20 i 3 1 20 i 4 0 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/flashtxt.csd.xml0000644000000000000000000000335712262561504020520 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o flashtxt.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 instr 1 flashtxt 1, "Instr 1 live" ao oscil 4000, 440, 1 out ao endin </CsInstruments> <CsScore> ; Table 1: an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/buzz.csd.xml0000644000000000000000000000343312262561504017650 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o buzz.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 110 ifn = 1 knh line p4, p3, p5 asig buzz 1, kcps, knh, ifn outs asig, asig endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 3 20 20 i 1 + 3 3 3 i 1 + 3 10 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/bitshift.csd.xml0000644000000000000000000000603112262561504020467 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in ;-odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -o bitshift.wav -W --nosound ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 2 instr 1 ;bit shift right ival = p4>>p5 printf_i "%i>>%i = %i\n", 1, p4, p5, ival endin instr 2 ;bit shift left ival = p4<<p5 printf_i "%i<<%i = %i\n", 1, p4, p5, ival endin </CsInstruments> <CsScore> i 1 0 0.1 2 1 i 1 + . 3 1 i 1 + . 7 2 i 1 + . 16 1 i 1 + . 16 2 i 1 + . 16 3 i 2 5 0.1 1 1 i 2 + . 1 2 i 2 + . 1 3 i 2 + . 1 4 i 2 + . 2 1 i 2 + . 2 2 i 2 + . 2 3 i 2 + . 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluidCCi.csd.xml0000644000000000000000000000526412262561504020344 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidCCi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "19Trumpet.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 56 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 fluidCCi giengine, 1, 93, 127 ;full chorus & fluidCCi giengine, 1, 91, 127 ;full reverb! fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 7 asigl, asigr fluidOut giengine outs asigl*imvol, asigr*imvol endin </CsInstruments> <CsScore> i 1 0 5 60 100 ;play one note from score and... i 99 0 60 ;play virtual keyboard for 60 sec. e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/octmidi.csd.xml0000644000000000000000000000341212262561504020303 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages MIDI in -odac -iadc -d -M0 ;;;RT audio I/O with MIDI in ; For Non-realtime ouput leave only the line below: ; -o octmidi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; This example expects MIDI note inputs on channel 1 i1 octmidi print i1 endin </CsInstruments> <CsScore> ;Dummy f-table to give time for real-time MIDI events f 0 8000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/samphold.csd.xml0000644000000000000000000000401112262561504020456 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o samphold.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kx line -1, p3, 1 ; between -1 and +1 ktrig metro 1 ; triggers 1 time per second kval samphold kx, ktrig ; change value whenever ktrig = 1 printk2 kval ; will print every time kval changes asig diskin2 "flute.aiff", 1+kval, 0, 1 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 11 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/db.csd.xml0000644000000000000000000000333612262561504017245 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o db.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 idec = p4 iamp = db(idec) print iamp asig vco2 iamp, 110 ;sawtooth outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 50 i 1 + 1 > i 1 + 1 > i 1 + 1 85 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wgbowedbar.csd.xml0000644000000000000000000000377312262561504021010 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o wgbowedbar.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kp = p6 asig wgbowedbar p4, cpspch(p5), 1, kp, 0.995 outs asig, asig endin </CsInstruments> <CsScore> s i1 0 .5 .5 7.00 .1 ;short sound i1 + . .3 8.00 .1 i1 + . .5 9.00 .1 s i1 0 .5 .5 7.00 1 ;longer sound i1 + . .3 8.00 1 i1 + . .5 9.00 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLroller.csd.xml0000644000000000000000000000730612262561504020402 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLroller.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; A sine with oscillator with flroller controlled frequency sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Frequency Roller", 900, 400, 50, 50 ; Minimum value output by the roller imin = 200 ; Maximum value output by the roller imax = 5000 ; Increment with each pixel istep = 1 ; Logarithmic type roller selected iexp = -1 ; Roller graphic type (1=horizontal) itype = 1 ; Display handle (-1=not used) idisp = -1 ; Width of the roller in pixels iwidth = 300 ; Height of the roller in pixels iheight = 50 ; Distance of the left edge of the knob ; from the left edge of the panel ix = 300 ; Distance of the top edge of the knob ; from the top edge of the panel iy = 50 gkfreq, ihandle FLroller "Frequency", imin, imax, istep, iexp, itype, idisp, iwidth, iheight, ix, iy ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 iamp = 15000 ifn = 1 asig oscili iamp, gkfreq, ifn out asig endin </CsInstruments> <CsScore> ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/resonr.csd.xml0000644000000000000000000001322712262561504020170 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o resonr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Sean Costello */ ; Orchestra file for resonant filter sweep of a sawtooth-like waveform. ; The outputs of reson, resonr, and resonz are scaled by coefficients ; specified in the score, so that each filter can be heard on its own ; from the same instrument. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 idur = p3 ibegfreq = p4 ; beginning of sweep frequency iendfreq = p5 ; ending of sweep frequency ibw = p6 ; bandwidth of filters in Hz ifreq = p7 ; frequency of gbuzz that is to be filtered iamp = p8 ; amplitude to scale output by ires = p9 ; coefficient to scale amount of reson in output iresr = p10 ; coefficient to scale amount of resonr in output iresz = p11 ; coefficient to scale amount of resonz in output ; Frequency envelope for reson cutoff kfreq linseg ibegfreq, idur * .5, iendfreq, idur * .5, ibegfreq ; Amplitude envelope to prevent clicking kenv linseg 0, .1, iamp, idur - .2, iamp, .1, 0 ; Number of harmonics for gbuzz scaled to avoid aliasing iharms = (sr*.4)/ifreq asig gbuzz 1, ifreq, iharms, 1, .9, 1 ; "Sawtooth" waveform ain = kenv * asig ; output scaled by amp envelope ares reson ain, kfreq, ibw, 1 aresr resonr ain, kfreq, ibw, 1 aresz resonz ain, kfreq, ibw, 1 out ares * ires + aresr * iresr + aresz * iresz endin </CsInstruments> <CsScore> /* Written by Sean Costello */ f1 0 8192 9 1 1 .25 ; cosine table for gbuzz generator i1 0 10 1 3000 200 100 4000 1 0 0 ; reson output with bw = 200 i1 10 10 1 3000 200 100 4000 0 1 0 ; resonr output with bw = 200 i1 20 10 1 3000 200 100 4000 0 0 1 ; resonz output with bw = 200 i1 30 10 1 3000 50 200 8000 1 0 0 ; reson output with bw = 50 i1 40 10 1 3000 50 200 8000 0 1 0 ; resonr output with bw = 50 i1 50 10 1 3000 50 200 8000 0 0 1 ; resonz output with bw = 50 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/oscil1i.csd.xml0000644000000000000000000000551012262561504020217 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil1i.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipanfn = p4 asig vco2 .3, 220 kpan oscil1i 0, 1, p3, ipanfn ;create panning & kleft = sqrt(kpan) ;start right away kright = sqrt(1-kpan) outs kleft*asig, kright*asig endin </CsInstruments> <CsScore> f 1 0 3 -7 .5 3 .5 ;remain in center (.5 CONSTANT) f 2 0 129 7 1 129 0 ;left-->right f 3 0 129 7 .5 32 1 64 0 33 .5 ;center-->left-->right-->center i 1 0 2 1 ;use table 1 i 1 3 2 2 ;use table 2 i 1 6 2 3 ;use table 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vadd.csd.xml0000644000000000000000000000506512262561504017577 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vadd ifn1, ival, ielements, idstoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 5 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 8 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1 10 12 i2 1.6 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vpow_i.csd.xml0000644000000000000000000000507012262561504020160 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vpow_i ifn1, ival, ielements, idstoffset endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/plate.csd.xml0000644000000000000000000000365412262561504017770 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ;-odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: -o plate.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Note: this example is heavy on CPU sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ab diskin2 "beats.wav", 1 al,ar platerev 1, 2, p4, 0.73, 1.0, 5.0, 0.001, ab,ab outs al*.25,ar*.25 endin </CsInstruments> <CsScore> f1 0 8 -2 0.3 0.3875 0.39274 0.32 0.85714 0.78548 f2 0 8 -2 0.2 0.666667 1.57097 0.24 0.75 0.78548 i1 0 4 1 i1 + 4 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKTubeBell.csd.xml0000644000000000000000000000375012262561504020740 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKTubeBell.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv2 line p6, p3, p7 ;Crossfade of Outputs kv1 = p5 ;(FM) Modulator Index One asig STKTubeBell cpspch(p4), 1, 2, kv1, 4, kv2, 11, 10, 1, 70, 128,50 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 7.05 0 100 100 i 1 + 4 9.00 127 127 30 i 1 + 1 10.00 127 12 30 i 1 + 3 6.08 127 1 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/clip.csd.xml0000644000000000000000000000440012262561504017600 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o clip.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; white noise arnd rand 1 ; full amlitude ; Clip the noisy waveform's amplitude to 0.5 a1 clip arnd, 2, 0.5 outs a1, a1 endin instr 2 ; white noise arnd rand 1 ; full amlitude ; Clip the noisy waveform's amplitude to 0.1 a1 clip arnd, 2, 0.1 outs a1, a1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 1 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vdelayxwq.csd.xml0000644000000000000000000000535112262561504020703 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vdelayxwq.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 ims = .5 ;maximum delay time in seconds iws = 1024 ;best quality adl = .5 ;delay time aout1 diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav aout2 diskin2 "fox.wav", 1, 0, 1 ;loop fox.wav aout3 diskin2 "Church.wav", 1, 0, 1 ;loop Church.wav aout4 diskin2 "flute.aiff", 1, 0, 1 ;loop flute.aiff a2 poscil3 .2, .1, 1 ;make an LFO, 1 cycle per 2 seconds adl = a2 + ims/2 ;offset the LFO so that it is positive aout1, aout2, aout3, aout4 vdelayxwq aout1, aout2, aout3, aout4, adl, ims, iws ;use the LFO to control delay time outq aout1, aout2, aout3, aout4 endin </CsInstruments> <CsScore> f1 0 8192 10 1 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fmmetal.csd.xml0000644000000000000000000000445012262561504020303 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmmetal.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 440 kvdepth = 0 kvrate = 0 ifn1 = 1 ifn2 = 2 ifn3 = 2 ifn4 = 1 ivfn = 1 kc2 = p5 kc1 line p4, p3, 1 asig fmmetal .5, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn outs asig, asig endin </CsInstruments> <CsScore> ; sine wave. f 1 0 32768 10 1 ; the "twopeaks.aiff" audio file. f 2 0 256 1 "twopeaks.aiff" 0 0 0 i 1 0 4 6 5 i 1 5 4 .2 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vpow.csd.xml0000644000000000000000000000507112262561504017651 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vpow ifn1, ival, ielements, idstoffset, 1 endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/display.csd.xml0000644000000000000000000000353712262561504020330 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o display.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 110 ifn = 1 knh line p4, p3, p5 asig buzz 1, kcps, knh, ifn outs asig, asig display asig, .1, 3 endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 3 20 20 i 1 + 3 3 3 i 1 + 3 150 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpstuni.csd.xml0000644000000000000000000000477212262561504020352 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cpstuni.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Table #1, a normal 12-tone equal temperament scale. ; numgrades = 12 (twelve tones) ; interval = 2 (one octave) ; basefreq = 261.659 (Middle C) ; basekeymidi = 60 (Middle C) gitemp ftgen 1, 0, 64, -2, 12, 2, 261.659, 60, 1.00, \ 1.059, 1.122, 1.189, 1.260, 1.335, 1.414, \ 1.498, 1.588, 1.682, 1.782, 1.888, 2.000 ; Instrument #1. instr 1 ; Use Table #1. ifn = 1 ; If the base key (note #60) is C, then 9 notes ; above it (note #60 + 9 = note #69) should be A. index = 69 i1 cpstuni index, ifn print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cauchyi.csd.xml0000644000000000000000000000345112262561504020303 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o exprand.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 klambda cauchyi 100, 1, 3 printk2 klambda ; look aout oscili 0.8, 440+klambda, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/csbeats.csd.xml0000644000000000000000000001337312262561504020306 0ustar rootroot <CsoundSynthesizer> <CsOptions> </CsOptions> <CsInstruments> sr = 44100 nchnls = 2 gi1 ftgen 1, 0, 4096, 10, 1 gi2 ftgen 2, 0, 4096, 7, -1, 4096, 1 ; sawtooth gi3 ftgen 3, 0, 4096, 7, 0, 1024, 1, 2048, -1, 1024, 0 ;triangle instr 101,102,103 iamp = ampdbfs(p5) a1 oscil iamp, p4, p1-100 kenv expseg 1, p3, .01 a1 = a1 * kenv outs a1, a1 endin </CsInstruments> <CsScore> ; by Brian Baughn 3-14-05 ; bbaughn@berklee.net beats = 100 permeasure = 4 i101 m1 b1 q mp D3 i101 F3 i101 D3 i101 m2 b1 D3 i101 b3 D3 i101 m3 b1 D3 i101 F3 i101 D3 i101 m4 b1 D3 i101 b3 D3 i101 m5 b1 D3 i101 b4 G5 i101 m6 b1 E5 i101 b2 F5 i101 b3 e Eb5 i101 b3.5 e i101 b4 q i101 m7 b1 e D5 i101 i101 q i101 e Db5 i101 i101 q i101 m8 b1 q D5 i101 E5 i101 D5 i102 m1 b2 q D4 i102 b4 E4 i102 b4 Bb3 i102 m2 b2 F4 i102 b2 B3 i102 b4 C#4 i102 b4 Bb3 i102 m3 b2 q D4 i102 b4 E4 i102 b4 Bb3 i102 m4 b2 F4 i102 b2 B3 i102 b4 C#4 i102 b4 Bb3 i103 m5 b2 e F6 i103 b2 e A5 i103 b2.5 e D6 i103 b3 e F6 i103 b3 e A5 i103 b4 e E6 i103 m6 b1 q C#6 i103 q D6 i103 e C6 i103 i103 q i103 m7 b1 e B5 i103 i103 q i103 e Bb5 i103 i103 q i103 m8 b1 e F5 i103 b1 e A5 i103 b1.5 e D6 i103 b2 e Bb5 i103 b2.5 e D6 i103 b3 q F5 i103 b3 A5 end </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/clfilt_lowpass.csd.xml0000644000000000000000000000372612262561504021710 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o clfilt_lowpass.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; white noise asig rand 0.5 outs asig, asig endin instr 2 ; filtered noise asig rand 0.9 ; Lowpass filter signal asig with a ; 10-pole Butterworth at 500 Hz. a1 clfilt asig, 500, 0, 10 outs a1, a1 endin </CsInstruments> <CsScore> i 1 0 2 i 2 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/massign.csd.xml0000644000000000000000000000763612262561504020330 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o massign.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "19Trumpet.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 56 massign 0,0 ;disable triggering of all instruments on all channels, but massign 12,10 ;assign instr. 10 to midi channel 12 massign 3,30 ;assign instr. 30 to midi channel 3 instr 10 ; soundfont only on midi channel 12 mididefault 60, p3 midinoteonkey p4, p5 ; in midi notes ikey init p4 ivel init p5 fluidNote giengine, 1, ikey, ivel endin instr 30 ; FM-oscilator only on midi channel 3 mididefault 60, p3 midinoteoncps p4, p5 ; in Hertz icps init p4 iamp init p5 iamp = iamp/127 kenv madsr 0.5, 0, 1, 0.5 asig foscil iamp*kenv, icps, 1, 1.414, 2, 1 outs asig, asig endin instr 99 ; output sound from fluidengine imvol init 7 aL, aR fluidOut giengine outs aL*imvol, aR*imvol endin </CsInstruments> <CsScore> ; sine f 1 0 16384 10 1 i 10 0 2 60 100 ;one note on the trumpet in midi and... i 30 2 2 220 80 ;one FM note in Hz i 99 0 60 ;stay active for 60 sec. e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ptablei.csd.xml0000644000000000000000000000673612262561504020307 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tablei.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 ;generate new values every time the instr is played instr 1 ifn = p4 isize = p5 ithresh = 0.5 itemp ftgen ifn, 0, isize, 21, 2 iwrite_value = 0 i_index = 0 loop_start: iread_value ptablei i_index, ifn if iread_value > ithresh then iwrite_value = 1 else iwrite_value = -1 endif ptableiw iwrite_value, i_index, ifn loop_lt i_index, 1, isize, loop_start turnoff endin instr 2 ifn = p4 isize = ftlen(ifn) prints "Index\tValue\n" i_index = 0 loop_start: ivalue tablei i_index, ifn prints "%d:\t%f\n", i_index, ivalue loop_lt i_index, 1, isize, loop_start ;read table 1 with our index aout oscili .5, 100, ifn ;use table to play the polypulse outs aout, aout endin </CsInstruments> <CsScore> i 1 0 1 100 16 i 2 0 2 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/readk3.csd.xml0000644000000000000000000000726512262561504020036 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o readk3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes three control signals to a file kfreq randh 100, 1, 2, 1, 500 ;generates one random number between 400 and 600 per second kdb randh 12, 1, 2, 1, -12 ;amplitudes in dB between -24 and 0 kpan randh .5, 1, 2, 1, .5 ;panning between 0 and 1 dumpk3 kfreq, kdb, kpan, "dumpk3.txt", 8, 1 ;writes the control signals prints "WRITING:\n" printks "kfreq = %f, kdb = %f, kpan = %f\n", 1, kfreq, kdb, kpan ;prints them endin instr 2 ;reads the file written by instr 1 kf,kdb,kp readk3 "dumpk3.txt", 8, 1 prints "READING:\n" printks "kfreq = %f, kdb = %f, kpan = %f\n", 1, kf, kdb, kp ;prints again kdb lineto kdb, .1 ;smoothing amp transition kp lineto kp, .1 ;smoothing pan transition aout poscil ampdb(kdb), kf, giSine aL, aR pan2 aout, kp outs aL, aR endin </CsInstruments> <CsScore> i 1 0 5 i 2 5 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cps2pch_ftable.csd.xml0000644000000000000000000000403112262561504021530 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cps2pch_ftable.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ipch = 8.02 ; Use Table #1, a table of frequency multipliers. icps cps2pch ipch, -1 print icps endin </CsInstruments> <CsScore> ; Table #1: a table of frequency multipliers. ; Creates a 10-note scale of unequal divisions. f 1 0 16 -2 1 1.1 1.2 1.3 1.4 1.6 1.7 1.8 1.9 ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKBlowHole.csd.xml0000644000000000000000000000356012262561504020754 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKBlowHole.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = p4 kv1 = p7 ;stiffness of reed kv3 line p5, p3, p6 ;state of tonehole asig STKBlowHole cpspch(ipch), 1, 2, kv1, 4, 100, 11, kv3, 1, 10, 128, 100 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 4 10.00 20 127 100 i 1 + 7 6.09 120 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/spat3d_UHJ.csd.xml0000644000000000000000000001457212262561504020570 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o spat3d_UHJ.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Istvan Varga */ sr = 48000 kr = 750 ksmps = 64 nchnls = 2 itmp ftgen 1, 0, 64, -2, \ /* depth1, depth2, max delay, IR length, idist, seed */ \ 3, 48, -1, 0.01, 0.25, 123, \ 1, 21.982, 0.05, 0.87, 4000.0, 0.6, 0.7, 2, /* ceil */ \ 1, 1.753, 0.05, 0.87, 3500.0, 0.5, 0.7, 2, /* floor */ \ 1, 15.220, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* front */ \ 1, 9.317, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* back */ \ 1, 17.545, 0.05, 0.87, 5000.0, 0.8, 0.7, 2, /* right */ \ 1, 12.156, 0.05, 0.87, 5000.0, 0.8, 0.7, 2 /* left */ instr 1 p3 = p3 + 1.0 kazim line 0.0, 4.0, 360.0 ; azimuth kelev line 40, p3 - 1.0, -20 ; elevation kdist = 2.0 ; distance ; convert coordinates kX = kdist * cos(kelev * 0.01745329) * sin(kazim * 0.01745329) kY = kdist * cos(kelev * 0.01745329) * cos(kazim * 0.01745329) kZ = kdist * sin(kelev * 0.01745329) ; source signal a1 phasor 160.0 a2 delay1 a1 a1 = a1 - a2 kffrq1 port 200.0, 0.8, 12000.0 affrq upsamp kffrq1 affrq pareq affrq, 5.0, 0.0, 1.0, 2 kffrq downsamp affrq aenv4 phasor 3.0 aenv4 limit 2.0 - aenv4 * 8.0, 0.0, 1.0 a1 butterbp a1 * aenv4, kffrq, 160.0 aenv linseg 1.0, p3 - 1.0, 1.0, 0.04, 0.0, 1.0, 0.0 a_ = 4000000 * a1 * aenv + 0.00000001 ; spatialize a_W, a_X, a_Y, a_Z spat3d a_, kX, kY, kZ, 1.0, 1, 2, 2.0, 2 ; convert to UHJ format (stereo) aWre, aWim hilbert a_W aXre, aXim hilbert a_X aYre, aYim hilbert a_Y aWXre = 0.0928*aXre + 0.4699*aWre aWXim = 0.2550*aXim - 0.1710*aWim aL = aWXre + aWXim + 0.3277*aYre aR = aWXre - aWXim - 0.3277*aYre outs aL, aR endin </CsInstruments> <CsScore> /* Written by Istvan Varga */ t 0 60 i 1 0.0 8.0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/oscilikts.csd.xml0000644000000000000000000000472612262561504020670 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o oscilikts.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1: oscilikts example. instr 1 ; Frequency envelope. kfrq expon 400, p3, 1200 ; Phase. kphs line 0.1, p3, 0.9 ; Sync 1 atmp1 phasor 100 ; Sync 2 atmp2 phasor 150 async diff 1 - (atmp1 + atmp2) a1 oscilikts 14000, kfrq, 1, async, 0 a2 oscilikts 14000, kfrq, 1, async, -kphs out a1 - a2 endin </CsInstruments> <CsScore> ; Table #1: Sawtooth wave f 1 0 3 -2 1 0 -1 ; Play Instrument #1 for four seconds. i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fmbell.csd.xml0000644000000000000000000000503012262561504020112 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmbell.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp = p4 kfreq = 880 kc1 = p5 kc2 = p6 kvdepth = 0.005 kvrate = 6 asig fmbell kamp, kfreq, kc1, kc2, kvdepth, kvrate outs asig, asig endin instr 2 kamp = p4 kfreq = 880 kc1 = p5 kc2 = p6 kvdepth = 0.005 kvrate = 6 asig fmbell kamp, kfreq, kc1, kc2, kvdepth, kvrate, 1, 1, 1, 1, 1, p7 outs asig, asig endin </CsInstruments> <CsScore> ; sine wave. f 1 0 32768 10 1 i 1 0 3 .2 5 5 i 1 + 4 .3 .5 1 s i 2 0 12 .2 5 5 16 i 2 + 12 .3 .5 1 12 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tambourine.csd.xml0000644000000000000000000000316412262561504021024 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tambourine.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idamp = p4 asig tambourine .8, 0.01, 30, idamp outs asig, asig endin </CsInstruments> <CsScore> i 1 0 .2 0 i 1 + .2 > i 1 + 1 .7 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/resony.csd.xml0000644000000000000000000000533612262561504020201 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o resonx.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; unfiltered noise kenv linseg 0, p3*.5, 1, p3*.5, 0 ;envelope asig rand 0.7 ;white noise outs asig*kenv, asig*kenv endin instr 2 ; filtered noise ksep = p4 ;vary seperation of center frequency of filters in octaves kenv linseg 0, p3*.5, 1, p3*.5, 0 ;envelope asig rand 0.7 kbf line 300, p3, 2000 ;vary base frequency afilt resony asig, kbf, 300, 4, ksep asig balance afilt, asig outs asig*kenv, asig*kenv endin </CsInstruments> <CsScore> i 1 0 2 i 2 3 2 1 i 2 6 2 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tablekt.csd.xml0000644000000000000000000000536012262561504020305 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tablekt.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 gift1 ftgen 1, 0, 1024, 10, 1 ;sine wave gift2 ftgen 2, 0, 1024, 10, 1, 1, 1, 1, 1, 1, 1, 1, 1 ;pulse instr 1 andx phasor 400 ;phasor for reading the index kfn init 1 ;initialize the choice of the function table kmetro init 1 ;initialize the frequency of the metro knewft metro kmetro ;make a new choice for selecting the function table once a second if knewft == 1 then kfn = (kfn == 1 ? 2 : 1) ;switch between 1 and 2 kmetro random .5, 2 ;create new metro frequency printk2 kfn endif ares tablekt andx, kfn, 1 outs ares, ares endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hvs2.csd.xml0000644000000000000000000001153212262561504017537 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O </CsOptions> <CsInstruments> sr=44100 ksmps=100 nchnls=2 0dbfs = 1 ginumLinesX init 4 ginumLinesY init 4 ginumParms init 3 giOutTab ftgen 5,0,8, -2, 0 giPosTab ftgen 6,0,32, -2, 3,2,1,0,4,5,6,7,8,9,10, 11, 15, 14, 13, 12 giSnapTab ftgen 8,0,64, -2, 1,1,1, 2,0,0, 3,2,0, 2,2,2, 5,2,1, 2,3,4, 6,1,7, 0,0,0, \ 1,3,5, 3,4,4, 1,5,8, 1,1,5, 4,3,2, 3,4,5, 7,6,5, 7,8,9 tb0_init giOutTab FLpanel "Prova HVS2",600,400,10,100,0 gk1, gk2, ih1, ih2 FLjoy "HVS controller XY", 0, 1, 1, 0, 0, 0, -1, -1, 300, 300, 0, 50 ; *ihandle, *numlinesX, *numlinesY, *iwidth, *iheight, *ix, *iy,*image; gihandle FLhvsBox ginumLinesX, ginumLinesY, 300, 300, 300, 50, 1 FLpanel_end FLrun instr 1 ; Smooth control signals to avoid clicks kx portk gk1, 0.02 ky portk gk2, 0.02 ; kx, ky, inumParms, inumlinesX, inumlinesY, iOutTab, iPosTab, iSnapTab [, iConfigTab] hvs2 kx, ky, ginumParms, ginumLinesX, ginumLinesY, giOutTab, giPosTab, giSnapTab ;, iConfigTab ; *kx, *ky, *ihandle; FLhvsBoxSetValue gk1, gk2, gihandle k0 init 0 k1 init 1 k2 init 2 printk2 tb0(k0) printk2 tb0(k1), 10 printk2 tb0(k2), 20 kris init 0.003 kdur init 0.02 kdec init 0.007 ; Make parameters of synthesis depend on the table values produced by hvs ares1 fof 0.2, tb0(k0)*100 + 50, tb0(k1)*100 + 200, 0, tb0(k2) * 10 + 50, 0.003, 0.02, 0.007, 20, \ 1, 2, p3 ares2 fof 0.2, tb0(k1)*100 + 50, tb0(k2)*100 + 200, 0, tb0(k0) * 10 + 50, 0.003, 0.02, 0.007, 20, \ 1, 2, p3 outs ares1, ares2 endin </CsInstruments> <CsScore> f 1 0 1024 10 1 ;Sine wave f 2 0 1024 19 0.5 0.5 270 0.5 ;Grain envelope table f0 3600 i1 0 3600 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/insremotM.csd.xml0000644000000000000000000000300512262561504020626 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o insremotM.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> nchnls = 1 insremot "192.168.1.100", "192.168.1.101", 1 instr 1 aq barmodel 1, 1, p4, 0.001, 0.23, 5, p5, p6, p7 out aq endin </CsInstruments> <CsScore> i1 0 0.5 3 0.2 500 0.05 i1 0.5 0.5 -3 0.3 1000 0.05 i1 1.0 0.5 -3 0.4 1000 0.1 i1 1.5 4.0 -3 0.5 800 0.05 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/inletk.csd.xml0000644000000000000000000000426712262561504020152 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o inletk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 connect "bend", "bendout", "guitar", "bendin" instr bend kbend line p4, p3, p5 outletk "bendout", kbend endin instr guitar kbend inletk "bendin" kpch pow 2, kbend/12 printk2 kpch asig oscili .4, 440*kpch, 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 1024 10 1 i"guitar" 0 5 8.00 i"bend" 3 .2 -12 12 i"bend" 4 .1 -17 40 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lowresx.csd.xml0000644000000000000000000000461512262561504020364 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lowresx.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kres = p4 inumlayer = 4 kenv linseg 0, p3*.1, 1, p3*.8, 1, p3*.1, 0 ;envelope asig vco .3 * kenv, 220, 1 ;sawtooth kcut line 30, p3, 1000 ;note: kcut is not in Hz alx lowresx asig, kcut, kres, inumlayer ;note: kres is not in dB aout balance alx, asig ;avoid very loud sounds outs aout, aout endin </CsInstruments> <CsScore> ;sine wave f 1 0 16384 10 1 i 1 0 5 1 i 1 + 5 3 i 1 + 5 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ptable.csd.xml0000644000000000000000000000457112262561504020131 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o table.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Vary our index linearly from 0 to 1. kndx line 0, p3, 1 ; Read Table #1 with our index. ifn = 1 ixmode = 1 kfreq ptable kndx, ifn, ixmode ; Generate a sine waveform, use our table values ; to vary its frequency. a1 oscil 20000, kfreq, 2 out a1 endin </CsInstruments> <CsScore> ; Table #1, a line from 200 to 2,000. f 1 0 1025 -7 200 1024 2000 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/kill.csd.xml0000644000000000000000000000416512262561504017614 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o kill.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 giHandleinstance 1, 0, 4 ; Instrument #1. instr 1 k1 expon 440, p3/10,880 ; begin gliss and continue if k1 < sr/2 kgoto contin ; until Nyquist detected kill giHandle ; then quit contin: a1 oscili 10000, k1 out a1 endin </CsInstruments> <CsScore> </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/jitter.csd.xml0000644000000000000000000000404612262561504020160 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o jitter.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp init p4 kcpsmin init 4 kcpsmax init 8 kj2 jitter kamp, kcpsmin, kcpsmax aout pluck 1, 200+kj2, 1000, 0, 1 aout dcblock aout ;remove DC outs aout, aout endin </CsInstruments> <CsScore> i 1 0 15 2 ;a bit jitter i 1 8 15 10 ;some more i 1 16 15 20 ;lots more e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen03.csd.xml0000644000000000000000000001153412262561504017573 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen03.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;example by Russell Pinkston - Univ. of Texas (but slightly modified) gisine ftgen 1, 0, 16384, 10, 1 ;sine wave instr 1 ihertz = cpspch(p4) ipkamp = p5 iwsfn = p6 ;waveshaping function inmfn = p7 ;normalization function aenv linen 1, .01, p3, .1 ;overall amp envelope actrl linen 1, 2, p3, 2 ;waveshaping index control aindex poscil actrl/2, ihertz, gisine ;sine wave to be distorted asignal tablei .5+aindex, iwsfn, 1 ;waveshaping anormal tablei actrl, inmfn,1 ;amplitude normalization asig = asignal*anormal*ipkamp*aenv asig dcblock2 asig ;get rid of possible DC outs asig, asig endin </CsInstruments> <CsScore> ; first four notes are specific Chebyshev polynomials using gen03. The values were obtained from Dodge page 147 f4 0 513 3 -1 1 0 1 ; First-order Chebyshev: x f5 0 257 4 4 1 ; Normalizing function for fn4 f6 0 513 3 -1 1 -1 0 2 ; Second-order Chebyshev: 2x2 - 1 f7 0 257 4 6 1 ; Normalizing function for fn6 f8 0 513 3 -1 1 0 -3 0 4 ; Third-order Chebyshev: 4x3 - 3x f9 0 257 4 8 1 ; Normalizing function for fn8 f10 0 513 3 -1 10 0 -7 0 56 0 -112 0 64 ; Seventh-order Chebyshev: 64x7 - 112x5 + 56x3 - 7x f11 0 257 4 10 1 ; Normalizing function for fn10 f12 0 513 3 -1 1 5 4 3 2 2 1 ; a 4th order polynomial function over the x-interval -1 to 1 f13 0 257 4 12 1 ; Normalizing function for fn12 ; five notes with same fundamental, different waveshape & normalizing functions ; pch amp wsfn nmfn i1 0 3 8.00 .7 4 5 i1 + . . . 6 7 i1 + . . . 8 9 i1 + . . . 10 11 i1 + . . . 12 13 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/poisson.csd.xml0000644000000000000000000000455012262561504020351 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o poisson.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 ksmps = 441 ;ksmps set deliberately high to have few k-periods per second nchnls = 1 ; Instrument #1. instr 1 ; Generates a random number in a poisson distribution. ; klambda = 1 i1 poisson 1 print i1 endin instr 2 kres poisson p4 printk (ksmps/sr),kres ;prints every k-period endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 i 2 1 0.2 0.5 i 2 2 0.2 4 ;average 4 events per k-period i 2 3 0.2 20 ;average 20 events per k-period e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zakinit.csd.xml0000644000000000000000000000761312262561504020333 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zakinit.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 4410 nchnls = 1 ; Initialize the ZAK space. ; Create 3 a-rate variables and 5 k-rate variables. zakinit 2, 3 instr 1 ;a simple waveform. ; Generate a simple sine waveform. asin oscil 20000, 440, 1 ; Send the sine waveform to za variable #1. zaw asin, 1 endin instr 2 ;generates audio output. ; Read za variable #1. a1 zar 1 ; Generate audio output. out a1 ; Clear the za variables, get them ready for ; another pass. zacl 0, 3 endin instr 3 ;increments k-type channels k0 zkr 0 k1 zkr 1 k2 zkr 2 zkw k0+1, 0 zkw k1+5, 1 zkw k2+10, 2 endin instr 4 ;displays values from k-type channels k0 zkr 0 k1 zkr 1 k2 zkr 2 ; The total count for k0 is 30, since there are 10 ; control blocks per second and intruments 3 and 4 ; are on for 3 seconds. printf "k0 = %i\n",k0, k0 printf "k1 = %i\n",k1, k1 printf "k2 = %i\n",k2, k2 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 i 1 0 1 i 2 0 1 i 3 0 3 i 4 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/follow.csd.xml0000644000000000000000000000476112262561504020165 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o follow.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig soundin "beats.wav" outs asig, asig endin instr 2 ;envelope follower as soundin "beats.wav" as = as*.7 ;reduce volume a bit at tone as, 500 ;smooth estimated envelope af follow at, p4 asin poscil3 .5, 440, 1 ; "beats.wav" provides amplitude for poscil asig balance asin, af outs asig, asig endin </CsInstruments> <CsScore> ;sine wave. f 1 0 32768 10 1 i 1 0 2 i 2 2 2 0.001 ;follow quickly i 2 5 3 0.2 ;follow slowly e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vco2.csd.xml0000644000000000000000000001075112262561504017530 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vco2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 10 nchnls = 1 ; user defined waveform -1: trapezoid wave with default parameters (can be ; accessed at ftables starting from 10000) itmp ftgen 1, 0, 16384, 7, 0, 2048, 1, 4096, 1, 4096, -1, 4096, -1, 2048, 0 ift vco2init -1, 10000, 0, 0, 0, 1 ; user defined waveform -2: fixed table size (4096), number of partials ; multiplier is 1.02 (~238 tables) itmp ftgen 2, 0, 16384, 7, 1, 4095, 1, 1, -1, 4095, -1, 1, 0, 8192, 0 ift vco2init -2, ift, 1.02, 4096, 4096, 2 instr 1 kcps expon p4, p3, p5 ; instr 1: basic vco2 example a1 vco2 12000, kcps ; (sawtooth wave with default out a1 ; parameters) endin instr 2 kcps expon p4, p3, p5 ; instr 2: kpw linseg 0.1, p3/2, 0.9, p3/2, 0.1 ; PWM example a1 vco2 10000, kcps, 2, kpw out a1 endin instr 3 kcps expon p4, p3, p5 ; instr 3: vco2 with user a1 vco2 14000, kcps, 14 ; defined waveform (-1) aenv linseg 1, p3 - 0.1, 1, 0.1, 0 ; de-click envelope out a1 * aenv endin instr 4 kcps expon p4, p3, p5 ; instr 4: vco2ft example, kfn vco2ft kcps, -2, 0.25 ; with user defined waveform a1 oscilikt 12000, kcps, kfn ; (-2), and sr/4 bandwidth out a1 endin </CsInstruments> <CsScore> i 1 0 3 20 2000 i 2 4 2 200 400 i 3 7 3 400 20 i 4 11 2 100 200 f 0 14 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gausstrig-2.csd.xml0000644000000000000000000000400512262561504021021 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfrq0 oscil 8, 0.25, 1 ktrig metro 1 kfrq samphold kfrq0+8, ktrig seed 20120125 aimp gausstrig 0.5, kfrq, 0.5, 1 aenv filter2 aimp, 1, 1, 0.993, 0.993 anoi fractalnoise 0.2, 1.7 al = anoi*aenv ar delay al, 0.02 outs al, ar endin </CsInstruments> <CsScore> f1 0 8192 10 1 i1 0 16 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/system.csd.xml0000644000000000000000000000507412262561504020205 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ; -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o system.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 ; Waits for command to execute before continuing ires system_i 1,{{ ps date cd ~/Desktop pwd ls -l whois csounds.com }} print ires turnoff endin instr 2 ; Runs command in a separate thread ires system_i 1,{{ ps date cd ~/Desktop pwd ls -l whois csounds.com }}, 1 print ires turnoff endin </CsInstruments> <CsScore> ; Play Instrument #1 for thirty seconds. i 1 0 1 i 2 5 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/readscore.csd.xml0000644000000000000000000000213212262561504020620 0ustar rootroot <CsoundSynthesizer> <CsOptions> -d -o dac </CsOptions> <CsInstruments> instr 1 a1 flooper2 1000,p4,2,5,0.1,1 out a1 endin instr 2 ires readscore {{ {12 COUNT i1 $COUNT 1 [1 + $COUNT/12] } }} endin </CsInstruments> <CsScore> f0 12 f 1 0 0 1 "fox.wav" 0 0 1 i2 0 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/diskin.csd.xml0000644000000000000000000000310612262561504020134 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o diskin.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktrans linseg 1, 5, 2, 10, -2 a1 diskin "beats.wav", ktrans, 0, 1, 0, 32 outs a1, a1 endin </CsInstruments> <CsScore> i 1 0 15 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/until.csd.xml0000644000000000000000000000416012262561504020007 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ifthen.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 lab99: if p4<0 goto lab100 p4 = p4-1 print p4 goto lab99 lab100: endin instr 2 until p4<0 do p4 = p4-1 print p4 od endin </CsInstruments> <CsScore> i 1 1 1 4 i 2 2 1 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/delay.csd.xml0000644000000000000000000000476112262561504017761 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o delay.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 adel init 0 ilev = p4 ;level of direct sound idelay = p5 *.001 ;Delay in ms ifd = p6 ;feedback ain diskin2 "fox.wav", 1, 1 adel delay ain + (adel*ifd), idelay;ifd = amount of feedback asig moogvcf adel, 1500, .6, 1 ;color feedback outs asig*ilev, ain endin </CsInstruments> <CsScore> ;Delay is in ms i 1 0 15 2 200 .95 ;with feedback i 1 4 5 2 20 .95 i 1 + 3 2 5 .95 i 1 + 3 3 5 0 ;no feedback e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/partikkelsync.csd.xml0000644000000000000000000003323512262561504021544 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out -odac ;;;RT audio ; For Non-realtime ouput leave only the line below: ; -o partikkel_softsync.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 20 nchnls = 2 ; Example by Oeyvind Brandtsegg 2007, revised 2008 giSine ftgen 0, 0, 65537, 10, 1 giCosine ftgen 0, 0, 8193, 9, 1, 1, 90 giSigmoRise ftgen 0, 0, 8193, 19, 0.5, 1, 270, 1 ; rising sigmoid giSigmoFall ftgen 0, 0, 8193, 19, 0.5, 1, 90, 1 ; falling sigmoid ; ************************************************* ; example of soft synchronization of two partikkel instances ; ************************************************* instr 1 /*score parameters*/ igrainrate = p4 ; grain rate igrainsize = p5 ; grain size in ms igrainFreq = p6 ; fundamental frequency of source waveform iosc2Dev = p7 ; partikkel instance 2 grain rate deviation factor iMaxSync = p8 ; max soft sync amount (increasing to this value during length of note) /*overall envelope*/ iattack = 0.001 idecay = 0.2 isustain = 0.7 irelease = 0.2 amp linsegr 0, iattack, 1, idecay, isustain, 1, isustain, irelease, 0 kgrainfreq = igrainrate ; grains per second kdistribution = 0 ; periodic grain distribution idisttab = -1 ; (default) flat distribution used ; for grain distribution async = 0 ; no sync input kenv2amt = 0 ; no secondary enveloping ienv2tab = -1 ; default secondary envelope (flat) ienv_attack = giSigmoRise ; default attack envelope (flat) ienv_decay = giSigmoFall ; default decay envelope (flat) ksustain_amount = 0.3 ; time (in fraction of grain dur) at ; sustain level for each grain ka_d_ratio = 0.2 ; balance between attack and decay time kduration = igrainsize ; set grain duration in ms kamp = 0.2*0dbfs ; amp igainmasks = -1 ; (default) no gain masking kwavfreq = igrainFreq ; fundamental frequency of source waveform ksweepshape = 0 ; shape of frequency sweep (0=no sweep) iwavfreqstarttab = -1 ; default frequency sweep start ; (value in table = 1, which give ; no frequency modification) iwavfreqendtab = -1 ; default frequency sweep end ; (value in table = 1, which give ; no frequency modification) awavfm = 0 ; no FM input ifmamptab = -1 ; default FM scaling (=1) kfmenv = -1 ; default FM envelope (flat) icosine = giCosine ; cosine ftable kTrainCps = kgrainfreq ; set trainlet cps equal to grain ; rate for single-cycle trainlet in ; each grain knumpartials = 3 ; number of partials in trainlet kchroma = 1 ; balance of partials in trainlet ichannelmasks = -1 ; (default) no channel masking, ; all grains to output 1 krandommask = 0 ; no random grain masking kwaveform1 = giSine ; source waveforms kwaveform2 = giSine ; kwaveform3 = giSine ; kwaveform4 = giSine ; iwaveamptab = -1 ; mix of 4 source waveforms and ; trainlets (set to default) asamplepos1 = 0 ; phase offset for reading source waveform asamplepos2 = 0 ; asamplepos3 = 0 ; asamplepos4 = 0 ; kwavekey1 = 1 ; original key for source waveform kwavekey2 = 1 ; kwavekey3 = 1 ; kwavekey4 = 1 ; imax_grains = 100 ; max grains per k period iopcode_id = 1 ; id of opcode, linking partikkel ; to partikkelsync a1 partikkel kgrainfreq, kdistribution, idisttab, async, kenv2amt, \ ienv2tab,ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, \ kduration, kamp, igainmasks, kwavfreq, ksweepshape, \ iwavfreqstarttab, iwavfreqendtab, awavfm, ifmamptab, kfmenv, \ icosine, kTrainCps, knumpartials, kchroma, ichannelmasks, \ krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, asamplepos4, \ kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains, iopcode_id async1 partikkelsync iopcode_id ; clock pulse output of the ; partikkel instance above ksyncGravity line 0, p3, iMaxSync ; strength of synchronization aphase2 init 0 asyncPolarity limit (int(aphase2*2)*2)-1, -1, 1 ; use the phase of partikkelsync instance 2 to find sync ; polarity for partikkel instance 2. ; If the phase of instance 2 is less than 0.5, we want to ; nudge it down when synchronizing, ; and if the phase is > 0.5 we want to nudge it upwards. async1 = async1*ksyncGravity*asyncPolarity ; prepare sync signal ; with polarity and strength kgrainfreq2 = igrainrate * iosc2Dev ; grains per second for second partikkel instance iopcode_id2 = 2 a2 partikkel kgrainfreq2, kdistribution, idisttab, async1, kenv2amt, \ ienv2tab, ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, \ kduration, kamp, igainmasks, kwavfreq, ksweepshape, \ iwavfreqstarttab, iwavfreqendtab, awavfm, ifmamptab, kfmenv, \ icosine, kTrainCps, knumpartials, kchroma, ichannelmasks, \ krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, \ asamplepos4, kwavekey1, kwavekey2, kwavekey3, kwavekey4, \ imax_grains, iopcode_id2 async2, aphase2 partikkelsync iopcode_id2 ; clock pulse and phase ; output of the partikkel instance above, ; we will only use the phase outs a1*amp, a2*amp endin </CsInstruments> <CsScore> /*score parameters igrainrate = p4 ; grain rate igrainsize = p5 ; grain size in ms igrainFreq = p6 ; frequency of source wave within grain iosc2Dev = p7 ; partikkel instance 2 grain rate deviation factor iMaxSync = p8 ; max soft sync amount (increasing to this value during length of note) */ ; GrRate GrSize GrFund Osc2Dev MaxSync i1 0 10 2 20 880 1.3 0.3 s i1 0 10 5 20 440 0.8 0.3 s i1 0 6 55 15 660 1.8 0.45 s i1 0 6 110 10 440 0.6 0.6 s i1 0 6 220 3 660 2.6 0.45 s i1 0 6 220 3 660 2.1 0.45 s i1 0 6 440 3 660 0.8 0.22 s e e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gainslider.csd.xml0000644000000000000000000000365712262561504021007 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in Silent -odac -iadc -d ;;;realtime output </CsOptions> <CsInstruments> sr = 48000 ksmps = 100 nchnls = 2 instr 1 ; gainslider test ; uncomment for realtime midi ;kmod ctrl7 1, 1, 0, 127 ; uncomment for non realtime km0d phasor 1/10 kmod scale km0d, 127, 0 kout gainslider kmod printks "kmod = %f kout = %f\\n", 0.1, kmod, kout aout diskin2 "fox.wav", 1, 0, 1 aout = aout*kout outs aout, aout endin </CsInstruments> <CsScore> i1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/toot3.csd.xml0000644000000000000000000000331212262561504017722 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> instr 3 ; p3=duration of note k1 linen p4, p6, p3, p7 ; p4=amp a1 oscil k1, p5, 1 ; p5=freq out a1 ; p6=attack time endin </CsInstruments> <CsScore> f1 0 4096 10 1 ; sine wave ;ins strt dur amp(p4) freq(p5) attack(p6) release(p7) i3 0 1 10000 440 0.5 0.7 i3 1.5 1 10000 440 0.9 0.1 i3 3 1 5000 880 0.02 0.99 i3 4.5 1 5000 880 0.7 0.01 i3 6 2 20000 220 0.5 0.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/v.csd.xml0000644000000000000000000000463412262561504017127 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o v.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aenv expseg .01, p3*0.25, 1, p3*0.75, 0.01 asig poscil3 .4*aenv, p4, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave ;because note 3 and 5 are played simultaneously and are nearly of the same frequency, ;played together they will create a "beating" sound. i 1 0 2 110 ; note1 v2 i 1 3 . 220 ; note2 i 1 6 . 110 ; note3 v1 i 1 9 . 880 ; note4 i 1 12 . 100 ; note5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen18.csd.xml0000644000000000000000000000600712262561504017600 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen18.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ilen = ftlen(ifn) aphase phasor 220 asig tablei aphase*ilen, ifn outs asig, asig endin </CsInstruments> <CsScore> f 1 0 4096 10 1 ;sine f 2 0 4096 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ;sawtooth f 3 0 4096 11 10 5 2 ;cosine partials f 11 0 8192 18 1 1 0 4095 2 1 4096 8191 ;sine+sawtooth f 12 0 8192 18 1 1 0 4095 3 1 4096 8191 ;sine+cosine partials f 13 0 1024 18 1 0.7 0 767 3 0.7 512 1023 ;sine+cosine partials overlapped, shorter table i 1 0 2 2 ;play sawtooth i 1 + 2 3 ;then cosine partials i 1 5 2 11 ;now sine+sawtooth i 1 + 2 12 ;and sine+cosine partials i 1 + 2 13 ;and sine+cosine partials overlapped, shorter table e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/puts.csd.xml0000644000000000000000000000366412262561504017657 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o puts.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcount init 440 ktrig metro 10 kcount = kcount + ktrig String sprintfk "frequency in Hertz : %d \n", kcount puts String, kcount asig poscil .7, kcount, 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 16384 10 1 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sqrt.csd.xml0000644000000000000000000000413012262561504017642 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sqrt.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 asig pluck 0.7, 55, 55, 0, 1 kpan line 0,p3,1 kleft = sqrt(1-kpan) kright = sqrt(kpan) printks "square root of left channel = %f\\n", 1, kleft ;show coarse of sqaure root values outs asig*kleft, asig*kright ;where 0.707126 is between 2 speakers endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mxadsr.csd.xml0000644000000000000000000000343312262561504020154 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mxadsr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .5 kenv mxadsr 0.5, 0, 1, 0.5 asig pluck kenv, icps, icps, 2, 1 outs asig, asig endin </CsInstruments> <CsScore> f 2 0 4096 10 1 f0 30 ;runs 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpuprc.csd.xml0000644000000000000000000000503412262561504020151 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpuprc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 cpuprc 1, 2 cpuprc 2, 30 instr 1 ;cpu processing-time percent usage is set to 2% for each note asig oscil 0.2, 440, 1 outs asig, asig endin instr 2 ;cpu processing-time percent usage is set to 30% for each note ;so the 4 notes of the score exceeds 100% by far asig oscil 0.2, 440, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 32768 10 1 ; sine wave i 1 0 1 i 1 0 1 i 1 0 1 i 1 0 1 ;too many notes to process, ;check Csound output! i 2 3 1 i 2 3 1 i 2 3 1 i 2 3 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vtablei.csd.xml0000644000000000000000000000377212262561504020312 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac -B441 -b441 </CsOptions> <CsInstruments> sr = 44100 kr = 100 ksmps = 441 nchnls = 2 gindx init 0 instr 1 kindex init 0 ktrig metro 0.5 if ktrig = 0 goto noevent event "i", 2, 0, 0.5, kindex kindex = kindex + 1 noevent: endin instr 2 iout1 init 0 iout2 init 0 iout3 init 0 iout4 init 0 indx = p4 vtablei indx, 1, 1, 0, iout1,iout2, iout3, iout4 print iout1, iout2, iout3, iout4 turnoff endin </CsInstruments> <CsScore> f 1 0 32 10 1 i 1 0 20 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/random.csd.xml0000644000000000000000000000475312262561504020144 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o random.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;same values every time krnd random 100, 1000 printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin instr 2 ;different values every time seed 0 krnd random 100, 1000 ; seed from system clock printk .5, krnd ; look aout oscili 0.8, 440+krnd, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave. i 1 0 1 i 2 2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvslock.csd.xml0000644000000000000000000000376012262561504020342 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvslock.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifil ftgen 0, 0, 0, 1, "fox.wav", 0, 0, 1 instr 1 klock = p4 fsig pvstanal 1, 1, 1, gifil ; no further transformations fsigout pvslock fsig, klock ; lock frequency aout pvsynth fsigout outs aout, aout endin </CsInstruments> <CsScore> i 1 0 2.6 1 ; locked i 1 3 2.6 0 ; not locked e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vco.csd.xml0000644000000000000000000000646412262561504017454 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o vco.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1 instr 1 ; Set the amplitude. kamp = p4 ; Set the frequency. kcps = cpspch(p5) ; Select the wave form. iwave = p6 ; Set the pulse-width/saw-ramp character. kpw init 0.5 ; Use Table #1. ifn = 1 ; Generate the waveform. asig vco kamp, kcps, iwave, kpw, ifn ; Output and amplification. out asig endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 65536 10 1 ; Define the score. ; p4 = raw amplitude (0-32767) ; p5 = frequency, in pitch-class notation. ; p6 = the waveform (1=Saw, 2=Square/PWM, 3=Tri/Saw-Ramp-Mod) i 1 00 02 20000 05.00 1 i 1 02 02 20000 05.00 2 i 1 04 02 20000 05.00 3 i 1 06 02 20000 07.00 1 i 1 08 02 20000 07.00 2 i 1 10 02 20000 07.00 3 i 1 12 02 20000 09.00 1 i 1 14 02 20000 09.00 2 i 1 16 02 20000 09.00 3 i 1 18 02 20000 11.00 1 i 1 20 02 20000 11.00 2 i 1 22 02 20000 11.00 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/integ.csd.xml0000644000000000000000000000517512262561504017771 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o integ.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig diskin2 "fox.wav", 1 outs asig, asig endin instr 2 ; with diff asig diskin2 "fox.wav", 1 ares diff asig outs ares, ares endin instr 3 ; with integ asig diskin2 "fox.wav", 1 aint integ asig aint = aint*.05 ;way too loud outs aint, aint endin instr 4 ; with diff and integ asig diskin2 "fox.wav", 1 ares diff asig aint integ ares outs aint, aint endin </CsInstruments> <CsScore> i 1 0 1 i 2 1 1 i 3 2 1 i 4 3 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outiat.csd.xml0000644000000000000000000000360312262561504020162 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out and midi in and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outiat.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 10 nchnls = 2 instr 1 ikey notnum ivel veloc ivib = 25 ;low value. outiat 1, ivib, 0, 127 ;assign aftertouch on print ivib ;external synth for example to midion 1, ikey, ivel ;change depth of filter modulation endin </CsInstruments> <CsScore> f0 30 ;play for 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/schedwhen.csd.xml0000644000000000000000000000434312262561504020627 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o schedwhen.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kmtr metro 100 ;produce 100 triggers per second schedwhen kmtr, 2, 1, .3 ;but schedwhen plays instr. 2 only once endin instr 2 aenv linseg 0, p3*.1, 1, p3*.3, 1, p3*.6, 0 ;envelope a1 poscil .3*aenv, 1000, 1 outs a1, a1 endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine i 1 0 3 i 1 3 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pop.csd.xml0000644000000000000000000000444012262561504017453 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pop.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 stack 100000 instr 1 a1 oscils 0.7, 220, 0 k1 line 0, p3, 1 push "blah", 123.45, a1, k1 push rnd(k1) k_rnd pop S01, i01, a01, k01 pop printf_i "S01 = '%s', i01 = %g\n", 1, S01, i01 ktrig metro 5.0 printf "k01 = %.3f, k_rnd = %.3f\n", ktrig, k01, k_rnd outs a01, a01 endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/oscil1.csd.xml0000644000000000000000000000423212262561504020046 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil1.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giPan ftgen 0, 0, 8, -2, .5, .2, .8, .1, .9, 0, 1, .5 instr 1 istay = 2 ;how many seconds to stay on the first table value asig vco2 .3, 220 kpan oscil1 istay, 1, p3-istay, giPan ;create panning printk2 kpan ;print when new value aL, aR pan2 asig, kpan ;apply panning outs aL, aR endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/binit.csd.xml0000644000000000000000000000373212262561504017765 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o binit.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ;ain inch 1 ; for live input ain diskin "beats.wav", 1 ; input signal fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .003, 1, 3, 500 ; partial tracking fbins binit fst, 2048 ; convert it back to bins aout pvsynth fbins ; overlap-add resynthesis outs aout, aout endin </CsInstruments> <CsScore> i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/nreverb.csd.xml0000644000000000000000000000445012262561504020321 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o nreverb.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 gaout init 0 instr 1 a1 oscil 15000, 440, 1 out a1 gaout = gaout+a1 endin instr 99 a2 nreverb gaout, 2, .3 out a2*.15 ;volume of reverb gaout = 0 endin </CsInstruments> <CsScore> ; Table 1: an ordinary sine wave. f 1 0 32768 10 1 i 1 0 .5 i 1 1 .5 i 1 2 .5 i 1 3 .5 i 1 4 .5 i 99 0 9 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/deltap3.csd.xml0000644000000000000000000000635312262561504020216 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o deltap3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasig init 0 gidel = 1 ;delay time in seconds instr 1 ain pluck .7, 440, 1000, 0, 1 outs ain, ain vincr gasig, ain ;send to global delay endin instr 2 ifeedback = p4 abuf2 delayr gidel adelL deltap3 .4 ;first tap (on left channel) adelM deltap3 1 ;second tap (on middle channel) delayw gasig + (adelL * ifeedback) abuf3 delayr gidel kdel line 1, p3, .01 ;vary delay time adelR deltap3 .65 * kdel ;one pitch changing tap (on the right chn.) delayw gasig + (adelR * ifeedback) ;make a mix of all deayed signals outs adelL + adelM, adelR + adelM clear gasig endin </CsInstruments> <CsScore> i 1 0 1 i 1 3 1 i 2 0 3 0 ;no feedback i 2 3 8 .8 ;lots of feedback e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/filesr.csd.xml0000644000000000000000000000522012262561504020136 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o filesr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;after an example from Jonathan Murphy instr 1 ;load sound into an ftable Sfile strcpy "beats.wav" ilen filelen Sfile isr filesr Sfile isamps = ilen * isr ;adjust the length of the table to be a power of two closest ;to the actual size of the sound isize init 1 loop: isize = isize * 2 if (isize < isamps) igoto loop itab ftgen 0, 0, isize, 1, Sfile, 0, 0, 0 prints "sample rate = %f, size = %f\n", isr, isize ;prints them endin </CsInstruments> <CsScore> i1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/times_complex.csd.xml0000644000000000000000000001400612262561504021524 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o times_complex.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;by joachim heintz and rory walsh sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giWave ftgen 0, 0, 1024, 10, 1, .5, .25 instr again instance = p4 ;reset the duration of this instance iDur rnd31 5, 3 ;shorter values are more probable iDur = abs(iDur) + 0.2 p3 = iDur ;trigger the effect instrument of this instance event_i "i", "fx_processor", 0, iDur, instance ;print the status quo kTime times prints "instance = %d, start = %f, duration = %f\n", instance, i(kTime), iDur ;make sound iamp active 1 ;scale amplitudes iOct random 5, 10 ;find pitch aEnv transeg 0, 0.02, 0, 1/iamp, p3-0.02, -6, 0 ;output envelope aSend poscil aEnv, cpsoct(iOct), giWave ;audio signal ;send signal to effect instrument Sbus sprintf "audio_%d", instance ;create unique software bus chnset aSend/2, Sbus ;send audio on this bus ;get the last k-cycle of this instance and trigger the successor in it kLast release schedkwhen kLast, 0, 0, "again", 0, 1, instance+1 endin instr fx_processor ;apply feedback delay to the above instrument iwhich = p4 ;receive instance number ... Sbus sprintf "audio_%d", iwhich ; ... and related software bus audio chnget Sbus ;receive audio on this bus irvbtim random 1, 5 ;find reverb time p3 = p3+irvbtim ;adjust instrument duration iltptmL random .1, .5 ;find looptime left ... iltptmR random .1, .5 ;...and right ipan random 0, 1 ; pan and ... imix random 0, 1 ;... mix audio aL,aR pan2 audio, ipan ;create stereo awetL comb aL, irvbtim, iltptmL ;comb filter awetR comb aR, irvbtim, iltptmR aoutL ntrpol aL, awetL, imix ;wet-dry mix aoutR ntrpol aR, awetR, imix outs aoutL/2, aoutR/2 endin </CsInstruments> <CsScore> i "again" 0 1 1 e 3600 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/bformenc.csd.xml0000644000000000000000000000544612262561504020457 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages ;-odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: -o bformenc.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 8 ;bformenc is deprecated, please use bformenc1 instr 1 ; generate pink noise anoise pinkish 1000 ; two full turns kalpha line 0, p3, 720 kbeta = 0 ; fade ambisonic order from 2nd to 0th during second turn kord0 = 1 kord1 linseg 1, p3 / 2, 1, p3 / 2, 0 kord2 linseg 1, p3 / 2, 1, p3 / 2, 0 ; generate B format aw, ax, ay, az, ar, as, at, au, av bformenc anoise, kalpha, kbeta, kord0, kord1, kord2 ; decode B format for 8 channel circle loudspeaker setup a1, a2, a3, a4, a5, a6, a7, a8 bformdec 4, aw, ax, ay, az, ar, as, at, au, av ; write audio out outo a1, a2, a3, a4, a5, a6, a7, a8 endin </CsInstruments> <CsScore> ; Play Instrument #1 for 20 seconds. i 1 0 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsbufread2.csd.xml0000644000000000000000000000351212262561504021077 0ustar rootroot <CsoundSynthesizer> <CsOptions> </CsOptions> <CsInstruments> ksmps = 64 0dbfs = 1 nchnls = 2 instr 1 kcnt init 0 ifftsize = 2048 ihop = ifftsize/4 a1 diskin2 "beats.wav", 1, 0, 1 a1 = a1*0.5 fsig1 pvsanal a1, ifftsize, ihop, ifftsize, 1 ih, kt pvsbuffer fsig1, 10 fsig2 pvsbufread2 kt, ih, 1, 1 fsig3 pvsbufread2 kt, ih, 2, 2 a2 pvsynth fsig3 a3 pvsynth fsig2 outs a2, a3 endin </CsInstruments> <CsScore> f1 0 2048 -7 0 128 1.1 128 0.5 256 1.8 512 1.1 1024 0.1 f2 0 2048 -7 1 128 0.2 128 0.05 256 0.5 512 0.9 1024 0.1 i1 0 60 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fout_poly.csd.xml0000644000000000000000000000665712262561504020711 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o fout_poly.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1 - Play an audio file. instr 1 ; Generate an audio signal using ; the audio file "beats.wav". asig soundin "beats.wav" out asig endin ; Instrument #2 - Create a basic tone. instr 2 iamp = 5000 icps = 440 iphs = 0 ; Create an audio signal. asig oscils iamp, icps, iphs out asig endin ; Instrument #99 - Save the global signal to a file. instr 99 ; Read the csound output buffer aoutput monitor ; Write the output of csound to a headerless ; audio file called "fout_poly.raw". fout "fout_poly.raw", 1, aoutput endin </CsInstruments> <CsScore> ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 every quarter-second. i 2 0.00 0.1 i 2 0.25 0.1 i 2 0.50 0.1 i 2 0.75 0.1 i 2 1.00 0.1 i 2 1.25 0.1 i 2 1.50 0.1 i 2 1.75 0.1 ; Make sure the global instrument, #99, is running ; during the entire performance (2 seconds). i 99 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/checkbox.csd.xml0000644000000000000000000000345012262561504020443 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o checkbox.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 10 nchnls = 2 instr 1 ; Get the value from the checkbox. k1 checkbox 1 ; If the checkbox is selected then k2=440, otherwise k2=880. k2 = (k1 == 0 ? 440 : 880) a1 oscil 10000, k2, 1 outs a1, a1 endin </CsInstruments> <CsScore> ; sine wave. f 1 0 32768 10 1 i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvspitch.csd.xml0000644000000000000000000000430612262561504020516 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvspitch.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 1 giwave ftgen 0, 0, 4096, 10, 1, 0.5, 0.333, 0.25, 0.2, 0.1666 instr 1 ifftsize = 1024 iwtype = 1 /* cleaner with hanning window */ a1 inch 1 ;Realtime audio input ;a1 soundin "input.wav" ;Use this line for file input fsig pvsanal a1, ifftsize, ifftsize/4, ifftsize, iwtype kfr, kamp pvspitch fsig, 0.01 adm oscil kamp, kfr * 1.5, giwave ;Generate note a fifth above detected pitch out adm endin </CsInstruments> <CsScore> i 1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLslidBnk2Setk.csd.xml0000644000000000000000000001150112262561504021372 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O </CsOptions> <CsInstruments> sr = 44100 ksmps = 10 nchnls = 2 ;Example by Gabriel Maldonado 2007 giElem init 8 giOutTab ftgen 1,0,128, 2, 0 giSine ftgen 3,0,256,10, 1 giOutTab2 ftgen 4,0,128, 2, 0 itab ftgen 29, 0, 129, 5, .002, 128, 1 ;** exponential ascending curve for slider mapping giExpTab ftgen 30, 0, 129, -24, itab, 0, 1 ;** rescaled curve for slider mapping giConfigTab ftgen 2,0,128,-2, 1, 500, -1, 13, \ 1, 500, -1, 13, \ 1, 500, -1, 13, \ 1, 5000, -1, 13, \ \ 1, 1000, -1, 5, \ 1, 1000, -1, 5, \ 1, 1000, -1, 5, \ 1, 5000, -1, 5 FLpanel "Multiple FM",600,600 FLslidBnk2 "mod1@mod2@mod3@amp@freq1@freq2@freq3@freqPo", giElem, giOutTab2, giConfigTab, 400, 500, 100, 10 giHandle FLslidBnkGetHandle FLpanel_end FLrun instr 1 ktrig slider8table 1, giOutTab, 0, \ ; ctl min max init func 27, 1, 500, 3, -1, \ ;1 repeat rate 28, 1, 500, 4, -1, \ ;2 random freq. amount 29, 1, 500, 1, -1, \ ;3 random amp. amount 30, 1, 5000, 1, -1, \ ;4 number of concurrent loop points \ 31, 1, 1000, 1, -1, \;5 kloop1 32, 1, 1000, 1, -1, \;6 kloop2 33, 1, 1000, 1, -1, \;7 kloop3 34, 1, 1000, 1, -1 ;8 kloop4 kmodindex1 init 0 kmodindex2 init 0 kmodindex3 init 0 kamp init 0 kfreq1 init 0 kfreq2 init 0 kfreq3 init 0 kfreq4 init 0 vtable1k giOutTab2, kmodindex1, kmodindex2, kmodindex3, kamp, kfreq1, kfreq2, kfreq3, kfreq4 ; *kflag, *ihandle, *ifn, *startInd, *startSlid, *numSlid; FLslidBnk2Setk ktrig, giHandle, giOutTab, 0, 0, giElem printk2 kmodindex1 printk2 kmodindex2,10 printk2 kmodindex3,20 printk2 kamp,30 amod1 oscili kmodindex1, kfreq1, giSine amod2 oscili kmodindex2, kfreq2, giSine amod3 oscili kmodindex3, kfreq3, giSine aout oscili kamp, kfreq4+amod1+amod2+amod3, giSine outs aout, aout endin </CsInstruments> <CsScore> i1 0 3600 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midiout.csd.xml0000644000000000000000000000301012262561504020317 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -Ma -Q1 ;;;realtime audio out and midi out and midi in (all midi inputs) </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 midiout 192, 1, 21, 0 ;program change to instr. 21 inum notnum ivel veloc midion 1, inum, ivel endin </CsInstruments> <CsScore> i 1 0 3 80 100 ;play note for 3 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpumeter.csd.xml0000644000000000000000000000431312262561504020500 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpumeter.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 1 0dbfs = 1 instr 1 ;cpu metering; stop when too large k0 cpumeter 0.1 printk2 k0 if k0>70 then event "i", 3, 0.1, 1 endif endin instr 2 event_i "i", 2, 1, 1000 asig oscil 0.2, 440, 1 out asig endin instr 3 exitnow endin </CsInstruments> <CsScore> f 1 0 32768 10 1 ; sine wave i 1 0 1000 i 2 0 1000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftmorf.csd.xml0000644000000000000000000000434712262561504020160 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftmorf.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs =1 instr 1 kndx line 0, p3, 7 ftmorf kndx, 1, 2 asig oscili .8, 440, 2 outs asig, asig endin </CsInstruments> <CsScore> f1 0 8 -2 3 4 5 6 7 8 9 10 f2 0 1024 10 1 /*contents of f2 dont matter */ f3 0 1024 10 1 f4 0 1024 10 0 1 f5 0 1024 10 0 0 1 f6 0 1024 10 0 0 0 1 f7 0 1024 10 0 0 0 0 1 f8 0 1024 10 0 0 0 0 0 1 f9 0 1024 10 0 0 0 0 0 0 1 f10 0 1024 10 1 1 1 1 1 1 1 i1 0 15 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pconvolve.csd.xml0000644000000000000000000001437412262561504020677 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out -iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pconvolve.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kmix = .5 ; Wet/dry mix. Vary as desired. kvol = .05*kmix ; Overall volume level of reverb. May need to adjust ; when wet/dry mix is changed, to avoid clipping. ; do some safety checking to make sure we the parameters a good kmix = (kmix < 0 || kmix > 1 ? .5 : kmix) kvol = (kvol < 0 ? 0 : .5*kvol*kmix) ; size of each convolution partion -- for best performance, this parameter needs to be tweaked ipartitionsize = p4 ; calculate latency of pconvolve opcode idel = (ksmps < ipartitionsize ? ipartitionsize + ksmps : ipartitionsize)/sr prints "Convolving with a latency of %f seconds%n", idel ; actual processing al, ar ins ;get live input awetl, awetr pconvolve kvol*(al+ar), "kickroll.wav", ipartitionsize ; Delay dry signal, to align it with the convoled sig adryl delay (1-kmix)*al, idel adryr delay (1-kmix)*al, idel outs adryl+awetl, adryr+awetr endin instr 2 imix = 0.5 ; Wet/dry mix. Vary as desired. ivol = .05*imix ; Overall volume level of reverb. May need to adjust ; when wet/dry mix is changed, to avoid clipping. ipartitionsize = 1024 ; size of each convolution partion idel = (ksmps < ipartitionsize ? ipartitionsize + ksmps : ipartitionsize)/sr ; latency of pconvolve opcode kcount init idel*kr ; since we are using a soundin [instead of ins] we can ; do a kind of "look ahead" by looping during one k-pass ; without output, creating zero-latency loop: asig soundin p4, 0 awetl, awetr pconvolve ivol*(asig),"rv_stereo.wav", ipartitionsize adry delay (1-imix)*asig,idel ; Delay dry signal, to align it with kcount = kcount - 1 if kcount > 0 kgoto loop outs awetl+adry, awetr+adry endin </CsInstruments> <CsScore> i 1 0 20 1024 ;play live for 20 seconds i 2 20 5 "fox.wav" i 2 25 5 "flute.aiff" e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen01.csd.xml0000644000000000000000000000560512262561504017573 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen01.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;plays deferred and non-deferred sounds with loscil ifn = p4 ibas = 1 asig loscil 1, 1, ifn, ibas outs asig, asig endin instr 2 ;plays only non-deffered sound isnd = p4 aread line sr*p3, p3, 0 ;play this backward asig tablei aread, isnd ;use table 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 131072 1 "beats.wav" 0 0 0 ;non-deferred sound f 2 0 0 1 "flute.aiff" 0 0 0 ;& deferred sounds in f 3 0 0 1 "beats.ogg" 0 0 0 ;different formats i 1 0 1 1 i 1 + 1 2 i 1 + 1 3 i 2 4 2 1 ;non-deffered sound for instr. 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/statvar.csd.xml0000644000000000000000000000371512262561504020345 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 ;;;RT audio I/O, midi in ; For Non-realtime ouput leave only the line below: ; -o statevar.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 kenv linseg 0,0.1,1, p3-0.2,1, 0.1, 0 asig buzz 16000*kenv, 100, 100, 1 kf expseg 100, p3/2, 5000, p3/2, 1000 ahp,alp,abp,abr statevar asig, kf, 200 outs alp,ahp ; lowpass left, highpass right endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 i1 0 2 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sflooper.csd.xml0000644000000000000000000000566512262561504020520 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -+rtmidi=virtual -M0 ;;;realtime audio in, midi in ; For Non-realtime ouput leave only the line below: ; -o sflooper.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 isf sfload "07AcousticGuitar.sf2" sfpassign 0, isf instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 print ivel ivel init ivel/127 ;velocity dependent kamp linsegr 1,1,1,.1,0 ;envelope kamp = kamp * .0002 ;scale amplitude (= kamp/5000) kfreq init 1 ;do not change freq from sf ;"07AcousticGuitar.sf2" contains 2 samples, on notes E1 and C#4 ;start loop from beginning, loop .2 seconds - on the root key of these samples aL,aR sflooper ivel, inum, kamp*ivel, kfreq, 0, 0, .2, .05 outs aL, aR endin </CsInstruments> <CsScore> f0 60 ; stay active for 1 minute i1 0 1 60 100 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/define_args.csd.xml0000644000000000000000000000420512262561504021122 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o define_args.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Define the oscillator macro. #define OSCMACRO(VOLUME'FREQ'TABLE) #oscil $VOLUME, $FREQ, $TABLE# ; Instrument #1 instr 1 ; Use the oscillator macro. ; This will be expanded to "a1 oscil 5000, 440, 1". a1 $OSCMACRO(5000'440'1) ; Send it to the output. out a1 endin </CsInstruments> <CsScore> ; Define Table #1 with an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/setscorepos.csd.xml0000644000000000000000000000421612262561504021227 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o setscorepos.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 8192, 10, 1 instr 1 asig poscil 0.5, p4, giSine ;play something outs asig, asig endin instr 11 setscorepos 8.5 endin </CsInstruments> <CsScore> i1 0 2 220 ;this one will be played i11 2.5 1 ;start setscorepos now i1 3 2 330 ;skip this note i1 6 2 440 ;and this one i1 9 2 550 ;play this one </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/taninv.csd.xml0000644000000000000000000000327112262561504020155 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o taninv.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 0.5 i1 = taninv(irad) print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outrg.csd.xml0000644000000000000000000000547012262561504020021 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outrg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 instr 1 kleft init 1 asig vco2 .5, 220 ;sawtooth idur = p3/(nchnls-1) knext init idur kpos init 0 krate init 1/idur kbase init 0 ktime timeinsts if ktime>=knext then kleft = kleft + 1 knext = knext + idur kpos = 0 kbase = ktime else kpos = (ktime-kbase)/idur endif printks "speaker %d position %f\n", 0, kleft, kpos a1,a2 pan2 asig, kpos outrg kleft, a1, a2 kpos = kbase/idur endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ATSsinnoi-2.csd.xml0000644000000000000000000001370612262561504020670 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac -d -m128 </CsOptions> <CsInstruments> ;example by joachim heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 1024, 10, 1 gSfile = "fox.ats" giNumParts ATSinfo gSfile, 3 ;overall number of partials giDur ATSinfo gSfile, 7 ;duration seed 0 instr PlayList event_i "i", "PlayAll", 0, 1, 1, 0, .5 ;sine only, half speed event_i "i", "PlayAll", giDur*2+1, 1, 0, 1, .5 ;noise only event_i "i", "PlayAll", giDur*4+2, 1, .5, .5, .5 ;half sine, half noise endin instr PlayAll iSinAmnt = p4 ;sinee amount (0-1) iNzAmnt = p5 ;noise amount (0-1) iSpeed = p6 ;speed p3 = giDur/iSpeed ktime line 0, giDur/iSpeed, giDur prints "Resynthesizing all partials with tone = %.1f and noise = %.1f.\n", iSinAmnt, iNzAmnt aOut ATSsinnoi ktime, iSinAmnt, iNzAmnt, 1, gSfile, giNumParts outs aOut, aOut endin instr PlayBand iOffset = p4 ;offset in partials iSpeed = p5 ;speed p3 = giDur/iSpeed ktime line 0, giDur/iSpeed, giDur prints "Resynthesizing partials %d to %d with related noise.\n", iOffset+1, iOffset+10 aOut ATSsinnoi ktime, 1, 1, 1, gSfile, 10, iOffset, 1 outs aOut, aOut ;call itself again if iOffset < giNumParts - 20 then event_i "i", "PlayBand", giDur/iSpeed+1, 1, iOffset+10, iSpeed endif endin instr PlayWeighted ;sine amount, noise amount and speeed are varying kSinAmnt randomi 0, 1, 1, 3 kNzAmnt = 1-kSinAmnt kSpeed randomi .01, .3, 1, 3 async init 0 atime, aEnd syncphasor kSpeed/giDur, async kTrig metro 100 kEnd max_k aEnd, kTrig, 1 ;1 if phasor signal crosses zero ktime downsamp atime aOut ATSsinnoi ktime*giDur, kSinAmnt, kNzAmnt, 1, gSfile, giNumParts outs aOut, aOut ;exit if file is at the end if kEnd == 1 then event "i", "End", 0, 1 endif endin instr End exitnow endin </CsInstruments> <CsScore> i "PlayList" 0 1 i "PlayBand" 20 1 0 .5 i "PlayWeighted" 110 100 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/bbcuts.csd.xml0000644000000000000000000000416412262561504020142 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o bbcuts.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ;Play an audio file aleft, aright diskin2 "kickroll.wav", 1, 0 outs aleft, aright endin instr 2 ;Cut-up stereo audio file. ibps = 16 isubdiv = 2 ibarlength = 2 iphrasebars = 1 inumrepeats = 8 aleft, aright diskin2 "kickroll.wav", 1, 0 aleft, aright bbcuts aleft, aright, ibps, isubdiv, ibarlength, iphrasebars, inumrepeats outs aleft, aright endin </CsInstruments> <CsScore> i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lessequal.csd.xml0000644000000000000000000000500412262561504020650 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o <=.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kval randomh 0, 1.2, 20 ;choose between 0 and 1.2 if kval >0 && kval<=.5 then ;3 possible outcomes kvl = 1 elseif kval >.5 && kval<=1 then kvl =2 elseif kval >1 then kvl =3 endif printks "random number = %f, result = %f\n", .1, kval, kvl asig poscil .7, 440*kvl, 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 16384 10 1 i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLpanel-sensekey.csd.xml0000644000000000000000000000372712262561504022031 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLpanel-sensekey.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Example by Johnathan Murphy sr = 44100 ksmps = 128 nchnls = 2 ; ikbdcapture flag set to 1 ikey init 1 FLpanel "sensekey", 740, 340, 100, 250, 2, ikey gkasc, giasc FLbutBank 2, 16, 8, 700, 300, 20, 20, -1 FLpanelEnd FLrun instr 1 kkey sensekey kprint changed kkey FLsetVal kprint, kkey, giasc endin </CsInstruments> <CsScore> i1 0 60 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/expsega.csd.xml0000644000000000000000000000450612262561504020314 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o expsega.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Define a short percussive amplitude envelope that ; goes from 0.01 to 20,000 and back. aenv expsega 0.01, 0.1, 20000, 0.1, 0.01 a1 oscil aenv, 440, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #1 for one second. i 1 1 1 ; Play Instrument #1 for one second. i 1 2 1 ; Play Instrument #1 for one second. i 1 3 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/opcode_example.csd.xml0000644000000000000000000001354112262561504021643 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o opcode_example.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 /* example opcode 1: simple oscillator */ opcode Oscillator, a, kk kamp, kcps xin ; read input parameters a1 vco2 kamp, kcps ; sawtooth oscillator xout a1 ; write output endop /* example opcode 2: lowpass filter with local ksmps */ opcode Lowpass, a, akk setksmps 1 ; need sr=kr ain, ka1, ka2 xin ; read input parameters aout init 0 ; initialize output aout = ain*ka1 + aout*ka2 ; simple tone-like filter xout aout ; write output endop /* example opcode 3: recursive call */ opcode RecursiveLowpass, a, akkpp ain, ka1, ka2, idep, icnt xin ; read input parameters if (icnt >= idep) goto skip1 ; check if max depth reached ain RecursiveLowpass ain, ka1, ka2, idep, icnt + 1 skip1: aout Lowpass ain, ka1, ka2 ; call filter xout aout ; write output endop /* example opcode 4: de-click envelope */ opcode DeClick, a, a ain xin aenv linseg 0, 0.02, 1, p3 - 0.05, 1, 0.02, 0, 0.01, 0 xout ain * aenv ; apply envelope and write output endop /* instr 1 uses the example opcodes */ instr 1 kamp = .7 ; amplitude kcps expon 50, p3, 500 ; pitch a1 Oscillator kamp, kcps ; call oscillator kflt linseg 0.4, 1.5, 0.4, 1, 0.8, 1.5, 0.8 ; filter envelope a1 RecursiveLowpass a1, kflt, 1 - kflt, 10 ; 10th order lowpass a1 DeClick a1 outs a1, a1 endin </CsInstruments> <CsScore> i 1 0 4 e5 ;extra second before quitting </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/diskin2.csd.xml0000644000000000000000000000311012262561504020211 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o diskin2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktrans linseg 1, 5, 2, 10, -2 a1 diskin2 "beats.wav", ktrans, 0, 1, 0, 32 outs a1, a1 endin </CsInstruments> <CsScore> i 1 0 15 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/octcps.csd.xml0000644000000000000000000000350612262561504020152 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o octcps.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Convert a cycles-per-second value into an ; octave value. icps = 440 ioct = octcps(icps) print ioct endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvbufread.csd.xml0000644000000000000000000000403512262561504020633 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvbufread.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; analyze "fox.wav" and "flute.aiff" with PVANAL first ktime1 line 0, p3, .8 ; use a part of "flute.pvx" file ktime2 line 0, p3, 1.2 ; use a part of "beats.pvx" file kcross expon .03, p3, 1 pvbufread ktime1, "flute.pvx" asig pvcross ktime2, 1, "beats.pvx", 1-kcross, kcross outs asig, asig endin </CsInstruments> <CsScore> i 1 0 3 i 1 + 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pwd.csd.xml0000644000000000000000000000320612262561504017446 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ; -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o system.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 ; Waits for command to execute before continuing Swd pwd puts Swd, 1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for minimal time. i 1 0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/rnd.csd.xml0000644000000000000000000000432412262561504017441 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o rnd.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Andres Cabrera 2010 sr = 44100 ksmps = 4410 nchnls = 1 0dbfs = 1 instr 1 ; Generate a random number from 0 to 10. irand = rnd(10) print irand endin instr 2 klimit init 10 krand = rnd(klimit) printk 0, krand endin </CsInstruments> <CsScore> i 1 0 1 ; Generate 1 number i 1 0 1 ; Generate another number i 1 0 1 ; yet another number i 2 2 1 ; 1 second prints 9 values (kr = 10) e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dcblock2.csd.xml0000644000000000000000000000411012262561504020332 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o dcblock2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ;add DC to "beats.wav" asig soundin "beats.wav" asig = asig+5000 ;adds DC of 5000 outs asig, asig endin instr 2 ;dcblock audio asig soundin "beats.wav" asig = asig+5000 ;adds DC adc dcblock2 asig ;remove DC again outs adc, adc endin </CsInstruments> <CsScore> i 1 0 2 i 2 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/clockoff.csd.xml0000644000000000000000000000502012262561504020436 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o clockoff.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Start clock #1. clockon 1 ; Do something that keeps Csound busy. a1 oscili 10000, 440, 1 out a1 ; Stop clock #1. clockoff 1 ; Print the time accumulated in clock #1. i1 readclock 1 print i1 endin </CsInstruments> <CsScore> ; Initialize the function tables. ; Table 1: an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for one second starting at 0:00. i 1 0 1 ; Play Instrument #1 for one second starting at 0:01. i 1 1 1 ; Play Instrument #1 for one second starting at 0:02. i 1 2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/event.csd.xml0000644000000000000000000000606512262561504020003 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o event.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - an oscillator with a high note. instr 1 ; Create a trigger and set its initial value to 1. ktrigger init 1 ; If the trigger is equal to 0, continue playing. ; If not, schedule another event. if (ktrigger == 0) goto contin ; kscoreop="i", an i-statement. ; kinsnum=2, play Instrument #2. ; kwhen=1, start at 1 second. ; kdur=0.5, play for a half-second. event "i", 2, 1, 0.5 ; Make sure the event isn't triggered again. ktrigger = 0 contin: a1 oscils 10000, 440, 1 out a1 endin ; Instrument #2 - an oscillator with a low note. instr 2 a1 oscils 10000, 220, 1 out a1 endin </CsInstruments> <CsScore> ; Make sure the score plays for two seconds. f 0 2 ; Play Instrument #1 for a half-second. i 1 0 0.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/maca.csd.xml0000644000000000000000000000765212262561504017566 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mac.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;4 band equalizer klow = p4 ;low gain1 aenv1 oscil 0.2, 1, 4 aenv1 = aenv1 + klow klmid = p5 ;low gain2 aenv2 oscil 0.21, 1.1, 4 aenv2 = aenv2 + klmid kmidh = p6 ;high gain1 aenv3 oscil 0.19, 1.2, 4 aenv3 = aenv3 + kmidh khigh = p7 ;high gain2 aenv4 oscil 0.18, 1.3, 4 aenv4 = aenv4 + khigh ifn = p8 ;table ilc1 table 0, ifn ;low freqency range ilc2 table 1, ifn ;low-mid ihc1 table 2, ifn ;mid-high ihc2 table 3, ifn ;high asig diskin2 "fox.wav", 1 alow1 butterlp asig, ilc1 ;lowpass 1 almid butterlp asig, ilc2 ;lowpass 2 amidh butterhp asig, ihc1 ;highpass 1 ahigh butterhp asig, ihc2 ;highpass 2 aout maca aenv1, alow1, aenv2, almid, aenv3, amidh, aenv4, ahigh outs aout, aout endin </CsInstruments> <CsScore> f1 0 4 -2 150 300 600 5000 f2 0 4 -2 75 500 1000 10000 f3 0 4 -2 200 700 1500 3000 f4 0 4096 10 1 ; low lowmid midhigh high table i 1 0 2.8 2 1 1 1 1 i 1 3 2.8 2 3 1 1 2 i 1 6 2.8 2 1 2 3 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/nrpn.csd.xml0000644000000000000000000000350612262561504017634 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -Q1 ;;;realtime audio out with MIDI out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; change attack time of external synth initc7 1, 6, 0 ; set controller 6 to 0 nrpn 1, 99, 1 ; set MSB nrpn 1, 98, 99 ; set LSB katt ctrl7 1, 6, 1, 127 ; DataEntMSB idur = 2 noteondur2 1, 60, 100, idur ; play note on synth endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lpshold.csd.xml0000644000000000000000000000636412262561504020331 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpshold.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfrq init p4 ; frequency of the loop ifrac = p5 ; fraction of frequency at which to force retrigger ipan = p6 ; pan position ktrig metro kfrq * ifrac ; trigger to force restart the loop iphs = 0 ; initial phase of the loop ; a loop of midi note numbers: note duration etc... knote lpshold kfrq, ktrig, iphs, 61, 0.0625, 60, 0.9375, 61, 1, 58, 1, \ 63, 2, 65, 3 aenv linseg 0,0.01,1,p3-0.11,1,0.1,0 ; amplitude envelope krnd rspline -0.05,0.05,0.5,1 ; random detune asig gbuzz 0.2*aenv, cpsmidinn(knote+krnd), 30, 1, 0.5, 1 outs asig*ipan, asig*(1-ipan) endin </CsInstruments> <CsScore> ; cosine wave. f 1 0 16384 11 1 ; 3 layers of the loop are played, each at a different speed, ; - with different retriggering rate, and pan location. i 1 0 60 0.5 [8/10] 0.5 i 1 0 60 0.375 [8/11] 0.1 i 1 0 60 0.25 [8/13] 0.9 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outkat.csd.xml0000644000000000000000000000406312262561504020165 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -Q1 -M0 ;;;realtime audio out and midi in and out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outkat.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 10 nchnls = 2 instr 1 ikey notnum ivel veloc kvib linseg 100, .5, 120 ;vary aftertouch in .5 second kvbr = int(kvib) ;whole numbers only outkat 1, kvbr, 0, 127 ;assign aftertouch on printk2 kvbr ;external synth for example to midion 1, ikey, ivel ;change depth of filter modulation endin </CsInstruments> <CsScore> f0 30 ;play for 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outkpc_fltk.csd.xml0000644000000000000000000000341712262561504021205 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1;;;RT audio I/O with MIDI in </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 FLpanel "outkpc",200,100,90,90;start of container gkpg, gihandle FLcount "Midi-Program change",0,127,1,5,1,152,40,16,23,-1 FLpanelEnd FLrun instr 1 ktrig changed gkpg outkpc ktrig,gkpg,0,127 endin </CsInstruments> <CsScore> ; Run instrument 1 for 60 seconds i 1 0 60 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLsetText.csd.xml0000644000000000000000000000622712262561504020544 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLsetText.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 2 ; Example by Giorgio Zucco and Andres Cabrera 2007 FLpanel "FLsetText",250,100,50,50 gk1,giha FLcount "", 1, 20, 1, 20, 1, 200, 40, 20, 20, 0, 1, 0, 1 FLpanelEnd FLrun instr 1 ; This instrument is triggered by FLcount above each time ; its value changes iname = i(gk1) print iname ; Must use FLsetText on the init pass! if (iname == 1) igoto text1 if (iname == 2) igoto text2 if (iname == 3) igoto text3 igoto end text1: FLsetText "FM",giha igoto end text2: FLsetText "GRANUL",giha igoto end text3: FLsetText "PLUCK",giha igoto end end: endin </CsInstruments> <CsScore> f 0 3600 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/phaser2.csd.xml0000644000000000000000000000705012262561504020221 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o phaser2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 2 ; demonstration of phase shifting abilities of phaser2. ; Input mixed with output of phaser2 to generate notches. ; Demonstrates the interaction of imode and ksep. idur = p3 iamp = p4 * .04 iorder = p5 ; number of 2nd-order stages in phaser2 network ifreq = p6 ; not used ifeed = p7 ; amount of feedback for phaser2 imode = p8 ; mode for frequency scaling isep = p9 ; used with imode to determine notch frequencies kamp linseg 0, .2, iamp, idur - .2, iamp, .2, 0 iharms = (sr*.4) / 100 ; "Sawtooth" waveform exponentially decaying function, to control notch frequencies asig gbuzz 1, 100, iharms, 1, .95, 2 kline expseg 1, idur, .005 aphs phaser2 asig, kline * 2000, .5, iorder, imode, isep, ifeed out (asig + aphs) * iamp endin </CsInstruments> <CsScore> ; cosine wave for gbuzz f2 0 8192 9 1 1 .25 ; phaser2, imode=1 i2 00 10 7000 8 .2 .9 1 .33 i2 11 10 7000 8 .2 .9 1 2 ; phaser2, imode=2 i2 22 10 7000 8 .2 .9 2 .33 i2 33 10 7000 8 .2 .9 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLmouse.csd.xml0000644000000000000000000000523112262561504020226 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 ;Example by Andres Cabrera 2007 giwidth = 400 giheight = 300 FLpanel "FLmouse", giwidth, giheight, 10, 10 FLpanelEnd FLrun 0dbfs = 1 instr 1 kx, ky, kb1, kb2, kb3 FLmouse 2 ktrig changed kx, ky ;Print only if coordinates have changed printf "kx = %f ky = %f \n", ktrig, kx, ky kfreq = ((giwidth - ky)*1000/giwidth) + 300 ; y coordinate determines frequency, x coordinate determines amplitude ; Left mouse button (kb1) doubles the frequency ; Right mouse button (kb3) activates sound on channel 2 aout oscil kx /giwidth , kfreq * (kb1 + 1), 1 outs aout, aout * kb3 endin </CsInstruments> <CsScore> f 1 0 1024 10 1 i 1 0 120 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pdhalf.csd.xml0000644000000000000000000001033012262561504020106 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pdhalf.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 4 idur = p3 iamp = p4 ifreq = p5 itable = p6 aenv linseg 0, .001, 1.0, idur - .051, 1.0, .05, 0 aosc phasor ifreq kamount linseg 0.0, 0.02, -0.99, 0.05, -0.9, idur-0.06, 0.0 apd pdhalf aosc, kamount aout tablei apd, itable, 1 outs aenv*aout*iamp, aenv*aout*iamp endin </CsInstruments> <CsScore> f1 0 16385 10 1 f2 0 16385 10 1 .5 .3333 .25 .5 f3 0 16385 9 1 1 270 ; inverted cosine ; descending "just blues" scale ; pdhalf with cosine table ; (imitates the CZ-101 "sawtooth waveform") t 0 100 i4 0 3 .6 512 3 i. + . . 448 i. + . . 384 i. + . . 358.4 i. + . . 341.33 i. + . . 298.67 i. + 5 . 256 s ; pdhalf with a sine table t 0 120 i4 0 3 .6 512 1 i. + . . 448 i. + . . 384 i. + . . 358.4 i. + . . 341.33 i. + . . 298.67 i. + 5 . 256 s ; pdhalf with a sawtooth-like table t 0 150 i4 0 3 .6 512 2 i. + . . 448 i. + . . 384 i. + . . 358.4 i. + . . 341.33 i. + . . 298.67 i. + 5 . 256 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/areson.csd.xml0000644000000000000000000000412512262561504020144 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o areson.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; unfiltered noise asig rand 0.5 ; white noise signal. outs asig, asig endin instr 2 ; filtered noise kcf init 1000 kbw init 100 asig rand 0.5 afil areson asig, kcf, kbw afil balance afil,asig ; afil = very loud outs afil, afil endin </CsInstruments> <CsScore> i 1 0 2 i 2 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvswarp.csd.xml0000644000000000000000000000405312262561504020357 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pvswarp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kscal = p4 asig soundin "fox.wav" ; get the signal in fsig pvsanal asig, 1024, 256, 1024, 1 ; analyse it ftps pvswarp fsig, kscal, 0 ; warp it atps pvsynth ftps ; synthesise it outs atps, atps endin </CsInstruments> <CsScore> i 1 0 3 1 i 1 + 3 1.5 i 1 + 3 3 i 1 + 3 .25 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fmb3.csd.xml0000644000000000000000000000356212262561504017510 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmb3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 220 kc1 = p4 kc2 = p5 kvrate = 6 kvdpth line 0, p3, p6 asig fmb3 .4, kfreq, kc1, kc2, kvdpth, kvrate outs asig, asig endin </CsInstruments> <CsScore> ;sine wave. f 1 0 32768 10 1 i 1 0 2 5 5 0.1 i 1 3 2 .5 .5 0.01 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/moscil.csd.xml0000644000000000000000000000360712262561504020147 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1;;;RT audio I/O with MIDI in </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 instr 1 ;Triggered by MIDI notes on channel 1 inote notnum ivel veloc kpitch = 40 kfreq = 2 kdur = .04 kpause = .1 k1 lfo kpitch, kfreq,5 ;plays a stream of notes of kdur duration on MIDI channel 1 moscil 1, inote + k1, ivel, kdur, kpause endin </CsInstruments> <CsScore> ; Dummy ftable f0 60 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen52.csd.xml0000644000000000000000000001237312262561504017601 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen52.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 garvb init 0 gaW init 0 gaX init 0 gaY init 0 itmp ftgen 1, 0, 64, -2, 2, 40, -1, -1, -1, 123, \ 1, 13.000, 0.05, 0.85, 20000.0, 0.0, 0.50, 2, \ 1, 2.000, 0.05, 0.85, 20000.0, 0.0, 0.25, 2, \ 1, 16.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 9.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 12.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2, \ 1, 8.000, 0.05, 0.85, 20000.0, 0.0, 0.35, 2 itmp ftgen 2, 0, 262144, -2, 0 spat3dt 2, -0.2, 1, 0, 1, 1, 2, 0.005 itmp ftgen 3, 0, 262144, -52, 3, 2, 0, 4, 2, 1, 4, 2, 2, 4 instr 1 a1 vco2 1, 440, 10 kfrq port 100, 0.008, 20000 a1 butterlp a1, kfrq a2 linseg 0, 0.003, 1, 0.01, 0.7, 0.005, 0, 1, 0 a1 = a1 * a2 * 2 denorm a1 vincr garvb, a1 aw, ax, ay, az spat3di a1, p4, p5, p6, 1, 1, 2 vincr gaW, aw vincr gaX, ax vincr gaY, ay endin instr 2 denorm garvb ; skip as many samples as possible without truncating the IR arW, arX, arY ftconv garvb, 3, 2048, 2048, (65536 - 2048) aW = gaW + arW aX = gaX + arX aY = gaY + arY garvb = 0 gaW = 0 gaX = 0 gaY = 0 aWre, aWim hilbert aW aXre, aXim hilbert aX aYre, aYim hilbert aY aWXr = 0.0928*aXre + 0.4699*aWre aWXiYr = 0.2550*aXim - 0.1710*aWim + 0.3277*aYre aL = aWXr + aWXiYr aR = aWXr - aWXiYr outs aL, aR endin </CsInstruments> <CsScore> i 1 0 0.5 0.0 2.0 -0.8 i 1 1 0.5 1.4 1.4 -0.6 i 1 2 0.5 2.0 0.0 -0.4 i 1 3 0.5 1.4 -1.4 -0.2 i 1 4 0.5 0.0 -2.0 0.0 i 1 5 0.5 -1.4 -1.4 0.2 i 1 6 0.5 -2.0 0.0 0.4 i 1 7 0.5 -1.4 1.4 0.6 i 1 8 0.5 0.0 2.0 0.8 i 2 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/serialWrite.csd.xml0000644000000000000000000000541012262561504021145 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n ;;;no output ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o serialWrite.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 500 ; the default krate can be too fast for the arduino to handle nchnls = 2 0dbfs = 1 instr 1 iPort serialBegin "/dev/cu.usbmodemfa131", 9600 ;connect to the arduino with baudrate = 9600 kGain init 16 ;read our knob value kVal serialRead iPort if (kVal != -1) then kGain = kVal/128 endif aSig in ;get our audio input and get its rms kRms rms aSig*kGain kRms = kRms*kRms*255 ;scale the rms to a good value for the LED and send it out serialWrite iPort, (kRms < 255 ? kRms : 255) ;must be in range: 0-255 endin </CsInstruments> <CsScore> f 1 0 1024 10 1 1 1 1 1 1 i 1 0 200 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/scans-2.csd.xml0000644000000000000000000001003312262561504020116 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o scans-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 strset 1, "mary.wav" strset 2, "fox.wav" instr 2 ;show 2 different trajectories, with samples as excitation signal ismp = p6 iamp = p7 itrj = p8 aout soundin p6 ;choose wave file scanu ismp, .01, 6, 2, 33, 44, 5, 2, .01, .05, -.05, .1, .5, 0, 0, aout, 1, 0 asig scans iamp, cpspch(p5), itrj , 0 outs asig, asig endin </CsInstruments> <CsScore> f1 0 128 7 0 64 1 64 0 ; Initial condition f2 0 128 -7 1 128 0.3 ; Masses f33 0 16384 -23 "cylinder-128,8" ; Spring matrices f44 0 128 -7 2 4 0 124 2 ; Centering force f5 0 128 -7 1 128 0 ; Damping f6 0 128 -7 -.0 128 0 ; Initial velocity f7 0 128 -5 .001 128 128 ; Trajectories f77 0 128 -23 "spiral-8,16,128,2,1over2" s i2 0 5 63 6.00 1 .9 7 ;"mary.wav" & i2 6 5 60 7.00 ;trajectory table 7 i2 10 5 60 8.00 s i2 0 5 63 6.00 2 .08 7 ;"fox.wav", at much lower volume i2 6 5 60 7.00 i2 10 5 60 8.00 s i2 0 5 63 6.00 1 .9 77 ;"mary.wav" & i2 6 5 60 7.00 ;trajectory table 77 i2 10 5 60 8.00 s i2 0 5 63 6.00 2 .08 77 ;"fox.wav", at much lower volume i2 6 5 60 7.00 i2 10 5 60 8.00 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/bexprnd.csd.xml0000644000000000000000000000463312262561504020323 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o bexprnd.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values kexp bexprnd 100 printk .2, kexp ; look aout oscili 0.8, 440+kexp, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 kexp bexprnd 100 printk .2, kexp ; look aout oscili 0.8, 440+kexp, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ziwm.csd.xml0000644000000000000000000000547412262561504017653 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ziwm.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a simple instrument. instr 1 ; Add 20.5 to zk variable #1. ziwm 20.5, 1 endin ; Instrument #2 -- another simple instrument. instr 2 ; Add 15.25 to zk variable #1. ziwm 15.25, 1 endin ; Instrument #3 -- prints out zk variable #1. instr 3 ; Read zk variable #1 at i-rate. i1 zir 1 ; Print out the value of zk variable #1. ; It should be 35.75 (20.5 + 15.25) print i1 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 ; Play Instrument #3 for one second. i 3 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outq4.csd.xml0000644000000000000000000000354712262561504017740 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outq4.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 asig vco2 .05, 30 ; sawtooth waveform at low volume kcut line 100, p3, 30 ; Vary cutoff frequency kresonance = 7 inumlayer = 2 asig lowresx asig, kcut, kresonance, inumlayer outq4 asig ; output channel 4 endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/prints.csd.xml0000644000000000000000000000416512262561504020200 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o prints.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Init-time print. prints "%2.3f\\t%!%!%!%!%!%!semicolons! %%\\n", 1234.56789 endin </CsInstruments> <CsScore> /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Play instrument #1. i 1 0 0.004 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/expsegba.csd.xml0000644000000000000000000000450712262561504020457 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o expsega.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Define a short percussive amplitude envelope that ; goes from 0.01 to 20,000 and back. aenv expsegba 0.01, 0.1, 20000, 0.2, 0.01 a1 oscil aenv, 440, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #1 for one second. i 1 1 1 ; Play Instrument #1 for one second. i 1 2 1 ; Play Instrument #1 for one second. i 1 3 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/abs.csd.xml0000644000000000000000000000302312262561504017416 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o abs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ix = p4 iabs = abs(ix) print iabs endin </CsInstruments> <CsScore> i 1 0 1 0 i 1 + 1 -.15 i 1 + 1 -13 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/greaterthan.csd.xml0000644000000000000000000000456212262561504021166 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o greaterthan.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Get the 4th p-field from the score. k1 = p4 ; Is it greater than 3? (1 = true, 0 = false) k2 = (p4 > 3 ? 1 : 0) ; Print the values of k1 and k2. printks "k1 = %f, k2 = %f\\n", 1, k1, k2 endin </CsInstruments> <CsScore> ; Call Instrument #1 with a p4 = 2. i 1 0 0.5 2 ; Call Instrument #1 with a p4 = 3. i 1 1 0.5 3 ; Call Instrument #1 with a p4 = 4. i 1 2 0.5 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mac.csd.xml0000644000000000000000000000646412262561504017425 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mac.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;4 band equalizer klow = p4 ;low gain1 klmid = p5 ;low gain2 kmidh = p6 ;high gain1 khigh = p7 ;high gain2 ifn = p8 ;table ilc1 table 0, ifn ;low freqency range ilc2 table 1, ifn ;low-mid ihc1 table 2, ifn ;mid-high ihc2 table 3, ifn ;high asig diskin2 "fox.wav", 1 alow1 butterlp asig, ilc1 ;lowpass 1 almid butterlp asig, ilc2 ;lowpass 2 amidh butterhp asig, ihc1 ;highpass 1 ahigh butterhp asig, ihc2 ;highpass 2 aout mac klow, alow1, klmid, almid, kmidh, amidh, khigh, ahigh outs aout, aout endin </CsInstruments> <CsScore> f1 0 4 -2 150 300 600 5000 f2 0 4 -2 75 500 1000 10000 f3 0 4 -2 200 700 1500 3000 ; low lowmid midhigh high table i 1 0 2.8 2 1 1 1 1 i 1 3 2.8 2 3 1 1 2 i 1 6 2.8 2 1 2 3 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/inleta.csd.xml0000644000000000000000000002170012262561504020127 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o inleta.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 ; Connect up instruments and effects to create the signal flow graph. connect "SimpleSine", "leftout", "Reverberator", "leftin" connect "SimpleSine", "rightout","Reverberator", "rightin" connect "Moogy", "leftout", "Reverberator", "leftin" connect "Moogy", "rightout","Reverberator", "rightin" connect "Reverberator", "leftout", "Compressor", "leftin" connect "Reverberator", "rightout","Compressor", "rightin" connect "Compressor", "leftout", "Soundfile", "leftin" connect "Compressor", "rightout","Soundfile", "rightin" ; Turn on the "effect" units in the signal flow graph. alwayson "Reverberator", 0.91, 12000 alwayson "Compressor" alwayson "Soundfile" ; Define instruments and effects in order of signal flow. instr SimpleSine ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Default values: p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 pset 0, 0, 10, 0, 0, 0, 0.5 iattack = 0.015 idecay = 0.07 isustain = p3 irelease = 0.3 p3 = iattack + idecay + isustain + irelease adamping linsegr 0.0, iattack, 1.0, idecay + isustain, 1.0, irelease, 0.0 iHz = cpsmidinn(p4) ; Rescale MIDI velocity range to a musically usable range of dB. iamplitude = ampdb(p5 / 127 * 15.0 + 60.0) ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. icosine ftgenonce 0, 0, 65537, 11, 1 aoscili oscili iamplitude, iHz, icosine aadsr madsr iattack, idecay, 0.6, irelease asignal = aoscili * aadsr aleft, aright pan2 asignal, p7 ; Stereo audio output to be routed in the orchestra header. outleta "leftout", aleft outleta "rightout", aright endin instr Moogy ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Default values: p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 pset 0, 0, 10, 0, 0, 0, 0.5 iattack = 0.003 isustain = p3 irelease = 0.05 p3 = iattack + isustain + irelease adamping linsegr 0.0, iattack, 1.0, isustain, 1.0, irelease, 0.0 iHz = cpsmidinn(p4) ; Rescale MIDI velocity range to a musically usable range of dB. iamplitude = ampdb(p5 / 127 * 20.0 + 60.0) print iHz, iamplitude ; Use ftgenonce instead of ftgen, ftgentmp, or f statement. isine ftgenonce 0, 0, 65537, 10, 1 asignal vco iamplitude, iHz, 1, 0.5, isine kfco line 2000, p3, 200 krez = 0.8 asignal moogvcf asignal, kfco, krez, 100000 asignal = asignal * adamping aleft, aright pan2 asignal, p7 ; Stereo audio output to be routed in the orchestra header. outleta "leftout", aleft outleta "rightout", aright endin instr Reverberator ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" idelay = p4 icutoff = p5 aleft, aright reverbsc aleftin, arightin, idelay, icutoff ; Stereo output. outleta "leftout", aleft outleta "rightout", aright endin instr Compressor ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" kthreshold = 25000 icomp1 = 0.5 icomp2 = 0.763 irtime = 0.1 iftime = 0.1 aleftout dam aleftin, kthreshold, icomp1, icomp2, irtime, iftime arightout dam arightin, kthreshold, icomp1, icomp2, irtime, iftime ; Stereo output. outleta "leftout", aleftout outleta "rightout", arightout endin instr Soundfile ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ; Stereo input. aleftin inleta "leftin" arightin inleta "rightin" outs aleftin, arightin endin </CsInstruments> <CsScore> ; It is not necessary to activate "effects" or create f-tables in the score! ; Overlapping notes create new instances of instruments with proper connections. i "SimpleSine" 1 5 60 85 i "SimpleSine" 2 5 64 80 i "Moogy" 3 5 67 75 i "Moogy" 4 5 71 70 ;1 extra second after the performance e 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fprintks.csd.xml0000644000000000000000000000456512262561504020525 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o fprintks.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a score generator example. instr 1 ; K-rate stuff. kstart init 0 kdur linrand 10 kpitch linrand 8 ; Printing to to a file called "my.sco". fprintks "my.sco", "i1\\t%2.2f\\t%2.2f\\t%2.2f\\n", kstart, kdur, 4+kpitch knext linrand 1 kstart = kstart + knext endin </CsInstruments> <CsScore> /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Play Instrument #1. i 1 0 0.001 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/loscil.csd.xml0000644000000000000000000000461612262561504020147 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ichnls = ftchnls(p4) print ichnls if (ichnls == 1) then asigL loscil .8, 1, p4, 1 asigR = asigL elseif (ichnls == 2) then asigL, asigR loscil .8, 1, p4, 1 ;safety precaution if not mono or stereo else asigL = 0 asigR = 0 endif outs asigL, asigR endin </CsInstruments> <CsScore> f 1 0 0 1 "mary.wav" 0 0 0 f 2 0 0 1 "kickroll.wav" 0 0 0 i 1 0 3 1 ;mono file i 1 + 2 2 ;stereo file e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/jspline.csd.xml0000644000000000000000000000405112262561504020317 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o jspline.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kamp init p4 kcpsmin init 2 kcpsmax init 20 ksp jspline kamp, kcpsmin, kcpsmax aout pluck 1, 200+ksp, 1000, 0, 1 aout dcblock aout ;remove DC outs aout, aout endin </CsInstruments> <CsScore> i 1 0 10 2 ;a bit jitter i 1 8 10 10 ;some more i 1 16 10 20 ;lots more e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/rewindscore.csd.xml0000644000000000000000000000445412262561504021206 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o rewindscore.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kenv expon 1, p3, 0.0001 aout poscil .5*kenv, cpspch(p4), 1 outs aout, aout endin instr 100 rewindscore endin </CsInstruments> <CsScore> f1 0 4096 10 1 ;sine wave i1 0 1 8.00 i1 + 1 8.03 i1 + 1 8.04 i1 + 1 8.07 i1 + 1 8.09 i1 + 1 8.10 i1 + 1 8.09 i1 + 1 8.07 i1 + 1 8.03 i100 9 1 ;rewind from 9th second e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/maxtab.csd.xml0000644000000000000000000000132212262561504020125 0ustar rootroot <CsoundSynthesizer> <CsInstruments> instr 1 t1 init 10 t1[3] = 42 k1 maxtab t1 printk2 k1 endin </CsInstruments> <CsScore> i1 0 0.1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/atone.csd.xml0000644000000000000000000000354412262561504017767 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o atone.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;white noise asig rand 1 outs asig, asig endin instr 2 ;filtered noise asig rand 1 khp init 4000 asig atone asig, khp outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 i 2 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/genfarey-2.csd.xml0000644000000000000000000001574012262561504020621 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc for RT audio input as well ; For Non-realtime ouput leave only the line below: ; -o genfarey.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; GENfarey creates table gidelta. ; The table contains the delta values of Farey Sequence 7 (p5=7). ; They are used as Inter Onset Intervals (IOIs) or event durations. ; If p6 is set to 1 for IOI output then the length of the table (p3=-18) is -(|F_7| - 1) ; Remember that a negative sign is for non-power-of-2 table lengths. ; The negative sign in front of the GEN number prevents post-normalisation of its values. gidelta ftgen 0,0,-18,"farey",7,1 ; Use GENfarey with p6 set to 2 to generate the denominators of fractions of F_7 ; this is used in this example as factors to create a series of pitches: gimult ftgen 0,0,-18,"farey",7,2 ;-------- loop and trigger instrument 901 using a Farey Sequence polyrhythm instr 1 kindx init 0 kindx2 init 0 ktrigger init 0 ktime_unit init p6 kstart init p4 kloop init p5 kinitndx init 0 kfn_times init gidelta knote init 60 kbasenote init p8 ifundam init p7 ktrigger seqtime ktime_unit, kstart, kloop, kinitndx, kfn_times if (ktrigger > 0 ) then kpitch = cpspch(ifundam) kmult tab kindx2, gimult kpitch = kpitch * kmult knote = kbasenote + kmult event "i", 901, 0, .4, .10, kpitch, kpitch * .9, 0.4, 5, .75, .8, 1.0, .15, .0, .125, .125, .25, .5, 1.0, .0, .0, .0, .0, .125, .25, .25, .25, knote kindx = kindx + 1 kindx = kindx % kloop kindx2 = kindx2 + 1 kindx2 = kindx2 % kloop endif endin ;------ basic 2 Operators FM algorithm ---------------- instr 901 inotedur = p3 imaxamp = p4 ;ampdb(p4) icarrfreq = p5 imodfreq = p6 ilowndx = p7 indxdiff = p8-p7 knote = p27 aampenv linseg p9, p14*p3, p10, p15*p3, p11, p16*p3, p12, p17*p3, p13 adevenv linseg p18, p23*p3, p19, p24*p3, p20, p25*p3, p21, p26*p3, p22 amodosc oscili (ilowndx+indxdiff*adevenv)*imodfreq, imodfreq, 10 acarosc oscili imaxamp*aampenv, icarrfreq+amodosc, 10 outs acarosc, acarosc endin </CsInstruments> <CsScore> f10 0 4096 10 1 ;sine wave ; p4 kstart := index offset into the Farey Sequence ; p5 kloop := end index into Farey Seq. ; p6 timefac := time in seconds for one loop to complete ; p7 fundam := fundamental of the FM instrument ; p8 basenote:= root pitch of the midi voice output ; note that pitch structures of the midi file output are not equivalent to the ; ones used for the FM real-time synthesis. ; start dur kstart kloop timefac fundam. basenote i1 0.0 44 0 18 2 6.05 60 i1 4 30 0 18 3 7.05 72 i1 34 12 9 18 3 7.05 72 i1 10 12 0 18 1.5 8 84 i1 22 12 0 9 1.5 8 84 i1 15 16 0 18 1 5 48 i1 22 20 5 17 1.7 4 36 i1 46 20 3 11 2.5 7.04 71 i1 51 20 5 13 2.5 7.06 72 i1 73.5 1.5 11 18 1.5 5.05 48 i1 75 1 12 18 1 6.03 58 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pgmassign_advanced.csd.xml0000644000000000000000000001031512262561504022470 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -F pgmassign_advanced.mid ;;;realtime audio out with midifile in ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pgmassign_advanced.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 massign 1, 1 ; channels 1 to 4 use instr 1 by default massign 2, 1 massign 3, 1 massign 4, 1 ; pgmassign_advanced.mid can be found in /manual/examples ; pgmassign.mid has 4 notes with these parameters: ; ; Start time Channel Program ; ; note 1 0.5 1 10 ; note 2 1.5 2 11 ; note 3 2.5 3 12 ; note 4 3.5 4 13 pgmassign 0, 0 ; disable program changes pgmassign 11, 3 ; program 11 uses instr 3 pgmassign 12, 2 ; program 12 uses instr 2 ; waveforms for instruments itmp ftgen 1, 0, 1024, 10, 1 itmp ftgen 2, 0, 1024, 10, 1, 0.5, 0.3333, 0.25, 0.2, 0.1667, 0.1429, 0.125 itmp ftgen 3, 0, 1024, 10, 1, 0, 0.3333, 0, 0.2, 0, 0.1429, 0, 0.10101 instr 1 /* sine */ kcps cpsmidib 2 ; note frequency asnd oscili .6, kcps, 1 outs asnd, asnd endin instr 2 /* band-limited sawtooth */ kcps cpsmidib 2 ; note frequency asnd oscili .6, kcps, 2 outs asnd, asnd endin instr 3 /* band-limited square */ kcps cpsmidib 2 ; note frequency asnd oscili .6, kcps, 3 outs asnd, asnd endin </CsInstruments> <CsScore> t 0 120 f 0 8.5 2 -2 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/getcfg.csd.xml0000644000000000000000000001174112262561504020116 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out -iadc ;;;uncomment -iadc if realtime audio input is needed too </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 S1 getcfg 1 ; -+max_str_len S2 getcfg 2 ; -i S3 getcfg 3 ; -o S4 getcfg 4 ; RTaudio S5 getcfg 5 ; -t S6 getcfg 6 ; os system host S7 getcfg 7 ; callback prints "------------------------------" prints "\nMax string len : " prints S1 prints "\nInput file name (-i) : " prints S2 prints "\nOutput file name (-o) : " prints S3 prints "\nRTaudio (-odac) : " prints S4 prints "\nBeat mode (-t)? : " prints S5 prints "\nHost Op. Sys. : " prints S6 prints "\nCallback ? : " prints S7 prints "\n" prints "------------------------------\n" endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/logcurve.csd.xml0000644000000000000000000000312612262561504020503 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in Silent -odac -iadc -d ;;;realtime output </CsOptions> <CsInstruments> sr = 48000 ksmps = 100 nchnls = 2 instr 1 ; logcurve test kmod phasor 1/p3 kout logcurve kmod, p4 printks "kmod = %f kout = %f\\n", 0.1, kmod, kout endin </CsInstruments> <CsScore> i1 0 10 2 i1 10 10 30 i1 20 10 0.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/insremot.csd.xml0000644000000000000000000000247112262561504020517 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o insremot.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> nchnls = 1 insremot "192.168.1.100", "192.168.1.101", 1 instr 1 aq barmodel 1, 1, p4, 0.001, 0.23, 5, p5, p6, p7 out aq endin </CsInstruments> <CsScore> f0 360 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsarp2.csd.xml0000644000000000000000000000603612262561504020255 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsarp2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "fox.wav" ain1 soundin Sfile1 fftin pvsanal ain1, ifftsize, ioverlap, iwinsize, iwinshape ;make 3 independently moving accentuations in the spectrum kbin1 linseg 0.05, p3/2, .05, p3/2, .05 farp1 pvsarp fftin, kbin1, .9, 10 kbin2 linseg 0.075, p3/2, .1, p3/2, .075 farp2 pvsarp fftin, kbin2, .9, 10 kbin3 linseg 0.02, p3/2, .03, p3/2, .04 farp3 pvsarp fftin, kbin3, .9, 10 ;resynthesize and add them aout1 pvsynth farp1 aout2 pvsynth farp2 aout3 pvsynth farp3 aout = aout1*.3 + aout2*.3 + aout3*.3 out aout endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tone.csd.xml0000644000000000000000000000367512262561504017633 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tone.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 asig diskin2 "beats.wav", 1 outs asig, asig endin instr 2 kton line 10000, p3, 0 ;all the way down to 0 Hz asig diskin2 "beats.wav", 1 asig tone asig, kton ;half-power point at 500 Hz outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 i 2 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pan.csd.xml0000644000000000000000000000504512262561504017435 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pan.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 kcps = p4 k1 phasor kcps ; "fraction" of circle - controls speed of rotation - can be negative k2 tablei k1, 1, 1 ; sin of angle (sinusoid in f1) k3 tablei k1, 1, 1, .25, 1 ; cos of angle (sin offset 1/4 circle) arnd randomi 400, 1000, 50 ; produce random values asig poscil .7, arnd, 1 ; audio signal.. a1,a2,a3,a4 pan asig, k2/2, k3/2, 2, 1, 1 ; sent in a circle (f2=1st quad sin) outq a1, a2, a3, a4 endin </CsInstruments> <CsScore> f1 0 8192 10 1 f2 0 8193 9 .25 1 0 i1 0 10 .2 ;move to the tight i1 11 10 -.2 ;move to the left e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftcps.csd.xml0000644000000000000000000000374212262561504020000 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o ftlen.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the base frequency of Table #1. ; if it has been set in the original file. icps = ftcps(1) print icps endin </CsInstruments> <CsScore> ; Table #1: Use an audio file, Csound will determine its base frequency, if set. f 1 0 0 1 "sample.wav" 0 0 0 ; Play Instrument #1 for 1 second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vaddv.csd.xml0000644000000000000000000001045712262561504017766 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc -nm0 ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 opcode TableDumpSimp, 0, ijo ;prints the content of a table in a simple way ifn, iprec, ippr xin; function table, float precision while printing (default = 3), parameters per row (default = 10, maximum = 32) iprec = (iprec == -1 ? 3 : iprec) ippr = (ippr == 0 ? 10 : ippr) iend = ftlen(ifn) indx = 0 Sformat sprintf "%%.%df\t", iprec Sdump = "" loop: ival tab_i indx, ifn Snew sprintf Sformat, ival Sdump strcat Sdump, Snew indx = indx + 1 imod = indx % ippr if imod == 0 then puts Sdump, 1 Sdump = "" endif if indx < iend igoto loop puts Sdump, 1 endop instr 1 ifn1 = p4 ifn2 = p5 ielements = p6 idstoffset = p7 isrcoffset = p8 kval init 25 vaddv ifn1, ifn2, ielements, idstoffset, isrcoffset, 1 turnoff endin instr 2 TableDumpSimp p4, 3, 16 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 15 16 f 2 0 16 -7 1 15 2 i2 0.0 0.2 1 i2 0.2 0.2 2 i1 0.4 0.01 1 2 5 3 8 i2 0.8 0.2 1 i1 1.0 0.01 1 2 5 10 -2 i2 1.2 0.2 1 i1 1.4 0.01 1 2 8 14 0 i2 1.6 0.2 1 i1 1.8 0.01 1 2 8 0 14 i2 2.0 0.2 1 i1 2.2 0.002 1 1 8 5 2 i2 2.4 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/timedseq.csd.xml0000644000000000000000000001133212262561504020466 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o timedseq.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giseq ftgen 0,0,128,-2, 2, 0, 0.5, 8.00,\ ;first note 2, 1, 0.5, 8.02,\ ;second note 2, 2, 0.5, 8.04,\ ;third 2, 3, 0.5, 8.05,\ ;fourth 2, 4, 0.5, 8.07,\ ;fifth 2, 5, 0.5, 8.09,\ ;sixth 2, 6, 0.5, 8.11,\ ;seventh 2, 7, 0.5, 9.00,\ ;eight note 2, 8, 0.5, 8.00,\ ;due to a quirk in the opcode, it needs an extra note - a copy of the first note -1, 8, -1, -1 ;last line is a dummy event that indicates to timedseq when to loop back to the beginning instr 1 ibeats = 8 ;lengths of sequence in beats itempo = p4 ;tempo iBPS = itempo/60 ;beats per second kphase phasor iBPS/ibeats ;phasor to move through table kpointer = kphase*ibeats ;multiply phase (range 0 - 1) by the number of beats contained within the sequence kp1 init 0 kp2 init 0 kp3 init 0 kp4 init 0 ktrigger timedseq kpointer, giseq, kp1, kp2,kp3, kp4 schedkwhen ktrigger, 0, 0, 2, 0, kp3/abs(iBPS), kp4 ;p3 values have been scaled according to tempo so that they maesure beats rather than seconds endin ;abs(iBPS)(absolute value) is used because the tempo provided by the fourth note of the score is negative. ;Durations here should be positive, because negative values for duration would indicate a held note. instr 2 aenv linseg 0,0.01,1,p3-0.01,0 ;amplitude envelope asig vco2 0.4, cpspch(p4), 4, 0.5 outs asig*aenv, asig*aenv endin </CsInstruments> <CsScore> i 1 0 4 120 i 1 + . 240 i 1 + . 480 i 1 + . -480 ;when negative it plays backwards e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vpow_i-2.csd.xml0000644000000000000000000000411712262561504020320 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vpow_i-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gifil ftgen 1, 0, 0, 1, "fox.wav", 0, 0, 1 instr 1 ival = p4 ;different distortion settings ielements = p5 idstoffset = p6 ;index offset vpow_i 1, ival, ielements, idstoffset asig lposcil 1, 1, 0, 0, 1 outs asig, asig endin </CsInstruments> <CsScore> i1 0 2.7 .5 70000 0 ;no offset i1 3 2.7 .01 50000 70000 ;add another period of distortion, starting at sample 70000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tradsyn.csd.xml0000644000000000000000000000405412262561504020342 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tradsyn.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = p4 ain diskin2 "beats.wav", 1 fs1,fsi2 pvsifd ain,2048,512,1 ; ifd analysis fst partials fs1,fsi2,.003,1,3,500 ; partial tracking aout tradsyn fst, 1, ipch, 500, 1 ; resynthesis outs aout, aout endin </CsInstruments> <CsScore> f1 0 8192 10 1 i 1 0 2 1.5 ;up a 5th i 1 + 2 .5 ;octave down e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen24.csd.xml0000644000000000000000000000462412262561504017600 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen24.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 ;choose between tables kcps init 1/p3 ;create index over duration of note. kndx phasor kcps ixmode = 1 ;normalize to 0-1 kval table kndx, ifn, ixmode asig poscil .7, 440 + kval, 1 ;add to frequency outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave f 10 0 16384 -24 1 0 400;scale sine wave from table 1 from 0 to 400 f 11 0 16384 -24 1 0 50 ;and from 0 to 50 i 1 0 3 10 i 1 4 3 11 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outch.csd.xml0000644000000000000000000000413512262561504020000 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outch.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 5 0dbfs = 1 instr 1 asig vco2 .05, 100 ; sawtooth waveform at low volume kcut line 100, p3, 30 ; Vary cutoff frequency kresonance = .7 inumlayer = 3 asig lowresx asig, kcut, kresonance, inumlayer klfo lfo 4, .5, 4 klfo = klfo+1 ; offset of 1 printks "signal is sent to channel %d\\n", .1, klfo outch klfo,asig endin </CsInstruments> <CsScore> i 1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/diskgrain.csd.xml0000644000000000000000000000514312262561504020631 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o diskgrain.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iolaps = 2 igrsize = 0.04 ifreq = iolaps/igrsize ips = 1/iolaps istr = p4 /* timescale */ ipitch = p5 /* pitchscale */ a1 diskgrain "mary.wav", 1, ifreq, ipitch, igrsize, ips*istr, 1, iolaps outs a1, a1 endin </CsInstruments> <CsScore> f 1 0 8192 20 2 1 ;Hanning function ; timescale pitchscale i 1 0 5 1 1 i 1 + 5 2 1 i 1 + 5 1 0.75 i 1 + 5 1.5 1.5 i 1 + 5 0.5 1.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/trlowest.csd.xml0000644000000000000000000000416012262561504020537 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o trlowest.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ain diskin2 "beats.wav", 1 fs1,fsi2 pvsifd ain, 2048, 512, 1 ; ifd analysis fst partials fs1, fsi2, .003, 1, 3, 500 ; partial tracking flow,kfr,kamp trlowest fst, 1 ; lowest freq-track aout tradsyn flow, 1, 1, 1, 1 ; resynthesis of lowest frequency outs aout*2, aout*2 endin </CsInstruments> <CsScore> f1 0 8192 10 1 ;sine wave i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vibr.csd.xml0000644000000000000000000000351012262561504017614 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vibr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kaverageamp init 500 kaveragefreq init 4 kvib vibr kaverageamp, kaveragefreq, 1 asig poscil .8, 220+kvib, 1 ;add vibrato outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKStifKarp.csd.xml0000644000000000000000000000362312262561504020764 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKStifKarp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p6, p3, p7 ;Pickup Position kv2 = p5 ;String Sustain asig STKStifKarp cpspch(p4), 1, 4, kv1, 11, kv2, 1, 10 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 5.00 0 100 100 i 1 + 40 5.00 127 1 127 i 1 10 32 5.00 127 1 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midipolyaftertouch.csd.xml0000644000000000000000000000602212262561504022566 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midipolyaftertouch.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1;midi keyboard and score are used midinoteoncps p4, p5 ;puts MIDI key translated to cycles per second into p4, and MIDI velocity into p5 kafter init 127 ;full open midipolyaftertouch kafter, 1 printk2 kafter ;display the key value when it changes and when key is pressed kvel = p5/127 ;scale midi velocity to 0-1 kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by ain pluck kenv*kvel, p4, p4, 2, 1 ;velocity value asig moogvcf2 ain, 300+(kafter*100), .5 ;use value of aftertouch to control frequency of filter outs asig, asig ;base freq of filter = 300 Hz endin </CsInstruments> <CsScore> f 0 30 ;runs for 30 seconds f 2 0 4096 10 1 ;sine wave i 1 0 2 440 100 ;play these notes from score as well i 1 + 2 1440 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/Mixer.csd.xml0000644000000000000000000000651212262561504017743 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ; -iadc ;;;RT audio out ; For Non-realtime ouput leave only the line below: ; -o Mixer.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 katt expon 0.01, p3, 1 ;create an attack aout poscil .7, 440,1 MixerSetLevel 1, 3, katt ;impose attack on the gain level MixerSend aout, 1, 3, 0 ;send to channel 0 endin instr 2 aout vco2 .5, 110 ;saw wave MixerSetLevel 2, 3, .25 ;set level to .25 of vco2 MixerSend aout, 2, 3, 1 ;send to channel 1 endin instr 3 ;mix instr.1 and 2 with reverb kgain1 MixerGetLevel 1,3 ;get level form buss 3 kgain2 MixerGetLevel 2,3 ;get level form buss 3 a1 MixerReceive 3,0 ;receive channel 0 a2 MixerReceive 3,1 ;receive channel 1 aout = a1*kgain1+a2*kgain2 ;mix them aoutL, aoutR reverbsc aout, aout, 0.85, 12000 ;add a nice reverb outs aoutL, aoutR MixerClear endin </CsInstruments> <CsScore> f1 0 4096 10 1 i1 0 2 i2 0 2 i3 0 8 ;reverb stays on for 8 sec. e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/locsend_stereo.csd.xml0000644000000000000000000000531212262561504021664 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o locsend_stereo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ga1 init 0 ga2 init 0 instr 1 krevsend = p4 aout diskin2 "beats.wav", 1, 0, 1 kdegree line 0, p3, 180 ;left to right kdistance line 1, p3, 30 a1, a2 locsig aout, kdegree, kdistance, p4 ar1, ar2 locsend ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a2 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 outs a1, a2 ga1 = 0 ga2 = 0 endin </CsInstruments> <CsScore> ; sine wave. f 1 0 16384 10 1 i 1 0 4 .1 ;with reverb i 1 + 4 0 ;no reverb i99 0 7 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dssilist.csd.xml0000644000000000000000000000212012262561504020504 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out -odac </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ; list all DSSI and LADSPA plugins dssilist endin </CsInstruments> <CsScore> i 1 0 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mp3len.csd.xml0000644000000000000000000000331512262561504020053 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o mp3len.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ilen mp3len p4 ;calculate length of mp3 file print ilen asigL, asigR mp3in p4 outs asigL, asigR endin </CsInstruments> <CsScore> i 1 0 30 "XORNOT_jul-14-05.mp3" ; long signal e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftfree.csd.xml0000644000000000000000000000345512262561504020135 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ftfree.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gitempTable ftgen 0, 0, 65537, 10, 1 instr 1 aout oscili .5, 440, gitempTable outs aout, aout ;free temp table at deinit time ftfree gitempTable, 1 print gitempTable endin </CsInstruments> <CsScore> f 0 5 i 1 0 .1 i 1 3 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/inch.csd.xml0000644000000000000000000000402412262561504017574 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -idac ;;;realtime audio I/O ; For Non-realtime ouput leave only the line below: ; inch.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;nchnls channels in, two channels out ain1, ainr inch 1, 2 ;grab your mic and sing adel linseg 0, p3*.5, 0.02, p3*.5, 0 ;max delay time = 20ms aoutl flanger ain1, adel, .7 aoutr flanger ain1, adel*2, .8 fout "in_ch.wav", 14, aoutl, aoutr ;write to stereo file, outs aoutl, aoutr ;16 bits with header endin </CsInstruments> <CsScore> i 1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hrtfstat-2.csd.xml0000644000000000000000000000521512262561504020654 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o hrtfstat-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iAz = p4 iElev = p5 itim = ftlptim(1) ; transeg a dur ty b dur ty c dur ty d kamp transeg 0, p3*.1, 0, .9, p3*.3, -3, .5, p3*.3, -2, 0 ain loscil3 kamp, 50, 1 aleft,aright hrtfstat ain, iAz, iElev, "hrtf-44100-left.dat","hrtf-44100-right.dat" outs aleft, aright endin </CsInstruments> <CsScore> f 1 0 0 1 "Church.wav" 0 0 0 ;Csound computes tablesize ; Azim Elev i1 0 7 90 0 ;to the right i1 3 7 -90 -40 ;to the left and below i1 6 7 180 90 ;behind and up e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dispfft.csd.xml0000644000000000000000000000355012262561504020315 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dispfft.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 110 ifn = 1 knh line p4, p3, p5 asig buzz 1, kcps, knh, ifn outs asig, asig dispfft asig, .1, 2048, 0, 1 endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 3 20 20 i 1 + 3 3 3 i 1 + 3 150 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/strsub.csd.xml0000644000000000000000000000365512262561504020206 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc ;;;-d RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o strsub.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; By: Jonathan Murphy 2007 instr 1 Smember strget p4 ; Parse Smember istrlen strlen Smember idelimiter strindex Smember, ":" S1 strsub Smember, 0, idelimiter ; "String1" S2 strsub Smember, idelimiter + 1, istrlen ; "String2" printf "First string: %s\nSecond string: %s\n", 1, S1, S2 endin </CsInstruments> <CsScore> i 1 0 1 "String1:String2" </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/xscanmap.csd.xml0000644000000000000000000001071212262561504020466 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o xscanmap.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;the matrices can be found in /manual/examples instr 1 ; Plain scanned syntnesis ; note - scanu display is turned off a0 = 0 xscanu 1, .01, 6, 2, "128-stringcircularX", 4, 5, 2, .1, .1, -.01, .1, .5, 0, 0, a0, 0, 0 a1 xscans p4, cpspch(p5), 7, 0, 3 k1,k2 xscanmap 0, 1000, 1000, 64 display k1, .25 ; note - display is updated every second outs a1, a1 endin instr 2 ; Scan synthesis with audio injection and dual scan paths ; note - scanu display is turned off ain diskin2 "fox.wav",1,0,1 ain in a0 = ain/10000 xscanu 1, .01, 6, 2, "128,8-gridX", 14, 5, 2, .01, .05, -.05, .1, .5, 0, 0, a0, 0, 0 a1 xscans p4, cpspch(p5), 7, 0, 2 a2 xscans p4, cpspch(p6), 77, 0, 3 k1,k2 xscanmap 0, 1000, 1000, 127 display k2, .5 ; note - display is updated ever 500ms outs a1,a2 endin </CsInstruments> <CsScore> ; Initial condition ;f1 0 16 7 0 8 1 8 0 f1 0 128 7 0 64 1 64 0 ; Masses f2 0 128 -7 1 128 1 ; Centering force f4 0 128 -7 0 128 2 f14 0 128 -7 2 64 0 64 2 ; Damping f5 0 128 -7 1 128 1 ; Initial velocity f6 0 128 -7 -.0 128 .0 ; Trajectories f7 0 128 -5 .001 128 128 f77 0 128 -23 "128-spiral-8,16,128,2,1over2" ; Sine f9 0 1024 10 1 ;-------------------------------- ; Note list i1 0 10 .9 7.00 s i2 0 10 1 8.00 6.00 i2 0 10 1 7.00 8.05 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cauchy.csd.xml0000644000000000000000000000463512262561504020137 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cauchy.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; every run time same values kalpha cauchy 100 printk .2, kalpha ; look aout oscili 0.8, 440+kalpha, 1 ; & listen outs aout, aout endin instr 2 ; every run time different values seed 0 kalpha cauchy 100 printk .2, kalpha ; look aout oscili 0.8, 440+kalpha, 1 ; & listen outs aout, aout endin </CsInstruments> <CsScore> ; sine wave f 1 0 16384 10 1 i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/balance.csd.xml0000644000000000000000000000402412262561504020240 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o balance.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; Generate a band-limited pulse train. asrc buzz 0.9, 440, sr/440, 1 ; Send the source signal through 2 filters. a1 reson asrc, 1000, 100 a2 reson a1, 3000, 500 ; Balance the filtered signal with the source. afin balance a2, asrc outs afin, afin endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen13.csd.xml0000644000000000000000000002467012262561504017601 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen13.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ;example by Russell Pinkston - Univ. of Texas (but slightly modified) gisine ftgen 0, 0, 16384, 10, 1 ;sine wave instr 1 ihertz = cpspch(p4) ipkamp = p5 iwsfn = p6 ;waveshaping function inmfn = p7 ;normalization function agate linen 1, .01, p3, .1 ;overall amp envelope kctrl linen .99, 2, p3, 2 ;waveshaping index control aindex poscil kctrl/2, ihertz, gisine ;sine wave to be distorted asignal tablei .5+aindex, iwsfn, 1 ;waveshaping knormal tablei kctrl, inmfn, 1 ;amplitude normalization asig = asignal*knormal*ipkamp*agate outs asig, asig endin </CsInstruments> <CsScore> ; This proves the statement in Dodge (p. 147) that Chebyshev polynomials ; of order K have "only the kth harmonic." This is only true when the ; waveshaping index is at the maximum - i.e., when the entire transfer ; function is being accessed. RP. ;-------------------------------------------------------------------------------------------------------------------------------------------- ; quasi sawtooth transfer function: ; h0 h1 h2 h3 h4 h5 h6 h7 h8 h9 h10 h11 h12 h13 h14 h15 h16 h17 h18 h19 h20 f1 0 513 13 1 1 0 100 -50 -33 25 20 -16.7 -14.2 12.5 11.1 -10 -9.09 8.333 7.69 -7.14 -6.67 6.25 5.88 -5.55 -5.26 5 f2 0 257 4 1 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 0 4 6.00 .7 1 2 i1 4 . 7.00 . i1 8 . 8.00 . ;-------------------------------------------------------------------------------------------------------------------------------------------- ; quasi square wave transfer function: ; h0 h1 h2 h3 h4 h5 h6 h7 h8 h9 h10 h11 h12 h13 h14 h15 h16 h17 h18 h19 f3 0 513 13 1 1 0 100 0 -33 0 20 0 -14.2 0 11.1 0 -9.09 0 7.69 0 -6.67 0 5.88 0 -5.26 f4 0 257 4 3 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 16 4 6.00 .7 3 4 i1 20 . 7.00 . i1 24 . 8.00 . ;-------------------------------------------------------------------------------------------------------------------------------------------- ; quasi triangle wave transfer function: ; h0 h1 h2 h3 h4 h5 h6 h7 h8 h9 h10 h11 h12 h13 h14 h15 h16 h17 h18 h19 f5 0 513 13 1 1 0 100 0 -11.11 0 4 0 -2.04 0 1.23 0 -.826 0 .59 0 -.444 0 .346 0 -.277 f6 0 257 4 5 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 32 4 6.00 .7 5 6 i1 36 . 7.00 . i1 40 . 8.00 . ;-------------------------------------------------------------------------------------------------------------------------------------------- ; transfer function1: h0 h1 h2 h3 h4 h5 h6 h7 h8 h9 h10 h11 h12 h13 h14 h15 h16 f7 0 513 13 1 1 0 1 -.8 0 .6 0 0 0 .4 0 0 0 0 .1 -.2 -.3 .5 f8 0 257 4 7 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 48 4 5.00 .7 7 8 i1 52 . 6.00 . i1 56 . 7.00 . ;-------------------------------------------------------------------------------------------------------------------------------------------- ;=========================================================================; ; This demonstrates the use of high partials, sometimes without a ; ; fundamental, to get quasi-inharmonic spectra from waveshaping. ; ;=========================================================================; ; transfer function2: h0 h1 h2 h3 h4 h5 h6 h7 h8 h9 h10 h11 h12 h13 h14 h15 h16 f9 0 513 13 1 1 0 0 0 -.1 0 .3 0 -.5 0 .7 0 -.9 0 1 0 -1 0 f10 0 257 4 9 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 64 4 5.00 .7 9 10 i1 68 . 6.00 . i1 72 . 7.00 . ;-------------------------------------------------------------------------------------------------------------------------------------------- ; transfer function3: h0 h1 h2 h3 h4 h5 h6 h7 h8 h9 h10 h11 h12 h13 h14 h15 h16 h17 h18 h19 h17 h18 h19 h20 f11 0 513 13 1 1 0 0 0 0 0 0 0 -1 0 1 0 0 -.1 0 .1 0 -.2 .3 0 -.7 0 .2 0 -.1 f12 0 257 4 11 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 80 4 5.00 .7 11 12 i1 84 . 5.06 . i1 88 . 6.00 . ;-------------------------------------------------------------------------------------------------------------------------------------------- ;=========================================================================; ; split a sinusoid into 3 odd-harmonic partials of relative strength 5:3:1 ;=========================================================================; ;-------------------------------------------------------------------------------------------------------------------------------------------- ; transfer function4: h0 h1 h2 h3 h4 h5 f13 0 513 13 1 1 0 5 0 3 0 1 f14 0 257 4 13 1 ; normalizing function with midpoint bipolar offset ; st dur pch amp wsfn nmfn i1 96 4 5.00 .7 13 14 i1 100 . 5.06 . i1 104 . 6.00 . e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKClarinet.csd.xml0000644000000000000000000000355012262561504021001 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKclarinet.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv5 = p5 ;breath pressure kv1 line p6, p3, p7 ;reed stiffness asig STKClarinet cpspch(p4), 1, 2, kv1, 4, 100, 11, 60, 1, 10, 128, kv5 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 3 8.00 100 127 10 i 1 + 10 8.08 80 60 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/delay1.csd.xml0000644000000000000000000000452012262561504020033 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o delay.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Instrument #1 -- Silence on one channel instr 1 ; Make a basic sound. abeep vco 20000, 440, 1 ; Delay the beep by 1 sample. idlt = 1/sr adel delay abeep, idlt adel1 delay1 abeep ; Send the beep to the left speaker and ; the difference in the delayes to the right speaker. outs abeep, adel-adel1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1. i 1 0.0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/mincer.csd.xml0000644000000000000000000000402412262561504020130 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o mincer.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = p3 ilock = p4 ipitch = 1 itimescale = 0.5 iamp = 0.8 atime line 0,idur,idur*itimescale asig mincer atime, iamp, ipitch, 1, ilock outs asig, asig endin </CsInstruments> <CsScore> f 1 0 0 1 "fox.wav" 0 0 0 i 1 0 5 0 ;not locked i 1 6 5 1 ;locked e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tableseg.csd.xml0000644000000000000000000000377712262561504020457 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tableseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; analyze "fox.wav" with PVANAL first iend = p4 ktime line 0, p3, iend tableseg p5, p3, p6 ;morph from table 1 asig vpvoc ktime, 1, "fox.pvx" ;to table 2 outs asig*3, asig*3 endin </CsInstruments> <CsScore> f 1 0 512 9 .5 1 0 f 2 0 512 7 0 20 1 30 0 230 0 232 1 i 1 0 10 2.7 1 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pinkish.csd.xml0000644000000000000000000000361112262561504020321 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pinkish.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 awhite unirand 2.0 ; Normalize to +/-1.0 awhite = awhite - 1.0 apink pinkish awhite, 1, 0, 0, 1 out apink * 30000 endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hrtferstat.csd.xml0000644000000000000000000000422312262561504021042 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select flags here ; realtime audio out ; -o dac ; For Non-realtime ouput leave only the line below: ;-o hrtf.wav </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 gasrc init 0 instr 1 ;a plucked string kamp = p4 kcps = cpspch(p5) icps = cpspch(p5) a1 pluck kamp, kcps, icps, 0, 1 gasrc = a1 endin instr 10 ;uses output from instr1 as source aleft,aright hrtfstat gasrc, 90,0, "hrtf-44100-left.dat","hrtf-44100-right.dat" outs aleft, aright endin </CsInstruments> <CsScore> ; Play Instrument 1: a plucked string i 1 0 2 20000 8.00 ; Play Instrument 10 for 2 seconds. i 10 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/transegr.csd.xml0000644000000000000000000000375412262561504020511 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o transegr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .2 ; st,dur1,typ1,val,dur2,typ2,end kenv transegr 0, .2, 2, .5, 1, - 3, 0 asig pluck kenv*iamp, icps, icps, 1, 1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 4096 10 1 ;sine f0 30 ;runs 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tanh.csd.xml0000644000000000000000000000407412262561504017612 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o tanh.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 0dbfs = 1 ; Instrument #1. instr 1 asig1 vco 1, 440, 2, 0.4, 1 asig2 vco 1, 800, 3, 0.5, 1 asig = asig1+asig2 ;; will go out of range out tanh(asig) ;; but tanh is a limiter endin </CsInstruments> <CsScore> f1 0 65536 10 1 ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fractalnoise.csd.xml0000644000000000000000000000333612262561504021332 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kbeta linseg 0, p3/4, 2, p3/4, 0, p3*0.1, 2, p3*0.15, 0 seed 20120124 aout fractalnoise 0.05, kbeta outs aout, aout endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/tival.csd.xml0000644000000000000000000000634012262561504017775 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o tival.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idur = abs(p3) ;make p3 positive even if p3 is negative in score itiv tival i1 = -1 ;assume this is tied note, so keep fase of oscili tigoto slur ;no reinitialisation on tied notes i1 = 0 ;first note, so reset phase aatt line p4, idur, 0 ;primary envelope slur: if itiv==0 kgoto note ;no expression on first and second note aslur linseg 0, idur*.3, p4, idur*.7, 0 ;envelope for slurred note aatt = aatt + aslur note: asig oscili aatt, p5, 1, i1 outs asig, asig endin </CsInstruments> <CsScore> f1 0 4096 10 1 ;sine wave i1 0 -5 .8 451 ;p3 = 5 seconds i1 1.5 -1.5 .1 512 i1 3 2 .7 440 ;3 notes together--> duration = 5 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/toot4.csd.xml0000644000000000000000000000504312262561504017726 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> instr 4 iamp = ampdb(p4) ; convert decibels to linear amp iscale = iamp * .333 ; scale the amp at initialization inote = cpspch(p5) ; convert octave.pitch to cps k1 linen iscale, p6, p3, p7 ; p4=amp a3 oscil k1, inote*.996, 1 ; p5=freq a2 oscil k1, inote*1.004, 1 ; p6=attack time a1 oscil k1, inote, 1 ; p7=release time a1 = a1+a2+a3 out a1 endin </CsInstruments> <CsScore> f1 0 4096 10 1 ; sine wave ;ins strt dur amp freq attack release i4 0 1 75 8.04 0.1 0.7 i4 1 1 70 8.02 0.07 0.6 i4 2 1 75 8.00 0.05 0.5 i4 3 1 70 8.02 0.05 0.4 i4 4 1 85 8.04 0.1 0.5 i4 5 1 80 8.04 0.05 0.5 i4 6 2 90 8.04 0.03 1. </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gendyx-2.csd.xml0000644000000000000000000000371312262561504020314 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kenv expseg 0.001, p3*0.05, 0.5, p3*0.9, 0.5, p3*0.05, 0.001 kc1 linseg 1, p3/2, 12, p3/2, 3 kc2 random 0, 4 seed 20120124 asig gendyx kenv, 1, 3, 0.7, 0.8, 120, 4300, 0.2, 0.7, kc1, kc2, 12, kc1 aout dcblock asig outs aout, aout endin </CsInstruments> <CsScore> i1 0 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/rspline.csd.xml0000644000000000000000000000417012262561504020331 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o jspline.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 krangeMin init p4 krangeMax init p5 kcpsmin init 2 kcpsmax init 3 ksp rspline krangeMin, krangeMax, kcpsmin, kcpsmax aout pluck 1, 200+ksp, 1000, 0, 1 aout dcblock aout ;remove DC outs aout, aout endin </CsInstruments> <CsScore> i 1 0 10 2 5 ;a bit jitter i 1 8 10 10 20 ;some more i 1 16 10 20 50 ;lots more e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/guiro.csd.xml0000644000000000000000000000277312262561504020011 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o guiro.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 a1 guiro .8, p4 outs a1, a1 endin </CsInstruments> <CsScore> i1 0 1 1 i1 + 1 .01 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/slicearray.csd.xml0000644000000000000000000000662112262561504021016 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;create and fill an array kArr[] genarray_i 1, 9 ;print the content printf "%s", 1, "kArr = whole array\n" kndx = 0 until kndx == lenarray(kArr) do printf "kArr[%d] = %f\n", kndx+1, kndx, kArr[kndx] kndx += 1 od ;build new arrays for the slices kArr1[] init 5 kArr2[] init 4 ;put in first five and last four elements kArr1 slicearray kArr, 0, 4 kArr2 slicearray kArr, 5, 8 ;print the content printf "%s", 1, "\nkArr1 = slice from index 0 to index 4\n" kndx = 0 until kndx == lenarray(kArr1) do printf "kArr1[%d] = %f\n", kndx+1, kndx, kArr1[kndx] kndx += 1 od printf "%s", 1, "\nkArr2 = slice from index 5 to index 8\n" kndx = 0 until kndx == lenarray(kArr2) do printf "kArr2[%d] = %f\n", kndx+1, kndx, kArr2[kndx] kndx += 1 od turnoff endin </CsInstruments> <CsScore> i 1 0 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midinoteoncps.csd.xml0000644000000000000000000000470712262561504021536 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M1 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o midinoteoncps.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 midinoteoncps p4, p5 ;puts MIDI key translated to cycles per second into p4, and MIDI velocity into p5 print p4 ;display the key value when it changes and when key is pressed kvel = p5/127 ;scale midi velocity to 0-1 kenv madsr 0.5, 0.8, 0.8, 0.5 ;amplitude envelope multiplied by asig pluck kenv*kvel, p4, p4, 2, 1 ;velocity value outs asig, asig endin </CsInstruments> <CsScore> f 0 30 ;runs for 30 seconds f 2 0 4096 10 1 i 1 0 2 440 100 ; play these notes from score as well i 1 + 2 1440 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLsetVal.csd.xml0000644000000000000000000000444412262561504020341 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLsetVal.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 441 ksmps = 100 nchnls = 1 ; By Andres Cabrera 2007 FLpanel "Frequency Slider", 340, 100, 50, 50 gkvalue1, gihandle1 FLslider "Slider A", 200, 5000, -1, 1, -1, 320, 20, 10, 10 gkvalue2, gihandle2 FLslider "Slider B", 200, 5000, -1, 1, -1, 320, 20, 10, 50 ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun ;Set the widget's initial value FLsetVal_i 300, gihandle1 instr 1 endin </CsInstruments> <CsScore> ; Function table that defines a single cycle ; of a sine wave. f 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKWhistle.csd.xml0000644000000000000000000000406712262561504020663 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKWhistle.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = p4 kv1 line p5, p3, p6 ;Blowing Frequency Modulation kv3 = p7 ;Fipple Modulation Frequency asig STKWhistle cpspch(p4), 1, 4, 20, 11, kv3, 1, 100, 2, kv1, 128, 127 asig = asig*.7 ;too loud outs asig, asig endin </CsInstruments> <CsScore> i 1 0 .5 9.00 100 30 30 i 1 1 3 9.00 100 0 20 i 1 4.5 . 9.00 1 0 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/minaccum.csd.xml0000644000000000000000000000517612262561504020460 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o minaccum.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;saw asig vco2 .2, p4 outs asig, asig gasaw = asig endin instr 2 ;sine aout poscil .3, p4, 1 outs aout, aout gasin = aout endin instr 10 accum init 0 minaccum accum, gasaw + gasin ;saw and sine accumulated outs accum, accum clear accum endin </CsInstruments> <CsScore> f 1 0 4096 10 1 i 1 0 7 330 i 2 3 3 440 i 1 10 7 330 ;same notes but without minaccum, for comparison i 2 13 3 440 i 10 0 6 ;accumulation note stops after 6 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ptrack.csd.xml0000644000000000000000000000435312262561504020144 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ptrack.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ihop = p4 aout diskin2 "fox.wav",1, 0, 1 kf,ka ptrack aout, ihop ; pitch track with different hopsizes kcps port kf, 0.01 ; smooth freq kamp port ka, 0.01 ; smooth amp ; drive an oscillator asig poscil ampdb(kamp)*0dbfs, kcps, 1 outs asig, asig endin </CsInstruments> <CsScore> ; simple sine wave f 1 0 4096 10 1 i 1 0 5 128 i 1 6 5 512 i 1 12 5 1024 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/chnset.csd.xml0000644000000000000000000001154612262561504020146 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o chnset.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;Example by Joachim Heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1; send i-values chnset 1, "sio" chnset -1, "non" endin instr 2; send k-values kfreq randomi 100, 300, 1 chnset kfreq, "cntrfreq" kbw = kfreq/10 chnset kbw, "bandw" endin instr 3; send a-values anois rand .1 chnset anois, "noise" loop: idur random .3, 1.5 timout 0, idur, do reinit loop do: ifreq random 400, 1200 iamp random .1, .3 asig oscils iamp, ifreq, 0 aenv transeg 1, idur, -10, 0 asine = asig * aenv chnset asine, "sine" endin instr 11; receive some chn values and send again ival1 chnget "sio" ival2 chnget "non" print ival1, ival2 kcntfreq chnget "cntrfreq" kbandw chnget "bandw" anoise chnget "noise" afilt reson anoise, kcntfreq, kbandw afilt balance afilt, anoise chnset afilt, "filtered" endin instr 12; mix the two audio signals amix1 chnget "sine" amix2 chnget "filtered" chnmix amix1, "mix" chnmix amix2, "mix" endin instr 20; receive and reverb amix chnget "mix" aL, aR freeverb amix, amix, .8, .5 outs aL, aR endin instr 100; clear chnclear "mix" endin </CsInstruments> <CsScore> i 1 0 20 i 2 0 20 i 3 0 20 i 11 0 20 i 12 0 20 i 20 0 20 i 100 0 20 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/farey7shuffled.csd.xml0000644000000000000000000001351212262561504021573 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac -+rtaudio=alsa --midioutfile=farey7.mid </CsOptions> <CsInstruments> sr=48000 ksmps=10 nchnls=1 0dbfs = 1 gidelta init 100 gimult init 101 ;-------- loop and trigger instrument 901 using a Farey Sequence polyrhythm instr 1 kindx init 0 kindx2 init 0 ktrigger init 0 ktime_unit init p6 kstart init p4 kloop init p5 kinitndx init 0 kfn_times init gidelta knote init 60 kbasenote init p8 ifundam init p7 ktrigger seqtime ktime_unit, kstart, kloop, kinitndx, kfn_times if (ktrigger > 0 ) then kpitch = cpspch(ifundam) kmult tab kindx2, gimult kpitch = kpitch * kmult knote = kbasenote + kmult event "i", 901, 0, .4, .1, kpitch, kpitch * .9, .4, 5, .75, .8, 1.0, .15, .0, .125, .125, .25, .5, 1.0, .0, .0, .0, .0, .125, .25, .25, .25, knote kindx = kindx + 1 kindx = kindx % kloop kindx2 = kindx2 + 1 kindx2 = kindx2 % kloop if (kindx2 == 0) then tableshuffle gimult endif endif endin ; 1 ;------ basic 2 Operators FM algorithm ---------------- instr 901 inotedur = p3 imaxamp = p4 ;ampdb(p4) icarrfreq = p5 imodfreq = p6 ilowndx = p7 indxdiff = p8-p7 knote = p27 aampenv linseg p9, p14*p3, p10, p15*p3, p11, p16*p3, p12, p17*p3, p13 adevenv linseg p18, p23*p3, p19, p24*p3, p20, p25*p3, p21, p26*p3, p22 amodosc oscili (ilowndx+indxdiff*adevenv)*imodfreq, imodfreq, 10 acarosc oscili imaxamp*aampenv, icarrfreq+amodosc, 10 out acarosc ;------ we also write down a midi track here ---------- midion 1, knote, 100 endin ; 901 </CsInstruments> <CsScore> f10 0 4096 10 1 f100 0 -18 "farey" 7 1 f101 0 -18 "farey" 7 2 ; p4 kstart := index offset into the Farey Sequence ; p5 kloop := end index into Farey Seq. ; p6 timefac := time in seconds for one loop to complete ; p7 fundam := fundamental of the FM instrument ; p8 basenote:= root pitch of the midi voice output ; note that pitch structures of the midi file output are not equivalent to the ; ones used for the FM real-time synthesis. ; start dur kstart kloop timefac fundam. basenote i1 0.0 44 0 18 1 6.05 60 i1 4 40 0 18 3 7.05 72 i1 10 38 0 18 1.5 8 84 i1 15 50 0 18 1 5 48 i1 22 75 5 17 1.7 4 36 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/passign.csd.xml0000644000000000000000000000504312262561504020321 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o passign.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 8 idur,iamp,iskiptime,iattack,irelease,irvbtime,irvbgain passign 3 kamp linen iamp, iattack, idur, irelease asig soundin "fox.wav", iskiptime arampsig = kamp * asig aeffect reverb asig, irvbtime arvbretrn = aeffect * irvbgain ;mix dry & wet signals outs arampsig + arvbretrn, arampsig + arvbretrn endin </CsInstruments> <CsScore> ;ins strt dur amp skip atk rel rvbt rvbgain i8 0 4 .3 0 .03 .1 1.5 .3 i8 4 4 .3 1.6 .1 .1 3.1 .7 i8 8 4 .3 0 .5 .1 2.1 .2 i8 12 4 .4 0 .01 .1 1.1 .1 i8 16 4 .5 0.1 .01 .1 0.1 .1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midichn.csd.xml0000644000000000000000000000250012262561504020263 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M0 -+rtmidi=virtual ;;; midi file input </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;change channel on virtual midi keyboard i1 midichn print i1 endin </CsInstruments> <CsScore> f 0 20 ;runs for 20 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/instance.csd.xml0000644000000000000000000000460712262561504020466 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o instance.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - oscillator with a high note. instr 1 ; Play Instrument #2 at the same time. iHandleinstance 2, 0, p3 ; Play a high note. a1 oscils 10000, 880, 1 out a1 endin ; Instrument #2 - oscillator with a low note. instr 2 ; Play a low note. a1 oscils 10000, 220, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for half a second. i 1 0 0.5 ; Play Instrument #1 for half a second. i 1 1 0.5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pdclip.csd.xml0000644000000000000000000000436012262561504020131 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o abs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; test instrument for pdclip opcode instr 3 idur = p3 iamp = p4 ifreq = p5 ifn = p6 kenv linseg 0, .05, 1.0, idur - .1, 1.0, .05, 0 aosc oscil 1.0, ifreq, ifn kmod expseg 0.00001, idur, 1.0 aout pdclip aosc, kmod, 0.0, 1.0 out kenv*aout*iamp endin </CsInstruments> <CsScore> f1 0 16385 10 1 f2 0 16385 10 1 .5 .3333 .25 .5 ; pdclipped sine wave i3 0 3 15000 440 1 i3 + 3 15000 330 1 i3 + 3 15000 220 1 s ; pdclipped composite wave i3 0 3 15000 440 2 i3 + 3 15000 330 2 i3 + 3 15000 220 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/lposcil3.csd.xml0000644000000000000000000000425012262561504020404 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lposcil3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = 1.5 ; a fifth up kloop = 0 ; loop start time (in samples) kend line 45000, p3, 10000 ; vary loop end time (in samples) asig lposcil3 1, kcps, kloop, kend, 1 outs asig, asig endin </CsInstruments> <CsScore> ; Its table size is deferred, ; and format taken from the soundfile header. f 1 0 0 1 "beats.wav" 0 0 0 ; Play Instrument #1 for 6 seconds. ; This will loop the drum pattern several times. i 1 0 6 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vibrato.csd.xml0000644000000000000000000000450012262561504020320 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vibrato.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kaverageamp init .5 kaveragefreq init 5 krandamountamp line p4, p3, p5 ;increase random amplitude of vibrato krandamountfreq init .3 kampminrate init 3 kampmaxrate init 5 kcpsminrate init 3 kcpsmaxrate init 5 kvib vibrato kaverageamp, kaveragefreq, krandamountamp, krandamountfreq, kampminrate, kampmaxrate, kcpsminrate, kcpsmaxrate, 1 asig poscil .8, 220+kvib, 1 ;add vibrato outs asig, asig endin </CsInstruments> <CsScore> f 1 0 16384 10 1 ;sine wave i 1 0 15 .01 20 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/bqrez.csd.xml0000644000000000000000000000455312262561504020005 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o bqrez.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;sawtooth waveform. kfco line 200, p3, 2000;filter-cutoff frequency from .2 to 5 KHz. kres = p4 ;resonance imode = p5 ;mode asig vco 0.2, 220, 1 afilt bqrez asig, kfco, kres, imode asig balance afilt, asig outs asig, asig endin </CsInstruments> <CsScore> ;sine wave f 1 0 16384 10 1 i 1 0 3 1 0 ; low pass i 1 + 3 30 0 ; low pass i 1 + 3 1 1 ; high pass i 1 + 3 30 1 ; high pass e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dumpk-2.csd.xml0000644000000000000000000000454312262561504020140 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dumpk-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes a control signal to a file kfreq randh 100, 1, 2, 1, 500 ;generates one random number between 400 and 600 per second dumpk kfreq, "dumpk.txt", 8, 1 ;writes the control signal printk 1, kfreq ;prints it endin instr 2 ;reads the file written by instr 1 kfreq readk "dumpk.txt", 8, 1 printk 1, kfreq ;prints it aout poscil .2, kfreq, giSine outs aout, aout endin </CsInstruments> <CsScore> i 1 0 5 i 2 5 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/alpass.csd.xml0000644000000000000000000000446712262561504020151 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o alpass.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 100 nchnls = 2 0dbfs = 1 gamix init 0 instr 1 acps expon p4, p3, p5 asig vco 0.6, acps, 1 outs asig, asig gamix = gamix + asig endin instr 99 arvt1 line 3.5*1.5, p3, 6 arvt2 line 3.5, p3, 4 ilpt = 0.1 aleft alpass gamix, arvt1, ilpt aright alpass gamix, arvt2, ilpt*2 outs aleft, aright gamix = 0 ; clear mixer endin </CsInstruments> <CsScore> f1 0 4096 10 1 i 1 0 3 20 2000 i 99 0 8 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/peak.csd.xml0000644000000000000000000000365612262561504017605 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o peak.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1 - play an audio file. instr 1 ; Capture the highest amplitude in the "beats.wav" file. asig soundin "beats.wav" kp peak asig ; Print out the peak value once per second. printk 1, kp out asig endin </CsInstruments> <CsScore> ; Play Instrument #1, the audio file, for three seconds. i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen04.csd.xml0000644000000000000000000001007712262561504017575 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen04.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisine ftgen 0, 0, 16384, 10, 1 ;sine wave instr 1 ihertz = cpspch(p4) ipkamp = p5 iwsfn = p6 ;waveshaping function inmfn = p7 ;normalization function agate linen 1, .01, p3, .1 ;overall amp envelope kctrl linen .9, 2, p3, 2 ;waveshaping index control aindex poscil kctrl/2, ihertz, gisine ;sine wave to be distorted asignal tablei .5+aindex, iwsfn, 1 ;waveshaping knormal tablei 1/kctrl, inmfn , 1 ;amplitude normalization outs asignal*knormal*ipkamp*agate, asignal*knormal*ipkamp*agate endin </CsInstruments> <CsScore> f1 0 64 21 6 ;Gaussian (random) distribution f2 0 33 4 1 1 ;normalizing function with midpoint bipolar offset s ; st dur pch amp wsfn inmfn i1 0 4 6.00 .7 1 2 i1 4 . 7.00 . i1 8 . 8.00 . ;------------------------------------------------------------------------------------- f3 0 1025 13 1 1 0 5 0 5 0 10 ;Chebyshev algorithm f4 0 513 4 3 1 ;normalizing function with midpoint bipolar offset s ; st dur pch amp wsfn inmfn i1 0 4 6.00 .9 3 4 i1 4 . 7.00 . i1 8 . 8.00 . e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sfplay3m.csd.xml0000644000000000000000000000533312262561504020415 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfplay3m.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf sfload "07AcousticGuitar.sf2" sfplist gisf sfpassign 10, gisf instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/7000 ;scale amplitude kfreq init 1 ;do not change freq from sf aout sfplay3m ivel, inum, kamp*ivel, kfreq, 10 ;preset index = 10 outs aout, aout endin </CsInstruments> <CsScore> f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pitchamdf.csd.xml0000644000000000000000000000470112262561504020614 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o pitchamdf.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;clean audio asig soundin p4 outs asig, asig endin instr 2 ;use pitch asig soundin p4 asig tone asig, 1000 ;lowpass-filter kcps, krms pitchamdf asig, 100, 500, 200 asig poscil krms, kcps, 1 ;re-synthesize with sawtooth outs asig, asig endin </CsInstruments> <CsScore> f1 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ;sawtooth i 1 0 3 "fox.wav" i 2 3 3 "fox.wav" i 1 6 3 "mary.wav" i 2 9 3 "mary.wav" i 1 12 3 "beats.wav" i 2 15 3 "beats.wav" e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/equals.csd.xml0000644000000000000000000000460412262561504020151 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o equals.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ienv = p4 ;choose envelope in score if (ienv == 0) kthen kenv adsr 0.05, 0.05, 0.95, 0.05 ;sustained envelope elseif (ienv == 1) kthen kenv adsr 0.5, 1, 0.5, 0.5 ;triangular envelope elseif (ienv == 2) kthen kenv adsr 1, 1, 1, 0 ;ramp up endif aout vco2 .1, 110, 10 aout = aout * kenv outs aout, aout endin </CsInstruments> <CsScore> i1 0 2 0 i1 3 2 1 i1 6 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hrtfer.csd.xml0000644000000000000000000000435712262561504020156 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o hrtfer.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 kaz linseg 0, p3, -360 ; move the sound in circle kel linseg -40, p3, 45 ; around the listener, changing ; elevation as its turning asrc soundin "beats.wav" aleft,aright hrtfer asrc, kaz, kel, "HRTFcompact" aleftscale = aleft * 200 arightscale = aright * 200 outs aleftscale, arightscale endin </CsInstruments> <CsScore> i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/marimba.csd.xml0000644000000000000000000000621612262561504020270 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o marimba.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 ksmps = 128 nchnls = 2 ; Instrument #1. instr 1 ifreq = cpspch(p4) ihrd = 0.1 ipos = 0.561 imp = 1 kvibf = 6.0 kvamp = 0.05 ivibfn = 2 idec = 0.6 a1 marimba 20000, ifreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec, 20, 10 outs a1, a1 endin </CsInstruments> <CsScore> ; Table #1, the "marmstk1.wav" audio file. f 1 0 256 1 "marmstk1.wav" 0 0 0 ; Table #2, a sine wave for the vibrato. f 2 0 128 10 1 ; Play Instrument #1 for one second. i 1 0 1 8.09 i 1 + 0.5 8.00 i 1 + 0.5 7.00 i 1 + 0.25 8.02 i 1 + 0.25 8.01 i 1 + 0.25 7.09 i 1 + 0.25 8.02 i 1 + 0.25 8.01 i 1 + 0.25 7.09 i 1 + 0.3333 8.09 i 1 + 0.3333 8.02 i 1 + 0.3334 8.01 i 1 + 0.25 8.00 i 1 + 0.3333 8.09 i 1 + 0.3333 8.02 i 1 + 0.25 8.01 i 1 + 0.3333 7.00 i 1 + 0.3334 6.00 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/toot2.csd.xml0000644000000000000000000000254312262561504017726 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> instr 2 a1 oscil p4, p5, 1 ; p4=amp out a1 ; p5=freq endin </CsInstruments> <CsScore> f1 0 4096 10 1 ; sine wave ;ins strt dur amp(p4) freq(p5) i2 0 1 2000 880 i2 1.5 1 4000 440 i2 3 1 8000 523.251 i2 4.5 1 16000 659.255 i2 6 1 32000 783.991 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gain.csd.xml0000644000000000000000000000423612262561504017576 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gain.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asrc buzz .8, 440, sr/440, 1 ; band-limited pulse train. a1 reson asrc, 1000, 100 ; Sent through a2 reson a1, 3000, 500 ; 2 filters krms rms asrc ; then balanced afin gain a2, krms ; with source outs afin, afin endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/linenr.csd.xml0000644000000000000000000001126512262561504020147 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc -M0 ;;;RT audio I/O with MIDI in </CsOptions> <CsInstruments> ; Example by Jonathan Murphy and Charles Gran 2007 sr = 44100 ksmps = 10 nchnls = 2 ; new, and important. Make sure that midi note events are only ; received by instruments that actually need them. ; turn default midi routing off massign 0, 0 ; route note events on channel 1 to instr 1 massign 1, 1 ; Define your midi controllers #define C1 #21# #define C2 #22# #define C3 #23# ; Initialize MIDI controllers initc7 1, 21, 0.5 ;delay send initc7 1, 22, 0.5 ;delay: time to zero initc7 1, 23, 0.5 ;delay: rate gaosc init 0 ; Define an opcode to "smooth" the MIDI controller signal opcode smooth, k, k kin xin kport linseg 0, 0.0001, 0.01, 1, 0.01 kin portk kin, kport xout kin endop instr 1 ; Generate a sine wave at the frequency of the MIDI note that triggered the intrument ifqc cpsmidi iamp ampmidi 10000 aenv linenr iamp, .01, .1, .01 ;envelope a1 oscil aenv, ifqc, 1 ; All sound goes to the global variable gaosc gaosc = gaosc + a1 endin instr 198 ; ECHO kcmbsnd ctrl7 1, 21, 0, 1 ;delay send ktime ctrl7 1, 22, 0.01, 6 ;time loop fades out kloop ctrl7 1, 23, 0.01, 1 ;loop speed ; Receive MIDI controller values and then smooth them kcmbsnd smooth kcmbsnd ktime smooth ktime kloop smooth kloop imaxlpt = 1 ;max loop time ; Create a variable reverberation (delay) of the gaosc signal acomb vcomb gaosc, ktime, kloop, imaxlpt, 1 aout = (acomb * kcmbsnd) + gaosc * (1 - kcmbsnd) outs aout, aout gaosc = 0 endin </CsInstruments> <CsScore> f1 0 16384 10 1 i198 0 10000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/limit.csd.xml0000644000000000000000000000656612262561504020006 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o limit.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 ; Limit / Mirror / Wrap igain = p4 ;gain ilevl1 = p5 ; + level ilevl2 = p6 ; - level imode = p7 ;1 = limit, 2 = mirror, 3 = wrap ain soundin "fox.wav" ain = ain*igain if imode = 1 goto limit if imode = 2 goto mirror asig wrap ain, ilevl2, ilevl1 goto outsignal limit: asig limit ain, ilevl2, ilevl1 goto outsignal mirror: asig mirror ain, ilevl2, ilevl1 outsignal: outs asig*.5, asig*.5 ;mind your speakers endin </CsInstruments> <CsScore> ; Gain +Levl -Levl Mode i1 0 3 4.00 .25 -1.00 1 ;limit i1 4 3 4.00 .25 -1.00 2 ;mirror i1 8 3 4.00 .25 -1.00 3 ;wrap e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vtablewa.csd.xml0000644000000000000000000000245112262561504020462 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac -b441 -B441 </CsOptions> <CsInstruments> sr=44100 kr=4410 ksmps=10 nchnls=2 0dbfs = 1 instr 1 vcopy 2, 1, 262144 ar random 0, 1 vtablewa ar,2,1,ar out ar,ar endin </CsInstruments> <CsScore> f1 0 262144 -1 "beats.wav" 0 4 0 f2 0 262144 2 0 i1 0 4 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/opa.csd.xml0000644000000000000000000000415512262561504017437 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o a.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; hear the difference between instr.1 and 2 sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;sine wave at k-rate ksig oscil 0.8, 440, 1 ; k-rate to the audio-rate conversion asig = a(ksig) outs asig, asig endin instr 2 ;sine wave at a-rate asig oscil 0.8, 440, 1 outs asig, asig endin </CsInstruments> <CsScore> ; sine wave. f 1 0 16384 10 1 i 1 0 2 i 2 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fmwurlie.csd.xml0000644000000000000000000000452112262561504020507 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fmwurlie.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq = 440 kc1 = p4 kc2 = 1 kvdepth = 0.05 kvrate = 6 ifn1 = 1 ifn2 = 1 ifn3 = 1 ifn4 = 2 ivfn = 1 asig fmwurlie .5, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn outs asig, asig endin </CsInstruments> <CsScore> ; sine wave f 1 0 32768 10 1 ; audio file f 2 0 256 1 "fwavblnk.aiff" 0 0 0 i 1 0 3 6 i 1 + 3 30 i 1 + 2 60 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pareq.csd.xml0000644000000000000000000000562312262561504017771 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pareq.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 15 ifc = p4 ; Center / Shelf kq = p5 ; Quality factor sqrt(.5) is no resonance kv = ampdb(p6) ; Volume Boost/Cut imode = p7 ; Mode 0=Peaking EQ, 1=Low Shelf, 2=High Shelf kfc linseg ifc*2, p3, ifc/2 asig rand 5000 ; Random number source for testing aout pareq asig, kfc, kv, kq, imode ; Parmetric equalization outs aout, aout ; Output the results endin </CsInstruments> <CsScore> ; SCORE: ; Sta Dur Fcenter Q Boost/Cut(dB) Mode i15 0 1 10000 .2 12 1 i15 + . 5000 .2 12 1 i15 . . 1000 .707 -12 2 i15 . . 5000 .1 -12 0 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fin.csd.xml0000644000000000000000000000364012262561504017432 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fin.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asnd init 0 ;input of fin must be initialized fin "fox.wav", 0, 0, asnd ;read audiofile aenv follow asnd, 0.01 ;envelope follower kenv downsamp aenv asig rand kenv ;gate the noise with audiofile outs asig, asig endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/else.csd.xml0000644000000000000000000000436612262561504017614 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o else.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ipch = cpspch(p4) ienv = p5 if (ienv == 0) then kenv adsr 0.01, 0.95, .7, .5 else kenv linseg 0, p3 * .5, 1, p3 * .5, 0 endif aout vco2 .8, ipch, 10 aout moogvcf aout, ipch + (kenv * 6 * ipch) , .5 aout = aout * kenv outs aout, aout endin </CsInstruments> <CsScore> i 1 0 2 8.00 0 i 1 3 2 8.00 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/aftouch.csd.xml0000644000000000000000000000315112262561504020304 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out No messages MIDI in -odac -d -M0 ;;;RT audio out with MIDI in </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kaft aftouch 0, 1 printk2 kaft ;aftertouch from music keyboard used for volume control asig oscil 0.7 * kaft, 220, 1 outs asig, asig endin </CsInstruments> <CsScore> ;sine wave. f 1 0 16384 10 1 i 1 0 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpsmidinn2.csd.xml0000644000000000000000000000411012262561504020715 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ;;;RT audio out, midi in, note=p4 and velocity=p5 -odac -+rtmidi=virtual -M0d --midi-key=4 --midi-velocity-amp=5 ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cpsmidinn.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 massign 0, 1 ;assign all midi to instr. 1 instr 1 ;play virtual keyboard inote = p4 icps = cpsmidinn(inote) asig oscil 0.6, icps, 1 print icps outs asig, asig endin </CsInstruments> <CsScore> f0 20 ;sine wave. f 1 0 16384 10 1 ;play note from score too i1 0 1 60 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/madsr-2.csd.xml0000644000000000000000000000343312262561504020123 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o madsr-2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 icps cpsmidi iamp ampmidi .5 kenv madsr 0.5, 0, 1, 0.5 asig pluck kenv, icps, icps, 2, 1 outs asig, asig endin </CsInstruments> <CsScore> f 2 0 4096 10 1 f0 30 ;runs 30 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/strlower.csd.xml0000644000000000000000000000456112262561504020542 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n ;;;no sound output ;-iadc ;;;uncomment -iadc if realtime audio input is needed too </CsOptions> <CsInstruments> ;example of Joachim Heintz opcode FilSuf, S, So ;returns the suffix of a filename or path, optional in lower case Spath,ilow xin ipos strrindex Spath, "." ;look for the rightmost '.' Suf strsub Spath, ipos+1 ;extract the substring after "." if ilow != 0 then ;if ilow input is not 0 then Suf strlower Suf ;convert to lower case endif xout Suf endop instr suff ilow = p4 prints "Printing suffix:\n" Suf FilSuf "/my/dir/my/file.WAV", ilow puts Suf, 1 endin </CsInstruments> <CsScore> i "suff" 0 1 0 ;do not convert to lower case i "suff" 3 1 1 ;convert to lower case e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/divides.csd.xml0000644000000000000000000000424012262561504020302 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o /.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idiv = 1/p3 * p4 ktrm oscil 1, idiv, 1 ;use oscil as an envelope printf "retrigger rate per note duration = %f\n",1, idiv kndx line 5, p3, 1 ;vary index of FM asig foscil ktrm, 200, 1, 1.4, kndx, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 4096 10 1 ;sine wave i 1 0 3 10 i 1 4 3 15 i 1 8 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/compress.csd.xml0000644000000000000000000000456412262561504020517 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -iadc ;;;RT audio out and in ; For Non-realtime ouput leave only the line below: ; -o compress.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 ; uncompressed signal asig diskin2 "beats.wav", 1, 0, 1 outs asig, asig endin instr 2 ; compressed signal. ; Use the "beats.wav" audio file and a mic avoice in asig diskin2 "beats.wav", 1, 0, 1 ; duck the audio signal "beats.wav" with your voice. kthresh = 0 kloknee = 40 khiknee = 60 kratio = 3 katt = 0.1 krel = .5 ilook = .02 asig compress asig, avoice, kthresh, kloknee, khiknee, kratio, katt, krel, ilook ; voice-activated compressor outs asig, asig endin </CsInstruments> <CsScore> i 1 0 5 i 2 6 21 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/timeinsts.csd.xml0000644000000000000000000000406512262561504020677 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o timeinsts.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kvib init 1 ktim timeinsts ;read time if ktim > 2 then ;do something after 2 seconds kvib oscili 2, 3, giSine ;make a vibrato endif asig poscil .5, 600+kvib, giSine ;add vibrato outs asig, asig endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsynth.csd.xml0000644000000000000000000000407612262561504020375 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsynth.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1 ; von-Hann window Sfile = "fox.wav" ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape; fft-analysis of the audio-signal aout pvsynth fftin; resynthesis out aout endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/toot7.csd.xml0000644000000000000000000001241412262561504017731 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac </CsOptions> <CsInstruments> instr 7 ifunc1 = p11 ; initial waveform ifunc2 = p12 ; crossfade waveform ifad1 = p3 * p13 ; calculate initial fade (% of dur) ifad2 = p3 - ifad1 ; calculate remaining duration irel = .01 ; set vibrato release time idel1 = p3 * p10 ; calculate initial delay (% of dur) isus = p3 - (idel1 + irel) ; calculate remaining duration iamp = ampdb(p4) iscale = iamp * .166 ; p4=amp inote = cpspch(p5) ; p5=freq k3 linseg 0, idel1, p9, isus, p9, irel, 0 ; p6=attack time k2 oscil k3, p8, 1 ; p7=release time k1 linen iscale, p6, p3, p7 ; p8=vib rate a6 oscil k1, inote*.998+k2, ifunc2 ; p9=vib depth a5 oscil k1, inote*1.002+k2, ifunc2 ; p10=vib delay (0-1) a4 oscil k1, inote+k2, ifunc2 ; p11=initial wave a3 oscil k1, inote*.997+k2, ifunc1 ; p12=cross wave a2 oscil k1, inote*1.003+k2, ifunc1 ; p13=fade time a1 oscil k1, inote+k2, ifunc1 kfade linseg 1, ifad1, 0, ifad2, 1 afunc1 = kfade * (a1+a2+a3) afunc2 = (1 - kfade) * (a4+a5+a6) out afunc1 + afunc2 endin </CsInstruments> <CsScore> f1 0 2048 10 1 ; Sine f2 0 2048 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth f3 0 2048 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square f4 0 2048 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse ;in st dur amp frq atk rel vbrt vbdp vbdl stw ndw crstm i7 0 5 96 8.07 .03 .1 5 6 .99 1 2 .1 i7 6 5 96 8.09 .03 .1 5 6 .99 1 3 .1 i7 12 8 96 8.07 .03 .1 5 6 .99 1 4 . e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/genfarey.csd.xml0000644000000000000000000000335612262561504020462 0ustar rootroot <CsoundSynthesizer> <CsOptions> </CsOptions> <CsInstruments> sr=44100 ksmps=10 nchnls=1 instr 4 kndx init 0 ; read out elements of F_8 one by one and print to file if (kndx < 23) then kelem tab kndx, 1 fprintks "farey8table.txt", "%2.6f\\n", kelem kndx = kndx+1 endif endin </CsInstruments> <CsScore> ; initialise integer for Farey Sequence F_8 f1 0 -23 "farey" 8 0 ; if mode=0 then the table stores all elements of the Farey Sequence ; as fractions in the range [0,1] i4 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cosseg.csd.xml0000644000000000000000000000607212262561504020143 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o linseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 kcps = cpspch(p4) kenv cosseg 0, 0.25, 1, 0.75, 0 ; together = 1 sec asig poscil kenv, kcps, giSine outs asig, asig endin instr 2 ; scaling to duration kcps = cpspch(p4) kenv cosseg 0, p3*0.25, 1, p3*0.75, 0 asig poscil kenv, kcps, giSine outs asig, asig endin instr 3 ; with negative value kcps = cpspch(p4) aenv cosseg 0, 0.1, 1, 0.5, -0.9, 0.4, 0 asig poscil aenv, kcps, giSine outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 7.00 ; = 1 sec, p3 fits exactly i 1 2 2 7.00 ; = 2 sec, p3 truncated at 1 sec i 2 4 1 7.00 ; scales to duration i 2 6 2 7.00 ; of p3 i 3 9 2 7.00 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outq1.csd.xml0000644000000000000000000000354712262561504017735 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o outq1.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 0dbfs = 1 instr 1 asig vco2 .05, 30 ; sawtooth waveform at low volume kcut line 60, p3, 300 ; Vary cutoff frequency kresonance = 7 inumlayer = 2 asig lowresx asig, kcut, kresonance, inumlayer outq1 asig ; output channel 1 endin </CsInstruments> <CsScore> i 1 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/resonx.csd.xml0000644000000000000000000000474412262561504020202 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o resonx.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; unfiltered noise kenv linseg 0, p3*.5, 1, p3*.5, 0 ;envelope asig rand 0.7 ;white noise outs asig*kenv, asig*kenv endin instr 2 ; filtered noise kenv linseg 0, p3*.5, 1, p3*.5, 0 ;envelope asig rand 0.7 kcf line 300, p3, 2000 afilt resonx asig, kcf, 300, 4 asig balance afilt, asig outs asig*kenv, asig*kenv endin </CsInstruments> <CsScore> i 1 0 2 i 2 3 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen25.csd.xml0000644000000000000000000000440712262561504017600 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen25.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 1, 0, 32768, 10, 1 gienv ftgen 2, 0, 1025, 25, 0, 0.01, 200, 1, 400, 1, 513, 0.01 ; y value must be >= 0 instr 1 kcps init 3/p3 ;play 3x over duration of note kndx phasor kcps ixmode = 1 ;normalize to 0-1 kval table kndx, gienv, ixmode kval =kval*100 ;scale up to 0-100 asig poscil 1, 220+kval, gisin ;use table for amplitude outs asig, asig endin </CsInstruments> <CsScore> i 1 0 4 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/partikkel-2.csd.xml0000644000000000000000000003137512262561504021011 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out -odac ;;;RT audio ; For Non-realtime ouput leave only the line below: ; -o partikkel.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 20 nchnls = 2 ; Example by Joachim Heintz and Oeyvind Brandtsegg 2008 giCosine ftgen 0, 0, 8193, 9, 1, 1, 90 ; cosine giDisttab ftgen 0, 0, 32768, 7, 0, 32768, 1 ; for kdistribution giFile ftgen 0, 0, 0, 1, "fox.wav", 0, 0, 0 ; soundfile for source waveform giWin ftgen 0, 0, 4096, 20, 9, 1 ; grain envelope giPan ftgen 0, 0, 32768, -21, 1 ; for panning (random values between 0 and 1) ; ************************************************* ; partikkel example, processing of soundfile ; uses the file "fox.wav" ; ************************************************* instr 1 /*score parameters*/ ispeed = p4 ; 1 = original speed igrainrate = p5 ; grain rate igrainsize = p6 ; grain size in ms icent = p7 ; transposition in cent iposrand = p8 ; time position randomness (offset) of the pointer in ms icentrand = p9 ; transposition randomness in cents ipan = p10 ; panning narrow (0) to wide (1) idist = p11 ; grain distribution (0=periodic, 1=scattered) /*get length of source wave file, needed for both transposition and time pointer*/ ifilen tableng giFile ifildur = ifilen / sr /*sync input (disabled)*/ async = 0 /*grain envelope*/ kenv2amt = 1 ; use only secondary envelope ienv2tab = giWin ; grain (secondary) envelope ienv_attack = -1 ; default attack envelope (flat) ienv_decay = -1 ; default decay envelope (flat) ksustain_amount = 0.5 ; no meaning in this case (use only secondary envelope, ienv2tab) ka_d_ratio = 0.5 ; no meaning in this case (use only secondary envelope, ienv2tab) /*amplitude*/ kamp = 0.4*0dbfs ; grain amplitude igainmasks = -1 ; (default) no gain masking /*transposition*/ kcentrand rand icentrand ; random transposition iorig = 1 / ifildur ; original pitch kwavfreq = iorig * cent(icent + kcentrand) /*other pitch related (disabled)*/ ksweepshape = 0 ; no frequency sweep iwavfreqstarttab = -1 ; default frequency sweep start iwavfreqendtab = -1 ; default frequency sweep end awavfm = 0 ; no FM input ifmamptab = -1 ; default FM scaling (=1) kfmenv = -1 ; default FM envelope (flat) /*trainlet related (disabled)*/ icosine = giCosine ; cosine ftable kTrainCps = igrainrate ; set trainlet cps equal to grain rate for single-cycle trainlet in each grain knumpartials = 1 ; number of partials in trainlet kchroma = 1 ; balance of partials in trainlet /*panning, using channel masks*/ imid = .5; center ileftmost = imid - ipan/2 irightmost = imid + ipan/2 giPanthis ftgen 0, 0, 32768, -24, giPan, ileftmost, irightmost ; rescales giPan according to ipan tableiw 0, 0, giPanthis ; change index 0 ... tableiw 32766, 1, giPanthis ; ... and 1 for ichannelmasks ichannelmasks = giPanthis ; ftable for panning /*random gain masking (disabled)*/ krandommask = 0 /*source waveforms*/ kwaveform1 = giFile ; source waveform kwaveform2 = giFile ; all 4 sources are the same kwaveform3 = giFile kwaveform4 = giFile iwaveamptab = -1 ; (default) equal mix of source waveforms and no amplitude for trainlets /*time pointer*/ afilposphas phasor ispeed / ifildur /*generate random deviation of the time pointer*/ iposrandsec = iposrand / 1000 ; ms -> sec iposrand = iposrandsec / ifildur ; phase values (0-1) krndpos linrand iposrand ; random offset in phase values /*add random deviation to the time pointer*/ asamplepos1 = afilposphas + krndpos; resulting phase values (0-1) asamplepos2 = asamplepos1 asamplepos3 = asamplepos1 asamplepos4 = asamplepos1 /*original key for each source waveform*/ kwavekey1 = 1 kwavekey2 = kwavekey1 kwavekey3 = kwavekey1 kwavekey4 = kwavekey1 /* maximum number of grains per k-period*/ imax_grains = 100 aL, aR partikkel igrainrate, idist, giDisttab, async, kenv2amt, ienv2tab, \ ienv_attack, ienv_decay, ksustain_amount, ka_d_ratio, igrainsize, kamp, igainmasks, \ kwavfreq, ksweepshape, iwavfreqstarttab, iwavfreqendtab, awavfm, \ ifmamptab, kfmenv, icosine, kTrainCps, knumpartials, \ kchroma, ichannelmasks, krandommask, kwaveform1, kwaveform2, kwaveform3, kwaveform4, \ iwaveamptab, asamplepos1, asamplepos2, asamplepos3, asamplepos4, \ kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains outs aL, aR endin </CsInstruments> <CsScore> ;i1 st dur speed grate gsize cent posrnd cntrnd pan dist i1 0 2.757 1 200 15 0 0 0 0 0 s i1 0 2.757 1 200 15 400 0 0 0 0 s i1 0 2.757 1 15 450 400 0 0 0 0 s i1 0 2.757 1 15 450 400 0 0 0 0.4 s i1 0 2.757 1 200 15 0 400 0 0 1 s i1 0 5.514 .5 200 20 0 0 600 .5 1 s i1 0 11.028 .25 200 15 0 1000 400 1 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/outvalue.csd.xml0000644000000000000000000000262712262561504020526 0ustar rootroot <CsoundSynthesizer> <CsOptions> </CsOptions> <CsInstruments> ;run this example in CsoundQt, a Csound editor that provides widgets ;make the Widgets-panel visible, by clicking the Widgets symbol in the menu or pressing (Alt+1). sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; written by Alex Hofmann instr 1 ;move fader kMoveUp linseg 0, 3, 1, 1, 1, 0.5, 0 outvalue "movefader", kMoveUp endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cpsxpch_105et.csd.xml0000644000000000000000000000361312262561504021244 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cpsxpch_105et.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a 10.5ET scale. ipch = 4.02 iequal = 21 irepeat = 4 ibase = 16.35160062496 icps cpsxpch ipch, iequal, irepeat, ibase print icps endin </CsInstruments> <CsScore> ; Play Instrument #1 for one second. i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vtablek.csd.xml0000644000000000000000000000406112262561504020304 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac -B441 -b441 </CsOptions> <CsInstruments> sr = 44100 kr = 100 ksmps = 441 nchnls = 2 gkindx init -1 instr 1 kindex init 0 ktrig metro 0.5 if ktrig = 0 goto noevent gkindx = gkindx + 1 noevent: endin instr 2 kout1 init 0 kout2 init 0 kout3 init 0 kout4 init 0 vtablek gkindx, 1, 1, 0, kout1,kout2, kout3, kout4 printk2 kout1 printk2 kout2 printk2 kout3 printk2 kout4 endin </CsInstruments> <CsScore> f 1 0 32 10 1 i 1 0 20 i 2 0 20 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/butterbr.csd.xml0000644000000000000000000000366712262561504020520 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o butterbr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; White noise asig rand 0.5 outs asig, asig endin instr 2 ; filtered noise asig rand 0.7 abr butterbr asig, 3000, 2000 ;center frequency = 3000, bandwidth = +/- (2000)/2, so 2000-4000 outs abr, abr endin </CsInstruments> <CsScore> i 1 0 2 i 2 2.5 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsbuffer.csd.xml0000644000000000000000000000647112262561504020665 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsbufread.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 opcode FileToPvsBuf, iik, Siiii ;;writes an audio file at the first k-cycle to a fft-buffer (via pvsbuffer) Sfile, ifftsize, ioverlap, iwinsize, iwinshape xin ktimek timeinstk if ktimek == 1 then ilen filelen Sfile kcycles = ilen * kr; number of k-cycles to write the fft-buffer kcount init 0 loop: ain soundin Sfile fftin pvsanal ain, ifftsize, ioverlap, iwinsize, iwinshape ibuf, ktim pvsbuffer fftin, ilen + (ifftsize / sr) loop_lt kcount, 1, kcycles, loop xout ibuf, ilen, ktim endif endop instr 1 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window ibuffer, ilen, k0 FileToPvsBuf "fox.wav", ifftsize, ioverlap, iwinsize, iwinshape ktmpnt linseg ilen, p3, 0; reads the buffer backwards in p3 seconds fread pvsbufread ktmpnt, ibuffer aout pvsynth fread out aout endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/aresonk.csd.xml0000644000000000000000000000413612262561504020321 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o aresonk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 0, 0, 2^10, 10, 1 instr 1 ksig randomh 400, 1800, 150 aout poscil .2, 1000+ksig, gisin outs aout, aout endin instr 2 ksig randomh 400, 1800, 150 kbw line 1, p3, 600 ; vary bandwith ksig aresonk ksig, 800, kbw aout poscil .2, 1000+ksig, gisin outs aout, aout endin </CsInstruments> <CsScore> i 1 0 5 i 2 5.5 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/FLtext.csd.xml0000644000000000000000000000703512262561504020066 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o FLtext.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; A sine with oscillator with fltext box controlled ; frequency either click and drag or double click and ; type to change frequency value sr = 44100 kr = 441 ksmps = 100 nchnls = 1 FLpanel "Frequency Text Box", 270, 600, 50, 50 ; Minimum value output by the text box imin = 200 ; Maximum value output by the text box imax = 5000 ; Step size istep = 1 ; Text box graphic type itype = 1 ; Width of the text box in pixels iwidth = 70 ; Height of the text box in pixels iheight = 30 ; Distance of the left edge of the text box ; from the left edge of the panel ix = 100 ; Distance of the top edge of the text box ; from the top edge of the panel iy = 300 gkfreq,ihandle FLtext "Enter the frequency", imin, imax, istep, itype, iwidth, iheight, ix, iy ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 iamp = 15000 ifn = 1 asig oscili iamp, gkfreq, ifn out asig endin </CsInstruments> <CsScore> ; Function table that defines a single cycle ; of a sine wave. f 1 0 1024 10 1 ; Instrument 1 will play a note for 1 hour. i 1 0 3600 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ftgentmp.csd.xml0000644000000000000000000000266412262561504020507 0ustar rootroot <CsoundSynthesizer> <CsOptions> </CsOptions> <CsInstruments> sr = 44100 ksmps = 128 nchnls = 2 0dbfs = 1 instr 1 ifno ftgentmp 0, 0, 512, 10, 1 print ifno endin instr 2 print ftlen(p4) endin </CsInstruments> <CsScore> i 1 0 10 i 2 2 1 101 i 1 5 10 i 2 7 1 102 i 2 12 1 101 i 2 17 1 102 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/loopxseg.csd.xml0000644000000000000000000000522012262561504020512 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loopxseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kfreq rspline 0.01,20,0.2,1 ; freq. of loop repetition created by random spline ktrig init 0 ; loop restart trigger (not used) iphase = 0 ; initial phase ; loop of filter cutoff values (oct format). Rescaled further down. kcfoct loopxseg kfreq, ktrig, iphase, 1,1,0,0 kenv linseg 0,0.01,1,p3-5.01,1,5,0 asig vco2 0.2*kenv,cpsmidinn(48),0 kdep rspline 5,8,0.2,1 ; filter depth created by a random spline kcf port cpsoct((kcfoct*kdep)+4), 0.001 ; smooth filter changes asig moogladder asig,kcf,rnd(0.6) aL,aR pan2 asig,rnd(1) outs aL, aR endin </CsInstruments> <CsScore> i 1 0 60 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/max.csd.xml0000644000000000000000000000441412262561504017443 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o max.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 k1 oscili 1, 10.0, 1 ;combine 3 sinusses k2 oscili 1, 1.0, 1 ;at different rates k3 oscili 1, 3.0, 1 kmax max k1, k2, k3 kmax = kmax*250 ;scale kmax printk2 kmax ;check the values aout vco2 .5, 220, 6 ;sawtooth asig moogvcf2 aout, 600+kmax, .5 ;change filter around 600 Hz outs asig, asig endin </CsInstruments> <CsScore> f1 0 32768 10 1 i1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dbamp.csd.xml0000644000000000000000000000335412262561504017743 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dbamp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 instr 1 iamp = p4 idb = dbamp(iamp) print idb asig vco2 iamp, 110 ;sawtooth outs asig, asig endin </CsInstruments> <CsScore> i 1 0 1 100 i 1 + 1 1000 i 1 + 1 10000 i 1 + 1 20000 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen49.csd.xml0000644000000000000000000000337012262561504017604 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen49.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kcps = sr/ftlen(1) asig oscil .8, kcps, 1 outs asig, asig endin </CsInstruments> <CsScore> f 1 0 131072 49 "beats.mp3" 0 1 ;read an audio file (using GEN49). i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/modulus.csd.xml0000644000000000000000000000645412262561504020354 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -Ma --midi-key=4 --midi-velocity-amp=5 -m0 ;;;realtime audio out and midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o %.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giScale ftgen 1, 0, -12, -2, 0, -34, -7, 10, -14, 3, -21, -3, -27, -10, 7, -17 ;12 note scale with detuned keys instr 1 ikey = p4 ivel = p5 indx = ikey % 12 ;work on the twelftone scale icent tab_i indx, giScale ;load the scale ifreqeq = cpsmidinn(ikey) ifreq = ifreqeq * cent(icent) ;change frequency by cents from table prints "Key %d modulus 12 = %d. ", ikey, indx prints "Equal-tempered frequency of this key = %f,", ifreqeq prints " but here with cent deviation %d = %f%n", icent, ifreq asig vco2 ivel*.5, ifreq outs asig, asig endin </CsInstruments> <CsScore> f 0 60 ;run for 60 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/clfilt_highpass.csd.xml0000644000000000000000000000377612262561504022033 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o clfilt_highpass.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; white noise asig rand 0.6 outs asig, asig endin instr 2 ;filtered noise asig rand 0.7 ; Highpass filter signal asig with a 6-pole Chebyshev ; Type I at 20 Hz with 3 dB of passband ripple. a1 clfilt asig, 20, 1, 6, 1, 3 outs a1, a1 endin </CsInstruments> <CsScore> i 1 0 2 i 2 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/upsamp.csd.xml0000644000000000000000000001211112262561504020154 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o upsamp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;;with code from Steven Cook / David Akbari, Menno Knevel and Joachim Heintz sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 seed 0 opcode Decimator, a, akk ;UDO Sample rate / Bit depth reducer ;see http://www.csounds.com/udo/displayOpcode.php?opcode_id=73 setksmps 1 ain, kbit, ksrate xin kbits = 2^kbit ;bit depth (1 to 16) kfold = (sr/ksrate) ;sample rate kin downsamp ain ;convert to kr kin = (kin+0dbfs) ;add DC to avoid (-) kin = kin*(kbits/(0dbfs*2)) ;scale signal level kin = int(kin) ;quantise aout upsamp kin ;convert to sr aout = aout*(2/kbits)-0dbfs ;rescale and remove DC a0ut fold aout, kfold ;resample xout a0ut endop instr 1 ;avoid playing this too loud kbit = p4 ksr = 44100 asig diskin "fox.wav", 1 aout Decimator asig, kbit, ksr printks "bitrate = %d, ", 3, kbit printks "with samplerate = %d\\n", 3, ksr outs aout*.7, aout*.7 endin instr 2 ;moving randomly between different bit values (1 - 6) kbit randomi 1, 6, .5, 1 asig diskin "fox.wav", 1, 0, 1 ;loop play aout Decimator asig, kbit, 44100 printks "bitrate = %f\n", .3, kbit outs aout*.7, aout*.7 endin </CsInstruments> <CsScore> i 1 0 3 16 ;sounds allright but i 1 + 3 5 ;it's getting worse i 1 + 3 2 ;and worse... i 2 9 22 ;or quality moves randomly e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gendy.csd.xml0000644000000000000000000000275012262561504017765 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aout gendy 0.7, 1, 1, 1, 1, 20, 1000, 0.5, 0.5 outs aout, aout endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/STKSaxofony.csd.xml0000644000000000000000000000424312262561504021046 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o STKSaxofony.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifreq = p4 kv1 = p5 ;reed stiffness kv3 line p6, p3, p7 ;blow position kv6 line 0, p3, 127 ;depth of low-frequency oscillator asig STKSaxofony cpspch(p4), 1, 2, kv1, 4, 100, 26, 70, 11, kv3, 1, kv6, 29, 100 asig = asig * .5 ;too loud outs asig, asig endin </CsInstruments> <CsScore> i 1 0 3 6.00 30 100 10 i 1 + . 8.00 30 100 100 i 1 + . 7.00 90 127 30 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sndloop.csd.xml0000644000000000000000000000444512262561504020340 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sndloop.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 instr 1 itrig = p4 asig diskin2 "beats.wav", 1, 0, 1 ;get the signal in, loop it ktrig line 0, itrig, 1 ;when to trigger signal = p4 kpitch line 1.2, p3, .5 ;vary pitch of recorded signal aout,krec sndloop asig, kpitch, ktrig, .4, 0.05 ;rec starts at p4 sec, for .4 secs and 0.05 crossfade printk2 krec ; prints the recording signal outs aout, aout endin </CsInstruments> <CsScore> i1 0 5 .5 ;trigger in seconds (=p4) i1 + 5 .8 i1 + 10 1.2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/monitor.csd.xml0000644000000000000000000000527612262561504020354 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o monitor.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 asig poscil3 .5, 880, giSine ;write a raw file: 32 bits with header fout "fout_880.wav", 15, asig outs asig, asig endin instr 2 klfo lfo 1, 2, 0 asig poscil3 .5*klfo, 220, giSine ;write an aiff file: 32 bits with header fout "fout_aif.aiff", 25, asig ; fout "fout_all3.wav", 14, asig outs asig, asig endin instr 99 ;read the stereo csound output buffer allL, allR monitor ;write the output of csound to an audio file ;to a wav file: 16 bits with header fout "fout_all.wav", 14, allL, allR endin </CsInstruments> <CsScore> i 1 0 2 i 2 0 3 i 99 0 3 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fareyleni.csd.xml0000644000000000000000000001166212262561504020637 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc for RT audio input as well ; For Non-realtime ouput leave only the line below: ; -o fareyleni.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; initialise integer n for Farey Sequence F_8 gifarn init 8 ; calculate length of F_8, it should return 23 for |F_8| gires fareyleni gifarn ; convert to negative number for the GEN routine because ; the table length won't be a power of 2 ; (The length of a Farey Sequence of n > 1 is always an odd number) gilen init gires * -1 ; create F_8 with GENfarey, the negative table number prevents ; unnecessary normalisation (F_8 IS already normalised) ; n mode: gifarey ftgen 100, 0, gilen, "farey", gifarn, 4 ; if mode=4 then 1 is added to each element of F_n. ; Useful for creating just tuning tables that can be read by the cps2pch opcode. instr 1 ; the very last element of F_n is not needed in the case of tuning tables ires = gires - 1 ; read out and print to file kndx init 0 if (kndx < ires) then kelem tab kndx, gifarey fprintks "farey8_tuning.txt", "%2.6f\\n", kelem kndx = kndx+1 endif endin instr 2 ip cps2pch p4, -100 asig poscil .5, ip, 1 aenv linseg 0, 0.1, 1, p3-0.2, 1, 0.1, 0 outs asig * aenv, asig * aenv endin </CsInstruments> <CsScore> f1 0 8192 10 1 ;sine wave i1 0 .1 i2 1 .5 8.00 i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + . > i2 + 1 8.22 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen21.csd.xml0000644000000000000000000001003712262561504017570 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen21.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifn = p4 isize = ftlen(ifn) prints "TABLE NUMBER: %d\n", ifn prints "Index\tValue\n" iindex = 0 ;start loop begin_loop: ivalue tab_i iindex, ifn prints "%d:\t%f\n", iindex, ivalue iindex = iindex + 1 if (iindex < isize) igoto begin_loop ;produce sound - and repeat it 10 times so you can hear the patterns: aphase phasor 10/10 ;play all 32 values 10x over 10 seconds aphase = aphase*isize ;step through table afrq table aphase, p4 ;read table number asig poscil .5, (afrq*500)+1000,10 ;scale values of table 500 times, add 1000 Hz outs asig , asig ;so we can distinguish the different tables endin </CsInstruments> <CsScore> f1 0 32 21 1 ;Uniform (white noise) f2 0 32 21 6 ;Gaussian (mu=0.0, sigma=1/3.83=0.261) f3 0 32 21 6 5.745 ;Gaussian (mu=0.0, sigma=5.745/3.83=1.5) f4 0 32 21 9 1 1 2 ;Beta (note that level precedes arguments) f5 0 32 21 10 1 2 ;Weibull f10 0 8192 10 1 ;Sine wave i 1 0 10 1 i 1 11 10 2 i 1 22 10 3 i 1 33 10 4 i 1 44 10 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/cent.csd.xml0000644000000000000000000000557112262561504017614 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o cent.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; original pitch iroot = 440 ; root note = A (440 Hz) print iroot ;print out asig oscili 0.6, iroot, 1 outs asig, asig endin instr 2 iroot = 440 ; root note = A (440 Hz) icents = p4 ; change root note by 300 and 1200 cents ifactor = cent(icents) ; calculate new note inew = iroot * ifactor print iroot ; Print all print ifactor print inew asig oscili 0.6, inew, 1 outs asig, asig endin </CsInstruments> <CsScore> ; sine wave f1 0 32768 10 1 i 1 0 2 0 ;no change i 2 2.5 2 300 ;note = C above A i 2 5 2 1200 ;1 octave higher e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/hrtfmove2.csd.xml0000644000000000000000000000553712262561504020601 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select flags here ; realtime audio out -o dac ; For Non-realtime ouput leave only the line below: ; -o hrtf.wav </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 gasrc init 0 instr 1 ;a plucked string kamp = p4 kcps = cpspch(p5) icps = cpspch(p5) a1 pluck kamp, kcps, icps, 0, 1 gasrc = a1 endin instr 10 ;uses output from instr1 as source kaz linseg 0, p3, 720 ;2 full rotations aleft,aright hrtfmove2 gasrc, kaz,0, "hrtf-44100-left.dat","hrtf-44100-right.dat" outs aleft, aright endin </CsInstruments> <CsScore> ; Play Instrument 1: a simple arpeggio i1 0 .2 15000 8.00 i1 + .2 15000 8.04 i1 + .2 15000 8.07 i1 + .2 15000 8.11 i1 + .2 15000 9.02 i1 + 1.5 15000 8.11 i1 + 1.5 15000 8.07 i1 + 1.5 15000 8.04 i1 + 1.5 15000 8.00 i1 + 1.5 15000 7.09 i1 + 1.5 15000 8.00 ; Play Instrument 10 for 10 seconds. i10 0 10 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sekere.csd.xml0000644000000000000000000000316612262561504020137 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sekere.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 idamp = p4 ;vary damping amount asig sekere 1, 0.01, 64, idamp outs asig, asig endin </CsInstruments> <CsScore> i1 0 1 .1 i1 + 1 .9 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/interp.csd.xml0000644000000000000000000000565312262561504020165 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o interp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 8000 kr = 8 ksmps = 1000 nchnls = 1 ; Instrument #1 - a simple instrument. instr 1 ; Create an amplitude envelope. kamp linseg 0, p3/2, 20000, p3/2, 0 ; The amplitude envelope will sound rough because it ; jumps every ksmps period, 1000. a1 oscil kamp, 440, 1 out a1 endin ; Instrument #2 - a smoother sounding instrument. instr 2 ; Create an amplitude envelope. kamp linseg 0, p3/2, 25000, p3/2, 0 aamp interp kamp ; The amplitude envelope will sound smoother due to ; linear interpolation at the higher a-rate, 8000. a1 oscil aamp, 440, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 256 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 2 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/loopsegp.csd.xml0000644000000000000000000000662012262561504020507 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loopsegp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ktrig metro 22/8 ; triggers used to generate new direction values kdir trandom ktrig,-2.99,2.99 kdir = 0.5*int(kdir) ; kdir will either -1, -0.5, 0, 0.5 or 1 ; kphase - looping pointer kphase phasor kdir ; a loop sequence of midi note numbers and durations knote loopsegp kphase, 40,1,40,0, 43,1,43,0, 49,2,48,0, \ 47,1,47,0, 46,1,46,0, 46,1,47,0, 49,1,49,0, 43,1,43,0, 46,1,46,0, 40,1,39,0 kmul rspline 0.1,0.8,0.5,5 ; modulation of buzz tone asig gbuzz 0.2, cpsmidinn(knote), 30, 3, kmul, 1 ; buzz tone outs asig, asig schedkwhen ktrig,0,0,2,0,0.1 ; play metronome endin instr 2 ; metronome acps expon 180+rnd(40),p3,50 aamp expon 0.05+rnd(0.05),p3,0.001 asig poscil aamp-0.001,acps,2 outs asig,asig endin </CsInstruments> <CsScore> ; cosine wave. f 1 0 16384 11 1 ; sine wave. f 2 0 16384 10 1 i 1 0 360 0.25 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vincr.csd.xml0000644000000000000000000000422312262561504017775 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vincr.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gaReverb init 0 instr 1 idur = p3 kpitch = p4 a1 diskin2 "fox.wav", kpitch a1 = a1*.5 ;reduce volume vincr gaReverb, a1 endin instr 99 ; global reverb al, ar reverbsc gaReverb, gaReverb, .8, 10000 outs gaReverb+al, gaReverb+ar clear gaReverb endin </CsInstruments> <CsScore> i1 0 3 1 i99 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/genwave.csd.xml0000644000000000000000000001620212262561504020310 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o 0dbfs.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 kr = 4410 nchnls = 1 0dbfs = 1 zakinit 3,1 instr 1 ; wavelet synth instrument iamp = p4; scaling factor of wavelets ifreq = p5; frequency of wavelets itab = p6; selected wavelet function inum = p7; number of wavelets to be created a1 osciln p4, p5, p6, p7 out a1 endin instr 2 ; wavelet analysis intrument a1 soundin "fox.wav" ; Decomposition Structure: ; 1 LEVEL 2 LEVEL ; HP->ah1 ; a1->| HP(up2)->ah2 ; LP->al1->| ; LP(up2)->al2 ; ain = a1*.5; attenuate input signal ; since wavelet coefficients ; could reach big values ah1 dconv ain,ftlen(8),8 al1 dconv ain,ftlen(7),7 ah2 dconv al1,ftlen(10),10 al2 dconv al1,ftlen(9),9 zaw ah1,0 zaw al1,1 zaw ah2,2 zaw al2,3 aout zar p4 out aout zacl 0,3 endin </CsInstruments> <CsScore> ; First of all, we need several FIR filters which are capable ; to produce wavelet families. ; One can input filter coefficients manualy using GEN02 ; or read them from text file. ; Most of compact-supported wavelet coefficients can be obtained from ; Wavelet Browser by PyWavelets wavelets.pybytes.com ; You can select family and order of filter ; then copy desired coefficients into txt file. ; Notice that for correct interpretation of results you should use ; coeffs of Decomposition low-pass filter. ; Daubechies 2 f 1 0 4 -2 -0.1294095226 0.2241438680 0.8365163037 0.4829629131 ; Symlet 10 f 2 0 0 -23 "sym10.txt" ; Now we want to produce some wavelet granules. ; They can be used in wavelet synthesis etc. ; Tables of large sizes should produce smoother wavelets. ; We take array of filter coefficients from ftable 1 ; and deconvolve it until output length of 16384. ; The order of filters through the deconvolution process ; is given by 14 which is 1110 in binary. ; So the first filter is LP ('0') and others are HP ('1'). f 3 0 16384 "wave" 1 14 0 f 4 0 16384 "wave" 2 1 0 f 5 0 16384 "wave" 2 7 0 f 6 0 16384 "wave" 2 6 0 ; The main purpose of using wavelets is wavelet transform. ; It is not that easy to perform a classic DWT in Csound since downsampling ; of audio signal is needed at each step of wavelet decomposition. ; Anyway, using GENwave it is possible to create a number of upsampled ; wavelets and perform a so-called undecimated wavelet transform ; aka stationary wavelet transform (and it is even better). ; So we need some upsampled childs of mother wavelet. f 7 0 16 "wave" 1 0 -1 ;db2 scaling function for 1st iteration f 8 0 16 "wave" 1 1 -1 ;db2 wavelet function for 1st iteration f 9 0 32 "wave" 1 0 -1 ;db2 scaling function for 2nd iteration f 10 0 32 "wave" 1 1 -1 ;db2 wavelet function for 2nd iteration ; Let's hear how some wavelets could sound.. ; amp frq wave times i 1 0 1 0.6 15 3 8 i 1 0.5 . 0.9 20 4 5 i 1 0.9 . 0.7 8 5 . i 1 1.1 . 0.4 30 6 9 ; Now try to decompose input file using wavelets i 2 2 4 1; approximation 1st level i 2 5 . 2; details 2nd level </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/midion_simple.csd.xml0000644000000000000000000000376012262561504021511 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d -M0 -Q1 ;;;RT audio I/O with MIDI in </CsOptions> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Example by Giorgio Zucco 2007 instr 1 ;Triggered by MIDI notes on channel 1 ifund notnum ivel veloc knote1 init ifund knote2 init ifund + 3 knote3 init ifund + 5 ;minor chord on MIDI out channel 1 ;Needs something plugged to csound's MIDI output midion 1, knote1,ivel midion 1, knote2,ivel midion 1, knote3,ivel endin </CsInstruments> <CsScore> ; Dummy ftable f0 60 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/downsamp.csd.xml0000644000000000000000000000401612262561504020504 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o downsamp.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ifrq = cpspch(p4) ain diskin2 "beats.wav", 1 aenv follow ain, .001 ;take the amplitude every 1/1000th of a second alow tone aenv, 25 ;lowpass-filter (25 Hz) for a clean signal kenv downsamp alow asig pluck kenv, ifrq, 15, 0, 1 outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 9 i 1 + . 7 i 1 + . 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vexpseg.csd.xml0000644000000000000000000000512312262561504020335 0ustar rootroot <CsoundSynthesizer> <CsOptions> -odac -B441 -b441 </CsOptions> <CsInstruments> sr=44100 ksmps=10 nchnls=2 gilen init 32 gitable1 ftgen 0, 0, gilen, 10, 1 gitable2 ftgen 0, 0, gilen, 10, 1 gitable3 ftgen 0, 0, gilen, -7, 30, gilen, 35 gitable4 ftgen 0, 0, gilen, -7, 400, gilen, 450 gitable5 ftgen 0, 0, gilen, -7, 5000, gilen, 5500 instr 1 vcopy gitable2, gitable1, gilen turnoff endin instr 2 vexpseg gitable2, 16, gitable3, 2, gitable4, 2, gitable5 endin instr 3 kcount init 0 if kcount < 16 then kval table kcount, gitable2 printk 0,kval kcount = kcount +1 else turnoff endif endin </CsInstruments> <CsScore> i1 0 1 s i2 0 10 i3 0 1 i3 1 1 i3 1.5 1 i3 2 1 i3 2.5 1 i3 3 1 i3 3.5 1 i3 4 1 i3 4.5 1 </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/temposcal.csd.xml0000644000000000000000000000403412262561504020643 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o temposcal.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ilock = p4 itab = 1 ipitch = 1 iamp = 0.8 ktime linseg 0.3, p3/2, 0.8, p3/2, 0.3 asig temposcal ktime, iamp, ipitch, itab, ilock outs asig, asig endin </CsInstruments> <CsScore> f 1 0 0 1 "fox.wav" 0 4 0 i 1 0 3.8 0 ;no locking i 1 4 3.8 1 ;locking e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/locsig_stereo.csd.xml0000644000000000000000000000532512262561504021521 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o locsig_stereo.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ga1 init 0 ga2 init 0 ; Instrument #1 instr 1 ; Use the "beats.wav" audio file. aout diskin2 "beats.wav", 1, 0, 1 kdegree line 0, p3, 180 kdistance line 1, p3 , 30 a1, a2 locsig aout, kdegree, kdistance, .1 ar1, ar2 locsend ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a2 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 outs a1, a2 ga1=0 ga2=0 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for five seconds. i 1 0 2 i99 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/prepiano.csd.xml0000644000000000000000000000435112262561504020473 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if real audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o prepiano.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ;; fund NS detune stiffness decay loss (bndry) (hammer) scan prep aa,ab prepiano 60, 3, 10, p4, 3, 0.002, 2, 2, 1, 5000, -0.01, p5, p6, 0, 0.1, 1, 2 outs aa*.2, ab*.2 endin </CsInstruments> <CsScore> f1 0 8 2 1 0.6 10 100 0.001 ;; 1 rattle f2 0 8 2 1 0.7 50 500 1000 ;; 1 rubber i1 0.0 1 1 0.09 20 i1 1 . -1 0.09 40 ;; 1 -> skip initialisation i1 2 . -1 0.09 60 i1 3 . -1 0.09 80 i1 4 1.8 -1 0.09 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dssi4cs.csd.xml0000644000000000000000000000562112262561504020233 0ustar rootroot <CsoundSynthesizer> <CsOptions> ;use appropriate realtime options </CsOptions> <CsInstruments> ksmps = 256 nchnls = 2 dssilist gihandle dssiinit "amp.so", 0, 1 ;gihandle dssiinit "cmt.so", 30 , 2 ;gihandle2 dssiinit "cmt.so", 8 , 1 ;gihandle dssiinit "delayorama_1402", 0 gihandle2 dssiinit "cmt.so", 49 , 1 ;gihandle dssiinit "freq_tracker_1418.so", 0 , 1, 1 ;gihandle dssiinit "g2reverb.so", 0, 1 ;gihandle2 dssiinit "declip_1195.so", 0, 1 ;gihandle2 dssiinit "revdelay_1605.so", 0, 1 ;gihandle2 dssiinit "tap_chorusflanger.so", 0, 1 ;gihandle2 dssiinit "plate_1423.so", 0, 1 gihandle3 dssiinit "gate_1410.so", 0, 1 ;gihandle3 dssiinit "hexter.so", 0, 1 instr 1 print p4 dssiactivate gihandle, p4 dssiactivate gihandle2, p4 dssiactivate gihandle3, p4 endin instr 2 ain1 inch 1 ain2 inch 2 ;aout1,aout2 dssiaudio gihandle, ain1, ain2 aout1 dssiaudio gihandle, ain1 outs aout1,aout1 endin instr 3 kval linen 1, p3 /3, p3, p3/ 3 dssictls gihandle, p4, kval, 1 endin instr 4 ain1 inch 1 aout1 dssiaudio gihandle2, ain1 outs aout1,aout1 endin </CsInstruments> <CsScore> i 1 1 1 1 i 2 2 15 ;plugin 1 i 3 3 12 0 ;Control port 0 i 4 8 2 ;plugin 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/multitap.csd.xml0000644000000000000000000000374712262561504020525 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o multitap.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ga1 init 0 instr 1 asig diskin2 "beats.wav", 1,0 outs asig, asig ga1 = ga1+asig endin instr 2 asig multitap ga1, 1.2, .5, 1.4, .2 outs asig, asig ga1 = 0 endin </CsInstruments> <CsScore> i 1 .5 .2 ; short sound i 2 0 3 ; echoes e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/clockon.csd.xml0000644000000000000000000000501712262561504020306 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o clockon.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Start clock #1. clockon 1 ; Do something that keeps Csound busy. a1 oscili 10000, 440, 1 out a1 ; Stop clock #1. clockoff 1 ; Print the time accumulated in clock #1. i1 readclock 1 print i1 endin </CsInstruments> <CsScore> ; Initialize the function tables. ; Table 1: an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for one second starting at 0:00. i 1 0 1 ; Play Instrument #1 for one second starting at 0:01. i 1 1 1 ; Play Instrument #1 for one second starting at 0:02. i 1 2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/exitnow.csd.xml0000644000000000000000000000657712262561504020367 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -n ;;;no sound ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o exitnow.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;after an example by Iain McCurdy sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 FLcolor 200, 200, 200, 0, 0, 0 ; LABEL | WIDTH | HEIGHT | X | Y FLpanel "rtclock", 500, 130, 0, 0 ; ON,OFF,TYPE,WIDTH, HEIGHT, X, Y, OPCODE, INS,START,IDUR gkOnOff,ihOnOff FLbutton "On/Off", 1, 0, 22, 150, 25, 5, 5, 0, 1, 0, 3600 gkExit,ihExit FLbutton "exitnow",1, 0, 21, 150, 25, 345, 5, 0, 999, 0, 0.001 FLsetColor2 255, 0, 50, ihOnOff ;reddish color ;VALUE DISPLAY BOXES WIDTH,HEIGHT,X, Y gidclock FLvalue "clock", 100, 25, 200, 60 FLsetVal_i 1, ihOnOff FLpanel_end FLrun instr 1 if gkOnOff !=0 kgoto CONTINUE ;sense if FLTK on/off switch is not off (in which case skip the next line) turnoff ;turn this instr. off now CONTINUE: ktime rtclock ;clock continues to run even FLprintk2 ktime, gidclock ;after the on/off button was used to stop endin instr 999 exitnow ;exit Csound as fast as possible endin </CsInstruments> <CsScore> f 0 60 ;runs 60 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vbap4.csd.xml0000644000000000000000000000413212262561504017667 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o vbap4.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 4 ;quad 0dbfs = 1 vbaplsinit 2, 4, 0, 90, 180, 270 instr 1 asig diskin2 "beats.wav", 1, 0, 1 ;loop beats.wav kaz line 0, p3, p4 ;come from right rear speaker & a1,a2,a3,a4 vbap4 asig, 180, 100, kaz ;change spread of soundsource printks "spread of source = %d\n", 1, kaz ;print spread value outq a1,a2,a3,a4 endin </CsInstruments> <CsScore> i 1 0 12 100 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/loscil3.csd.xml0000644000000000000000000000462112262561504020226 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o loscil3.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ichnls = ftchnls(p4) print ichnls if (ichnls == 1) then asigL loscil3 .8, 1, p4, 1 asigR = asigL elseif (ichnls == 2) then asigL, asigR loscil3 .8, 1, p4, 1 ;safety precaution if not mono or stereo else asigL = 0 asigR = 0 endif outs asigL, asigR endin </CsInstruments> <CsScore> f 1 0 0 1 "mary.wav" 0 0 0 f 2 0 0 1 "kickroll.wav" 0 0 0 i 1 0 3 1 ;mono file i 1 + 2 2 ;stereo file e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/zawm.csd.xml0000644000000000000000000000655412262561504017643 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o zawm.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 1 a-rate variable and 1 k-rate variable. zakinit 1, 1 ; Instrument #1 -- a basic instrument. instr 1 ; Generate a simple sine waveform. asin oscil 15000, 440, 1 ; Mix the sine waveform with za variable #1. zawm asin, 1 endin ; Instrument #2 -- another basic instrument. instr 2 ; Generate another waveform with a different frequency. asin oscil 15000, 880, 1 ; Mix this sine waveform with za variable #1. zawm asin, 1 endin ; Instrument #3 -- generates audio output. instr 3 ; Read za variable #1, containing both waveforms. a1 zar 1 ; Generate the audio output. out a1 ; Clear the za variables, get them ready for ; another pass. zacl 0, 1 endin </CsInstruments> <CsScore> ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 ; Play Instrument #3 for one second. i 3 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dumpk4.csd.xml0000644000000000000000000001041212262561504020055 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dumpk4.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giSine ftgen 0, 0, 2^10, 10, 1 instr 1 ;writes four control signals to a file kcf randh 950, 1, 2, 1, 1050 ;generates one random number between 100 and 2000 per second kq randh 10, 1, 2, 1, 11 ;generates another random number between 1 and 21 per second kdb randh 9, 1, 2, 1, -15 ;amplitudes in dB between -24 and -6 kpan randh .5, 1, 2, 1, .5 ;panning between 0 and 1 dumpk4 kcf, kq, kdb, kpan, "dumpk4.txt", 8, 1 ;writes the control signals prints "WRITING:\n" printks "kcf = %f, kq = %f, kdb = %f, kpan = %f\n", 1, kcf, kq, kdb, kpan ;prints them endin instr 2 ;reads the file written by instr 1 kcf,kq,kdb,kp readk4 "dumpk4.txt", 8, 1 prints "READING:\n" printks "kcf = %f, kq = %f, kdb = %f, kpan = %f\n", 1, kcf, kq, kdb, kp ;prints values kdb lineto kdb, .1 ;smoothing amp transition kp lineto kp, .1 ;smoothing pan transition anoise rand ampdb(kdb), 2, 1 kbw = kcf/kq ;bandwidth of resonant filter abp reson anoise, kcf, kbw aout balance abp, anoise aL, aR pan2 aout, kp outs aL, aR endin </CsInstruments> <CsScore> i 1 0 5 i 2 5 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/readf.csd.xml0000644000000000000000000000317412262561504017741 0ustar rootroot <CsoundSynthesizer> <CsOptions> -n </CsOptions> <CsInstruments> instr 1 Swd pwd printf_i "Working directory is '%s'\n", 1, Swd prints "Reading myself =):\n" read: Sline, kLinNum readf "readf.csd" printf "Line %d: %s", kLinNum, kLinNum, Sline if kLinNum != -1 then kgoto read else turnoff endif endin </CsInstruments> <CsScore> i1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/vexp_i.csd.xml0000644000000000000000000000507012262561504020147 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o cigoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr=44100 ksmps=128 nchnls=2 instr 1 ifn1 = p4 ival = p5 ielements = p6 idstoffset = p7 kval init 25 vexp_i ifn1, ival, ielements, idstoffset endin instr 2 ;Printtable itable = p4 isize = ftlen(itable) kcount init 0 kval table kcount, itable printk2 kval if (kcount == isize) then turnoff endif kcount = kcount + 1 endin </CsInstruments> <CsScore> f 1 0 16 -7 1 16 17 i2 0.0 0.2 1 i1 0.4 0.01 1 2 3 4 i2 0.8 0.2 1 i1 1.0 0.01 1 0.5 5 -3 i2 1.2 0.2 1 i1 1.4 0.01 1 1.5 10 12 i2 1.6 0.2 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sprintfk.csd.xml0000644000000000000000000000431212262561504020513 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o sprintfk.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 48000 ksmps = 16 nchnls = 2 0dbfs = 1 ; Example by Jonathan Murphy 2007 instr 1 S1 = "1" S2 = " + 1" ktrig init 0 kval init 2 if (ktrig == 1) then S1 strcatk S1, S2 kval = kval + 1 endif String sprintfk "%s = %d", S1, kval puts String, kval ktrig metro 1 endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/follow2.csd.xml0000644000000000000000000000431412262561504020241 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o follow2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 asig soundin "beats.wav" outs asig, asig endin instr 2 ;using follow2 as soundin "beats.wav" af follow2 as, p4, p5 ar rand 44100 ;noise ; "beats.wav" provides amplitude for noise asig balance ar, af outs asig, asig endin </CsInstruments> <CsScore> i 1 0 2 i 2 2 2 0.001 0.01 ;quick attack & deacy i 2 5 2 0.1 0.5 ;slow attack & deacy e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/ampmidid.csd.xml0000644000000000000000000000475212262561504020447 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ;;;RT audio out, note=p4 and velocity=p5 -odac --midi-key=4 --midi-velocity-amp=5 ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o ampmidid.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 massign 0, 1 ;assign all midi to instr. 1 instr 1 isine ftgenonce 0, 0, 4096, 10, 1 ;sine wave ihz = cpsmidinn(p4) ivelocity = p5 idb ampmidid ivelocity, 20 ;map to dynamic range of 20 dB. idb = idb + 60 ;limit range to 60 to 80 decibels iamplitude = ampdb(idb) ;loudness in dB to signal amplitude a1 oscili iamplitude, ihz, isine aenv madsr 0.05, 0.1, 0.5, 0.2 asig = a1 * aenv outs asig, asig endin </CsInstruments> <CsScore> ; note velocity i 1 0 2 61 100 i 1 + 2 65 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/kgoto.csd.xml0000644000000000000000000000615112262561504020001 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o kgoto.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Change kval linearly from 0 to 2 over ; the period set by the third p-field. kval line 0, p3, 2 ; If kval is greater than or equal to 1 then play the high note. ; If not then play the low note. if (kval >= 1) kgoto highnote kgoto lownote highnote: kfreq = 880 goto playit lownote: kfreq = 440 goto playit playit: ; Print the values of kval and kfreq. printks "kval = %f, kfreq = %f\\n", 1, kval, kfreq a1 oscil 10000, kfreq, 1 out a1 endin </CsInstruments> <CsScore> ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fluidSetInterpMethod.csd.xml0000644000000000000000000000540512262561504022761 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out and realtime midi in ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ;-o fluidSetInterpMethod.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giengine fluidEngine ; soundfont path to manual/examples isfnum fluidLoad "07AcousticGuitar.sf2", giengine, 1 fluidProgramSelect giengine, 1, isfnum, 0, 0 instr 1 mididefault 60, p3 midinoteonkey p4, p5 ikey init p4 ivel init p5 iInterpMethod = p6 fluidSetInterpMethod giengine, 1, iInterpMethod fluidNote giengine, 1, ikey, ivel endin instr 99 imvol init 7 asigl, asigr fluidOut giengine outs asigl*imvol, asigr*imvol endin </CsInstruments> <CsScore> ;hear the difference i 1 0 2 60 120 0 ;no interpolation i 1 3 2 72 120 0 i 1 6 2 60 120 7 ;7th order interpolation i 1 9 2 72 120 7 i 99 0 12 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/octave.csd.xml0000644000000000000000000000434212262561504020137 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o octave.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 iroot = 440 ; root note is A above middle-C (440 Hz) koct lfo 5, 1, 5 ; generate sawtooth, go from 5 octaves higher to root koc = int(koct) ; produce only whole numbers kfactor = octave(koc) ; for octave knew = iroot * kfactor printk2 knew asig pluck 1, knew, 1000, 0, 1 asig dcblock asig ;remove DC outs asig, asig endin </CsInstruments> <CsScore> i 1 0 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/date.csd.xml0000644000000000000000000000267012262561504017575 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac -iadc ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o date.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> instr 1 ii date print ii Sa dates ii prints Sa Ss dates -1 prints Ss St dates 1 prints St endin </CsInstruments> <CsScore> i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/deltapi.csd.xml0000644000000000000000000000635312262561504020304 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o deltapi.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gasig init 0 gidel = 1 ;delay time in seconds instr 1 ain pluck .7, 440, 1000, 0, 1 outs ain, ain vincr gasig, ain ;send to global delay endin instr 2 ifeedback = p4 abuf2 delayr gidel adelL deltapi .4 ;first tap (on left channel) adelM deltapi 1 ;second tap (on middle channel) delayw gasig + (adelL * ifeedback) abuf3 delayr gidel kdel line 1, p3, .01 ;vary delay time adelR deltapi .65 * kdel ;one pitch changing tap (on the right chn.) delayw gasig + (adelR * ifeedback) ;make a mix of all deayed signals outs adelL + adelM, adelR + adelM clear gasig endin </CsInstruments> <CsScore> i 1 0 1 i 1 3 1 i 2 0 3 0 ;no feedback i 2 3 8 .8 ;lots of feedback e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/sfinstr3m.csd.xml0000644000000000000000000000517112262561504020607 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -+rtmidi=virtual -M0 ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o sfinstr3m.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisf sfload "07AcousticGuitar.sf2" sfilist isf instr 1 ; play from score and midi keyboard mididefault 60, p3 midinoteonkey p4, p5 inum init p4 ivel init p5 ivel init ivel/127 ;make velocity dependent kamp linsegr 1, 1, 1, .1, 0 kamp = kamp/7000 ;scale amplitude kfreq init 1 ;do not change freq from sf aout sfinstr3m ivel, inum, kamp*ivel, kfreq, 0, gisf outs aout, aout endin </CsInstruments> <CsScore> f0 60 ; stay active for 1 minute i1 0 1 60 127 i1 + 1 62 < i1 + 1 65 < i1 + 1 69 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dust2.csd.xml0000644000000000000000000000301212262561504017710 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o oscil.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 kdens expon 2, p3, 20000 aout dust2 0.5, kdens outs aout, aout endin </CsInstruments> <CsScore> i1 0 10 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/pvsmorph.csd.xml0000644000000000000000000000532712262561504020540 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in -odac ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o pvsmorph.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 16 nchnls = 1 0dbfs = 1 ;; example written by joachim heintz 2009 giSine ftgen 0, 0, 4096, 10, 1 instr 1 iampint1 = p4 iampint2 = p5 ifrqint1 = p6 ifrqint2 = p7 kampint linseg iampint1, p3, iampint2 kfrqint linseg ifrqint1, p3, ifrqint2 ifftsize = 1024 ioverlap = ifftsize / 4 iwinsize = ifftsize iwinshape = 1; von-Hann window Sfile1 = "fox.wav" ain1 soundin Sfile1 ain2 buzz .2, 50, 100, giSine fftin1 pvsanal ain1, ifftsize, ioverlap, iwinsize, iwinshape fftin2 pvsanal ain2, ifftsize, ioverlap, iwinsize, iwinshape fmorph pvsmorph fftin1, fftin2, kampint, kfrqint aout pvsynth fmorph out aout * .5 endin </CsInstruments> <CsScore> i 1 0 3 0 0 1 1 i 1 3 3 1 0 1 0 i 1 6 3 0 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/gen51.csd.xml0000644000000000000000000001352212262561504017575 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac -M0 ;;;realtime audio out and midi input ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o gen51.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;example by Iain McCurdy sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 giEqTmp12 ftgen 1,0,128,-51,12,2,cpsoct(8),60,1,2^(1/12),2^(2/12),2^(3/12),2^(4/12),2^(5/12),2^(6/12),2^(7/12),2^(8/12),2^(9/12),2^(10/12),2^(11/12),2^(12/12) giEqTmp10 ftgen 2,0,128,-51,10,2,cpsoct(8),60,1,2^(1/10),2^(2/10),2^(3/10),2^(4/10),2^(5/10),2^(6/10),2^(7/10),2^(8/10),2^(9/10),2^(10/10) giEqTmp24 ftgen 3,0,128,-51,24,2,cpsoct(8),60,1,2^(1/24),2^(2/24),2^(3/24),2^(4/24),2^(5/24),2^(6/24),2^(7/24),2^(8/24),2^(9/24),2^(10/24),2^(11/24), \ 2^(12/24),2^(13/24),2^(14/24),2^(15/24),2^(16/24),2^(17/24),2^(18/24),2^(19/24),2^(20/24),2^(21/24),2^(22/24),2^(23/24),2^(24/24) instr 1 ;midi input instrument /*USE PITCH BEND TO MODULATE NOTE NUMBER UP OR DOWN ONE STEP - ACTUAL INTERVAL IT WILL MODULATE BY WILL BE DEPENDENT UPON THE GEN51 SCALE USED*/ ;kbend pchbend 0,2 /*ALTERNATIVELY IF USING VIRTUAL MIDI DEVICE OR A KEYBOARD WITH NO PITCH BEND WHEEL, USE CONTROLLERS 1 AND 2 TO MODULATE PITCH UP OR DOWN 1 STEP*/ kup ctrl7 1, 1, 0, 1 kdown ctrl7 1, 2, 0, -1 kbend = kup+kdown inum notnum kcps tablei inum+kbend, giEqTmp24 ;read cps values from GEN51, scale table using a combination of note played and pitch bend/controllers 1 and 2 a1 vco2 0.2, kcps, 4, 0.5 outs a1, a1 endin instr 2 ;score input instrument knum line p4, p3, p5 ;gliss using a straight line bewteen p4 and p5 for the entire note duration kcps tablei knum, giEqTmp24 ;read cps values from GEN51 scale table a1 vco2 0.2, kcps, 4, 0.5 outs a1, a1 endin </CsInstruments> <CsScore> f 0 3600 ;instr 2. Score input. Gliss from step number p4 to step number p5 ;p4 - starting note number ;p5 - ending note number i 2 0 2 60 61 i 2 + 2 70 58 i 2 + 2 66 66.5 i 2 + 2 71.25 71 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/wguide2.csd.xml0000644000000000000000000000470712262561504020231 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o wguide2.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 aout diskin2 "beats.wav", 1, 0, 1 ;in signal afreq1 line 100, p3, 2000 afreq2 line 1200, p3, p4 ;vary second frequency in the score kcutoff1 = 3000 kcutoff2 = 1500 kfeedback1 = 0.25 ;the sum of the two feedback kfeedback2 = 0.25 ;values should not exceed 0.5 asig wguide2 aout, afreq1, afreq2, kcutoff1, kcutoff2, kfeedback1, kfeedback2 asig dcblock2 asig ;get rid of DC outs asig, asig endin </CsInstruments> <CsScore> i 1 0 8 1200 ;freqency of afreq2 remains the same i 1 9 8 100 ;freqency of afreq2 gets lower e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/atonek.csd.xml0000644000000000000000000000412712262561504020140 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;RT audio out ;-iadc ;;;uncomment -iadc if RT audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o atonek.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gisin ftgen 0, 0, 2^10, 10, 1 instr 1 ksig randomh 400, 1800, 150 aout poscil .2, 1000+ksig, gisin outs aout, aout endin instr 2 ksig randomh 400, 1800, 150 khp line 1, p3, 400 ;vary high-pass ksig atonek ksig, khp aout poscil .2, 1000+ksig, gisin outs aout, aout endin </CsInstruments> <CsScore> i 1 0 5 i 2 5.5 5 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/dripwater.csd.xml0000644000000000000000000000363412262561504020662 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o dripwater.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 imaxshake = p4 ifreq = p5 ifreq1 = p6 ifreq2 = p7 ;low amplitude adrp dripwater .1, 0.09, 10, .9, imaxshake, ifreq, ifreq1, ifreq2 asig clip adrp, 2, 0.9 ; avoid drips that drip too loud outs asig, asig endin </CsInstruments> <CsScore> {100 CNT i1 [0.1 * $CNT] 0.5 0.5 430 1000 800 } e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/reverb.csd.xml0000644000000000000000000000470312262561504020144 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o reverb.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 0dbfs = 1 nchnls = 2 ga1 init 0 instr 1 asig poscil .2, cpspch(p4), 1 outs asig, asig ga1 += asig ;add direct signal to global reverb endin instr 99 ;(highest instr number executed last) arev reverb ga1, 1.5 outs arev, arev ga1 = 0 ;clear endin </CsInstruments> <CsScore> f 1 0 128 10 1 ;sine i 1 0 0.1 7.00 ;short sounds i 1 1 0.1 8.02 i 1 2 0.1 8.04 i 1 3 0.1 8.06 i 99 0 6 ;reverb runs for 6 seconds e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/fouti.csd.xml0000644000000000000000000000313712262561504020005 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o fouti.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 gihand fiopen "test.txt", 0 instr 1 ires random 0, 10 fouti gihand, 0, 1, ires ficlose gihand endin </CsInstruments> <CsScore> i 1 0 1 e </CsScore> </CsoundSynthesizer> manual_src~dfsg/examples-xml/svfilter.csd.xml0000644000000000000000000001102012262561504020503 0ustar rootroot <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o svfilter.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ; Orchestra file for resonant filter sweep of a sawtooth-like waveform. ; The seperate outputs of the filter are scaled by values from the score, ; and are mixed together. sr = 44100 kr = 2205 ksmps = 20 nchnls = 1 instr 1 idur = p3 ifreq = p4 iamp = p5 ilowamp = p6 ; determines amount of lowpass output in signal ihighamp = p7 ; determines amount of highpass output in signal ibandamp = p8 ; determines amount of bandpass output in signal iq = p9 ; value of q iharms = (sr*.4) / ifreq asig gbuzz 1, ifreq, iharms, 1, .9, 1 ; Sawtooth-like waveform kfreq linseg 1, idur * 0.5, 4000, idur * 0.5, 1 ; Envelope to control filter cutoff alow, ahigh, aband svfilter asig, kfreq, iq aout1 = alow * ilowamp aout2 = ahigh * ihighamp aout3 = aband * ibandamp asum = aout1 + aout2 + aout3 kenv linseg 0, .1, iamp, idur -.2, iamp, .1, 0 ; Simple amplitude envelope out asum * kenv endin </CsInstruments> <CsScore> f1 0 8192 9 1 1 .25 i1 0 5 100 1000 1 0 0 5 ; lowpass sweep i1 5 5 200 1000 1 0 0 30 ; lowpass sweep, octave higher, higher q i1 10 5 100 1000 0 1 0 5 ; highpass sweep i1 15 5 200 1000 0 1 0 30 ; highpass sweep, octave higher, higher q i1 20 5 100 1000 0 0 1 5 ; bandpass sweep i1 25 5 200 1000 0 0 1 30 ; bandpass sweep, octave higher, higher q i1 30 5 200 2000 .4 .6 0 ; notch sweep - notch formed by combining highpass and lowpass outputs e </CsScore> </CsoundSynthesizer> manual_src~dfsg/zak/0000755000000000000000000000000012262561504013532 5ustar rootrootmanual_src~dfsg/zak/top.xml0000644000000000000000000000513712262561504015064 0ustar rootroot Zak Patch System The zak opcodes are used to create a system for i-rate, k-rate or a-rate patching. The zak system can be thought of as a global array of variables. These opcodes are useful for performing flexible patching or routing from one instrument to another. The system is similar to a patching matrix on a mixing console or to a modulation matrix on a synthesizer. It is also useful whenever an array of variables is required. The zak system is initialized by the zakinit opcode, which is usually placed just after the other global initializations: sr, kr, ksmps, nchnls. The zakinit opcode defines two areas of memory, one area for i- and k-rate patching, and the other area for a-rate patching. The zakinit opcode may only be called once. Once the zak space is initialized, other zak opcodes can be used to read from, and write to the zak memory space, as well as perform various other tasks. Zak channels count from 0, so if you define 1 channel, the only valid channel is channel 0. Opcodes for the zak patch system are: Audio Rate: zacl, zakinit, zamod, zar, zarg, zaw and zawm. Control Rate: zkcl, zkmod, zkr, zkw, and zkwm. At initialization: zir, ziw and ziwm manual_src~dfsg/scoregens/0000755000000000000000000000000012262561504014735 5ustar rootrootmanual_src~dfsg/scoregens/gen07.xml0000644000000000000000000001147412262561504016406 0ustar rootroot GEN07 GEN07 GEN07 Constructs functions from segments of straight lines. Description Constructs functions from segments of straight lines. Syntax f # time size 7 a n1 b n2 c ... Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). a, b, c, etc. -- ordinate values, in odd-numbered pfields p5, p7, p9, . . . n1, n2, etc. -- length of segment (no. of storage locations), in even-numbered pfields. Cannot be negative, but a zero is meaningful for specifying discontinuous waveforms (e.g. in the example below). The sum n1 + n2 + .... will normally equal size for fully specified functions. If the sum is smaller, the function locations not included will be set to zero; if the sum is greater, only the first size locations will be stored. Note If p4 is positive, functions are post-normalized (rescaled to a maximum absolute value of 1 after generation). A negative p4 will cause rescaling to be skipped. Discrete-point linear interpolation implies an increase or decrease along a segment by equal differences between adjacent locations; exponential interpolation implies that the progression is by equal ratio. In both forms the interpolation from a to b is such as to assume that the value b will be attained in the n + 1th location. For discontinuous functions, and for the segment encompassing the end location, this value will not actually be reached, although it may eventually appear as a result of final scaling. Examples Here is an example of the GEN07 routine. It uses the files gen07.csd. An example of the GEN07 routine. These are the diagrams of the waveforms of the GEN07 routines, as used in the example: f 2 0 1024 7 0 512 1 0 -1 512 0 - sawtooth up and down, starting and ending at 0 f 2 0 1024 7 0 512 1 0 -1 512 0 - sawtooth up and down, starting and ending at 0 f 3 0 1024 7 1 512 1 0 -1 512 -1 - a square from positive to negative f 3 0 1024 7 1 512 1 0 -1 512 -1 - a square from positive to negative f 4 0 1024 7 1 1024 -1 - sawtooth down, a straight line from positive to negative f 4 0 1024 7 1 1024 -1 - sawtooth down, a straight line from positive to negative See Also GEN05, GEN06, and GEN08 manual_src~dfsg/scoregens/f.xml0000644000000000000000000001541012262561504015705 0ustar rootroot f statement function table statement f Statement (or Function Table Statement) f Statement (or Function Table Statement) Causes a GEN subroutine to place values in a stored function table. Description This causes a GEN subroutine to place values in a stored function table for use by instruments. Syntax f p1 p2 p3 p4 p5 ... PMAX Performance p1 -- Table number by which the stored function will be known. A negative number requests that the table be destroyed. p2 -- Action time of function generation (or destruction) in beats. p3 -- Size of function table (i.e. number of points) Must be a power of 2, or a power-of-2 plus 1 if this number is positive. Maximum table size is 16777216 (224) points. p4 -- Number of the GEN routine to be called (see GEN ROUTINES). A negative value will cause rescaling to be omitted. p5 ... PMAX -- Parameters whose meaning is determined by the particular GEN routine. Special Considerations Function tables are arrays of floating-point values. You can create a simple sine wave using the line: f 1 0 1024 10 1 This table uses GEN10 to fill the table. Historically, due to older platform constraints, Csound could only accept tables whose size was a power of two. This limitation has been removed in recent versions, and you can freely create tables of any size. However, to create a table whose size is not a power of two (or power of two plus one), you must specify the size as a negative number. Not all opcodes accept tables whose size is not a power of two, since they may depend on this for internal optimization. For arrays whose length is a power of 2, space allocation always provides for 2n points plus an additional guard point. The guard point value, used during interpolated lookup, can be automatically set to reflect the table's purpose: If size is an exact power of 2, the guard point will be a copy of the first point; this is appropriate for interpolated wrap-around lookup as in oscili, etc., and should even be used for non-interpolating oscil for safe consistency. If size is set to 2 n + 1, the guard point value automatically extends the contour of table values; this is appropriate for single-scan functions such in envplx, oscil1, oscil1i, etc. The size of the table is used as a code to tell Csound how to fill this guard-point. If the size is exactly power-of-two, then the guard point contains a copy of the first point on the table. If the size is power-of-two plus one, Csound will extend the contour of the function stored in the table for one extra point. Table space is allocated in primary memory, along with instrument data space. The maximum table number used to be 200. This has been changed to be limited by memory only. (Currently there is an internal soft limit of 300, this is automatically extended as required.) An existing function table can be removed by an f statement containing a negative p1 and an appropriate action time. A function table can also be removed by the generation of another table with the same p1. Functions are not automatically erased at the end of a score section. p2 action time is treated in the same way as in i statements with respect to sorting and modification by t statements. If an f statement and an i statement have the same p2, the sorter gives the f statement precedence so that the function table will be available during note initialization. The maximum number of p-fields accepted in the score is determined by PMAX (a compilation time varible). PMAX is currently set to 1000. This may discard values entered when using GEN02. To overcome this, use GEN23 or GEN28 to read the values from a file. An f 0 statement (zero p1, positive p2) may be used to create an action time with no associated action. Such time markers are useful for padding out a score section (see s statement) and for letting Csound run from realtime events only (e.g. using only MIDI input without score events). The time given is the number of seconds Csound will run. If you want Csound to run for 10 hours, use: f0 36000 The simplest way to fill a table (f1) with 0's is: f1 0 xx 2 0 where xx = table size. The simplest way to fill a table (f1) with *any* single value is: f1 0 xx -7 yy xx yy where xx = table size and yy = any single value In both of the above examples, table size (p3) must be a power of 2 or power-of-2 plus 1. See also GEN ROUTINES Credits Updated August 2002 thanks to a note from &namerasmus;. There is no longer a hard limit of 200 function tables. manual_src~dfsg/scoregens/gen21.xml0000644000000000000000000001664612262561504016410 0ustar rootroot GEN21 GEN21 GEN21 Generates tables of different random distributions. Description This generates tables of different random distributions. (See also betarand, bexprnd, cauchy, exprand, gauss, linrand, pcauchy, poisson, trirand, unirand, and weibull) Syntax f # time size 21 type level [arg1 [arg2]] Initialization time and size are the usual GEN function arguments. level defines the amplitude. Note that GEN21 is not self-normalizing as are most other GEN functions. type defines the distribution to be used as follow: 1 = Uniform (positive numbers only) 2 = Linear (positive numbers only) 3 = Triangular (positive and negative numbers) 4 = Exponential (positive numbers only) 5 = Biexponential (positive and negative numbers) 6 = Gaussian (positive and negative numbers) 7 = Cauchy (positive and negative numbers) 8 = Positive Cauchy (positive numbers only) 9 = Beta (positive numbers only) 10 = Weibull (positive numbers only) 11 = Poisson (positive numbers only) Of all these cases only 9 (Beta) and 10 (Weibull) need extra arguments. Beta needs two arguments and Weibull one. If type = 6, the random numbers in the ftable follow a normal distribution centered around 0.0 (mu = 0.0) with a variance (sigma) of level / 3.83. Thus more than 99.99% of the random values generated are in the range -level to +level. The default value for level is 1 (sigma = 0.261). If a mean value different of 0.0 is desired, this mean value has to be added to the generated numbers. Examples f1 0 1024 21 1 ; Uniform (white noise) f1 0 1024 21 6 ; Gaussian (mu=0.0, sigma=1/3.83=0.261) f1 0 1024 21 6 5.745 ; Gaussian (mu=0.0, sigma=5.745/3.83=1.5) f1 0 1024 21 9 1 1 2 ; Beta (note that level precedes arguments) f1 0 1024 21 10 1 2 ; Weibull All of the above additions were designed by the author between May and December 1994, under the supervision of Dr. Richard Boulanger. Here is a complete example of the GEN21 routine. It uses the file gen21.csd. Example of the GEN21 routine. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. These are the diagrams of the waveforms of the GEN21 routines, as used in the example: f 1 0 32 21 1 - positive numbers only f 1 0 32 21 1 - positive numbers only f 2 0 32 21 6 f 2 0 32 21 6 f 3 0 32 21 6 5.745 f 3 0 32 21 6 5.745 f 4 0 32 21 9 1 1 2 - positive numbers only f 4 0 32 21 9 1 1 2 - positive numbers only f 5 0 32 21 10 1 2 - positive numbers only f 5 0 32 21 10 1 2 - positive numbers only Credits Author: &nameparis; MIT, Cambridge 1995 Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK Precisions about mu and sigma added by &namepinot; after a discussion with Joachim Heintz on the Csound List, December 2010. New in Csound version 3.2 manual_src~dfsg/scoregens/gen25.xml0000644000000000000000000000714512262561504016406 0ustar rootroot GEN25 GEN25 GEN25 Construct functions from segments of exponential curves in breakpoint fashion. Description These subroutines are used to construct functions from segments of exponential curves in breakpoint fashion. Syntax f # time size 25 x1 y1 x2 y2 x3 ... Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). x1, x2, x3, etc. -- locations in table at which to attain the following y value. Must be in increasing order. If the last value is less than size, then the rest will be set to zero. Should not be negative but can be zero. y1, y2, y3,, etc. -- Breakpoint values attained at the location specified by the preceding x value. These must be non-zero and must be alike in sign. Note If p4 is positive, functions are post-normalized (rescaled to a maximum absolute value of 1 after generation). A negative p4 will cause rescaling to be skipped. Examples Here is an example of the GEN25 opcode. It uses the file gen25.csd. Example of the GEN25 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This is the diagram of the waveform of the GEN25 routine, as used in the example: f 2 0 1025 25 0 0.01 200 1 400 1 513 0.01 - a function which begins at 0.01, rises to 1 at the 200th table location, makes a straight line to the 400th location, and returns to 0.01 by the end of the table f 2 0 1025 25 0 0.01 200 1 400 1 513 0.01 - a function which begins at 0.01, rises to 1 at the 200th table location, makes a straight line to the 400th location, and returns to 0.01 by the end of the table See Also f statement, GEN27 Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK New in Csound version 3.49 manual_src~dfsg/scoregens/gen43.xml0000644000000000000000000000501412262561504016377 0ustar rootroot GEN43 GEN43 GEN43 Loads a PVOCEX file containing a PV analysis. Description This subroutine loads a PVOCEX file containing the PV analysis (amp-freq) of a soundfile and calculates the average magnitudes of all analysis frames of one or all audio channels. It then creates a table with these magnitudes for each PV bin. Syntax f # time size 43 filecod channel Initialisation size -- number of points in the table, power-of-two or power-of-two plus 1. GEN 43 does not make any distinction between these two sizes, but it requires the table to be at least the fftsize/2. PV bins cover the positive spectrum from 0Hz (table index 0) to the Nyquist (table index fftsize/2+1) in equal-size frequency increments (of size sr/fftsize). filcod -- a pvocex file (which can be generated by pvanal). channel -- audio channel number from which the magnitudes will be extracted; a 0 will average the magnitudes from all channels. Reading stops at the end of the file. Note if p4 is positive, the table will be post-normalised. A negative p4 will cause post-normalisation to be skipped. Examples f1 0 512 43 "viola.pvx" 1 f1 0 -1024 -43 "noiseprint.pvx" 0 This table can be used as a masking table for pvstencil and pvsmaska. The first example uses a 1024-point FFT phase vocoder analysis file from which the first channel is used. The second uses all channels of a 2048-point file, without post-normalisation. For noise reduction applications with pvstencil, it is easiest to skip table normalisation (negative GEN code). Credits Author: &namevictor; manual_src~dfsg/scoregens/gen20.xml0000644000000000000000000001371012262561504016374 0ustar rootroot GEN20 GEN20 GEN20 Generates functions of different windows. Description This subroutine generates functions of different windows. These windows are usually used for spectrum analysis or for grain envelopes. Syntax f # time size 20 window max [opt] Initialization size -- number of points in the table. Must be a power of 2 ( + 1). window -- Type of window to generate: 1 = Hamming 2 = Hanning 3 = Bartlett ( triangle) 4 = Blackman ( 3-term) 5 = Blackman - Harris ( 4-term) 6 = Gaussian 7 = Kaiser 8 = Rectangle 9 = Sync max -- For negative p4 this will be the absolute value at window peak point. If p4 is positive or p4 is negative and p6 is missing the table will be post-rescaled to a maximum value of 1. opt -- Optional argument required by the Gaussian window and the Kaiser window. Examples f 1 0 1024 20 5 This creates a function which contains a 4 - term Blackman - Harris window with maximum value of 1. f 1 0 1024 -20 2 456 This creates a function that contains a Hanning window with a maximum value of 456. f 1 0 1024 -20 1 This creates a function that contains a Hamming window with a maximum value of 1. f 1 0 1024 20 7 1 2 This creates a function that contains a Kaiser window with a maximum value of 1. The extra argument specifies how "open" the window is, for example a value of 0 results in a rectangular window and a value of 10 in a Hamming like window. f 1 0 1024 20 6 1 2 This creates a function that contains a Gaussian window with a maximum value of 1. The extra argument specifies how broad the window is, as the standard deviation of the curve; in this example the s.d. is 2. The default value is 1. For all diagrams, see Window Functions Here is an example of the GEN20 routine. It uses the file gen20.csd. Example of the GEN20 routine. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. These are the diagrams of the waveforms of the GEN20 routines, as used in the example: f 1 0 512 20 2 - Hanning window f 1 0 512 20 2 - Hanning window f 2 0 512 20 6 1 - Gaussian window f 2 0 512 20 6 1 - Gaussian window Credits Author: &nameparis; MIT, Cambridge 1995 Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK New in Csound version 3.2 Optional argument to Gaussian added in 5.10 manual_src~dfsg/scoregens/gen24.xml0000644000000000000000000000414612262561504016403 0ustar rootroot GEN24 GEN24 GEN24 Reads numeric values from another allocated function-table and rescales them. Description This subroutine reads numeric values from another allocated function-table and rescales them according to the max and min values given by the user. Syntax f # time size -24 ftable min max Initialization #, time, size -- the usual GEN parameters. See f statement. ftable -- ftable must be an already allocated table with the same size as this function. min, max -- the rescaling range. Note This GEN is useful, for example, to eliminate the starting offset in exponential segments allowing a real starting from zero. Examples Here is an example of the GEN24 opcode. It uses the file gen24.csd. Example of the GEN24 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; New in Csound version 4.16 manual_src~dfsg/scoregens/genfarey.xml0000644000000000000000000002334512262561504017266 0ustar rootroot GENfarey GENfarey "farey" Fills a table with the Farey Sequence Fn of the integer n. Description A Farey Sequence Fn of order n is a list of fractions in their lowest terms between 0 and 1 and in ascending order. Their denominators do not exceed n. This means a fraction a/b belongs to Fn if 0 ≤ a ≤ b ≤ n. The numerator and denominator of each fraction are always coprime. 0 and 1 are included in Fn as the fractions 0/1 and 1/1. For example F5 = {0/1, 1/5, 1/4, 1/3, 2/5, 1/2, 3/5, 2/3, 3/4, 4/5, 1/1} Some properties of the Farey Sequence: If a/b and c/d are two successive terms of Fn, then bc - ad = 1. If a/b, c/d, e/f are three successive terms of Fn, then: c/d = (a+e) / (b+f). In this case c/d is called the mediant fraction between a/b and e/f. If n > 1, then no two successive terms ofFn have the same denominator. The length of any Farey Sequence Fn is determined by |Fn| = 1 + SUM over n (phi(m)) where phi(m) is Euler's totient function, which gives the number of integers ≤ m that are coprime to m. Some values for the length of Fn given n: n Fn 1 2 2 3 3 5 4 7 5 11 6 13 7 19 8 23 9 29 10 33 11 43 12 47 13 59 14 65 15 73 16 81 17 97 18 103 19 121 20 129 Syntax f # time size "farey" fareynum mode Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). fareynum -- the integer n for generating Farey Sequence Fn mode -- integer to trigger a specific output to be written into the table: 0 -- outputs floating point numbers representing the elements of Fn. 1 -- outputs delta values of successive elements of Fn, useful for generating note durations for example. 2 -- outputs only the denominators of the integer ratios, useful for indexing other tables or instruments for example. 3 -- same as mode 2 but with normalised output. 4 -- same as mode 0 but with 1 added to each number, useful for generating tables for tuning opcodes, for example cps2pch. Examples f1 0 -23 "farey" 8 0 Generates generates Farey Sequence F8. The table contains all 23 elements of F8 as floating point numbers. f1 0 -18 "farey" 7 1 This generates Farey Sequence F7. The table contains 18 delta values of F7, i.e. the difference between ri+1 - ri, where r is the ith element of Fn. f1 0 -43 "farey" 11 2 This generates Farey Sequence F11. The table contains the denominators of all 43 fractions in F11. f1 0 -43 "farey" 11 3 This generates Farey Sequence F11. The table contains the denominators of all 43 fractions in F11, each of those divided by 11, i.e. normalised. f1 0 -18 "farey" 7 4 This generates Farey Sequence F7. The table contains all fractions of F7, same as mode 0, but this time '1' is added to each table element. A simple example of the GENfarey routine. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is a complete example of the GENfarey routine. It uses the files genfarey-2.csd. Another example of the GENfarey routine. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. These are the diagrams of the waveforms of the GENfarey routines, as used in the example: gidelta ftgen 100,0,-18,"farey",7,1 - delta values of Farey Sequence 7 gidelta ftgen 100,0,-18,"farey",7,1 - delta values of Farey Sequence 7 gimult ftgen 101,0,-18,"farey",7,2 - generate the denominators of fractions of F_7 gimult ftgen 101,0,-18,"farey",7,2 - generate the denominators of fractions of F_7 Credits Author: Georg Boenn University of Glamorgan 2010 New in Csound version 5.13 manual_src~dfsg/scoregens/gen06.xml0000644000000000000000000001217512262561504016404 0ustar rootroot GEN06 GEN06 GEN06 Generates a function comprised of segments of cubic polynomials. Description This subroutine will generate a function comprised of segments of cubic polynomials, spanning specified points just three at a time. Syntax f # time size 6 a n1 b n2 c n3 d ... Initialization size -- number of points in the table. Must be a power off or power-of-2 plus 1 (see f statement). a, c, e, ... -- local maxima or minima of successive segments, depending on the relation of these points to adjacent inflexions. May be either positive or negative. b, d, f, ... -- ordinate values of points of inflexion at the ends of successive curved segments. May be positive or negative. n1, n2, n3 ... -- number of stored values between specified points. Cannot be negative, but a zero is meaningful for specifying discontinuities. The sum n1 + n2 + ... will normally equal size for fully specified functions. (for details, see GEN05). Note GEN06 constructs a stored function from segments of cubic polynomial functions. Segments link ordinate values in groups of 3: point of inflexion, maximum/minimum, point of inflexion. The first complete segment encompasses b, c, d and has length n2 + n3, the next encompasses d, e, f and has length n4 + n5, etc. The first segment (a, b with length n1) is partial with only one inflexion; the last segment may be partial too. Although the inflexion points b, d, f ... each figure in two segments (to the left and right), the slope of the two segments remains independent at that common point (i.e. the 1st derivative will likely be discontinuous). When a, c, e... are alternately maximum and minimum, the inflexion joins will be relatively smooth; for successive maxima or successive minima the inflexions will be comb-like. Examples Here is an example of the GEN06 routine. It uses the files gen06.csd. An example of the GEN06 routine. These are the diagrams of the waveforms of the GEN06 routines, as used in the example: f 2 0 513 6 1 128 -1 128 1 64 -.5 64 .5 16 -.5 8 1 16 -.5 8 1 16 -.5 84 1 16 -.5 8 .1 16 -.1 17 0 - a not-so-smooth curve f 2 0 513 6 1 128 -1 128 1 64 -.5 64 .5 16 -.5 8 1 16 -.5 8 1 16 -.5 84 1 16 -.5 8 .1 16 -.1 17 0 - a not-so-smooth curve f 3 0 513 6 0 128 0.5 128 1 128 0 129 -1 - a curve running 0 to 1 to -1, with a minimum, maximum and minimum at these values respectively. Inflexions are at .5 and 0 and are relatively smooth f 3 0 513 6 0 128 0.5 128 1 128 0 129 -1 - a curve running 0 to 1 to -1, with a minimum, maximum and minimum at these values respectively. Inflexions are at .5 and 0 and are relatively smooth See Also GEN05, GEN07, and GEN08 manual_src~dfsg/scoregens/v.xml0000644000000000000000000000641112262561504015726 0ustar rootroot v statement v Statement v Provides for locally variable time warping of score events. Description The v statement provides for locally variable time warping of score events. Syntax v p1 Initialization p1 -- Time warp factor (must be positive). Performance The v statement takes effect with the following i statement, and remains in effect until the next v statement, s statement, or e statement. Examples The value of p1 is used as a multiplier for the start times (p2) of subsequent i statements. i1 0 1 ; note1 v2 i1 1 1 ; note2 In this example, the second note occurs two beats after the first note, and is twice as long. Although the v statement is similar to the t statement, the v statement is local in operation. That is, v affects only the following notes, and its effect may be cancelled or changed by another v statement. Carried values are unaffected by the v statement (see Carry). i1 0 1 v2 i. + . i. . . In this example, the v statement has no effect. Examples Here is an example of the v statement. It uses the file v.csd. Example of the v statement. In this example, note3 and note5 occur simultaneously, while note4 actually occurs before note3, that is, at its original place. Durations are unaffected. manual_src~dfsg/scoregens/gen41.xml0000644000000000000000000000554112262561504016402 0ustar rootroot GEN41 GEN41 GEN41 Generates a random list of numerical pairs. Description Generates a discrete random distribution function by giving a list of numerical pairs. Syntax f # time size -41 value1 prob1 value2 prob2 value3 prob3 ... valueN probN Performance The first number of each pair is a value, and the second is the probability of that value to be chosen by a random algorithm. Even if any number can be assigned to the probability element of each pair, it is suggested to give it a percent value, in order to make it clearer for the user. This subroutine is designed to be used together with duserrnd and urd opcodes (see duserrnd for more information). Examples Here is an example of the GEN41 opcode. It uses the file gen41.csd. Example of the GEN41 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i 1 time 0.00067: 8.00000 i 1 time 0.00133: 8.00000 i 1 time 0.00200: 8.00000 i 1 time 0.00267: 8.00000 i 1 time 0.00333: 2.00000 i 1 time 0.00400: 8.00000 i 1 time 0.00533: 8.00000 i 1 time 0.00600: 8.00000 .......... This is the diagram of the waveform of the GEN41 routine, as used in the example: f 1 0 -20 -41 2 .1 8 .9 f 1 0 -20 -41 2 .1 8 .9 Credits Author: &namegabriel; manual_src~dfsg/scoregens/gen03.xml0000644000000000000000000001472412262561504016403 0ustar rootroot GEN03 GEN03 GEN03 Generates a stored function table by evaluating a polynomial. Description This subroutine generates a stored function table by evaluating a polynomial in x over a fixed interval and with specified coefficients. Syntax f # time size 3 xval1 xval2 c0 c1 c2 ... cn Initialization size -- number of points in the table. Must be a power of 2 or a power-of-2 plus 1. xval1, xval2 -- left and right values of the x interval over which the polynomial is defined (xval1 < xval2). These will produce the 1st stored value and the (power-of-2 plus l)th stored value respectively in the generated function table. c0, c1, c2, ..., cn -- coefficients of the nth-order polynomial C0 + C1x + C2x2 + . . . + Cnxn Coefficients may be positive or negative real numbers; a zero denotes a missing term in the polynomial. The coefficient list begins in p7, providing a current upper limit of 144 terms. Note The defined segment [fn(xval1), fn(xval2)] is evenly distributed. Thus a 512-point table over the interval [-1,1] will have its origin at location 257 (at the start of the 2nd half). Provided the extended guard point is requested, both fn(-1) and fn(1) will exist in the table. GEN03 is useful in conjunction with table or tablei for audio waveshaping (sound modification by non-linear distortion). Coefficients to produce a particular formant from a sinusoidal lookup index of known amplitude can be determined at preprocessing time using algorithms such as Chebyshev formulae. See also GEN13. Examples Here is an example of the GEN03 opcode. It uses the file gen03.csd. Example of the GEN03 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. These are the diagrams of the waveforms of the GEN03 routines, as used in the example: f4 0 513 3 1 1 0 1 - first-order Chebyshev: x f4 0 513 3 1 1 0 1 - first-order Chebyshev: x f6 0 513 3 -1 1 -1 0 2 - second-order Chebyshev: 2x2 - 1 f6 0 513 3 -1 1 -1 0 2 - second-order Chebyshev: 2x2 - 1 f8 0 513 3 -1 1 0 -3 0 4 - third-order Chebyshev: 4x3 - 3x f8 0 513 3 -1 1 0 -3 0 4 - third-order Chebyshev: 4x3 - 3x f10 0 513 3 -1 10 0 -7 0 56 0 -112 0 64 - seventh-order Chebyshev: 64x7 - 112x5 + 56x3 - 7x f10 0 513 3 -1 10 0 -7 0 56 0 -112 0 64 - seventh-order Chebyshev: 64x7 - 112x5 + 56x3 - 7x f12 0 513 3 -1 1 5 4 3 2 2 1 - a 4th order polynomial function over the x-interval -1 to 1 f12 0 513 3 -1 1 5 4 3 2 2 1 - a 4th order polynomial function over the x-interval -1 to 1 See Also GEN13, GEN14, and GEN15. Information about the Chebyshev polynomials on Wikipedia: http://en.wikipedia.org/wiki/Chebyshev_polynomials manual_src~dfsg/scoregens/a.xml0000644000000000000000000000523012262561504015677 0ustar rootroot a statement advance statement a Statement (or Advance Statement) a Advance score time by a specified amount. Description This causes score time to be advanced by a specified amount without producing sound samples. Syntax a p1 p2 p3 Performance p1 Carries no meaning. Usually zero. p2 Action time, in beats, at which advance is to begin. p3 Number of beats to advance without producing sound. p4 | p5 | These carry no meaning. p6 | . . Special Considerations This statement allows the beat count within a score section to be advanced without generating intervening sound samples. This can be of use when a score section is incomplete (the beginning or middle is missing) and the user does not wish to generate and listen to a lot of silence. p2, action time, and p3, number of beats, are treated as in i statements, with respect to sorting and modification by t statements. An a statement will be temporarily inserted in the score by the Score Extract feature when the extracted segment begins later than the start of a Section. The purpose of this is to preserve the beat count and time count of the original score for the benefit of the peak amplitude messages which are reported on the user console. Whenever an a statement is encountered by a performing orchestra, its presence and effect will be reported on the user's console. Examples Here is an example of the a statement. It uses the file a.csd. Example of the a statement. manual_src~dfsg/scoregens/gen51.xml0000644000000000000000000001335712262561504016407 0ustar rootroot GEN51 GEN51 GEN51 This subroutine fills a table with a fully customized micro-tuning scale, in the manner of Csound opcodes cpstun, cpstuni and cpstmid. Description This subroutine fills a table with a fully customized micro-tuning scale, in the manner of Csound opcodes cpstun, cpstuni et cpstmid. Syntax f # time size -51 numgrades interval basefreq basekey tuningRatio1 tuningRatio2 .... tuningRationN Performance The first four parameters (i.e. p5, p6, p7 and p8) define the following generation directives: p5 (numgrades) -- the number of grades of the micro-tuning scale p6 (interval) -- the frequency range covered before repeating the grade ratios, for example 2 for one octave, 1.5 for a fifth etcetera p7 (basefreq) -- the base frequency of the scale in cps p8 (basekey) -- the integer index of the scale to which to assign basefreq unmodified The other parameters define the ratios of the scale: p9...pN (tuningRatio1...etc.) -- the tuning ratios of the scale For example, for a standard 12-grade scale with the base-frequency of 261 cps assigned to the key-number 60, the corresponding f-statement in the score to generate the table should be: ; numgrades basefreq tuning-ratios (eq.temp) ....... ; interval basekey f1 0 64 -51 12 2 261 60 1 1.059463 1.12246 1.18920 ..etc... After the gen has been processed, the table f1 is filled with 64 different frequency values. The 60th element is filled with the frequency value of 261, and all other elements (preceding and subsequents) of the table are filled according to the tuning ratios Another example with a 24-grade scale with a base frequency of 440 assigned to the key-number 48, and a repetition interval of 1.5: ; numgrades basefreq tuning-ratios ..... ; interval basekey f1 0 64 -51 24 1.5 440 48 1 1.01 1.02 1.03 ..etc... Examples Here is an example of the GEN51 routine. It uses the files gen51.csd. An example of the GEN51 routine. These are the diagrams of the waveforms of the GEN51 routines, as used in the example: f 1 0 128 -51 12 2 cpsoct(8) 60 1 2^(1/12) 2^(2/12) 2^(3/12) 2^(4/12) 2^(5/12) 2^(6/12) 2^(7/12) 2^(8/12) 2^(9/12) 2^(10/12) 2^(11/12) 2^(12/12) f 1 0 128 -51 12 2 cpsoct(8) 60 1 2^(1/12) 2^(2/12) 2^(3/12) 2^(4/12) 2^(5/12) 2^(6/12) 2^(7/12) 2^(8/12) 2^(9/12) 2^(10/12) 2^(11/12) 2^(12/12) f 2 0 128 -51 10 2 cpsoct(8) 60 1 2^(1/10) 2^(2/10) 2^(3/10) 2^(4/10) 2^(5/10) 2^(6/10) 2^(7/10) 2^(8/10) 2^(9/10) 2^(10/10) f 2 0 128 -51 10 2 cpsoct(8) 60 1 2^(1/10) 2^(2/10) 2^(3/10) 2^(4/10) 2^(5/10) 2^(6/10) 2^(7/10) 2^(8/10) 2^(9/10) 2^(10/10) f 3 0 128 -51 24 2 cpsoct(8) 60 1 2^(1/24) 2^(2/24) 2^(3/24) 2^(4/24) 2^(5/24) 2^(6/24) 2^(7/24) 2^(8/24) 2^(9/24) 2^(10/24) 2^(11/24) 2^(12/24) 2^(13/24) 2^(14/24) 2^(15/24) 2^(16/24) 2^(17/24) 2^(18/24) 2^(19/24) 2^(20/24) 2^(21/24) 2^(22/24) 2^(23/24) 2^(24/24) f 3 0 128 -51 24 2 cpsoct(8) 60 1 2^(1/24) 2^(2/24) 2^(3/24) 2^(4/24) 2^(5/24) 2^(6/24) 2^(7/24) 2^(8/24) 2^(9/24) 2^(10/24) 2^(11/24) 2^(12/24) 2^(13/24) 2^(14/24) 2^(15/24) 2^(16/24) 2^(17/24) 2^(18/24) 2^(19/24) 2^(20/24) 2^(21/24) 2^(22/24) 2^(23/24) 2^(24/24) Credits Author: &namegabriel; manual_src~dfsg/scoregens/gensone.xml0000644000000000000000000000526312262561504017123 0ustar rootroot GENsone GENsone "sone" Generate a table with values of the sone function. Description Creates an ftable with values of the sone function for equal power. Syntax f # time size "sone" start end equalpoint rescale Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). start, end -- first and last value to be stored. The points stored are uniformly spaced between these to the table size. equalpoint -- the point on the curve when the input and output values are equal. rescale -- if not zero the table is not rescaled The table is filled with the function x*POWER(x/eqlp, FL(33.0)/FL(78.0)) for x between the start and end points. This is the Sone loudness curve. Examples Here is an example of the GENsone routine. It uses the files gensone.csd. An example of the GENsone routine. This is the diagram of the waveforms of the GENsone routine, as used in the example: f 2 0 16385 "sone" 0 32000 32000 0 f 2 0 16385 "sone" 0 32000 32000 0 See Also More information on Sone: http://en.wikipedia.org/wiki/Sone Credits Written by &namevictor; manual_src~dfsg/scoregens/gen05.xml0000644000000000000000000001116412262561504016400 0ustar rootroot GEN05 GEN05 GEN05 Constructs functions from segments of exponential curves. Description Constructs functions from segments of exponential curves. Syntax f # time size 5 a n1 b n2 c ... Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). a, b, c, etc. -- ordinate values, in odd-numbered pfields p5, p7, p9, . . . These must be nonzero and must be alike in sign. n1, n2, etc. -- length of segment (no. of storage locations), in even-numbered pfields. Cannot be negative, but a zero is meaningful for specifying discontinuous waveforms. The sum n1 + n2 + .... will normally equal size for fully specified functions. If the sum is smaller, the function locations not included will be set to zero; if the sum is greater, only the first size locations will be stored. Note If p4 is positive, functions are post-normalized (rescaled to a maximum absolute value of 1 after generation). A negative p4 will cause rescaling to be skipped. Discrete-point linear interpolation implies an increase or decrease along a segment by equal differences between adjacent locations; exponential interpolation implies that the progression is by equal ratio. In both forms the interpolation from a to b is such as to assume that the value b will be attained in the n + 1th location. For discontinuous functions, and for the segment encompassing the end location, this value will not actually be reached, although it may eventually appear as a result of final scaling. Examples Here is a simple example of the GEN05 routine. It uses the files gen05.csd. An example of the GEN05 routine. These are the diagrams of the waveforms of the GEN05 routines, as used in the example: f 2 0 129 5 1 100 0.0001 29 - waveform that goes over 100 points from 1 to 0.0001, stay there for 29 points f 2 0 129 5 1 100 0.0001 29 - waveform that goes over 100 points from 1 to 0.0001, stay there for 29 points f 3 0 129 5 0.00001 87 1 22 .5 20 0.0001 - waveform that goes from 0.00001 to 1 in 87 points, then from 1 to .5 in 22 points and then from .5 to 0.0001 in 20 points f 3 0 129 5 0.00001 87 1 22 .5 20 0.0001 - waveform that goes from 0.00001 to 1 in 87 points, then from 1 to .5 in 22 points and then from .5 to 0.0001 in 20 points See Also GEN06, GEN07, and GEN08 manual_src~dfsg/scoregens/gen31.xml0000644000000000000000000000375012262561504016401 0ustar rootroot GEN31 GEN31 GEN31 Mixes any waveform specified in an existing table. Description This routine is similar to GEN09, but allows mixing any waveform specified in an existing table. Syntax f # time size 31 src pna stra phsa pnb strb phsb ... Performance src -- source table number pna, pnb, ... -- partial number, must be a positive integer stra, strb, ... -- amplitude scale phsa, phsb, ... -- start phase (0 to 1) GEN31 does not support tables with an extended guard point (ie. table size = power of two + 1). Although such tables will work both for input and output, when reading source table(s), the guard point is ignored, and when writing the output table, guard point is simply copied from the first sample (table index = 0). The reason of this limitation is that GEN31 uses FFT, which requires power of two table size. GEN32 allows using linear interpolation for resampling and phase shifting, which makes it possible to use any table size (however, for partials calculated with FFT, the power of two limitation still exists). Credits Author: &nameistvan; New in version 4.15 manual_src~dfsg/scoregens/gen09.xml0000644000000000000000000001302212262561504016377 0ustar rootroot GEN09 GEN09 GEN09 Generate composite waveforms made up of weighted sums of simple sinusoids. Description These subroutines generate composite waveforms made up of weighted sums of simple sinusoids. The specification of each contributing partial requires 3 p-fields using GEN09. Syntax f # time size 9 pna stra phsa pnb strb phsb ... Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). pna, pnb, etc. -- partial no. (relative to a fundamental that would occupy size locations per cycle) of sinusoid a, sinusoid b, etc. Must be positive, but need not be a whole number, i.e., non-harmonic partials are permitted. Partials may be in any order. stra, strb, etc. -- strength of partials pna, pnb, etc. These are relative strengths, since the composite waveform may be rescaled later. Negative values are permitted and imply a 180 degree phase shift. phsa, phsb, etc. -- initial phase of partials pna, pnb, etc., expressed in degrees (0-360). Note These subroutines generate stored functions as sums of sinusoids of different frequencies. The two major restrictions on GEN10 that the partials be harmonic and in phase do not apply to GEN09 or GEN19. In each case the composite wave, once drawn, is then rescaled to unity if p4 was positive. A negative p4 will cause rescaling to be skipped. Examples Here is an example of the GEN09 routine. It uses the files gen09.csd. Example of the GEN09 routine. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. These are the diagrams of the waveforms of the GEN09 routines, as used in the example: gi1 ftgen 1,0,2^10,9, 1,3,0, 3,1,0, 9,0.333,180 - approximation of a square wave gi1 ftgen 1,0,2^10,9, 1,3,0, 3,1,0, 9,0.333,180 - approximation of a square wave gi2 ftgen 2,0,2^10,9, 1,3,180, 3,1,0, 9,0.333,0 - same values as gi1, except phase value gi2 ftgen 2,0,2^10,9, 1,3,180, 3,1,0, 9,0.333,0 - same values as gi1, except phase value gi3 ftgen 3,0,2^10,9, 1,2,0, 3,2,0, 9,0.333,180 - inharmonic partials, but with distortion due to the sudden jump in ending and beginning of the wave gi3 ftgen 3,0,2^10,9, 1,2,0, 3,2,0, 9,0.333,180 - inharmonic partials, but with distortion due to the sudden jump in ending and beginning of the wave gi4 ftgen 4,0,2^10,9, 1,2,180, 3,2,0, 9,0.333,0 - same ratio as gi3, except with less artefacts gi4 ftgen 4,0,2^10,9, 1,2,180, 3,2,0, 9,0.333,0 - same ratio as gi3, except with less artefacts See Also GEN10, GEN19 manual_src~dfsg/scoregens/gen15.xml0000644000000000000000000001451512262561504016404 0ustar rootroot GEN15 GEN15 GEN15 Creates two tables of stored polynomial functions. Description This subroutine creates two tables of stored polynomial functions, suitable for use in phase quadrature operations. Syntax f # time size 15 xint xamp h0 phs0 h1 phs1 h2 phs2 ... Initialization size -- number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement). The normal value is power-of-2 plus 1. xint -- provides the left and right values [-xint, +xint] of the x interval over which the polynomial is to be drawn. This subroutine will eventually call GEN03 to draw both functions; this p5 value is therefor expanded to a negative-positive p5, p6 pair before GEN03 is actually called. The normal value is 1. xamp -- amplitude scaling factor of the sinusoid input that is expected to produce the following spectrum. h0, h1, h2, ... hn -- relative strength of partials 0 (DC), 1 (fundamental), 2 ... that will result when a sinusoid of amplitude xamp * int(size/2)/xint is waveshaped using this function table. These values thus describe a frequency spectrum associated with a particular factor xamp of the input signal. phs0, phs1, ... -- phase in degrees of desired harmonics h0, h1, ... when the two functions of GEN15 are used with phase quadrature. Notes GEN15 creates two tables of equal size, labeled f # and f # + 1. Table # will contain a Chebyshev function of the first kind, drawn using GEN13 with partial strengths h0cos(phs0), h1cos(phs1), ... Table #+1 will contain a Chebyshev function of the 2nd kind by calling GEN14 with partials h1sin(phs1), h2sin(phs2),... (note the harmonic displacement). The two tables can be used in conjunction in a waveshaping network that exploits phase quadrature. Before version 5.16 there was a bug (pointed out by Menno Knevel and fixed by &namepinot;) on the number of pfields transmitted to gen13 and gen14 by gen15. The consequence is that all the csd, or orc and sco files that used gen15 before this bug was fixed, are likely to sound different now. Examples Here is an example of the GEN15 routine. It uses the files gen15.csd. An example of the GEN15 routine. These are the diagrams of the waveforms of the GEN15 routine, as used in the example (in each diagram, the curve in blue is for ftable 33 and the curve in green is for table 34): f 33 0 8193 -15 1 1 1 0 1 180 .8 45 .6 270 .5 90 .4 225 .2 135 .1 315 f 33 0 8193 -15 1 1 1 0 1 180 .8 45 .6 270 .5 90 .4 225 .2 135 .1 315 f 33 0 8193 -15 1 1 1 0 1 0 1 180 1 180 1 0 1 0 1 180 1 180 1 0 1 0 1 180 1 180 f 33 0 8193 -15 1 1 1 0 1 0 1 180 1 180 1 0 1 0 1 180 1 180 1 0 1 0 1 180 1 180 f 33 0 8193 -15 1 1 1 0 1 0 .9 180 .5 270 .75 90 .4 45 .2 225 .1 0 f 33 0 8193 -15 1 1 1 0 1 0 .9 180 .5 270 .75 90 .4 45 .2 225 .1 0 f 33 0 8193 -15 1 1 1 0 1 0 .5 0 .9 0 .3 0 .75 0 .2 180 .6 180 .15 180 .5 180 .1 180 f 33 0 8193 -15 1 1 1 0 1 0 .5 0 .9 0 .3 0 .75 0 .2 180 .6 180 .15 180 .5 180 .1 180 f 33 0 8193 -15 1 1 1 180 1 180 .5 180 .9 180 .3 180 .75 180 .2 0 .6 0 .15 0 .5 0 .1 0 f 33 0 8193 -15 1 1 1 180 1 180 .5 180 .9 180 .3 180 .75 180 .2 0 .6 0 .15 0 .5 0 .1 0 See Also GEN13, and GEN14. manual_src~dfsg/scoregens/n.xml0000644000000000000000000000507012262561504015716 0ustar rootroot n statement n Statement n Repeats a section. Description Repeats a section from the referenced m statement. Syntax n p1 Initialization p1 -- Name of mark to repeat. Performance This can be helpful in setting a up verse and chorus structure in the score. Names may contain letters and numerals. For example, the following score: m foo i1 0 1 i1 1 1.5 i1 2.5 2 s i1 0 10 s n foo e Will be passed from the preprocessor to Csound as: i1 0 1 i1 1 1.5 i1 2.5 2 s i1 0 10 s ;; this is named section repeated i1 0 1 i1 1 1.5 i1 2.5 2 s ;; end of named section e Examples See the example for the m statement. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April 1998 New in Csound version 3.48 manual_src~dfsg/scoregens/t.xml0000644000000000000000000000656412262561504015735 0ustar rootroot t statement tempo statement t Statement (Tempo Statement) t Sets the tempo. Description This statement sets the tempo and specifies the accelerations and decelerations for the current section. This is done by converting beats into seconds. Syntax t p1 p2 p3 p4 ... (unlimited) Initialization p1 -- Must be zero. p2 -- Initial tempo on beats per minute. p3, p5, p7,... -- Times in beats (in non-decreasing order). p4, p6, p8,... -- Tempi for the referenced beat times. Performance Time and Tempo-for-that-time are given as ordered couples that define points on a "tempo vs. time" graph. (The time-axis here is in beats so is not necessarily linear.) The beat-rate of a Section can be thought of as a movement from point to point on that graph: motion between two points of equal height signifies constant tempo, while motion between two points of unequal height will cause an accelarando or ritardando accordingly. The graph can contain discontinuities: two points given equal times but different tempi will cause an immediate tempo change. Motion between different tempos over non-zero time is inverse linear. That is, an accelerando between two tempos M1 and M2 proceeds by linear interpolation of the single-beat durations from 60/M1 to 60/M2. The first tempo given must be for beat 0. A tempo, once assigned, will remain in effect from that time-point unless influenced by a succeeding tempo, i.e. the last specified tempo will be held to the end of the section. A t statement applies only to the score section in which it appears. Only one t statement is meaningful in a section; it can be placed anywhere within that section. If a score section contains no t statement, then beats are interpreted as seconds (i.e. with an implicit t 0 60 statement). N.B. If the CSound command includes a -t flag, the interpreted tempo of all score t statements will be overridden by the command-line tempo. Examples Here is an example of the t statement. It uses the file t.csd. Example of the t statement. manual_src~dfsg/scoregens/topXO.xml0000644000000000000000000003046412262561504016537 0ustar rootroot Score Statements and GEN Routines

Score Statements The statements used in scores are: a - Advance score time by a specified amount b - Resets the clock e - Marks the end of the last section of the score f - Causes a GEN subroutine to place values in a stored function table i - Makes an instrument active at a specific time and for a certain duration m - Sets a named mark in the score n - Repeats a marked section q - Used to quiet an instrument r - Starts a repeated section s - Marks the end of a section t - Sets the tempo v - Provides for locally variable time warping of score events x - Skip the rest of the current section { - Begins a non-sectional, nestable loop } - Ends a non-sectional, nestable loop
&scoregensa; &scoregensb; &scoregense; &scoregensf; &scoregensi; &scoregensm; &scoregensn; &scoregensq; &scoregensr; &scoregenss; &scoregenst; &scoregensv; &scoregensx; &scoregensleftbrace; &scoregensrightbrace;
GEN Routines GEN routines are used as data generators for function tables. When a function table is created using the f score statement the GEN function is given as its fourth argument. A negative GEN number implies that the function is not rescaled, and maintains its original values. Sine/Cosine Generators: GEN09 - Composite waveforms made up of weighted sums of simple sinusoids. GEN10 - Composite waveforms made up of weighted sums of simple sinusoids. GEN11 - Additive set of cosine partials. GEN19 - Composite waveforms made up of weighted sums of simple sinusoids. GEN30 - Generates harmonic partials by analyzing an existing table. GEN33 - Generate composite waveforms by mixing simple sinusoids. GEN34 - Generate composite waveforms by mixing simple sinusoids. Line/Exponential Segment Generators: GEN05 - Constructs functions from segments of exponential curves. GEN06 - Generates a function comprised of segments of cubic polynomials. GEN07 - Constructs functions from segments of straight lines. GEN08 - Generate a piecewise cubic spline curve. GEN16 - Creates a table from a starting value to an ending value. GEN25 - Construct functions from segments of exponential curves in breakpoint fashion. GEN27 - Construct functions from segments of straight lines in breakpoint fashion. File Access GEN Routines: GEN01 - Transfers data from a soundfile into a function table. GEN23 - Reads numeric values from a text file. GEN28 - Reads a text file which contains a time-tagged trajectory. Numeric Value Access GEN Routines GEN02 - Transfers data from immediate pfields into a function table. GEN17 - Creates a step function from given x-y pairs. GEN52 - Creates an interleaved multichannel table from the specified source tables, in the format expected by the ftconv opcode. Window Function GEN Routines GEN20 - Generates functions of different windows. Random Function GEN Routines GEN21 - Generates tables of different random distributions. GEN40 - Generates a random distribution using a distribution histogram. GEN41 - Generates a random list of numerical pairs. GEN42 - Generates a random distribution of discrete ranges of values. GEN43 - Loads a PVOCEX file containing a PV analysis. Waveshaping GEN Routines GEN03 - Generates a stored function table by evaluating a polynomial. GEN13 - Stores a polynomial whose coefficients derive from the Chebyshev polynomials of the first kind. GEN14 - Stores a polynomial whose coefficients derive from Chebyshevs of the second kind. GEN15 - Creates two tables of stored polynomial functions. Amplitude Scaling GEN Routines GEN04 - Generates a normalizing function. GEN12 - Generates the log of a modified Bessel function of the second kind. GEN24 - Reads numeric values from another allocated function-table and rescales them. Mixing GEN Routines GEN18 - Writes composite waveforms made up of pre-existing waveforms. GEN31 - Mixes any waveform specified in an existing table. GEN32 - Mixes any waveform, resampled with either FFT or linear interpolation. Pitch and Tuning GEN Routines GEN51 - fills a table with a fully customized micro-tuning scale, in the manner of Csound opcodes cpstun, cpstuni and cpstmid. Named GEN Routines Csound's GEN routines can be extended with GEN function plugins. There is currently a simple GEN plugin that provides exponential and hyperbolic tangent functions. These GEN functions are not called by number, but by name. "tanh" - fills a table from a hyperbolic tangent formula. "exp" - fills a table from a hyperbolic tangent formula.
&scoregensgen01; &scoregensgen02; &scoregensgen03; &scoregensgen04; &scoregensgen05; &scoregensgen06; &scoregensgen07; &scoregensgen08; &scoregensgen09; &scoregensgen10; &scoregensgen11; &scoregensgen12; &scoregensgen13; &scoregensgen14; &scoregensgen15; &scoregensgen16; &scoregensgen17; &scoregensgen18; &scoregensgen19; &scoregensgen20; &scoregensgen21; &scoregensgen23; &scoregensgen24; &scoregensgen25; &scoregensgen27; &scoregensgen28; &scoregensgen30; &scoregensgen31; &scoregensgen32; &scoregensgen33; &scoregensgen34; &scoregensgen40; &scoregensgen41; &scoregensgen42; &scoregensgen43; &scoregensgen51; &scoregensgen52; manual_src~dfsg/scoregens/gen11.xml0000644000000000000000000001160112262561504016371 0ustar rootroot GEN11 GEN11 GEN11 Generates an additive set of cosine partials. Description This subroutine generates an additive set of cosine partials, in the manner of Csound generators buzz and gbuzz. Syntax f # time size 11 nh [lh] [r] Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). nh -- number of harmonics requested. Must be positive. lh(optional) -- lowest harmonic partial present. Can be positive, zero or negative. The set of partials can begin at any partial number and proceeds upwards; if lh is negative, all partials below zero will reflect in zero to produce positive partials without phase change (since cosine is an even function), and will add constructively to any positive partials in the set. The default value is 1 r(optional) -- multiplier in an amplitude coefficient series. This is a power series: if the lhth partial has a strength coefficient of A the (lh + n)th partial will have a coefficient of A * rn, i.e. strength values trace an exponential curve. r may be positive, zero or negative, and is not restricted to integers. The default value is 1. Note This subroutine is a non-time-varying version of the CSound buzzand gbuzz generators, and is similarly useful as a complex sound source in subtractive synthesis. With lh and r present it parallels gbuzz; with both absent or equal to 1 it reduces to the simpler buzz (i.e. nh equal-strength harmonic partials beginning with the fundamental). Sampling the stored waveform with an oscillator is more efficient than using the dynamic buzz units. However, the spectral content is invariant and care is necessary, lest the higher partials exceed the Nyquist during sampling to produce fold-over. Examples Here is an example of the GEN11 routine. It uses the files gen11.csd. An example of the GEN11 routine. These are the diagrams of the waveforms of the GEN11 routines, as used in the example: f 1 0 16384 11 1 1 f 1 0 16384 11 1 1 f 2 0 16384 11 10 1 .7 f 2 0 16384 11 10 1 .7 f 3 0 16384 11 10 5 2 f 3 0 16384 11 10 5 2 See Also GEN10 manual_src~dfsg/scoregens/gen17.xml0000644000000000000000000000666012262561504016410 0ustar rootroot GEN17 GEN17 GEN17 Creates a step function from given x-y pairs. Description This subroutine creates a step function from given x-y pairs. Syntax f # time size 17 x1 a x2 b x3 c ... Initialization size -- number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement). The normal value is power-of-2 plus 1. x1, x2, x3, etc. -- x-ordinate values, in ascending order, 0 first. a, b, c, etc. -- y-values at those x-ordinates, held until the next x-ordinate. Note This subroutine creates a step function of x-y pairs whose y-values are held to the right. The right-most y-value is then held to the end of the table. The function is useful for mapping one set of data values onto another, such as MIDI note numbers onto sampled sound ftable numbers ( see loscil). Examples f 1 0 128 -17 0 1 12 2 24 3 36 4 48 5 60 6 72 7 84 8 This describes a step function with 8 successively increasing levels, each 12 locations wide except the last which extends its value to the end of the table. Rescaling is inhibited. Indexing into this table with a MIDI note-number would retrieve a different value every octave up to the eighth, above which the value returned would remain the same. Here is a complete example of the GEN17 routine. It uses the files gen17.csd. An example of the GEN17 routine. This is the diagram of the waveform of the GEN17 routine, as used in the example: f 2 0 128 -17 0 10 32 20 64 30 96 40 - a step function with 4 equal levels, each 32 locations wide except the last which extends its value to the end of the table f 2 0 128 -17 0 10 32 20 64 30 96 40 - a step function with 4 equal levels, each 32 locations wide except the last which extends its value to the end of the table See Also GEN02 manual_src~dfsg/scoregens/gen34.xml0000644000000000000000000001210012262561504016371 0ustar rootroot GEN34 GEN34 GEN34 Generate composite waveforms by mixing simple sinusoids. Description These routines generate composite waveforms by mixing simple sinusoids, similarly to GEN09, but the parameters of the partials are specified in an already existing table, which makes it possible to calculate any number of partials in the orchestra. The difference between GEN33 and GEN34 is that GEN33 uses inverse FFT to generate output, while GEN34 is based on the algorithm used in oscils opcode. GEN33 allows integer partials only, and does not support power of two plus 1 table size, but may be significantly faster with a large number of partials. On the other hand, with GEN34, it is possible to use non-integer partial numbers and extended guard point, and this routine may be faster if there is only a small number of partials (note that GEN34 is also several times faster than GEN09, although the latter may be more accurate). Syntax f # time size 34 src nh scl [fmode] Initialization size -- number of points in the table. Must be power of two or a power of two plus 1. src -- source table number. This table contains the parameters of each partial in the following format: stra, pna, phsa, strb, pnb, phsb, ... the parameters are: stra, strb, etc.: relative strength of partials. The actual amplitude depends on the value of scl, or normalization (if enabled). pna, pnb, etc.: partial number, or frequency, depending on fmode (see below); zero and negative values are allowed, however, if the absolute value of the partial number exceeds (size / 2), the partial will not be rendered. phsa, phsb, etc.: initial phase, in the range 0 to 1. Table length (not including the guard point) should be at least 3 * nh. If the table is too short, the number of partials (nh) is reduced to (table length) / 3, rounded towards zero. nh -- number of partials. Zero or negative values are allowed, and result in an empty table (silence). The actual number may be reduced if the source table (src) is too short, or some partials have too high frequency. scl -- amplitude scale. fmode (optional, default = 0) -- a non-zero value can be used to set frequency in Hz instead of partial numbers in the source table. The sample rate is assumed to be fmode if it is positive, or -(sr * fmode) if any negative value is specified. Examples Here is an example of the GEN34 routine. It uses the file gen34.csd. Example of the gen34 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This is the diagram of the waveform generated by GEN34, as used in the example: f 2 0 262144 -34 1 inumh 1 -1 f 2 0 262144 -34 1 inumh 1 -1 See Also GEN09, GEN33 Credits Programmer: &nameistvan; March 2002 New in version 4.19 manual_src~dfsg/scoregens/gen23.xml0000644000000000000000000000607612262561504016406 0ustar rootroot GEN23 GEN23 GEN23 Reads numeric values from a text file. Description This subroutine reads numeric values from an external ASCII file. Syntax f # time size -23 "filename.txt" Initialization "filename.txt" -- numeric values contained in "filename.txt" (which indicates the complete pathname of the character file to be read), can be separated by spaces, tabs, newline characters or commas. Also, words that contains non-numeric characters can be used as comments since they are ignored. size -- number of points in the table. Must be a power of 2 , power of 2 + 1, or zero. If size = 0, table size is determined by the number of numeric values in filename.txt. (New in Csound version 3.57) Note All characters following ';' or '#' (comment) are ignored until next line (numbers too). Examples Here is an example of the GEN23 opcode. It uses the file gen23.csd and spectrum.txt. Example of the GEN23 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This is the diagram of the waveform of the GEN23 routine, as used in the example: f 2 0 128 -23 "spectrum.txt" - not normalized f 2 0 128 -23 "spectrum.txt" - not normalized Credits Author: &namegabriel; Italy February, 1998 New in Csound version 3.47. Comments starting with '#' are ignored from Csound version 5.12. manual_src~dfsg/scoregens/genwave.xml0000644000000000000000000000667312262561504017127 0ustar rootroot GENwave GENwave "wave" Generates a compactly supported wavelet function. Description Creates a compactly supported wavelet, scaling function or wavelet packet. The output function is obtained by deconvolution of corresponding mirror filter impulse response. This procedure is applied in an iterative fashion. The filter bank used in classic discrete wavelet transform grows only towards low frequencies. Meanwhile wavelet packet transform introduces any possible direction of tree's growth. The sequence of mirror filters used in deconvolution is determined by binary form of seq value. "0" corresponds to low-pass filter and "1" to high-pass. The number of iteration steps is determined by filter's length and size of function table. I.e. for filter length 8 and table size 256 there are log2(256/8) = 5 iterations available. Syntax f # time size "wave" fnsf seq rescale Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). fnsf -- pre-existing table with scaling function coefficients. seq -- non-negative integer number which corresponds to sequence of low-pass and high-pass mirror filters during deconvolution procedure. rescale -- if not zero the table is not rescaled Examples Here is an example of the GENwave routine. It uses the file genwave.csd. Example of the GENwave routine. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also Ingrid Daubechies. Ten Lectures on Wavelets, SIAM 1992. Pedro A.G. Batista. An Introduction to Sound Synthesis with Wavelet Packets, Csound Book CD Chapters, MIT 2000. Victor M. Wickerhauser. Acoustic Signal Compression with Wavelet Packets, Yale 1992. Credits Written by Gleb Rogozinsky Saint-Petersburg University of Film and Television Saint-Petersburg, Russia April 2012 manual_src~dfsg/scoregens/gen32.xml0000644000000000000000000000446312262561504016404 0ustar rootroot GEN32 GEN32 GEN32 Mixes any waveform, resampled with either FFT or linear interpolation. Description This routine is similar to GEN31, but allows specifying source ftable for each partial. Tables can be resampled either with FFT, or linear interpolation. Syntax f # time size 32 srca pna stra phsa srcb pnb strb phsb ... Performance srca, srcb -- source table number. A negative value can be used to read the table with linear interpolation (by default, the source waveform is transposed and phase shifted using FFT); this is less accurate, but faster, and allows non-integer and negative partial numbers. pna, pnb, ... -- partial number, must be a positive integer if source table number is positive (i.e. resample with FFT). stra, strb, ... -- amplitude scale phsa, phsb, ... -- start phase (0 to 1) Examples Here is an example of the GEN32 routine. It uses the file gen32.csd. Example of the gen32 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namerasmus; Programmer: &nameistvan; New in version 4.17 manual_src~dfsg/scoregens/gen14.xml0000644000000000000000000001563212262561504016404 0ustar rootroot GEN14 GEN14 GEN14 Stores a polynomial whose coefficients derive from Chebyshevs of the second kind. Description Uses Chebyshev coefficients to generate stored polynomial functions which, under waveshaping, can be used to split a sinusoid into harmonic partials having a pre-definable spectrum. Syntax f # time size 14 xint xamp h0 h1 h2 ... Initialization size -- number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement). The normal value is power-of-2 plus 1. xint -- provides the left and right values [-xint, +xint] of the x interval over which the polynomial is to be drawn. These subroutines both call GEN03 to draw their functions; the p5 value here is therefore expanded to a negative-positive p5, p6 pair before GEN03 is actually called. The normal value is 1. xamp -- amplitude scaling factor of the sinusoid input that is expected to produce the following spectrum. h0, h1, h2, etc. -- relative strength of partials 0 (DC), 1 (fundamental), 2 ... that will result when a sinusoid of amplitude xamp * int(size/2)/xint is waveshaped using this function table. These values thus describe a frequency spectrum associated with a particular factor xamp of the input signal. Note GEN13 is the function generator normally employed in standard waveshaping. It stores a polynomial whose coefficients derive from the Chebyshev polynomials of the first kind, so that a driving sinusoid of strength xamp will exhibit the specified spectrum at output. Note that the evolution of this spectrum is generally not linear with varying xamp. However, it is bandlimited (the only partials to appear will be those specified at generation time); and the partials will tend to occur and to develop in ascending order (the lower partials dominating at low xamp, and the spectral richness increasing for higher values of xamp). A negative hn value implies a 180 degree phase shift of that partial; the requested full-amplitude spectrum will not be affected by this shift, although the evolution of several of its component partials may be. The pattern +,+,-,-,+,+,... for h0,h1,h2... will minimize the normalization problem for low xamp values (see above), but does not necessarily provide the smoothest pattern of evolution. GEN14 stores a polynomial whose coefficients derive from Chebyshevs of the second kind. Examples Here is an example of the GEN14 opcode. It uses the file gen14.csd. Example of the GEN14 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. These are the diagrams of the waveforms of the GEN14 routines, as used in the example: f28 0 4097 13 1 1 1 0 .8 0 .5 0 .2 - waveshaping function: GEN13, odd harmonics f28 0 4097 13 1 1 1 0 .8 0 .5 0 .2 - waveshaping function: GEN13, odd harmonics f29 0 4097 14 1 1 1 0 .8 0 .5 0 .2 - waveshaping function: GEN14, the same odd harmonics f29 0 4097 14 1 1 1 0 .8 0 .5 0 .2 - waveshaping function: GEN14, the same odd harmonics f30 0 4097 13 1 1 0 1 0 .6 0 .4 0 .1 - waveshaping function: GEN13, even harmonics f30 0 4097 13 1 1 0 1 0 .6 0 .4 0 .1 - waveshaping function: GEN13, even harmonics f31 0 4097 14 1 1 0 1 0 .6 0 .4 0 .1 - waveshaping function: GEN14, the same even harmonics f31 0 4097 14 1 1 0 1 0 .6 0 .4 0 .1 - waveshaping function: GEN14, the same even harmonics See Also GEN03, GEN13, and GEN15. Information about the Chebyshev polynomials on Wikipedia: http://en.wikipedia.org/wiki/Chebyshev_polynomials manual_src~dfsg/scoregens/gen10.xml0000644000000000000000000001127412262561504016376 0ustar rootroot GEN10 GEN10 GEN10 Generate composite waveforms made up of weighted sums of simple sinusoids. Description These subroutines generate composite waveforms made up of weighted sums of simple sinusoids. The specification of each contributing partial requires 1 pfield using GEN10. Syntax f # time size 10 str1 str2 str3 str4 ... Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). str1, str2, str3, etc. -- relative strengths of the fixed harmonic partial numbers 1,2,3, etc., beginning in p5. Partials not required should be given a strength of zero. Note These subroutines generate stored functions as sums of sinusoids of different frequencies. The two major restrictions on GEN10 that the partials be harmonic and in phase do not apply to GEN09 or GEN19. In each case the composite wave, once drawn, is then rescaled to unity if p4 was positive. A negative p4 will cause rescaling to be skipped. Examples Here is an example of the GEN10 routine. It uses the files gen10.csd. An example of the GEN10 routine. These are the diagrams of the waveforms of the GEN10 routines, as used in the example: f 1 0 16384 10 1 - sine wave with only the fundamental frequency f 1 0 16384 10 1 - sine wave with only the fundamental frequency f 2 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 - sawtooth, with a fundamental and 8 harmonics f 2 0 16384 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 - sawtooth, with a fundamental and 8 harmonics f 3 0 16384 10 1 0 0.3 0 0.2 0 0.14 0 .111 - square wave, with a fundamental and 8 harmonics but 4 have 0 strength f 3 0 16384 10 1 0 0.3 0 0.2 0 0.14 0 .111 - square wave, with a fundamental and 8 harmonics but 4 have 0 strength f 4 0 16384 10 1 1 1 1 0.7 0.5 0.3 0.1 - pulse wave, with a fundamental and 8 harmonics f 4 0 16384 10 1 1 1 1 0.7 0.5 0.3 0.1 - pulse wave, with a fundamental and 8 harmonics See Also GEN09, GEN11, and GEN19. manual_src~dfsg/scoregens/genexp.xml0000644000000000000000000000633512262561504016754 0ustar rootroot GENexp GENexp "exp" Generate a table with values on the exp function. Description Creates an ftable with values of the exp function. Syntax f # time size "exp" start end rescale Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). start, end -- first and last value to be stored; the GEN draws a curve that goes from start to end: exp(start) .... exp(end). The points stored are uniformly spaced between these to the table size rescale -- if not zero the table is not rescaled Examples Here is a simple example of the GENexp routine. It uses the file genexp.csd. Example of the GENexp routine. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. These are the diagrams of the waveforms of the GENexp routines, as used in the example: f 2 0 8192 "exp" 0 15 0 f 2 0 8192 "exp" 0 15 0 f 3 0 8192 "exp" 0 3 0 f 3 0 8192 "exp" 0 3 0 See Also GENexp and GENsone. More information on this routine: http://www.csounds.com/journal/issue11/distortionSynthesis, written by Victor Lazzarini Credits Written by &namevictor; manual_src~dfsg/scoregens/gen49.xml0000644000000000000000000001026312262561504016407 0ustar rootroot GEN49 GEN49 GEN49 Transfers data from an MP3 soundfile into a function table. Description This subroutine transfers data from an MP3 soundfile into a function table. Syntax f# time size 49 filcod skiptime format Performance size -- number of points in the table. Ordinarily a power of 2 or a power-of-2 plus 1 (see f statement); the maximum tablesize is 16777216 (224) points. The allocation of table memory can be deferred by setting this parameter to 0; the size allocated is then the number of points in the file (probably not a power-of-2), and the table is not usable by normal oscillators, but it is usable by a loscil unit. The soundfile can also be mono or stereo. filcod -- integer or character-string denoting the source soundfile name. An integer denotes the file soundin.filcod ; a character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the file is sought first in the current directory, then in that given by the environment variable SSDIR (if defined) then by SFDIR. See also soundin. skiptime -- begin reading at skiptime seconds into the file. format -- specifies the audio data-file format required: 1 - Mono file 3 - First channel (left) 2 - Stereo file 4 - Second channel (right) If format = 0 the sample format is taken from the soundfile header. Note Reading stops at end-of-file or when the table is full. Table locations not filled will contain zeros. If p4 is positive, the table will be post-normalized (rescaled to a maximum absolute value of 1 after generation). A negative p4 will cause rescaling to be skipped. Examples Here is an example of the GEN49 routine. It uses the files gen49.csd. An example of the GEN49 routine. This is the diagram of the waveforms of the GEN49 routine, as used in the example: f 1 0 131072 49 "beats.mp3" 0 1 f 1 0 131072 49 "beats.mp3" 0 1 Credits Written by &namejohn; February 2009. manual_src~dfsg/scoregens/gen04.xml0000644000000000000000000001153112262561504016375 0ustar rootroot GEN04 GEN04 GEN04 Generates a normalizing function. Description This subroutine generates a normalizing function by examining the contents of an existing table. Syntax f # time size 4 source# sourcemode Initialization size -- number of points in the table. Should be power-of-2 plus 1. Must not exceed (except by 1) the size of the source table being examined; limited to just half that size if the sourcemode is of type offset (see below). source # -- table number of stored function to be examined. sourcemode -- a coded value, specifying how the source table is to be scanned to obtain the normalizing function. Zero indicates that the source is to be scanned from left to right. Non-zero indicates that the source has a bipolar structure; scanning will begin at the mid-point and progress outwards, looking at pairs of points equidistant from the center. Note The normalizing function derives from the progressive absolute maxima of the source table being scanned. The new table is created left-to-right, with stored values equal to 1/(absolute maximum so far scanned). Stored values will thus begin with 1/(first value scanned), then get progressively smaller as new maxima are encountered. For a source table which is normalized (values <= 1), the derived values will range from 1/(first value scanned) down to 1. If the first value scanned is zero, that inverse will be set to 1. The normalizing function from GEN04 is not itself normalized. GEN04 is useful for scaling a table-derived signal so that it has a consistent peak amplitude. A particular application occurs in waveshaping when the carrier (or indexing) signal is less than full amplitude. Examples Here is a simple example of the GEN04 routine. It uses the files gen04.csd. An example of the GEN04 routine. These are the diagrams of the waveforms of the GEN04 routines, as used in the example: f1 0 64 21 6 - Gaussian (random) distribution f1 0 64 21 6 - Gaussian (random) distribution f2 0 33 4 1 1 - AND its normalizing function with midpoint bipolar offset f2 0 33 4 1 1 - AND its normalizing function with midpoint bipolar offset f3 0 1025 13 1 1 0 5 0 5 0 10 - Chebyshev algorithm f3 0 1025 13 1 1 0 5 0 5 0 10 - Chebyshev algorithm f4 0 513 4 3 1 - AND its normalizing function with midpoint bipolar offset f4 0 513 4 3 1 - AND its normalizing function with midpoint bipolar offset manual_src~dfsg/scoregens/gen01.xml0000644000000000000000000001324712262561504016400 0ustar rootroot GEN01 GEN01 GEN01 Transfers data from a soundfile into a function table. Description This subroutine transfers data from a soundfile into a function table. Syntax f# time size 1 filcod skiptime format channel Performance size -- number of points in the table. Ordinarily a power of 2 or a power-of-2 plus 1 (see f statement); the maximum tablesize is 16777216 (224) points. The allocation of table memory can be deferred by setting this parameter to 0; the size allocated is then the number of points in the file (probably not a power-of-2), and the table is not usable by normal oscillators, but it is usable by a loscil unit. The soundfile can also be mono or stereo. filcod -- integer or character-string denoting the source soundfile name. An integer denotes the file soundin.filcod ; a character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the file is sought first in the current directory, then in that given by the environment variable SSDIR (if defined) then by SFDIR. See also soundin. skiptime -- begin reading at skiptime seconds into the file. channel -- channel number to read in. 0 denotes read all channels. format -- specifies the audio data-file format: 1 - 8-bit signed character 4 - 16-bit short integers 2 - 8-bit A-law bytes 5 - 32-bit long integers 3 - 8-bit U-law bytes 6 - 32-bit floats If format = 0 the sample format is taken from the soundfile header, or by default from the CSound -o command-line flag. Note Reading stops at end-of-file or when the table is full. Table locations not filled will contain zeros. If p4 is positive, the table will be post-normalized (rescaled to a maximum absolute value of 1 after generation). A negative p4 will cause rescaling to be skipped. GEN01 also works with WAV and OGG and a dozen and more other sound formats; these file formats depend on libsndfile, see http://www.mega-nerd.com/libsndfile/ Examples Here is an example of the GEN01 routine. It uses the files gen01.csd and several sound files. An example of the GEN01 routine. These are the diagrams of the waveforms of the GEN01 routines, as used in the example: f 1 0 131072 1 "beats.wav" 0 0 0 - non-deferred sound f 1 0 131072 1 "beats.wav" 0 0 0 - non-deferred sound f 2 0 0 1 "flute.aiff" 0 0 0 - deferred sound f 2 0 0 1 "flute.aiff" 0 0 0 - deferred sound f 3 0 0 1 "beats.ogg" 0 0 0 - deferred sound f 3 0 0 1 "beats.ogg" 0 0 0 - deferred sound Credits September 2003. Thanks goes to Dr. Richard Boulanger for pointing out the references to the AIFF file format. manual_src~dfsg/scoregens/gen13.xml0000644000000000000000000002026312262561504016377 0ustar rootroot GEN13 GEN13 GEN13 Stores a polynomial whose coefficients derive from the Chebyshev polynomials of the first kind. Description Uses Chebyshev coefficients to generate stored polynomial functions which, under waveshaping, can be used to split a sinusoid into harmonic partials having a pre-definable spectrum. Syntax f # time size 13 xint xamp h0 h1 h2 ... Initialization size -- number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement). The normal value is power-of-2 plus 1. xint -- provides the left and right values [-xint, +xint] of the x interval over which the polynomial is to be drawn. These subroutines both call GEN03 to draw their functions; the p5 value here is therefor expanded to a negative-positive p5, p6 pair before GEN03 is actually called. The normal value is 1. xamp -- amplitude scaling factor of the sinusoid input that is expected to produce the following spectrum. h0, h1, h2, etc. -- relative strength of partials 0 (DC), 1 (fundamental), 2 ... that will result when a sinusoid of amplitude xamp * int(size/2)/xint is waveshaped using this function table. These values thus describe a frequency spectrum associated with a particular factor xamp of the input signal. GEN13 is the function generator normally employed in standard waveshaping. It stores a polynomial whose coefficients derive from the Chebyshev polynomials of the first kind, so that a driving sinusoid of strength xamp will exhibit the specified spectrum at output. Note that the evolution of this spectrum is generally not linear with varying xamp. However, it is bandlimited (the only partials to appear will be those specified at generation time); and the partials will tend to occur and to develop in ascending order (the lower partials dominating at low xamp, and the spectral richness increasing for higher values of xamp). A negative hn value implies a 180 degree phase shift of that partial; the requested full-amplitude spectrum will not be affected by this shift, although the evolution of several of its component partials may be. The pattern +,+,-,-,+,+,... for h0,h1,h2... will minimize the normalization problem for low xamp values (see above), but does not necessarily provide the smoothest pattern of evolution. Examples Here is an example of the GEN13 opcode. It uses the file gen13.csd. Example of the GEN13 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. These are the diagrams of the waveforms of the GEN13 routines, as used in the example: f1 0 513 13 1 1 0 100 -50 -33 25 20 -16.7 -14.2 12.5 11.1 -10 -9.09 8.333 7.69 -7.14 -6.67 6.25 5.88 -5.55 -5.26 5 - quasi sawtooth transfer function f1 0 513 13 1 1 0 100 -50 -33 25 20 -16.7 -14.2 12.5 11.1 -10 -9.09 8.333 7.69 -7.14 -6.67 6.25 5.88 -5.55 -5.26 5 - quasi sawtooth transfer function f3 0 513 13 1 1 0 100 0 -33 0 20 0 -14.2 0 11.1 0 -9.09 0 7.69 0 -6.67 0 5.88 0 -5.26 - quasi square wave transfer function f3 0 513 13 1 1 0 100 0 -33 0 20 0 -14.2 0 11.1 0 -9.09 0 7.69 0 -6.67 0 5.88 0 -5.26 - quasi square wave transfer function f5 0 513 13 1 1 0 100 0 -11.11 0 4 0 -2.04 0 1.23 0 -.826 0 .59 0 -.444 0 .346 0 -.277 - quasi triangle wave transfer function f5 0 513 13 1 1 0 100 0 -11.11 0 4 0 -2.04 0 1.23 0 -.826 0 .59 0 -.444 0 .346 0 -.277 - quasi triangle wave transfer function f7 0 513 13 1 1 0 1 -.8 0 .6 0 0 0 .4 0 0 0 0 .1 -.2 -.3 .5 - transfer function 1 f7 0 513 13 1 1 0 1 -.8 0 .6 0 0 0 .4 0 0 0 0 .1 -.2 -.3 .5 - transfer function 1 f9 0 513 13 1 1 0 0 0 -.1 0 .3 0 -.5 0 .7 0 -.9 0 1 0 -1 0 - transfer function 2 f9 0 513 13 1 1 0 0 0 -.1 0 .3 0 -.5 0 .7 0 -.9 0 1 0 -1 0 - transfer function 2 f11 0 513 13 1 1 0 0 0 0 0 0 0 -1 0 1 0 0 -.1 0 .1 0 -.2 .3 0 -.7 0 .2 0 -.1 - transfer function 3 f11 0 513 13 1 1 0 0 0 0 0 0 0 -1 0 1 0 0 -.1 0 .1 0 -.2 .3 0 -.7 0 .2 0 -.1 - transfer function 3 f13 0 513 13 1 1 0 5 0 3 0 1 - split a sinusoid into 3 odd-harmonic partials of relative strength 5:3:1 f13 0 513 13 1 1 0 5 0 3 0 1 - split a sinusoid into 3 odd-harmonic partials of relative strength 5:3:1 See Also GEN03, GEN14, and GEN15. Information about the Chebyshev polynomials on Wikipedia: http://en.wikipedia.org/wiki/Chebyshev_polynomials manual_src~dfsg/scoregens/gen22.xml0000644000000000000000000000075112262561504016377 0ustar rootroot GEN22 GEN22 GEN22 Deprecated. Description Deprecated as of version 4.19. Use the GEN18 routine instead. manual_src~dfsg/scoregens/gen28.xml0000644000000000000000000001017312262561504016404 0ustar rootroot GEN28 GEN28 GEN28 Reads a text file which contains a time-tagged trajectory. Description This function generator reads a text file which contains sets of three values representing the xy coordinates and a time-tag for when the signal should be placed at that location, allowing the user to define a time-tagged trajectory. The file format is in the form: time1 X1 Y1 time2 X2 Y2 time3 X3 Y3 The configuration of the xy coordinates in space places the signal in the following way: a1 is -1, 1 a2 is 1, 1 a3 is -1, -1 a4 is 1, -1 This assumes a loudspeaker set up as a1 is left front, a2 is right front, a3 is left back, a4 is right back. Values greater than 1 will result in sounds being attenuated as if in the distance. GEN28 creates values to 10 milliseconds of resolution. Syntax f # time size 28 ifilcod Initialization size -- number of points in the table. Must be 0. GEN28 takes 0 as the size and automatically allocates memory. ifilcod -- character-string denoting the source file name. A character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the named file is sought in the current directory. Examples f1 0 0 28 "move" The file "move" should look like: 0 -1 1 1 1 1 2 4 4 2.1 -4 -4 3 10 -10 5 -40 0 Since GEN28 creates values to 10 milliseconds of resolution, there will be 500 values created by interpolating X1 to X2 to X3 and so on, and Y1 to Y2 to Y3 and so on, over the appropriate number of values that are stored in the function table. The sound will begin in the left front, over 1 second it will move to the right front, over another second it move further into the distance but still in the right front, then in just 1/10th of a second it moves to the left rear, a bit distant. Finally over the last .9 seconds the sound will move to the right rear, moderately distant, and it comes to rest between the two left channels (due west!), quite distant. Here is an example of the GEN28 routine. It uses the file gen28.csd. Example of the gen28 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Richard Karpen Seattle, Wash 1998 New in Csound version 3.48 manual_src~dfsg/scoregens/top.xml0000644000000000000000000003241112262561504016262 0ustar rootroot Score Statements and GEN Routines
Score Statements The statements used in scores are: a - Advance score time by a specified amount b - Resets the clock e - Marks the end of the last section of the score f - Causes a GEN subroutine to place values in a stored function table i - Makes an instrument active at a specific time and for a certain duration m - Sets a named mark in the score n - Repeats a marked section q - Used to quiet an instrument r - Starts a repeated section s - Marks the end of a section t - Sets the tempo v - Provides for locally variable time warping of score events x - Skip the rest of the current section { - Begins a non-sectional, nestable loop } - Ends a non-sectional, nestable loop
&scoregensa; &scoregensb; &scoregense; &scoregensf; &scoregensi; &scoregensm; &scoregensn; &scoregensq; &scoregensr; &scoregenss; &scoregenst; &scoregensv; &scoregensx; &scoregensleftbrace; &scoregensrightbrace;
GEN Routines GEN routines are used as data generators for function tables. When a function table is created using the f score statement the GEN function is given as its fourth argument. A negative GEN number implies that the function is not rescaled, and maintains its original values. Sine/Cosine Generators: GEN09 - Composite waveforms made up of weighted sums of simple sinusoids. GEN10 - Composite waveforms made up of weighted sums of simple sinusoids. GEN11 - Additive set of cosine partials. GEN19 - Composite waveforms made up of weighted sums of simple sinusoids. GEN30 - Generates harmonic partials by analyzing an existing table. GEN33 - Generate composite waveforms by mixing simple sinusoids. GEN34 - Generate composite waveforms by mixing simple sinusoids. Line/Exponential Segment Generators: GEN05 - Constructs functions from segments of exponential curves. GEN06 - Generates a function comprised of segments of cubic polynomials. GEN07 - Constructs functions from segments of straight lines. GEN08 - Generate a piecewise cubic spline curve. GEN16 - Creates a table from a starting value to an ending value. GEN25 - Construct functions from segments of exponential curves in breakpoint fashion. GEN27 - Construct functions from segments of straight lines in breakpoint fashion. File Access GEN Routines: GEN01 - Transfers data from a soundfile into a function table. GEN23 - Reads numeric values from a text file. GEN28 - Reads a text file which contains a time-tagged trajectory. GEN49 - Transfers data from an MP3 soundfile into a function table. Numeric Value Access GEN Routines GEN02 - Transfers data from immediate pfields into a function table. GEN17 - Creates a step function from given x-y pairs. GEN52 - Creates an interleaved multichannel table from the specified source tables, in the format expected by the ftconv opcode. Window Function GEN Routines GEN20 - Generates functions of different windows. Random Function GEN Routines GEN21 - Generates tables of different random distributions. GEN40 - Generates a random distribution using a distribution histogram. GEN41 - Generates a random list of numerical pairs. GEN42 - Generates a random distribution of discrete ranges of values. GEN43 - Loads a PVOCEX file containing a PV analysis. Waveshaping GEN Routines GEN03 - Generates a stored function table by evaluating a polynomial. GEN13 - Stores a polynomial whose coefficients derive from the Chebyshev polynomials of the first kind. GEN14 - Stores a polynomial whose coefficients derive from Chebyshevs of the second kind. GEN15 - Creates two tables of stored polynomial functions. Amplitude Scaling GEN Routines GEN04 - Generates a normalizing function. GEN12 - Generates the log of a modified Bessel function of the second kind. GEN24 - Reads numeric values from another allocated function-table and rescales them. Mixing GEN Routines GEN18 - Writes composite waveforms made up of pre-existing waveforms. GEN31 - Mixes any waveform specified in an existing table. GEN32 - Mixes any waveform, resampled with either FFT or linear interpolation. Pitch and Tuning GEN Routines GEN51 - fills a table with a fully customized micro-tuning scale, in the manner of Csound opcodes cpstun, cpstuni and cpstmid. Named GEN Routines Csound's GEN routines can be extended with GEN function plugins. There is currently a simple GEN plugin that provides exponential and hyperbolic tangent functions, and the sone function. There is also a generator called farey for the Farey sequence operations. These GEN functions are not called by number, but by name. "tanh" - fills a table from a hyperbolic tangent formula. "exp" - fills a table from an exponential formula. "sone" - fills a table from a sone function formula. "farey" - fills a table from a Farey sequence. "wave" - fills a table from a wavelet transform.
&scoregensgen01; &scoregensgen02; &scoregensgen03; &scoregensgen04; &scoregensgen05; &scoregensgen06; &scoregensgen07; &scoregensgen08; &scoregensgen09; &scoregensgen10; &scoregensgen11; &scoregensgen12; &scoregensgen13; &scoregensgen14; &scoregensgen15; &scoregensgen16; &scoregensgen17; &scoregensgen18; &scoregensgen19; &scoregensgen20; &scoregensgen21; &scoregensgen23; &scoregensgen24; &scoregensgen25; &scoregensgen27; &scoregensgen28; &scoregensgen30; &scoregensgen31; &scoregensgen32; &scoregensgen33; &scoregensgen34; &scoregensgen40; &scoregensgen41; &scoregensgen42; &scoregensgen43; &scoregensgen49; &scoregensgen51; &scoregensgen52; &scoregenstanh; &scoregensexp; &scoregenssone; &scoregensfarey; &scoregenswave; manual_src~dfsg/scoregens/x.xml0000644000000000000000000000210512262561504015724 0ustar rootroot x statement x Statement x Skip the rest of the current section. Description This statement may be used to skip the rest of the current section. Syntax x anything Initialization All pfields are ignored. Examples Here is an example of the x statement. It uses the file x.csd. Example of the x statement. manual_src~dfsg/scoregens/r.xml0000644000000000000000000000461212262561504015723 0ustar rootroot r statement repeat statement r Statement (Repeat Statement) r Starts a repeated section. Description Starts a repeated section, which lasts until the next s, r or e statement. Syntax r p1 p2 Initialization p1 -- Number of times to repeat the section. p2 -- Macro(name) to advance with each repetition (optional). Performance In order that the sections may be more flexible than simple editing, the macro named p2 is given the value of 1 for the first time through the section, 2 for the second, and 3 for the third. This can be used to change p-field parameters, or ignored. Warning Because of serious problems of interaction with macro expansion, sections must start and end in the same file, and not in a macro. Examples Here is an example of the r statement. It uses the file r.csd. Example of the r statement. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April, 1998 New in Csound version 3.48 Example written by &namekevin; manual_src~dfsg/scoregens/gen16.xml0000644000000000000000000000622512262561504016404 0ustar rootroot GEN16 GEN16 GEN16 Creates a table from a starting value to an ending value. Description Creates a table from beg value to end value of dur steps. Syntax f # time size 16 val1 dur1 type1 val2 [dur2 type2 val3 ... typeX valN] Initialization size -- number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement). The normal value is power-of-2 plus 1. The end value is only reached when the table length is power-of-2 plus 1. (This length is crucial for use with the tab opcode.) beg -- starting value dur -- number of segments type -- if 0, a straight line is produced. If non-zero, then GEN16 creates the following curve, for dur steps: beg + (end - beg) * (1 - exp( i*type/(dur-1) )) / (1 - exp(type)) end -- value after dur segments Here are some examples of the curves generated for different values of type: Tables generated by GEN16 for different values of type. Tables generated by GEN16 for different values of type. Note If type > 0, there is a slowly rising (concave) or slowly decaying (convex) curve, while if itype < 0, the curve is fast rising (convex) or fast decaying (concave). See also transeg. A simple example of the GEN16 routine. Credits Author: &namejohn; University of Bath, Codemist. Ltd. Bath, UK October, 2000 New in Csound version 4.09 manual_src~dfsg/scoregens/i.xml0000644000000000000000000002425612262561504015720 0ustar rootroot i statement instrument statement note statement i Statement (Instrument or Note Statement) i Makes an instrument active at a specific time and for a certain duration. Description This statement calls for an instrument to be made active at a specific time and for a certain duration. The parameter field values are passed to that instrument prior to its initialization, and remain valid throughout its Performance. Syntax i p1 p2 p3 p4 ... Initialization p1 -- Instrument number, usually a non-negative integer. An optional fractional part can provide an additional tag for specifying ties between particular notes of consecutive clusters. A negative p1 (including tag) can be used to turn off a particular held note. p2 -- Starting time in arbitrary units called beats. p3 -- Duration time in beats (usually positive). A negative value will initiate a held note (see also ihold). A negative value can also be used for 'always on' instruments like reverberation. These notes are not terminated by s statements A zero value will invoke an initialization pass without performance (see also instr). p4 ... -- Parameters whose significance is determined by the instrument. Performance Beats are evaluated as seconds, unless there is a t statement in this score section or a -t flag in the command-line. Starting or action times are relative to the beginning of a section ( see s statement), which is assigned time 0. Note statements within a section may be placed in any order. Before being sent to an orchestra, unordered score statements must first be processed by Sorter, which will reorder them by ascending p2 value. Notes with the same p2 value will be ordered by ascending p1; if the same p1, then by ascending p3. Notes may be stacked, i.e., a single instrument can perform any number of notes simultaneously. (The necessary copies of the instrument's data space will be allocated dynamically by the orchestra loader.) Each note will normally turn off when its p3 duration has expired, or on receipt of a MIDI noteoff signal. An instrument can modify its own duration either by changing its p3 value during note initialization, or by prolonging itself through the action of a linenr or xtratim unit. An instrument may be turned on and left to perform indefinitely either by giving it a negative p3 or by including an ihold in its i-time code. If a held note is active, an i statement with matching p1 will not cause a new allocation but will take over the data space of the held note. The new pfields (including p3) will now be in effect, and an i-time pass will be executed in which the units can either be newly initialized or allowed to continue as required for a tied note (see tigoto). A held note may be succeeded either by another held note or by a note of finite duration. A held note will continue to perform across section endings (see s statement). It is halted only by turnoff or by an i statement with negative matching p1 or by an e statement. It is possible to have multiple instances (usually, but not necessarily, notes of different pitches) of the same instrument, held simultaneously, via negative p3 values. The instrument can then be fed new parameters from the score. This is useful for avoiding long hard-coded linsegs, and can be accomplished by adding a decimal part to the instrument number. For example, to hold three copies of instrument 10 in a simple chord: i10.1 0 -1 7.00 i10.2 0 -1 7.04 i10.3 0 -1 7.07 Subsequent i statements can refer to the same sounding note instances, and if the instrument definition is done properly, the new p-fields can be used to alter the character of the notes in progress. For example, to bend the previous chord up an octave and release it: i10.1 1 1 8.00 i10.2 1 1 8.04 i10.3 1 1 8.07 When turning off notes, bear in mind that i 1.1 == i 1.10 and i 1.1 != i 1.01. The maximum number of decimal places that can be used depends on the precision Csound was compiled with (See Csound Double (64-bit) vs. Float (32-bit)) The instrument definition has to take this into account, however, especially if clicks are to be avoided (see the example below). Note that the decimal instrument number notation cannot be used in conjunction with real-time MIDI. In this case, the instrument would be monophonic while a note was held. Notes being tied to previous instances of the same instrument, should skip most initialization by means of tigoto, except for the values entered in score. For example, all table reading opcodes in the instrument, should usually be skipped, as they store their phase internally. If this is suddenly changed, there will be audible clicks in the output. Note that many opcodes (such as delay and reverb) are prepared for optional initialization. To use this feature, the tival opcode is suitable. Therefore, they need not be hidden by a tigoto jump. Beginning with Csound version 3.53, strings are recognized in p-fields for opcodes that accept them (convolve, adsyn, diskin, etc.). There may be only one string per score line. You can also turnoff notes from the score by using a negative number for the instrument (p1). This is equivalent to using the turnoff2 opcode. When a note is turned off from the score, it is allowed to release (if xtratim or opcodes with release section like linenr are used) and only notes with the same fractional part are turned off. Also, only the last instance of the instrument will be turned off, so there have to be as many negative instrument numbers as positive ones for all notes to be turned off. i 1.1 1 300 8.00 i 1.2 1 300 8.04 i 1.3 1 -300 8.07 i 1.3 1 -300 8.09 ; notice that p-fields after p2 will be ignored if ; instrument number is negative i -1.1 3 1 4.00 i -1.2 4 51 4.04 i -1.3 5 1 4.07 i -1.3 6 10 4.09 Special Considerations The maximum instrument number used to be 200. This has been changed to be limited by memory only (currently there is an internal soft limit of 200; this is automatically extended as required). Examples Here is an example of the i statement. It uses the file i_statement.csd. Example of the i statement. Credits Additional text (Csound Version 4.07) explaining tied notes, edited by &namerasmus; from a note by David Kirsh, posted to the Csound mailing list. Example instrument by &namerasmus;. Updated August 2002 thanks to a note from &namerasmus;. There is no longer a hard limit of 200 instruments. manual_src~dfsg/scoregens/gen40.xml0000644000000000000000000000712612262561504016402 0ustar rootroot GEN40 GEN40 GEN40 Generates a random distribution using a distribution histogram. Description Generates a continuous random distribution function starting from the shape of a user-defined distribution histogram. Syntax f # time size 40 shapetab Performance The shape of histogram must be stored in a previously defined table, in fact shapetab argument must be filled with the number of such table. Histogram shape can be generated with any other GEN routines. Since no interpolation is used when GEN40 processes the translation, it is suggested that the size of the table containing the histogram shape to be reasonably big, in order to obtain more precision (however after the processing the shaping-table can be destroyed in order to re-gain memory). This subroutine is designed to be used together with cuserrnd opcode (see cuserrnd for more information). Examples Here is an example of the GEN40 opcode. It uses the file gen40.csd. Example of the GEN40 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i 1 time 0.00067: 53.14918 i 1 time 0.20067: 0.00000 i 1 time 0.40067: 0.00000 i 1 time 0.60067: 96.80406 i 1 time 0.80067: 94.20729 i 1 time 1.00000: 0.00000 i 1 time 1.20067: 86.13032 i 1 time 1.40067: 31.37096 i 1 time 1.60067: 70.35889 i 1 time 1.80000: 0.00000 i 1 time 2.00000: 49.18914 WARNING: Seeding from current time 2006647442 i 2 time 3.00067: 21.45002 i 2 time 3.20067: 44.32333 i 2 time 3.40067: 46.05420 i 2 time 3.60000: 0.00000 i 2 time 3.80067: 41.32175 i 2 time 4.00000: 0.00000 i 2 time 4.20000: 63.72019 i 2 time 4.40067: 0.00000 i 2 time 4.60067: 0.00000 i 2 time 4.80067: 0.00000 i 2 time 5.00000: 74.49330 This is the diagram of the waveform of the GEN40 routine, as used in the example: f 2 0 16384 40 1 f 2 0 16384 40 1 Credits Author: &namegabriel; manual_src~dfsg/scoregens/e.xml0000644000000000000000000000435012262561504015705 0ustar rootroot e statement e Statement e statement This statement may be used to mark the end of the last section of the score. Description This statement may be used to mark the end of the last section of the score. Syntax e [time] Performance The first p-field time is optional and if present determines the end time (length in beats) of the final section of the score. This time must be after the end of the last event otherwise it will have no effect. "Always on" instruments will end at the given time. Extending the section in this way is useful to avoid prematurely cutting off reverb tails or other effects. Special Considerations The e statement is contextually identical to an s statement. Additionally, the e statement terminates all signal generation (including indefinite performance) and closes all input and output files. If an e statement occurs before the end of a score, all subsequent score lines will be ignored. The e statement is optional in a score file yet to be sorted. If a score file has no e statement, then Sort processing will supply one. Examples Here is an example of the e statement. It uses the file e.csd. Example of the e statement. manual_src~dfsg/scoregens/gen27.xml0000644000000000000000000000703612262561504016407 0ustar rootroot GEN27 GEN27 GEN27 Construct functions from segments of straight lines in breakpoint fashion. Description Construct functions from segments of straight lines in breakpoint fashion. Syntax f # time size 27 x1 y1 x2 y2 x3 ... Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). x1, x2, x3, etc. -- locations in table at which to attain the following y value. Must be in increasing order. If the last value is less than size, then the rest will be set to zero. Should not be negative but can be zero. y1, y2, y3,, etc. -- Breakpoint values attained at the location specified by the preceding x value. Note If p4 is positive, functions are post-normalized (rescaled to a maximum absolute value of 1 after generation). A negative p4 will cause rescaling to be skipped. Examples Here is an example of the GEN27 opcode. It uses the file gen27.csd. Example of the GEN27 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This is the diagram of the waveform of the GEN27 routine, as used in the example: f 2 0 1025 27 0 0 200 1 400 -1 513 0 - a function which begins at 0, rises to 1 at the 200th table location, falls to -1, by the 400th location, and returns to 0 by the end of the table. The interpolation is linear f 2 0 1025 27 0 0 200 1 400 -1 513 0 - a function which begins at 0, rises to 1 at the 200th table location, falls to -1, by the 400th location, and returns to 0 by the end of the table. The interpolation is linear See Also f statement, GEN25 Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK New in Csound version 3.49 manual_src~dfsg/scoregens/gen30.xml0000644000000000000000000000602712262561504016400 0ustar rootroot GEN30 GEN30 GEN30 Generates harmonic partials by analyzing an existing table. Description Extracts a range of harmonic partials from an existing waveform. Syntax f # time size 30 src minh maxh [ref_sr] [interp] Performance src -- source ftable minh -- lowest harmonic number maxh -- highest harmonic number ref_sr (optional) -- maxh is scaled by (sr / ref_sr). The default value of ref_sr is sr. If ref_sr is zero or negative, it is now ignored. interp (optional) -- if non-zero, allows changing the amplitude of the lowest and highest harmonic partial depending on the fractional part of minh and maxh. For example, if maxh is 11.3 then the 12th harmonic partial is added with 0.3 amplitude. This parameter is zero by default. GEN30 does not support tables with an extended guard point (ie. table size = power of two + 1). Although such tables will work both for input and output, when reading source table(s), the guard point is ignored, and when writing the output table, guard point is simply copied from the first sample (table index = 0). The reason of this limitation is that GEN30 uses FFT, which requires power of two table size. GEN32 allows using linear interpolation for resampling and phase shifting, which makes it possible to use any table size (however, for partials calculated with FFT, the power of two limitation still exists). Examples Here is an example of the GEN30 routine. It uses the files gen30.csd. Example of the GEN30 routine. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &nameistvan; New in version 4.16 manual_src~dfsg/scoregens/s.xml0000644000000000000000000000666312262561504015734 0ustar rootroot s statement s Statement s Marks the end of a section. Description The s statement marks the end of a section. Syntax s [time] Initialization The first p-field time is optional and if present determines the end time (length in beats) of the section. This time must be after the end of the last event in the section otherwise it will have no effect. This can be used to create a pause before the beginning of the next section or to allow "always on" instruments such as effects to play by themselves for some length of time. Performance Sorting of the i statement, f statement and a statement by action time is done section by section. Time warping for the t statement is done section by section. All action times within a section are relative to its beginning. A section statement establishes a new relative time of 0, but has no other reinitializing effects (e.g. stored function tables are preserved across section boundaries). A section is considered complete when all action times and finite durations have been satisfied (i.e., the "length" of a section is determined by the last occurring action or turn-off). A section can be extended by the use of an f0 statement or by supplying the optional p1 value to the s statement. A section ending automatically invokes a purge of inactive instrument and data spaces. Note Since score statements are processed section by section, the amount of memory required depends on the maximum number of score statements in a section. Memory allocation is dynamic, and the user will be informed as extra memory blocks are requested during score processing. For the end of the final section of a score, the s statement is optional; the e statement may be used instead. Examples Here is an example of the s statement. It uses the file s.csd. Example of the s statement. manual_src~dfsg/scoregens/gen18.xml0000644000000000000000000001101212262561504016374 0ustar rootroot GEN18 GEN18 GEN18 Writes composite waveforms made up of pre-existing waveforms. Description Writes composite waveforms made up of pre-existing waveforms. Each contributing waveform requires 4 pfields and can overlap with other waveforms. Syntax f # time size 18 fna ampa starta finisha fnb ampb startb finishb ... Initialization size -- number of points in the table. Must be a power-of-2 (see f statement). fna, fnb, etc. -- pre-existing table number to be written into the table. ampa, ampb, etc. -- strength of wavefoms. These are relative strengths, since the composite waveform may be rescaled later. Negative values are permitted and imply a 180 degree phase shift. starta, startb, etc. -- where to start writing the fn into the table. finisha, finishb, etc. -- where to stop writing the fn into the table. The last available location is the power of two - 1. Examples f 1 0 4096 10 1 f 2 0 1024 18 1 1 0 511 1 1 512 1023 f2 consists of two copies of f1 written in to locations 0-511 and 512-1023. Here is an example of the GEN18 routine. It uses the files gen18.csd. An example of the GEN18 routine. These are the diagrams of the waveforms generated by GEN18, as used in the example: f 11 0 8193 18 1 1 0 4096 2 1 4097 8192 - composite waveform made up of a sine wave and a sawtooth f 11 0 8193 18 1 1 0 4096 2 1 4097 8192 - composite waveform made up of a sine wave and a sawtooth f 12 0 8192 18 1 1 0 4096 3 1 4097 8192 - composite waveform made up of a sine wave and a wave of cosine partials f 12 0 8192 18 1 1 0 4096 3 1 4097 8192 - composite waveform made up of a sine wave and a wave of cosine partials f 13 0 1024 18 1 0.7 0 767 3 0.7 512 1023 - sine+cosine partials overlapped, also shorter table than f12 f 13 0 1024 18 1 0.7 0 767 3 0.7 512 1023 - sine+cosine partials overlapped, also shorter table than f12 Deprecated Names GEN18 was called GEN22 in version 4.18. The name was changed due to a conflict with DirectCsound. Credits Author: William Pete Moss University of Texas at Austin Austin, Texas USA January 2002 New in version 4.18, changed in version 4.19 manual_src~dfsg/scoregens/m.xml0000644000000000000000000000561712262561504015724 0ustar rootroot m statement mark statement m Statement (Mark Statement) m Sets a named mark in the score. Description Sets a named mark in the score, which can be used by an n statement. Syntax m p1 Initialization p1 -- Name of mark. Performance This can be helpful in setting a up verse and chorus structure in the score. Names may contain letters and numerals. For example, the following score: m foo i1 0 1 i1 1 1.5 i1 2.5 2 s i1 0 10 s n foo e Will be passed from the preprocessor to Csound as: i1 0 1 i1 1 1.5 i1 2.5 2 s i1 0 10 s ;; this is named section repeated i1 0 1 i1 1 1.5 i1 2.5 2 s ;; end of named section e Examples Here is an example of the m statement. It uses the file m.csd. Example of the m statement. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April, 1998 New in Csound version 3.48 manual_src~dfsg/scoregens/gen12.xml0000644000000000000000000000747512262561504016410 0ustar rootroot GEN12 GEN12 GEN12 Generates the log of a modified Bessel function of the second kind. Description This generates the log of a modified Bessel function of the second kind, order 0, suitable for use in amplitude-modulated FM. Syntax f # time size 12 xint Initialization size -- number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement). The normal value is power-of-2 plus 1. xint -- specifies the x interval [0 to +xint] over which the function is defined. Note This subroutine draws the natural log of a modified Bessel function of the second kind, order 0 (commonly written as I subscript 0), over the x-interval requested. The call should have rescaling inhibited. The function is useful as an amplitude scaling factor in cycle-synchronous amplitude-modulated FM. (See Palamin & Palamin, J. Audio Eng. Soc., 36/9, Sept. 1988, pp.671-684.) The algorithm is interesting because it permits the normally symmetric FM spectrum to be made asymmetric around a frequency other than the carrier, and is thereby useful for formant positioning. By using a table lookup index of I(r - 1/r), where I is the FM modulation index and r is an exponential parameter affecting partial strengths, the Palamin algorithm becomes relatively efficient, requiring only oscil's, table lookups, and a single exp call. Examples Here is an example of the GEN12 opcode. It uses the file gen12.csd. Example of the GEN12 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This is the diagram of the waveform of the GEN12 routines, as used in the example: f 37 0 1024 -12 40 - Bessel function-defined from 0 to 40 f 37 0 1024 -12 40 - Bessel function-defined from 0 to 40 Credits Example is, with minor modifications, taken from The Csound Book (page 87). manual_src~dfsg/scoregens/gen19.xml0000644000000000000000000000773412262561504016415 0ustar rootroot GEN19 GEN19 GEN19 Generate composite waveforms made up of weighted sums of simple sinusoids. Description These subroutines generate composite waveforms made up of weighted sums of simple sinusoids. The specification of each contributing partial requires 4 p-fields using GEN19. Syntax f # time size 19 pna stra phsa dcoa pnb strb phsb dcob ... Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). pna, pnb, etc. -- partial no. (relative to a fundamental that would occupy size locations per cycle) of sinusoid a, sinusoid b, etc. Must be positive, but need not be a whole number, i.e., non-harmonic partials are permitted. Partials may be in any order. stra, strb, etc. -- strength of partials pna, pnb, etc. These are relative strengths, since the composite waveform may be rescaled later. Negative values are permitted and imply a 180 degree phase shift. phsa, phsb, etc. -- initial phase of partials pna, pnb, etc., expressed in degrees. dcoa, dcob, etc. -- DC offset of partials pna, pnb, etc. This is applied after strength scaling, i.e. a value of 2 will lift a 2-strength sinusoid from range [-2,2] to range [0,4] (before later rescaling). Note These subroutines generate stored functions as sums of sinusoids of different frequencies. The two major restrictions on GEN10 that the partials be harmonic and in phase do not apply to GEN09 or GEN19. In each case the composite wave, once drawn, is then rescaled to unity if p4 was positive. A negative p4 will cause rescaling to be skipped. Examples Here is an example of the GEN19 routine. It uses the files gen19.csd. An example of the GEN19 routine. This is the diagram of the waveform of the GEN19 routines, as used in the example: f 2 0 1024 19 .5 .5 270 .5 - a rising sigmoid f 2 0 1024 19 .5 .5 270 .5 - a rising sigmoid See Also GEN09 and GEN10 manual_src~dfsg/scoregens/leftbrace.xml0000644000000000000000000001442712262561504017416 0ustar rootroot { statement { Statement { Begins a non-sectional, nestable loop. Description The { and } statements can be used to repeat a group of score statements. These loops do not constitute independent score sections and thus may repeat events within the same section. Multiple loops may overlap in time or be nested within each other. Syntax { p1 p2 Initialization p1 -- Number of times to repeat the loop. p2 -- A macro name that is automatically defined at the beginning of the loop and whose value is advanced with each repetition (optional). The initial value is zero and the final value is (p1 - 1). Performance The { statement is used in conjunction with the } statement to define repeating groups of other score events. A score loop begins with the { statement which defines the number of repetitions and a unique macro name that will contain the current loop counter. The body of a loop can contain any number of other events (including sectional breaks) and is terminated by a } statement on its own line. The } statement takes no parameters. The use of the term "loop" here does not imply any sort of temporal succession to the loop iterations. In other words, the p2 values of the events inside of the loop are not automatically incremented by the length of the loop in each repetition. This is actually an advantage since it allows groups of simulataneous events to be easily defined as well. The loop macro can be used along with score expressions to increase the start times of events or to vary the events in any other way desired for each iteration. The macro is incremented by one for each repetition. Note that unlike the r statement, the value of the macro the first time through the loop is zero (0), not one (1). Therefore the final value is one less than the number of repetitions. Score loops are a very powerful tool. While similar to the section repeat facility (the r statement), their chief advantage is that the score events in successive iterations of the loop are not separated by a section termination. Thus, it is possible to create multiple loops that overlap in time. Loops also can be nested within each other to a depth of 39 levels. Warning Because of serious problems of interaction with macro expansion, loops must start and end in the same file, and not in a macro. Examples Here are some examples of the { and } statements. Sequentially repeat a three-note phrase four times. { 4 CNT i1 [0.00 + 0.75 * $CNT.] 0.2 220 i1 [0.25 + 0.75 * $CNT.] . 440 i1 [0.50 + 0.75 * $CNT.] . 880 } is interpreted as i1 0.00 0.2 220 i1 0.25 . 440 i1 0.50 . 880 i1 0.75 0.2 220 i1 1.00 . 440 i1 1.25 . 880 i1 1.50 0.2 220 i1 1.75 . 440 i1 2.00 . 880 i1 2.25 0.2 220 i1 2.50 . 440 i1 2.75 . 880 Create a group of simultaneous harmonic partials. In this example, p4 is assumed to be the frequency of the note event. { 8 PARTIAL i1 0 1 [100 * ($PARTIAL. + 1)] } is interpreted as i1 0 1 100 i1 0 1 200 i1 0 1 300 i1 0 1 400 i1 0 1 500 i1 0 1 600 i1 0 1 700 i1 0 1 800 Here is a full example of the { and } statements. It uses the file leftbrace.csd. An example of nested loops to create several inharmonic sine clusters. Credits Author: &namegabriel; New in Csound version 3.52 (?). (Fixed in version 5.08). manual_src~dfsg/scoregens/b.xml0000644000000000000000000000370312262561504015703 0ustar rootroot b statement b Statement b Statement This statement resets the clock. Description This statement resets the clock. Syntax b p1 Performance p1 -- Specifies how the clock is to be set. Special Considerations p1 is the number of beats by which p2 values of subsequent i statements are modified. If p1 is positive, the clock is reset forward, and subsequent notes appear later, the number of beats specified by p1 being added to the note's p2. If p1 is negative, the clock is reset backward, and subsequent notes appear earlier, the number of beats specified by p1 being subtracted from the note's p2. There is no cumulative affect. The clock is reset with each b statement. If p1 = 0, the clock is returned to its original position, and subsequent notes appear at their specified p2. Examples Here is an example of the b statement. It uses the file b.csd. Example of the b statement. Credits Explanation suggested and example provided by Paul Winkler. (Csound Version 4.07) manual_src~dfsg/scoregens/q.xml0000644000000000000000000000305612262561504015723 0ustar rootroot q statement q Statement q statement This statement may be used to quiet an instrument. Description This statement may be used to quiet an instrument. Syntax q p1 p2 p3 Performance p1 -- Instrument number to mute/unmute. p2 -- Action time in beats. p3 -- determines whether the instrument is muted/unmuted. The value of 0 means the instrument is muted, other values mean it is unmuted. Note that this does not affect instruments that are already running at time p2. It blocks any attempt to start one afterwards. Examples Here is an example of the q statement. It uses the file q.csd. Example of the q statement. manual_src~dfsg/scoregens/gen08.xml0000644000000000000000000001065312262561504016405 0ustar rootroot GEN08 GEN08 GEN08 Generate a piecewise cubic spline curve. Description This subroutine will generate a piecewise cubic spline curve, the smoothest possible through all specified points. Syntax f # time size 8 a n1 b n2 c n3 d ... Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). a, b, c, etc. -- ordinate values of the function. n1, n2, n3 ... -- length of each segment measured in stored values. May not be zero, but may be fractional. A particular segment may or may not actually store any values; stored values will be generated at integral points from the beginning of the function. The sum n1 + n2 + ... will normally equal size for fully specified functions. Note GEN08 constructs a stored table from segments of cubic polynomial functions. Each segment runs between two specified points but depends as well on their neighbors on each side. Neighboring segments will agree in both value and slope at their common point. (The common slope is that of a parabola through that point and its two neighbors). The slope at the two ends of the function is constrained to be zero (flat). Hint: to make a discontinuity in slope or value in the function as stored, arrange a series of points in the interval between two stored values; likewise for a non-zero boundary slope. Examples Here is an example of the GEN08 routine. It uses the files gen08.csd. An example of the GEN08 routine. These are the diagrams of the waveforms of the GEN08 routines, as used in the example: f 2 0 65 8 0 16 1 16 1 16 0 17 0 - a curve with a smooth hump in the middle, going briefly negative outside the hump then flat at its ends f 2 0 65 8 0 16 1 16 1 16 0 17 0 - a curve with a smooth hump in the middle, going briefly negative outside the hump then flat at its ends f 3 0 65 8 -1 32 1 2 0 14 0 17 0 - from a negative value,a curve with a smooth hump, going negative creating a small hump then flat at its ends f 3 0 65 8 -1 32 1 2 0 14 0 17 0 - from a negative value,a curve with a smooth hump, going negative creating a small hump then flat at its ends See Also GEN05, GEN06, and GEN07 manual_src~dfsg/scoregens/gen02.xml0000644000000000000000000000756312262561504016405 0ustar rootroot GEN02 GEN02 GEN02 Transfers data from immediate pfields into a function table. Description This subroutine transfers data from immediate pfields into a function table. Syntax f # time size 2 v1 v2 v3 ... Initialization size -- number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement). The maximum tablesize is 16777216 (224) points. v1, v2, v3, etc. -- values to be copied directly into the table space. The number of values is limited by the compile-time variable PMAX, which controls the maximum pfields (currently 1000). The values copied may include the table guard point; any table locations not filled will contain zeros. Note If p4 (the GEN routine number is positive, the table will be post-normalized (rescaled to a maximum absolute value of 1 after generation). A negative p4 will cause rescaling to be skipped. You will usually want to use -2 with this GEN function, so that your values are not normalized. Examples Here is an example of the GEN02 routine. It uses the files gen02.csd. Example of the GEN02 routine. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. These are the diagrams of the waveforms of the GEN02 routines, as used in the example: f 2 0 5 2 0 2 0 f 2 0 5 2 0 2 0 f 3 0 5 2 0 2 10 0 f 3 0 5 2 0 2 10 0 f 4 0 9 2 0 2 10 100 0 f 4 0 9 2 0 2 10 100 0 See Also GEN17 Credits December 2002. Thanks to &namerasmus;, corrected the limit of the PMAX variable. manual_src~dfsg/scoregens/gen52.xml0000644000000000000000000001012212262561504016373 0ustar rootroot GEN52 GEN52 GEN52 Creates an interleaved multichannel table from the specified source tables, in the format expected by the ftconv opcode. Description GEN52 creates an interleaved multichannel table from the specified source tables, in the format expected by the ftconv opcode. It can also be used to extract a channel from a multichannel table and store it in a normal mono table, copy tables with skipping some samples, adding delay, or store in reverse order, etc. Three parameters must be given for each channel to be processed. fsrc declares the source f-table number. The parameter offset specifies an offset for the source file. If different to 0, the source file is not read from the beginning, but the offset number of values are skipped. The offset is used to determine the channel number to be read from interleaved f-tables, e.g. for channel 2, offset must be 1. It can also be used to set a read offset on the source table. This parameter gives absolute values, so if a skip of 20 sample frames for a 2 channel f-table is desired, offset must be set to 40. The srcchnls parameter is used to declare the number of channels in the source f-table. This parameter sets the skip size when reading the source f-table. When more than one channel (nchannels > 1) is given, source f-tables are interleaved in the newly created table. If the source f-table is finished before the destination f-table is full, the remaining values are set to 0. Syntax f # time size 52 nchannels fsrc1 offset1 srcchnls1 [fsrc2 offset2 srcchnls2 ... fsrcN offsetN srcchnlsN] Example ; source tables f 1 0 16384 10 1 f 2 0 16384 10 0 1 ; create 2 channel interleaved table f 3 0 32768 -52 2 1 0 1 2 0 1 ; extract first channel from table 3 f 4 0 16384 -52 1 3 0 2 ; extract second channel from table 3 f 5 0 16384 -52 1 3 1 2 Here is a complete example of the GEN52 opcode. It uses the file gen52.csd. Example of the GEN52 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This is the diagram of the waveform of the GEN52 routine, as used in the example: f 3 0 262144 -52 3 2 0 4 2 1 4 2 2 4 f 3 0 262144 -52 3 2 0 4 2 1 4 2 2 4 Credits Author: &nameistvan; manual_src~dfsg/scoregens/gen33.xml0000644000000000000000000001220712262561504016400 0ustar rootroot GEN33 GEN33 GEN33 Generate composite waveforms by mixing simple sinusoids. Description These routines generate composite waveforms by mixing simple sinusoids, similarly to GEN09, but the parameters of the partials are specified in an already existing table, which makes it possible to calculate any number of partials in the orchestra. The difference between GEN33 and GEN34 is that GEN33 uses inverse FFT to generate output, while GEN34 is based on the algorithm used in oscils opcode. GEN33 allows integer partials only, and does not support power of two plus 1 table size, but may be significantly faster with a large number of partials. On the other hand, with GEN34, it is possible to use non-integer partial numbers and extended guard point, and this routine may be faster if there is only a small number of partials (note that GEN34 is also several times faster than GEN09, although the latter may be more accurate). Syntax f # time size 33 src nh scl [fmode] Initialization size -- number of points in the table. Must be power of two and at least 4. src -- source table number. This table contains the parameters of each partial in the following format: stra, pna, phsa, strb, pnb, phsb, ... the parameters are: stra, strb, etc.: relative strength of partials. The actual amplitude depends on the value of scl, or normalization (if enabled). pna, pnb, etc.: partial number, or frequency, depending on fmode (see below); zero and negative values are allowed, however, if the absolute value of the partial number exceeds (size / 2), the partial will not be rendered. With GEN33, partial number is rounded to the nearest integer. phsa, phsb, etc.: initial phase, in the range 0 to 1. Table length (not including the guard point) should be at least 3 * nh. If the table is too short, the number of partials (nh) is reduced to (table length) / 3, rounded towards zero. nh -- number of partials. Zero or negative values are allowed, and result in an empty table (silence). The actual number may be reduced if the source table (src) is too short, or some partials have too high frequency. scl -- amplitude scale. fmode (optional, default = 0) -- a non-zero value can be used to set frequency in Hz instead of partial numbers in the source table. The sample rate is assumed to be fmode if it is positive, or -(sr * fmode) if any negative value is specified. Examples Here is an example of the GEN33 routine. It uses the file gen33.csd. Example of the gen33 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This is the diagram of the waveform generated by GEN33, as used in the example: f 2 0 262144 -33 1 inumh 1 -1 f 2 0 262144 -33 1 inumh 1 -1 See Also GEN09, GEN34 Credits Programmer: &nameistvan; March 2002 New in version 4.19 manual_src~dfsg/scoregens/gen42.xml0000644000000000000000000000601012262561504016373 0ustar rootroot GEN42 GEN42 GEN42 Generates a random distribution of discrete ranges of values. Description Generates a random distribution function of discrete ranges of values by giving a list of groups of three numbers. Syntax f # time size -42 min1 max1 prob1 min2 max2 prob2 min3 max3 prob3 ... minN maxN probN Performance The first number of each group is a the minimum value of the range, the second is the maximum value and the third is the probability of that an element belonging to that range of values can be chosen by a random algorithm. Probabilities for a range should be a fraction of 1, and the sum of the probabilities for all the ranges should total 1.0. This subroutine is designed to be used together with duserrnd and urd opcodes (see duserrnd for more information). Since both duserrnd and urd do not use any interpolation, it is suggested to give a size reasonably big. Examples Here is an example of the GEN42 opcode. It uses the file gen42.csd. Example of the GEN42 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i1 184.61538 i1 130.76923 i1 169.23077 i1 12.00000 ....... WARNING: Seeding from current time 3751086165 i2 138.46154 i2 12.00000 i2 123.07692 i2 161.53846 i2 123.07692 i2 153.84615 ...... This is the diagram of the waveform of the GEN42 routine, as used in the example: f 1 0 -20 -42 10 20 .3 100 200 .7 f 1 0 -20 -42 10 20 .3 100 200 .7 Credits Author: &namegabriel; manual_src~dfsg/scoregens/rightbrace.xml0000644000000000000000000000421512262561504017573 0ustar rootroot } statement } Statement } Ends a non-sectional, nestable loop. Description The { and } statements can be used to repeat a group of score statements. These loops do not constitute independent score sections and thus may repeat events within the same section. Multiple loops may overlap in time or be nested within each other. Syntax } Initialization All pfields are ignored. Performance The } statement is used in conjunction with the { statement to define repeating groups of other score events. A score loop begins with the { statement which defines the number of repetitions and a unique macro name that will contain the current loop counter. The body of a loop can contain any number of other events (including sectional breaks) and is terminated by a } statement on its own line. The } statement takes no parameters. See the documentation for the { statement for further details. Examples See the examples in the entry for the { statement. Credits Author: &namegabriel; New in Csound version 3.52 (?). (Fixed in version 5.08). manual_src~dfsg/scoregens/gentanh.xml0000644000000000000000000000727312262561504017114 0ustar rootroot GENtanh GENtanh "tanh" Generate a table with values on the tanh function. Description Creates an ftable with values of the tanh function. Syntax f # time size "tanh" start end rescale Initialization size -- number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). start, end -- first and last value to be stored; the GEN draws a curve that goes from start to end: tanh(start) .... tanh(end). The points stored are uniformly spaced between these to the table size rescale -- if not zero the table is not rescaled Examples Here is a simple example of the GENtanh routine. It uses the file gentanh.csd. Example of the GENtanh routine. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. These are the diagrams of the waveforms of the GENtanh routines, as used in the example: f 2 0 8192 "tanh" -100 100 0 - lots of distortion f 2 0 8192 "tanh" -100 100 0 - lots of distortion f 3 0 8192 "tanh" -10 10 0 - less distortion than f2 f 3 0 8192 "tanh" -10 10 0 - less distortion than f2 f 4 0 8192 "tanh" -10 15 0 f 4 0 8192 "tanh" -10 15 0 See Also GENexp and GENsone. More information on this routine: http://www.csounds.com/journal/issue11/distortionSynthesis, written by Victor Lazzarini Credits Written by &namejohn; manual_src~dfsg/math/0000755000000000000000000000000012262561504013676 5ustar rootrootmanual_src~dfsg/math/amp.xml0000644000000000000000000000166012262561504015200 0ustar rootroot
Amplitude Converters Opcodes to convert between different amplitude measurements are: ampdb ampdbfs db dbamp dbfsamp Use rms to find the rms value of a signal. See also 0dbfs for another way to handle amplitudes in csound.
manual_src~dfsg/math/trig.xml0000644000000000000000000000163212262561504015367 0ustar rootroot
Trigonometric Functions Opcodes that perform trigonometric functions are: cos, cosh and cosinv sin, sinh and sininv tan, tanh, taninv, and taninv2.
manual_src~dfsg/math/opeqfunc.xml0000644000000000000000000000223712262561504016244 0ustar rootroot
Opcode Equivalents of Functions Opcodes that perform the equivalent of mathematical functions are: chebyshevpoly divz mac maca polynomial pow product sum taninv2
manual_src~dfsg/math/linearalgebra.xml0000644000000000000000000006242612262561504017222 0ustar rootroot Linear algebra opcodes Mathematical Operations Linear Algebra Opcodes Linear Algebra Opcodes Scalar, vector, and matrix arithmetic on real and complex values. Description These opcodes implement many linear algebra operations, from scalar, vector, and matrix arithmetic up to and including QR based eigenvalue decompositions. The opcodes are designed for digital signal processing, and of course other mathematical operations, in the Csound orchestra language. The numerical implementation uses the gmm++ library from home.gna.org/getfem/gmm_intro. For applications with f-sig variables, array arithmetic must be performed only when the f-sig is "current," because f-rate is some fraction of k-rate; currency can be determined with the la_k_current_f opcode. For applications using assignments between real vectors and a-rate variables, array arithmetic must be performed only when the vectors are "current", because the size of the vector may be some integral multiple of ksmps; currency can be determined by means of the la_k_current_vr opcode. Linear Algebra Data Types Mathematical Type Code Corresponding Csound Type or Types real scalar r i-rate or k-rate variable complex scalar c pair of i-rate or k-rate variables, e.g. "kr, ki" real vector vr i-rate variable holding address of array real vector a a-rate variable real vector t function table number complex vector vc i-rate variable holding address of array complex vector f fsig variable real matrix mr i-rate variable holding address of array complex matrix mc i-rate variable holding address of array
All arrays are 0-based; the first index iterates rows to give columns, the second index iterates columns to give elements. All arrays are general and dense; banded, Hermitian, symmetric and sparse routines are not implemented. An array can be of type code vr, vc, mr, or mc and is stored in an i-rate object. In orchestra code, an array is passed as a MYFLT i-rate variable that contains the address of the array object, which is actually stored in the allocator opcode instance. Although array variables are i-rate, of course their values and even shapes may change at i-rate or k-rate. All operands must be pre-allocated; except for the creation opcodes, no opcode ever allocates any arrays. This is true even if the array appears on the left-hand side of an opcode! However, some operations may reshape arrays to hold results. Arrays are automatically deallocated when their instrument is deallocated. Not only for more efficient performance, but also to make it easier to remember opcode names, the performance rate, output value types, operation names, and input value types are deterministically encoded into the opcode name: "la" for "linear algebra opcode family". "i" or "k" for performance rate. Type code(s) (see above table) for output value(s), but only if the type is not implicit from the input values. Operation name: common mathematical name (preferred) or abbreviation. Type code(s) for input values, if not implicit. For additional details, see the gmm++ documentation at http://download.gna.org/getfem/doc/gmmuser.pdf. Syntax Array Creation ivr la_i_vr_create irows Create a real vector with irows rows. ivc la_i_vc_create irows Create a complex vector with irows rows. imr la_i_mr_create irows, icolumns [, odiagonal] Create a real matrix with irows rows and icolumns columns, with an optional value on the diagonal. imc la_i_mc_create irows, icolumns [, odiagonal_r, odiagonal_i] Create a complex matrix with irows rows and icolumns columns, with an optional complex value on the diagonal. Array Introspection irows la_i_size_vr ivr Return the number of rows in real vector ivr. irows la_i_size_vc ivc Return the number of rows in complex vector ivc. irows, icolumns la_i_size_mr imr Return the number of rows and columns in real matrix imr. irows, icolumns la_i_size_mc imc Return the number of rows and columns in complex matrix imc. kfiscurrent la_k_current_f fsig Return 1 if fsig is current, that is, if the value of fsig will change on the next kperiod. kvriscurrent la_k_current_vr ivr Return 1 if the real vector ivr is current, that is, if Csound's current audio sample frame stands at index 0 of the vector. la_i_print_vr ivr Print the value of real vector ivr. la_i_print_vc ivc Print the value of complex vector ivc. la_i_print_mr imr Print the value of real matrix imr. la_i_print_mc imc Print the value of complex matrix imc. Array Assignment and Conversion ivr la_i_assign_vr ivr Assign the value of the real vector on the right-hand side to the real vector on the left-hand side, at i-rate. ivr la_k_assign_vr ivr Assign the value of the real vector on the right-hand side to the real vector on the left-hand side, at k-rate. ivc la_i_assign_vc ivc ivc la_k_assign_vc ivr imr la_i_assign_mr imr imr la_k_assign_mr imr imc la_i_assign_mc imc imc la_k_assign_mc imr Assignments to vectors from tables or fsigs may resize the vectors. Assignments to vectors from a-rate variables, or to a-rate variables from vectors, will be performed incrementally, one chunk of ksmps elements per kperiod. Therefore, array arithmetic on such vectors should only be performed when the vectors are current, as determined by the la_k_currrent_vr opcode. ivr la_k_assign_a asig ivr la_i_assign_t itablenumber ivr la_k_assign_t itablenumber ivc la_k_assign_f fsig asig la_k_a_assign ivr itablenum la_i_t_assign ivr itablenum la_k_t_assign ivr fsig la_k_f_assign ivc Fill Arrays with Random Elements ivr la_i_random_vr [ifill_fraction] ivr la_k_random_vr [kfill_fraction] ivc la_i_random_vc [ifill_fraction] ivc la_k_random_vc [kfill_fraction] imr la_i_random_mr [ifill_fraction] imr la_k_random_mr [kfill_fraction] imc la_i_random_mc [ifill_fraction] imc la_k_random_mc [kfill_fraction] Array Element Access ivr la_i_vr_set irow, ivalue kvr la_k_vr_set krow, kvalue ivc la_i_vc_set irow, ivalue_r, ivalue_i kvc la_k_vc_set krow, kvalue_r, kvalue_i imr la_i mr_set irow, icolumn, ivalue kmr la_k mr_set krow, kcolumn, ivalue imc la_i_mc_set irow, icolumn, ivalue_r, ivalue_i kmc la_k_mc_set krow, kcolumn, kvalue_r, kvalue_i ivalue la_i_get_vr ivr, irow kvalue la_k_get_vr ivr, krow ivalue_r, ivalue_i la_i_get_vc ivc, irow kvalue_r, kvalue_i la_k_get_vc ivc, krow ivalue la_i_get_mr imr, irow, icolumn kvalue la_k_get_mr imr, krow, kcolumn ivalue_r, ivalue_i la_i_get_mc imc, irow, icolumn kvalue_r, kvalue_i la_k_get_mc imc, krow, kcolumn Single Array Operations imr la_i_transpose_mr imr imr la_k_transpose_mr imr imc la_i_transpose_mc imc imc la_k_transpose_mc imc ivr la_i_conjugate_vr ivr ivr la_k_conjugate_vr ivr ivc la_i_conjugate_vc ivc ivc la_k_conjugate_vc ivc imr la_i_conjugate_mr imr imr la_k_conjugate_mr imr imc la_i_conjugate_mc imc imc la_k_conjugate_mc imc Scalar Operations ir la_i_norm1_vr ivr kr la_k_norm1_vr ivc ir la_i_norm1_vc ivc kr la_k_norm1_vc ivc ir la_i_norm1_mr imr kr la_k_norm1_mr imr ir la_i_norm1_mc imc kr la_k_norm1_mc imc ir la_i_norm_euclid_vr ivr kr la_k_norm_euclid_vr ivr ir la_i_norm_euclid_vc ivc kr la_k_norm_euclid_vc ivc ir la_i_norm_euclid_mr mvr kr la_k_norm_euclid_mr mvr ir la_i_norm_euclid_mc mvc kr la_k_norm_euclid_mc mvc ir la_i_distance_vr ivr kr la_k_distance_vr ivr ir la_i_distance_vc ivc kr la_k_distance_vc ivc ir la_i_norm_max imr kr la_k_norm_max imc ir la_i_norm_max imr kr la_k_norm_max imc ir la_i_norm_inf_vr ivr kr la_k_norm_inf_vr ivr ir la_i_norm_inf_vc ivc kr la_k_norm_inf_vc ivc ir la_i_norm_inf_mr imr kr la_k_norm_inf_mr imr ir la_i_norm_inf_mc imc kr la_k_norm_inf_mc imc ir la_i_trace_mr imr kr la_k_trace_mr imr ir, ii la_i_trace_mc imc kr, ki la_k_trace_mc imc ir la_i_lu_det imr kr la_k_lu_det imr ir la_i_lu_det imc kr la_k_lu_det imc Elementwise Array-Array Operations ivr la_i_add_vr ivr_a, ivr_b ivc la_k_add_vc ivc_a, ivc_b imr la_i_add_mr imr_a, imr_b imc la_k_add_mc imc_a, imc_b ivr la_i_subtract_vr ivr_a, ivr_b ivc la_k_subtract_vc ivc_a, ivc_b imr la_i_subtract_mr imr_a, imr_b imc la_k_subtract_mc imc_a, imc_b ivr la_i_multiply_vr ivr_a, ivr_b ivc la_k_multiply_vc ivc_a, ivc_b imr la_i_multiply_mr imr_a, imr_b imc la_k_multiply_mc imc_a, imc_b ivr la_i_divide_vr ivr_a, ivr_b ivc la_k_divide_vc ivc_a, ivc_b imr la_i_divide_mr imr_a, imr_b imc la_k_divide_mc imc_a, imc_b Inner Products ir la_i_dot_vr ivr_a, ivr_b kr la_k_dot_vr ivr_a, ivr_b ir, ii la_i_dot_vc ivc_a, ivc_b kr, ki la_k_dot_vc ivc_a, ivc_b imr la_i_dot_mr imr_a, imr_b imr la_k_dot_mr imr_a, imr_b imc la_i_dot_mc imc_a, imc_b imc la_k_dot_mc imc_a, imc_b ivr la_i_dot_mr_vr imr_a, ivr_b ivr la_k_dot_mr_vr imr_a, ivr_b ivc la_i_dot_mc_vc imc_a, ivc_b ivc la_k_dot_mc_vc imc_a, ivc_b Matrix Inversion imr, icondition la_i_invert_mr imr imr, kcondition la_k_invert_mr imr imc, icondition la_i_invert_mc imc imc, kcondition la_k_invert_mc imc Matrix Decompositions and Solvers ivr la_i_upper_solve_mr imr [, j_1_diagonal] ivr la_k_upper_solve_mr imr [, j_1_diagonal] ivc la_i_upper_solve_mc imc [, j_1_diagonal] ivc la_k_upper_solve_mc imc [, j_1_diagonal] ivr la_i_lower_solve_mr imr [, j_1_diagonal] ivr la_k_lower_solve_mr imr [, j_1_diagonal] ivc la_i_lower_solve_mc imc [, j_1_diagonal] ivc la_k_lower_solve_mc imc [, j_1_diagonal] imr, ivr_pivot, isize la_i_lu_factor_mr imr imr, ivr_pivot, ksize la_k_lu_factor_mr imr imc, ivr_pivot, isize la_i_lu_factor_mc imc imc, ivr_pivot, ksize la_k_lu_factor_mc imc ivr_x la_i_lu_solve_mr imr, ivr_b ivr_x la_k_lu_solve_mr imr, ivr_b ivc_x la_i_lu_solve_mc imc, ivc_b ivc_x la_k_lu_solve_mc imc, ivc_b imr_q, imr_r la_i_qr_factor_mr imr imr_q, imr_r la_k_qr_factor_mr imr imc_q, imc_r la_i_qr_factor_mc imc imc_q, imc_r la_k_qr_factor_mc imc ivr_eig_vals la_i_qr_eigen_mr imr, i_tolerance ivr_eig_vals la_k_qr_eigen_mr imr, k_tolerance ivr_eig_vals la_i_qr_eigen_mc imc, i_tolerance ivr_eig_vals la_k_qr_eigen_mc imc, k_tolerance Matrix must be Hermitian in order to compute eigenvectors. ivr_eig_vals, imr_eig_vecs la_i_qr_sym_eigen_mr imr, i_tolerance ivr_eig_vals, imr_eig_vecs la_k_qr_sym_eigen_mr imr, k_tolerance ivc_eig_vals, imc_eig_vecs la_i_qr_sym_eigen_mc imc, i_tolerance ivc_eig_vals, imc_eig_vecs la_k_qr_sym_eigen_mc imc, k_tolerance Credits Michael Gogins New in Csound version 5.09 manual_src~dfsg/math/mathfunc.xml0000644000000000000000000000317012262561504016226 0ustar rootroot
Mathematical Functions Opcodes that perform mathematical functions are: abs ceil exp floor frac int log log10 logbtwo pow powershape powoftwo round sqrt
manual_src~dfsg/math/array.xml0000644000000000000000000003215012262561504015537 0ustar rootroot Array opcodes Mathematical Operations Array Opcodes Array Opcodes Variable Name An array must be created (via init or fillarray) as kMyName plus ending brackets. The brackets determine the dimensions of the array. So, kArr[] init 10 creates a one-dimensional array of length 10, whereas kArr[][] init 10, 10 creates a two-dimensional array with 10 rows and 10 columns. After the initalization of the array, referring to the array as a whole is done without any brackets. Brackets are only used if an element is indexed: kArr[] init 10 ;with brackets because of initialization kLen = lenarray(kArr) ;without brackets kFirstEl = kArr[0] ;indexing with brackets The same syntax is used for a simple copy via the '=' operator: kArr1[] fillarray 1, 2, 3, 4, 5 kArr2[] = kArr1 ;creates kArr2 as copy of kArr1 k-rate Note that most array operations are currently k-rate only. So like any other k-rate opcode, an operation on arrays will be automatically repeated every k-cycle. For instance, this code will repeat re-writing the array with different random values every k-cycle, as long as the instrument runs: kArr[] init 10 kIndx = 0 until kIndx == lenarray(kArr) do kArr[kIndx] rnd31 10, 0 kIndx += 1 od If you want to avoid this, you must organize it in one of the usual ways, for instance by using a trigger: kArr[] init 10 kTrig metro 1 if kTrig == 1 then ;do the following once a second kIndx = 0 until kIndx == lenarray(kArr) do kArr[kIndx] rnd31 10, 0 kIndx += 1 od endif Creation/Initialization The usual way to create an array is with init: kArr[] init 10 ;creates one-dimensional array with length 10 kArr[][] init 10, 10 ;creates two-dimensional array A one-dimensional array can also be created and filled with distinct values by the opcode fillarray. This line creates a vector with length 4 and puts in the numbers [1, 2, 3, 4]: kArr[] fillarray 1, 2, 3, 4 Length The function lenarray(kArr) reports the length of an array. See example for function lenarray. Copy Arrays to/from Tables copyf2array kArr, kfn copies data from an ftable to a vector. copya2ftab kArr, kfn copies data from a vector to an function table. See examples for opcodes copyf2array and copya2ftab. Array Operations: Math +, -, *, / on a Number If the four basic math operators are used between an array and a scalar (number), the operation is applied to each element. The safest way to do this is to store the result in a new array: kArr1[] fillarray 1, 2, 3 kArr2[] = kArr1 + 10 ;(kArr2 is now [11, 12, 13]) Here is an example of array/scalar operations. It uses the file array_scalar_math.csd. Example of array operations +, -, *, / on a Second Array If the four basic math operators are used between two arrays, the operation is applied element by element. The result can be straightforward stored in a new array: kArr1[] fillarray 1, 2, 3 kArr2[] fillarray 10, 20, 30 kArr3[] = kArr1 + kArr2 ;(kArr3 is now [11, 22, 33]) Here is an example of array operations. It uses the file array_array_math.csd. Example of array operations Map a Function to an Array kArrRes maparray kArrSrc, "fun" maps the k-rate 1-arg function in the string to every element of the vector. Possible functions are for instance abs, ceil, exp, floor, frac, int, log, log10, round, sqrt. This is a simple example: kArrSrc[] fillarray 1, 2, 3, 4, 5 kArrRes[] init 5 kArrRes maparray kArrSrc, "sqrt" See example for opcode maparray. Array Operations: min, max, sum, scale, slice Minimum and Maximum kMin [,kMinIndx] minarray kArr returns the smallest value in an array, and optionally its index. kMax [,kMaxIndx] maxarray kArr returns the largest value in an array, and optionally its index. See examples for opcodes minarray and maxarray. Sum kSum sumarray kArr returns the sum of all values in kArr. See example for opcode sumarray. Scale scalearray kArr, kMin, kMax scales all values in kArr between kMin and kMax. kArr[] fillarray 1, 3, 9, 5, 6 scalearray kArr, 1, 3 changes kArr to [1, 1.5, 3, 2, 2.25]. See example for opcode scalearray. Slice slicearray kArr, iStart, iEnd returns a slice of kArr from index iStart to index iEnd (included). The array for receiving the slice must have been created in advance: kArr[] fillarray 1, 2, 3, 4, 5, 6, 7, 8, 9 kArr1[] init 5 kArr2[] init 4 kArr1 slicearray kArr, 0, 4 ;[1, 2, 3, 4, 5] kArr2 slicearray kArr, 5, 8 ;[6, 7, 8, 9] See example for opcode slicearray. Arrays in UDOs The dimension of an input array must be declared in two places: as k[] or k[][] in the type input list as kName[], kName[][] etc in the xin list. For instance : opcode FirstEl, k, k[] ;returns the first element of vector kArr kArr[] xin xout kArr[0] endop Here is an example of an array in an UDO. It uses the file array_udo.csd. Example of an array in an UDO Credits This manual page has been written by Joachim Heintz. July 2013 New in Csound 6.00 manual_src~dfsg/math/artlogic.xml0000644000000000000000000000166512262561504016234 0ustar rootroot
Arithmetic and Logic Operations Opcodes that perform arithmetic and logic operations are -, +, &&, ||, *, /, ˆ, and %. See the Conditional Values section and the if family of opcodes for usage of logical operators.
manual_src~dfsg/math/top.xml0000644000000000000000000000033212262561504015220 0ustar rootroot Mathematical Operations &mathamp; &mathartlogic; &sigmodcompaccum; &mathmathfunc; &mathopeqfunc; &mathrndfunc; &mathtrig; &linearalgebra; &array; manual_src~dfsg/math/rndfunc.xml0000644000000000000000000000104112262561504016053 0ustar rootroot
Random Functions Opcodes that perform random functions are: birnd rnd See the section Random (Noise) Generators for opcodes that generate random signals.
manual_src~dfsg/oscnetwork/0000755000000000000000000000000012262561504015143 5ustar rootrootmanual_src~dfsg/oscnetwork/OSC.xml0000644000000000000000000000152412262561504016313 0ustar rootroot
OSC OSC enables interaction between different audio processes, and in particular between Csound and other synthesis engines. The following opcodes are available: OSCinit - Start an OSC listener thread. OSClisten - Receive OSC messages. OSCsend - Send an OSC message. Credits By: John ffitch with the liblo library as inspiration and support.
manual_src~dfsg/oscnetwork/top.xml0000644000000000000000000000014512262561504016467 0ustar rootroot OSC and Network &OSC; &Network; &Remote; manual_src~dfsg/oscnetwork/Network.xml0000644000000000000000000000063112262561504017316 0ustar rootroot
Network The following opcodes can stream or receive audio through UDP: sockrecv socksend
manual_src~dfsg/oscnetwork/Remote.xml0000644000000000000000000000225512262561504017124 0ustar rootroot
Remote Opcodes The Remote opcodes enable transmission of score or MIDI events through a network, so remote instances (or a different local instance) can process them. The following opcodes are available: insglobal - Used to implement a remote orchestra. insremot - Used to implement a remote orchestra. midiglobal - Used to implement a remote MIDI orchestra. midiremot - Used to implement a remote MIDI orchestra. remoteport - Defines the port for use with the remote system.
manual_src~dfsg/makeframes.py0000644000000000000000000003062412262561504015437 0ustar rootroot# This python script generates the extra files needed for the frames # version of the manual. It generates the files: # indexframes.html, contents.html and title.html # Output is based on html files created by Michael Rhoades # Script written by Andres Cabrera June 2006 # Licensed under the GPL licence version 3 or later import sys from xml.dom import minidom version = sys.argv[1:] if (len(version)==0): version=[''] indexframesFile = open('html/indexframes.html','w') creditsFile = open('html/credits.html','w') titleFile = open('html/title.html','w') contentsFile = open('html/contents.html','w') indexFile = open('html/index.html','r') index = indexFile.read() indexFile.seek(0) indexparse = minidom.parse(indexFile) frameLink = ''' ''' s = '''

The Canonical Csound Reference Manual

''' titleEnd = index.find(s) + len(s) + 1 if (index[titleEnd:titleEnd+len(frameLink)]==frameLink): print '''************************************** Warning: index.html already processed! **************************************''' # indexframes.html ----------------------------------------- indexframesFile.write('''\n\n Csound ''' +version[0]+ ''' Manual - Frames Version <body> <p>This page uses frames, but your browser doesn\'t support them.</p> </body> ''') indexframesFile.close() # credits.html ----------------------------------------- creditsFile.write('''\n \n ''') creditsFile.write(index[index.find('
')+106]) creditsFile.write('
\n \n') creditsFile.close() # title.html ----------------------------------------- titleFile.write('''\n\n The Canonical Csound Reference Manual\n

The Canonical Csound Reference Manual
| A | B | C | D | E | F | G | H | I | J | K | L | M | N | O | P | R | S | T | U | V | W | X | Z |
Score Statements | GEN Routines | Quick Reference

\n\n''') titleFile.close() # contents.html ----------------------------------------- contentsFile.write(''' Contents
''') #routine to find the toc navheader = indexparse.getElementsByTagName('div') i = 0 while (navheader[i]!=''): #print navheader[i].attributes.keys() if (len(navheader[i].attributes.keys())!=0): if (navheader[i].attributes.keys()[0]=='class'): #print navheader[i].attributes['class'].value if(navheader[i].attributes['class'].value == 'toc'): break i += 1 toc = navheader[i] sections = toc.getElementsByTagName('span') a = toc.getElementsByTagName('a') for j in range(len(a)): if (a[j].attributes['href'].value=='OpcodesTop.html'): overview=a[j].parentNode.parentNode.nextSibling.nextSibling elif (a[j].attributes['href'].value=='ScoregensTop.html'): scoregens=a[j].parentNode.parentNode.nextSibling.nextSibling break #print overview.toxml().encode('utf-8') # Routine to make titles bigger and black, except Appendices for j in range(len(sections)): #print "---------\n", sections[j].toxml() #raw_input() if ((sections[j].attributes['class'].value!='section')and (sections[j].attributes['class'].value!='chapter')and (sections[j].attributes['class'].value!='appendix')and (sections[j].attributes['class'].value!='refentrytitle')and (sections[j].attributes['class'].value!='refpurpose')and (sections[j].attributes['class'].value!='quote')and (sections[j].attributes['class'].value!='emphasis')): print "--------Title:", sections[j].childNodes[1].firstChild.toxml().encode('utf-8') font = minidom.Element('font') font.setAttribute('size','4') font.setAttribute('color','Black') font.appendChild(minidom.Element('br')) if (sections[j].childNodes[1].childNodes[0].toxml().encode('utf-8')=='Glossary'): font.appendChild(sections[j].childNodes[1].cloneNode(True)) else: font.appendChild(sections[j].childNodes[1].childNodes[0].cloneNode(True)) font.appendChild(minidom.Element('br')) #print sections[j].firstChild.toxml().encode('latin-1') #print font.toxml().encode('latin-1') parent = sections[j].parentNode parent.appendChild(font) parent.removeChild(sections[j]) #spanElement = sections[j].getElementsByTagName('span') #try: #sections[j].removeChild(spanElement) #except: #print "No span for", sections[j].toxml() #print sections[j].toxml().encode('latin-1') #print font.toxml().encode('latin-1') elif (sections[j].attributes['class'].value=='appendix'): font = minidom.Element('font') font.setAttribute('size','4') font.appendChild(sections[j].childNodes[1].cloneNode(True)) font.appendChild(minidom.Element('br')) #print sections[j].firstChild.toxml().encode('latin-1') #print font.toxml().encode('latin-1') parent = sections[j].parentNode parent.appendChild(font) parent.removeChild(sections[j]) #print sections[j].toxml().encode('latin-1') #print font.toxml().encode('latin-1') else: #print sections[j].toxml().encode('latin-1') parent = sections[j].parentNode newNode = sections[j].getElementsByTagName('a') font = minidom.Element('font') font.setAttribute('size','2') #font.setAttribute('color','Black') #font.appendChild(minidom.Element('br')) #if (sections[j].attributes['class'].value!='refpurpose' and #sections[j].attributes['class'].value!='refentrytitle' and #sections[j].attributes['class'].value!='quote' and #sections[j].attributes['class'].value!='emphasis'): if newNode != []: # font.appendChild(newNode[0].cloneNode(True)) parent.appendChild(newNode[0].cloneNode(True)) # parent.appendChild(font) parent.removeChild(sections[j]) #print sections[j].toxml().encode('latin-1') #print toc.toxml() #.encode('latin-1') # routine to remove text description from opcode entries and # add alphabetical references and letters unwantedtext = overview.getElementsByTagName('dt') alphabetcaps = [] alphabet = [] for num in range(65,93): alphabetcaps.extend(chr(num)) alphabet.extend(chr(num+32)) count = -1 for text in unwantedtext: if count==15 or count==23: count = count + 1 #There is no 'Q' or 'Y' #print "text: ", text.toxml() text.removeChild(text.childNodes[0]) #text.appendChild(minidom.Element('br')) #print "Unwanted: ", text.toxml() node = text.getElementsByTagName('a') if ((node[0].firstChild.toxml().encode('utf-8')[0]==alphabet[count+1]) or(node[0].toxml().encode('utf-8')[0]==alphabetcaps[count+1])): count += 1 string = '' + alphabetcaps[count] + '' #print "string:", string letter = minidom.parseString(string) text.insertBefore(minidom.Element('br'), text.firstChild) text.insertBefore(letter.firstChild, text.firstChild) print "unwanted2: ", text.firstChild #print text.toxml().encode('latin-1') # routine to remove text description from score statement and GEN entries unwantedtext = scoregens.getElementsByTagName('dt') for text in unwantedtext: node = text.firstChild if node.nodeType == minidom.Node.ELEMENT_NODE and node.tagName == 'a': text.removeChild(text.childNodes[1]) text.appendChild(minidom.Element('br')) # Routines to remove indentation #dt = toc.getElementsByTagName('dt') #for j in range(len(dt)): #spanElement = dt.getElementsByTagName('span') #if spanElement.attributes['class'].value == 'section': #break #parent = dt[j].parentNode #print parent.toxml() #for k in range(len(dt[j].childNodes)): #parent.insertBefore(dt[j].firstChild ,dt[j]) #parent.removeChild(dt[j]) #dl = toc.getElementsByTagName('dl') #for j in range(len(dl)): #parent = dl[j].parentNode #for k in range(len(dl[j].childNodes)): #parent.insertBefore(dl[j].firstChild ,dl[j]) #parent.removeChild(dl[j]) #dd = toc.getElementsByTagName('dd') #for j in range(len(dd)): #parent = dd[j].parentNode #if (parent.nodeName!='dd'): #for k in range(len(dd[j].childNodes)): #parent.insertBefore(dd[j].firstChild ,dd[j]) #parent.removeChild(dd[j]) #print toc #print toc.toxml().encode('latin-1') toc = toc.toxml().encode('utf-8') contentsFile.write(toc) contentsFile.close() # Modify index.html to include a link to the frames version if (index[titleEnd:titleEnd+len(frameLink)]!=frameLink): newindex = index[0:titleEnd]+ frameLink newindex += index[titleEnd:] indexFile.close() indexFile = open('html/index.html','w') indexFile.seek(0) indexFile.write(newindex) indexFile.close() manual_src~dfsg/table/0000755000000000000000000000000012262561504014034 5ustar rootrootmanual_src~dfsg/table/readwrit.xml0000644000000000000000000000532212262561504016401 0ustar rootroot
Read/Write Operations Opcodes that read and write to a table are: ftloadk ftload ftsavek ftsave tablecopy tablegpw tableicopy tableigpw tableimix tableiw tablemix tablera tablew tablewa tablewkt tabmorph tabmorpha tabmorphak tabmorphi tabrec tabplay ftmorf Table values can be accessed within expressions using the tb family of opcodes. Many oscillators are in fact specialized table readers. See the Basic oscillators section.
manual_src~dfsg/table/top.xml0000644000000000000000000000263512262561504015366 0ustar rootroot Function Table Control Refer to the f score statement, ftgen, ftgentmp, ftgenonce and the GEN Routines section for information on creating tables. Tables can be removed from memory using the ftfree opcode. Tables by default, require a size which is a power of two. However tables with any size can be generated by specifying the size as a negative number (see f score statement). Not all opcodes accept tables whose size is not a power of two, as this may be a requirement for internal processing. For information on table access, consult the section Table Access. Tables for use with the loscilx opcode can be loaded using sndload. &tablequeries; &tablereadwrit; &tableselect; manual_src~dfsg/table/select.xml0000644000000000000000000000102412262561504016032 0ustar rootroot
Table Reading with Dynamic Selection Opcodes that let one dynamically (at k-rate) select tables are: tableikt tablekt tablexkt
manual_src~dfsg/table/queries.xml0000644000000000000000000000177112262561504016241 0ustar rootroot
Table Queries Opcodes the query tables for information are: For tables loaded from a sound file (using GEN01): ftchnls, ftcps,ftlen, ftlptim and ftsr For any table: nsamp, ftlen, tableng The opcode tabsum calculates the sum of values in a table.
manual_src~dfsg/utility/0000755000000000000000000000000012262561504014450 5ustar rootrootmanual_src~dfsg/utility/scale.xml0000644000000000000000000001057112262561504016265 0ustar rootroot Utilities scale scale Scale the amplitude of a sound file. scale Description Takes a sound file and scales it by applying a gain, either constant or variable. The scale can be specified as a multiplier, a maximum or a percentage of 0db. Syntax scale [OPTIONS ... ] infile Initialization Flags: -A = Generate an AIFF outout file. -W = Generate an WAV outout file. -h = Generate an outout file with no header. -c = Generate 8-bit signed_char sound samples. -a = Generate alaw sound samples. -u = Generate ulaw sound samples. -s = Generate short integer sound samples. -l = Generate long (32 bit) integer sound samples. -f = Generate floating point samples. -F arg = Specifies the gain to be applied. If arg is a floating point number that gain is applied uniformly to the input. Alternatively it could be a file name which specifies a breakpoint file for varying the gain for different periods. -M fpnum = Scales the input so the maximum absolute displacement is the value given. -P fpnum = Scales the input so the maximum absolute displacement is the pencentage given of 0db. -R = Continually rewrite the header while writing soundfile (WAV/AIFF). -H integer = Show a "heart-beat" to indicate progress, in style 1, 2 or 3. -N = Alert call (usually ringing the bell) when finished. -o fname = output file name (default: test.wav) Examples scale -s -W -F 0.96 -o out.wav sound.wav This creates a new sound file with a constant gain of 0.96. It is particularly useful if the input file is in floating point format. Credits Author: &namejohn; 1994 manual_src~dfsg/utility/sdif2ad.xml0000644000000000000000000001030512262561504016505 0ustar rootroot Utilities sdif2ad sdif2ad Converts SDIF files to files usable by adsyn. sdif2ad Description Convert files Sound Description Interchange Format (SDIF) to the format usable by Csound's adsyn opcode. As of Csound version 4.10, sdif2ad was available only as a standalone program for Windows console and DOS. Syntax sdif2ad [flags] infilename outfilename Initialization Flags: -sN -- apply amplitude scale factor N -pN -- keep only the first N partials. Limited to 1024 partials. The source partial track indices are used directly to select internal storage. As these can be arbitrary values, the maximum of 1024 partials may not be realized in all cases. -r -- byte-reverse output file data. The byte-reverse option is there to facilitate transfer across platforms, as Csound's adsyn file format is not portable. If the filename passed to hetro has the extension .sdif, data will be written in SDIF format as 1TRC frames of additive synthesis data. The utility program sdif2ad can be used to convert any SDIF file containing a stream of 1TRC data to the Csound adsyn format. sdif2ad allows the user to limit the number of partials retained, and to apply an amplitude scaling factor. This is often necessary, as the SDIF specification does not, as of the release of sdif2ad, require amplitudes to be within a particular range. sdif2ad reports information about the file to the console, including the frequency range. The main advantages of SDIF over the adsyn format, for Csound users, is that SDIF files are fully portable across platforms (data is big-endian), and do not have the duration limit of 32.76 seconds imposed by the 16 bit adsyn format. This limit is necessarily imposed by sdif2ad. Eventually, SDIF reading will be incorporated directly into adsyn, thus enabling files of any length (subject to system memory limits) to be analysed and processed. Users should remember that the SDIF formats are still under development. While the 1TRC format is now fairly well established, it can still change. For detailed information on the Sound Description Interchange Format, refer to the CNMAT website: http://cnmat.CNMAT.Berkeley.EDU/SDIF Some other SDIF resources (including a viewer) are available via the NC_DREAM website: http://www.bath.ac.uk/~masjpf/NCD/dreamhome.html Credits Author: Richard Dobson Somerset, England August, 2000 New in Csound version 4.08 manual_src~dfsg/utility/sndinfo.xml0000644000000000000000000000547512262561504016645 0ustar rootroot Utilities sndinfo sndinfo Displays information about a soundfile. sndinfo Description Get basic information about one or more soundfiles. Syntax csound -U sndinfo [options] soundfilenames ... sndinfo [options] soundfilenames ... Initialization sndinfo will attempt to find each named file, open it for reading, read in the soundfile header, then print a report on the basic information it finds. The order of search across soundfile directories is as described above. If the file is of type AIFF, some further details are listed first. There are two option types: -i or -i1 will print instrument information, which includes looping. The option continues until a -i0 option. The other option is -b which prints the broadcast information for WAV files. It can similarly be negated with -b0. Examples csound -U sndinfo test Bosendorfer/"BOSEN mf A0 st" foo foo2 where the environment variables SFDIR = /u/bv/sound, and SSDIR = /so/bv/Samples, might produce the following: util SNDINFO: /u/bv/sound/test: srate 22050, monaural, 16 bit shorts, 1.10 seconds headersiz 1024, datasiz 48500 (24250 sample frames) /so/bv/Samples/Bosendorfer/BOSEN mf A0 st: AIFF, 197586 stereo samples, base Frq 261.6 (MIDI 60), sustnLp: mode 1, 121642 to 197454, relesLp: mode 0 AIFF soundfile, looping with modes 1, 0 srate 44100, stereo, 16 bit shorts, 4.48 seconds headersiz 402, datasiz 790344 (197586 sample frames) /u/bv/sound/foo: no recognizable soundfile header /u/bv/sound/foo2: couldn't find manual_src~dfsg/utility/pv_import.xml0000644000000000000000000000256412262561504017220 0ustar rootroot Utilities pv_import pv_import Converts a comma-separated text file to a .pvx file. pv_import Syntax pv_import cstext_file pv_file csound -U pv_import cstext_file pv_file Initialization cstext_file - Name of the input comma-separated text file. pv_file - Name of the output .pvx file. The pv_import utility generates a .pvx file usable with the pvoc generator. It can be used in combination with pv_export to modify sound analysis made by the PVANAL utility. Credits Author: John ffitch 1995 manual_src~dfsg/utility/convertXO.xml0000644000000000000000000000462612262561504017131 0ustar rootroot
File Conversion (DNOISE, HET_IMPORT, HET_EXPORT, PVLOOK, PV_EXPORT, PV_IMPORT, SDIF2AD, SRCONV) The following utilities exist for file conversion: DNOISE: Reduces noise in a file. HET_EXPORT: Exports a .het (produced by HETRO) to a comma separated text file. HET_IMPORT: Generates a .het (in the format produced by HETRO) from a comma separated text file for usage with the adsyn generator. PVLOOK: View formatted text output of STFT analysis files. PV_EXPORT: Converts a file generated by PVANAL to a text file. PV_IMPORT: Converts a text file (in the format generated by PV_EXPORT) to a PVANAL format file to be used with the pvoc opcode. SDIF2AD: Converts SDIF files to files usable by adsynt. SRCONV: Converts the sample rate of an audio file. &utilityhet_export; &utilityhet_import; &utilitypvlook; &utilitypv_export; &utilitypv_import; &utilitysdif2ad; &utilitysrconv;
manual_src~dfsg/utility/envext.xml0000644000000000000000000000373412262561504016512 0ustar rootroot Utilities envext envext Extracts the envelope of a file to a text file. envext Syntax envext [-flags] soundfile csound -U envext [-flags] soundfile Initialization soundfile - Name of the input soundfile. The following flags are available for envext (The default values are stated in parenthesis): -o fnam Name of output filename (newenv) -w size (in seconds) of analysis window (0.25) The envext utility generates a text file containing time and amplitude pairs by finding the absolute peak within each window. Example Using the command (while in the manual directory): csound -U envext examples/mary.wav will produce the a text file containing the following: 0.000 0.000 0.000 0.000 0.250 0.000 0.500 0.000 0.750 0.000 1.249 0.170 1.499 0.269 1.530 0.307 1.872 0.263 2.056 0.304 2.294 0.241 2.570 0.216 2.761 0.178 3.077 0.011 3.251 0.001 3.500 0.000 Which shows the time for the peak amplitude within each measured window. Credits Author: John ffitch 1995 manual_src~dfsg/utility/hetro.xml0000644000000000000000000001575612262561504016331 0ustar rootroot Utilities hetro hetro Decomposes an input soundfile into component sinusoids. hetro Description Hetrodyne filter analysis for the Csound adsyn generator. Syntax csound -U hetro [flags] infilename outfilename hetro [flags] infilename outfilename Initialization hetro takes an input soundfile, decomposes it into component sinusoids, and outputs a description of the components in the form of breakpoint amplitude and frequency tracks. Analysis is conditioned by the control flags below. A space is optional between flag and value. -s srate -- sampling rate of the audio input file. This will over-ride the srate of the soundfile header, which otherwise applies. If neither is present, the default is 10000. Note that for adsyn synthesis the srate of the source file and the generating orchestra need not be the same. -c channel -- channel number sought. The default is 1. -b begin -- beginning time (in seconds) of the audio segment to be analyzed. The default is 0.0 -d duration -- duration (in seconds) of the audio segment to be analyzed. The default of 0.0 means to the end of the file. Maximum length is 32.766 seconds. -f begfreq -- estimated starting frequency of the fundamental, necessary to initialize the filter analysis. The default is 100 (cps). -h partials -- number of harmonic partials sought in the audio file. Default is 10, maximum is a function of memory available. -M maxamp -- maximum amplitude summed across all concurrent tracks. The default is 32767. -m minamp -- amplitude threshold below which a single pair of amplitude/frequency tracks is considered dormant and will not contribute to output summation. Typical values: 128 (48 db down from full scale), 64 (54 db down), 32 (60 db down), 0 (no thresholding). The default threshold is 64 (54 db down). -n brkpts -- initial number of analysis breakpoints in each amplitude and frequency track, prior to thresholding (-m) and linear breakpoint consolidation. The initial points are spread evenly over the duration. The default is 256. -l cutfreq -- substitute a 3rd order Butterworth low-pass filter with cutoff frequency cutfreq (in Hz), in place of the default averaging comb filter. The default is 0 (don't use). Performance As of Csound 4.08, hetro can write SDIF output files if the output file name ends with ".sdif" or ".SDIF". See the sdif2ad utility for more information about the Csound's SDIF support. Examples hetro -s44100 -b.5 -d2.5 -h16 -M24000 audiofile.test adsynfile7 This will analyze 2.5 seconds of channel 1 of a file "audiofile.test", recorded at 44.1 kHz, beginning .5 seconds from the start, and place the result in a file "adsynfile7". We request just the first 16 harmonics of the sound, with 256 initial breakpoint values per amplitude or frequency track, and a peak summation amplitude of 24000. The fundamental is estimated to begin at 100 Hz. Amplitude thresholding is at 54 db down. The Butterworth LPF is not enabled. File Format The output file contains time-sequenced amplitude and frequency values for each partial of an additive complex audio source. The information is in the form of breakpoints (time, value, time, value, ....) using 16-bit integers in the range 0 - 32767. Time is given in milliseconds, and frequency in Hertz (cps). The breakpoint data is exclusively non-negative, and the values -1 and -2 uniquely signify the start of new amplitude and frequency tracks. A track is terminated by the value 32767. Before being written out, each track is data-reduced by amplitude thresholding and linear breakpoint consolidation. A component partial is defined by two breakpoint sets: an amplitude set, and a frequency set. Within a composite file these sets may appear in any order (amplitude, frequency, amplitude ....; or amplitude, amplitude..., then frequency, frequency,...). During adsyn resynthesis the sets are automatically paired (amplitude, frequency) from the order in which they were found. There should be an equal number of each. A legal adsyn control file could have following format: -1 time1 value1 ... timeK valueK 32767 ; amplitude breakpoints for partial 1 -2 time1 value1 ... timeL valueL 32767 ; frequency breakpoints for partial 1 -1 time1 value1 ... timeM valueM 32767 ; amplitude breakpoints for partial 2 -2 time1 value1 ... timeN valueN 32767 ; frequency breakpoints for partial 2 -2 time1 value1 .......... -2 time1 value1 .......... ; pairable tracks for partials 3 and 4 -1 time1 value1 .......... -1 time2 value1 .......... Credits Author: Tom Sullivan 1992 Author: &namejohn; 1994 Author: Richard Dobson 2000 October 2002. Thanks to &namerasmus;, added a note about the SDIF format. manual_src~dfsg/utility/atsa.xml0000644000000000000000000002300312262561504016120 0ustar rootroot Utilities atsa atsa Performs ATS analysis on a soundfile. atsa Description ATS analysis for use with the Csound ATS Resynthesis opcodes. Syntax csound -U atsa [flags] infilename outfilename Initialization The following flags can be set for atsa (The default values are stated in parenthesis): -b start (0.000000 seconds) -e duration (0.000000 seconds or end) -l lowest frequency (20.000000 Hertz) -H highest frequency (20000.000000 Hertz) -d frequency deviation (0.100000 of partial freq.) -c window cycles (4 cycles) -w window type (type: 1) (Options: 0=BLACKMAN, 1=BLACKMAN_H, 2=HAMMING, 3=VONHANN) -h hop size (0.250000 of window size) -m lowest magnitude (-60.000000) -t track length (3 frames) -s min. segment length (3 frames) -g min. gap length (3 frames) -T SMR threshold (30.000000 dB SPL) -S min. segment SMR (60.000000 dB SPL) -P last peak contribution (0.000000 of last peak's parameters) -M SMR contribution (0.500000) -F File Type (type: 4) (Options: 1=amp.and freq. only, 2=amp.,freq. and phase, 3=amp.,freq. and residual, 4=amp.,freq.,phase, and residual) Parameters ATS analysis was devised by Juan Pampin. For complete information on ATS visit: http://www-ccrma.stanford.edu/~juan/ATS.html. Analysis parameters must be carefully tuned for the Analysis Algorithm (ATSA) to properly capture the nature of the signal to be analyzed. As there are a significant number of them, ATSH offers the possibility of Saving/Loading them in a Binary File carrying the extension "*.apf". The extension is not mandatory, but recommended. A brief explanation of each Analysis Parameters follows: Start (secs.): the starting time of the analysis in seconds. Duration (secs.): the duration time of the analysis in seconds. A zero means the whole duration of the input sound file. Lowest Frequency (Hz.): this parameter will partially determine the size of the Analysis Window to be used. To compute the size of the Analysis Window, the period of the Lowest Frequency in samples (SR / LF) is multiplied by the number of cycles of it the user wants to fit in the Analysis Window (see parameter 6). This value is rounded to the next power of two to determine the size of the FFT for the analysis. The remaining samples are zero-padded. If the signal is a single, harmonic sound, then the value of the Lowest Frequency should be its fundamental frequency or a sub-harmonic of it. If it is not harmonic, then its lowest significant frequency component may be a good starting value. Highest Frequency (Hz.): highest frequency to be taken into account for Peak Detection. Once it is determined that no relevant information is found beyond a certain frequency, the analysis may be faster and more accurate setting the Highest Frequency parameter to that value. Frequency Deviation (Ratio): frequency deviation allowed for each peak in the Peak Continuation Algorithm, as a ratio of the frequency involved. For instance, considering a peak at 440 Hz and a Deviation of .1 will produce that the Peak Continuation Algorithm will only try to find candidates for its continuation between 396 and 484 Hz (10% above and below the frequency of the peak). A small value is likely to produce more trajectories whilst a large value will reduce them, but at the cost of rendering information difficult to be further processed. Number of Cycles of Lowest Frequency to fit in Analysis Window: this will also partially determine the size of the Fourier Analysis Window to be used. See Parameter 3. For single harmonic signals, it is supposed to be more than one (typically 4). Hop Size (Ratio): size of the gap between one Analysis Window and the next expressed as a ratio of the Window Size. For instance, a Hop Size value of .25 will "jump" by 512 samples (Windows will overlap for a 75% of their size). This parameter will also determine the size of the analysis frames obtained. Signals that change their spectra very fast (such as Speech sounds) may need a high frame rate in order to properly track their changes. Amplitude Threshold (dB): the highest amplitude value to be taken into account for Peak Detection. Window Type: the shape of the smoothing function to be used for the Fourier Analysis. There are four choices available at present: Blackman, Blackman-Harris, Von Hann, and Hanning. Precise specifications about them are easily found on D.S.P. bibliography. Track Length (Frames): The Peak Continuation Algorithm will "look-back" by Length frames in order to do its job better, preventing frequency trajectories from curving too much and loosing stability. However, a large value for this parameter will slow down the analysis significantly. Minimal Segment Length (Frames): once the analysis is done, the spectral data can be further "cleaned" up during post-processing. Trajectories shorter than this value are suppressed if their average SMR is below Minimal Segment SMR (see parameters 16 and 14). This might help to avoid non-relevant sudden changes while keeping a high frame rate, reducing also the number of intermittent sinusoids during synthesis. Minimal Gap Length (Frames): as parameter 11, this one is also used to clean up the data during post-processing. In this case, gaps (zero amplitude values, i.e. theoretical "silence") longer than Length frames are filled up with amplitude/frequency values obtained by linear interpolation of the adjacent active frames. This parameter prevents sudden interruptions of stable trajectories while keeping a high frame rate. SMR Threshold (dB SPL): also a post-processing parameter, the SMR Threshold is used to eliminate partials with low averages. Minimal Segment SMR (dB SPL): this parameter is used in combination with parameter 11. Short segments with SMR average below this value will be removed during post-processing. Last Peak Contribution (0 to 1): as explained in Parameter 10, the Peak Continuation Algorithm "looks-back" several number of frames to do its job better. This parameter will help to weight the contribution of the first precedent peak over the others. A zero value means that all precedent peaks (to the size of Parameter 10) are equally taken in account. SMR Contribution (0 to 1): In addition to the proximity in frequency of the peaks, the ATS Peak Continuation Algorithm may use psycho-acoustic information (the Signal-to-Mask-Ratio, or SMR) to improve the perceptual results. This parameter indicates how much the SMR information is used during tracking. For instance, a value of .5 makes the Peak Continuation Algorithm to use a 50% of SMR information and a 50% of Frequency Proximity information to decide which is the best candidate to continue a sinusoidal track. Examples The following command: atsa -b0.1 -e1 -l100 -H10000 -w2 audiofile.wav audiofile.ats Generates the ATS analysis file 'audiofile.ats' from the original 'audiofile.wav' file. It begins analysis from second 0.1 of the file and the analysis is performed for 1 second thereafter. The lowest frequency stored is 100 Hz and the highest is 10kHz. A Hamming window is used for each analysis frame. manual_src~dfsg/utility/topXO.xml0000644000000000000000000001132012262561504016240 0ustar rootroot Dan Ellis MIT Media Lab
Cambridge Massachussetts
The Utility Programs The Csound Utilities are soundfile preprocessing programs that return information on a soundfile or create some analyzed version of it for use by certain Csound generators. Though different in goals, they share a common soundfile access mechanism and are describable as a set. The Soundfile Utility programs can be invoked in two equivalent forms: csound -U utilname filenames utilname filenames In the first, the utility is invoked as part of the Csound executable, while in the second it is called as a standalone program. The second is smaller by about 200K, but the two forms are identical in function. The first is convenient in not requiring the maintenance and use of several independent programs - one program does all. When using this form, a -U flag detected in the command line will cause all subsequent flags and names to be interpreted as per the named utility; i.e. Csound generation will not occur, and the program will terminate at the end of utility processing.
Directories. Filenames are of two kinds, source soundfiles and resultant analysis files. Each has a hierarchical naming convention, influenced by the directory from which the Utility is invoked. Source soundfiles with a full pathname (begins with dot (.), slash (/), or for ThinkC includes a colon (:)), will be sought only in the directory named. Soundfiles without a path will be sought first in the current directory, then in the directory named by the SSDIR environment variable (if defined), then in the directory named by SFDIR. An unsuccessful search will return a "cannot open" error. Resultant analysis files are written into the current directory, or to the named directory if a path is included. It is tidy to keep analysis files separate from sound files, usually in a separate directory known to the SADIR variable. Analysis is conveniently run from within the SADIR directory. When an analysis file is later invoked by a Csound generator it is sought first in the current directory, then in the directory defined by SADIR.
Soundfile Formats. Csound can read and write audio files in a variety of formats. Write formats are described by Csound command flags. On reading, the format is determined from the soundfile header, and the data automatically converted to floating-point during internal processing. When Csound is installed on a host with local soundfile conventions (SUN, NeXT, Macintosh) it may conditionally include local packaging code which creates soundfiles not portable to other hosts. However, Csound on any host can always generate and read AIFF files, which is thus a portable format. Sampled sound libraries are typically AIFF, and the variable SSDIR usually points to a directory of such sounds. If defined, the SSDIR directory is in the search path during soundfile access. Note that some AIFF sampled sounds have an audio looping feature for sustained performance; the analysis programs will traverse any loop segment once only. For soundfiles without headers, an SR value may be supplied by the -r flag (or its default). If both the SR header and the command-line flag are present, the flag value will override the header. When sound is accessed by the audio Analysis programs, only a single channel is read. For stereo or quad files, the default is channel one; alternate channels may be obtained on request. &utilityanalysis; &utilityqueries; &utilityconvert; &utilityother; Credits Dan Ellis MIT Media Lab Cambridge, Massachussetts
manual_src~dfsg/utility/cs.xml0000644000000000000000000001644312262561504015607 0ustar rootroot Utilities cs cs Starts Csound with a set of options that can be controlled by environment variables, and input and output files determined by the specified filename stem. cs Description Starts Csound with a set of options that can be controlled by environment variables, and input and output files determined by the specified filename stem. Syntax cs [-OPTIONS] <name> [CSOUND OPTIONS ... ] Initialization Flags: - OPTIONS = OPTIONS is a sequence of alphabetic characters that can be used for selecting the Csound executable to be run, as well as the command line flags (see below). There is a default for the option 'r' (selects real-time output), but it can be overridden. <name> = this is the filename stem for selecting input files; it may contain a path. Files that have .csd, .orc, or .sco extension are searched, and either a CSD or an orc/sco pair that matches <name> the best are selected. MIDI files with a .mid extension are also searched, and if one that matches <name> at least as close as the CSD or orc/sco pair, it is used with the -F flag. NOTE The MIDI file is not used if any -M or -F flag is specified by the user - new in version 4.24.0) Unless there is any option (-n or -o) related to audio output, an output file name with the appropriate extension is automatically generated (based on the name of selected input files and format options). The output file is always written to the current directory. NOTE file name extensions are not case sensitive. [CSOUND OPTIONS ... ] = any number of additional options for Csound that are simply copied to the final command line to be executed. The command line that is executed is generated from four parts: Csound executable (possibly with options). This is exactly one of the following (the last one has the highest precedence): a built-in default the value of the CSOUND environment variable environment variables with a name in the format of CSOUND_x where x is an uppercase letter selected by characters of the -OPTIONS string. Thus, if the -dcba option is used, and the environment variables CSOUND_B and CSOUND_C are defined, the value of CSOUND_B will take effect. Any number of option lists, added in the following order: either some built-in defaults, or the value of the CSFLAGS environment variable if it is defined. environment variables with a name in the format of CSFLAGS_x where x is an uppercase letter selected by characters of the -OPTIONS string. Thus, if the -dcba option is used, and the environment variables CSFLAGS_A and CSFLAGS_C are defined as '-M 1 -o dac' and '-m231 -H0', respectively, the string '-m231 -H0 -M 1 -o dac' will be added. The explicit options of [CSOUND OPTIONS ... ]. Any options and file names generated from <name>. NOTE Quoted options that contain spaces are allowed. Examples Assuming the following environment variables: CSOUND = csoundfltk.exe -W CSOUND_D = csound64.exe -J CSOUND_R = csoundfltk.exe -h CSFLAGS = -d -m135 -H1 -s CSFLAGS_D = -f CSFLAGS_R = -m0 -H0 -o dac1 -M "MIDI Yoke NT: 1" -b 200 -B 6000 And a directory that contains: foo.orc piano.csd foo.sco piano.mid im.csd piano2.mid ImproSculpt2_share.csd foobar.csd The following commands will execute as shown: cs foo => csoundfltk.exe -W -d -m135 -H1 -s -o foo.wav \ foo.orc foo.sco cs foob => csoundfltk.exe -W -d -m135 -H1 -s \ -o foobar.wav foobar.csd cs -r imp -i adc => csoundfltk.exe -h -d -m135 -H1 -s -m0 -H0 \ -o dac1 -M "MIDI Yoke NT: 1" \ -b 200 -B 6000 -i adc \ ImproSculpt2_share.csd cs -d im => csound64.exe -J -d -m135 -H1 -s -f -o im.sf \ im.csd cs piano => csoundfltk.exe -W -d -m135 -H1 -s \ -F piano.mid -o piano.wav \ piano.csd cs piano2 => csoundfltk.exe -W -d -m135 -H1 -s \ -F piano2.mid -o piano2.wav \ piano.csd Credits Author: Istvan Varga Jan 2003 manual_src~dfsg/utility/het_import.xml0000644000000000000000000000260112262561504017343 0ustar rootroot Utilities het_import het_import Converts a comma-separated text file to a .het file. het_import Syntax het_import cstext_file het_file csound -U het_import cstext_file het_file Initialization cstext_file - Name of the input comma-separated text file. het_file - Name of the output .het file. The het_import utility generates a .het file usable with the adsyn generator. It can be used in combination with het_export to modify sound analysis made by the HETRO utility. Credits Author: John ffitch 1995 manual_src~dfsg/utility/pvlook.xml0000644000000000000000000004050012262561504016503 0ustar rootroot Utilities pvlook pvlook View formatted text output of STFT analysis files. pvlook Description View formatted text output of STFT analysis files created with pvanal. Syntax csound -U pvlook [flags] infilename pvlook [flags] infilename Initialization pvlook reads a file, and frequency and amplitude trajectories for each of the analysis bins, in readable text form. The file is assumed to be an STFT analysis file created by pvanal. By default, the entire file is processed. -bb n -- begin at analysis bin number n, numbered from 1. Default is 1. -eb n -- end at analysis bin number n. Defaults to the highest. -bf n -- begin at analysis frame number n, numbered from 1. Defaults to 1. -ef n -- end at analysis frame number n. Defaults to the highest. -i -- prints values as integers. Defaults to floating point. 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0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.140 1.265 2.766 3.289 3.296 3.293 3.296 3.296 3.290 3.293 3.292 3.291 3.297 3.295 3.294 3.296 3.291 3.292 3.294 3.291 3.296 3.297 3.292 3.295 3.292 3.290 3.295 3.293 3.294 3.297 3.292 3.293 3.294 3.290 3.295 3.295 3.292 3.296 3.293 3.291 3.294 3.291 3.293 3.297 3.292 3.295 3.294 3.288 3.293 3.293 3.292 3.297 3.294 3.292 3.295 3.290 3.292 3.295 3.292 3.295 3.295 3.290 3.294 3.292 3.292 3.297 3.293 3.293 3.295 3.290 3.292 3.293 3.290 3.296 3.296 3.292 3.295 3.291 3.290 3.294 3.291 3.294 3.296 3.291 3.293 3.293 3.290 3.295 3.294 3.293 3.296 3.291 3.291 3.293 3.290 3.294 3.296 3.292 3.295 3.293 3.288 3.293 3.292 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Credits Author: Richard Karpen Seattle, Wash 1993 (New in Csound version 3.57) manual_src~dfsg/utility/dnoise.xml0000644000000000000000000001551412262561504016461 0ustar rootroot Utilities dnoise dnoise Reduces noise in a file. dnoise Description This is a noise reduction scheme using frequency-domain noise-gating. Syntax dnoise [flags] -i noise_ref_file -o output_soundfile input_soundfile Initialization Dnoise specific flags: (no flag) input soundfile to be denoised -i fname input reference noise soundfile -o fname output soundfile -N fnum # of bandpass filters (default: 1024) -w fovlp filter overlap factor: {0,1,(2),3} DON'T USE -w AND -M -M awlen analysis window length (default: N-1 unless -w is specified) -L swlen synthesis window length (default: M) -D dfac decimation factor (default: M/8) -b btim begin time in noise reference soundfile (default: 0) -B smpst starting sample in noise reference soundfile (default: 0) -e etim end time in noise reference soundfile (default: end of file) -E smpend final sample in noise reference soundfile (default: end of file) -t thr threshold above noise reference in dB (default: 30) -S gfact sharpness of noise-gate turnoff, range: 1 to 5 (default: 1) -n numfrm number of FFT frames to average over (default: 5) -m mingain minimum gain of noise-gate when off in dB (default: -40) Soundfile format options: -A AIFF format output -W WAV format output -J IRCAM format output -h skip soundfile header (not valid for AIFF/WAV output) -8 8-bit unsigned char sound samples -c 8-bit signed_char sound samples -a alaw sound samples -u ulaw sound samples -s short_int sound samples -l long_int sound samples -f float sound samples. Floats also supported for WAV files. (New in Csound 3.47.) Additional options: -R verbose - print status info -H [N] print a heartbeat character at each soundfile write. -- fname output to log file fname -V verbose - print status info Note DNOISE also looks at the environment variable SFOUTYP to determine soundfile output format. The -i flag is used for a reference noise file (normally created from a short section of the denoised file, where only noise is audible). The input soundfile to be denoised can be given anywhere on the command line, without a flag. Performance This is a noise reduction scheme using frequency-domain noise-gating. This should work best in the case of high signal-to-noise with hiss-type noise. The algorithm is that suggested by Moorer & Berger in Linear-Phase Bandsplitting: Theory and Applications presented at the 76th Convention 1984 October 8-11 New York of the Audio Engineering Society (preprint #2132) except that it uses the Weighted Overlap-Add formulation for short-time Fourier analysis-synthesis in place of the recursive formulation suggested by Moorer & Berger. The gain in each frequency bin is computed independently according to gain = g0 + (1-g0) * [avg / (avg + th*th*nref)] ˆ sh where avg and nref are the mean squared signal and noise respectively for the bin in question. (This is slightly different than in Moorer & Berger.) The critical parameters th and g0 are specified in dB and internally converted to decimal values. The nref values are computed at the start of the program on the basis of a noise_soundfile (specified in the command line) which contains noise without signal. The avg values are computed over a rectangular window of m FFT frames looking both ahead and behind the current time. This corresponds to a temporal extent of m*D/R (which is typically (m*N/8)/R). The default settings of N, M, and D should be appropriate for most uses. A higher sample rate than 16 Khz might indicate a higher N. Credits Author: Mark Dolson August 26, 1989 Author: &namejohn; December 30, 2000 Updated by &namerasmus; on March 11, 2002. manual_src~dfsg/utility/pv_export.xml0000644000000000000000000000260412262561504017222 0ustar rootroot Utilities pv_export pv_export Converts a .pvx file to a comma separated text file. pv_export Syntax pv_export pv_file cstext_file csound -U pv_export pv_file cstext_file Initialization pv_file - Name of the input .pvx file. cstext_file - Name of the output comma-separated text file. The pv_export utility generates a comma-separated text file for manual editing of a .pvx file produced by the PVANAL utility. It can be used in combination with pv_import to produce data for the pvoc generator. Credits Author: John ffitch 1995 manual_src~dfsg/utility/mkdb.xml0000644000000000000000000000150312262561504016106 0ustar rootroot Utilities mkdb mkdb Outputs a catalog of opcodes and plugin modules. mkdb Description Outputs a catalog of opcodes and plugin modules, sorted by opcode name or by module. Syntax mkdb [-m] [base_directory] Credits Author: &namejohn; Aug 2011 New in version 5.14 manual_src~dfsg/utility/extractor.xml0000644000000000000000000000664212262561504017215 0ustar rootroot Utilities extractor extractor Extract a section of audio from an audio file. extractor Description Extract a section of audio, by time or sample, from an existing sound file. Syntax extractor [OPTIONS ... ] infile Initialization Flags: -S integer = Start the extract at the given sample number. -Z integer = End the extract at the given sample number. - Q integer = Extract given number of samples. -T fpnum = Start the extract at the given time in seconds. -E fpnum = End the extract at the given time in seconds. -D fpnum = Extract given time in seconds. -v = Verbose mode. -R = Continually rewrite the header while writing soundfile (WAV/AIFF). -H integer = Show a "heart-beat" to indicate progress, in style 1, 2 or 3. -N = Alert call (usually ringing the bell) when finished. -v = Verbose mode. -o fname = output file name (default: test.wav) Examples The default values are extractor -S 0 -Z end-of-file -otest For example extractor -S 10234 -D 2.13 in.aiff -o out.wav This creates a new sound file taken from sample 10234 and lasting 2.13 seconds. Credits Author: &namejohn; 1994 manual_src~dfsg/utility/analysisXO.xml0000644000000000000000000000316512262561504017271 0ustar rootroot
Analysis File Generation (<!--ATSA,--> CVANAL, HETRO, LPANAL, PVANAL) The following utilities exist for Soundfile analysis: CVANAL: Impulse Response Fourier Analysis for convolve operator. HETRO: Heterodyne analysis for the Csound adsyn generator. LPANAL: Linear predicitive coding analysis for the Csound Linear Predictive Coding (LPC) Resynthesis opcodes. PVANAL: Phase vocoder analysis for the Csound pvoc generator. &utilitycvanal; &utilityhetro; &utilitylpanal; &utilitypvanal;
manual_src~dfsg/utility/top.xml0000644000000000000000000001172412262561504016001 0ustar rootroot Dan Ellis MIT Media Lab
Cambridge Massachussetts
The Utility Programs The Csound Utilities are soundfile preprocessing programs that return information on a soundfile or create some analyzed version of it for use by certain Csound generators. Though different in goals, they share a common soundfile access mechanism and are describable as a set. The Soundfile Utility programs can be invoked in two equivalent forms: csound -U utilname filenames utilname filenames In the first, the utility is invoked as part of the Csound executable, while in the second it is called as a standalone program. The second is smaller by about 200K, but the two forms are identical in function. The first is convenient in not requiring the maintenance and use of several independent programs - one program does all. When using this form, a -U flag detected in the command line will cause all subsequent flags and names to be interpreted as per the named utility; i.e. Csound generation will not occur, and the program will terminate at the end of utility processing.
Directories. Filenames are of two kinds, source soundfiles and resultant analysis files. Each has a hierarchical naming convention, influenced by the directory from which the Utility is invoked. Source soundfiles with a full pathname (begins with dot (.), slash (/), or for ThinkC includes a colon (:)), will be sought only in the directory named. Soundfiles without a path will be sought first in the current directory, then in the directory named by the SSDIR environment variable (if defined), then in the directory named by SFDIR. An unsuccessful search will return a "cannot open" error. Resultant analysis files are written into the current directory, or to the named directory if a path is included. It is tidy to keep analysis files separate from sound files, usually in a separate directory known to the SADIR variable. Analysis is conveniently run from within the SADIR directory. When an analysis file is later invoked by a Csound generator it is sought first in the current directory, then in the directory defined by SADIR.
Soundfile Formats. Csound can read and write audio files in a variety of formats. Write formats are described by Csound command flags. On reading, the format is determined from the soundfile header, and the data automatically converted to floating-point during internal processing. When Csound is installed on a host with local soundfile conventions (SUN, NeXT, Macintosh) it may conditionally include local packaging code which creates soundfiles not portable to other hosts. However, Csound on any host can always generate and read AIFF files, which is thus a portable format. Sampled sound libraries are typically AIFF, and the variable SSDIR usually points to a directory of such sounds. If defined, the SSDIR directory is in the search path during soundfile access. Note that some AIFF sampled sounds have an audio looping feature for sustained performance; the analysis programs will traverse any loop segment once only. For soundfiles without headers, an SR value may be supplied by the -r flag (or its default). If both the SR header and the command-line flag are present, the flag value will override the header. When sound is accessed by the audio Analysis programs, only a single channel is read. For stereo or quad files, the default is channel one; alternate channels may be obtained on request. &utilityanalysis; &utilityqueries; &utilityconvert; &utilityother; Credits Dan Ellis MIT Media Lab Cambridge, Massachussetts
manual_src~dfsg/utility/csb64enc.xml0000644000000000000000000000415512262561504016606 0ustar rootroot Utilities csb64enc csb64enc Converts a binary file to a Base64 encoded text file. csb64enc Description The csb64enc utility generates a Base64 encoded text file from a binary file, such as a standard MIDI file (.mid) or any type of audio file. It is useful to convert a file in the format accepted by the <CsFileB> section of a csd file, to include the file within it. Syntax csb64enc [OPTIONS ... ] infile1 [ infile2 [ ... ]] Initialization Flags: - w n = set line width of the output file to n (default: 72) - o fname = output file name (default: stdout) Examples csb64enc -w 78 -o file.txt file.mid This command produces a Base64 encoded text file from the standard MIDI file file.mid. This file can now be pasted within a csd file's <CsFileB> section. See also makecsd Credits Author: Istvan Varga Jan 2003 manual_src~dfsg/utility/other.xml0000644000000000000000000000370212262561504016315 0ustar rootroot
Other Csound Utilities (CS, CSB64ENC, ENVEXT, EXTRACTOR, MAKECSD, MIXER, SCALE, MKDB) The following miscellaneous utilities are available: CS: Starts Csound with a set of options that can be controlled by environment variables, and input and output files determined by the specified filename stem. CSB64ENC: Converts a binary file to a Base64 enconded text file. ENVEXT: Extract the envelope of a file to a text list. EXTRACTOR: Extract a section of audio from an audio file. MAKECSD: Creates a CSD file from the specified input files. MIXER: Mixes together a number of soundfiles. SCALE: Scale the amplitude of a sound file. MKDB: Creates a plugin opcode catalog. &utilitycs; &utilitycsb64enc; &utilityenvext; &utilityextractor; &utilitymakecsd; &utilitymixer; &utilityscale; &utilitymkdb;
manual_src~dfsg/utility/convert.xml0000644000000000000000000000463312262561504016660 0ustar rootroot
File Conversion (<!--DNOISE, -->HET_IMPORT, HET_EXPORT, PVLOOK, PV_EXPORT, PV_IMPORT, SDIF2AD, SRCONV) The following utilities exist for file conversion: HET_EXPORT: Exports a .het (produced by HETRO) to a comma separated text file. HET_IMPORT: Generates a .het (in the format produced by HETRO) from a comma separated text file for usage with the adsyn generator. PVLOOK: View formatted text output of STFT analysis files. PV_EXPORT: Converts a file generated by PVANAL to a text file. PV_IMPORT: Converts a text file (in the format generated by PV_EXPORT) to a PVANAL format file to be used with the pvoc opcode. SDIF2AD: Converts SDIF files to files usable by adsynt. SRCONV: Converts the sample rate of an audio file. &utilitydnoise; &utilityhet_export; &utilityhet_import; &utilitypvlook; &utilitypv_export; &utilitypv_import; &utilitysdif2ad; &utilitysrconv;
manual_src~dfsg/utility/makecsd.xml0000644000000000000000000001174112262561504016605 0ustar rootroot Utilities makecsd makecsd Creates a CSD file from the specified input files. makecsd Description Creates a CSD file from the specified input files. The first input file that has a .orc extension (case is not significant) is put to the <CsInstruments> section, and the first input file that has a .sco extension becomes <CsScore>. Any remaining files are Base64 encoded and added as <CsFileB> tags. An empty <CsOptions> section is always added. Some text filtering is performed on the orchestra and score file: newlines are converted to the native format of the system on which makecsd is being run. blank lines are removed from the beginning and end of files. any trailing whitespace is removed from the end of lines. optionally, tabs can be expanded to spaces with an user specified tabstop size. optionally, a MIDI file can be included. optionally, a Licence can be specified as either a file or a common one. if required a score processor can be specified for the <CsScore> section. Syntax makecsd [OPTIONS ... ] infile1 [ infile2 [ ... ]] Initialization Flags: - t n = expand tabs to spaces using tabstop size n (default: disabled). This applies only to the orchestra and score file. - w n = set Base64 line width to n (default: 72). Note: the orchestra and score are not wrapped. - o fname = output file name (default: stdout) - m fname = specify a MIDI file to include (default: none) - b progname = specify the program to process the score (default: none) - L fname = file name containing the licence text (default: none) - l integer = specify a standard licence (default: none). The ones available are: 0: All rights reserved 1: CC BY-NC-ND 2: CC BY-NC-SA 3: CC BY-NC 4: CC BY-ND 5: CC BY-SA 6: CC BY 7: Licenced under BSD Examples makecsd -t 6 -w 78 -o file.csd file.mid file.orc file.sco sample.aif This creates a CSD from file.orc and file.sco (tabs are expanded to spaces assuming a tabstop size of 6 characters), and file.mid and sample.aif are added as <CsFileB> tags containing Base64 encoded data with a line width of 78 characters. The output file is file.csd. Credits Author: Istvan Varga Jan 2003 Author: &namejohn; Feb 2011 Options for MIDI, score processing and licence new in version 5.14 manual_src~dfsg/utility/lpanal.xml0000644000000000000000000001137712262561504016452 0ustar rootroot Utilities lpanal lpanal Performs both linear predictive and pitch-tracking analysis on a soundfile. lpanal Description Linear predictive analysis for the Csound Linear Predictive Coding (LPC) Resynthesis opcodes. Syntax csound -U lpanal [flags] infilename outfilename lpanal [flags] infilename outfilename Initialization lpanal performs both lpc and pitch-tracking analysis on a soundfile to produce a time-ordered sequence of frames of control information suitable for Csound resynthesis. Analysis is conditioned by the control flags below. A space is optional between the flag and its value. -a -- [alternate storage] asks lpanal to write a file with filter poles values rather than the usual filter coefficient files. When lpread / lpreson are used with pole files, automatic stabilization is performed and the filter should not get wild. (This is the default in the Windows GUI) - Changed by Marc Resibois. -s srate -- sampling rate of the audio input file. This will over-ride the srate of the soundfile header, which otherwise applies. If neither is present, the default is 10000. -c channel -- channel number sought. The default is 1. -b begin -- beginning time (in seconds) of the audio segment to be analyzed. The default is 0.0 -d duration -- duration (in seconds) of the audio segment to be analyzed. The default of 0.0 means to the end of the file. -p npoles -- number of poles for analysis. The default is 34, the maximum 50. -h hopsize -- hop size (in samples) between frames of analysis. This determines the number of frames per second (srate / hopsize) in the output control file. The analysis framesize is hopsize * 2 samples. The default is 200, the maximum 500. -C string -- text for the comments field of the lpfile header. The default is the null string. -P mincps -- lowest frequency (in Hz) of pitch tracking. -P0 means no pitch tracking. -Q maxcps -- highest frequency (in Hz) of pitch tracking. The narrower the pitch range, the more accurate the pitch estimate. The defaults are -P70, -Q200. -v verbosity -- level of terminal information during analysis. 0 = none 1 = verbose 2 = debug The default is 0. -X -- write machine independent format analysis file. Examples lpanal -a -p26 -d2.5 -P100 -Q400 audiofile.test lpfil22 will analyze the first 2.5 seconds of file "audiofile.test", producing srate/200 frames per second, each containing 26-pole filter coefficients and a pitch estimate between 100 and 400 Hertz. Stabilized (-a) output will be placed in "lpfil22" in the current directory. File Format Output is a file comprised of an identifiable header plus a set of frames of floating point analysis data. Each frame contains four values of pitch and gain information, followed by npoles filter coefficients. The file is readable by Csound's lpread. lpanal is an extensive modification of Paul Lanksy's lpc analysis programs. manual_src~dfsg/utility/cvanal.xml0000644000000000000000000000677112262561504016451 0ustar rootroot Utilities cvanal cvanal Converts a soundfile into a single Fourier transform frame. cvanal Description Impulse Response Fourier Analysis for convolve operator Syntax csound -U cvanal [flags] infilename outfilename cvanal [flags] infilename outfilename Initialization cvanal -- converts a soundfile into a single Fourier transform frame. The output file can be used by the convolve operator to perform Fast Convolution between an input signal and the original impulse response. Analysis is conditioned by the flags below. A space is optional between the flag and its argument. -s rate -- sampling rate of the audio input file. This will over-ride the srate of the soundfile header, which otherwise applies. If neither is present, the default is 10000. -c channel -- channel number sought. If omitted, the default is to process all channels. If a value is given, only the selected channel will be processed. -b begin -- beginning time (in seconds) of the audio segment to be analyzed. The default is 0.0 -d duration -- duration (in seconds) of the audio segment to be analyzed. The default of 0.0 means to the end of the file. -X -- write machine independent format analysis file. Examples cvanal asound cvfile will analyze the soundfile "asound" to produce the file "cvfile" for the use with convolve. To use data that is not already contained in a soundfile, a soundfile converter that accepts text files may be used to create a standard audio file, e.g., the .DAT format for SOX. This is useful for implementing FIR filters. Files The output file has a special convolve header, containing details of the source audio file. The analysis data is stored as float, in rectangular (real/imaginary) form. Note The analysis file is not system independent! Ensure that the original impulse recording/data is retained. If/when required, the analysis file can be recreated. Credits Author: Greg Sullivan Based on algorithm given in Elements Of Computer Music, by F. Richard Moore. manual_src~dfsg/utility/srconv.xml0000644000000000000000000001116712262561504016512 0ustar rootroot Utilities srconv srconv Converts the sample rate of an audio file. srconv Description Converts the sample rate of an audio file at sample rate Rin to a sample rate of Rout. Optionally the ratio (Rin / Rout) may be linearly time-varying according to a set of (time, ratio) pairs in an auxiliary file. Syntax srconv [flags] infile Initialization Flags: -P num = pitch transposition ratio (srate / r) [don't specify both P and r] -P num = pitch transposition ratio (srate / r) [don't specify both P and r] -Q num =quality factor (1, 2, 3, or 4: default = 2) -i filnam = auxiliary breakpoints file (no breakpoint by default. i.e. No ratio change) -r num = output sample rate (must be specified) -o fnam = sound output filename -A = create an AIFF format output soundfile -J = create an IRCAM format output soundfile -W = create a WAV format output soundfile -h = no header on output soundfile -c = 8-bit signed_char sound samples -a = alaw sound samples -8 = 8-bit unsigned_char sound samples -u = ulaw sound samples -s = short_int sound samples -l = long_int sound samples -f = float sound samples -r N = orchestra srate override -K = Do not generate PEAK chunks -R = continually rewrite header while writing soundfile (WAV/AIFF) -H# = print a heartbeat style 1, 2 or 3 at each soundfile write -N = notify (ring the bell) when score or miditrack is done -- fnam = log output to file This program performs arbitrary sample-rate conversion with high fidelity. The method is to step through the input at the desired sampling increment, and to compute the output points as appropriately weighted averages of the surrounding input points. There are two cases to consider: sample rates are in a small-integer ratio - weights are obtained from table. sample rates are in a large-integer ratio - weights are linearly interpolated from table. Calculate increment: if decimating, then window is impulse response of low-pass filter with cutoff frequency at half of output sample rate; if interpolating, then window is impulse response of lowpass filter with cutoff frequency at half of input sample rate. Credits Author: Mark Dolson August 26, 1989 Author: &namejohn; December 30, 2000 manual_src~dfsg/utility/pvanal.xml0000644000000000000000000001420612262561504016456 0ustar rootroot Utilities pvanal pvanal Converts a soundfile into a series of short-time Fourier transform frames. pvanal Description Fourier analysis for the Csound pvoc generator Syntax csound -U pvanal [flags] infilename outfilename pvanal [flags] infilename outfilename Pvanal extension to create a PVOC-EX file. The standard Csound utility program pvanal has been extended to enable a PVOC-EX format file to be created, using the existing interface. To create a PVOC-EX file, the file name must be given the required extension, .pvx, e.g test.pvx. The requirement for the FFT size to be a power of two is here relaxed, and any positive value is accepted; odd numbers are rounded up internally. However, power-of-two sizes are still to be preferred for all normal applications. The channel select flags are ignored, and all source channels will be analysed and written to the output file, up to a compiler-set limit of eight channels. The analysis window size (iwinsize) is set internally to double the FFT size. Initialization pvanal converts a soundfile into a series of short-time Fourier transform (STFT) frames at regular timepoints (a frequency-domain representation). The output file can be used by pvoc to generate audio fragments based on the original sample, with timescales and pitches arbitrarily and dynamically modified. Analysis is conditioned by the flags below. A space is optional between the flag and its argument. -s srate -- sampling rate of the audio input file. This will over-ride the srate of the soundfile header, which otherwise applies. If neither is present, the default is 10000. -c channel -- channel number sought. The default is 1. -b begin -- beginning time (in seconds) of the audio segment to be analyzed. The default is 0.0 -d duration -- duration (in seconds) of the audio segment to be analyzed. The default of 0.0 means to the end of the file. -n frmsiz -- STFT frame size, the number of samples in each Fourier analysis frame. Must be a power of two, in the range 16 to 16384. For clean results, a frame must be larger than the longest pitch period of the sample. However, very long frames result in temporal "smearing" or reverberation. The bandwidth of each STFT bin is determined by sampling rate / frame size. The default framesize is the smallest power of two that corresponds to more than 20 milliseconds of the source (e.g. 256 points at 10 kHz sampling, giving a 25.6 ms frame). -w windfact -- Window overlap factor. This controls the number of Fourier transform frames per second. Csound's pvoc will interpolate between frames, but too few frames will generate audible distortion; too many frames will result in a huge analysis file. A good compromise for windfact is 4, meaning that each input point occurs in 4 output windows, or conversely that the offset between successive STFT frames is framesize/4. The default value is 4. Do not use this flag with -h. -h hopsize -- STFT frame offset. Converse of above, specifying the increment in samples between successive frames of analysis (see also lpanal). Do not use with -w. -H -- Use a Hamming window instead of the default von Hann window. -K -- Use a Kaiser window instead of the default von Hann window. The Kaiser parameter default is 6.8, but can be set with the -B option. -B beta -- Set the beta parameter for any Kaiser window used to the floating point value beta. Examples pvanal asound pvfile will analyze the soundfile "asound" using the default frmsiz and windfact to produce the file "pvfile" suitable for use with pvoc. Files The output file has a special pvoc header containing details of the source audio file, the analysis frame rate and overlap. Frames of analysis data are stored as float, with the magnitude and frequency (in Hz) for the first N/2 + 1 Fourier bins of each frame in turn. Frequency encodes the phase increment in such a way that for strong harmonics it gives a good indication of the true frequency. For low amplitude or rapidly moving harmonics it is less meaningful. Diagnostics Prints total number of frames, and frames completed every 20th frame. Credits Author: Dan Ellis MIT Media Lab Cambridge, Massachussetts 1990 manual_src~dfsg/utility/queries.xml0000644000000000000000000000061012262561504016644 0ustar rootroot
File Queries (SNDINFO) The following utilities exist for Soundfile quuery: SNDINFO: Displays information about a soundfile. &utilitysndinfo;
manual_src~dfsg/utility/het_export.xml0000644000000000000000000000262112262561504017354 0ustar rootroot Utilities het_export het_export Converts a .het file to a comma separated text file. het_export Syntax het_export het_file cstext_file csound -U het_export het_file cstext_file Initialization het_file - Name of the input .het file. cstext_file - Name of the output comma-separated text file. The het_export utility generates a comma-separated text file for manual editing of a .het file produced by the HETRO utility. It can be used in combination with het_import to produce data for the adsyn generator. Credits Author: John ffitch 1995 manual_src~dfsg/utility/mixer.xml0000644000000000000000000001260312262561504016320 0ustar rootroot Utilities mixer mixer Mixes together a number of soundfiles. mixer Description Mixes together a number of soundfiles, starting at different times and with individual channel selection from the input files. Syntax mixer [OPTIONS ... ] infile [[OPTIONS... ] infile] ... Initialization Flags: -A = Generate an AIFF output file. -W = Generate an WAV output file. -h = Generate an output file with no header. -c = Generate 8-bit signed_char sound samples. -a = Generate alaw sound samples. -u = Generate ulaw sound samples. -s = Generate short integer sound samples. -l = Generate long (32 bit) integer sound samples. -f = Generate floating point samples. -F arg = Specifies the gain to be applied to the following input file. If arg is a floating point number that gain is applied uniformly to the input. Alternatively it could be a file name which specifies a breakpoint file for varying the gain for different periods. -S integer = Indicate at which sample to start to mix in the next input file. -T fpnum = Indicate at which time (in seconds) to start to mix in the next input file. -1 = Mix in channel 1 from next sound file. -2 = Mix in channel 2 from next sound file. -3 = Mix in channel 3 from next sound file. -4 = Mix in channel 4 from next sound file. -^ intx inty = Mix in channel x from next sound file as channel y in the output. -v = Verbose mode. -R = Continually rewrite the header while writing soundfile (WAV/AIFF). -H integer = Show a "heart-beat" to indicate progress, in style 1, 2 or 3. -N = Alert call (usually ringing the bell) when finished. -o fname = output file name (default: test.wav) Examples The default values are mixer -s -otest -F 1.0 -S 0 For example mixer -F 0.96 in1.wav -S 300 -2 in2.aiff -S 300 -^4 1 in3.wav -o out.wav This creates a new sound file with a constant gain of 0.96 from in1.wav with the second channel of in2.aiff mixed in after 300 samples and channel 4 of in3.wav outpout as channel 1 after 300 samples. Credits Author: &namejohn; 1994 manual_src~dfsg/utility/analysis.xml0000644000000000000000000000313412262561504017016 0ustar rootroot
Analysis File Generation (ATSA, CVANAL, HETRO, LPANAL, PVANAL) The following utilities exist for Soundfile analysis: ATSA: ATS analysis for use with the Csound ATS Resynthesis opcodes. CVANAL: Impulse Response Fourier Analysis for convolve operator. HETRO: Heterodyne analysis for the Csound adsyn generator. LPANAL: Linear predicitive coding analysis for the Csound Linear Predictive Coding (LPC) Resynthesis opcodes. PVANAL: Phase vocoder analysis for the Csound pvoc generator. &utilityatsa; &utilitycvanal; &utilityhetro; &utilitylpanal; &utilitypvanal;
manual_src~dfsg/manual.xml0000644000000000000000000025742712262561504014765 0ustar rootroot ]> The Canonical Csound Reference Manual Version &csoundversion; Barry Vercoe MIT Media Lab et. al. &csoundversion; &pface; Overview &oviewintroduction; &oviewrecentdev; &command; &oviewusing; &oviewconfiguring; &orch; &score; &oviewfrontends; &oviewbuilding; &oviewlinks; Opcodes Overview &siggen; &sigio; &sigmod; &control; &table; &math; &pitch; &midi; &spectral; &strings; &vectorial; &zak; &plugin; &oscnetwork; &mixer; &signalflowgraph; &jacko; &lua; &py; &imageopcodes; &stkopcodes; &miscopcodes; Reference &opcodes; &scoregens; &deprecated; &utility; &cscore; &csbeats; &misc; manual_src~dfsg/xsl/0000755000000000000000000000000012262561504013553 5ustar rootrootmanual_src~dfsg/xsl/customXHTMLOneChunk.xsl.in0000644000000000000000000000225512262561504020476 0ustar rootroot appendix toc,title article/appendix nop article toc,title book toc,title chapter title part toc,title preface toc,title qandadiv toc qandaset toc reference toc,title sect1 toc sect2 toc sect3 toc sect4 toc sect5 toc section toc set toc,title manual_src~dfsg/xsl/customHTMLHelp.xsl.in0000644000000000000000000000244212262561504017522 0ustar rootroot appendix toc,title article/appendix nop article toc,title book toc,title chapter title part toc,title preface toc,title qandadiv toc qandaset toc reference toc,title sect1 toc sect2 toc sect3 toc sect4 toc sect5 toc section toc set toc,title manual_src~dfsg/xsl/customXHTMLchunk.xsl.in0000644000000000000000000000243312262561504020072 0ustar rootroot appendix toc,title article/appendix nop article toc,title book toc,title chapter title part toc,title preface toc,title qandadiv toc qandaset toc reference toc,title sect1 toc sect2 toc sect3 toc sect4 toc sect5 toc section toc set toc,title manual_src~dfsg/xsl/customPDF.xsl.in0000644000000000000000000001053612262561504016561 0ustar rootroot appendix toc,title article/appendix nop article toc,title book toc,title chapter title part toc,title preface toc,title qandadiv toc qandaset toc reference toc,title sect1 toc sect2 toc sect3 toc sect4 toc sect5 toc section toc set toc,title 8pt manual_src~dfsg/midi/0000755000000000000000000000000012262561504013667 5ustar rootrootmanual_src~dfsg/midi/virtual.xml0000644000000000000000000001703112262561504016101 0ustar rootroot
Virtual MIDI Keyboard [Virtual MIDI keyboard] Virtual MIDI keyboard. The virtual MIDI keyboard module (activated using -+rtmidi=virtual on the command line flags) provides a way of sending realtime MIDI information to Csound without the need of a MIDI device. It can send note information, control changes, bank and program changes on a specified channel. The MIDI information from the virtual keyboard is processed by Csound in exactly the same way as MIDI information that comes from the other MIDI drivers, so if your Csound orchestra is designed to work with hardware MIDI devices, this will also work. For the device flag (-M), the virtual keyboard uses this to take in the name of a keyboard mapping files. Like all MIDI drivers, a device must be given to activate the driver. If you would like to just use the default settings of the keyboard, simply passing in 0 (i.e. -M0) and the virtual keyboard will use its default settings. If instead of the 0 a name of a file is given, the keyboard will attempt to load the file as a keyboard mapping. If the file could not be opened or read correctly, the default settings will be used. Keyboard Mapping files allow the user to customize the name and number of banks as well as the name and number of programs per bank. The following example keyboard mapping (named keyboard.map) has inline comments on the file format. This file is also available with the Csound source distribution in the InOut/virtual_keyboard folder. # Custom Keyboard Map for Virtual Keyboard # Steven Yi # # USAGE # # When using the Virtual Keyboard, you can supply a filename for a mapping # of banks and programs via the -M flag, for example: # # csound -+rtmidi=virtual -Mkeyboard.map my_project.csd # # INFORMATION ON THE FORMAT # # -lines that start with '#' are comments # -lines that have [] start new bank definitions, # the contents are bankNum=bankName, with bankNum=[1,16384] # -lines following bank statements are program definitions # in the format programNum=programName, with programNum=[1,128] # -bankNumbers and programNumbers are defined in this file # starting with 1, but are converted to midi values (starting # with 0) when read # # NOTES # # -if an invalid bank definition is found, all program # defintions that follow will be ignored until a new # valid bank definition is found # -if a valid bank is defined by no valid programs found # for that bank, it will default to General MIDI program # definitions # -if an invalid program definition is found, it will be # ignored [1=My Bank] 1=My Test Patch 1 2=My Test Patch 2 30=My Test Patch 30 [2=My Bank2] 1=My Test Patch 1(bank2) 2=My Test Patch 2(bank2) 30=My Test Patch 30(bank3) The ten sliders up top are by default set to MIDI Controller number 1-10 though they can be changed to whatever one wishes to use. The controller numbers and values of each slider are set per channel, so one may use different settings and values for each channel. By default there are 128 banks and for each bank 128 patches defaulting to General Midi names. The MIDI bank standard uses 14-bit resolution to support 16384 possible banks, but the bank numbers by default are 0-127. To use values higher than 127, one should use a custom keyboard map and set the desired bank number value for the bank name. The virtual keyboard will correctly transmit the bank number as MSB and LSB with controller numbers 0 and 32. Beyond the input available from interacting with the GUI via mouse, one may also trigger off MIDI notes by using the ASCII keyboard when the virtual keyboard window is focused. The layout is done much like a tracker and offers two octaves and a major third to trigger, starting from Middle-C (MIDI note 60). The ASCII keyboard MIDI note values are given in the following table. ASCII Keyboard MIDI Note Values Keyboard Key MIDI Value z 60 s 61 x 62 d 63 c 64 v 65 g 66 b 67 h 68 n 69 j 70 m 71 q 72 2 73 w 74 3 75 e 76 r 77 5 78 t 79 6 80 y 81 7 82 u 83 i 84 9 85 o 86 0 87 p 88
Here's an example of usage of the virtual MIDI keyboard. It uses the file virtual.csd.
manual_src~dfsg/midi/generic.xml0000644000000000000000000000044112262561504016024 0ustar rootroot
Generic Input and Output Opcodes for generic MIDI input and output are midiin and midiout.
manual_src~dfsg/midi/extender.xml0000644000000000000000000000063412262561504016232 0ustar rootroot
Event Extenders Opcodes that let one extend the duration of an event are: release xtratim
manual_src~dfsg/midi/topXO.xml0000644000000000000000000000701012262561504015460 0ustar rootroot Real-time MIDI Support Csound supports realtime MIDI input and output, as well as input from MIDI files. Realtime MIDI input is activated using the -M (or --midi-device=DEVICE) command line flag. You must specify the device number or name after the -M. For example to use device number 2, you would use something like: csound -M2 myrtmidi.csd You can find out the available alsa devices by using: cat /proc/asound/cards You must then use something like: csound -+rtmidi=alsa -M hw:3 myrtmidi.csd Realtime MIDI output is activated using -Q, using device number or names as shown above. You can also load a MIDI file using the -F or --midifile=FILE command line flag. The MIDI file is read in realtime, and behaves as if it was being performed or recieved in realtime. So the csound program is not aware if MIDI input comes from a MIDI file or directly from a MIDI interface. Once realtime MIDI input and/or output has been activated, opcodes like MIDI Input and MIDI Output will have effect. When MIDI input is enabled (with -M or -F), each incoming noteon message will generate a note event for an instrument which has the same number as the channel of the event (see massign and pgmassign to change this behavior). This means that MIDI controlled instruments are polyphonic by default, since each note will generate a new instance of the instrument. See the MIDI/Score Interoperability opcodes for information on designing instruments which can be used from the score or driven by MIDI. &midiinput; &midioutput; &midigeneric; &midiconvert; &midiextender; &midionoff; &midiinterop; &midirealtime; &midisliderbk; manual_src~dfsg/midi/sliderbk.xml0000644000000000000000000000470612262561504016217 0ustar rootroot
Slider Banks Opcodes for slider banks of MIDI controls are: slider8 slider8f slider16 slider16f slider32 slider32f slider64 slider64f s16b14 s32b14 sliderKawai Opcodes for storing slider banks of MIDI controls to tables are: slider8table slider8tablef slider16table slider16tablef slider32table slider32tablef slider64table slider64tablef
manual_src~dfsg/midi/output.xml0000644000000000000000000000321312262561504015750 0ustar rootroot
MIDI Message Output Opcodes that produce MIDI output are: mdelay nrpn outiat outic outic14 outipat outipb outipc outkat outkc outkc14 outkpat outkpb outkpc
manual_src~dfsg/midi/interop.xml0000644000000000000000000000731612262561504016100 0ustar rootroot
MIDI/Score Interoperability opcodes The following opcodes can be used to design instruments that work interchangably for real-time MIDI and score events: midichannelaftertouch midichn midicontrolchange mididefault midinoteoff midinoteoncps midinoteonkey midinoteonoct midinoteonpch midipitchbend midipolyaftertouch midiprogramchange. Adapting a score-activated Csound instrument. To adapt an ordinary Csound instrument designed for score activation for score/MIDI interoperability: Change all linen, linseg, and expseg opcodes to linenr, linsegr, and expsegr, respectively, except for a de-clicking or damping envelope. This will not materially change score-driven performance. Add the following lines at the beginning of the instrument definition: ; Ensures that a MIDI-activated instrument ; will have a positive p3 field. mididefault 60, p3 ; Puts MIDI key translated to cycles per ; second into p4, and MIDI velocity into p5 midinoteoncps p4, p5 Obviously, midinoteoncps could be changed to midinoteonoct or any of the other options, and the choice of p-fields is arbitrary. MIDI Realtime Input/Ouput command line options New MIDI I/O flags in Csound 5.02, can replace most uses of these MIDI interop opcodes, and make usage easier.
manual_src~dfsg/midi/top.xml0000644000000000000000000001102712262561504015214 0ustar rootroot Real-time MIDI Support Csound supports realtime MIDI input and output, as well as input from MIDI files. Realtime MIDI input is activated using the -M (or --midi-device=DEVICE) command line flag. You must specify the device number or name after the -M. For example to use device number 2, you would use something like: csound -M2 myrtmidi.csd You can find out the available devices by using an out of range device: csound -M99 myrtmidi.csd This will only work if the MIDI module can be accessed by device number. For alsa, you must first find the device name using: cat /proc/asound/cards You must then use something like: csound -+rtmidi=alsa -M hw:3 myrtmidi.csd Realtime MIDI output is activated using -Q, using device number or names as shown above. You can also load a MIDI file using the -F or --midifile=FILE command line flag. The MIDI file is read in realtime, and behaves as if it was being performed or recieved in realtime. So the csound program is not aware if MIDI input comes from a MIDI file or directly from a MIDI interface. Once realtime MIDI input and/or output has been activated, opcodes like MIDI Input and MIDI Output will have effect. When MIDI input is enabled (with -M or -F), each incoming noteon message will generate a note event for an instrument which has the same number as the channel of the event (This means that MIDI controlled instruments are polyphonic by default, since each note will generate a new instance of the instrument.) If you have 1 instrument only, Csound works in omni mode, ie. it responds to all channels into that single instrument. If you have more than one instrument and instrs 1 - 16 , then by default instr 1 -> chn 1, instr 2 -> chn 2, unless you alter the mapping (see massign and pgmassign to change this behavior). If you have more than one instrument, but instr N in between 1 - 16 is missing, then chn N will be routed by default to the lowest order instrument. See the MIDI/Score Interoperability opcodes for information on designing instruments which can be used from the score or driven by MIDI. There are several realtime MIDI modules available, you must use the -+rtmidi flag (See -+rtmidi), to specify the module. The default module is portmidi which provides adecuate MIDI I/O on all platforms, however for improved performance and reliablity some platform specific modules are also provided. Currently the midi modules available are: alsa - To use the ALSA midi system (Linux only) winmme - To use the windows MME system (Windows only) portmidi - To use the portmidi system (all platforms). This is the default setting. virtual - To use a virtual graphical keyboard (See below) as MIDI input (all platforms) When csound runs, it will process the score and then quit. If there are no events in the score, Csound will exit immediately. If you want to use only MIDI events instead of score events, you need to tell Csound to run for a certain amount of time. This can be done with a dummy f-statement like "f 0 3600". &midivirtual; &midiinput; &midioutput; &midigeneric; &midiconvert; &midiextender; &midionoff; &midiinterop; &midirealtime; &midisliderbk; manual_src~dfsg/midi/convert.xml0000644000000000000000000000201512262561504016067 0ustar rootroot
Converters The following opcodes can convert MIDI information from a MIDI-triggered instrument instance: MIDI note number to frequency converters: cpsmidi, cpsmidib, cpstmid, octmidi, octmidib, pchmidi and pchmidib. MIDI velocity to amplitude converters: ampmidi and ampmidid.
manual_src~dfsg/midi/onoff.xml0000644000000000000000000000166012262561504015523 0ustar rootroot
Note-on/Note-off Output Opcodes to output MIDI note on or off messages are: midion midion2 moscil noteoff noteon noteondur noteondur2
manual_src~dfsg/midi/realtime.xml0000644000000000000000000000043712262561504016217 0ustar rootroot
System Realtime Messages Opcodes for System Realtime MIDI messages are: mclock and mrtmsg.
manual_src~dfsg/midi/sliderbkXO.xml0000644000000000000000000000472412262561504016466 0ustar rootroot
Slider Banks Opcodes for slider banks of MIDI controls are: slider8 slider8f slider16 slider16f slider32 slider32f slider64 slider64f s16b14 s32b14
manual_src~dfsg/midi/input.xml0000644000000000000000000000407312262561504015554 0ustar rootroot
MIDI input The following opcodes can recieve MIDI information: MIDI information for any instruments: aftouch, chanctrl and polyaft, pchbend. MIDI information for MIDI-triggered instruments: veloc , midictrl and notnum. See also Converters. MIDI Controller input for any instrument: ctrl7, ctrl14 and ctrl21. MIDI Controller input for MIDI-triggered instruments only: midic7, midic14 and midic21. MIDI controller value initialization: initc7, initc14, initc21 and ctrlinit. massign can be used to specify the csound instrument to be triggered by a particular MIDI channel. pgmassign can be use to assign a csound instrument to a specific MIDI program.
manual_src~dfsg/categories.py0000644000000000000000000001255012262561504015447 0ustar rootroot#categories and their order are defined here: categories = ['Orchestra Syntax:Header', 'Orchestra Syntax:Block Statements', 'Orchestra Syntax:Macros', 'Signal Generators:Additive Synthesis/Resynthesis', 'Signal Generators:Basic Oscillators', 'Signal Generators:Dynamic Spectrum Oscillators', 'Signal Generators:FM Synthesis', 'Signal Generators:Granular Synthesis', 'Signal Generators:Hyper Vectorial Synthesis', 'Signal Generators:Linear and Exponential Generators', 'Signal Generators:Envelope Generators', 'Signal Generators:Models and Emulations', 'Signal Generators:Phasors', 'Signal Generators:Random (Noise) Generators', 'Signal Generators:Sample Playback', 'Signal Generators:Scanned Synthesis', 'Signal Generators:STK Opcodes', 'Signal Generators:Table Access', 'Signal Generators:Wave Terrain Synthesis', 'Signal Generators:Waveguide Physical Modeling', 'Signal I/O:File I/O', 'Signal I/O:Signal Input', 'Signal I/O:Signal Output', 'Signal I/O:Software Bus', 'Signal I/O:Printing and Display', 'Signal I/O:Soundfile Queries', 'Signal Modifiers:Amplitude Modifiers', 'Signal Modifiers:Convolution and Morphing', 'Signal Modifiers:Delay', 'Signal Modifiers:Panning and Spatialization', 'Signal Modifiers:Reverberation', 'Signal Modifiers:Sample Level Operators', 'Signal Modifiers:Signal Limiters', 'Signal Modifiers:Special Effects', 'Signal Modifiers:Standard Filters', 'Signal Modifiers:Standard Filters:Resonant', 'Signal Modifiers:Standard Filters:Control', 'Signal Modifiers:Specialized Filters', 'Signal Modifiers:Waveguides', 'Signal Modifiers:Waveshaping', 'Signal Modifiers:Comparators and Accumulators', 'Instrument Control:Clock Control', 'Instrument Control:Conditional Values', 'Instrument Control:Compilation', 'Instrument Control:Duration Control', 'Instrument Control:Invocation', 'Instrument Control:Program Flow Control', 'Instrument Control:Realtime Performance Control', 'Instrument Control:Initialization and Reinitialization', 'Instrument Control:Sensing and Control', 'Instrument Control:Stacks', 'Instrument Control:Subinstrument Control', 'Instrument Control:Time Reading', 'Jacko Opcodes', 'Lua Opcodes', 'Serial I/O', 'Table Control', 'Table Control:Table Queries', 'Table Control:Dynamic Selection', 'Table Control:Read/Write Opreations', 'FLTK:Containers', 'FLTK:Valuators', 'FLTK:Other', 'FLTK:Appearance', 'Mathematical Operations:Arithmetic and Logic Operations', 'Mathematical Operations:Comparators and Accumulators', 'Mathematical Operations:Mathematical Functions', 'Mathematical Operations:Trigonometric Functions', 'Mathematical Operations:Amplitude Functions', 'Mathematical Operations:Random Functions', 'Mathematical Operations:Opcode Equivalents of Functions', 'Mathematical Operations:Vectors', 'Pitch Converters:Functions', 'Pitch Converters:Tuning Opcodes', 'Real-time MIDI:Input', 'Real-time MIDI:Output', 'Real-time MIDI:Converters', 'Real-time MIDI:Generic I/O', 'Real-time MIDI:Event Extenders', 'Real-time MIDI:Note Output', 'Real-time MIDI:MIDI/Score Interoperability', 'Real-time MIDI:System Realtime', 'Real-time MIDI:Slider Banks', 'Signal Flow Graph Opcodes', 'Spectral Processing:STFT', 'Spectral Processing:LPC', 'Spectral Processing:Non-Standard', 'Spectral Processing:Streaming', 'Spectral Processing:ATS', 'Spectral Processing:Loris', 'Spectral Processing:Other', 'Strings:Definition', 'Strings:Manipulation', 'Strings:Conversion', 'Vectorial:Tables', 'Vectorial:Scalar operations', 'Vectorial:Vectorial operations', 'Vectorial:Envelopes', 'Vectorial:Limiting and Wrapping', 'Vectorial:Delay Paths', 'Vectorial:Random', 'Vectorial:Cellular Automata', 'Zak Patch System', 'Plugin Hosting:DSSI and LADSPA', 'Plugin Hosting:VST', 'OSC', 'Faust Opcodes', 'Network', 'Remote Opcodes', 'Mixer Opcodes', 'Python Opcodes', 'Image Processing Opcodes', 'Array Opcodes', 'Miscellaneous', 'Testing', 'Utilities'] manual_src~dfsg/spectral/0000755000000000000000000000000012262561504014562 5ustar rootrootmanual_src~dfsg/spectral/realtimeXO.xml0000644000000000000000000002232012262561504017354 0ustar rootroot
Tools for Real-time Spectral Processing (pvs opcodes) With these opcodes, two new core facilities are added to Csound. They offer improved audio quality, and fast performance, enabling high-quality analysis and resynthesis (together with transformations) to be applied in real-time to live signals. The original Csound phase vocoder remains unaltered; the new opcodes use an entirely separate set of functions based on pvoc.c in the CARL distribution, written by Mark Dolson. Support for the new PVOC-EX analysis file format. This is a fully portable (cross-platform) open file format, supporting three analysis formats, and multi-channel signals. Currently only the standard amplitude+frequency format has been implemented in the opcodes, but the file format itself supports amplitude+phase and complex (real-imaginary) formats. In addition to the new opcodes, the original Csound pvoc opcodes have been extended (and thereby with enhanced audio quality in some cases) to read PVOC-EX files as well as the original (non-portable) format. Full details of the structure of a PVOC-EX file are available via the website: http://www.cs.bath.ac.uk/~jpff/NOS-DREAM/researchdev/pvocex/pvocex.html. This site also gives details of the freely available console programs pvocex and pvocex2 which can be used to create PVOC-EX files in all supported formats. A new frequency-domain signal type, fully streamable, with f as the leading character. In this document it is conveniently referred to as an fsig. Primary support for fsigs is provided by the opcodes pvsanal and pvsynth, which perform conventional phase vocoder overlap-add analysis and resynthesis, independently of the orchestra control-rate. The only requirement is that the control-rate kr be higher than or equal to the analysis rate, whch can be expressed by the requirement that ksmps <= overlap, where overlap is the distance in samples between analysis frames, as specified for pvsanal. As overlap is typically at least 128, and more usually 256, this is not an onerous restriction in practice. The opcode pvsinfo can be used at init time to acquire the properties of an fsig. The fsig enables the nominal separation between the analysis and resynthesis stages of the phase vocoder to be exposed to the Csound programmer, so that not only can alternatives be employed for either or both of these stages (not only oscillator-bank resynthesis, but also the generation of synthetic fsig streams), but opcodes, operating on the fsig stream, can themselves become more elemental. Thus the fsig enables the creation of a true streaming plugin framework for frequency domain signals. With the old pvoc opcodes, each opcode is required to act as a resynthesiser, so that facilities such as pitch scaling are duplicated in each opcode; and in many cases the opcodes are parameter-rich. The separation of analysis and synthesis stages by means of the fsig encourages the development of a wide range of simple building-block opcodes implementing one or two functions, with which more elaborate processes can be constructed. This is very much a preliminary and experimental release, and it is possible that the precise definition of the opcodes may change, in response to user feedback. Also, clearly, many new possibilities for opcodes are opened up; these factors may also have a retrospective influence on the opcodes presented here. Note that some opcode parameters currently have restricted or missing implementation. This is at least in part in order to keep the opcodes simple at this stage, and also because they highlight important design issues on which no decision has yet been made, and on which opinions from users are sought. One important point about the new signal type is that because the analysis rate is typically much lower than kr, new analysis frames are not available on each k-cycle. Internally, the opcodes track ksmps, and also maintain a frame counter, so that frames are read and written at the correct times; this process is generally transparent to the user. However, it means that k-rate signals only act on an fsig at the analysis rate, not at each k-cycle. The opocde pvsftw returns a k-rate flag that is set when new fsig data is valid. Because of the nature of the overlap-add system, the use of these opcodes incurs a small but significant delay, or latency, determined by the window size (max(ifftsize,iwinsize)). This is typically around 23msecs. In this first release, the delay is slightly in excess of the theoretical minimum, and it is hoped that it can be reduced, as the opcodes are further optimized for real-time streaming. The opcodes for real-time spectral processing are pvsadsyn, pvsanal, pvscross, pvsfread, pvsftr, pvsftw, pvsinfo, pvsmaska, and pvsynth. In addition there are a number of opcodes available as plugins in Csound5. These are pvsdiskin, pvscent, pvsdemix, pvsfreeze, pvsbuffer, pvsbufread, pvscale, pvshift, pvsifd, pvsinit, pvsin, pvsout, pvsosc, pvsbin, pvsdisp, pvsfwrite, pvsmix, pvsmooth, pvsfilter, pvsblur, pvstencil, pvsarp, pvsvoc, pvsmorph pvsbandp pvsbandr A number of opcodes are designed to generate and process streaming partials tracks data. these are partials, trcross, trfilter, trsplit, trmix, trscale, trshift, trlowest, trhighest tradsyn, sinsyn, resyn, binit See the Stacks section for information on the stack opcodes which can stack f-signals.
manual_src~dfsg/spectral/stft.xml0000644000000000000000000000504112262561504016264 0ustar rootroot
Short-time Fourier Transform (STFT) Resynthesis Use of PVOC-EX files with the old Csound pvoc opcodes All the original pvoc opcodes can now read a PVOC-EX file, as well as the native non-portable file format. As the PVOC-EX file uses a double-size analysis window, users may find that this gives a useful improvement in quality, for some sounds and processes, despite the fact that the resynthesis does not use the same window size. Apart from the window size parameter, the main difference between the original .pv format and PVOC-EX is in the amplitude range of analysis frames. While rescaling is applied, so that no significant difference in output level is experienced, whichever file format is used, some slight loss of amplitude can still arise, as the double window usage itself modifies frame amplitudes, of which the resynthesis code is unaware. Note that all the original pvoc opcodes expect a mono analysis file, and multi-channel PVOC-EX files will accordingly be rejected. Opcodes the implement STFT resynthesis are: mincer temposcal tableseg pvadd pvbufread pvcross pvinterp pvoc pvread tableseg tablexseg vpvoc Use the utility PVANAL to generate pv analysis files.
manual_src~dfsg/spectral/topXO.xml0000644000000000000000000000054612262561504016362 0ustar rootroot Spectral Processing See the section Additive Synthesis/Resynthesis for the basic resynthesis opcodes. &spectralstft; &spectrallpcresyn; &spectralnonstand; &spectralrealtime; manual_src~dfsg/spectral/nonstand.xml0000644000000000000000000000237612262561504017140 0ustar rootroot
Non-standard Spectral Processing These units generate and process non-standard signal data types, such as down-sampled time-domain control signals and audio signals, and their frequency-domain (spectral) representations. The data types (d-, w-) are self-defining, and the contents are not processable by any other Csound units. These unit generators are experimental, and subject to change between releases, they will also be joined by others later. The opcodes for non-standard spectral processing are specaddm, specdiff, specdisp, specfilt, spechist, specptrk, specscal, specsum, and spectrum.
manual_src~dfsg/spectral/lpcresyn.xml0000644000000000000000000000153412262561504017146 0ustar rootroot
Linear Predictive Coding (LPC) Resynthesis The linear predictive coding resynthesis opcodes are: lpfreson lpinterp lpread lpreson lpslot LPC analysis files can be created using the LPANAL utility.
manual_src~dfsg/spectral/top.xml0000644000000000000000000000053712262561504016113 0ustar rootroot Spectral Processing See the section Additive Synthesis/Resynthesis for the basic resynthesis opcodes. &spectralstft; &spectrallpcresyn; &spectralnonstand; &spectralrealtime; &spectralATS; &spectralloris; manual_src~dfsg/spectral/ATS.xml0000644000000000000000000000414612262561504015740 0ustar rootroot
ATS Spectral Processing These opcodes can read, transform and resynthesize ATS analysis files. Please note that you need the ATS application to produce analysis files. From the ATS Reference Manual: "ATS is a software library of functions for spectral Analysis, Transformation, and Synthesis of sound based on a sinusoidal plus critical-band noise model. A sound in ATS is a symbolic object representing a spectral model that can be sculpted using a variety of transformation functions." For more information on ATS visit: http://www-ccrma.stanford.edu/~juan/ATS.html. ATS analysis files can be produced using the ATS software or the csound utility ATSA. The opcodes for ATS processing are: ATSinfo: reads data out of the header of an ATS file. ATSread, ATSreadnz, ATSbufread, ATSinterpread, ATSpartialtap: read data from an ATS file or buffer. ATSadd, ATSaddnz, ATScross, ATSsinnoi: Resynthesize sound. Credits Author: Alex Norman Seattle,Washington 2004
manual_src~dfsg/spectral/realtime.xml0000644000000000000000000002310012262561504017102 0ustar rootroot
Tools for Real-time Spectral Processing (pvs opcodes) With these opcodes, two new core facilities are added to Csound. They offer improved audio quality, and fast performance, enabling high-quality analysis and resynthesis (together with transformations) to be applied in real-time to live signals. The original Csound phase vocoder remains unaltered; the new opcodes use an entirely separate set of functions based on pvoc.c in the CARL distribution, written by Mark Dolson. The Csound dnoise and srconv utilities (also by Dolson, from CARL) also use this pvoc engine. CARL pvoc is also the basis for the phase vocoder included in the Composer's Desktop Project. A few small but important modifications have been made to the original CARL code to support real-time streaming. Support for the new PVOC-EX analysis file format. This is a fully portable (cross-platform) open file format, supporting three analysis formats, and multi-channel signals. Currently only the standard amplitude+frequency format has been implemented in the opcodes, but the file format itself supports amplitude+phase and complex (real-imaginary) formats. In addition to the new opcodes, the original Csound pvoc opcodes have been extended (and thereby with enhanced audio quality in some cases) to read PVOC-EX files as well as the original (non-portable) format. Full details of the structure of a PVOC-EX file are available via the website: http://www.cs.bath.ac.uk/~jpff/NOS-DREAM/researchdev/pvocex/pvocex.html. This site also gives details of the freely available console programs pvocex and pvocex2 which can be used to create PVOC-EX files in all supported formats. A new frequency-domain signal type, fully streamable, with f as the leading character. In this document it is conveniently referred to as an fsig. Primary support for fsigs is provided by the opcodes pvsanal and pvsynth, which perform conventional phase vocoder overlap-add analysis and resynthesis, independently of the orchestra control-rate. The only requirement is that the control-rate kr be higher than or equal to the analysis rate, whch can be expressed by the requirement that ksmps <= overlap, where overlap is the distance in samples between analysis frames, as specified for pvsanal. As overlap is typically at least 128, and more usually 256, this is not an onerous restriction in practice. The opcode pvsinfo can be used at init time to acquire the properties of an fsig. The fsig enables the nominal separation between the analysis and resynthesis stages of the phase vocoder to be exposed to the Csound programmer, so that not only can alternatives be employed for either or both of these stages (not only oscillator-bank resynthesis, but also the generation of synthetic fsig streams), but opcodes, operating on the fsig stream, can themselves become more elemental. Thus the fsig enables the creation of a true streaming plugin framework for frequency domain signals. With the old pvoc opcodes, each opcode is required to act as a resynthesiser, so that facilities such as pitch scaling are duplicated in each opcode; and in many cases the opcodes are parameter-rich. The separation of analysis and synthesis stages by means of the fsig encourages the development of a wide range of simple building-block opcodes implementing one or two functions, with which more elaborate processes can be constructed. This is very much a preliminary and experimental release, and it is possible that the precise definition of the opcodes may change, in response to user feedback. Also, clearly, many new possibilities for opcodes are opened up; these factors may also have a retrospective influence on the opcodes presented here. Note that some opcode parameters currently have restricted or missing implementation. This is at least in part in order to keep the opcodes simple at this stage, and also because they highlight important design issues on which no decision has yet been made, and on which opinions from users are sought. One important point about the new signal type is that because the analysis rate is typically much lower than kr, new analysis frames are not available on each k-cycle. Internally, the opcodes track ksmps, and also maintain a frame counter, so that frames are read and written at the correct times; this process is generally transparent to the user. However, it means that k-rate signals only act on an fsig at the analysis rate, not at each k-cycle. The opocde pvsftw returns a k-rate flag that is set when new fsig data is valid. Because of the nature of the overlap-add system, the use of these opcodes incurs a small but significant delay, or latency, determined by the window size (max(ifftsize,iwinsize)). This is typically around 23msecs. In this first release, the delay is slightly in excess of the theoretical minimum, and it is hoped that it can be reduced, as the opcodes are further optimized for real-time streaming. The opcodes for real-time spectral processing are pvsadsyn, pvsanal, pvscross, pvsfread, pvsftr, pvsftw, pvsinfo, pvsmaska, and pvsynth. In addition there are a number of opcodes available as plugins in Csound5. These are pvstanal, pvsdiskin, pvscent, pvsdemix, pvsfreeze, pvsbuffer, pvsbufread, pvsbufread2, pvscale, pvshift, pvsifd, pvsinit, pvsin, pvsout, pvsosc, pvsbin, pvsdisp, pvsfwrite, pvslock, pvsmix, pvsmooth, pvsfilter, pvsblur, pvstencil, pvsarp, pvsvoc, pvsmorph, pvsbandp, pvsbandr, pvswarp, pvsgain, pvs2tab, tab2pvs. A number of opcodes are designed to generate and process streaming partials tracks data. these are partials, trcross, trfilter, trsplit, trmix, trscale, trshift, trlowest, trhighest tradsyn, sinsyn, resyn, binit See the Stacks section for information on the stack opcodes which can stack f-signals.
manual_src~dfsg/spectral/loris.xml0000644000000000000000000003201712262561504016437 0ustar rootroot
Loris Opcodes These opcodes are an optional component of Csound5. You can check if they are installed by using the command 'csound -z' which lists all available opcodes. The Loris family of opcodes wraps: lorisread which imports a set of bandwidth-enhanced partials from a SDIF-format data file, applying control-rate frequency, amplitude, and bandwidth scaling envelopes, and stores the modified partials in memory; lorismorph, which morphs two stored sets of bandwidth-enhanced partials and stores a new set of partials representing the morphed sound. The morph is performed by linearly interpolating the parameter envelopes (frequency, amplitude, and bandwidth, or noisiness) of the bandwidth-enhanced partials according to control-rate frequency, amplitude, and bandwidth morphing functions, and lorisplay, which renders a stored set of bandwidth-enhanced partials using the method of Bandwidth-Enhanced Additive Synthesis implemented in the Loris software, applying control-rate frequency, amplitude, and bandwidth scaling envelopes. For more information about sound morphing and manipulation using Loris and the Reassigned Bandwidth-Enhanced Additive Model, visit the Loris web site at www.cerlsoundgroup.org/Loris. Examples Play the partials wihtout modification ; ; Play the partials in clarinet.sdif ; from 0 to 3 sec with 1 ms fadetime ; and no frequency , amplitude, or ; bandwidth modification. ; instr 1 ktime linseg 0, p3, 3.0 ; linear time function from 0 to 3 seconds lorisread ktime, "clarinet.sdif", 1, 1, 1, 1, .001 asig lorisplay 1, 1, 1, 1 out asig endin Add tuning and vibrato ; Play the partials in clarinet.sdif ; from 0 to 3 sec with 1 ms fadetime ; adding tuning and vibrato, increasing the ; "breathiness" (noisiness) and overall ; amplitude, and adding a highpass filter. ; instr 2 ktime linseg 0, p3, 3.0 ; linear time function from 0 to 3 seconds ; compute frequency scale for tuning ; (original pitch was G#4) ifscale = cpspch(p4)/cpspch(8.08) ; make a vibrato envelope kvenv linseg 0, p3/6, 0, p3/6, .02, p3/3, .02, p3/6, 0, p3/6, 0 kvib oscil kvenv, 4, 1 ; table 1, sinusoid kbwenv linseg 1, p3/6, 1, p3/6, 2, 2*p3/3, 2 lorisread ktime, "clarinet.sdif", 1, 1, 1, 1, .001 a1 lorisplay 1, ifscale+kvib, 2, kbwenv a2 atone a1, 1000 ; highpass filter, cutoff 1000 Hz out a2 endin The instrument in the first example synthesizes a clarinet tone from beginning to end using partials derived from reassigned bandwidth-enhanced analysis of a three-second clarinet tone, stored in a file, clarinet.sdif. The instrument in Example 2 adds tuning and vibrato to the clarinet tone synthesized by instr 1, boosts its amplitde and noisiness, and applies a highpass filter to the result. The following score can be used to test both of the instruments described above. ; make sinusoid in table 1 f 1 0 4096 10 1 ; play instr 1 ; strt dur i 1 0 3 i 1 + 1 i 1 + 6 s ; play instr 2 ; strt dur ptch i 2 1 3 8.08 i 2 3.5 1 8.04 i 2 4 6 8.00 i 2 4 6 8.07 e Morph partials ; Morph the partials in clarinet.sdif into the ; partials in flute.sdif over the duration of ; the sustained portion of the two tones (from ; .2 to 2.0 seconds in the clarinet, and from ; .5 to 2.1 seconds in the flute). The onset ; and decay portions in the morphed sound are ; specified by parameters p4 and p5, respectively. ; The morphing time is the time between the ; onset and the decay. The clarinet partials are ; shfited in pitch to match the pitch of the flute ; tone (D above middle C). ; instr 1 ionset = p4 idecay = p5 itmorph = p3 - (ionset + idecay) ipshift = cpspch(8.02)/cpspch(8.08) ; clarinet time function, morph from .2 to 2.0 seconds ktcl linseg 0, ionset, .2, itmorph, 2.0, idecay, 2.1 ; flute time function, morph from .5 to 2.1 seconds ktfl linseg 0, ionset, .5, itmorph, 2.1, idecay, 2.3 kmurph linseg 0, ionset, 0, itmorph, 1, idecay, 1 lorisread ktcl, "clarinet.sdif", 1, ipshift, 2, 1, .001 lorisread ktfl, "flute.sdif", 2, 1, 1, 1, .001 lorismorph 1, 2, 3, kmurph, kmurph, kmurph asig lorisplay 3, 1, 1, 1 out asig endin More morphing ; Morph the partials in trombone.sdif into the ; partials in meow.sdif. The start and end times ; for the morph are specified by parameters p4 ; and p5, respectively. The morph occurs over the ; second of four pitches in each of the sounds, ; from .75 to 1.2 seconds in the flutter-tongued ; trombone tone, and from 1.7 to 2.2 seconds in ; the cat's meow. Different morphing functions are ; used for the frequency and amplitude envelopes, ; so that the partial amplitudes make a faster ; transition from trombone to cat than the frequencies. ; (The bandwidth envelopes use the same morphing ; function as the amplitudes.) ; instr 2 ionset = p4 imorph = p5 - p4 irelease = p3 - p5 kttbn linseg 0, ionset, .75, imorph, 1.2, irelease, 2.4 ktmeow linseg 0, ionset, 1.7, imorph, 2.2, irelease, 3.4 kmfreq linseg 0, ionset, 0, .75*imorph, .25, .25*imorph, 1, irelease, 1 kmamp linseg 0, ionset, 0, .75*imorph, .9, .25*imorph, 1, irelease, 1 lorisread kttbn, "trombone.sdif", 1, 1, 1, 1, .001 lorisread ktmeow, "meow.sdif", 2, 1, 1, 1, .001 lorismorph 1, 2, 3, kmfreq, kmamp, kmamp asig lorisplay 3, 1, 1, 1 out asig endin The instrument in the first morphing example performs a sound morph between a clarinet tone and a flute tone using reassigned bandwidth-enhanced partials stored in clarinet.sdif and flute.sdif. The morph is performed over the sustain portions of the tones, 2. seconds to 2.0 seconds in the case of the clarinet tone and .5 seconds to 2.1 seconds in the case of the flute tone. The time index functions, ktcl and ktfl, align the onset and decay portions of the tones with the specified onset and decay times for the morphed sound, specified by parameters p4 and p5, respectively. The onset in the morphed sounds is purely clarinet partial data, and the decay is purely flute data. The clarinet partials are shifted in pitch to match the pitch of the flute tone (D above middle C). The instrument in the second morphing example performs a sound morph between a flutter-tongued trombone tone and a cat's meow using reassigned bandwidth-enhanced partials stored in trombone.sdif and meow.sdif. The data in these SDIF files have been channelized and distilled to establish correspondences between partials. The two sets of partials are imported and stored in memory locations labeled 1 and 2, respectively. Both of the original sounds have four notes, and the morph is performed over the second note in each sound (from .75 to 1.2 seconds in the flutter-tongued trombone tone, and from 1.7 to 2.2 seconds in the cat's meow). The different time index functions, kttbn and ktmeow, align those segments of the source and target partial sets with the specified morph start, morph end, and overall duration parameters. Two different morphing functions are used, so that the partial ammplitudes and bandwidth coefficients morph quickly from the trombone values to the cat's-meow values, and the frequencies make a more gradual transition. The morphed partials are stored in a memory location labeled 3 and rendered by the subsequent lorisplay instruction. They could also have been used as a source for another morph in a three-way morphing instrument. The following score can be used to test both of the instruments described above. ; play instr 1 ; strt dur onset decay i 1 0 3 .25 .15 i 1 + 1 .10 .10 i 1 + 6 1. 1. s ; play instr 2 ; strt dur morph_start morph_end i 2 0 4 .75 2.75 e Credits This implementation of the Loris unit generators was written by Kelly Fitz (loris@cerlsoundgroup.org). It is patterned after a prototype implementation of the lorisplay unit generator written by Corbin Champion, and based on the method of Bandwidth-Enhanced Additive Synthesis and on the sound morphing algorithms implemented in the Loris library for sound modeling and manipulation. The opcodes were further adapted as a plugin for Csound 5 by Michael Gogins.
manual_src~dfsg/pitch/0000755000000000000000000000000012262561504014054 5ustar rootrootmanual_src~dfsg/pitch/funcs.xml0000644000000000000000000000270012262561504015713 0ustar rootroot
Functions Opcodes that provide common pitch functions are: cent cpsmidinn cpsoct cpspch octave octcps octmidinn octpch pchmidinn pchoct semitone
manual_src~dfsg/pitch/top.xml0000644000000000000000000000015012262561504015374 0ustar rootroot Pitch Converters &pitchfuncs; &pitchtuning; manual_src~dfsg/pitch/tuning.xml0000644000000000000000000000121012262561504016074 0ustar rootroot
Tuning Opcodes Opcodes that provide tuning functions are: cps2pch cpsxpch cpstun cpstuni
manual_src~dfsg/opcodeparser.py0000644000000000000000000001643012262561504016011 0ustar rootroot# This script generates the opcode.xml file # by Andres Cabrera June 2006-2010 # Licensed under the GPL licence version 3 or later # modification for empty arg in command and links on opcodes by Francois Pinot February 2007 from xml.dom import minidom import os, glob, sys # categories holds the list of valid categories for opcodes from categories import categories XO = False opcodelist = [] outfilename = 'opcodes.xml' quickref = open(outfilename,'w') quickref.write("\n") quickref.write('''\n''') entries = [] for i in categories: entries.append([]) manualfilename = 'manual.xml' manual = open(manualfilename, 'r') text = manual.read() manual.close() files = glob.glob('opcodes/*.xml') files[len(files):]=glob.glob('opcodes/*/*.xml') files[len(files):]=glob.glob('vectorial/*.xml') files[len(files):]=glob.glob('utility/*.xml') files.sort() if files.index('opcodes/topXO.xml') >= 0: files.remove('opcodes/topXO.xml') headerText = text[0:text.find(' entryText = source.read().replace("\xef\xbb\xbf","") newfile = headerText + entryText newfile = newfile.replace("\r", "") source.close() #print text xmldoc = minidom.parseString(newfile) xmldocId = xmldoc.documentElement.getAttribute('id') # Some files need special treatment (adds, dollar, divides, modulus, multiplies, # opbitor, opor, raises, subtracts, assign, ifdef, ifndef, define, include, undef) # There must be a better way to avoid loosing the entities when parsing the XML # file. Anyone??? if (filename == 'opcodes' + os.sep + 'adds.xml'): entry = 'a + b (no rate restriction)\n' elif (filename == 'opcodes' + os.sep + 'dollar.xml'): entry = '$NAME\n' elif (filename == 'opcodes' + os.sep + 'divides.xml'): entry = 'a / b (no rate restriction)\n' elif (filename == 'opcodes' + os.sep + 'modulus.xml'): entry = 'a % b (no rate restriction)\n' elif (filename == 'opcodes' + os.sep + 'multiplies.xml'): entry = 'a * b (no rate restriction)\n' elif (filename == 'opcodes' + os.sep + 'opbitor.xml'): entry = 'a '+'| b (bitwise OR)\n' elif (filename == 'opcodes' + os.sep + 'opor.xml'): entry = 'a '+'|| b (logical OR; not audio-rate)\n' elif (filename == 'opcodes' + os.sep + 'raises.xml'): entry = 'a ^ b (b not audio-rate)\n' elif (filename == 'opcodes' + os.sep + 'subtracts.xml'): entry = 'a - b (no rate restriction)\n' # elif (filename == 'opcodes' + os.sep + 'assign.xml'): # entry = ''' '+'−a (no rate restriction) # '+'+a (no rate restriction)\n''' elif (filename == 'opcodes' + os.sep + 'ifdef.xml'): entry = '#ifdef NAME ....' + \ '#else ....' + \ '#end\n' elif (filename == 'opcodes' + os.sep + 'ifndef.xml'): entry='#ifndef NAME ....' + \ '#else ....' + \ '#end\n' elif (filename == 'opcodes' + os.sep + 'define.xml'): entry='#define NAME # replacement text #\n' + \ '#define NAME(a' b' c') # replacement text #\n' elif (filename == 'opcodes' + os.sep + 'include.xml'): entry='#include "filename"\n' elif (filename == 'opcodes' + os.sep + 'undef.xml'): entry='#undef NAME\n' elif (filename == 'opcodes' + os.sep + '0dbfs.xml'): entry = '0dbfs = iarg\n' else: synopsis = xmldoc.getElementsByTagName('synopsis') if (len(synopsis) != 0): # There can be more than 1 synopsis per file for num in range(len(synopsis)): tmp = synopsis[num].toxml() if XO: opcodename = tmp[tmp.find('') + 9:tmp.find('')] else: opcodename = "" tmp = tmp.replace('', '') entry += tmp.replace('', '') #if entry != '': #entry += '' else: #print "no synopsis tag for file: " + file entry = '' #print "Entry ------ ", entry info = xmldoc.getElementsByTagName('refentryinfo') if (len(info)!=0 and entry != ''): category = info[0].toxml() category = category[21:-23] #print filename, category else: print "no refentryinfo tag for file " + filename category = "Miscellaneous" if (entry!=''): print filename + " sent to Miscellaneous" desc = xmldoc.getElementsByTagName('refpurpose') description = "" if (len(desc)!=0 and entry != ''): description = desc[0].firstChild.toxml().strip() #print filename, category else: print "no refpurpose tag for file " + filename #print category match = False for j, thiscategory in enumerate(categories): if (category == thiscategory): entries[j].append([entry, description]) match = True if match == False: print filename + "---- WARNING! No Category Match!" for i in range(len(categories)): if (len(entries[i])==0): print "No entries for category: "+categories[i]+"...Skipping" continue #quickref.write("\n") quickref.write("\n") count = 0 for j in range(len(entries[i])): newentry = entries[i].pop(0) #entry = entry.replace("$", "$") #entry = entry.replace(" ", " ") quickref.write("" + description) # + '\n') quickref.write(newentry[1] + "") # + '\n') quickref.write(newentry[0] + "\n") # + '\n') count += 1 #quickref.write("\n") quickref.write("\n") print str(count) + " entries in category: " + categories[i] quickref.write('\n') quickref.close() print entries manual_src~dfsg/images/0000755000000000000000000000000012262561504014212 5ustar rootrootmanual_src~dfsg/images/gen21_4.png0000644000000000000000000001712112262561504016061 0ustar rootrootPNG  IHDR<ٴe zTXtRaw profile type exifxڥir;s s9|j[fԶUb@d"PG6>z|QG|~rLq9}}z?ge|?χ_,DԇJ"ߟφ]vOϙ-+Dk?N^fR.wZlH!߮?=q>÷8Eyu]~|r՟os`so;Y/̎gA÷nBN\CK,kRPM3ITI3YH&os״Hy?PT%RF9ErCl>yNʓObiNl+E2s9"O}Wv\a,d'?^?}/Vyθ-VK/Χ`[XxFg7j=:吳D"fK RkdtQ/Gҳ6^G@Q6L f+zmK޻t)E4ADeԢU":t< F2t1'Ϝ<{\YyUW[}5gv߲u=O>N?r3oPWq5+VY71a )__A-?H:B"{x&@\9fqelIqk(,}ar.r?{p }7 V4ϚZ엘lv,ٻ;5eveWEV,{Co(h)i,lh5ooNh'}ieHc<]ۜArԩZZ=+H&w42⥙zJO@UT:[{?QIu%=߶K?jejWh\kkҋI#iC'&AdeGs<<=,;7gЁB&٦9wG0iqbnq0ЭX}ҽDߥE@K4[Od-kK= c[kNhHUc9:dofHL?`K4B ak5?3eKsvvNߎC%xYR\dmp[גl܉xP /2ncQO!gԋ]pU--\8Mzpk}[6 xvh$qZ!{k ?5*ama6z)e9ub%IjZ RmY Jm*!JΖ'?K^jJƨ4 /K+4CY/:ՋDT3Q+Ouiyq}*!e]yTWQMG]TM4I5; BW?ӎ5;=ؙ^KBA5.:n g5snU{c^Cw1$nzŅ;tѨj9"ݔL!۵8.P65E"qq7Jݕnd|\pܵ<LF#9pQW%zTzk,}2t&z\C#2Հ>kzR NNj!>~ݽid|(${_b>irk8=9mus$SǘʣiJV.T:{]a$f'/)Y+To&0P}jRdA؁}sxϩ!M!U(6ULiȂ.[8.rWiQ`eΗ֡%5Wʅ"~ u!6⃋l+.dm9p!%E,Ƒ.D«T25Aݢ(h. y4aށQ}u)y`TTۍ.4Błi8 ˬVֈm\15fe #}Bጱ9"uDE 32 f塀f%H@ $1A)@&)Ey#CRth(9"#G8KX z`Qw&au6pkCDlטw:ٻcjAŚOܙDjZk[%C-]RaD=ikS^sL@.#!< {*QJ$ͪD^6x_F}**C b<ilE8[ԮD{4e1c|ϲ0o=:IPgh 1\LUĠY$X?N qMFD9s[W'RYKj4`D|Fr>>6/7LPDBX,d2FZ~q{{;9TWW;VPQQQÁd BZMռ[VFVDDoY^^&-S}}=ySx^j=˵X,!!!0sLBdd$gz.==ɓ'}׮]*j|>0k֬!/c?۷o|||3]i4h9֬Ynݺm۶MÑ#Gfn߸qㅌw{{;WN$ڵk:66TUUm߾^ydl,|*`& g;xGj L6 YfTܺu+G#d2T`i4f}||X,2䍠t?- LůB\\hf޼y:!!wDGG=`X,,>/|><c2555-X8TTK222#a؜9s㭛{%vrFX,vq8$%$$$PBP& §\}}D"yddd?Ps7n47L-( ~樨(Bo3lddmM`rkf0ofX,6?!dE"***d2 ..=xChm޼\HP9~Yӏ?Q0ՊC>>>=%KXo.((H=/:uuuRs\NJ6, j4@QQQVV|ɒ%...o>Ooo/ p8TWW_~j P*dHB !.` Z]UUwXaaazTAA{"вpkgee;w4aa;C/Xcc#t |brJܹs[laY,//OJJ—,ZZ555Px.ؘ?p8>""?N:A5gΜywV-\oeffZFvdB̟?_ȑ#qqq41>;;;00I9d; P10t=͗ E9ZM DÇx=:: Wlٲms>?::J_G8Drݻw%/^ɹ{.j|>;v8{,=H$"R N=Hҁ*V&00pqF RSSFRr%K]Fn /#::W^y_+ 7pmmmK믰37Wi!rp>O0p :Bܹs υu0ΆFV%HLt:+0 $$믿Fpptt4Al. -gAE㵷Ӵٰa}}}aعs'Mw)))pF=z> N'!8y5kO\^RRBš ӌfrVLL Nϟ?Od4VVPP@`JVs1uuuxeڵ?b`Fd, T*I@rr2#_sٲe4NtNF!,MMMx]"@jZ_sssbb"T(} m6!H*}'_~%*\bŅ d7oˣ !,,tkSRRe CFhh(%m0KkffShX|>b\6ϏXCd?\*"**_m7 ?pYTj***cn34Mݖ>|xgbŊ7o˳jjjU4D"b*쒒Z[[i,^P.9'N޽J8<nݢ"oh{H$#4tdd$44)bx\9vKGl?'ó2&QDB ߻w$|/|Scj*<GcǎMN !W 6[n=w nS vZplB!JA0.,,ZvztJJJݻ oB y @TTj'===111CM3]w˗9 L&9s&~çݮ6m\.i<&!BBB`6OYYT*e=܎!Z._Lv \ rS ŒC#SއD"a6.]"dfӒyŧ9j4&#((HR͝;w2`0y 7Bl6Bu0 v$D౱1hro5FGG_ )bеw$~+oó˩ƾ>'2M%a V+gx^l6whH[ZZ.\gLʕ+=R~gYYYL u;vI|ݺuOHt:7lR8p#RZWHyT*`2:::\~<IENDB`manual_src~dfsg/images/gen02_1.png0000644000000000000000000001013512262561504016053 0ustar rootrootPNG  IHDR<ٴeazTXtRaw profile type exifxڥiݺ sY'r8VeY~>HHH ?_\9Z䪽OEߺ> zuXτM6SiTbK2FW*TLyscxƇ>`w[~/'_W{fOϪ駾os`?AO+ˋo|]n< w"Z 6n܆Oqɽ}Ʃ7մH74VZU3PgEs끢n\({ǖ۟V2Vމ91Yz'w+N>]q ~*ܿ\ዣafc#w)ETl K'- FF8)HR$7FZJo4=cM )`4U"DŤhФmhѪvVb`fFϽ}[n1Xs0ha0~3,NmXg%-];ݶn}T:H8ϸp[vw|>!7PK򃔏/U} 8f kqu tv̢Z#Ş )Nqk(,7}a?n?-UC"7mOABi,Dߕzc=OmǾN׶z-,iƄ)^nYӭs榘m52- mCHsg}^ 3ϙoQ P:.6i;q -! j^ՐTcP'hSm8MJ¶U Ϊi<_a#0_ӄţ%"A$ʆ:܄LJ8Z6.NaMj/gH5&dU D@zδ1mb!5Mw(ԑMtrDBZfɔV1Uz#ھxMgx2T 吣WɶJC. qNI?^O=D(6;BbK0tߜs@El5$lZ(.g e=J߰_{߂$g&WSb M"< ;cJ^@w7c,ݪ ~%x>[~;OI#0~v" /gڴ%|BR-exQ6gp~ĹoC(,#ywPGRytjW:6?L$zAn{^EXaQSҦhT u"^G5r W Z4n\{#d&%ly/l$ʬup/ AЍH(Sc$XNwsNo%&_j R{I3^(tFu^ƍ40MO.M >? lzB.?0!-b{JgKʁlד?>^tw<_'f'Y$4X(ksrL2~D OD(u/{TjȌ홞$:]Wy)؛sBITOIDATxoHSoǟm\[mT61 c$E E,7 BP_E6CWf&ᴬmNgksskwwz$Yg۽ۼy79sp,Hd|:LUy]\ liia20֨P(t}bQt:֭K.%!@Qp(=Qe,R isyyn(=Q:Ea rnwԴo51l ;P( aQ*^"0 A`0HsHAAf HkmmeXxESXXXTTWE,X,;,m# F1 X,=&s-o,l6hBP(j@@DpNsxx "I*vkRcA;v7}+(ήOqq1'8lkkYD~b}zyyy91Z>_I̽P(}p.>999~?9ҐZnppprroAmmmSSS<epppnn` ?~:ǏM-cll TQ(χc\QQ1==d@ L&kkkK~z'Nxi]PchrrԩS3x` Cb8rML4U uaad2K(~Ν0l,3ͽsssFӹ~A_@bBͰv/..D)ѤQzjH|>FH$I;qR__?44D[埵\B!aZN*[zST$hZ۔4xa0{umP VPtng2 %g LLLHTB~ACQY,ݻWʕ+=J=ZZZTB쵕:7'_*X<22h```MҐeHR)7x)++cX}}}@Vv/D9#mÇf0 χ}S]dógϪRΠ(\J_nҭ+s}V+=c۽w4J-WVV׷qxJRP:b8ydFĎlw`6wD|8PT*ADSYwelٲTN:)Ν;Oٮ];o޼){BCC>|PLʛ7ooU1шOUN>-{5k֤I]vy6lVpq1)RY➞M6x`N{ KXPP:+xCI *)SL_~DD .(z'NAA!!!۷/ E>zuQ#>LJE}}}$z"A vZZof_XZZ?5z BP^hooo>d }#:ё T{x$ڭ[ ˆдi3g BB1`^D9GMUC^8RSq Ah n߾5c D'ք ѫW1 Pv*fݺغYY/I! uN, ;qB ۶ũS/a d1ҸЭAȎT/tq4KK23O>gϞZ\f 8Ag˖-~|.N^h B!z#L0!BCCc9>5jTQQ>WPx</tqqq"?OP<'R}QXXXhwA6Tajݺ5AN,WW׬, ck5r'1xzz޿_$ rʌ-H8@B"IJJsαcd9ڵ^m8@}nP8VINNΝA6kN=>|h%PS#"--M6VիWwss3W4=###))/ܶm? &dѣ.]zMeo۷_x177wr. ? 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There are many ways to do this, each affording a different manner of control. Direct synthesis generates waveforms by sampling a stored function representing a single cycle; additive synthesis generates the many partials of a complex tone, each with its own loudness envelope; subtractive synthesis begins with a complex tone and filters it. Non-linear synthesis uses frequency modulation and waveshaping to give simple signals complex characteristics, while sampling and storage of a natural sound allows it to be used at will. Since comprehensive moment-by-moment specification of sound can be tedious, control is gained in two ways: 1) from the instruments in an orchestra, and 2) from the events within a score. An orchestra is really a computer program that can produce sound, while a score is a body of data which that program can react to. Whether a rise-time characteristic is a fixed constant in an instrument, or a variable of each note in the score, depends on how the user wants to control it. The instruments in a Csound orchestra (see Syntax of the Orchestra) are defined in a simple syntax that invokes complex audio processing routines. A score (see The Standard Numeric Score) passed to this orchestra contains numerically coded pitch and control information, in standard numeric score format. Although many users are content with this format, higher level score processing languages are often convenient. The programs making up the Csound system have a long history of development, beginning with the Music 4 program written at Bell Telephone Laboratories in the early 1960's by Max Mathews. That initiated the stored table concept and much of the terminology that has since enabled computer music researchers to communicate. Valuable additions were made at Princeton by the late Godfrey Winham in Music 4B; my own Music 360 (1968) was very indebted to his work. With Music 11 (1973) I took a different tack: the two distinct networks of control and audio signal processing stemmed from my intensive involvement in the preceding years in hardware synthesizer concepts and design. This division has been retained in Csound. Because it is written entirely in C, Csound is easily installed on any machine running Unix or C. At MIT it runs on VAX/DECstations under Ultrix 4.2, on SUNs under OS 4.1, SGI's under 5.0, on IBM PC's under DOS 6.2 and Windows 3.1, and on the Apple Macintosh under ThinkC 5.0. With this single language for defining the audio signal processing, and portable audio formats like AIFF and WAV, users can move easily from machine to machine. The 1991 version added phase vocoder, FOF, and spectral data types. 1992 saw MIDI converter and control units, enabling Csound to be run from MIDI score-files and external keyboards. In 1994 the sound analysis programs (lpc, pvoc) were integrated into the main load module, enabling all Csound processing to be run from a single executable, and Cscore could pass scores directly to the orchestra for iterative performance. The 1995 release introduced an expanded MIDI set with MIDI-based linseg, butterworth filters, granular synthesis, and an improved spectral-based pitch tracker. Of special importance was the addition of run-time event generating tools (Cscore and MIDI) allowing run-time sensing and response setups that enable interactive composition and experiment. It appeared that real-time software synthesis was now showing some real promise. manual_src~dfsg/preface/gettingstartedXO.xml0000644000000000000000000002321712262561504020400 0ustar rootroot

Getting Started with Csound What you need Csound comes preinstalled in the XO and it is used internally by other activities like TamTam, so the XO comes ready to use all the material in this manual. This manual can be browsed online here, but it might be a good idea to download it as a content bundle here, because it will be accessible fast and offline from the Web Activity under the "Media" category in the initial page. Running When Csound runs, it reads a Csound file and produces audio according to the instructions provided by that file. Csound can be run in different ways. It can be called from a terminal, or from a graphical front-end. A front-end is a graphical program that assists running Csound and can usually help edit csound files, like the Csound Editor Activity . Using the Csound Editor Activity [The Csound Editor Activity] The Csound Editor Activity The Csound Editor Activity provides an easy way of editing and playing Csound files. You can load and save Csound files using this panel. When you select load from this menu, you will find the directory containing the Csound Examples. There are several directories there that contain various example files. Valid files are the ones that end in .csd. You can also navigate to this manual's examples directory which is located (by default) in the directory /home/olpc/Library/csound5.08_manual_xo/examples. The time and transport bar shows the current time of the file playing and provides buttons for starting, pausing or stopping a performance. After you have loaded a Csound file, you will want to try the Play button (the one with the triangle). The upper text panel of the editor shows the orchestra section of the Csound file. This is where instruments and sound processes are defined. The lower text panel contains the score. The score lists a set of events which trigger the instruments defined in the orchestra section. This are the "notes" the intruments play. The buttons select what is displayed in the lower panel. The lower panel can show the Score section, the Csound console output (a report of the internal processes of Csound) and a MIDI keyboard for realtime generation of note events. Using OLPCsound from the Terminal Activity To use Csound from the command line, open the Terminal Activity. You can try the example files included with this manual line by moving to the examples directory of the manual like this: cd /home/olpc/Library/csound5.08_manual_xo/examples and then typing: csound oscils.csd This will make csound play the file "oscils.csd". You can see the long list of examples available by typing: ls or: ls|more which lists all the contents in the current directory. Editing and writing your own csd files Csound reads text files which contain instructions for producing sound. These files usually have the extension ".csd" and are called csd files. They can be created and edited using simple text editors like Pippy or the Csound Editor Activity. Csound's .csd files contain 3 main sections contained within <CsSynthesizer> and </CsSynthesizer> tags: <CsOptions> - Includes the Command Line flags specific to this particular file. These options can also be set using the .csoundrc file or directly in the command line. Some frontends also provide ways to specify global or local options. <CsInstruments> - Contains the instruments or processes available in the file. Instruments are defined using the instr and endin opcodes. The CsInstruments section also contains the Orchestra Header which defines things like sample rate, the number of samples in a control period and the number of output channels. <CsScore> - Contains the 'notes' to be played optionally the definition of f-tables. Notes are created using the i statement, and f-tables are created using the f statement. There are several other score statements available. Note that anything after a semicolon (;) until the end of the line is a comment, and is ignored by Csound. When you use the Csound Editor Activity, you will not see the tags mentioned above, as it hides them for easier editing. You will only see two panels containing the Orchestra and Score sections of the csd file. In other words, if you are using the Csound Editor Activity, you don't need to worry about these tags and the structure of the file, as the activity will take care of that for you. To use the csound examples in this manual, you can open them directly from the Csound Editor Activity, or you can copy (CTRL-C) and paste (CTRL-V) directly from the page. When you copy and paste, remember you must copy/paste the Orchestra (<CsInstruments>) and Score (<CsScore>) sections separately. Remember to exclude the tags (e.g. <CsInstruments>) in the Csound Editor Activity.
manual_src~dfsg/preface/toot2XO.xml0000644000000000000000000000231712262561504016415 0ustar rootroot
Toot 2 The first instrument was not interesting because it could play only one note at one amplitude level. We can make things more interesting by allowing the pitch and amplitude to be defined by parameters in the score. Each column in the score constitutes a parameter field, numbered from the left. The first three parameter fields of the i statement have a reserved function: p1 = instrument number p2 = start time p3 = duration All other parameter fields are determined by the way the sound designer defines his instrument. In the instrument below, the oscillator's amplitude argument is replaced by p4 and the frequency argument by p5. Now we can change these values at i-time, i.e. with each note in the score. The csd file toot2.csd now looks like:
manual_src~dfsg/preface/toot1XO.xml0000644000000000000000000000664212262561504016421 0ustar rootroot
Toot 1 In the Csound language, the words are called opcodes. Opcodes are instructions that Csound understands and performs actions with. This manual contains a long list of opcodes in alphabetical order in the Reference Section. Opcodes can produce results and receive parameters. The output of an opcode will always be to the left, while the inputs or parameters will be to the right. This first csd file, called toot1.csd contains a single instrument which uses the oscil opcode to play a 440Hz sine wave (defined by f1 in the score) at an amplitude of 10000. If you are using the Csound Editor Activity, remember that you can open csd files directly from the File Menu. You can also copy and paste from this page, but you must paste the orchestra section and the score section separately. To do this, select the text between the <CsInstruments> and </CsInstruments> tags, without including them. Then press CTRL-C, to copy to the clipboard. Switch to the Csound Editor Activity and click on the Instrument panel, then press CTRL-V. To paste the score section, do the same for all text contained between the <CsScore> and </CsScore> tags, and paste it on the Editor's Score panel. The opcode oscil produces an oscillation which is stored in variable a1. That is the output or result of the opcode as it is to the left. On the right we have the parameters the opcode uses to determine it output. In the case of oscil, we need to specify the amplitude of oscillation, which is the first parameter, the frequency of oscillation, which is the second one, and the shape or waveform of the oscillation. The shape of the oscillation is created not in the orchestra, but in the score section. The shape is stored in the computer's memory using what Csound calls an f-table, which is created using an f statement. There are many different shapes you can create using the different GEN routines provided by Csound. The oscillation produced by the oscil opcode is made audible by the out opcode. Notice that this opcode doesn't have any outputs. Its output is actually the sound you hear! In the Orchestra section of a csd file we define the instruments, but no instrument sounds if it is not played. To play an instrument, the i statement in the score must be used.
manual_src~dfsg/preface/topXO.xml0000644000000000000000000000050112262561504016141 0ustar rootroot Preface &pfacepreface; &pfacegettingstarted; &pfacetootorials; manual_src~dfsg/preface/gettingstarted.xml0000644000000000000000000001725312262561504020134 0ustar rootroot
Getting Started with Csound Downloading In case you don't already have Csound (or have an older version) download the appropriate Csound version for your platform from the Sourceforge Csound Download Page. Installers for Windows have '.exe' extension and for Mac '.dmg'. If the installer's filename ends in '-d' it means the installer has been built with double precision (64-bit) which provides higher quality output than the ordinary float precision (32-bit). The float versions provide quicker output, which may be important if you're using Csound in a real-time setting. You can also download the sources and build them, but this requires more expertise (See the section Building Csound). It may also be useful to download the most recent version of this manual, which you will also find there. Running Csound can be run in different ways. Since Csound is a command line program (DOS in Windows terms), just clicking on the csound executable will have no effect. Csound must be called either from the computer's command line or from a front end. To use Csound from the command line, you must open a Terminal (Command Prompt or DOS Prompt on Windows, or Terminal on MacOS). Using Csound from the command line can be difficult if you've never used a terminal, so you may want to try to use one of the front ends, either QuteCsound, which is included with the latest distributions, or another front end. A front end is a window-based (not necessarily Windows-based) program that assists running Csound. Most front ends include text editors with which you can edit csound files, and many include other useful features. Whether being run from a front end or being executed from the command line, Csound needs two things: A Csound file ('.csd' or possibly an '.orc' and a '.sco' file) A list of command line flags (or configuration options) that configure execution. They determine things like output filename and format, whether real-time audio and MIDI are enabled, which audio output to use for real-time audio, the buffer size, the types of messages printed, etc. These options can be included in the '.csd' file itself, so for the examples included in this manual you shouldn't need to worry about them. Front end programs often have dialog boxes in which the command line flags can be set. The complete and very long list of available command flags can be found here, but you might want to have a look there later... See the section Configuring if Csound is giving you trouble. This documentation includes many '.csd' files which you can try out, and which should work directly from the command line or from any front end. A simple example is oscil.csd, which can be found in the examples folder of this documentation. Your front end should allow you to load the file, and the front end should have a 'play' or 'render' button that will allow you to hear the file. If you want to experiment with the file, you're well advised to use the front end's 'Save As..." command to copy it to some other directory on your hard drive, such as a 'csound scores' directory that you create. Note for MacCsound users You might need to remove all the lines from the command options slot in order for the manual examples to work. You can also try the manual examples from the command line. To do this, navigate to the examples directory of the manual using something like this on Windows (assuming the manual is located at c:\Program Files\Csound\manual\): cd "c:\Program Files\Csound\doc\manual\examples" or something like: cd /manualdirectory/manual/examples for the Mac or linux Terminal. Then type: csound oscil.csd The example files are configured to run in real time by default, so with this command you should hear a two-second sine wave. Writing your own .csd files A .csd file looks like this (this file is oscils.csd): Csound's .csd files have three main sections between the <CsSynthesizer> and </CsSynthesizer> tags: CsOptions - Includes the Command Line flags specific to this particular file. These options can also be set using the .csoundrc file, which you can edit in a text editor, or directly in the command line. Some front ends also provide ways to specify global or local options. CsInstruments - Contains the instruments or processes available in the file. Instruments are defined using the instr and endin opcodes. The CsInstruments section also contains the Orchestra Header, which defines things like sample rate, the number of samples in a control period, and the number of output channels. CsScore - Contains the 'notes' to be played, and optionally the definition of f-tables. Notes are created using the i statement, and f-tables are created using the f statement. Several other score statements are available. Anything after a semicolon (;) until the end of the line is a comment, and is ignored by Csound. You can write .csd files in any plain text editor, such as Notepad or Textedit. If you use a word processor (not recommended), be sure to save the file as plain text (not rich text). Many front ends include advanced editing capabilities, such as syntax highlighting and auto-completion of code. You can find an in-depth tutorial on getting started with Csound written by Michael Gogins here.
manual_src~dfsg/preface/historyXO.xml0000644000000000000000000000736312262561504017055 0ustar rootroot History of the Canonical Csound Reference Manual This initial version of this manual for early versions of Csound was started at MIT by &namebarry; and maintained there during the 1980's and start of the 1990's. Some of the manual comes from documents for programs like Music11 from the 1970's. This original manual was improved and worked on by Richard Boulanger, &namejohn;, Jean Piché and &namerasmus;. This manual led to the Official Csound Reference Manual, still located at: http://www.lakewoodsound.com/csound, for Csound version 4.16, November, 1999, which was maintained by &namedavid;. A parallel version of the manual called the Alternative Csound Reference Manual, was developed by &namekevin; using DocBook/SGML. This version later became the Canonical version. When Csound was licenced as LGPL by MIT in 2003, the manual was licenced GFDL and placed on Sourceforge along with the sources of Csound. In the winter of 2004, the Canonical Manual was converted to DocBook/XML by &namesteven; to allow for more people to be able to compile and maintain the manual. The manual is currently maintained by &nameandres; with continuous contributions from the Csound Community. The manual continues to be a community run project that depends on the contributions of developers and users to help refine the coverage and accuracy of its contents. All contributions are welcome and appreciated. Other Contributors Mike Berry Eli Breder Michael Casey Michael Clark Perry Cook Sean Costello Richard Dobson Mark Dolson Dan Ellis Tom Erbe Bill Gardner &namemichael;Matt Ingalls Richard Karpen &nameanthony;&namevictor; Allan LeeDavid Macintyre &namegabriel;Max Mathews Hans Mikelson Peter Neubäcker Peter Nix Ville Pulkki Maurizio Umberto Puxeddu John Ramsdell Marc Resibois Rob Shaw Paris Smaragdis Greg Sullivan &nameistvan;Bill Verplank Robin Whittle &namesteven;&namepinot;
This list is by no means complete. More information can be gathered from the Changelog file in the manual's sources repository. manual_src~dfsg/preface/prefaceXO.xml0000644000000000000000000000243312262561504016752 0ustar rootroot
Preface to the OLPCsound Manual Welcome to the OLPCsound manual. Csound is a programmable computer music system. It was originally written by Barry Vercoe at the Massachusetts Institute of Technology in 1984 in the C programming language, based on earlier systems like Music IV by Max Mathews. Since then, Csound has received numerous contributions from researchers, programmers, and musicians from around the world, and is still being actively developed by a group of passionate programmers. Csound runs on many different computer platforms including Linux, Microsoft Windows and Apple Mac OS X. OLPCsound is the name of the special version of Csound optimized for the XO laptop built by the OLPC Foundation. This manual is released under the GNU Free Documentation License. Csound and OLPCsound are released under the GNU Lesser Public License.
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Toot 10 The following instrument uses the Csound rand opcode to produce noise, and a reson opcode to filter it. The bandwidth of reson will be set at i-time, but its center frequency will be swept via a line unit through a wide range of frequencies during each note. We add reverb as in Toot 9.
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Toot 7 Now we will add the ability to do a linear crossfade between any two of our four basic waveforms. We will employ our delayed vibrato scheme to regulate the speed of the crossfade. The schematic drawing below is the graphical representation of the connection of the opcodes in this tootorial. [TOOT 7 Schematic] TOOT 7 Schematic
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What's new in Csound &csoundversion; Release Notes for Csound 6.02 This new version has a large number of bug fixes (including clearing all general tickets on SourceForge). It also introduces some major new facilities such as use as a server, code to run Csound in a browser and a large generalisation of filter opcodes to have parameters changeable at audio rate. New opcodes: instance opcode schedules a new instrument instance, storing the instance handle. kill opcode stops a given instrument instance. strfromurl to set a string from a URL. Orchestra: If building supports it, a #include string can be a URL or a file. A space is again permitted between a function name and the opening bracket for all functions allowed in Csound5 (but not in general). The Csound command can start with an empty CSD in daemon mode (--daemon): do not exit if CSD/orchestra is not given, is empty or does not compile). Score: If building supports it, a #include string can be a URL or a file. Modified Opcodes and Gens: Many filters generalised to allow k- or a-rate parameters. In particular it includes these: areson atonex butterworth filters fofilter lowres lowresx lpf18 mode moogladder moogvcf reson resonr resonx resonz statevar tonex The maximum number of presets in sfont increased to 16384. cpsmidinn is now more accurate. max_k now behaves like the documentation. There were caseswhen it gave strange results. The vst4cs opcodes have been re-factored. FLTK code has been encapsulated. The build system has been updated for Csound 6. In alwayson opcode changes for better handling of pfields, more reliable insert of an instrument instance for repeating or re-started score sections. The signal flow graph opcodes have replaced OpenMP multi-threading with pthreads, using one-time initialization of static structures. Frontends: PNaCl is now supported as a platform, allowing Csound to run under the Chrome browser in all enabled operating systems. Bugs fixed: adsynt2 opcode fixed. ftgentmp opcode fixed. dates opcode fixed. fixed a bug in pvsfilter. fixed stereo out in temposcal and mincer. pan2 opcode fixed. index overflow in randh and randi fixed. A number of fixes to CsoundVST: initialization, score handling, and MIDI driver initialization, so it now works for Csound 6. fixed pycalln for no inputs. fixed/revised setting and use of ksmps and kr in UDOs. fixed problem in sending a score event from max to csound via csound~ (Ticket #58). If itype in chn_k was set to 3 and values are set less than 1, Csound6 used to give an INIT Error. (Ticket #67). A number of reported seg faults have been dealt with. xtratim opcode was using incorrect ekr value from csound instead of from instance; when used in conjunction with setksmps, was causing notes to have very long xtratim set and thus notes were effectively not getting turned off System changes: A server mode is now available, accepting input via UDP (with --port option). A longstanding bug in extract was detected and fixed. It does suggest that this facility is little used! The way the external score generator was coded is substantially changed. In particular this should fix a very strange bug in Windows. Fixed crashing bug with invalue channel callback due to wrong data object being pulled from csound host data. Fixed bug in UDOs with no local ksmps where kcounter was being used incorrectly. Better checking in channels. (Experimental) If the environment variable CS_UDO_DIR is set then any files in the directory that have an .udo extension are automatically included at the start of the orchestra. This facility needs review to see if it is what is required. (Experimental) There are new cuda GPGPU opcodes (source only): cudasynth (3 versions for additive synthesis, additive synthesis of fsigs and phase vocoder resynthesis) and cudanal (a GPGPU version of pvsanal). Internal changes: Many attempts at faster code. Type inference and parsing still improving. iOS: Fixed crash where no csoundSetHostImplementedMIDIIO is used on iOS and no _RTMIDI value is set. OSX: Fixed input device name for auhal. Release Notes for Csound6 Csound6 is a significant rewrite of much of the code. In particular the API is not compatible, although all orc/sco/csd works should still run. There are new facilities, like sample accuracy and realtime mode, described below. IMPORTANT: The environment variable to find plugins are called OPCODE6DIR64 or OPCODE6DIR (note the 6) so it can co-exist with Csound5. Similarly .csoundrc is renamed .csound6rc. Arrays are now mainstream, with syntax and opcode support. They also exist in multidimensional format. They are created (usually) with init opcode or fillarray. k1[] init 4 generates a k-rate 1-D array of length 4. Similarly a2[][] init 4, 4 creates a square 4x4 a-rate array. k2[] fillarray 1, 2, 3, 4 creates a 4-element vector filled with 1,..4, which also defines the length. Elements are used via indexing in [] such as k1[2] or a2[2][3]. One dimensional arrays replace tvars, and can be used in opcodes like maxtab, mintab and sumtab (see below). Array setting can be done in left-hand side of opcodes, i.e.: aSigs[0] vco2 .1, 440 aSigs[1] vco2 .1, 880 The new realtime priority mode can be switched on with by passing the --realtime or setting the CSOUND_PARAMS field realtime_mode to 1. This has the following effects: all opcode audio file reading/writing is handled asynchronously by a separate thread. all init-pass operations are also performed asynchronously. Multicore support is totally rewritten using a different algorithm for task-dispatch, which should use less memory and fewer locks. New opcodes: faustgen array -- many new or revised opcodes -- see Array Opcodes. compileorc takes a filename containing a collection of instrument definitions and compiles them, replacing existing versions. It returns 0 on success. compilestr is like compileorc but takes a string. readscore runs the score preprocessor on a string and then schedules new events via the RT event mechanism, returning 0 if successful. Orchestra Note events can start and end in mid-kcycle. As this is an incompatible change it is only invoked when the command-line option --sample-accurate is specified. Note that this does not work for tied notes, and use of skipping initialisation has questionable use. Instruments can run at local ksmps values using setksmps iksmps as in Csound 5 UDOs. Compilation can be done at any stage, new instruments are added or replace old ones. Running instances of old instrument definitions are not affected. Only limitation is that header constants in instr 0 are read only once at the time of the first compilation. Init-time code can be placed outside instruments in the global space, and this will be executed once-only following the compilation. In this case, score event generation can be completely replaced by orchestra code. See also new opcodes compileorc and compilestr. New syntax operators +=, -=, *= and /=. These are more than syntactic sugar; please use += and -= for accumulating reverbs as it gives better multicore behaviour. The opcodes add, sub, mul and div have been deleted; use the forms + - * /. Not many people were aware of these opcodes. Any opcode with a single output or with no outputs can be used as a function. Some opcodes might require type annotation to resolve ambiguities, more details on the Function syntax in Csound 6. A statement can be broken across lines after a , = or arithmetic operation. There are a range of new or recoded operations on k-valued arrays, most restricted to 1 dimensional arrays (vectors): kans minarray ktab returns the smallest value in the (possibly) multidimensional array kans maxarray ktab is like mintab kabs sumarray ktab returns sum of all values in the array ktab genarray imin, imax[, inc] generates vector of values from imin to imax by increments of inc (default 1) ktab2 maparray ktab1, "sin" maps the k-rate 1-arg function in the string to every element of the vector ktab2 maparray_i ktab1, "sin" maps the i-rate 1-arg function in the string to every element of the vector ktab2 slicearray ktab1, istart, iend returns a slice of ktab1 from ktab1[istart] to ktab1[iend] copyf2array ktab, kfn copies data from an ftable to a vector copya2ftab ktab, kfn copies data from a vector to an ftable Arithmetic on arrays is allowed. In particular addition, subtraction, multiplication, division on a element-by-element version is provided in arithmetic format. Similar operations between an array and a scalar are allowed. Each instance of any instrument has a scratchpad of 4 values that persist; allows values to carry to next use of the instrument; hope it may be useful in legato etc. If a table number is given as -1 then an internal sine wave equivalent to f. 0 16382 10 1 is used. Attempts to write to this table will give unpredictable results, but is not policed. The 16382 can be change by command line option --sine-size=# where the # is rounded up to a power of two. A number of oscil opcodes now have the f-table parameter as optional, defaulting to the internal sine wave. (oscil1, oscil1i, oscil, oscil3, oscili, foscil, foscil1, loscil, loscil3). Score: Score lines can have multiple strings. Change to escape characters in score strings -- they do not happen. Also note the readscore opcode. Modified Opcodes and Gens: The k() function can take an a-rate argument in which case it is a call to downsamp. Utilities Hetro/adsyn analysis files can be machine byte-order independent if created with -X. Down side is a longer file and a little slower loading. The het_export utility will create the independent format from the old, and het_import is no longer necessary. cvanal and lpanal will produce machine independent files if -X option is used. The convolve and lpread etc opcodes will accept either format. You are encouraged to use the machine independent form. Analysis files produced with -X can be used on other systems. Frontends Bugs fixed: System Changes: In Linux and OSX the treatment of locales is now thread-safe and local. Platform Changes: API: New API functions... new configuration/parameter setting functions PUBLIC int csoundSetOption(CSOUND *csound, char *option); PUBLIC void csoundSetParams(CSOUND *csound, CSOUND_PARAMS *p); PUBLIC void csoundGetParams(CSOUND *csound, CSOUND_PARAMS *p); PUBLIC void csoundSetOutput(CSOUND *csound, char *name, char *type, char *format); PUBLIC void csoundSetInput(CSOUND *csound, char *name); PUBLIC void csoundSetMIDIInput(CSOUND *csound, char *name); PUBLIC void csoundSetMIDIFileInput(CSOUND *csound, char *name); PUBLIC void csoundSetMIDIOutput(CSOUND *csound, char *name); PUBLIC void csoundSetMIDIFileOutput(CSOUND *csound, char *name); new parsing/compilation functions PUBLIC TREE *csoundParseOrc(CSOUND *csound, char *str); PUBLIC int csoundCompileTree(CSOUND *csound, TREE *root); PUBLIC int csoundCompileOrc(CSOUND *csound, const char *str); PUBLIC int csoundReadScore(CSOUND *csound, char *str); PUBLIC int csoundCompileArgs(CSOUND *, int argc, char **argv); new function for starting csound after first compilation PUBLIC int csoundStart(CSOUND *csound); new software bus threadsafe getters/setters PUBLIC MYFLT csoundGetControlChannel(CSOUND *csound, const char *name); PUBLIC void csoundSetControlChannel(CSOUND *csound, const char *name, MYFLT val); PUBLIC void csoundGetAudioChannel(CSOUND *csound, const char *name, MYFLT *samples); PUBLIC void csoundSetAudioChannel(CSOUND *csound, const char *name, MYFLT *samples); PUBLIC void csoundSetStringChannel(CSOUND *csound, const char *name, char *string); PUBLIC void csoundGetStringChannel(CSOUND *csound, const char *name, char *string); new table threadsafe copy functions PUBLIC void csoundTableCopyOut(CSOUND *csound, int table, MYFLT *dest); PUBLIC void csoundTableCopyIn(CSOUND *csound, int table, MYFLT *src); API has been made threadsafe so that performance and control can occur in separate threads (after a call to csoundStart() or csoundCompile()). Threadsafety is ensure by use of atomic read/writing to control channels spinlocks in audio and string channels mutexes protecting compilation, score events and table access. Internal: The build system is now cmake (and not scons as in Csound5). A number of table access opcodes have been rewritten but should behave the same. Similarly diskin and diskin2 now use the same code and so diskin should be more stable. The old parser is completely removed. New internal functions in Csound void (*FlushCircularBuffer)(CSOUND *, void *); void *(*FileOpenAsync)(CSOUND *, void *, int, const char *, void *, const char *, int, int, int); unsigned int (*ReadAsync)(CSOUND *, void *, MYFLT *, int); unsigned int (*WriteAsync)(CSOUND *, void *, MYFLT *, int); int (*FSeekAsync)(CSOUND *, void *, int, int); char *(*GetString)(CSOUND *, MYFLT); Extract a string originating from a score-event argument. Functions removed void *(*FileOpen)(CSOUND *, void*, int, const char*, void*, const char*); The "private" parts of the API have been changed considerably. Also structures like EVTBLK have changed. The LINKAGE1/FLINKAGE1 macros are renamed as LINKAGE_BUILTIN/FLINKAGE_BUILTIN. Template for a-rate perf-pass opcodes is int perf_myopcode(CSOUND *csound, MYOPCODE *p) { uint32_t offset = p->h.insdshead->ksmps_offset; uint32_t early = p->h.insdshead->ksmps_no_end; uint32_t nsmps = CS_KSMPS; ... if (UNLIKELY(offset)) memset(p->res, '\0', offset*sizeof(MYFLT)); if (UNLIKELY(early)) { nsmps -= early; memset(&p->res[nsmps], '\0', early*sizeof(MYFLT)); } for (n=offset; n<nsmps; n++) { ..... p->res[n] = .... } return OK; } String variables re-implemented OENTRY structure has changed and has a new dependency field; please use this field as it is required for multicore semantics. You could set it to -1 and disallow all parallelism, but at least it is safe. All opcodes that touch audio should take note of sample-accurate code. A number of previous API functions are removed; OpenFile and OpenFile2 both replaced by new OpenFile2 with additional argument. Additions have been made for arg type specifications for opcodes. Any-types have been added, as follows: '.' signifies a required arg of any-type '?' signifies an optional arg of any-type '*' signifies a var-arg list of any-type Arrays are now specified using "[x]" where x is a type-specifier. The type-specifier can be any of the of the current specifiers, including any-types. See Opcodes/arrays.c for example usage. New Type System A new type system has been added to Csound6, and significant changes have been made to the compiler. The previous system for handling types involved depending on the first-letter of a variable's name every time it was used to determine type. This meant there was a lot of re-checking of types. Also, adding new types was difficult, as there was a lot of custom code that had to be updated to check for new type letters. In Csound6, a separate system of types was added. Types are defined as CS_TYPE's. The creation of variables from types and the initialisation of memory has been encapsulated within the CS_TYPE's. This change allows easier addition of new types, as well as generic calculations of memory pools, amongst other things. The compiler has been modified since Csound5 to now use the type system as an integral part of its semantic checking phase. Variables are now registered into a CS_VAR_POOL when they are first defined, with the CS_VARIABLE having a reference to its CS_TYPE. After first time definition within the pool, the type information is then looked up in consequent variable lookups, rather than re-calculated from the variable name. This opens up possibilities for new variable naming and typing strategies, i.e. using "myVar:K" to denote a k-rate arg. This also opens up possibilities for user-defined types, such as "data myType kval, aval", then using "myVar:myType" to define a var of that type. (The previous is speculative, and is not an active proposal at this time.) The addition of the type system has formalised the static type system that has existed in Csound prior to Csound6. It has, arguably, simplified the code-base in terms of type handling, as well as laid the ground work for future type-related research to be integrated into Csound. New in Version 5.19 (2013 January 7) This is mainly a bug-fixing release but with a number of new opcodes and enhanced features. New opcodes: ipmidi module for MIDI over network. ppltrack opcode. combinv opcode. New Gen and Macros: Better checking in GEN28. Check range in outrg, and optionally allow wrapping. Orchestra: Change empty statement to a warning. Added line numbers to many input args message (new parser). Modified Opcodes and GENs: Better error and warning messages. loopseg now checks argument count. harmon2/3/4 improved. active: added the option to skip the instances in release phase. New and more tested implementation of ChordSpace. Bug fixed: Fix botched optimisation in lowpass filters. Chn opcodes fixed in Linux. Fix bug in loscil with silence. Correct GEN23 when comment does not end in newline. Correcting loopseg. Number of input and output channels fixed in new parser. Fixed GEN43 issue. Fixed fout. centroid was likely to crash. Minor bug in printing which lost %. An uninitialised value in fold fixed. Uninitalised values in dconv fixed. Assignment of fsigs now works. System Changes: Avoid seg fault on some user errors. Faster modal4 opcodes. Allow cabbage compilation. Made pfield size dynamic in event message csoundapi~. The default output format with pipe and double float is AU. Change to ircam with default format, '-o stdout' and pipe. Added double float precision for output format. Platform Changes: Linux: Spinlocks initialised (fixes bug in chn opcodes). OSX: Improved selection of devices in rtauhal module. Added a circular buffer interface and lock-free operation to rtauhal. Fixed MacOSX installer (creating symlinks to lib_csnd.dylib). Haiku: New platform Android: Using -B now in android to set circular buffer size. Added fluid synth opcodes for android. Added inputMessage method to CsoundObj. Allow CSDPlayer to be installed on SD Card. iOS: Improved audio routing. Bottom speaker the default for iOS. API: Added new API function csoundCompileFromStrings(). New in Version 5.18 (2012 August 29) This is mainly a bug-fixing release but with a number of new opcodes and enhanced features. New opcodes: centroid opcode like pvscent but acting on audio signals cosseg like linseg but with cosine interpolation cossegb like linsegb but with cosine interpolation cossegr like linsegr but with cosine interpolation joystick to read input values from an external joystick (Linux only) log2 function for logarithms base 2 platerev opcode to model a reverberating square plate pwd opcode to determine the current working directory readf opcode to read strings from a file readfi opcode to read strings from a file on initialisation vbap opcode like other vbap family but flexible about number of speakers and choice of layouts. vbapg opcode like vbap but only calculate the gains on the channels. New functionality Changes to <CsOptions> to allow spaces between words, and escaped characters. fout and fin use a better buffering strategy, and so are faster It is possible to specify just an orchestra with the --orc flag. This is useful when a score is not needed. A new command-line flag --ogg flag has been added for easy use of ogg/vorbis output. Added alsaseq real-time midi Bug fixes and improvements: dates opcode could crash on 64bit architecture; this is fixed Some multicore interlocks were wrong. It is believed that this was not actually a problem, but would be in the future. There were cases when a file was double closed, leading to a crash on exit. Two new features added in partikkel. Panning law for channelmasks can now be set using a function table (second optional argument to partikkel) and new support opcodes partikkelget and partikkelset, to access and modify the internal mask indices of partikkel. follow2 was reworked do the i-rate and k-rate calculations are the same. pvscent is corrected as it returned half the correct value. vbaplsinit can create more than one speaker layout which vbap/vbapg can use. Also much better diagnostics on incorrect layouts. Internal Changes: Code changed so bison 2.6 can be used. It is assumed that libsndfile version 1.0.19 or later is available. If the score is omitted a near-infinite wait is generated. New in Version 5.17 (March 2012) This is mainly a bug-fixing release with no major changes, but the number of fixes warrants a release. New opcodes: cell opcode, for cellular automata Modified Opcodes and Gens: active now will report total number of active or allocated instruments if argument in zero stsend and strecv the TCP socket opcodes reworked to a logical design DSSI system now will take up to 9 channels FLsavesnap works with other widgets where imin > imax Utilities: csbeats better documented and built by default; also more note lengths available Some security holes in utilities fixed Bug fixes: unirand opcode at a-rate fixed Locale fix for floating point literals in orchestra transegr fixed System Changes: Score can now last longer (change to size of time variable) An empty score gives a very long performance time (years and years) Android code released Changes to use of tmp files; now all are deleted at end of run (previously some were left) and the environment variable TMPDIR is used. interaction between Comments, end of line and end of file fixed Hexadecimal numbers now allowed in orchestra Empty orchestra now not a crash change to macro expansion inside a string avoid infinite loop when eof in malformed score macro fixed macroname-with-args diagnostics and memory leak change to preprocessor: {{ }} inside "..." and better diagnostics fix windows installer so it removes full $INSTDIR\bin from PATH during uninstall: this cleans up the PATH environment variable when uninstalling on Windows. Previously, it was leaving a trailing "\bin" on the PATH. CsoundAC MusicModel class more usable by C++ programs ftcps had been missed as a function Internal Changes: Many! Some messages quietened, code improvements etc New in Version 5.16 (February 2012) The major change is that the new parser is now the default. The old parser is still available in case of difficulty but the new has been given extensive testing since the start of the year, including complete restructuring of macro expansion. A side effect is that the runtime of most orchestras is faster, although parsing is slower. There are a few optimisations implemented like constant folding in simple cases. Line numbers and file names are traced better than before. Some memory leaks also fixed. New opcodes: Opcodes adapted from SuperCollider by &nametito;: dust, dust2, gausstrig, gendy, gendyc, and gendyx. Fractal noise generator by &nametito;: fractalnoise. Opcodes for accessing table values by direct indexing, by &namejohn;: ptable, ptablei, ptable3, and ptablew. These opcodes are respectively like table, tablei, table3, and tablew, but they do not require a power-of-2 table size. Modified Opcodes and Gens: There was a fence post problem in tab opcode that could falsely report a reference out of range. GEN15 mis-called gens 13 and 14 internally, using uninitialised values voice amplitude. Problem fixed. fmbell now takes an optional argument to control the sustain time. Change to pvsbasic for tab to table conversions. poscil is now polymorphic, allowing k- or a-rate amplitude and frequency. p() and i() changed when argument at k-rate. gen49 deferred now works. gen23 now available deferred. Utilities: Checked for use with the new parser in memory files. Frontends: Table access added to csoundapi~ via new get/set methods. Bug fixes and improvements: Many in new parser related to precedence and multicore. Better diagnostics when orchestra file/csd is missing. csd file: fix CsFileB and CsSampleB. Fixed score statement 'n'. Fixed bug in diskin2 leading to infinite loop. Fixed bug causing crossfade noise in hrtfmove. Fixed unlikely buffer overflows in some utilities. Avoid segfault in midicN. Bug in mp3in in skip=0 case fixed. 'r' score statement fixed with respect to macros. sndwarp could segfault. System Changes: Preprocessor #if #else #endin working. #includes depth now limited rather than infinite recursion. Really turn off all displays if --nodisplays or -d is used; fixes bug where using -d or --nodisplays would still cause the winFLTK.c csoundModuleInit to setup display callbacks; bug caused with python TK apps and CsoundYield_FLTK being called. Memory leak in mp3in and mp3len fixed. Internal Changes: Very, very, very many! And the new parser... New in Version 5.15 (December 2011) New opcodes: ftab2tab opcode. tab2pvs opcode. pvs2tab opcode. cpumeter opcode, (not really new but now available in OSX) minmax opcode. (EXPERIMENTAL) ftresize opcode. (EXPERIMENTAL) ftresizei opcode. hrtfearly opcode. hrtfreverb opcode. New Gen and Macros Code to allow GEN49 to be deferred [NB does not seem to work] Modified Opcodes and Gens socksend and sockrecv no longer uses MTFU check and work on Windows mpulse changed so if next event is at negative time use the absolute value serial opcode now runs on Windows as will as Un*x out, out2, outq, outh, outo outx and out32 are now identical opcodes and will take up to as many arguments as nchnls. This replaces the current remapping of opcodes turnoff2 now polymorphic wrt S and k types (ie accepts instrumnet names) Bugs fixed: GEN42 fixed jacko: fixed a segfault removing the unused JackSessionID option doppler memory leak fixed transegr fixed in release mode when skipping most of envelope FLPack now agrees with manual max_k now agrees with manual hrtfreverb fixed atsa code now works on Windows in more cases tabmorph bug fixed fixed problem with user-defined opcodes having no outputs Various fixes to * ... */ comments System Changes: Various licence issues sorted Loris is no longer part of the Csound tree Memory leaks fixed If no score is given a dummy that runs for over 100 years is created All score processing takes place in memory without temporary files String memory now expandable and no size limitation #if #else #end now in new parser Adjustments to MIDI file precision in output On OSX move from Coreaudio to AuHAL Multicore now safe for ZAK, Channels and modifying tables New coremidi module Virtual Keyboard improved: 1) Dropdown for choosing base octave (the one that starts with the virtual key mapped to physical key Z). Default value is 5 which is backwards compatible. 2) Shift-X mappings which add two octaves to X mappings for a total of 4 octaves playable from the physical keyboard (starting from selected base octave). 3) Control-N / Control-Shift-N mappings to increment / decrement slider for control N. 4) Mouse wheel now controls sliders. tsig type for vectors tsigs and fsigs allowed as arguments in UDOs API: Minor version upped Internal Changes: Very, very, very many! New in Version 5.14 (October 2011) New opcodes: mp3len opcode. qnan opcode. qinf opcode. exprandi opcode. cauchyi opcode. gaussi opcode. cpumeter opcode. linsegb opcode. expsegb opcode. transegb opcode. expsegba opcode. pvsgain opcode. pvsbufread2 opcode. serial opcodes. lua opcodes opcodes. plustab opcode. multtab opcode. maxarray opcode. minarray opcode. sumarray opcode. scalearray opcode. New functionality beats processor renamed to csbeats and distributed mkdb utility to provide a catalogue of plugin libraries/opcodes ladspa library build in default system macros are now expanded inside string in the score there in an until .. do .. od looping syntax (in the new parser only) SIGPIPE signals are ignored rather than causing Csound to exit It is possible to use vectors of k-rate values, named t-variable. They are initialised to a fixed sizw with init adncan be read with a simple [] syntax. assignment to elements is only via =. There are also a few new opcodes that provide wider functionality. Bug fixes and improvements: reading values to fill tables was broken with respect to comments internal error in wii_data fixed pvsshift fixed jacko fixed gen23 minor fixes wiimote fixed atsaadd fixed compress fixed to work with 0dbfs pvsbufread corrected with respect to position counting tempo opcode fixed CsFileB section in .csd files had a bug, now fixed deferred gen01 tables could have wrong size vbap_zak made to work(!) fixed memory issue in ATSsinoi various fixes to cscore various fixes to partials and tradsyn transegr could crash in some cases loris opcodes updated to latest version date opcode has new base in some platforms to avoid overflow pvsblur now works over reinit diskin, diskin2 and soundin now can read up to 40 channels prints behaves better with rounding fmpercfl now has working vibrato atreson now has gain parameter at k-rate comb opcode made safe if in and out arguments the same better accuracy in line and expon OSCsend recovers space previously lost OSCsend can send a table as a blob with the T tag -- experimental and untested. lpf18 now has an optional iskip argument i() will also accept an i-rate value in which case it is a no-op makecsd revised and extended to have options for MIDI and score processing and licenses lpanal reworked to remove bugs and oddities an issue with noise in alsa fixed and a click in portaudio fixed portaudio driver changed to be more robust on stop/exit Internal Changes: Many many changesto the new parser so it is now operational, but should be used with care The multicore system is distributed in an experimental mode and should be used with great care. New in Version 5.13 (January 2011) New opcodes: median opcode. filevalid opcode. pvstanal, pvswarp, temposcal, pvslock spectral processing opcodes. mincer opcode fareylen sequence opcodes. New functionality Real random number generators using /dev/random (Linux only). INF macro added to orchestras; z read as infinity in scores init changed to allow multiple inits in on statement GEN for support of farey sequences maxalloc,cpuprc, active now accept named instruments. If normalisation in pow opcodes is zero treat as 1 inch can take upto 20 inputs and outputs. pvscale, pvsvoc and pvsmix now have very good spectral envelope preservation modes (1 = filtered cepstrum, 2 = true envelope). oscil1 could be static if the duration was long; now there is a positive minimum increment. GEN49 now uses search paths. Bug fixes and improvements: Count of lines fixed in orchestras, and \ inside strings Fast tab opcodes made safe from crashes % in formated printing could crash Double free in fgen fixed sndwarp quietened (gave too many messages) gen41 deals with positive probabilities adsynt reworked removing many bugs adsynt2 phase error fixed Bug in max number of gens fixed Better checking in grain4 Better checking in adsyn modulus was wrong in new parser atonex/tonex did wrong operation mp3in could repeat sound at end of file changed opcode initialised to zero Serious bug in tabmorpha fixed GEN49 has serious bug removed, so no longer incorrect silences. partikkel opcode: fixed bug in sub-sample grain placement when using grain rate FM Internal Changes: In the new parser only there are operator @ and @@ to round up the next integer to a power of 2 or powerof2+1 Score sorting made much faster lineto improved Named gens allowed Various printing include instrument name if available Command option to omit loading a library Number of out channels no longer constrained to be number of in Many fixes to new parser More use of Warnings than Messages (allows for them to be switched off) csoundSetMessageCallback reset if callback set to null New in Version 5.12 (January 2010) New opcodes: transegr is a version of the transeg opcode which has a release section which is triggered by midi, a turnoff2 opcode or a negative instrument number i score event. ftgenonce generates a function table from within an instrument definition, without duplication of data. passign allows quick initialization of i-rate variables from p-fields crossfm implements crossed fm synthesis. loopxseg is like loopseg but with exponential envelope. looptseg is like loopseg but with a flexible envelope like transeg Bug fixes and improvements: pvshift would overwrite in double mode. pan2 case 3 fixed. clockon and clockoff now work again. cross2 and interp could have divided by zero linecount for error messages no longer includes text from .csoundrc p5gconnect changed to use a separate thread to avoid timeout problem. transeg checks argument count. sfload used to be limited to 10 sound fonts and was not policed. Now open-ended. Internal Changes: \" allowed as an escape in orchestral strings New parser fixed on optional arguments Better checking of f statement with negative number Soundfonts only initialise pitches array once, in the soundfont opcodes. Usual collection of gratuitous minor changes, layout and comments New in Version 5.11 (June 2009) New opcodes: mp3in allows reading of mp3 files directly in the orchestra. wiiconnect, wiidata, wiisend, wiirange opcodes by john ffitch to recieve and send data to a wiimote controller. New opcodes to receive data directly from a p5glove by john ffitch p5gdata tabsum sums sections of ftables MixerSetLevel_i an init-time only version of MixerSetLevel doppler implements a simulation of the doppler effect. filebit reports the file depth of a file. The new Signal Flow opcodes enable the usage of signal flow graphs in Csound. New functionality New panning type for pan2 opcode New csd score tag <CsExScore>. New -Ma option for ALSA RT MIDI module which listens to all devices. There is a gen49 to read mp3 files Added rounding bin code to pvscale Added non-power-of-2 table support for ftload and ftsave GEN23 totally rewritten to be more consistent in what constitutes a separator and comments. (Still no /* */ comments) Bug fixes and improvements: New examples for pvs opcodes by Joachim Heintz: pvsarp, pvscent, pvsbandp, pvsbandr, pvsbufread, pvsadsyn, pvsynth, pvsblur, pvscale, pvscross, pvsfilter, pvsfreeze, pvshift, pvsmaska, pvsmorph Use of automatic numbering of ftables reuses table numbers seed with positive argument was wrong sprintf with an empty string printed wrong data mute now works with both numeric and named instruments Small fixes in diskin, and in tablexkt Internal Changes: SConstruct now builds completely independent shared libraries for Python, Lua, and Java wrappers. New Parser almost usable Redrawing of graphs fixed so that only selected ones get redrawn. RT-alsa more forgiving on near sample rates It is possible to have the score generated by an external program rather than using standard score format using <CScore bin="translater"> to call the program translater on the score data lpc_export fixed Removed limit on macro names length PMAX, the number of arguments to a score event has been reduced by 2, and an overflow system introduced so GENs can have arbitrary numbers of arguments. Increased API version to 2.1. New API function pointer ldmemfile2withCB() which is a version of ldmemfile() allowing a callback to be set and called exactly once to process the MEMFIL buffer after it is loaded. csound->floatsize set; zero in earlier versions GetChannelLock added New in Version 5.10 (December 2008) New functionality New option to listen to all MIDI devices using the portmidi realtime module. To enable listening to all devices use "-+rtmidi=portmidi -Ma". Dither on output implemented; rectangular and triangular dither available in some cases GEN20 type 6 now has option to set variance Bug fixes and improvements: Locale set to C numeric to avoid , versus . problems. diskin fixed outo was broken regarding channel 6 pitchamdf fixed zfilter2 intialization fixed s32b14 fixed Fixed other bugs fixed that have not been reported publicly. Internal Changes: The major version of the Csound API is increased to 2; affected csound.so as well. This means that Csound 5.10 is incompatible with applications ("front ends", "clients", or "hosts") that were built for Csound 5.08 and earlier and that use API version 1.x. These applications will need to be rebuilt to work with the current and future versions of Csound. Csound front ends written in interpreted languages such as Python or Java may continue to work without modification. It may also be possible to keep both an earlier version of the Csound library and an API 2.0 version on the same machine together so that new and old Csound-based applications can run side-by-side. These changes do not in any way affect the compatibility of Csound orchestras and scores: all old documents should continue to work as before. Time now counted internally in samples, overcoming a longstanding bug with rounding of time to k-rate. Many internal changes related to branch prediction. Some opcodes are substantially quicker. New in Version 5.09 (October 2008) New opcodes: New vosim opcode by Rasmus Ekman which recreates the historic VOSIM (VOcal SIMulator) technique. New dcblock2 opcode by &namevictor;. New Chua's oscillator model: chuap by Michael Gogins. New Linear Algebra opcodes by Michael Gogins. Standard Linear algebra over real and complex vectors and matrices: elementwise arithmetic, norms, transpose and conjugate, inner products, matrix inverse, LU decomposition, QR decomposition, and QR-based eigenvalue decomposition. Includes copying vectors to and from a-rate signals, function tables, and f-signals. New ambisonic opcodes: bformdec1 and bformenc1. These opcodes deprecate the older bformdec and bformenc. New Score control opcodes by &namevictor;: rewindscore and setscorepos. New functionality: The vbap family of opcodes (vbap4, vbap8, vbap16 and vbapz) now accept k-rate variables for all their input arguments. New pulseaudio I/O module on Linux. New optional ienv parameter to generate envelopes for the soundfont opcodes: sfplay, sfplay3, sfplaym and sfplay3m. Added 'skip normalisation argument' to "tanh" named GEN routine. (See Named GEN Routines) Added scheduler priority option on alsa. Bug fixes and improvements: Allow scientific notation (as was in csound4!) in GEN23. Fixed bug in FLTK initialization. Should make FLTK usage more stable. Error on /* */ comments in orchestra fixed. poscil no longer overwrites frequency if variable is shared. printk and printks check that opcode is initialised. Deprecate soundout and soundouts in favour of fout. Fixed space opcode to accept non-pow-2 (deferred) tables. Fixed pvsmorph bug. Internal Changes: New parser has #include and argumentless macros. Less casting between floats and doubles in float version. Includes experimental multicore support. buzz opcode rewritten. Many other internal changes and small bug fixes. New in Version 5.08 (February 2008) New opcodes: imagecreate, imagesize, imagegetpixel, imagesetpixel, imagesave, imageload and imagefree: New image file processing opcodes by Cesare Marilungo to read/write png images from Csound. pvsbandp and pvsbandr by John ffitch, which perform band-pass and band-reject filtering in the spectral domain on a pvs signal. New HRTF opcodes by Brian Carty:hrtfmove, hrtfmove2 and hrtfstat. New waveshaping opcodes: powershape, polynomial, chebyshevpoly, pdclip, pdhalf, pdhalfy, and syncphasor New jack transport control opcode: jacktransport New functionality Added --csd-line-nums= command line option to select mode for error line reporting. New "no-carry" operator (! ) for score language that prevents implicit carrying of p-fields in i-statements. Added --syntax-check-only commandline flag (exclusive with --i-only) <CsLicence> tag for CSDs. <CsLicense> is accepted as an alternative to <CsLicence>. Bug fixes and improvements: Changed order of outputs for hilbert. This change breaks compatibility with previous versions, but fixes the opcode and now works as documented. Messages about loading opcode plugins modified so can be suppressed with message level flag. Major changes to score error reporting; now accurately reports the line numbers for the chain of inputs for most errors. Corrected pan2 so it agrees with documentation. <CsVersion> tag works again according to the manual. Fixed the { and } score looping statements. Added missing documentation for them and ~, &, |, and # operators in score expressions. hilbert had its outputs reversed, now correct. Manual example updated. Internal Changes: Change to gettext localisation; French and Columbian-Spanish translations available. Internal changes to partikkel, interpolation of waveform read and windowing, allowing more precise pitch synchronous granular synthesis. Updated examples for partikkel. pvscale: Improved algorithm for SDFT case so no ampltitude variation. New in Version 5.07 (October 2007) New opcodes: pan2: a stereo panning opcode cpsmidinn, pchmidinn, octmidinn: converters for MIDI note numbers fluidSetInterpMethod: interpolation in fluid sound fonts sflooper: a soundfont version of flooper2 pvsbuffer and pvsbufread: buffering/reading of fsigs for delays/timescale changes. New functionality SDFT - the Sliding Discrete Fourier Transform -- added seamlessly to pvsanal, etc opcodes if the overlap is less than the ksmps or less than 10. Some pvsXXX opcodes extended to take a-rate parameters when sliding. New feature (-O null / --logfile=null) that disables all messages and printing to the console. Bug fixes and improvements: partikkel -- particle synthesis had an inadvertent bug, now fixed. Closing of MIDI input on Windows(MM) failed; now fixed fluidEngine opcode now takes optional number of channels (range 16-256, default to 256) and polyphony (range 16-4096, default to 4096) to use. atsa utility safer when given silence. ATSaddnz: improved checking. Ambisonics (bformdec, bformenc) has more options for controlled opposites. Bug in turnoff2 fixed. het_export: invalid check caused export to fail. Internal Changes: Improved Windows installer. CsoundVST replaced by CsoundAC, that does not depend on the VST SDK headers. Less messages in Windows(MM) startup. P argument type added (k-rate defaults to 1) for opcode in and out types. New in Version 5.06 (June 2007) New granular opcodes: partikkel, partikkelsync and diskgrain. New opcode for event dispatch: scoreline. Many new opcodes from Gabriel Maldonado's CsoundAV: hvs1, hvs2, hvs3, vphaseseg, inrg, outrg, lposcila, lposcilsa, lposcilsa2, tabmorph, tabmorpha, tabmorphi, tabmorphak, trandom, vtable1k, slider8table, slider16table, slider32table, slider64table, slider8tablef, slider16tablef, slider32tablef, slider64tablef, sliderKawai and the a-rate version of ctrl7. Also from CsoundAV, many new FLTK widget opcodes: FLkeyIn, FLslidBnk2, FLvslidBnk, FLvslidBnk2, FLmouse, FLxyin, FLhvsBox, FLslidBnkSet, FLslidBnkSetk, FLslidBnk2Set, FLslidBnk2Setk, FLslidBnkGetHandle, New pvs opcodes: pvsdiskin, pvsmorph, eqfil New command line options (--m-warnings)to control messages csladspa: a CSD to LADSPA plugin kit. And many bug fixes including (but not limited to): fixed k-rate version of system; fixed scaling problems of vrandh and vrandi; fixed ocasional failure of turnoff; fixed OS X bug; fixed ATScross and fixed mod. Csound5GUI now works properly on all platforms and csoundapi~ (pd object) has been updated.
manual_src~dfsg/preface/top.xml0000644000000000000000000000042212262561504015674 0ustar rootroot Preface &pfacepreface; &pfacehistory; &pfacecopy; &pfacegettingstarted; &pfacewhatsnew; manual_src~dfsg/preface/toot4XO.xml0000644000000000000000000000211012262561504016406 0ustar rootroot
Toot 4 Next we'll animate the basic sound by mixing it with two slightly de-tuned copies of itself. We'll employ Csound's cpspch value converter which will allow us to specify the pitches by octave and pitch-class rather than by frequency, and we'll use the ampdb converter to specify loudness in decibels rather than linearly. Since we are adding the outputs of three oscillators, each with the same amplitude envelope, we'll scale the amplitude before we mix them. Both iscale and inote are arbitrary names to make the design a bit easier to read. Each is an i-rate variable, evaluated when the instrument is initialized. Here's the file toot4.csd:
manual_src~dfsg/preface/toot3XO.xml0000644000000000000000000000440412262561504016415 0ustar rootroot
Toot 3 Although in the second instrument we could control and vary the overall amplitude from note to note, it would be more musical if we could contour the loudness during the course of each note. To do this we'll need to employ an additional opcode linen, which is used as follows: kr linen kamp, irise, idur, idec ar linen xamp, irise, idur, idec linen produces a set of values which go linearly from 0 to kamp in the time (in seconds) specified by irise. It will stay there for some time and then it will return to 0 again taking idec time. The output of linen (k1) is patched into the kamp argument of an oscil. This applies an envelope to the oscil. The csd appears in file toot3.csd. [Envelope generated by the linen opcode] Envelope generated by the linen opcode
manual_src~dfsg/preface/toot6XO.xml0000644000000000000000000000654612262561504016431 0ustar rootroot
Toot 6 The first character in a score statement is an opcode, determining an action request; the remaining data consists of numeric parameter fields (p-fields) to be used by that action. So far we have been dealing with two different opcodes in our score: f and i. i statements, or note statements, invoke the p1 instrument at time p2 and turn it off after p3 seconds; all remaining p-fields are passed to the instrument. On the other hand, f statements, or lines with an opcode of f, invoke function-drawing subroutines called GENS. In Csound there are many GENS which fill tables in a variety of ways. For example, GEN01 transfers data from a soundfile; GEN07 allows you to construct functions from segments of straight lines; and GEN10, which we've been using in our scores so far, generates composite waveforms made up of a weighted sum of simple sinusoids. We have named the function "f1," invoked it at time 0, defined it to contain 512 points, and instructed GEN10 to fill that table with a single sinusoid whose amplitude is 1. GEN10 can in fact be used to approximate a variety of other waveforms, as illustrated by the following: f 1 0 2048 10 1 ; Sine f 2 0 2048 10 1 0.5 0.3 0.25 0.2 0.167 0.14 0.125 .111 ; Sawtooth f 3 0 2048 10 1 0 0.3 0 0.2 0 0.14 0 .111 ; Square f 4 0 2048 10 1 1 1 1 0.7 0.5 0.3 0.1 ; Pulse For the opcode f, the first four p-fields are interpreted as follows: p1 - table number - In the orchestra, you reference this table by its number. p2 - creation time - The time at which the function is generated. p3 - table size - Number of points in table - must be a power of 2, or that plus 1. p4 - generating subroutine - Which of the GEN soubroutine will you employ. p5 - meaning determined by the particular GEN subroutine. In the instrument and score below, we have added three additional functions to the score, and modified the orchestra so that the instrument can call them via p11. The csd file is toot6.csd.
manual_src~dfsg/preface/tootorialsXO.xml0000644000000000000000000000064112262561504017543 0ustar rootroot
Tootorials Learning Csound takes time and patience. In this section you will find a set of "tootorials" which will introduce you to the capabilities and the processes of working with Csound. &pfacetoot1; &pfacetoot2; &pfacetoot3; &pfacetoot4; &pfacetoot5; &pfacetoot6; &pfacetoot7; &pfacetoot8; &pfacetoot9; &pfacetoot10;
manual_src~dfsg/preface/toot8XO.xml0000644000000000000000000000457512262561504016433 0ustar rootroot
Toot 8 Now instead of continuing to enhance the same instrument, we will design a totally different one. In it we'll play a soundfile from disk in the orchestra, apply an amplitude envelope to it, and add some reverb. To do this we will employ Csound's soundin and reverb generators. The first is used like this: a1 soundin ifilcod[, iskiptime[, iformat]] soundin derives its signal from a pre-existing file. ifilcod is either the filename in double quotes, or an integer suffix (.n) to the name "soundin". Thus the file soundin.5 could be referenced either by the quoted name or by the integer 5. To read from 500ms into this file we might say (i.e. to skip the first 500ms): a1 soundin "soundin.5", .5 The Csound reverb generator is actually composed of four parallel comb filters plus two alpass filters in series. Although we could design a variant of our own using these same primitives, the preset reverb is convenient, and simulates a natural room response via internal parameter values. Only two arguments are required the input (asig) and the reverb time (krvt) ar reverb asig, krvt Here's a schematic representation of the instrument. [TOOT 8 Schematic] TOOT 8 Schematic The soundfile instrument with artificial envelope and a reverb (included directly) is as follows:
manual_src~dfsg/preface/history.xml0000644000000000000000000001020012262561504016566 0ustar rootroot
History of the Canonical Csound Reference Manual This initial version of this manual for early versions of Csound was started at MIT by &namebarry; and maintained there during the 1980's and start of the 1990's. Some of the manual comes from documents for programs like Music11 from the 1970's. This original manual was improved and worked on by Richard Boulanger, &namejohn;, Jean Piché and &namerasmus;. This manual led to the Official Csound Reference Manual, still located at: http://www.lakewoodsound.com/csound, for Csound version 4.16, November, 1999, which was maintained by &namedavid;. A parallel version of the manual called the Alternative Csound Reference Manual, was developed by &namekevin; using DocBook/SGML. This version later became the Canonical version. When Csound was licenced as LGPL by MIT in 2003, the manual was licenced GFDL and placed on Sourceforge along with the sources of Csound. In the winter of 2004, the Canonical Manual was converted to DocBook/XML by &namesteven; to allow for more people to be able to compile and maintain the manual. The manual is currently maintained by &nameandres; with continuous contributions from the Csound Community. The manual continues to be a community run project that depends on the contributions of developers and users to help refine the coverage and accuracy of its contents. All contributions are welcome and appreciated. Other Contributors Mike Berry Eli Breder Michael Casey Michael Clark Perry Cook Sean Costello Richard Dobson Mark Dolson Dan Ellis Tom Erbe Bill Gardner &namemichael;Matt Ingalls Richard Karpen &nameanthony;&namevictor; Allan LeeDavid Macintyre &namegabriel;Max Mathews Hans Mikelson Peter Neubäcker Peter Nix Ville Pulkki Maurizio Umberto Puxeddu John Ramsdell Marc Resibois Rob Shaw Paris Smaragdis Greg Sullivan &nameistvan;Bill Verplank Robin Whittle &namesteven;&namepinot;&nameandres;Gareth EdwardsJoachim Heintz&namejohn;Oeyvind BrandtseggMenno KnevelFelipe SatelerAnd many others.
This list is by no means complete. More information can be gathered from the Changelog file in the manual's sources repository.
manual_src~dfsg/preface/toot5XO.xml0000644000000000000000000000260112262561504016414 0ustar rootroot
Toot 5 To add some delayed vibrato to our chorusing instrument we use another oscillator for the vibrato and a line segment generator, linseg, as a means of controlling the delay. linseg is a k-rate or a-rate signal generator which traces a series of straight line segments between any number of specified points. It is used like this: kr linseg ia, idur1, ib[, idur2, ic[...]] ar linseg ia, idur1, ib[, idur2, ic[...]] The variables: ia, ib, ic, etc., are the values for the points. The i-rate variables: idur1, idur2, idur3, etc., set the duration, in seconds, between segments. Here's the file toot5.csd:
manual_src~dfsg/preface/toot9XO.xml0000644000000000000000000000473412262561504016431 0ustar rootroot
Toot 9 In the previous example you may have noticed the soundin source being "cut off" at ends of notes, because the reverb was inside the instrument itself. It is better to create a companion instrument, a global reverb instrument, to which the source signal can be sent. Let's also make this stereo (two channel). Variables are named cells which store numbers. In Csound, they can be either local or global, are available continuously, and can be updated at one of four rates - setup, i-rate, k-rate, or a-rate. Local variables (which begin with the letters p, i, k, or a) are private to a particular instrument. They cannot be read from, or written to, by any other instrument or by other notes from the same instrument. Global Variables are cells which are accessible by all instruments. Three of the same four variable types are supported (i, k, and a), but these letters are preceded by the letter "g" to identify them as "global." Global variables are used for "broadcasting" general values, for communicating between instruments, and for sending sound from one instrument to another. The reverb instr 99 below, receives input from instr 9 via the global a-rate variable garvbsig. Since instr 9 adds into this global, several copies of instr 9 can do this without losing any data. The addition requires garvbsig to be cleared before each k-rate pass through any active instruments. This is accomplished first with an init statement in the orchestra header, giving the reverb instrument a higher number than any other (instruments are performed in numerical order), and then clearing garvbsig within instr 99 once its data has been placed into the reverb. In the score we turn the global reverb on at time 0 and keep it on until irvbtime after the last note. [TOOT 9 Schematic] TOOT 9 Schematic
manual_src~dfsg/preface/copy.xml0000644000000000000000000001226012262561504016047 0ustar rootroot
Copyright Notice This version of the Csound Manual ("The Canonical Csound Manual") is released under the GNU Free Documentation Licence. Below are listed, for historical purposes, previous copyrights and requests for credit from previous authors. Previous copyright notices Copyright (c) 1986, 1992 by the Massachusetts Institute of Technology. All rights reserved. Developed by &namebarry; at the Experimental Music Studio, Media Laboratory, M.I.T., Cambridge, Massachusetts, with partial support from the System Development Foundation and from National Science Foundation Grant # IRI-8704665. Manual Copyright (c) 2003 by &namekevin; for modifications made to the Public Csound Reference Manual. Permission is granted to copy, distribute and/or modify this document under the terms of the GNU Free Documentation License, Version 1.2 or any later version published by the Free Software Foundation; with no Invariant Sections, no Front-Cover Texts, and no Back-Cover Texts. A copy of this license is available in the examples sub-directory or at: www.gnu.org/licenses/fdl.txt. This Csound language documentation in this manual is derived from Kevin Conder's Alternative Csound Reference Manual, which in turn is derived from the Public Csound Reference Manual. Copyright 2004-2005 by Michael Gogins for modifications made to the Alternative Csound Reference Manual. This legal notice is from the Public Csound Reference Manual: The original Hypertext Edition of the MIT Csound Manual was prepared for the World Wide Web by Peter J. Nix of the Department of Music at the University of Leeds and Jean Piché of the Faculté de musique de l'Université de Montréal. A Print Edition, in Adobe Acrobat format, was then maintained by &namedavid;. The editors fully acknowledge the rights of the authors of the original documentation and programs, as set out above, and further request that this notice appear wherever this material is held. The Public Csound Reference Manual's last known network location was http://www.lakewoodsound.com/csound/hypertext/manual.htm. The Alternative Csound Reference Manual's network location, for both the Transparent and Opaque copies, is http://kevindumpscore.com/download.html#csound-manual. The Csound and CsoundAC Manual's network location is http://sourceforge.net/projects/csound. Csound and CsoundAC Csound is copyright 1991-2008 by Barry Vercoe, John ffitch and others. CsoundAC is copyright 2001-2008 by Michael Gogins. Csound and CsoundAC (formerly CsoundVST) are free software; you can redistribute them and/or modify them under the terms of the GNU Lesser General Public License as published by the Free Software Foundation; either version 2.1 of the License, or (at your option) any later version. Csound and CsoundAC are distributed in the hope that they will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for more details. You should have received a copy of the GNU Lesser General Public License along with Csound and CsoundAC; if not, write to the Free Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. Virtual Synthesis Technology Virtual Synthesis Technology (VST) PlugIn interface technology by Steinberg Soft- und Hardware GmbH.
manual_src~dfsg/control/0000755000000000000000000000000012262561504014425 5ustar rootrootmanual_src~dfsg/control/clockctl.xml0000644000000000000000000000131712262561504016747 0ustar rootroot
Clock Control The opcodes to start and stop internal clocks are: clockoff clockon These clocks count CPU time. There are 32 independent clocks available. You can use the opcode readclock to read current values of a clock. See Time Reading for other timing opcodes.
manual_src~dfsg/control/timeread.xml0000644000000000000000000000254612262561504016750 0ustar rootroot
Time Reading Opcodes one can use to read time values are: readclock rtclock timeinstk timeinsts times timek You can obtain the system date using: date - Returns the number seconds since 1 January 1970. dates - Returns as a string the date and time specified. You can also set up counters using clockoff and clockon.
manual_src~dfsg/control/fltkvaluators.xml0000644000000000000000000000147612262561504020060 0ustar rootroot
FLTK Valuators The opcodes for FLTK valuators are: FLcount FLjoy FLknob FLroller FLslider FLtext
manual_src~dfsg/control/reinitn.xml0000644000000000000000000000321112262561504016614 0ustar rootroot
Initialization and Reinitialization Opcodes used for the initialization of variables: init tival = passign pset The opcodes that can generate another initialization pass are: reinit rigoto rireturn The opcode p can be used to find score p-fields at i- or k-rate. nstrnum returns the instrument number for a named instrument. Note that a instrument may modify the p3 (duration) parameter at initialisation time. For example statements like iattack = 0.02 irelease = 0.04 isustain = p3 p3 = iattack + isustain + irelease are valid.
manual_src~dfsg/control/fltkcontainers.xml0000644000000000000000000000242312262561504020176 0ustar rootroot
FLTK Containers The opcodes for FLTK containers are: FLgroup FLgroupEnd FLpack FLpackEnd FLpanel FLpanelEnd FLscroll FLscrollEnd FLtabs FLtabsEnd
manual_src~dfsg/control/durctl.xml0000644000000000000000000000153412262561504016447 0ustar rootroot
Duration Control Statements The opcodes one can use to manipulate a note's duration are: ihold turnoff turnoff2 turnon For other realtime instrument control see Real-time Performance Control and Instrument Invocation.
manual_src~dfsg/control/pgmctl.xml0000644000000000000000000000435512262561504016444 0ustar rootroot
Program Flow Control The opcodes to manipulate which orchestra statements are executed are: cggoto cigoto ckgoto cngoto elseif else endif goto if igoto kgoto tigoto timout Opcodes to create looping constructions are: loop_ge loop_gt loop_le loop_lt until Some of these opcodes work at i-rate even if they contain k- or a- rate comparisons. See the Reinitialization section.
manual_src~dfsg/control/topXO.xml0000644000000000000000000000050012262561504016213 0ustar rootroot Instrument Control &controlclockctl; &controlconditional; &controldurctl; &controlinvocat; &controlpgmctl; &controlrealtime; &controlreinitn; &controlsensing; &controlstacks; &controlsubinstr; &controltimeread; manual_src~dfsg/control/fltkgeneral.xml0000644000000000000000000000175312262561504017453 0ustar rootroot
General FLTK Widget-related Opcodes The general FLTK widget-related opcodes are: FLgetsnap FLloadsnap FLrun FLsavesnap FLsetsnap FLupdate FLsetSnapGroup
manual_src~dfsg/control/invocat.xml0000644000000000000000000000255212262561504016616 0ustar rootroot
Instrument Invocation The opcodes one can use to create score events from within a orchestra are: event event_i scoreline_i scoreline schedule schedwhen schedkwhen schedkwhennamed The mute opcode can be used to mute/unmute instruments during a performance. Instruments definitions can be removed using the remove opcode.
manual_src~dfsg/control/fltkother.xml0000644000000000000000000000516012262561504017153 0ustar rootroot
Other FLTK Widgets Other FLTK widget opcodes are: FLbox FLbutBank FLbutton FLexecButton FLkeyIn FLhvsBox FLhvsBoxSetValue FLmouse FLprintk FLprintk2 FLslidBnk FLslidBnk2 FLslidBnkGetHandle FLslidBnkSet FLslidBnk2Set FLslidBnk2Setk FLvalue FLvkeybd FLvslidBnk FLvslidBnk2 FLxyin
manual_src~dfsg/control/fltkappearance.xml0000644000000000000000000000420612262561504020131 0ustar rootroot
Modifying FLTK Widget Appearance The following opcodes modify FLTK widget appearance: FLcolor FLcolor2 FLhide FLlabel FLsetAlign FLsetBox FLsetColor FLsetColor2 FLsetFont FLsetPosition FLsetSize FLsetText FLsetTextColor FLsetTextSize FLsetTextType FLsetVal_i FLsetVal FLshow
manual_src~dfsg/control/top.xml0000644000000000000000000000046712262561504015760 0ustar rootroot Instrument Control &controlclockctl; &controlconditional; &controldurctl; &controlfltkintro; &controlinvocat; &controlpgmctl; &controlrealtime; &controlreinitn; &controlsensing; &controlstacks; &controlsubinstr; &controltimeread; manual_src~dfsg/control/sensingXO.xml0000644000000000000000000000732412262561504017072 0ustar rootroot
Sensing and Control Keyboard and mouse sensing sensekey (also called sense) xyin Envelope followers follow follow2 peak rms Tempo and Pitch estimation ptrack pitch pitchamdf tempest Tempo and Sequencing tempo miditempo tempoval seqtime seqtime2 trigger trigseq timedseq changed System getcfg Score control rewindscore setscorepos
manual_src~dfsg/control/realtime.xml0000644000000000000000000000154412262561504016755 0ustar rootroot
Real-time Performance Control Opcodes that monitor and control real-time performance are: active cpuprc maxalloc prealloc jacktransport The running csound process can be terminated using exitnow.
manual_src~dfsg/control/fltkintro.xml0000644000000000000000000002633312262561504017172 0ustar rootroot
FLTK Widgets and GUI controllers Widgets allow the design of a custom Graphical User Interface (GUI) to control an orchestra in real-time. They are derived from the open-source library FLTK (Fast Light Tool Kit). This library is one of the fastest graphic libraries available, supports OpenGL and should be source compatible with different platforms (Windows, Linux, Unix and Mac OS). The subset of FLTK implemented in Csound provides the following types of objects: Containers FLTK Containers are widgets that contain other widgets such as panels, windows, etc. Csound provides the following container objects: Panels Scroll areas Pack Tabs Groups Valuators The most useful objects are named FLTK Valuators. These objects allow the user to vary synthesis parameter values in real-time. Csound provides the following valuator objects: Sliders Knobs Rollers Text fields Joysticks Counters Other widgets There are other FTLK widgets that are not valuators nor containers: Buttons Button banks Labels Keyboard and Mouse sensing Also there are some other opcodes useful to modify the widget appearance: Updating widget value. Setting primary and selection colors of a widget. Setting font type, size and color of widgets. Resizing a widget. Hiding and showing a widget. There are also these general opcodes that allow the following actions: Running the widget thread: FLrun Loading snapshots containing the status of all valuators of an orchestra: FLgetsnap and FLloadsnap. Saving snapshots containing the status of all valuators of an orchestra: FLsavesnap and FLsetsnap Setting the snapshot group of a declared valuator: FLsetSnapGroup Below is a simple example of Csound code to create a window. Notice that all opcodes are init-rate and must be called only once per session. The best way to use them is to place them in the header section of an orchestra, before any instrument. Even though placing them inside an instrument is not prohibited, unpredictable results can occur if that instrument is called more than once. Each container is made up of a couple of opcodes: the first indicating the start of the container block and the last indicating the end of that container block. Some container blocks can be nested but they must not be crossed. After defining all containers, a widget thread must be run by using the special FLrun opcode that takes no arguments. <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o linseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;******************************* sr=48000 kr=480 ksmps=100 nchnls=1 ;*** It is recommended to put almost all GUI code in the ;*** header section of an orchestra FLpanel "Panel1",450,550 ;***** start of container ; some widgets should contained here FLpanelEnd ;***** end of container FLrun ;***** runs the widget thread, it is always required! instr 1 ;put some synthesis code here endin ;******************************* </CsInstruments> <CsScore> f 0 3600 ;dummy table for realtime input e </CsScore> </CsoundSynthesizer> The previous code simply creates a panel (an empty window because no widgets are defined inside the container). The following example creates two panels and inserts a slider inside each of them: <CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform ; Audio out Audio in No messages -odac -iadc ; -d ;;;RT audio I/O ; For Non-realtime ouput leave only the line below: ; -o linseg.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> ;******************************* sr=48000 kr=480 ksmps=100 nchnls=1 FLpanel "Panel1",450,550,100,100 ;***** start of container gk1,iha FLslider "FLslider 1", 500, 1000, 0 ,1, -1, 300,15, 20,50 FLpanelEnd ;***** end of container FLpanel "Panel2",450,550,100,100 ;***** start of container gk2,ihb FLslider "FLslider 2", 100, 200, 0 ,1, -1, 300,15, 20,50 FLpanelEnd ;***** end of container FLrun ;***** runs the widget thread, it is always required! instr 1 ; gk1 and gk2 variables that contain the output of valuator ; widgets previously defined, can be used inside any instrument printk2 gk1 printk2 gk2 ;print the values of the valuators whenever they change endin ;******************************* </CsInstruments> <CsScore> f 0 3600 ;dummy table for realtime input e </CsScore> </CsoundSynthesizer> All widget opcodes are init-rate opcodes, even if valuators output k-rate variables. This happens because an independent thread is run based on a callback mechanism. It consumes very few processing resources since there is no need of polling. (This differs from other MIDI based controller opcodes.) So you can use any number of windows and valuators without degrading the real-time performance. &controlfltkcontainers; &controlfltkvaluators; &controlfltkother; &controlfltkappearance; &controlfltkgeneral;
manual_src~dfsg/control/subinstr.xml0000644000000000000000000000110212262561504017012 0ustar rootroot
Sub-instrument Control These opcodes let one define and use a sub-instrument: subinstr subinstrinit See also the UDO and Orchestra Macros Macros section for similar functionality.
manual_src~dfsg/control/stacks.xml0000644000000000000000000000126712262561504016445 0ustar rootroot
Stacks Csound implements a global stack that can be accessed with the following opcodes: stack pop push pop_f push_f
manual_src~dfsg/control/conditional.xml0000644000000000000000000000104612262561504017453 0ustar rootroot
Conditional Values The opcodes for conditional values are ==, >=, >, <, <=, and !=.
manual_src~dfsg/control/sensing.xml0000644000000000000000000000731512262561504016623 0ustar rootroot
Sensing and Control TCL/TK widgets button checkbox control setctrl Keyboard and mouse sensing sensekey (also called sense) xyin Envelope followers follow follow2 peak rms Tempo and Pitch estimation ptrack pitch pitchamdf tempest Tempo and Sequencing tempo miditempo tempoval seqtime seqtime2 trigger trigseq timedseq changed System getcfg Score control rewindscore setscorepos
manual_src~dfsg/opcodes/0000755000000000000000000000000012270515241014375 5ustar rootrootmanual_src~dfsg/opcodes/fin.xml0000644000000000000000000000705312262561504015704 0ustar rootroot fin Signal I/O:File I/O fin fin Read signals from a file at a-rate. Description Read signals from a file at a-rate. Syntax fin ifilename, iskipframes, iformat, ain1 [, ain2] [, ain3] [,...] fin ifilename, iskipframes, iformat, arr[] Initialization ifilename -- input file name (can be a string or a handle number generated by fiopen). iskipframes -- number of frames to skip at the start (every frame contains a sample of each channel) iformat -- a number specifying the input file format for headerless files. If a header is found, this argument is ignored. 0 - 32 bit floating points without header 1 - 16 bit integers without header Performance fin (file input) is the complement of fout: it reads a multichannel file to generate audio rate signals. The user must be sure that the number of channels of the input file is the same as the number of ainX arguments. Please note that since this opcode generates its output using input parameters (on the right side of the opcode), these variables must be initialized before use, otherwise a 'used before defined' error will occur. You can use the init opcode for this. Examples Here is an example of the fin opcode. It uses the file fin.csd and fox.wav. Example of the fin opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also fini, fink Credits Author: &namegabriel; Italy 1999 Author: &namejohn; NUIM, 2013 New in Csound version 3.56 Array version new in 6.01 manual_src~dfsg/opcodes/ktrirand.xml0000644000000000000000000000077612262561504016753 0ustar rootroot ktrirand ktrirand ktrirand Deprecated. Description Deprecated as of version 3.49. Use the trirand opcode instead. manual_src~dfsg/opcodes/kfilter2.xml0000644000000000000000000000146112262561504016647 0ustar rootroot kfilter2 kfilter2 kfilter2 Deprecated. Description Deprecated as of version 3.49. Use the filter2 opcode instead. Credits Author: Michael A. Casey M.I.T. Cambridge, Mass. 1997 New in version 3.47 manual_src~dfsg/opcodes/mute.xml0000644000000000000000000000531412262561504016100 0ustar rootroot mute Instrument Control:Invocation mute mute Mutes/unmutes new instances of a given instrument. Description Mutes/unmutes new instances of a given instrument. Syntax mute insnum [, iswitch] mute "insname" [, iswitch] Initialization insnum -- instrument number. Equivalent to p1 in a score i statement. insname -- A string (in double-quotes) representing a named instrument. iswitch (optional, default=0) -- represents a switch to mute/unmute an instrument. A value of 0 will mute new instances of an instrument, other values will unmute them. The default value is 0. Performance All new instances of instrument inst will me muted (iswitch = 0) or unmuted (iswitch not equal to 0). There is no difficulty with muting muted instruments or unmuting unmuted instruments. The mechanism is the same as used by the score q statement. For example, it is possible to mute in the score and unmute in some instrument. Muting/Unmuting is indicated by a message (depending on message level). Examples Here is an example of the mute opcode. It uses the file mute.csd. Example of the mute opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits New in version 4.22 manual_src~dfsg/opcodes/gaussi.xml0000644000000000000000000000670712262561504016430 0ustar rootroot x-class noise generators gaussi Signal Generators:Random (Noise) Generators gaussi gaussi Gaussian distribution random number generator with interpolation. Description Gaussian distribution random number generator with controlled interpolation between values. This is an x-class noise generator. Syntax ares gaussi krange, xamp, xcps ires gaussi krange, xamp, xcps kres gaussi krange, xamp, xcps Performance krange -- the range of the random numbers (-krange to +krange). Outputs both positive and negative numbers. gauss returns random numbers following a normal distribution centered around 0.0 (mu = 0.0) with a variance (sigma) of krange / 3.83. Thus more than 99.99% of the random values generated are in the range -krange to +krange. If a mean value different of 0.0 is desired, this mean value has to be added to the generated numbers (see example below). For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. xamp -- range over which random numbers are distributed. xcps -- the frequency which new random numbers are generated. Examples Here is an example of the gaussi opcode. It uses the file gaussi.csd. Example of the gaussi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also gauss Credits Author: &namejohn; Bath May 2011 New in version 5.14 manual_src~dfsg/opcodes/resonx.xml0000644000000000000000000000771312262561504016451 0ustar rootroot resonx Signal Modifiers:Standard Filters:Resonant resonx resonx Emulates a stack of filters using the reson opcode. Description resonx is equivalent to a filters consisting of more layers of reson with the same arguments, serially connected. Using a stack of a larger number of filters allows a sharper cutoff. They are faster than using a larger number instances in a Csound orchestra of the old opcodes, because only one initialization and k- cycle are needed at time and the audio loop falls entirely inside the cache memory of processor. Syntax ares resonx asig, xcf, xbw [, inumlayer] [, iscl] [, iskip] Initialization inumlayer (optional) -- number of elements in the filter stack. Default value is 4. iscl (optional, default=0) -- coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. (This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise.) A zero value signifies no scaling of the signal, leaving that to some later adjustment (see balance). The default value is 0. iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance asig -- input signal xcf -- the center frequency of the filter, or frequency position of the peak response. xbw -- bandwidth of the filter (the Hz difference between the upper and lower half-power points) Examples Here is an example of the resonx opcodes. It uses the file resonx.csd. Example of the resonx opcodes. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also atonex, tonex Credits Author: &namegabriel; (adapted by &namejohn;) Italy New in Csound version 3.49 Audio rate parameters introduced in version 6.02 November 2013. manual_src~dfsg/opcodes/vstprogset.xml0000644000000000000000000000465512262561504017355 0ustar rootroot vstprogset Plugin Hosting:VST vstprogset vstprogset Loads parameter banks to a VST plugin. Syntax vstprogset instance, kprogram Description vstprogset sets one of the programs in an .fxb bank. Initialization instance - the number which identifies the plugin, to be passed to other vst4cs opcodes. kprogram - the number of the program to set. Examples Usage of vstprogset /* orc */ sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 gihandle1 vstinit "c:/vstplugins/cheeze/cheeze machine.dll", 1 instr 4 vstbankload gihandle1,"c:/vstplugins/cheeze/chengo'scheese.fxb" vstprogset gihandle1, 4 vstinfo gihandle1 endin /* sco */ i 3 0 21 i 4 1 1 57 32 e Credits By: &nameandres; and &namemichael; Uses code from Hermann Seib's VSTHost and Thomas Grill's vst~ object. VST is a trademark of Steinberg Media Technologies GmbH. VST Plug-In Technology by Steinberg. manual_src~dfsg/opcodes/tableikt.xml0000644000000000000000000001220412262561504016721 0ustar rootroot tableikt Table Control:Dynamic Selection tableikt tableikt Provides k-rate control over table numbers. Description k-rate control over table numbers. Function tables are read with linear interpolation. The standard Csound opcode tablei, when producing a k- or a-rate result, can only use an init-time variable to select the table number. tableikt accepts k-rate control as well as i-time. In all other respects they are similar to the original opcodes. Syntax ares tableikt xndx, kfn [, ixmode] [, ixoff] [, iwrap] kres tableikt kndx, kfn [, ixmode] [, ixoff] [, iwrap] Initialization ixmode -- if 0, xndx and ixoff ranges match the length of the table. if non-zero xndx and ixoff have a 0 to 1 range. Default is 0 ixoff -- if 0, total index is controlled directly by xndx, ie. the indexing starts from the start of the table. If non-zero, start indexing from somewhere else in the table. Value must be positive and less than the table length (ixmode = 0) or less than 1 (ixmode not equal to 0). Default is 0. iwrap -- if iwrap = 0, Limit mode: when total index is below 0, then final index is 0.Total index above table length results in a final index of the table length - high out of range total indexes stick at the upper limit of the table. If iwrap not equal to 0, Wrap mode: total index is wrapped modulo the table length so that all total indexes map into the table. For instance, in a table of length 8, xndx = 5 and ixoff = 6 gives a total index of 11, which wraps to a final index of 3. Default is 0. Performance kndx -- Index into table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode not equal to 0). xndx -- matching the table length (ixmode = 0) or a 0 to 1 range (ixmode not equal to 0) kfn -- Table number. Must be >= 1. Floats are rounded down to an integer. If a table number does not point to a valid table, or the table has not yet been loaded (GEN01) then an error will result and the instrument will be de-activated. Caution with k-rate table numbers At k-rate, if a table number of < 1 is given, or the table number points to a non-existent table, or to one which has a length of 0 (it is to be loaded from a file later) then an error will result and the instrument will be deactivated. kfn must be initialized at the appropriate rate using init. Attempting to load an i-rate value into kfn will result in an error. Examples Here is an example of the tableikt opcode. It uses the file tableikt.csd. Example of the tableikt opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: play table no: 8 play table no: 7 play table no: 6 ..... play table no: 2 play table no: 1 See Also tablekt Credits Author: Robin Whittle Australia May 1997 New in version 3.47 manual_src~dfsg/opcodes/kdump.xml0000644000000000000000000000075312262561504016250 0ustar rootroot kdump kdump kdump Deprecated. Description Deprecated as of version 3.49. Use the dumpk opcode instead. manual_src~dfsg/opcodes/slicearray.xml0000644000000000000000000000376312262561504017272 0ustar rootroot slicearray Table Control:Table Queries slicearray slicearray Take a slice of a vector. Description Take a slice of a vector (one-dimensional k-rate array). Syntax karray slicearray kinarray, istart, iend Initialization istart -- index of the first part of the answer. iend -- index of the first element of the answer. Examples Here is an example of the slicearray opcode. It uses the file slicearray.csd. Example of the slicearray opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also Vectorial opcodes Credits Author: &namejohn; Codemist Ltd 2013 New in version 6.00 manual_src~dfsg/opcodes/hvs2.xml0000644000000000000000000003070512262561504016012 0ustar rootroot hvs2 Signal Generators:Hyper Vectorial Synthesis hvs2 hvs2 Allows two-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables. Description hvs2 allows two-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables. Syntax hvs2 kx, ky, inumParms, inumPointsX, inumPointsY, iOutTab, iPositionsTab, iSnapTab [, iConfigTab] Initialization inumParms - number of parameters controlled by the HVS. Each HVS snapshot is made up of inumParms elements. inumPointsX, inumPointsY - number of points that each dimension of the HVS cube (or square in case of two-dimensional HVS; or line in case of one-dimensional HVS) is made up. iOutTab - number of the table receiving the set of output-parameter instant values of the HVS. The total amount of parameters is defined by the inumParms argument. iPositionsTab – a table filled with the individual positions of snapshots in the HVS matrix (see below for more information). iSnapTab – a table filled with all the snapshots. Each snapshot is made up of a set of parameter values. The amount of elements contained in each snapshots is specified by the inumParms argument. The set of elements of each snapshot follows (and is adjacent) to the previous one in this table. So the total size of this table should be >= to inumParms multiplied the number of snapshots you intend to store for the HVS. iConfigTab – (optional) a table containing the behavior of the HVS for each parameter. If the value of iConfigTab is zero (default), this argument is ignored, meaning that each parameter is treated with linear interpolation by the HVS. If iConfigTab is different than zero, then it must refer to an existing table whose contents are in its turn referring to a particolar kind of interpolation. In this table, a value of -1 indicates that corresponding parameter is leaved unchanged (ignored) by the HVS; a value of zero indicates that corresponding parameter is treated with linear-interpolation; each other values must be integer numbers indicating an existing table filled with a shape which will determine the kind of special interpolation to be used (table-based interpolation). Performance kx, ky - these are externally-modified variables which controls the motion of the pointer in the HVS matrix cube (or square or line in case of HVS matrices made up of less than 3 dimensions). The range of these input arguments must be 0 to 1. Hyper Vectorial Synthesis is a technique that allows control of a huge set of parameters by using a simple and global approach. The key concepts of the HVS are: The set of HVS parameters, whose amount is fixed and defined by the inumParms argument. During the HVS performance, all these parameters are variant and can be applied to any sound synthesis technique, as well as to any global control for algorithmic composition and any other kind of level. The user must previously define several sets of fixed values for each HVS parameter, each set corresponding to a determinate synthesis configuration. Each set of values is called snapshot, and can be considered as the coordinates of a bound of a multi-dimensional space. The HVS consists on moving a point in this multi-dimensional space (by using a special motion pointer, see below), according and inside the bounds defined by the snapshots. You can fix any amount of HVS parameters (each parameter being a dimension of the multi-dimensional space), even a huge number, the limit only depends on the processing power (and the memory) of your computer and on the complexity of the sound-synthesis you will use. The HVS cube (or square or line). This is the matrix (of 3, 2 or 1 dimensions, according to the hvs opcode you intend to use) of “mainstays” (or pivot) points of HVS. The total amount of pivot-points depends on the value of the inumPointsX, inumPointsY and inumPointsZ arguments. In the case of a 3-dimensional HVS matrix you can define, for instance, 3 points for the X dimension, 5 for the Y dimension and 2 for the Z dimension. In this case, the total number of pivot-points is 3 * 5 * 2 = 30. With this set of pivot points, the cube Is divided into smaller cubed zones each one bounded by eight nearby points. Each point is numbered. The numeral order of these points is enstabilished in the following way: number zero is the first point, number 1 the second and so on. Assuming you are using a 3-dimensional HVS cube having the number of points above mentioned (i.e. 3, 5 and 2 respectively for the X, Y and Z axis), the first point (point zero) is the upper-left-front vertex of the cube, by facing the XY plane of the cube. The second point is the middle point of the upper front edge of the cube and so on. You can refer to the figure below in order to understand how the numeral order of the pivot-points proceeds: For the 2-dimensional HVS, it is the same, by only omitting the rear cube face, so each zone is bounded by 4 pivot-points instead of 8. For the 1-dimensional HVS, the whole thing is even simpler because it is a line with the pivot-points proceeding from left to right. Each point is coupled with a snapshot. Snapshot order, as stored into the iSnapTab, can or cannot follow the order of the pivot-points numbers. In fact it is possible to alter this order by means the iPositionsTab, a table that remaps the position of each snapshot in relation to the pivot points. The iPositionsTab is made up of the positions of the snapshots (contained in the iSnapTab) in the two-dimensional grid. Each subsequent element is actually a pointer representing the position in the iSnapTab. For example, in a 2-dimensional HVS matrix such as the following (in this case having inumPointsX = 3 and inumPointsY = 5: 571 349 620 413 827
These numbers (to be stored in the iSnapTab table by using, for instance, the GEN02 function generator) represents the snapshot position within the grid (in this case a 3x5 matrix). So, the first element 5, has index zero and represents the 6th (element zero is the first) snapshot contained in the iSnapTab, the second element 7 represents the 8th element of iSnapTab and so on. Summing up, the vertices of each zone (a cubed zone is delimited by 8 vertices; a squared zone by 4 vertices and a linear zone by 2 points) are coupled with a determinate snapshot, whose number is remapped by the iSnapTab. Output values of the HVS are influenced by the motion pointer, a point whose position, in the HVS cube (or square or segment) is determined by the kx, ky and kz arguments. The values of these arguments, which must be in the 0 to 1 range, are externally set by the user. The output values, whose amount is equal to the inumParms argument, are stored in the iOutTab, a table that must be already allocated by the user, and must be at least inumParms size. In what way the motion pointer influences the output? Well, when the motion pointer falls in a determinate cubed zone, delimited, for instance, by 8 vertices (or pivot points), we assume that each vertex has associated a different snapshot (i.e. a set of inumParms values), well, the output will be the weighted average value of the 8 vertices, calculated according on the distance of the motion pointer from each of the 8 vertices. In the case of a default behavior, when the iConfigTab argument is not set, the exact output is calculated by using linear interpolation which is applied to each different parameter of the HVS. Anyway, it is possible to influence this behavior by setting the iConfigTab argument to a number of a table whose contents can affect one or more HVS parameters. The iConfigTab table elements are associated to each HVS parameter and their values affect the HVS output in the following way: If iConfigTab is equal to -1, corresponding output is skipped, i.e. the element is not calculated, leaving corresponding element value in the iOutTab unchanged; If iConfigTab is equal to zero, then the normal HVS output is calculated (by using weighted average of the nearest vertex of current zone where it falls the motion pointer); If iConfigTab element is equal to an integer number > zero, then the contents of a table having that number is used as a shape of a table-based interpolation. Examples Here is an example of the hvs2 opcode. It uses the file hvs2.csd. Example of the hvs2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is second example of the hvs2 opcode. It uses the file hvs2-2.csd. Second example of the hvs2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also hvs1, hvs3, vphaseseg Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/exprandi.xml0000644000000000000000000000617712262561504016750 0ustar rootroot x-class noise generators exprandi Signal Generators:Random (Noise) Generators exprandi exprandi Exponential distribution random number generator with interpolation (positive values only). Description Exponential distribution random number generator with controlled interpolation between values (positive values only). This is an x-class noise generator. Syntax ares exprandi klambda, xamp, xcps ires exprandi klambda, xamp, xcps kres exprandi klambda, xamp, xcps Performance klambda -- lambda parameter for the exponential distribution. The probablity density function of an exponential distribution is an exponential curve, whose mean is 0.69515/lambda. For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. xamp -- range over which random numbers are distributed. xcps -- the frequency which new random numbers are generated. Examples Here is an example of the exprandi opcode. It uses the file exprandi.csd. Example of the exprandi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also exprand Credits Author: &namejohn; Bath May 2011 New in version 5.14 manual_src~dfsg/opcodes/distort.xml0000644000000000000000000001232312262561504016614 0ustar rootroot distort Signal Modifiers:Special Effects distort distort Distort an audio signal via waveshaping and optional clipping. Description Distort an audio signal via waveshaping and optional clipping. Syntax ar distort asig, kdist, ifn[, ihp, istor] Initialization ifn -- table number of a waveshaping function with extended guard point. The function can be of any shape, but it should pass through 0 with positive slope at the table mid-point. The table size need not be large, since it is read with interpolation. ihp -- (optional) half-power point (in cps) of an internal low-pass filter. The default value is 10. istor -- (optional) initial disposition of internal data space (see reson). The default value is 0. Performance asig -- Audio signal to be processed kdist -- Amount of distortion (usually between 0 and 1) This unit distorts an incoming signal using a waveshaping function ifn and a distortion index kdist. The input signal is first compressed using a running rms, then passed through a waveshaping function which may modify its shape and spectrum. Finally it is rescaled to approximately its original power. The amount of distortion depends on the nature of the shaping function and on the value of kdist, which generally ranges from 0 to 1. For low values of kdist, we should like the shaping function to pass the signal almost unchanged. This will be the case if, at the mid-point of the table, the shaping function is near-linear and is passing through 0 with positive slope. A line function from -1 to +1 will satisfy this requirement; so too will a sigmoid (sinusoid from 270 to 90 degrees). As kdist is increased, the compressed signal is expanded to encounter more and more of the shaping function, and if this becomes non-linear the signal is increasingly bent on read-through to cause distortion. When kdist becomes large enough, the read-through process will eventually hit the outer limits of the table. The table is not read with wrap-around, but will stick at the end-points as the incoming signal exceeds them; this introduces clipping, an additional form of signal distortion. The point at which clipping begins will depend on the complexity (rms-to-peak value) of the input signal. For a pure sinusoid, clipping will begin only as kdist exceeds 0.7; for a more complex input, clipping might begin at a kdist of 0.5 or much less. kdist can exceed the clip point by any amount, and may be greater than 1. The shaping function can be made arbitrarily complex for extra effect. It should generally be continuous, though this is not a requirement. It should also be well-behaved near the mid-point, and roughly balanced positive-negative overall, else some excessive DC offset may result. The user might experiment with more aggressive functions to suit the purpose. A generally positive slope allows the distorted signal to be mixed with the source without phase cancellation. distort is useful as an effects process, and is usually combined with reverb and chorusing on effects busses. However, it can alternatively be used to good effect within a single instrument. Examples Here is an example of the distort opcode. It uses the file distort.csd. Example of the distort opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by &namebarry; for Extended Csound and released in Csound5. manual_src~dfsg/opcodes/slider32f.xml0000644000000000000000000001157312262561504016727 0ustar rootroot slider32f Real-time MIDI:Slider Banks slider32f slider32f Creates a bank of 32 different MIDI control message numbers, filtered before output. Description Creates a bank of 32 different MIDI control message numbers, filtered before output. Syntax k1,...,k32 slider32f ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1, \ ..., ictlnum32, imin32, imax32, init32, ifn32, icutoff32 Initialization ichan -- MIDI channel (1-16) ictlnum1 ... ictlnum32 -- MIDI control number (0-127) imin1 ... imin32 -- minimum values for each controller imax1 ... imax32 -- maximum values for each controller init1 ... init32 -- initial value for each controller ifn1 ... ifn32 -- function table for conversion for each controller icutoff1 ... icutoff32 -- low-pass filter cutoff frequency for each controller Performance k1 ... k32 -- output values slider32f is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider32f allows a bank of 32 different MIDI control message numbers. It filters the signal before output. This eliminates discontinuities due to the low resolution of the MIDI (7 bit). The cutoff frequency can be set separately for each controller (suggested range: .1 to 5 Hz). As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. slider32f opcodes do not output the required initial value immediately, but only after some k-cycles because the filter slightly delays the output. See Also s16b14, s32b14, slider16, slider16f, slider32, slider64, slider64f, slider8, slider8f Credits Author: &namegabriel; Italy December 1998 New in Csound version 3.50 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/taninv2.xml0000644000000000000000000000675312262561504016517 0ustar rootroot taninv2 Mathematical Operations:Opcode Equivalents of Functions taninv2 taninv2 Returns an arctangent. Description Returns the arctangent of iy/ix, ky/kx, or ay/ax. Syntax ares taninv2 ay, ax ires taninv2 iy, ix kres taninv2 ky, kx ...taninv2(ky, kx)... (no rate restriction) Returns the arctangent of iy/ix, ky/kx, or ay/ax. If y is zero, taninv2 returns zero regardless of the value of x. If x is zero, the return value is: π/2, if y is positive. -π/2, if y is negative. 0, if y is 0. Initialization iy, ix -- values to be converted Performance ky, kx -- control rate signals to be converted ay, ax -- audio rate signals to be converted Examples Here is an example of the taninv2 opcode. It uses the file taninv2.csd. Example of the taninv2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: i1 = 0.464 See Also taninv Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April 1998 Example written by &namekevin;. New in Csound version 3.48 Corrected on May 2002, thanks to &nameistvan;. Available as a function in version 6.00 manual_src~dfsg/opcodes/line.xml0000644000000000000000000000632112262561504016054 0ustar rootroot line Signal Generators:Linear and Exponential Generators line line Trace a straight line between specified points. Description Trace a straight line between specified points. Syntax ares line ia, idur, ib kres line ia, idur, ib Initialization ia -- starting value. ib -- value after idur seconds. idur -- duration in seconds of segment. A zero or negative value will cause all initialization to be skipped. Performance line generates control or audio signals whose values move linearly from an initial value to a final one. A common error with this opcode is to assume that the value of ib is held after the time idur1. line does not automatically end or stop at the end of the duration given. If your note length is longer than idur seconds, kres (or ares) will not come to rest at ib, but will instead continue to rise or fall with the same rate. If a rise (or fall) and then hold is required that the linseg opcode should be considered instead. Examples Here is an example of the line opcode. It uses the file line.csd. Example of the line opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also expon, expseg, expsegr, linseg, linsegr Credits manual_src~dfsg/opcodes/JackoMidiOut.xml0000644000000000000000000000566112262561504017455 0ustar rootroot JackoMidiOut Jacko Opcodes JackoMidiOut JackoMidiOut Sends a MIDI channel message to a Jack port. Description Sends a MIDI channel message to a Jack MIDI output port inside this instance of Csound, and in turn to its connected external Jack MIDI input port. Syntax JackoMidiOut ScsoundPortName, kstatus, kchannel, kdata1[, kdata2] Initialization ScsoundPortName -- The short name ("portname") of the internal Jack MIDI output port. Performance kstatus -- MIDI status byte; must indicate a MIDI channel message. kchannel -- MIDI channel (from 0 through 15). kdata1 -- First data byte of a MIDI channel message. kdata2 -- Optional second data byte of a MIDI channel message. This opcode can be called any number of times in the same kperiod. Messages from multiple instances of the opcode sending to the same port are collected before sending. Running status, system exclusive messages, and real-time messages are not supported. The granularity of timing is Csound's kperiod. See Also JackoInfo, JackoInfo, JackoFreewheel, JackoAudioOutConnect, JackoMidiInConnect, JackoMidiOutConnect, JackoOn, JackoAudioIn, JackoNoteOut, JackoTransport. Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/soundout.xml0000644000000000000000000000764312262561504017015 0ustar rootroot soundout Signal I/O:Signal Output soundout soundout Deprecated. Writes audio output to a disk file. Description The usage of soundout is discouraged. Please use fout instead. Writes audio output to a disk file. Syntax soundout asig1, ifilcod [, iformat] Initialization ifilcod -- integer or character-string denoting the destination soundfile name. An integer denotes the file soundin.filcod; a character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the named file is sought first in the current directory, then in that given by the environment variable SSDIR (if defined) then by SFDIR. See also GEN01. iformat (optional, default=0) -- specifies the audio data file format: 1 = 8-bit signed char (high-order 8 bits of a 16-bit integer) 2 = 8-bit A-law bytes 3 = 8-bit U-law bytes 4 = 16-bit short integers 5 = 32-bit long integers 6 = 32-bit floats If iformat = 0 it is taken from the soundfile header, and if no header from the Csound -o command-line flag. The default value is 0. Performance soundout writes audio output to a disk file. Use of fout is recommended instead of soundout See Also fout, out, outh, outo, outq, outq1, outq2, outq3, outq4, outs, outs1, outs2 soundouts Credits Author: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 manual_src~dfsg/opcodes/nreverb.xml0000644000000000000000000001115012262561504016564 0ustar rootroot nreverb Signal Modifiers:Reverberation nreverb nreverb A reverberator consisting of 6 parallel comb-lowpass filters. Description This is a reverberator consisting of 6 parallel comb-lowpass filters being fed into a series of 5 allpass filters. nreverb replaces reverb2 (version 3.48) and so both opcodes are identical. Syntax ares nreverb asig, ktime, khdif [, iskip] [,inumCombs] [, ifnCombs] \ [, inumAlpas] [, ifnAlpas] Initialization iskip (optional, default=0) -- Skip initialization if present and non-zero. inumCombs (optional) -- number of filter constants in comb filter. If omitted, the values default to the nreverb constants. New in Csound version 4.09. ifnCombs - function table with inumCombs comb filter time values, followed the same number of gain values. The ftable should not be rescaled (use negative fgen number). Positive time values are in seconds. The time values are converted internally into number of samples, then set to the next greater prime number. If the time is negative, it is interpreted directly as time in sample frames, and no processing is done (except negation). New in Csound version 4.09. inumAlpas, ifnAlpas (optional) -- same as inumCombs/ifnCombs, for allpass filter. New in Csound 4.09. Performance The input signal asig is reverberated for ktime seconds. The parameter khdif controls the high frequency diffusion amount. The values of khdif should be from 0 to 1. If khdif is set to 0 the all the frequencies decay with the same speed. If khdif is 1, high frequencies decay faster than lower ones. If ktime is inadvertently set to a non-positive number, ktime will be reset automatically to 0.01. (New in Csound version 4.07.) As of Csound version 4.09, nreverb may read any number of comb and allpass filter from an ftable. Examples Here is a simple example of the nreverb opcode. It uses the file nreverb.csd. Simple example of the nreverb opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an example of the nreverb opcode using an ftable for filter constants. It uses the file nreverb_ftable.csd, and beats.wav. An example of the nreverb opcode using an ftable for filter constants. Credits Authors: &nameparis; (reverb2) MIT, Cambridge 1995 Author: Richard Karpen (nreverb) Seattle, Wash 1998 manual_src~dfsg/opcodes/opor.xml0000644000000000000000000000751612262561504016113 0ustar rootroot || Mathematical Operations:Arithmetic and Logic Operations || || Logical OR operator. Description Arithmetic operators perform operations of change-sign (negate), don't-change-sign, logical AND logical OR, add, subtract, multiply and divide. Note that a value or an expression may fall between two of these operators, either of which could take it as its left or right argument, as in a + b * c. In such cases three rules apply: 1. * and / bind to their neighbors more strongly than + and −. Thus the above expression is taken as a + (b * c) with * taking b and c and then + taking a and b * c. 2. + and − bind more strongly than &&, which in turn is stronger than ||: a && b - c || d is taken as (a && (b - c)) || d 3. When both operators bind equally strongly, the operations are done left to right: a - b - c is taken as (a - b) - c Parentheses may be used as above to force particular groupings. Syntax a || b (logical OR; not audio-rate) where the arguments a and b may be further expressions. Examples Here is an example of the | operator. It uses the file logicOR.csd. Example of the | operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like these: Value:127 ChanNo:11 CtrlNo:62 Value:127 ChanNo:11 CtrlNo:60 Value:127 ChanNo:11 CtrlNo:60 Value:127 ChanNo:11 CtrlNo:60 Value:127 ChanNo:11 CtrlNo:60 .... Value:0 ChanNo:11 CtrlNo:62 Value:0 ChanNo:11 CtrlNo:62 Value:0 ChanNo:11 CtrlNo:62 Value:0 ChanNo:11 CtrlNo:62 .... Value:77 ChanNo:11 CtrlNo:23 Value:77 ChanNo:11 CtrlNo:23 Value:76 ChanNo:11 CtrlNo:23 Value:76 ChanNo:11 CtrlNo:23 .... See Also -, +, &&, *, /, ˆ, % manual_src~dfsg/opcodes/delay.xml0000644000000000000000000000547512262561504016234 0ustar rootroot delay Signal Modifiers:Delay delay delay Delays an input signal by some time interval. Description A signal can be read from or written into a delay path, or it can be automatically delayed by some time interval. Syntax ares delay asig, idlt [, iskip] Initialization idlt -- requested delay time in seconds. This can be as large as available memory will permit. The space required for n seconds of delay is 4n * sr bytes. It is allocated at the time the instrument is first initialized, and returned to the pool at the end of a score section. iskip (optional, default=0) -- initial disposition of delay-loop data space (see reson). The default value is 0. Performance asig -- audio signal delay is a composite of delayr and delayw, both reading from and writing into its own storage area. It can thus accomplish signal time-shift, although modified feedback is not possible. There is no minimum delay period. Examples Here is an example of the delay opcode. It uses the file delay.csd. Example of the delay opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also delay1, delayr, delayw manual_src~dfsg/opcodes/ftgen.xml0000644000000000000000000001043712262561504016233 0ustar rootroot ftgen Table Control ftgen ftgen Generate a score function table from within the orchestra. Description Generate a score function table from within the orchestra. Syntax gir ftgen ifn, itime, isize, igen, iarga [, iargb ] [...] Initialization gir -- either a requested or automatically assigned table number above 100. ifn -- requested table number If ifn is zero, the number is assigned automatically and the value placed in gir. Any other value is used as the table number itime -- is ignored, but otherwise corresponds to p2 in the score f statement. isize -- table size. Corresponds to p3 of the score f statement. igen -- function table GEN routine. Corresponds to p4 of the score f statement. iarga, iargb, ... -- function table arguments. Correspond to p5 through pn of the score f statement. Performance This is equivalent to table generation in the score with the f statement. Csound was originally designed to support tables with power of two sizes only. Though this has changed in recent versions (you can use any size by using a negative number), many opcodes will not accept them. Warning Although Csound will not protest if ftgen is used inside instr-endin statements, this is not the intended or supported use, and must be handled with care as it has global effects. (In particular, a different size usually leads to relocation of the table, which may cause a crash or otherwise erratic behaviour. Examples Here is an example of the ftgen opcode. It uses the file ftgen.csd. Example of the ftgen opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the ftgen opcode. It uses the file ftgen-2.csd. Example of the ftgen opcode. This example queries a file for it length to create an f-table of the appropriate size. See also GEN routine overview, ftgentmp Credits Author: &namebarry; M.I.T., Cambridge, Mass 1997 Added warning April 2002 by &namerasmus; manual_src~dfsg/opcodes/resyn.xml0000644000000000000000000000564212262561504016272 0ustar rootroot resyn Spectral Processing:Streaming resyn resyn Streaming partial track additive synthesis with cubic phase interpolation with pitch control and support for timescale-modified input Description The resyn opcode takes an input containg a TRACKS pv streaming signal (as generated, for instance by partials). It resynthesises the signal using linear amplitude and cubic phase interpolation to drive a bank of interpolating oscillators with amplitude and pitch scaling controls. Resyn is a modified version of sinsyn, allowing for the resynthesis of data with pitch and timescale changes. Syntax asig resyn fin, kscal, kpitch, kmaxtracks, ifn Performance asig -- output audio rate signal fin -- input pv stream in TRACKS format kscal -- amplitude scaling kpitch -- pitch scaling kmaxtracks -- max number of tracks in resynthesis. Limiting this will cause a non-linear filtering effect, by discarding newer and higher-frequency tracks (tracks are ordered by start time and ascending frequency, respectively) ifn -- function table containing one cycle of a sinusoid (sine or cosine) Examples Here is an example of the resyn opcode. It uses the file resyn.csd. Example of the resyn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of an ifd-analysis signal and cubic-phase additive resynthesis with pitch shifting. Credits Author: &namevictor; June 2005 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/ins.xml0000644000000000000000000000505612262561504015722 0ustar rootroot ins Signal I/O:Signal Input ins ins Reads stereo audio data from an external device or stream. Description Reads stereo audio data from an external device or stream. Warning This opcode is designed to be used only with orchestras that have nchnls_i=2. Doing so with orchestras with nchnls_i > 2 will cause incorrect audio input. Syntax ar1, ar2 ins Performance Reads stereo audio data from an external device or stream. If the command-line -i flag is set, sound is read continuously from the audio input stream (e.g. stdin or a soundfile) into an internal buffer. Any number of these opcodes can read freely from this buffer. Examples Here is an example of the ins opcode. It uses the file ins.csd. Example of the ins opcode. See Also diskin, in, inh, inh, ino, inq, soundin mp3in, Credits Authors: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 manual_src~dfsg/opcodes/diskgrain.xml0000644000000000000000000001265012262561504017102 0ustar rootroot diskgrain Signal Generators:Granular Synthesis diskgrain diskgrain Synchronous granular synthesis, using a soundfile as source. Description diskgrain implements synchronous granular synthesis. The source sound for the grains is obtained by reading a soundfile containing the samples of the source waveform. Syntax asig diskgrain Sfname, kamp, kfreq, kpitch, kgrsize, kprate, \ ifun, iolaps [,imaxgrsize , ioffset] Initialization Sfilename -- source soundfile. ifun -- grain envelope function table. iolaps -- maximum number of overlaps, max(kfreq)*max(kgrsize). Estimating a large value should not affect performance, but exceeding this value will probably have disastrous consequences. imaxgrsize -- max grain size in secs (default 1.0). ioffset -- start offset in secs from beginning of file (default: 0). Performance kamp -- amplitude scaling kfreq -- frequency of grain generation, or density, in grains/sec. kpitch -- grain pitch scaling (1=normal pitch, < 1 lower, > 1 higher; negative, backwards) kgrsize -- grain size in secs. kprate -- readout pointer rate, in grains. The value of 1 will advance the reading pointer 1 grain ahead in the source table. Larger values will time-compress and smaller values will time-expand the source signal. Negative values will cause the pointer to run backwards and zero will freeze it. The grain generator has full control of frequency (grains/sec), overall amplitude, grain pitch (a sampling increment) and grain size (in secs), both as fixed or time-varying (signal) parameters. An extra parameter is the grain pointer speed (or rate), which controls which position the generator will start reading samples in the file for each successive grain. It is measured in fractions of grain size, so a value of 1 (the default) will make each successive grain read from where the previous grain should finish. A value of 0.5 will make the next grain start at the midway position from the previous grain start and finish, etc.. A value of 0 will make the generator read always from a fixed position (wherever the pointer was last at). A negative value will decrement pointer positions. This control gives extra flexibility for creating timescale modifications in the resynthesis. Diskgrain will generate any number of parallel grain streams (which will depend on grain density/frequency), up to the olaps value (default 100). The number of streams (overlapped grains) is determined by grainsize*grain_freq. More grain overlaps will demand more calculations and the synthesis might not run in realtime (depending on processor power). Diskgrain can simulate FOF-like formant synthesis, provided that a suitable shape is used as grain envelope and a sinewave as the grain wave. For this use, grain sizes of around 0.04 secs can be used. The formant centre frequency is determined by the grain pitch. Since this is a sampling increment, in order to use a frequency in Hz, that value has to be scaled by tablesize/sr. Grain frequency will determine the fundamental. This opcode is a variation on the syncgrain opcode. Examples Here is an example of the diskgrain opcode. It uses the file diskgrain.csd. Example of the diskgrain opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; May 2007 New in Csound 5.06 manual_src~dfsg/opcodes/rand.xml0000644000000000000000000000732012262561504016051 0ustar rootroot rand Signal Generators:Random (Noise) Generators rand rand Generates a controlled random number series. Description Output is a controlled random number series between -amp and +amp Syntax ares rand xamp [, iseed] [, isel] [, ioffset] kres rand xamp [, iseed] [, isel] [, ioffset] Initialization iseed (optional, default=0.5) -- a seed value for the recursive pseudo-random formula. A value between 0 and 1 will produce an initial output of kamp * iseed. A value greater than 1 will be seeded from the system clock. A negative value will cause seed re-initialization to be skipped. The default seed value is .5. isel (optional, default=0) -- if zero, a 16-bit number is generated. If non-zero, a 31-bit random number is generated. Default is 0. ioffset (optional, default=0) -- a base value added to the random result. New in Csound version 4.03. Performance kamp, xamp -- range over which random numbers are distributed. ares, kres -- Random number produced. The internal pseudo-random formula produces values which are uniformly distributed over the range kamp to -kamp. rand will thus generate uniform white noise with an R.M.S value of kamp / root 2. The value ares or kres is within is a half-closed interval which contains -xamp, but never contains +xamp. Examples Here is an example of the rand opcode. It uses the file rand.csd. Example of the rand opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example will produce the following output: i 1 time 0.00067: 50.00305 i 1 time 0.50000: 62.71362 i 1 time 1.00000: -89.31885 WARNING: Seeding from current time 472230558 i 2 time 2.00067: -70.65735 i 2 time 2.50000: 69.15283 i 2 time 3.00000: -48.79761 See Also randh, randi Credits Thanks to a note from &namejohn;, I changed the names of the parameters. manual_src~dfsg/opcodes/biquad.xml0000644000000000000000000000705312262561504016375 0ustar rootroot biquad Signal Modifiers:Standard Filters biquad biquad A sweepable general purpose biquadratic digital filter. Description A sweepable general purpose biquadratic digital filter. Syntax ares biquad asig, kb0, kb1, kb2, ka0, ka1, ka2 [, iskip] Initialization iskip (optional, default=0) -- if non-zero, intialization will be skipped. Default value 0. (New in Csound version 3.50) Performance asig -- input signal biquad is a general purpose biquadratic digital filter of the form: a0*y(n) + a1*y[n-1] + a2*y[n-2] = b0*x[n] + b1*x[n-1] + b2*x[n-2] This filter has the following frequency response: B(Z) b0 + b1*Z-1 + b2*Z-2 H(Z) = ---- = ------------------ A(Z) a0 + a1*Z-1 + a2*Z-2 This type of filter is often encountered in digital signal processing literature. It allows six user-defined k-rate coefficients. Examples Here is an example of the biquad opcode. It uses the file biquad.csd. Example of the biquad opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the biquad opcode used for modal synthesis. It uses the file biquad-2.csd. See the Modal Frequency Ratios appendix for other frequency ratios. Example of the biquad opcode for modal synthesis. See Also biquada, moogvcf, rezzy Credits Author: &namehans; October 1998 New in Csound version 3.49 manual_src~dfsg/opcodes/equals.xml0000644000000000000000000000710212262561504016415 0ustar rootroot : ? conditional expressions == Instrument Control:Conditional Values == == Compares two values for equality. Description Compares two values for equality. Syntax (a == b ? v1 : v2) where a, b, v1 and v2 may be expressions, but a, b not audio-rate. Performance In the above conditional, a and b are first compared. If the indicated relation is true (a is equal to b), then the conditional expression has the value of v1; if the relation is false, the expression has the value of v2. (For convenience, a sole "=" will function as "= =".) NB.: If v1 or v2 are expressions, these will be evaluated before the conditional is determined. In terms of binding strength, all conditional operators (i.e. the relational operators (<, etc.), and ?, and : ) are weaker than the arithmetic and logical operators (+, -, *, /, && and ||). These are operators not opcodes. Therefore, they can be used within orchestra statements, but do not form complete statements themselves. Examples Here is an example of the == operator. It uses the file equals.csd. Example of the == operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also >=, >, <=, <, != s manual_src~dfsg/opcodes/lposcilsa.xml0000644000000000000000000000564212262561504017123 0ustar rootroot lposcilsa Signal Generators:Sample Playback lposcilsa lposcilsa Read stereo sampled sound from a table with looping and high precision. Description lposcilsa reads stereo sampled sound from a table with looping and high precision. Syntax ar1, ar2 lposcilsa aamp, kfreqratio, kloop, kend, ift [,iphs] Initialization ift -- function table number iphs -- initial phase (in samples) Performance ar1, ar2 -- output signal aamp -- amplitude kfreqratio -- multiply factor of table frequency (for example: 1 = original frequency, 1.5 = a fifth up , .5 = an octave down) kloop -- start loop point (in samples) kend -- end loop point (in samples) lposcilsa is the same as lposcila, but works with stereo files loaded with GEN01. Examples Here is an example of the lposcilsa opcode. It uses the file lposcilsa.csd. Example of the lposcilsa opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also lposcil, lposcila, lposcilsa2 Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/STKMoog.xml0000644000000000000000000000676612262561504016425 0ustar rootroot STKMoog Signal Generators:STK Opcodes STKMoog STKMoog STKMoog produces moog-like swept filter sounds. Description STKMoog produces moog-like swept filter sounds, using one attack wave, one looped wave, and an ADSR envelope and adds two sweepable formant filters. Syntax asignal STKMoog ifrequency, iamplitude, [kq, kv1[, krate, kv2[, klfo, kv3[, klfodepth, kv4[, kvol, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kq -- controller 2, Q filter. Value range of kv1 is 0-127. krate -- controller 4, rate of filter sweep. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kvol -- controller 128, volume. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the Moog class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKMoog opcode. It uses the file STKMoog.csd. Example of the STKMoog opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/sum.xml0000644000000000000000000000334312262561504015732 0ustar rootroot sum Mathematical Operations:Opcode Equivalents of Functions sum sum Sums any number of a-rate signals. Description Sums any number of a-rate signals. Syntax ares sum asig1 [, asig2] [, asig3] [...] Performance asig1, asig2, ... -- a-rate signals to be summed (mixed or added). Examples Here is an example of the sum opcode. It uses the file sum.csd. Example of the sum opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; Italy April 1999 New in Csound version 3.54 manual_src~dfsg/opcodes/tablekt.xml0000644000000000000000000001206412262561504016554 0ustar rootroot tablekt Table Control:Dynamic Selection tablekt tablekt Provides k-rate control over table numbers. Description k-rate control over table numbers. The standard Csound opcode table when producing a k- or a-rate result, can only use an init-time variable to select the table number. tablekt accepts k-rate control as well as i-time. In all other respects they are similar to the original opcodes. Syntax ares tablekt xndx, kfn [, ixmode] [, ixoff] [, iwrap] kres tablekt kndx, kfn [, ixmode] [, ixoff] [, iwrap] Initialization ixmode -- if 0, xndx and ixoff ranges match the length of the table. if non-zero xndx and ixoff have a 0 to 1 range. Default is 0 ixoff -- if 0, total index is controlled directly by xndx, ie. the indexing starts from the start of the table. If non-zero, start indexing from somewhere else in the table. Value must be positive and less than the table length (ixmode = 0) or less than 1 (ixmode not equal to 0). Default is 0. iwrap -- if iwrap = 0, Limit mode: when total index is below 0, then final index is 0.Total index above table length results in a final index of the table length - high out of range total indexes stick at the upper limit of the table. If iwrap not equal to 0, Wrap mode: total index is wrapped modulo the table length so that all total indexes map into the table. For instance, in a table of length 8, xndx = 5 and ixoff = 6 gives a total index of 11, which wraps to a final index of 3. Default is 0. Performance kndx -- Index into table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode not equal to 0). xndx -- matching the table length (ixmode = 0) or a 0 to 1 range (ixmode not equal to 0) kfn -- Table number. Must be >= 1. Floats are rounded down to an integer. If a table number does not point to a valid table, or the table has not yet been loaded (GEN01) then an error will result and the instrument will be de-activated. Caution with k-rate table numbers At k-rate, if a table number of < 1 is given, or the table number points to a non-existent table, or to one which has a length of 0 (it is to be loaded from a file later) then an error will result and the instrument will be deactivated. kfn must be initialized at the appropriate rate using init. Attempting to load an i-rate value into kfn will result in an error. Examples Here is an example of the tablekt opcode. It uses the file tablekt.csd. Example of the tablekt opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i1 2.00000 i1 1.00000 i1 2.00000 i1 1.00000 .... See Also tableikt Credits Author: Robin Whittle Australia May 1997 New in version 3.47 manual_src~dfsg/opcodes/spat3d.xml0000644000000000000000000003401512262561504016324 0ustar rootroot spat3d Signal Modifiers:Panning and Spatialization spat3d spat3d Positions the input sound in a 3D space and allows moving the sound at k-rate. Description This opcode positions the input sound in a 3D space, with optional simulation of room acoustics, in various output formats. spat3d allows moving the sound at k-rate (this movement is interpolated internally to eliminate "zipper noise" if sr not equal to kr). Syntax aW, aX, aY, aZ spat3d ain, kX, kY, kZ, idist, ift, imode, imdel, iovr [, istor] Initialization idist -- For modes 0 to 3, idist is the unit circle distance in meters. For mode 4, idist is the distance between microphones. The following formulas describe amplitude and delay as a function of sound source distance from microphone(s): amplitude = 1 / (0.1 + distance) delay = distance / 340 (in seconds) Distance can be calculated as: distance = sqrt(iXˆ2 + iYˆ2 + iZˆ2) In Mode 4, distance can be calculated as: distance from left mic = sqrt((iX + idist/2)ˆ2 + iYˆ2 + iZˆ2) distance from right mic = sqrt((iX - idist/2)ˆ2 + iYˆ2 + iZˆ2) With spat3d the distance between the sound source and any microphone should be at least (340 * 18) / sr meters. Shorter distances will work, but may produce artifacts in some cases. There is no such limitation for spat3di and spat3dt. Sudden changes or discontinuities in sound source location can result in pops or clicks. Very fast movement may also degrade quality. ift -- Function table storing room parameters (for free field spatialization, set it to zero or negative). Table size is 54. The values in the table are: Room Parameter Purpose 0 Early reflection recursion depth (0 is the sound source, 1 is the first reflection etc.) for spat3d and spat3di. The number of echoes for four walls (front, back, right, left) is: N = (2*R + 2) * R. If all six walls are enabled: N = (((4*R + 6)*R + 8)*R) / 3 1 Late reflection recursion depth (used by spat3dt only). spat3dt skips early reflections and renders echoes up to this level. If early reflection depth is negative, spat3d and spat3di will output zero, while spat3dt will start rendering from the sound source. 2 imdel for spat3d. Overrides opcode parameter if non-negative. 3 irlen for spat3dt. Overrides opcode parameter if non-negative. 4 idist value. Overrides opcode parameter if >= 0. 5 Random seed (0 - 65535) -1 seeds from current time. 6 - 53 wall parameters (w = 6: ceil, w = 14: floor, w = 22: front, w = 30: back, w = 38: right, w = 46: left) w + 0 Enable reflections from this wall (0: no, 1: yes) w + 1 Wall distance from listener (in meters) w + 2 Randomization of wall distance (0 - 1) (in units of 1 / (wall distance)) w + 3 Reflection level (-1 - 1) w + 4 Parametric equalizer frequency in Hz. w + 5 Parametric equalizer level (1.0: no filtering) w + 6 Parametric equalizer Q (0.7071: no resonance) w + 7 Parametric equalizer mode (0: peak EQ, 1: low shelf, 2: high shelf) imode -- Output mode 0: B format with W output only (mono) aout = aW 1: B format with W and Y output (stereo) aleft = aW + 0.7071*aY aright = aW - 0.7071*aY 2: B format with W, X, and Y output (2D). This can be converted to UHJ: aWre, aWim hilbert aW aXre, aXim hilbert aX aYre, aYim hilbert aY aWXr = 0.0928*aXre + 0.4699*aWre aWXiYr = 0.2550*aXim - 0.1710*aWim + 0.3277*aYre aleft = aWXr + aWXiYr aright = aWXr - aWXiYr 3: B format with all outputs (3D) 4: Simulates a pair of microphones (stereo output) aW butterlp aW, ifreq ; recommended values for ifreq aY butterlp aY, ifreq ; are around 1000 Hz aleft = aW + aX aright = aY + aZ Mode 0 is the cheapest to calculate, while mode 4 is the most expensive. In Mode 4, The optional lowpass filters can change the frequency response depending on direction. For example, if the sound source is located left to the listener then the high frequencies are attenuated in the right channel and slightly increased in the left. This effect can be disabled by not using filters. You can also experiment with other filters (tone etc.) for better effect. Note that mode 4 is most useful for listening with headphones, and is also more expensive to calculate than the B-format (0 to 3) modes. The idist parameter in this case sets the distance between left and right microphone; for headphones, values between 0.2 - 0.25 are recommended, although higher settings up to 0.4 may be used for wide stereo effects. More information about B format can be found here: http://www.york.ac.uk/inst/mustech/3d_audio/ambis2.htm imdel -- Maximum delay time for spat3d in seconds. This has to be longer than the delay time of the latest reflection (depends on room dimensions, sound source distance, and recursion depth; using this formula gives a safe (although somewhat overestimated) value: imdel = (R + 1) * sqrt(W*W + H*H + D*D) / 340.0 where R is the recursion depth, W, H, and D are the width, height, and depth of the room, respectively). iovr -- Oversample ratio for spat3d (1 to 8). Setting it higher improves quality at the expense of memory and CPU usage. The recommended value is 2. istor (optional, default=0) -- Skip initialization if non-zero (default: 0). Performance aW, aX, aY, aZ -- Output signals   mode 0 mode 1 mode 2 mode 3 mode 4 aW W out W out W out W out left chn / low freq. aX 0 0 X out X out left chn / high frq. aY 0 Y out Y out Y out right chn / low frq. aZ 0 0 0 Z out right chn / high fr. ain -- Input signal kX, kY, kZ -- Sound source coordinates (in meters) If you encounter very slow performance (up to 100 times slower), it may be caused by denormals (this is also true of many other IIR opcodes, including butterlp, pareq, hilbert, and many others). Underflows can be avoided by: Using the denorm opcode on ain before spat3d. mixing low level DC or noise to the input signal, e.g. atmp rnd31 1/1e24, 0, 0 aW, aX, aY, aZ spa3di ain + atmp, ... or aW, aX, aY, aZ spa3di ain + 1/1e24, ... reducing irlen in the case of spat3dt (which does not have an input signal). A value of about 0.005 is suitable for most uses, although it also depends on EQ settings. If the equalizer is not used, irlen can be set to 0. Examples Here is a example of the spat3d opcode that outputs a stereo file. It uses the file spat3d_stereo.csd. Stereo example of the spat3d opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is a example of the spat3d opcode that outputs a UHJ file. It uses the file spat3d_UHJ.csd. UHJ example of the spat3d opcode. Here is a example of the spat3d opcode that outputs a quadrophonic file. It uses the file spat3d_quad.csd. Quadrophonic example of the spat3d opcode. See Also spat3di, spat3dt Credits Author: &nameistvan; 2001 New in version 4.12 Updated April 2002 by &nameistvan; manual_src~dfsg/opcodes/mediank.xml0000644000000000000000000000516512262561504016542 0ustar rootroot mediank Signal Modifiers:Standard Filters mediank mediank A median filter, a variant FIR lowpass filter. Description Implementation of a median filter. Syntax kres mediank kin, ksize, imaxsize [, iskip] Initialization imaxsize -- the maximun size of the window used to select the data. iskip -- initial disposition of internal data space. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance kin -- krate value to be filtered ksize -- size of the window over which the input is to be filtered. It must not exceed the maximum window size; if it does it is truncated. mediank is a simple filter that retuns the median value of the last ksize values. It has a lowpass action. The efficiency decreases as the window size increases. Examples Here is an example of the mediank opcode. It uses the file mediank.csd. Example of the mediank opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; University of Bath May 2010 New in Csound version 5.13 manual_src~dfsg/opcodes/flscrollend.xml0000644000000000000000000000370512262561504017437 0ustar rootroot FLscrollEnd FLTK:Containers FLscrollEnd FLscrollEnd A FLTK opcode that marks the end of an area with scrollbars. Description A FLTK opcode that marks the end of an area with scrollbars. Syntax FLscrollEnd Performance Containers are useful to format the graphic appearance of the widgets. The most important container is FLpanel, that actually creates a window. It can be filled with other containers and/or valuators or other kinds of widgets. There are no k-rate arguments in containers. See Also FLgroup, FLgroupEnd, FLpack, FLpackEnd, FLpanel, FLpanelEnd, FLscroll, FLtabs, FLtabsEnd Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/oscilikts.xml0000644000000000000000000000642512262561504017136 0ustar rootroot oscilikts Signal Generators:Basic Oscillators oscilikts oscilikts A linearly interpolated oscillator with sync status that allows changing the table number at k-rate. Description oscilikts is the same as oscilikt. Except it has a sync input that can be used to re-initialize the oscillator to a k-rate phase value. It is slower than oscilikt and osciliktp. Syntax ares oscilikts xamp, xcps, kfn, async, kphs [, istor] Initialization istor (optional, defaults to 0) -- skip initialization. Performance xamp -- amplitude. xcps -- frequency in Hz. Zero and negative values are allowed. However, the absolute value must be less than sr (and recommended to be less than sr/2). kfn -- function table number. Can be varied at control rate (useful to morph waveforms, or select from a set of band-limited tables generated by GEN30). async -- any positive value resets the phase of oscilikts to kphs. Zero or negative values have no effect. kphs -- sets the phase, initially and when it is re-initialized with async. Examples Here is an example of the oscilikts opcode. It uses the file oscilikts.csd. Example of the oscilikts opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also oscilikt and osciliktp. Credits Author: &nameistvan; New in version 4.22 manual_src~dfsg/opcodes/scale.xml0000644000000000000000000000461112262561504016214 0ustar rootroot scale Signal Generators:Linear and Exponential Generators scale scale Arbitrary signal scaling. Description Scales incoming value to user-definable range. Similar to scale object found in popular dataflow languages. Syntax kscl scale kinput, kmax, kmin Performance kin -- Input value. Can originate from any k-rate source as long as that source's output is in range 0-1. kmin -- Minimum value of the resultant scale operation. kmax -- Maximum value of the resultant scale operation. Examples Here is an example of the scale opcode. It uses the file scale.csd. Example of the scale opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also gainslider, logcurve, expcurve Credits Author: David Akbari October 2006 manual_src~dfsg/opcodes/flpanel_end.xml0000644000000000000000000000162212262561504017373 0ustar rootroot FLpanel_end FLTK:Containers FLpanel_end FLpanel_end Marks the end of a group of FLTK widgets contained inside of a window (panel). Description Marks the end of a group of FLTK widgets contained inside of a window (panel). This is another name for FLpanelEnd provided for compatibility. See FLpanelEnd Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/expsega.xml0000644000000000000000000000650312262561504016563 0ustar rootroot expsega Signal Generators:Linear and Exponential Generators expsega expsega An exponential segment generator operating at a-rate. Description An exponential segment generator operating at a-rate. This unit is almost identical to expseg, but more precise when defining segments with very short durations (i.e., in a percussive attack phase) at audio rate. Syntax ares expsega ia, idur1, ib [, idur2] [, ic] [...] Initialization ia -- starting value. Zero is illegal. ib, ic, etc. -- value after idur1 seconds, etc. must be non-zero and must agree in sign with ia. idur1 -- duration in seconds of first segment. A zero or negative value will cause all initialization to be skipped. idur2, idur3, etc. -- duration in seconds of subsequent segments. A zero or negative value will terminate the initialization process with the preceding point, permitting the last defined line or curve to be continued indefinitely in performance. The default is zero. Performance This unit generate audio signals whose values can pass through two or more specified points. The sum of dur values may or may not equal the instrument's performance time. A shorter performance will truncate the specified pattern, while a longer one will cause the last defined segment to continue on in the same direction. Examples Here is an example of the expsega opcode. It uses the file expsega.csd. Example of the expsega opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also expseg, expsegr Credits Author: &namegabriel; Example written by &namekevin;. New in Csound 3.57 manual_src~dfsg/opcodes/doppler.xml0000644000000000000000000000753012262561504016575 0ustar rootroot doppler Signal Modifiers:Standard Filters doppler doppler A fast and robust method for approximating sound propagation, achieving convincing Doppler shifts without having to solve equations. Description A fast and robust method for approximating sound propagation, achieving convincing Doppler shifts without having to solve equations. The method computes frequency shifts based on reading an input delay line at a delay time computed from the distance between source and mic and the speed of sound. One instance of the opcode is required for each dimension of space through which the sound source moves. If the source sound moves at a constant speed from in front of the microphone, through the microphone, to behind the microphone, then the output will be frequency shifted above the source frequency at a constant frequency while the source approaches, then discontinuously will be shifted below the source frequency at a constant frequency as the source recedes from the microphone. If the source sound moves at a constant speed through a point to one side of the microphone, then the rate of change of position will not be constant, and the familiar Doppler frequency shift typical of a siren or engine approaching and receding along a road beside a listener will be heard. Syntax ashifted doppler asource, ksourceposition, kmicposition [, isoundspeed, ifiltercutoff] Initialization isoundspeed (optional, default=340.29) -- Speed of sound in meters/second. ifiltercutoff (optional, default=6) -- Rate of updating the position smoothing filter, in cycles/second. Performance asource -- Input signal at the sound source. ksourceposition -- Position of the source sound in meters. The distance between source and mic should not be changed faster than about 3/4 the speed of sound. kmicposition -- Position of the recording microphone in meters. The distance between source and mic should not be changed faster than about 3/4 the speed of sound. Examples Here is an example of the doppler opcode. It uses the file doppler.csd. Example of the doppler opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author of algorithm: &namepeter; Author of opcode: &namemichael; January 2010 New in Csound version 5.11 manual_src~dfsg/opcodes/itablew.xml0000644000000000000000000000077112262561504016557 0ustar rootroot itablew itablew itablew Deprecated. Description Deprecated as of version 3.52. Use the tableiw opcode instead. manual_src~dfsg/opcodes/tb.xml0000644000000000000000000001702712262561504015537 0ustar rootroot tb15_init tb14_init tb13_init tb12_init tb11_init tb10_init tb9_init tb8_init tb7_init tb6_init tb5_init tb4_init tb3_init tb2_init tb1_init tb0_init tb15 tb14 tb13 tb12 tb11 tb10 tb9 tb8 tb7 tb6 tb5 tb4 tb3 tb2 tb1 tb0 Table Control:Table Queries tb tb0, tb1, tb2, tb3, tb4, tb5, tb6, tb7, tb8, tb9, tb10, tb11, tb12, tb13, tb14, tb15 tb0_init, tb1_init, tb2_init, tb3_init, tb4_init, tb5_init, tb6_init, tb7_init, tb8_init, tb9_init, tb10_init, tb11_init, tb12_init, tb13_init, tb14_init, tb15_init Table Read Access inside expressions. Description Allow to read tables in function fashion, to be used inside expressions. At present time Csound only supports functions with a single input argument. However, to access table elements, user must provide two numbers, i.e. the number of table and the index of element. So, in order to allow to access a table element with a function, a previous preparation step should be done. Syntax tb0_init ifn tb1_init ifn tb2_init ifn tb3_init ifn tb4_init ifn tb5_init ifn tb6_init ifn tb7_init ifn tb8_init ifn tb9_init ifn tb10_init ifn tb11_init ifn tb12_init ifn tb13_init ifn tb14_init ifn tb15_init ifn iout = tb0(iIndex) kout = tb0(kIndex) iout = tb1(iIndex) kout = tb1(kIndex) iout = tb2(iIndex) kout = tb2(kIndex) iout = tb3(iIndex) kout = tb3(kIndex) iout = tb4(iIndex) kout = tb4(kIndex) iout = tb5(iIndex) kout = tb5(kIndex) iout = tb6(iIndex) kout = tb6(kIndex) iout = tb7(iIndex) kout = tb7(kIndex) iout = tb8(iIndex) kout = tb8(kIndex) iout = tb9(iIndex) kout = tb9(kIndex) iout = tb10(iIndex) kout = tb10(kIndex) iout = tb11(iIndex) kout = tb11(kIndex) iout = tb12(iIndex) kout = tb12(kIndex) iout = tb13(iIndex) kout = tb13(kIndex) iout = tb14(iIndex) kout = tb14(kIndex) iout = tb15(iIndex) kout = tb15(kIndex) Performance There are 16 different opcodes whose name is associated with a number from 0 to 15. User can associate a specific table with each opcode (so the maximum number of tables that can be accessed in function fashion is 16). Prior to access a table, user must associate the table with one of the 16 opcodes by means of an opcode chosen among tb0_init, ..., tb15_init. For example, tb0_init 1 associates table 1 with tb0( ) function, so that, each element of table 1 can be accessed (in function fashion) with: kvar = tb0(k_some_index_of_table1) * k_some_other_var ivar = tb0(i_some_index_of_table1) + i_some_other_var etc... By using these opcodes, user can drastically reduce the number of lines of an orchestra, improving its readability. Credits Written by &namegabriel;. manual_src~dfsg/opcodes/foscil.xml0000644000000000000000000001101612262561504016401 0ustar rootroot foscil Signal Generators:FM Synthesis foscil foscil A basic frequency modulated oscillator. Description A basic frequency modulated oscillator. Syntax ares foscil xamp, kcps, xcar, xmod, kndx, ifn [, iphs] Initialization ifn -- function table number. Requires a wrap-around guard point. iphs (optional, default=0) -- initial phase of waveform in table ifn, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is 0. Performance xamp -- the amplitude of the output signal. kcps -- a common denominator, in cycles per second, for the carrier and modulating frequencies. xcar -- a factor that, when multiplied by the kcps parameter, gives the carrier frequency. xmod -- a factor that, when multiplied by the kcps parameter, gives the modulating frequency. kndx -- the modulation index. foscil is a composite unit that effectively banks two oscil opcodes in the familiar Chowning FM setup, wherein the audio-rate output of one generator is used to modulate the frequency input of another (the carrier). Effective carrier frequency = kcps * xcar, and modulating frequency = kcps * xmod. For integral values of xcar and xmod, the perceived fundamental will be the minimum positive value of kcps * (xcar - n * xmod), n = 0,1,2,... The input kndx is the index of modulation (usually time-varying and ranging 0 to 4 or so) which determines the spread of acoustic energy over the partial positions given by n = 0,1,2,.., etc. ifn should point to a stored sine wave. Previous to version 3.50, xcar and xmod could be k-rate only. The actual formula used for this implementation of FM synthesis is xamp * cos(2π * t * kcps * xcar + kndx * sin(2π * t * kcps * xmod) - π), assuming that the table is a sine wave. Examples Here is an example of the foscil opcode. It uses the file foscil.csd. Example of the foscil opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also More information about frequency modulation on Wikipedia: http://en.wikipedia.org/wiki/Frequency_modulation_synthesis manual_src~dfsg/opcodes/pareq.xml0000644000000000000000000000742612262561504016244 0ustar rootroot pareq Signal Modifiers:Specialized Filters pareq pareq Implementation of Zoelzer's parametric equalizer filters. Description Implementation of Zoelzer's parametric equalizer filters, with some modifications by the author. The formula for the low shelf filter is: omega = 2*pi*f/sr K = tan(omega/2) b0 = 1 + sqrt(2*V)*K + V*K^2 b1 = 2*(V*K^2 - 1) b2 = 1 - sqrt(2*V)*K + V*K^2 a0 = 1 + K/Q + K^2 a1 = 2*(K^2 - 1) a2 = 1 - K/Q + K^2 The formula for the high shelf filter is: omega = 2*pi*f/sr K = tan((pi-omega)/2) b0 = 1 + sqrt(2*V)*K + V*K^2 b1 = -2*(V*K^2 - 1) b1 = 1 - sqrt(2*V)*K + V*K^2 a0 = 1 + K/Q + K^2 a1 = -2*(K^2 - 1) a2 = 1 - K/Q + K^2 The formula for the peaking filter is: omega = 2*pi*f/sr K = tan(omega/2) b0 = 1 + V*K/2 + K^2 b1 = 2*(K^2 - 1) b2 = 1 - V*K/2 + K^2 a0 = 1 + K/Q + K^2 a1 = 2*(K^2 - 1) a2 = 1 - K/Q + K^2 Syntax ares pareq asig, kc, kv, kq [, imode] [, iskip] Initialization imode (optional, default: 0) -- operating mode 0 = Peaking 1 = Low Shelving 2 = High Shelving iskip (optional, default=0) -- if non zero skip the initialisation of the filter. (New in Csound version 4.23f13 and 5.0) Performance kc -- center frequency in peaking mode, corner frequency in shelving mode. kv -- amount of boost or cut. A value less than 1 is a cut. A value greater than 1 is a boost. A value of 1 is a flat response. kq -- Q of the filter (sqrt(.5) is no resonance) asig -- the incoming signal Examples Here is an example of the pareq opcode. It uses the file pareq.csd. Example of the pareq opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits &namehans; December 1998 New in Csound version 3.50 manual_src~dfsg/opcodes/serialWrite.xml0000644000000000000000000000613412262561504017421 0ustar rootroot serialrite Serial I/O serialWrite serialWrite Write data to a serial port. Description Write data to a serial port for arduino. Syntax serialWrite iPort, iByte serialWrite iPort, kByte serialWrite iPort, SBytes Performance iPort -- port number optained from a serialBegin opcode. iByte -- a byte of data to write. Examples Here is an example of the serialWrite opcode. It uses the file serialWrite.csd. Example of the serialWrite opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This is the matching Arduino code : void setup() { // enable serial communication Serial.begin(9600); // declare pin 9 to be an output: pinMode(9, OUTPUT); } void loop() { // only do something if we received something (this should be at csound's k-rate) if (Serial.available()) { // set the brightness of LED (connected to pin 9) to our input value int brightness = Serial.read(); analogWrite(9, brightness); // while we are here, get our knob value and send it to csound int sensorValue = analogRead(A0); Serial.write(sensorValue/4); // scale to 1-byte range (0-255) } } ..... See Also serialBegin, serialEnd, serialWrite_i, serialRead, serialPrint, serialFlush. Credits Author: &namematt; 2011 New in version 5.14 manual_src~dfsg/opcodes/event_i.xml0000644000000000000000000000761212262561504016562 0ustar rootroot event_i Instrument Control:Invocation event_i event_i Generates a score event from an instrument. Description Generates a score event from an instrument. Syntax event_i "scorechar", iinsnum, idelay, idur, [, ip4] [, ip5] [, ...] event_i "scorechar", "insname", idelay, idur, [, ip4] [, ip5] [, ...] Initialization scorechar -- A string (in double-quotes) representing the first p-field in a score statement. This is usually e, f, or i. insname -- A string (in double-quotes) representing a named instrument. iinsnum -- The instrument to use for the event. This corresponds to the first p-field, p1, in a score statement. idelay -- When (in seconds) the event will occur from the current performance time. This corresponds to the second p-field, p2, in a score statement. idur -- How long (in seconds) the event will happen. This corresponds to the third p-field, p3, in a score statement. ip4, ip5, ... (optional) -- Parameters representing additional p-field in a score statement. It starts with the fourth p-field, p4. Performance The event is added to the queue at initialisation time. Note that the event_i opcode cannot accept string p-fields. If you need to pass strings when instantiating an instrument, use the scoreline or scoreline_i opcode. Examples Here is an example of the event_i opcode. It uses the file event_i.csd. Example of the event_i opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also event, schedule, schedwhen, schedkwhen, schedkwhennamed, scoreline, scoreline_i Credits Written by &nameistvan;. New in Csound5 manual_src~dfsg/opcodes/outiat.xml0000644000000000000000000000736412262561504016442 0ustar rootroot outiat Real-time MIDI:Output outiat outiat Sends MIDI aftertouch messages at i-rate. Description Sends MIDI aftertouch messages at i-rate. Syntax outiat ichn, ivalue, imin, imax Initialization ichn -- MIDI channel number (1-16) ivalue -- floating point value imin -- minimum floating point value (converted in MIDI integer value 0) imax -- maximum floating point value (converted in MIDI integer value 127 (7 bit)) Performance outiat (i-rate aftertouch output) sends aftertouch messages. It works only with MIDI instruments which recognize them. It can drive a different value of a parameter for each note currently active. It can scale an i-value floating-point argument according to the imin and imax values. For example, set imin = 1.0 and imax = 2.0. When the ivalue argument receives a 2.0 value, the opcode will send a 127 value to the MIDI OUT device. When the ivalue argument receives a 1.0 value, it will send a 0 value. i-rate opcodes send their message once during instrument initialization. Examples Here is an example of the outiat opcode. It uses the file outiat.csd. Example of the outiat opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also outic14, outic, outipat, outipb, outipc, outkat, outkc14, outkc, outkpat, outkpb, outkpc Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/pvsblur.xml0000644000000000000000000000616512262561504016630 0ustar rootroot pvsblur Spectral Processing:Streaming pvsblur pvsblur Average the amp/freq time functions of each analysis channel for a specified time. Description Average the amp/freq time functions of each analysis channel for a specified time (truncated to number of frames). As a side-effect the input pvoc stream will be delayed by that amount. Syntax fsig pvsblur fsigin, kblurtime, imaxdel Performance fsig -- output pv stream fsigin -- input pv stream. kblurtime -- time in secs during which windows will be averaged . imaxdel -- maximum delay time, used for allocating memory used in the averaging operation. This opcode will blur a pvstream by smoothing the amplitude and frequency time functions (a type of low-pass filtering); the amount of blur will depend on the length of the averaging period, larger blurtimes will result in a more pronounced effect. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the use of the pvsblur opcode. It uses the file pvsblur.csd. Example of the <emphasis>pvsblur</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namevictor; November 2004 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/aexprand.xml0000644000000000000000000000077612262561504016737 0ustar rootroot aexprand aexprand aexprand Deprecated. Description Deprecated as of version 3.49. Use the exprand opcode instead. manual_src~dfsg/opcodes/stix.xml0000644000000000000000000000727112262561504016121 0ustar rootroot stix Signal Generators:Models and Emulations stix stix Semi-physical model of a stick sound. Description stix is a semi-physical model of a stick sound. It is one of the PhISEM percussion opcodes. PhISEM (Physically Informed Stochastic Event Modeling) is an algorithmic approach for simulating collisions of multiple independent sound producing objects. Syntax ares stix iamp, idettack [, inum] [, idamp] [, imaxshake] Initialization iamp -- Amplitude of output. Note: As these instruments are stochastic, this is only a approximation. idettack -- period of time over which all sound is stopped inum (optional) -- The number of beads, teeth, bells, timbrels, etc. If zero, the default value is 30. idamp (optional) -- the damping factor, as part of this equation: damping_amount = 0.998 + (idamp * 0.002) The default damping_amount is 0.998 which means that the default value of idamp is 0. The maximum damping_amount is 1.0 (no damping). This means the maximum value for idamp is 1.0. The recommended range for idamp is usually below 75% of the maximum value. imaxshake (optional) -- amount of energy to add back into the system. The value should be in range 0 to 1. Examples Here is an example of the stix opcode. It uses the file stix.csd. Example of the stix opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cabasa, crunch, sandpaper, sekere Credits Author: &nameperry;, part of the PhOLIES (Physically-Oriented Library of Imitated Environmental Sounds) Adapted by &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.07 Added notes by &namerasmus; on May 2002. manual_src~dfsg/opcodes/vmap.xml0000644000000000000000000001014212262561504016064 0ustar rootroot vmap Vectorial:Vectorial operations vmap vmap Maps elements from a vector according to indexes contained in another vector. Description Maps elements from a vector onto another according to the indexes of a this vector. Syntax vmap ifn1, ifn2, ielements [,idstoffset, isrcoffset] Initialization ifn1 - number of the table where the vectorial signal will be copied, and which contains the mapping vector ifn2 - number of the table hosting the vectorial signal to be copied ielements - number of elements to process idstoffset - index offset for destination table (ifn1) isrcoffset - index offset for source table (ifn2) Performance vmap maps elements of ifn2 according to the values of table ifn1. Elements of ifn1 are treated as indexes of table ifn2, so element values of ifn1 must not exceed the length of ifn2 table otherwise a Csound will report an error. Elements of ifn1 are treated as integers, so any fractional part will be truncated. There is no interpolation performed on this operation. In practice, what happens is that the elements of ifn1 are used as indexes to ifn2, and then are replaced by the corresponding elements from ifn2. ifn1 must be different from ifn2, otherwise the results are unpredictable. Csound will produce an init error if they are not. All these operators (vaddv, vsubv, vmultv, vdivv, vpowv, vexpv, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2, etc. Note: bmscan not yet available on Canonical Csound Examples Here is an example of the vmap opcode. It uses the file vmap.csd. Example of the vmap opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/midinoteonkey.xml0000644000000000000000000001106012262561504017777 0ustar rootroot midinoteonkey Real-time MIDI:MIDI/Score Interoperability midinoteonkey midinoteonkey Gets a MIDI note number value. Description midinoteonkey is designed to simplify writing instruments that can be used interchangeably for either score or MIDI input, and to make it easier to adapt instruments originally written for score input to work with MIDI input. In general, it should be possible to write instrument definitions that work identically with both scores and MIDI, including both MIDI files and real-time MIDI input, without using any conditional statements, and that take full advantage of MIDI voice messages. Note that correlating Csound instruments with MIDI channel numbers is done using the massign opcode for real-time performance,. For file-driven performance, instrument numbers default to MIDI channel number + 1, but the defaults are overridden by any MIDI program change messages in the file. Syntax midinoteonkey xkey, xvelocity Performance xkey -- returns MIDI key during MIDI activation, remains unchanged otherwise. xvelocity -- returns MIDI velocity during MIDI activation, remains unchanged otherwise. If the instrument was activated by MIDI input, the opcode overwrites the values of xkey and xvelocity with the corresponding values from MIDI input. If the instrument was NOT activated by MIDI input, the values of xkey and xvelocity remain unchanged. This enables score p-fields to receive MIDI input data during MIDI activation, and score values otherwise. Adapting a score-activated Csound instrument. See the MIDI interop opcodes section for details on adapting score driven instruments for MIDI or vice-versa. Examples Here is an example of the midinoteonkey opcode. It uses the file midinoteonkey.csd. Example of the midinoteonkey opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i1 60.00000 i1 69.00000 See Also midichannelaftertouch, midicontrolchange, mididefault, midinoteoff, midinoteoncps, midinoteonoct, midinoteonpch, midipitchbend, midipolyaftertouch, midiprogramchange Credits Author: Michael Gogins New in version 4.20 manual_src~dfsg/opcodes/oscil1.xml0000644000000000000000000000660312262561504016322 0ustar rootroot oscil1 Signal Generators:Table Access oscil1 oscil1 Accesses table values by incremental sampling. Description Accesses table values by incremental sampling. Syntax kres oscil1 idel, kamp, idur [, ifn] Initialization idel -- delay in seconds before oscil1 incremental sampling begins. idur -- duration in seconds to sample through the oscil1 table just once. A zero or negative value will cause all initialization to be skipped. ifn -- (optional) function table number. tablei, oscil1i require the extended guard point. The number defaults to -1 which indicates a sinewave. Performance kamp -- amplitude factor. oscil1 accesses values by sampling once through the function table at a rate determined by idur. For the first idel seconds, the point of scan will reside at the first location of the table; it will then begin moving through the table at a constant rate, reaching the end in another idur seconds; from that time on (i.e. after idel + idur seconds) it will remain pointing at the last location. Each value obtained from sampling is then multiplied by an amplitude factor kamp before being written into the result. Examples Here is an example of the oscil1 opcode. It uses the file oscil1.csd. Example of the oscil1 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example will produce the following output: i1 0.50000 i1 0.20000 i1 0.80000 i1 0.10000 i1 0.90000 i1 0.00000 i1 1.00000 i1 0.50000 See Also table, tablei, table3, oscil1i, osciln manual_src~dfsg/opcodes/kbexprnd.xml0000644000000000000000000000077612262561504016752 0ustar rootroot kbexprnd kbexprnd kbexprnd Deprecated. Description Deprecated as of version 3.49. Use the bexprnd opcode instead. manual_src~dfsg/opcodes/koutat.xml0000644000000000000000000000076212262561504016437 0ustar rootroot koutat koutat koutat Deprecated. Description Deprecated as of version 3.52. Use the outkat opcode instead. manual_src~dfsg/opcodes/flpack.xml0000644000000000000000000001270112262561504016364 0ustar rootroot FLpack FLTK:Containers FLpack FLpack Provides the functionality of compressing and aligning FLTK widgets. Description FLpack provides the functionality of compressing and aligning widgets. Syntax FLpack iwidth, iheight, ix, iy, itype, ispace, iborder Initialization iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). itype -- an integer number that modifies the appearance of the target widget. The itype argument expresses the type of packing: 0 - vertical 1 - horizontal ispace -- sets the space between the widgets. iborder -- border type of the container. It is expressed by means of an integer number chosen from the following: 0 - no border 1 - down box border 2 - up box border 3 - engraved border 4 - embossed border 5 - black line border 6 - thin down border 7 - thin up border Performance FLpack provides the functionality of compressing and aligning widgets. Containers are useful to format the graphic appearance of the widgets. The most important container is FLpanel, that actually creates a window. It can be filled with other containers and/or valuators or other kinds of widgets. There are no k-rate arguments in containers. Examples The following example: FLpanel "Panel1",450,300,100,100 FLpack 400,300, 10,40,0,15,3 gk1, ihs1 FLslider "FLslider 1", 500, 1000, 2 ,1, -1, 300,15, 20,50 gk2, ihs2 FLslider "FLslider 2", 300, 5000, 2 ,3, -1, 300,15, 20,100 gk3, ihs3 FLslider "FLslider 3", 350, 1000, 2 ,5, -1, 300,15, 20,150 gk4, ihs4 FLslider "FLslider 4", 250, 5000, 1 ,11, -1, 300,30, 20,200 gk5, ihs5 FLslider "FLslider 5", 220, 8000, 2 ,1, -1, 300,15, 20,250 gk6, ihs6 FLslider "FLslider 6", 1, 5000, 1 ,13, -1, 300,15, 20,300 gk7, ihs7 FLslider "FLslider 7", 870, 5000, 1 ,15, -1, 300,30, 20,350 FLpackEnd FLpanelEnd ...will produce this result, when resizing the window: FLpack. FLpack. See Also FLgroup, FLgroupEnd, FLpackEnd, FLpanel, FLpanelEnd, FLscroll, FLscrollEnd, FLtabs, FLtabsEnd Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/filter2.xml0000644000000000000000000000766712262561504016512 0ustar rootroot filter2 Signal Modifiers:Specialized Filters filter2 filter2 Performs filtering using a transposed form-II digital filter lattice with no time-varying control. Description General purpose custom filter with no time-varying pole control. The filter coefficients implement the following difference equation: (1)*y(n) = b0*x[n] + b1*x[n-1] +...+ bM*x[n-M] - a1*y[n-1] -...- aN*y[n-N] the system function for which is represented by: B(Z) b0 + b1*Z-1 + ... + bM*Z-M H(Z) = ---- = -------------------------- A(Z) 1 + a1*Z-1 + ... + aN*Z-N Syntax ares filter2 asig, iM, iN, ib0, ib1, ..., ibM, ia1, ia2, ..., iaN kres filter2 ksig, iM, iN, ib0, ib1, ..., ibM, ia1, ia2, ..., iaN Initialization At initialization the number of zeros and poles of the filter are specified along with the corresponding zero and pole coefficients. The coefficients must be obtained by an external filter-design application such as Matlab and specified directly or loaded into a table via GEN01. Performance The filter2 opcodes perform filtering using a transposed form-II digital filter lattice with no time-varying control. Since filter2 implements generalized recursive filters, it can be used to specify a large range of general DSP algorithms. For example, a digital waveguide can be implemented for musical instrument modeling using a pair of delayr and delayw opcodes in conjunction with the filter2 opcode. Examples A first-order linear-phase lowpass FIR filter operating on a k-rate signal: k1 filter2 ksig, 2, 0, 0.5, 0.5 ;; k-rate FIR filter Here is another example of the filter2 opcode. It uses the file filter2.csd. Example of the filter2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zfilter2 Credits Author: Michael A. Casey M.I.T. Cambridge, Mass. 1997 New in version 3.47 manual_src~dfsg/opcodes/instimek.xml0000644000000000000000000000120312262561504016742 0ustar rootroot instimek instimek instimek Deprecated. Description Deprecated as of version 3.62. Use the timeinstk opcode instead. Credits &namedavid; originally pointed out this deprecated name. manual_src~dfsg/opcodes/cossegb.xml0000644000000000000000000000663112262561504016556 0ustar rootroot cossegb Signal Generators:Linear and Exponential Generators cossegb cossegb Trace a series of line segments between specified absolute points with cosine interpolation. Description Trace a series of line segments between specified absolute points with cosine interpolation. Syntax ares cossegb ia, itim1, ib [, itim2] [, ic] [...] kres cossegb ia, itim1, ib [, itim2] [, ic] [...] Initialization ia -- starting value. ib, ic, etc. -- value at tim1 seconds, etc. itim1 -- time in seconds of end of first segment. A zero or negative value will cause all initialization to be skipped. itim2, itim3, etc. -- time in seconds at the end of subsequent segments. Performance These units generate control or audio signals whose values can pass through 2 or more specified points. The last tim value may or may not equal the instrument's performance time: a shorter performance will truncate the specified pattern, while a longer one will cause the last value to be repeated until the end of the note. Examples Here is an example of the cossegb opcode. It uses the file cossegb.csd. Example of the cossegb opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cosseg, expon, expseg, expsegr, line, linseg linsegb linsegr transeg Credits Author: &namejohn; Aug 2012 New in version 5.18 manual_src~dfsg/opcodes/slider16table.xml0000644000000000000000000001404012262561504017563 0ustar rootroot slider16table Real-time MIDI:Slider Banks slider16table slider16table Stores a bank of 16 different MIDI control messages to a table. Description Stores a bank of 16 different MIDI control messages to a table. Syntax kflag slider16table ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, .... , ictlnum16, imin16, imax16, init16, ifn16 Initialization ichan -- MIDI channel (1-16) ioutTable -- number of the table that will contain the output ioffset -- output table offset. A zero means that the output of the first slider will affect the first table element. A 10 means that the output of the first slider will affect the 11th table element. ictlnum1 ... ictlnum16 -- MIDI control number (0-127) imin1 ... imin16 -- minimum values for each controller imax1 ... imax16 -- maximum values for each controller init1 ... init16 -- initial value for each controller ifn1 ... ifn16 -- function table for conversion for each controller Performance kflag -- a flag that informs if any control-change message in the bank has been received. In this case kflag is set to 1. Otherwise is set to zero. slider16table is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider16table allows a bank of 16 different MIDI control message numbers. As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. slider16table is very similar to slider16 and sliderN family of opcodes (see their manual for more information). The actual difference is that the output is not stored to k-rate variables, but to a table, denoted by the ioutTable argument. It is possible to define a starting index in order to use the same table for more than one slider bank (or other purposes). It is possible to use this opcode together with FLslidBnk2Setk and FLslidBnk2, so you can synchronize the position of the MIDI values to the position of the FLTK valuator widgets of FLslidBnk2. Notice that you have to specify the same min/max values as well the linear/exponential responses in both sliderNtable(f) and FLslidBnk2. The exception is when using table-indexed response instead of a lin/exp response. In this case, in order to achieve a useful result, the table-indexed response and actual min/max values must be set only in FLslidBnk2, whereas, in sliderNtable(f), you have to set a linear response and a minimum of zero and a maximum of one in all sliders. See Also slider16tablef, slider32table, slider32tablef, slider64table, slider64tablef, slidertable8, slider8tablef Credits Author: &namegabriel; New in Csound version 5.06 manual_src~dfsg/opcodes/chuap.xml0000644000000000000000000001216412262561504016227 0ustar rootroot chuap Signal Generators:Models and Emulations chuap chuap Simulates Chua's oscillator, an LRC oscillator with an active resistor, proved capable of bifurcation and chaotic attractors, with k-rate control of circuit elements. Description Simulates Chua's oscillator, an LRC oscillator with an active resistor, proved capable of bifurcation and chaotic attractors, with k-rate control of circuit elements. Syntax aI3, aV2, aV1 chuap kL, kR0, kC1, kG, kGa, kGb, kE, kC2, iI3, iV2, iV1, ktime_step Initialization iI3 -- Initial current at G iV2 -- Initial voltage at C2 iV1 -- Initial voltage at C1 Performance kL -- Inductor L kR0 -- Resistor R0 kC1 -- Capacitor C1 kG -- Resistor G kGa -- Resistor V (nonlinearity term) kGb -- Resistor V (nonlinearity term) kGb -- Resistor V (nonlinearity term) ktime_step -- Delta time in the difference equation, can be used to more or less control pitch. Chua's oscillator is a simple LRC oscillator with an active resistor. The oscillator can be driven into period bifurcation, and thus to chaos, because of the nonlinear response of the active resistor. Diagram of Chua's Oscillator Circuit The circuit is described by a set of three ordinary differential equations called Chua's equations: dI3 R0 1 --- = - -- I3 - - V2 dt L L dV2 1 G --- = -- I3 - -- (V2 - V1) dt C2 C2 dV1 G 1 --- = -- (V2 - V1) - -- f(V1) dt C1 C1 where f() is a piecewise discontinuity simulating the active resistor: f(V1) = Gb V1 + - (Ga - Gb)(|V1 + E| - |V1 - E|) A solution of these equations (I3,V2,V1)(t) starting from an initial state (I3,V2,V1)(0) is called a trajectory of Chua's oscillator. The Csound implementation is a difference equation simulation of Chua's oscillator with Runge-Kutta integration. This algorithm uses internal non linear feedback loops which causes audio result to depend on the orchestra sampling rate. For example, if you develop a project with sr=48000Hz and if you want to produce an audio CD from it, you should record a file with sr=48000Hz and then downsample the file to 44100Hz using the srconv utility. Be careful! Some sets of parameters will produce amplitude spikes or positive feedback that could damage your speakers. Examples Here is an example of the chuap opcode. It uses the file chuap.csd. Example of the chuap opcode. Credits Inventor of Chua's oscillator: Leon O. Chua Author of MATLAB simulation: James Patrick McEvoy MATLAB Adventures in Bifurcations and Chaos (ABC++) Author of Csound port: Michael Gogins New in Csound version 5.09 Note added by &namepinot;, August 2009 manual_src~dfsg/opcodes/active.xml0000644000000000000000000001053112262561504016376 0ustar rootroot active Instrument Control:Realtime Performance Control active active Returns the number of active instances of an instrument. Description Returns the number of active instances of an instrument. Syntax ir active insnum [,iopt] ir active Sinsname [,iopt] kres active kinsnum [,iopt] Initialization insnum -- number or string name of the instrument to be reported Sinsname -- instrument name iopt -- select currently active (zero, default), or all every active (non zero) Performance kinsnum -- number or string name of the instrument to be reported active returns the number of active instances of instrument number insnum/kinsnum (or named instrument Sinsname). As of Csound 4.17 the output is updated at k-rate (if input arg is k-rate), to allow running count of instr instances. As of Csound 5.17 if the instrument number is given as zero then all instruments are counted. Examples Here is a simple example of the active opcode. It uses the file active.csd. Simple example of the active opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 2: icount = 1.000 instr 2: icount = 2.000 Here is a more advanced example of the active opcode. It displays the results of the active opcode at k-rate instead of i-rate. It uses the file active_k.csd. Example of the active opcode at k-rate. Its output should include lines like: i2 1.00000 i2 2.00000 Here is another example of the active opcode, using the number of instances to calculate gain. It uses the file active_scale.csd. Example of the active opcode at k-rate. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK July, 1999 Examples written by &namekevin;. New in Csound version 3.57; named instruments added version 5.13 Option for all ever active new in 5.13 Count of all instruments new in 5.17 manual_src~dfsg/opcodes/peak.xml0000644000000000000000000000476612262561504016060 0ustar rootroot peak Instrument Control:Sensing and Control peak peak Maintains the output equal to the highest absolute value received. Description These opcodes maintain the output k-rate variable as the peak absolute level so far received. Syntax kres peak asig kres peak ksig Performance kres -- Output equal to the highest absolute value received so far. This is effectively an input to the opcode as well, since it reads kres in order to decide whether to write something higher into it. ksig -- k-rate input signal. asig -- a-rate input signal. Examples Here is an example of the peak opcode. It uses the file peak.csd, and beats.wav. Example of the peak opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: i 1 time 0.00002: 4835.00000 i 1 time 1.00002: 29312.00000 i 1 time 2.00002: 32767.00000 Credits Author: Robin Whittle Australia May 1997 Example written by &namekevin;. manual_src~dfsg/opcodes/faustaudio.xml0000644000000000000000000000344512262561504017275 0ustar rootroot faustaudio Faust Opcodes faustaudio faustaudio Instantiates and runs a compiled Faust program. Description Faustaudio will instantiate and run a Faust program compiled with faustcompile. Syntax ihandle,a1[,a2,...] faustaudio ifac[,ain1,...] Initialization ifac -- a handle to a compiled Faust program, produced by faustcompile. ihandle -- a handle to the Faust DSP instance, which can be used to access its controls with faustctl. Performance ain1,... -- input signals a1,... -- output signals Examples Here is an example of the faustaudio opcode, with a simple program that mixes two inputs: ifac faustcompile "process=+;", "-vec -lv 1" idsp,a1 faustaudio ifac,ain1,ain2 Credits Author: &namevictor;, 2013 manual_src~dfsg/opcodes/STKSitar.xml0000644000000000000000000000417312262561504016574 0ustar rootroot STKSitar Signal Generators:STK Opcodes STKSitar STKSitar STKSitar uses a plucked string physical model. Description STKSitar uses a plucked string physical model based on the Karplus-Strong algorithm. Syntax asignal STKSitar ifrequency, iamplitude Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Note The code for this opcode is taken directly from the Sitar class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html Examples Here is an example of the STKSitar opcode. It uses the file STKSitar.csd. Example of the STKSitar opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/ATSaddnz.xml0000644000000000000000000001242512262561504016577 0ustar rootroot ATSaddnz Spectral Processing:ATS ATSaddnz ATSaddnz uses the data from an ATS analysis file to perform noise resynthesis. Description ATSaddnz reads from an ATS analysis file and uses the data to perform additive synthesis using a modified randi function. Syntax ar ATSaddnz ktimepnt, iatsfile, ibands[, ibandoffset, ibandincr] Initialization iatsfile – the ATS number (n in ats.n) or the name in quotes of the analysis file made using ATS. ibands – number of noise bands that will be used in the resynthesis (the noise has a maximum of 25 bands) ibandoffset (optional) – is the first noise band used (defaults to 0). ibandincr (optional) – sets an increment by which these synthesis opcodes counts up from ibandoffset for ibins components in the re-synthesis (defaults to 1). Performance ktimepnt – The time pointer in seconds used to index the ATS file. Used for ATSaddnz exactly the same as for pvoc and ATSadd. ATSaddnz and ATSadd are based on pvadd by Richard Karpen and use files created by Juan Pampin's ATS (Analysis - Transformation - Synthesis). ATSaddnz also reads from an ATS file but it resynthesizes the noise from noise energy data contained in the ATS file. It uses a modified randi function to create band limited noise and modulates that with a cosine wave, to synthesize a user specified selection of frequency bands. Modulating the noise is required to put the band limited noise in the correct place in the frequency spectrum. Examples ktime line 0, p3, 2.5 asig ATSaddnz ktime, "clarinet.ats", 25 In the example above we're synthesizing all 25 noise bands from the data contained in the ATS analysis file called "clarinet.ats". Examples Here is a complete example of the ATSaddnz opcode. It uses the file ATSaddnz.csd. Example of the ATSaddnz opcode. Here we synthesize only the 25th noise band (ibandoffset of 24 and ibands of 1). Here is another example of the ATSaddnz opcode. It uses the file ATSaddnz-2.csd. Example 2 of the ATSaddnz opcode. See also ATSread, ATSreadnz, ATSinfo, ATSbufread, ATScross, ATSinterpread, ATSpartialtap, ATSaddnz, ATSsinnoi Credits Author: Alex Norman Seattle,Washington 2004 manual_src~dfsg/opcodes/diskin2.xml0000644000000000000000000002367512262561504016503 0ustar rootroot diskin2 Signal I/O:Signal Input diskin2 diskin2 Reads audio data from a file, and can alter its pitch using one of several available interpolation types, as well as convert the sample rate to match the orchestra sr setting. Description Reads audio data from a file, and can alter its pitch using one of several available interpolation types, as well as convert the sample rate to match the orchestra sr setting. diskin2 can also read multichannel files with any number of channels in the range 1 to 24 in versions before 5.14, and 40 after. Syntax a1[, a2[, ... aN]] diskin2 ifilcod, kpitch[, iskiptim \ [, iwrap[, iformat [, iwsize[, ibufsize[, iskipinit]]]]]] ar1[] diskin2 ifilcod, kpitch[, iskiptim \ [, iwrap[, iformat [, iwsize[, ibufsize[, iskipinit]]]]]] (in the array output version, the number of output channels does not have an upper limit.) Initialization ifilcod -- integer or character-string denoting the source soundfile name. An integer denotes the file soundin.ifilcod; a character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the named file is sought first in the current directory, then in those given by the environment variable SSDIR (if defined) then by SFDIR. See also GEN01. Note: files longer than 231-1 sample frames may not be played correctly on 32-bit platforms; this means a maximum length about 3 hours with a sample rate of 192000 Hz. iskiptim (optional, defaults to zero) -- time in seconds of input sound to be skipped, assuming kpitch=1. Can be negative, to add -iskiptim/kpitch seconds of delay instead of skipping sound. Note If iwrap is not 0 (locations are wrapped), iskiptim will not delay the sound if a negative value is used. Instead, the negative value will be "wrapped" from the end of the file. iwrap (optional, defaults to zero) -- if set to any non-zero value, read locations that are negative or are beyond the end of the file are wrapped to the duration of the sound file instead of assuming zero samples. Useful for playing a file in a loop. Note If iwrap is enabled, the file length should not be shorter than the interpolation window size (see below), otherwise there may be clicks in the sound output. iformat (optional, defaults to zero) -- sample format, for raw (headerless) files only. This parameter is ignored if the file has a header. Allowed values are: 0: 16-bit short integers 1: 8-bit signed char (high-order 8 bits of a 16-bit integer) 2: 8-bit A-law bytes 3: 8-bit U-law bytes 4: 16-bit short integers 5: 32-bit long integers 6: 32-bit floats 7: 8-bit unsigned int 8: 24-bit int 9: 64-bit doubles iwsize (optional, defaults to zero) -- interpolation window size, in samples. Can be one of the following: 1: round to nearest sample (no interpolation, for kpitch=1) 2: linear interpolation 4: cubic interpolation >= 8: iwsize point sinc interpolation with anti-aliasing (slow) Zero or negative values select the default, which is cubic interpolation. Note If interpolation is used, kpitch is automatically scaled by the ratio of the sample rate of the sound file and the orchestra, so that the file will always be played at the original pitch if kpitch is 1. However, the sample rate conversion is disabled if iwsize is 1. ibufsize (optional, defaults to 0) -- buffer size in mono samples (not sample frames). This is only the suggested value, the actual setting will be rounded so that the number of sample frames is an integer power of two and is in the range 128 (or iwsize if greater than 128) to 1048576. The default, which is 4096, and is enabled by zero or negative values, should be suitable for most uses, but for non-realtime mixing of many large sound files, a high buffer setting is recommended to improve the efficiency of disk reads. For real time audio output, reading the files from a fast RAM file system (on platforms where this option is available) with a small buffer size may be preferred. iskipinit (optional, defaults to 0) -- skip initialization if set to any non-zero value. Performance a1 ... a24 -- output signals, in the range -0dbfs to 0dbfs. Any samples before the beginning (i.e. negative location) and after the end of the file are assumed to be zero, unless iwrap is non-zero. The number of output arguments must be the same as the number of sound file channels - which can be determined with the filenchnls opcode, otherwise an init error will occur. ar1[] --- output signals, in a-rate array of size N, where N is the number of channels in the file. Arrays are automatically allocated to the correct size. Note It is more efficient to read a single file with many channels, than many files with only a single channel, especially with high iwsize settings. kpitch -- transpose the pitch of input sound by this factor (e.g. 0.5 means one octave lower, 2 is one octave higher, and 1 is the original pitch). Fractional and negative values are allowed (the latter results in playing the file backwards, however, in this case the skip time parameter should be set to some positive value, e.g. the length of the file, or iwrap should be non-zero, otherwise nothing would be played). If interpolation is enabled, and the sample rate of the file differs from the orchestra sample rate, the transpose ratio is automatically adjusted to make sure that kpitch=1 plays at the original pitch. Using a high iwsize setting (40 or more) can significantly improve sound quality when transposing up, although at the expense of high CPU usage. Examples Here is an example of the diskin2 opcode. It uses the file diskin2.csd, beats.wav. Example of the diskin2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also in, inh, ino, inq, ins, soundin and diskin Credits Author: Istvan Varga 2005 New in version 5.00 manual_src~dfsg/opcodes/jspline.xml0000644000000000000000000000675512262561504016604 0ustar rootroot jspline Signal Generators:Linear and Exponential Generators jspline jspline A jitter-spline generator. Description A jitter-spline generator. Syntax ares jspline xamp, kcpsMin, kcpsMax kres jspline kamp, kcpsMin, kcpsMax Performance kres, ares -- Output signal xamp -- Amplitude factor kcpsMin, kcpsMax -- Range of point-generation rate. Min and max limits are expressed in cps. jspline (jitter-spline generator) generates a smooth curve based on random points generated at [cpsMin, cpsMax] rate. This opcode is similar to randomi or randi or jitter, but segments are not straight lines, but cubic spline curves. Output value range is approximately > -xamp and < xamp. Actually, real range could be a bit greater, because of interpolating curves beetween each pair of random-points. At present time generated curves are quite smooth when cpsMin is not too different from cpsMax. When cpsMin-cpsMax interval is big, some little discontinuity could occurr, but it should not be a problem, in most cases. Maybe the algorithm will be improved in next versions. These opcodes are often better than jitter when user wants to naturalize or analogize digital sounds. They could be used also in algorithmic composition, to generate smooth random melodic lines when used together with samphold opcode. Note that the result is quite different from the one obtained by filtering white noise, and they allow the user to obtain a much more precise control. Examples Here is an example of the jspline opcode. It uses the file jspline.csd. Example of the jspline opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; New in Version 4.15 manual_src~dfsg/opcodes/JackoInit.xml0000644000000000000000000001042312262561504016776 0ustar rootroot JackoInit Jacko Opcodes JackoInit JackoInit Initializes Csound as a Jack client. Description Initializes this instance of Csound as a Jack client. Csound's sr must be equal to the Jack daemon's frames per second. Csound's ksmps must be equal to the Jack daemon's frames per period. Frames per period must not only (a) be a power of 2, but also (b) go evenly into the frames per second, e.g. 128 frames per period goes into 48000 frames per second 375 times, for a latency or MIDI time granularity of about 2.7 milliseconds (as good as or better than the absolute best human performers). The order of processing of all signals that pass from Jack input ports, through Csound processing, and to Jack output ports, must be properly determined by sequence of instrument and opcode definition within Csound. Syntax JackoInit SclientName, ServerName Initialization Sname -- String name of the inlet port. The name of the inlet is implicitly qualified by the instrument name or number, so it is valid to use the same inlet name in more than one instrument (but not to use the same inlet name twice in one instrument). SclientName -- The name of the Jack client; normally, should be "csound". ServerName -- The name of the Jack daemon; normally, will be "default". This opcode must be called once and only once in the orchestra header, and before any other Jack opcodes. If more than one instance of Csound is using the Jack opcodes at the same time, then each instance of Csound must use a different client name. Examples Here is an example of the JackoInit opcode. It uses the file JackoInit.csd. Example of the JackoInit opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also JackoInfo, JackoFreewheel, JackoAudioInConnect, JackoAudioOutConnect, JackoMidiInConnect, JackoMidiOutConnect, JackoOn, JackoAudioIn, JackoAudioOut, JackoMidiOut, JackoNoteOut, JackoTransport. Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/flgroup.xml0000644000000000000000000000774512262561504016616 0ustar rootroot FLgroup FLTK:Containers FLgroup FLgroup A FLTK container opcode that groups child widgets. Description A FLTK container opcode that groups child widgets. Syntax FLgroup "label", iwidth, iheight, ix, iy [, iborder] [, image] Initialization label -- a double-quoted string containing some user-provided text, placed near the corresponding widget. iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iborder (optional, default=0) -- border type of the container. It is expressed by means of an integer number chosen from the following: 0 - no border 1 - down box border 2 - up box border 3 - engraved border 4 - embossed border 5 - black line border 6 - thin down border 7 - thin up border If the integer number doesn't match any of the previous values, no border is provided as the default. image (optional) -- a handle referring to an eventual image opened with the bmopen opcode. If it is set, it allows a skin for that widget. Note about the bmopen opcode Although the documentation mentions the bmopen opcode, it has not been implemented in Csound 4.22. Performance Containers are useful to format the graphic appearance of the widgets. The most important container is FLpanel, that actually creates a window. It can be filled with other containers and/or valuators or other kinds of widgets. There are no k-rate arguments in containers. See Also FLgroupEnd, FLpack, FLpackEnd, FLpanel, FLpanelEnd, FLscroll, FLscrollEnd, FLtabs, FLtabsEnd Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/outc.xml0000644000000000000000000000441412262561504016100 0ustar rootroot outc Signal I/O:Signal Output outc outc Writes audio data with an arbitrary number of channels to an external device or stream. Description Writes audio data with an arbitrary number of channels to an external device or stream. Syntax outc asig1 [, asig2] [...] Performance outc outputs as many channels as provided. Any channels greater than nchnls are ignored. Zeros are added as necessary Examples Here is an example of the outc opcode. It uses the file outc.csd. Example of the outc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also out32, outch, outx, outz Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/inletkid.xml0000644000000000000000000000563212262561504016734 0ustar rootroot inletkid Signal Flow Graph Opcodes inletkid inletkid Receives a krate signal into an instrument from a named port. Description Receives a krate signal into an instrument from a named port. Syntax ksignal inletkid Sname, SinstanceID Initialization Sname -- String name of the inlet port. The name of the inlet is implicitly qualified by the instrument name or number, so it is valid to use the same inlet name in more than one instrument (but not to use the same inlet name twice in one instrument). SinstanceID -- String name of the outlet port's instance ID. This enables the inlet to discriminate between different instances of the outlet, e,g. one instance of the outlet might be created by a note specifying one instance ID, and another instance might be created by a note specifying another ID. This might be used, e.g., to situate difference instances of an instrument at different points in an Ambisonic space in a spatializing effects processor. Performance ksignal -- krate input signal During performance, the krate inlet signal is received from each instance of an instrument containing an outlet port to which this inlet has been connected using the connect opcode. The signals of all the outlets that are connected to an inlet, but only those share the specified instance ID, are summed in the inlet. See Also outleta outletk outletkid outletf inleta inletf connect alwayson ftgenonce Credits By: &namemichael; 2009 manual_src~dfsg/opcodes/envlpx.xml0000644000000000000000000001217112262561504016441 0ustar rootroot envlpx Signal Generators:Envelope Generators envlpx envlpx Applies an envelope consisting of 3 segments. Description envlpx -- apply an envelope consisting of 3 segments: stored function rise shape modified exponential pseudo steady state exponential decay Syntax ares envlpx xamp, irise, idur, idec, ifn, iatss, iatdec [, ixmod] kres envlpx kamp, irise, idur, idec, ifn, iatss, iatdec [, ixmod] Initialization irise -- rise time in seconds. A zero or negative value signifies no rise modification. idur -- overall duration in seconds. A zero or negative value will cause initialization to be skipped. idec -- decay time in seconds. Zero means no decay. An idec > idur will cause a truncated decay. ifn -- function table number of stored rise shape with extended guard point. iatss -- attenuation factor, by which the last value of the envlpx rise is modified during the note's pseudo steady state. A factor greater than 1 causes an exponential growth and a factor less than 1 creates an exponential decay. A factor of 1 will maintain a true steady state at the last rise value. Note that this attenuation is not by fixed rate (as in a piano), but is sensitive to a note's duration. However, if iatss is negative (or if steady state < 4 k-periods) a fixed attenuation rate of abs(iatss) per second will be used. 0 is illegal. iatdec -- attenuation factor by which the closing steady state value is reduced exponentially over the decay period. This value must be positive and is normally of the order of .01. A large or excessively small value is apt to produce a cutoff which is audible. A zero or negative value is illegal. ixmod (optional, between +- .9 or so) -- exponential curve modifier, influencing the steepness of the exponential trajectory during the steady state. Values less than zero will cause an accelerated growth or decay towards the target (e.g. subito piano). Values greater than zero will cause a retarded growth or decay. The default value is zero (unmodified exponential). Performance kamp, xamp -- input amplitude signal. Rise modifications are applied for the first irise seconds, and decay from time idur - idec. If these periods are separated in time there will be a steady state during which amp will be modified by the first exponential pattern. If rise and decay periods overlap then both modifications will be in effect for that time. If the overall duration idur is exceeded in performance, the final decay will continue on in the same direction, tending asymptotically to zero. Examples Here is an example of the envlpx opcode. It uses the file envlpx.csd. Example of the envlpx opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also envlpxr, linen, linenr Credits Thanks goes to &nameluis; for pointing out a mistake with iatss. manual_src~dfsg/opcodes/metro.xml0000644000000000000000000000562712262561504016263 0ustar rootroot metro Instrument Control:Sensing and Control metro metro Trigger Metronome Description Generate a metronomic signal to be used in any circumstance an isochronous trigger is needed. Syntax ktrig metro kfreq [, initphase] Initialization initphase - initial phase value (in the 0 to 1 range) Performance ktrig - output trigger signal kfreq - frequency of trigger bangs in cps metro is a simple opcode that outputs a sequence of isochronous bangs (that is 1 values) each 1/kfreq seconds. Trigger signals can be used in any circumstance, mainly to temporize realtime algorithmic compositional structures. metro will produce a trigger signal of 1 when its phase is exactly 0 or 1. If you want to skip the initial trigger, use a very small value like 0.00000001. Examples Here is an example of the metro opcode. It uses the file metro.csd Example of the metro opcode. Here is another example of the metro opcode. It uses the file metro-2.csd Another xample of the metro opcode. See Also More information on this opcode in the Floss Manuals: http://en.flossmanuals.net/csound/ch018_c-control-structures/ Credits Written by &namegabriel;. First Example written by &nameandres;. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/divz.xml0000644000000000000000000000501312262561504016076 0ustar rootroot divz Mathematical Operations:Opcode Equivalents of Functions divz divz Safely divides two numbers. Syntax ares divz xa, xb, ksubst ires divz ia, ib, isubst kres divz ka, kb, ksubst ...divz(ka, kb, ksubst)... (no rate restriction) Description Safely divides two numbers. Initialization Whenever b is not zero, set the result to the value a / b; when b is zero, set it to the value of subst instead. Examples Here is an example of the divz opcode. It uses the file divz.csd. Example of the divz opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: 200.000000 / 0.000000 = -1.000000 200.000000 / 19.999887 = 10.000056 200.000000 / 40.000027 = 4.999997 See Also =, init, tival Credits Author: &namejohn; after an idea by &namebarry; Example written by &namekevin;. Functional form available in csound Version 6.00 manual_src~dfsg/opcodes/notnum.xml0000644000000000000000000000514512262561504016450 0ustar rootroot notnum Real-time MIDI:Input notnum notnum Get a note number from a MIDI event. Description Get a note number from a MIDI event. Syntax ival notnum Performance Get the MIDI byte value (0 - 127) denoting the note number of the current event. Examples Here is an example of the notnum opcode. It uses the file notnum.csd. Example of the notnum opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also aftouch, ampmidi, cpsmidi, cpsmidib, midictrl, octmidi, octmidib, pchbend, pchmidi, pchmidib, veloc Credits Author: &namebarry; - &namemike; MIT - Mills May 1997 Example written by David Akbari. manual_src~dfsg/opcodes/fluidAllOut.xml0000644000000000000000000000552412262561504017355 0ustar rootroot fluidAllOut Signal Generators:Sample Playback fluidAllOut fluidAllOut Collects all audio from all Fluidsynth engines in a performance Syntax aleft, aright fluidAllOut Description Collects all audio from all Fluidsynth engines in a performance Performance aleft -- Left channel audio output. aright -- Right channel audio output. Invoke fluidAllOut in an instrument definition numbered higher than any fluidcontrol instrument definitions. All SoundFonts send their audio output to this one opcode. Send a note with an indefinite duration to this instrument to turn the SoundFonts on for as long as required. In this implementation, SoundFont effects such as chorus or reverb are used if and only if they are defaults for the preset. There is no means of turning such effects on or off, or of changing their parameters, from Csound. Examples Here is an example of the fluidAllOut opcodes. It uses the file fluidAllOut.csd. See Also fluidEngine, fluidNote, fluidLoad More information on soundfonts in the Floss Manuals: http://en.flossmanuals.net/csound/ch048_d-reading-midi-files Other information on soundfonts on Wikipedia: http://en.wikipedia.org/wiki/Soundfont Credits Opcode by Michael Gogins (gogins at pipeline dot com). Thanks to Peter Hanappe for Fluidsynth, and to Steven Yi for seeing that it is necessary to break up the Fluidsynth into several different Csound opcodes. manual_src~dfsg/opcodes/partials.xml0000644000000000000000000000730512262561504016747 0ustar rootroot partials Spectral Processing:Streaming partials partials Partial track spectral analysis. Description The partials opcode takes two input PV streaming signals containg AMP_FREQ and AMP_PHASE signals (as generated for instance by pvsifd or in the first case, by pvsanal) and performs partial track analysis, as described in Lazzarini et al, "Time-stretching using the Instantaneous Frequency Distribution and Partial Tracking", Proc.of ICMC05, Barcelona. It generates a TRACKS PV streaming signal, containing amplitude, frequency, phase and track ID for each output track. This type of signal will contain a variable number of output tracks, up to the total number of analysis bins contained in the inputs (fftsize/2 + 1 bins). The second input (AMP_PHASE) is optional, as it can take the same signal as the first input. In this case, however, all phase information will be NULL and resynthesis using phase information cannot be performed. Syntax ftrks partials ffr, fphs, kthresh, kminpts, kmaxgap, imaxtracks Performance ftrks -- output pv stream in TRACKS format ffr -- input pv stream in AMP_FREQ format fphs -- input pv stream in AMP_PHASE format kthresh -- analysis threshold. Tracks below ktresh*max_magnitude will be discarded (1 > ktresh >= 0). kminpoints -- minimum number of time points for a detected peak to make a track (1 is the minimum). Since this opcode works with streaming signals, larger numbers will increase the delay between input and output, as we have to wait for the required minimum number of points. kmaxgap -- maximum gap between time-points for track continuation (> 0). Tracks that have no continuation after kmaxgap will be discarded. imaxtracks -- maximum number of analysis tracks (number of bins >= imaxtracks) Example Here is an example of the partials opcodes. It uses the file partials.csd. Example of the partials opcodes. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of an ifd-analysis signal and cubic-phase additive resynthesis with pitch shifting. Credits Author: &namevictor; June 2005 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/0dbfs.xml0000644000000000000000000001131512262561504016122 0ustar rootroot 0dbfs Orchestra Syntax:Header 0dbfs 0dbfs Sets the value of 0 decibels using full scale amplitude. Description Sets the value of 0 decibels using full scale amplitude. Syntax 0dbfs = iarg 0dbfs Initialization iarg -- the value of 0 decibels using full scale amplitude. Performance The default is 32767, so all existing orcs should work. Amplitude values in Csound are always relative to a "0dbfs" value representing the peak available amplitude before clipping. In the original Csound, this value was always 32767, corresponding to the bipolar range of a 16bit soundfile or 16bit AD/DA codec. This remains the default peak amplitude for Csound, for backward compatibility. The 0dbfs value enables Csound to produce appropriately scaled values to whatever output format is being used, whether 16bit integer, 24bit integer, 32bit floats, or even 32bit integers. OdBFS can be defined in the header, to set the amplitude reference Csound will use, but it can also be used as a varible inside instruments like this: ipeak = 0dbfs asig oscil 0dbfs, freq, 1 out asig * 0.3 * 0dbfs The purpose of the 0dbfs opcode is for people to start to code 0dbfs-relatively (and use the ampdbfs() opcodes a lot more!), rather than use explicit sample values. Using 0dbfs=1 is in accordance to industry practice, as ranges from -1 to 1 are used in most commercial plugin formats and in most other synthesis systems like Pure Data. Floats written to a file, when 0dbfs = 1, will in effect go through no range translation at all. So the numbers in the file are exactly what the orc says they are. For more details on amplitude values in Csound, see the section Amplitude values in Csound Example Here is an example of the 0dbfs opcode. It uses the file 0dbfs.csd. Example of the 0dbfs opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also ampdbfs() Credits Author: Richard Dobson May 2002 New in version 4.10 manual_src~dfsg/opcodes/cauchyi.xml0000644000000000000000000000575312262561504016562 0ustar rootroot x-class noise generators cauchyi Signal Generators:Random (Noise) Generators cauchyi cauchyi Cauchy distribution random number generator with interpolation. Description Cauchy distribution random number generator with controlled interpolation between values. This is an x-class noise generator. Syntax ares cauchyi klambda, xamp, xcps ires cauchyi klambda, xamp, xcps kres cauchyi klambda, xamp, xcps Performance kalpha -- controls the spread from zero (big kalpha = big spread). Outputs both positive and negative numbers. For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. xamp -- range over which random numbers are distributed. xcps -- the frequency which new random numbers are generated. Examples Here is an example of the cauchyi opcode. It uses the file cauchyi.csd. Example of the cauchyi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cauchy Credits Author: &namejohn; Bath May 2011 New in version 5.14 manual_src~dfsg/opcodes/powershape.xml0000644000000000000000000001110012262561504017271 0ustar rootroot powershape Signal Modifiers:Waveshaping powershape powershape Waveshapes a signal by raising it to a variable exponent. Description The powershape opcode raises an input signal to a power with pre- and post-scaling of the signal so that the output will be in a predictable range. It also processes negative inputs in a symmetrical way to positive inputs, calculating a dynamic transfer function that is useful for waveshaping. Syntax aout powershape ain, kShapeAmount [, ifullscale] Initialization ifullscale -- optional parameter specifying the range of input values from -ifullscale to ifullscale. Defaults to 1.0 -- you should set this parameter to the maximum expected input value. Performance ain -- the input signal to be shaped. aout -- the output signal. kShapeAmount -- the amount of the shaping effect applied to the input; equal to the power that the input signal is raised. The powershape opcode is very similar to the pow unit generators for calculating the mathematical "power of" operation. However, it introduces a couple of twists that can make it much more useful for waveshaping audio-rate signals. The kShapeAmount parameter is the exponent to which the input signal is raised. To avoid discontinuities, the powershape opcode treats all input values as positive (by taking their absolute value) but preserves their original sign in the output signal. This allows for smooth shaping of any input signal while varying the exponent over any range. (powershape also (hopefully) deals intelligently with discontinuities that could arise when the exponent and input are both zero. Note though that negative exponents will usually cause the signal to exceed the maximum amplitude specified by the ifullscale parameter and should normally be avoided). The other adaptation involves the ifullscale parameter. The input signal is divided by ifullscale before being raised to kShapeAmount and then multiplied by ifullscale before being output. This normalizes the input signal to the interval [-1,1], guaranteeing that the output (before final scaling) will also be within this range for positive shaping amounts and providing a smoothly evolving transfer function while varying the amount of shaping. Values of kShapeAmount between (0,1) will make the signal more "convex" while values greater than 1 will make it more "concave". A value of exactly 1.0 will produce no change in the input signal. See Also pow, powoftwo Examples Here is an example of the powershape opcode. It uses the file powershape.csd. Example of the powershape opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &nameanthony; January 2008 New in Csound version 5.08 manual_src~dfsg/opcodes/gain.xml0000644000000000000000000000507112262561504016044 0ustar rootroot gain Signal Modifiers:Amplitude Modifiers gain gain Adjusts the amplitude audio signal according to a root-mean-square value. Description Adjusts the amplitude audio signal according to a root-mean-square value. Syntax ares gain asig, krms [, ihp] [, iskip] Initialization ihp (optional, default=10) -- half-power point (in Hz) of a special internal low-pass filter. The default value is 10. iskip (optional, default=0) -- initial disposition of internal data space (see reson). The default value is 0. Performance asig -- input audio signal gain provides an amplitude modification of asig so that the output ares has rms power equal to krms. rms and gain used together (and given matching ihp values) will provide the same effect as balance. Examples Here is an example of the gain opcode. It uses the file gain.csd. Example of the gain opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also balance, rms manual_src~dfsg/opcodes/chnclear.xml0000644000000000000000000000314312262561504016703 0ustar rootroot chnclear Signal I/O:Software Bus chnclear chnclear Clears an audio output channel of the named software bus. Description Clears an audio channel of the named software bus to zero. Implies declaring the channel with imode=2 (see also chn_a). Syntax chnclear Sname Initialization Sname -- a string that indicates which named channel of the software bus to clear. Examples Here is an example of the chnclear opcode. It uses the file chnclear.csd. Example of the chnclear opcode. Credits Author: &nameistvan; 2006 manual_src~dfsg/opcodes/rewindscore.xml0000644000000000000000000000340312262561504017447 0ustar rootroot rewindscore Instrument Control:Sensing and Control rewindscore rewindscore Rewinds the playback position of the current score performance. Description Rewinds the playback position of the current score performance.. Syntax rewindscore Examples Here is an example of the rewindscore opcode. It uses the file rewindscore.csd. An example of the rewindscore opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also setscorepos Credits Author: &namevictor; 2008 New in Csound version 5.09 manual_src~dfsg/opcodes/aweibull.xml0000644000000000000000000000077612262561504016741 0ustar rootroot aweibull aweibull aweibull Deprecated. Description Deprecated as of version 3.49. Use the weibull opcode instead. manual_src~dfsg/opcodes/vbapzmove.xml0000644000000000000000000001043412262561504017136 0ustar rootroot vbapzmove Signal Modifiers:Panning and Spatialization vbapzmove vbapzmove Writes a multi-channel audio signal to a ZAK array with moving virtual sources. Description Writes a multi-channel audio signal to a ZAK array with moving virtual sources. Syntax vbapzmove inumchnls, istartndx, asig, idur, ispread, ifldnum, ifld1, \ ifld2, [...] Initialization inumchnls -- number of channels to write to the ZA array. Must be in the range 2 - 256. istartndx -- first index or position in the ZA array to use idur -- the duration over which the movement takes place. ispread -- spreading of the virtual source (range 0 - 100). If value is zero, conventional amplitude panning is used. When ispread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. ifldnum -- number of fields (absolute value must be 2 or larger). If ifldnum is positive, the virtual source movement is a polyline specified by given directions. Each transition is performed in an equal time interval. If ifldnum is negative, specified angular velocities are applied to the virtual source during specified relative time intervals (see below). ifld1, ifld2, ... -- azimuth angles or angular velocities, and relative durations of movement phases (see below). Performance asig -- audio signal to be panned The opcode vbapzmove is the multiple channel analog of the opcodes like vbap4move, working on inumchnls and using a ZAK array for output. Please note that all vbap panning opcodes require the vbap system to be initialized using vbaplsinit. Examples See the entry for vbap8move for an example of usage of the vbapXmove opcodes. Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. See Also vbap16, vbap16move, vbap4, vbap4move, vbap8, vbap8move, vbaplsinit, vbapz, Credits &namejohn; University of Bath/Codemist Ltd. Bath, UK May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/moscil.xml0000644000000000000000000000703512262561504016416 0ustar rootroot moscil Real-time MIDI:Note Output moscil moscil Sends a stream of the MIDI notes. Description Sends a stream of the MIDI notes. Syntax moscil kchn, knum, kvel, kdur, kpause Performance kchn -- MIDI channel number (1-16) knum -- note number (0-127) kvel -- velocity (0-127) kdur -- note duration in seconds kpause -- pause duration after each noteoff and before new note in seconds moscil and midion are the most powerful MIDI OUT opcodes. moscil (MIDI oscil) plays a stream of notes of kdur duration. Channel, pitch, velocity, duration and pause can be controlled at k-rate, allowing very complex algorithmically generated melodic lines. When current instrument is deactivated, the note played by current instance of moscil is forcedly truncated. Any number of moscil opcodes can appear in the same Csound instrument, allowing a counterpoint-style polyphony within a single instrument. Examples Here is an example of the moscil opcode. It uses the file moscil.csd. Example of the moscil opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This example generates a stream of notes for every note received on the MIDI input. It generates MIDI notes on csound's MIDI output, so be sure to connect something. See Also midion, midion2, noteon, noteoff, noteondur, noteondur2 Credits Author: &namegabriel; Italy May 1997 New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/flashtxt.xml0000644000000000000000000000476512262561504016774 0ustar rootroot flashtxt Signal I/O:Printing and Display flashtxt flashtxt Allows text to be displayed from instruments like sliders Description Allows text to be displayed from instruments like sliders etc. (only on Unix and Windows at present) Syntax flashtxt iwhich, String Initialization iwhich -- the number of the window. String -- the string to be displayed. Performance Note that this opcode is not available on Windows due to the implimentation of pipes on that system A window is created, identified by the iwhich argument, with the text string displayed. If the text is replaced by a number then the window id deleted. Note that the text windows are globally numbered so different instruments can change the text, and the window survives the instance of the instrument. Examples Here is an example of the flashtxt opcode. It uses the file flashtxt.csd. Example of the flashtxt opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.11 manual_src~dfsg/opcodes/outq.xml0000644000000000000000000000616612262561504016124 0ustar rootroot outq Signal I/O:Signal Output outq outq Writes 4-channel audio data to an external device or stream. Description Writes 4-channel audio data to an external device or stream. Syntax outq asig1, asig2, asig3, asig4 Performance Sends 4-channel audio samples to an accumulating output buffer (created at the beginning of performance) which serves to collect the output of all active instruments before the sound is written to disk. There can be any number of these output units in an instrument. The type (mono, stereo, quad, hex, or oct) should agree with nchnls. But as of version 3.50, Csound will attempt to change an incorrect opcode to agree with the nchnls statement. Opcodes can be chosen to direct sound to any particular channel: outs1 sends to stereo channel 1, outq3 to quad channel 3, etc. Examples Here is an example of the outq opcode. It uses the file outq.csd. Example of the outq opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also out, outh, outo, outq1, outq2, outq3, outq4, outs, outs1, outs2, soundout Credits Author: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 manual_src~dfsg/opcodes/outkat.xml0000644000000000000000000000731712262561504016442 0ustar rootroot outkat Real-time MIDI:Output outkat outkat Sends MIDI aftertouch messages at k-rate. Description Sends MIDI aftertouch messages at k-rate. Syntax outkat kchn, kvalue, kmin, kmax Performance kchn -- MIDI channel number (1-16) kvalue -- floating point value kmin -- minimum floating point value (converted in MIDI integer value 0) kmax -- maximum floating point value (converted in MIDI integer value 127) outkat (k-rate aftertouch output) sends aftertouch messages. It works only with MIDI instruments which recognize them. It can drive a different value of a parameter for each note currently active. It can scale the k-value floating-point argument according to the kmin and kmax values. For example: set kmin = 1.0 and kmax = 2.0. When the kvalue argument receives a 2.0 value, the opcode will send a 127 value to the MIDI OUT device. When the kvalue argument receives a 1.0 value, it will send a 0 value. k-rate opcodes send a message each time the MIDI converted value of argument kvalue changes. Examples Here is an example of the outkat opcode. It uses the file outkat.csd. Example of the outkat opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also outiat, outic14, outic, outipat, outipb, outipc, outkc14, outkc, outkpat, outkpb, outkpc Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/midichannelaftertouch.xml0000644000000000000000000001152312262561504021465 0ustar rootroot midichannelaftertouch Real-time MIDI:MIDI/Score Interoperability midichannelaftertouch midichannelaftertouch Gets a MIDI channel's aftertouch value. Description midichannelaftertouch is designed to simplify writing instruments that can be used interchangeably for either score or MIDI input, and to make it easier to adapt instruments originally written for score input to work with MIDI input. In general, it should be possible to write instrument definitions that work identically with both scores and MIDI, including both MIDI files and real-time MIDI input, without using any conditional statements, and that take full advantage of MIDI voice messages. Note that correlating Csound instruments with MIDI channel numbers is done using the massign opcode for real-time performance,. For file-driven performance, instrument numbers default to MIDI channel number + 1, but the defaults are overridden by any MIDI program change messages in the file. Syntax midichannelaftertouch xchannelaftertouch [, ilow] [, ihigh] Initialization ilow (optional) -- optional low value after rescaling, defaults to 0. ihigh (optional) -- optional high value after rescaling, defaults to 127. Performance xchannelaftertouch -- returns the MIDI channel aftertouch during MIDI activation, remains unchanged otherwise. If the instrument was activated by MIDI input, the opcode overwrites the value of xchannelaftertouch with the corresponding value from MIDI input. If the instrument was NOT activated by MIDI input, the value of xchannelaftertouch remains unchanged. This enables score p-fields to receive MIDI input data during MIDI activation, and score values otherwise. Adapting a score-activated Csound instrument. See the MIDI interop opcodes section for details on adapting score driven instruments for MIDI or vice-versa. Examples Here is an example of the midichannelaftertouch opcode. It uses the file midichannelaftertouch.csd. Example of the midichannelaftertouch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i1 127.00000 i1 20.00000 i1 44.00000 See Also midicontrolchange, mididefault, midinoteoff, midinoteoncps, midinoteonkey, midinoteonoct, midinoteonpch, midipitchbend, midipolyaftertouch, midiprogramchange Credits Author: Michael Gogins New in version 4.20 manual_src~dfsg/opcodes/vpowv_i.xml0000644000000000000000000001024512262561504016616 0ustar rootroot vpowv_i Vectorial:Vectorial operations vpowv_i vpowv_i Performs power-of operations between two vectorial control signals at init time. Description Performs power-of operations between two vectorial control signals at init time. Syntax vpowv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Initialization ifn1 - number of the table hosting the first vector to be processed ifn2 - number of the table hosting the second vector to be processed ielements - number of elements of the two vectors idstoffset - index offset for the destination (ifn1) table isrcoffset - index offset for the source (ifn2) table Performance vpowv_i raises each element of ifn1 to the corresponding element of ifn2. Each vectorial signal is hosted by a table (ifn1 and ifn2). The number of elements contained in both vectors must be the same. The result is a new vectorial control signal that overrides old values of ifn1. If you want to keep the old ifn1 vector, use vcopy_i opcode to copy it in another table. You can use idstoffset and isrcoffset to specify vectors in any location of the tables. Negative values for idstoffset and isrcoffset are acceptable. If idstoffset is negative, the out of range section of the vector will be discarded. If isrcoffset is negative, the out of range elements will be assumed to be 1 (i.e. the destination elements will not be changed). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 1 (i.e. the destination vector will not be changed for these elements). Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at init time. There's an k-rate version of this opcode called vpowv. All these operators (vaddv_i,vsubv_i,vmultv_i,vdivv_i,vpowv_i,vexpv_i, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/oscili.xml0000644000000000000000000000657512262561504016422 0ustar rootroot oscili Signal Generators:Basic Oscillators oscili oscili A simple oscillator with linear interpolation. Description oscili reads table ifn sequentially and repeatedly at a frequency xcps. The amplitude is scaled by xamp. Linear interpolation is applied for table look up from internal phase values. Syntax ares oscili xamp, xcps, ifn [, iphs] kres oscili kamp, kcps, ifn [, iphs] Initialization ifn -- function table number. Requires a wrap-around guard point. iphs (optional) -- initial phase of sampling, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is 0. Performance kamp, xamp -- amplitude kcps, xcps -- frequency in cycles per second. oscili differs from oscil in that the standard procedure of using a truncated phase as a sampling index is here replaced by a process that interpolates between two successive lookups. Interpolating generators will produce a noticeably cleaner output signal, but they may take as much as twice as long to run. Adequate accuracy can also be gained without the time cost of interpolation by using large stored function tables of 2K, 4K or 8K points if the space is available. Table ifn is incrementally sampled modulo the table length and the value obtained is multiplied by amp. If you need to change the oscillator table with a k-rate signal, you can use oscilikt. Examples Here is an example of the oscili opcode. It uses the file oscili.csd. Example of the oscili opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also oscil, oscil3 Credits manual_src~dfsg/opcodes/STKHevyMetl.xml0000644000000000000000000000734112262561504017247 0ustar rootroot STKHevyMetl Signal Generators:STK Opcodes STKHevyMetl STKHevyMetl STKHevyMetl produces metal sounds. Description STKHevyMetl produces metal sounds, using FM synthesis. It uses 3 cascade operators with feedback modulation, also referred to as algorithm 3 of the TX81Z. Syntax asignal STKHevyMetl ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kmod -- controller2, total modulator index. Value range of kv1 is 0-127. kcross -- controller 4, crossfade of modulator. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kadsr -- controller 128, ADSR 2 and 4 target. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the HevyMetl class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKHevyMetl opcode. It uses the file STKHevyMetl.csd. Example of the STKHevyMetl opcode. See Also STKWurley. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/kread3.xml0000644000000000000000000000076212262561504016301 0ustar rootroot kread3 kread3 kread3 Deprecated. Description Deprecated as of version 3.52. Use the readk3 opcode instead. manual_src~dfsg/opcodes/kdump4.xml0000644000000000000000000000076212262561504016334 0ustar rootroot kdump4 kdump4 kdump4 Deprecated. Description Deprecated as of version 3.49. Use the dumpk4 opcode instead. manual_src~dfsg/opcodes/nsamp.xml0000644000000000000000000000633612262561504016251 0ustar rootroot nsamp Table Control:Table Queries nsamp nsamp Returns the number of samples loaded into a stored function table number. Description Returns the number of samples loaded into a stored function table number. Syntax nsamp(x) (init-rate args only) Performance Returns the number of samples loaded into stored function table number x by GEN01. This is useful when a sample is shorter than the power-of-two function table that holds it. New in Csound version 3.49. As of Csound version 5.02, nsamp works with deferred-length function tables (see GEN01). nsamp differs from ftlen in that nsamp gives the number of sample frames loaded, while ftlen gives the total number of samples. For example, with a stereo sound file of 10000 samples, ftlen() would return 19999 (i.e. a total of 20000 mono samples, not including a guard point), but nsamp() returns 10000. Examples Here is an example of the nsamp opcode. It uses the file nsamp.csd, kickroll.wav. and fox.wav. Example of the nsamp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Since the stereo audio file kickroll.wav has 37792 samples and the mono file fox.wav has 121569 samples, its output should include lines like these: actual numbers of samples = 37792 actual numbers of samples = 121569 See Also ftchnls, ftlen, ftlptim, ftsr Credits Author: &namegabriel; Italy October 1998 manual_src~dfsg/opcodes/delay1.xml0000644000000000000000000000465012262561504016307 0ustar rootroot delay1 Signal Modifiers:Delay delay1 delay1 Delays an input signal by one sample. Description Delays an input signal by one sample. Syntax ares delay1 asig [, iskip] Initialization iskip (optional, default=0) -- initial disposition of delay-loop data space (see reson). The default value is 0. Performance delay1 is a special form of delay that serves to delay the audio signal asig by just one sample. It is thus functionally equivalent to the delay opcode but is more efficient in both time and space. This unit is particularly useful in the fabrication of generalized non-recursive filters. Examples Here is an example of the delay and delay1 opcodes. It uses the file delay1.csd. Example of the delay1 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also delay, delayr, delayw Credits Author: Barry Vercoe Example written by &namejohn;. manual_src~dfsg/opcodes/ATSpartialtap.xml0000644000000000000000000000774112262561504017645 0ustar rootroot ATSpartialtap Spectral Processing:ATS ATSpartialtap ATSpartialtap returns a frequency, amplitude pair from an ATSbufread opcode. Description ATSpartialtap takes a partial number and returns a frequency, amplitude pair. The frequency and amplitude data comes from an ATSbufread opcode. Syntax kfrq, kamp ATSpartialtap ipartialnum Initialization ipartialnum - indicates the partial that the ATSpartialtap opcode should read from an ATSbufread. Performance kfrq - returns the frequency value for the requested partial. kamp - returns the amplitude value for the requested partial. ATSpartialtap takes a partial number and returns a frequency, amplitude pair. The frequency and amplitude data comes from an ATSbufread opcode. This is more restricted version of ATSread, since each ATSread opcode has its own independent time pointer, and ATSpartialtap is restricted to the data given by an ATSbufread. Its simplicity is its attractive feature. Examples Here is an example of the ATSpartialtap opcode. It uses the file ATSpartialtap.csd. Example of the ATSpartialtap opcode. This example here uses an ATSpartialtap, and an ATSbufread to read partials 5, 20 and 30 from 'beats.ats'. These amplitudes and frequencies could be used to re-synthesize those partials, or something all together different. See also ATSread, ATSreadnz, ATSinfo, ATSsinnoi, ATSbufread, ATScross, ATSinterpread, ATSadd, ATSaddnz Credits Author: Alex Norman Seattle,Washington 2004 manual_src~dfsg/opcodes/buzz.xml0000644000000000000000000001054612262561504016123 0ustar rootroot buzz Signal Generators:Dynamic Spectrum Oscillators buzz buzz Output is a set of harmonically related sine partials. Description Output is a set of harmonically related sine partials. Syntax ares buzz xamp, xcps, knh, ifn [, iphs] Initialization ifn -- table number of a stored function containing a sine wave. A large table of at least 8192 points is recommended. iphs (optional, default=0) -- initial phase of the fundamental frequency, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is zero Performance xamp -- amplitude xcps -- frequency in cycles per second The buzz units generate an additive set of harmonically related cosine partials of fundamental frequency xcps, and whose amplitudes are scaled so their summation peak equals xamp. The selection and strength of partials is determined by the following control parameters: knh -- total number of harmonics requested. New in Csound version 3.57, knh defaults to one. If knh is negative, the absolute value is used. buzz and gbuzz are useful as complex sound sources in subtractive synthesis. buzz is a special case of the more general gbuzz in which klh = kmul = 1; it thus produces a set of knh equal-strength harmonic partials, beginning with the fundamental. (This is a band-limited pulse train; if the partials extend to the Nyquist, i.e. knh = int (sr / 2 / fundamental freq.), the result is a real pulse train of amplitude xamp.) Although knh may be varied during performance, its internal value is necessarily integer and may cause pops due to discontinuities in the output. buzz can be amplitude- and/or frequency-modulated by either control or audio signals. N.B. This unit has its analog in GEN11, in which the same set of cosines can be stored in a function table for sampling by an oscillator. Although computationally more efficient, the stored pulse train has a fixed spectral content, not a time-varying one as above. Examples Here is an example of the buzz opcode. It uses the file buzz.csd. Example of the buzz opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also gbuzz Credits September 2003. Thanks to &namekanata; for correcting the mentions of the kmul parameter. manual_src~dfsg/opcodes/sfinstr.xml0000644000000000000000000001353112262561504016616 0ustar rootroot sfinstr Signal Generators:Sample Playback sfinstr sfinstr Plays a SoundFont2 (SF2) sample instrument, generating a stereo sound. Description Plays a SoundFont2 (SF2) sample instrument, generating a stereo sound. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. Syntax ar1, ar2 sfinstr ivel, inotenum, xamp, xfreq, instrnum, ifilhandle \ [, iflag] [, ioffset] Initialization ivel -- velocity value inotenum -- MIDI note number value instrnum -- number of an instrument of a SF2 file. ifilhandle -- unique number generated by sfload opcode to be used as an identifier for a SF2 file. Several SF2 files can be loaded and activated at the same time. iflag (optional) -- flag regarding the behavior of xfreq and inotenum ioffset (optional) -- start playing at offset, in samples. Performance xamp -- amplitude correction factor xfreq -- frequency value or frequency multiplier, depending by iflag. When iflag = 0, xfreq is a multiplier of a the default frequency, assigned by SF2 preset to the inotenum value. When iflag = 1, xfreq is the absolute frequency of the output sound, in Hz. Default is 0. When iflag = 0, inotenum sets the frequency of the output according to the MIDI note number used, and xfreq is used as a multiplier. When iflag = 1, the frequency of the output, is set directly by xfreq. This allows the user to use any kind of micro-tuning based scales. However, this method is designed to work correctly only with presets tuned to the default equal temperament. Attempts to use this method with a preset already having non-standard tunings, or with drum-kit-based presets, could give unexpected results. Adjustment of the amplitude can be done by varying the xamp argument, which acts as a multiplier. The ioffset parameter allows the sound to start from a sample different than the first one. The user should make sure that its value is within the length of the specific sound. Otherwise, Csound will probably crash. sfinstr plays an SF2 instrument instead of a preset (an SF2 instrument is the base of a preset layer). instrnum specifies the instrument number, and the user must be sure that the specified number belongs to an existing instrument of a determinate soundfont bank. Notice that both xamp and xfreq can operate at k-rate as well as a-rate, but both arguments must work at the same rate. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. Examples Here is an example of the sfinstr opcode. It uses the file sfinstr.csd. Example of the sfinstr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also sfilist, sfinstrm, sfload, sfpassign, sfplay, sfplaym, sfplist, sfpreset Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/pan2.xml0000644000000000000000000000465712262561504015777 0ustar rootroot pan2 Signal Modifiers:Panning and Spatialization pan2 pan2 Distribute an audio signal across two channels. Description Distribute an audio signal across two channels with a choice of methods. Syntax a1, a2 pan2 asig, xp [, imode] Initialization imode (optional) -- mode of the stereo positioning algorithm. 0 signifies equal power (harmonic) panning, 1 means the square root method, 2 means simple linear and 3 means an alternative equal-power pan (based on an UDO). The default value is 0. Performance pan2 takes an input signal asig and distributes it across two outputs (essentially stereo speakers) according to the control xp which can be k- or a-rate. A zero value for xp indicates hard left, and a 1 is hard right. Example Here is an example of the pan2 opcodes. It uses the file pan2.csd. Example of the pan2 opcodes. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; University of Bath, Codemist Ltd. Bath, UK September 2007 New in version 5.07 manual_src~dfsg/opcodes/sliderkawai.xml0000644000000000000000000000561212262561504017426 0ustar rootroot sliderKawai Real-time MIDI:Slider Banks sliderKawai sliderKawai Creates a bank of 16 different MIDI control message numbers from a KAWAI MM-16 midi mixer. Description Creates a bank of 16 different MIDI control message numbers from a KAWAI MM-16 midi mixer. Syntax k1, k2, ...., k16 sliderKawai imin1, imax1, init1, ifn1, \ imin2, imax2, init2, ifn2, ..., imin16, imax16, init16, ifn16 Initialization imin1 ... imin16 -- minimum values for each controller imax1 ... imax16 -- maximum values for each controller init1 ... init16 -- initial value for each controller ifn1 ... ifn16 -- function table for conversion for each controller Performance k1 ... k16 -- output values The opcode sliderKawai is equivalent to slider16, but it has the controller and channel numbers (ichan and ictlnum) hard-coded to make for quick compatiblity with the KAWAI MM-16 midi mixer. This device doesn't allow changing the midi message associated to each slider. It can only output on control 7 for each fader on a separate midi channel. This opcode is a quick way of assigning the mixer's 16 faders to k-rate variables in csound. See Also slider16, slider16f, slider32, slider32f, slider64, slider64f, slider8, slider8f Credits Author: &namegabriel; New in Csound version 5.06 manual_src~dfsg/opcodes/ATSinfo.xml0000644000000000000000000001335112262561504016431 0ustar rootroot ATSinfo Spectral Processing:ATS ATSinfo ATSinfo reads data out of the header of an ATS file. Description atsinfo reads data out of the header of an ATS file. Syntax idata ATSinfo iatsfile, ilocation Initialization iatsfile – the ATS number (n in ats.n) or the name in quotes of the analysis file made using ATS. ilocation – indicates which location in the header file to return. The data in the header gives information about the data contained in the rest of the ATS file. The possible values for ilocation are given in the following list: 0 - Sample rate (Hz) 1 - Frame Size (samples) 2 - Window Size (samples) 3 - Number of Partials 4 - Number of Frames 5 - Maximum Amplitude 6 - Maximum Frequency (Hz) 7 - Duration (seconds) 8 - ATS file Type Performance Macros can really improve the legibility of your csound code, I've provided my Macro Definitions below: #define ATS_SAMP_RATE #0# #define ATS_FRAME_SZ #1# #define ATS_WIN_SZ #2# #define ATS_N_PARTIALS #3# #define ATS_N_FRAMES #4# #define ATS_AMP_MAX #5# #define ATS_FREQ_MAX #6# #define ATS_DUR #7# #define ATS_TYPE #8# ATSinfo can be useful for writing generic instruments that will work with many ATS files, even if they have different lengths and different numbers of partials etc. Example 2 is a simple application of this. Examples Here is an example of the ATSinfo opcode. It uses the file ATSinfo.csd. Example of the ATSinfo opcode. In the example above we use ATSinfo to retrieve the number of partials in the ATS file Other examples imax_freq ATSinfo "cl.ats", $ATS_FREQ_MAX In the example above we get the maximum frequency value from the ATS file "cl.ats" and store it in imax_freq. We use the Csound Macro (defined above) $ATS_FREQ_MAX, which is equivalent to the number 6. i_npartials ATSinfo p4, $ATS_N_PARTIALS i_dur ATSinfo p4, $ATS_DUR ktimepnt line 0, p3, i_dur aout ATSadd ktimepnt, 1, p4, 1, i_npartials In the example above we use ATSinfo to retrieve the duration and number of partials in the ATS file indicated by p4. With this info we synthesize the partials using atsadd. Since the duration and number of partials are not "hard-coded" we can use this code with any ATS file. Here is another example of the ATSinfo opcode. It uses the file ATSinfo-2.csd. Example 2 of the ATSinfo opcode. See also ATSread, ATSreadnz, ATSbufread, ATScross, ATSinterpread, ATSpartialtap, ATSadd, ATSaddnz, ATSsinnoi Credits Author: Alex Norman Seattle,Washington 2004 manual_src~dfsg/opcodes/poscil.xml0000644000000000000000000001023512262561504016415 0ustar rootroot poscil Signal Generators:Basic Oscillators poscil poscil High precision oscillator. Description High precision oscillator. Syntax ares poscil aamp, acps [, ifn, iphs] ares poscil aamp, kcps [, ifn, iphs] ares poscil kamp, acps [, ifn, iphs] ares poscil kamp, kcps [, ifn, iphs] ires poscil kamp, kcps [, ifn, iphs] kres poscil kamp, kcps [, ifn, iphs] Initialization ifn -- (optional) function table number. This defaults to -1 which indicates a sinewave. iphs (optional, default=0) -- initial phase (normalized table index 0-1) Performance ares -- output signal kamp, aamp -- the amplitude of the output signal. kcps, acps -- the frequency of the output signal in cycles per second. poscil (precise oscillator) is the same as oscili, but allows much more precise frequency control, especially when using long tables and low frequency values. It uses floating-point table indexing, instead of integer math, like oscil and oscili. It is only a bit slower than oscili. Since Csound 4.22, poscil can accept also negative frequency values and use a-rate values both for amplitude and frequency. So both AM and FM are allowed using this opcode. The opcode poscil3 is the same as poscil, but uses cubic interpolation. Note that poscil can use deffered (non-power of two) length tables. Examples Here is an example of the poscil opcode. It uses the file poscil.csd. Example of the poscil opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also poscil3 Credits Author: &namegabriel; Italy 1998 November 2002. Added a note about the changes to Csound version 4.22, thanks to &namerasmus;. New in Csound version 3.52 Function table optional in Csound version 6.00 manual_src~dfsg/opcodes/vtable1k.xml0000644000000000000000000000464312262561504016643 0ustar rootroot vtable1k Vectorial:Tables vtable1k vtable1k Read a vector (several scalars simultaneously) from a table. Description This opcode reads vectors from tables at k-rate. Syntax vtable1k kfn,kout1 [, kout2, kout3, .... , koutN ] Performance kfn - table number kout1...koutN - output vector elements vtable1k is a reduced version of vtablek, it only allows to access the first vector (it is equivalent to vtablek with kndx = zero, but a bit faster). It is useful to easily and quickly convert a set of values stored in a table into a set of k-rate variables to be used in normal opocodes, instead of using individual table opcodes for each value. vtable1k is an unusual opcode as it produces its output on the right side arguments of the opcode. Examples Here is an example of the vtable1k opcode. It uses the files vtable1k.csd. Example of the vtable1k opcode. See Also vtablek Credits Written by Gabriel Maldonado. New in Csound 5.06 manual_src~dfsg/opcodes/sensekey.xml0000644000000000000000000001155312262561504016756 0ustar rootroot sensekey Instrument Control:Sensing and Control sensekey sensekey Returns the ASCII code of a key that has been pressed. Description Returns the ASCII code of a key that has been pressed, or -1 if no key has been pressed. Syntax kres[, kkeydown] sensekey Performance kres - returns the ASCII value of a key which is pressed or released. kkeydown - returns 1 if the key was pressed, 0 if it was released or if there is no key event. kres can be used to read keyboard events from stdin and returns the ASCII value of any key that is pressed or released, or it returns -1 when there is no keyboard activity. The value of kkeydown is 1 when a key was pressed, or 0 otherwise. This behavior is emulated by default, so a key release is generated immediately after every key press. To have full functionality, FLTK can be used to capture keyboard events. FLpanel can be used to capture keyboard events and send them to the sensekey opcode, by adding an additional optional argument. See FLpanel for more information. Note This opcode can also be written as sense. Examples Here is an example of the sensekey opcode. It uses the file sensekey.csd. Example of the sensekey opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is what the output should look like when the "q" button is pressed... q i1 113.00000 Here is an example of the sensekey opcode in conjucntion with FLpanel. It uses the file FLpanel-sensekey.csd. Example of the sensekey opcode using keyboard capture from an FLpanel. The lit button in the FLpanel window shows the last key pressed. Here is a more complex example of the sensekey opcode in conjucntion with FLpanel. It uses the file FLpanel-sensekey2.csd. Example of the sensekey opcode using keyboard capture from an FLpanel. The console output will look something like: new alloc for instr 1: Key code = 65, state = 1 Character = 'e' Key code = 65, state = 0 Key code = 72, state = 1 Character = 'r' Key code = 72, state = 0 Key code = 61, state = 1 Character = 'a' Key code = 61, state = 0 See also FLpanel, FLkeyIn Credits Author: &namejohn; University of Bath, Codemist. Ltd. Bath, UK October 2000 Examples written by &namekevin;, Johnathan Murphy and Istvan Varga. New in Csound version 4.09. Renamed in Csound version 4.10. manual_src~dfsg/opcodes/vdivv.xml0000644000000000000000000001402612262561504016264 0ustar rootroot vdivv Vectorial:Vectorial operations vdivv vdivv Performs division between two vectorial control signals Description Performs division between two vectorial control signals Syntax vdivv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] Initialization ifn1 - number of the table hosting the first vector to be processed ifn2 - number of the table hosting the second vector to be processed Performance kelements - number of elements of the two vectors kdstoffset - index offset for the destination (ifn1) table (Default=0) ksrcoffset - index offset for the source (ifn2) table (Default=0) kverbose - Selects whether or not warnings are printed (Default=0) vdivv divides two vectorial control signals, that is, each element of ifn1 is divided by the corresponding element of ifn2. Each vectorial signal is hosted by a table (ifn1 and ifn2). The number of elements contained in both vectors must be the same. The result is a new vectorial control signal that overrides old values of ifn1. If you want to keep the old ifn1 vector, use vcopy_i opcode to copy it in another table. You can use kdstoffset and ksrcoffset to specify vectors in any location of the tables. Negative values for kdstoffset and ksrcoffset are acceptable. If kdstoffset is negative, the out of range section of the vector will be discarded. If ksrcoffset is negative, the out of range elements will be assumed to be 0 (i.e. the destination elements will be set to 0). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 0 (i.e. the destination elements will be set to 0). If the optional kverbose argument is different to 0, the opcode will print warning messages every k-pass if table lengths are exceeded. Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at k-rate (this means that every k-pass the vectors are divided). There's an i-rate version of this opcode called vdivv_i. Please note that the elements argument has changed in version 5.03 from i-rate to k-rate. This will change the opcode's behavior in the unusual cases where the i-rate variable ielements is changed inside the instrument, for example in: instr 1 ielements = 10 vadd 1, 1, ielements ielements = 20 vadd 2, 1, ielements turnoff endin All these operators (vaddv,vsubv, vmultv, vdivv, vpowv, vexpv, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Examples Here is an example of the vdivv opcode. It uses the file vdivv.csd. Example of the vdivv opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/filelen.xml0000644000000000000000000000577612262561504016560 0ustar rootroot filelen Signal I/O:Soundfile Queries filelen filelen Returns the length of a sound file. Description Returns the length of a sound file. Syntax ir filelen ifilcod, [iallowraw] Initialization ifilcod -- sound file to be queried iallowraw -- Allow raw sound files (default=1) Performance filelen returns the length of the sound file ifilcod in seconds. filelen can return the length of convolve and PVOC files if the "allow raw sound file" flag is not zero (it is non-zero by default). Examples Here is an example of the filelen opcode. It uses the file filelen.csd, fox.wav, and kickroll.wav. Example of the filelen opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The mono audio file fox.wav is 2.8 seconds long, and the stereo file kickroll.wav is 0.9 seconds. So filelen's output should include a line for the mono and the stereo file like this: instr 1: ilen = 2.757 instr 1: ilen = 0.857 See Also filebit, filenchnls, filepeak, filesr Credits Author: &namematt; July 1999 New in Csound version 3.57 manual_src~dfsg/opcodes/kgoto.xml0000644000000000000000000000460312262561504016251 0ustar rootroot kgoto Instrument Control:Program Flow Control kgoto kgoto Transfer control during the performance-time passes. Description During the p-time passes only, unconditionally transfer control to the statement labeled by label. Syntax kgoto label where label is in the same instrument block and is not an expression. Examples Here is an example of the kgoto opcode. It uses the file kgoto.csd. Example of the kgoto opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: kval = 0.000000, kfreq = 440.000000 kval = 0.999732, kfreq = 440.000000 kval = 1.999639, kfreq = 880.000000 See Also cggoto, cigoto, ckgoto, goto, if, igoto, tigoto, timout Credits Example written by &namekevin;. manual_src~dfsg/opcodes/ctrl14.xml0000644000000000000000000000741112262561504016237 0ustar rootroot ctrl14 Real-time MIDI:Input ctrl14 ctrl14 Allows a floating-point 14-bit MIDI signal scaled with a minimum and a maximum range. Description Allows a floating-point 14-bit MIDI signal scaled with a minimum and a maximum range. Syntax idest ctrl14 ichan, ictlno1, ictlno2, imin, imax [, ifn] kdest ctrl14 ichan, ictlno1, ictlno2, kmin, kmax [, ifn] Initialization idest -- output signal ichan -- MIDI channel number (1-16) ictln1o -- most-significant byte controller number (0-127) ictlno2 -- least-significant byte controller number (0-127) imin -- user-defined minimum floating-point value of output imax -- user-defined maximum floating-point value of output ifn (optional) -- table to be read when indexing is required. Table must be normalized. Output is scaled according to imax and imin val. Performance kdest -- output signal kmin -- user-defined minimum floating-point value of output kmax -- user-defined maximum floating-point value of output ctrl14 (i- and k-rate 14 bit MIDI control) allows a floating-point 14-bit MIDI signal scaled with a minimum and a maximum range. The minimum and maximum values can be varied at k-rate. It can use optional interpolated table indexing. It requires two MIDI controllers as input. ctrl14 differs from midic14 because it can be included in score-oriented instruments without Csound crashes. It needs the additional parameter ichan containing the MIDI channel of the controller. MIDI channel is the same for all the controllers used in a single ctrl14 opcode. See Also ctrl7, ctrl21, initc7, initc14, initc21, midic7, midic14, midic21 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/vdelayxs.xml0000644000000000000000000000761612262561504016774 0ustar rootroot vdelayxs Signal Modifiers:Delay vdelayxs vdelayxs A stereo variable delay opcode with high quality interpolation. Description A stereo variable delay opcode with high quality interpolation. Syntax aout1, aout2 vdelayxs ain1, ain2, adl, imd, iws [, ist] Initialization imd -- max. delay time (seconds) iws -- interpolation window size (see below) ist (optional) -- skip initialization if not zero Performance aout1, aout2 -- output audio signals ain1, ain2 -- input audio signals adl -- delay time in seconds This opcode uses high quality (and slow) interpolation, that is much more accurate than the currently available linear and cubic interpolation. The iws parameter sets the number of input samples used for calculating one output sample (allowed values are any integer multiply of 4 in the range 4 - 1024); higher values mean better quality and slower speed. The multichannel opcodes (eg. vdelayxq) allow delaying 2 or 4 variables at once (stereo or quad signals); this is much more efficient than using separate opcodes for each channel. Notes Delay time is measured in seconds (unlike in vdelay and vdelay3), and must be a-rate. The minimum allowed delay is iws/2 samples. Using the same variables as input and output is allowed in these opcodes. In vdelayxw*, changing the delay time has some effects on output volume: a = 1 / (1 + dt) where a is the output gain, and dt is the change of delay time per seconds. These opcodes are best used in the double-precision version of Csound. Examples Here is an example of the use of the vdelayxs opcode. It uses the file vdelayxs.csd. Example of the <emphasis>vdelayxs</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vdelayx, vdelayxq, vdelayxw, vdelayxwq, vdelayxws manual_src~dfsg/opcodes/mixerreceive.xml0000644000000000000000000000765112262561504017623 0ustar rootroot MixerReceive Mixer Opcodes MixerReceive MixerReceive Receives an arate signal from a channel of a buss. Syntax asignal MixerReceive ibuss, ichannel Description Receives an arate signal that has been mixed onto a channel of a buss. Initialization ibuss -- The number of the buss, for example the number of the instrument receiving the signal. ichannel -- The number of the channel. Each buss has nchnls channels. Performance asignal -- The signal that has been mixed onto the indicated channel of the buss. Use of the mixer requires that instruments setting gains have smaller numbers than instruments sending signals, and that instruments sending signals have smaller numbers than instruments receiving those signals. However, an instrument may have any number of sends or receives. After the final signal is received, MixerClear must be invoked to reset the busses to 0 before the next kperiod. Examples instr 220 ; Master output ; It applies a bass enhancement, compression and fadeout ; to the whole piece, outputs signals, and clears the mixer. a1 MixerReceive 220, 0 a2 MixerReceive 220, 1 ; Bass enhancement al1 butterlp a1, 100 al2 butterlp a2, 100 a1 = al1*1.5 + a1 a2 = al2*1.5 + a2 ; Global amplitude shape kenv linseg 0., p5 / 2.0, p4, p3 - p5, p4, p5 / 2.0, 0. a1=a1*kenv a2=a2*kenv ; Compression a1 dam a1, 5000, 0.5, 1, 0.2, 0.1 a2 dam a2, 5000, 0.5, 1, 0.2, 0.1 ; Remove DC bias a1blocked dcblock a1 a2blocked dcblock a2 ; Output signals outs a1blocked, a2blocked MixerClear endin Here is a complete example of the MixerReceive opcode. It uses the file Mixer.csd Complete example of the MixerReceive opcode. Credits Michael Gogins (gogins at pipeline dot com). manual_src~dfsg/opcodes/vmultv_i.xml0000644000000000000000000001042312262561504016770 0ustar rootroot vmultv_i Vectorial:Vectorial operations vmultv_i vmultv_i Performs mutiplication between two vectorial control signals at init time. Description Performs mutiplication between two vectorial control signals at init time. Syntax vmultv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Initialization ifn1 - number of the table hosting the first vector to be processed ifn2 - number of the table hosting the second vector to be processed ielements - number of elements of the two vectors idstoffset - index offset for the destination (ifn1) table (Default=0) isrcoffset - index offset for the source (ifn2) table (Default=0) Performance vmultv_i multiplies two vectorial control signals, that is, each element of the first vector is processed (only) with the corresponding element of the other vector. Each vectorial signal is hosted by a table (ifn1 and ifn2). The number of elements contained in both vectors must be the same. The result is a new vectorial control signal that overrides old values of ifn1. If you want to keep the old ifn1 vector, use vcopy_i opcode to copy it in another table. You can use idstoffset and isrcoffset to specify vectors in any location of the tables. Negative values for idstoffset and isrcoffset are acceptable. If idstoffset is negative, the out of range section of the vector will be discarded. If isrcoffset is negative, the out of range elements will be assumed to be 1 (i.e. the destination elements will not be changed). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 1 (i.e. the destination vector will not be changed for these elements). Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at init time. There's an k-rate version of this opcode called vmultv. All these operators (vaddv_i,vsubv_i,vmultv_i,vdivv_i,vpowv_i,vexpv_i, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/fltext.xml0000644000000000000000000001217412262561504016436 0ustar rootroot FLtext FLTK:Valuators FLtext FLtext A FLTK widget opcode that creates a textbox. Description FLtext allows the user to modify a parameter value by directly typing it into a text field. Syntax kout, ihandle FLtext "label", imin, imax, istep, itype, iwidth, \ iheight, ix, iy Initialization ihandle -- a handle value (an integer number) that unequivocally references a corresponding widget. This is used by other opcodes that modify a widget's properties (see Modifying FLTK Widget Appearance). It is automatically output by FLtext and must not be set by the user label. (The user label is a double-quoted string containing some user-provided text placed near the widget.) label -- a double-quoted string containing some user-provided text, placed near corresponding widget. imin -- minimum value of output range. imax -- maximum value of output range. istep -- a floating-point number indicating the increment of valuator value corresponding to the mouse dragging. The istep argument allows the user to arbitrarily slow mouse motion, enabling arbitrary precision. itype -- an integer number denoting the appearance of the valuator. The itype argument can be set to the following values: 1 - normal behaviour 2 - dragging operation is suppressed, instead it will appear two arrow buttons. A mouse-click on one of these buttons can increase/decrease the output value. 3 - text editing is suppressed, only mouse dragging modifies the output value. iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). Performance kout -- output value FLtext allows the user to modify a parameter value by directly typing it into a text field: FLtext. FLtext. Its value can also be modified by clicking on it and dragging the mouse horizontally. The istep argument allows the user to arbitrarily set the response on mouse dragging. Examples Here is an example of the FLtext opcode. It uses the file FLtext.csd. Example of the FLtext opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLcount, FLjoy, FLknob, FLroller, FLslider Credits Author: &namegabriel; New in version 4.22 Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/mrtmsg.xml0000644000000000000000000000346012262561504016437 0ustar rootroot mrtmsg Real-time MIDI:System Realtime mrtmsg mrtmsg Send system real-time messages to the MIDI OUT port. Description Send system real-time messages to the MIDI OUT port. Syntax mrtmsg imsgtype Initialization imsgtype -- type of real-time message: 1 sends a START message (0xFA); 2 sends a CONTINUE message (0xFB); 0 sends a STOP message (0xFC); -1 sends a SYSTEM RESET message (0xFF); -2 sends an ACTIVE SENSING message (0xFE) Performance Sends a real-time message once, in init stage of current instrument. imsgtype parameter is a flag to indicate the message type. See Also mclock Credits Author: &namegabriel; Italy New in Csound version 3.47 manual_src~dfsg/opcodes/sleighbells.xml0000644000000000000000000001041212262561504017416 0ustar rootroot sleighbells Signal Generators:Models and Emulations sleighbells sleighbells Semi-physical model of a sleighbell sound. Description sleighbells is a semi-physical model of a sleighbell sound. It is one of the PhISEM percussion opcodes. PhISEM (Physically Informed Stochastic Event Modeling) is an algorithmic approach for simulating collisions of multiple independent sound producing objects. Syntax ares sleighbells kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ [, ifreq1] [, ifreq2] Initialization idettack -- period of time over which all sound is stopped inum (optional) -- The number of beads, teeth, bells, timbrels, etc. If zero, the default value is 32. idamp (optional) -- the damping factor, as part of this equation: damping_amount = 0.9994 + (idamp * 0.002) The default damping_amount is 0.9994 which means that the default value of idamp is 0. The maximum damping_amount is 1.0 (no damping). This means the maximum value for idamp is 0.03. The recommended range for idamp is usually below 75% of the maximum value. imaxshake (optional, default=0) -- amount of energy to add back into the system. The value should be in range 0 to 1. ifreq (optional) -- the main resonant frequency. The default value is 2500. ifreq1 (optional) -- the first resonant frequency. The default value is 5300. ifreq2 (optional) -- the second resonant frequency. The default value is 6500. Performance kamp -- Amplitude of output. Note: As these instruments are stochastic, this is only an approximation. Examples Here is an example of the sleighbells opcode. It uses the file sleighbells.csd. Example of the sleighbells opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also bamboo, dripwater, guiro, tambourine Credits Author: &nameperry;, part of the PhISEM (Physically Informed Stochastic Event Modeling) Adapted by &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.07 Added notes by &namerasmus; on May 2002. manual_src~dfsg/opcodes/noteondur.xml0000644000000000000000000000747112262561504017151 0ustar rootroot noteondur Real-time MIDI:Note Output noteondur noteondur Sends a noteon and a noteoff MIDI message both with the same channel, number and velocity. Description Sends a noteon and a noteoff MIDI message both with the same channel, number and velocity. Syntax noteondur ichn, inum, ivel, idur Initialization ichn -- MIDI channel number (1-16) inum -- note number (0-127) ivel -- velocity (0-127) idur -- how long, in seconds, this note should last. Performance noteondur (i-rate note on with duration) sends a noteon and a noteoff MIDI message both with the same channel, number and velocity. Noteoff message is sent after idur seconds are elapsed by the time noteondur was active. noteondur differs from noteondur2 in that noteondur truncates note duration when current instrument is deactivated by score or by real-time playing, while noteondur2 will extend performance time of current instrument until idur seconds have elapsed. In real-time playing, it is suggested to use noteondur also for undefined durations, giving a large idur value. Any number of noteondur opcodes can appear in the same Csound instrument, allowing chords to be played by a single instrument. Examples Here is an example of the noteondur opcode. It uses the file noteondur.csd. Example of the noteondur opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This example generates notes for every note received on the MIDI input. It generates MIDI notes on csound's MIDI output, so be sure to connect something. See Also noteoff, noteon, noteondur2, midion, midion2 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/abetarand.xml0000644000000000000000000000100312262561504017036 0ustar rootroot abetarand abetarand abetarand Deprecated. Description Deprecated as of version 3.49. Use the betarand opcode instead. manual_src~dfsg/opcodes/binit.xml0000644000000000000000000000530412262561504016232 0ustar rootroot binit Spectral Processing:Streaming binit binit PVS tracks to amplitude+frequency conversion. Description The binit opcode takes an input containg a TRACKS pv streaming signal (as generated, for instance by partials) and converts it into a equal-bandwidth bin-frame containing amplitude and frequency pairs (PVS_AMP_FREQ), suitable for overlap-add resynthesis (such as performed by pvsynth) or further PVS streaming phase vocoder signal transformations. For each frequency bin, it will look for a suitable track signal to fill it; if not found, the bin will be empty (0 amplitude). If more than one track fits a certain bin, the one with highest amplitude will be chosen. This means that not all of the input signal is actually 'binned', the operation is lossy. However, in many situations this loss is not perceptually relevant. Syntax fsig binit fin, isize Performance fsig -- output pv stream in PVS_AMP_FREQ format fin -- input pv stream in TRACKS format isize -- FFT size of output (N). Examples Here is an example of the binit opcode. It uses the file binit.csd. Example of the binit opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of an ifd-analysis signal, conversion to bin frames and overlap-add resynthesis. Credits Author: &namevictor; February 2006 New in Csound5.01 manual_src~dfsg/opcodes/betarand.xml0000644000000000000000000001200612262561504016702 0ustar rootroot x-class noise generators betarand Signal Generators:Random (Noise) Generators betarand betarand Beta distribution random number generator (positive values only). Description Beta distribution random number generator (positive values only). This is an x-class noise generator. Syntax ares betarand krange, kalpha, kbeta ires betarand krange, kalpha, kbeta kres betarand krange, kalpha, kbeta Performance krange -- range of the random numbers (0 - krange). kalpha -- alpha value. If kalpha is smaller than one, smaller values favor values near 0. kbeta -- beta value. If kbeta is smaller than one, smaller values favor values near krange. If both kalpha and kbeta equal one we have uniform distribution. If both kalpha and kbeta are greater than one we have a sort of Gaussian distribution. Outputs only positive numbers. For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the betarand opcode. It uses the file betarand.csd. Example of the betarand opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like: i 1 time 0.00267: 85.74227 i 1 time 0.20267: 12.07606 i 1 time 0.40267: 25.03239 i 1 time 0.60267: 0.42037 i 1 time 0.80267: 76.69589 i 1 time 1.00000: 29.73339 i 1 time 1.20267: 48.29811 i 1 time 1.40267: 75.46507 i 1 time 1.60267: 74.80686 i 1 time 1.80000: 81.37473 i 1 time 2.00000: 55.48827 Seeding from current time 3472120656 i 2 time 3.00267: 57.21408 i 2 time 3.20267: 30.95705 i 2 time 3.40267: 19.71687 i 2 time 3.60000: 64.48965 i 2 time 3.80267: 72.35818 i 2 time 4.00000: 49.65395 i 2 time 4.20000: 55.25888 i 2 time 4.40267: 3.98308 i 2 time 4.60267: 52.98075 i 2 time 4.80267: 58.07925 i 2 time 5.00000: 56.38914 See Also seed, bexprnd, cauchy, exprand, gauss, linrand, pcauchy, poisson, trirand, unirand, weibull Credits Author: &nameparis; MIT, Cambridge 1995 Existed in 3.30 manual_src~dfsg/opcodes/gendyc.xml0000644000000000000000000001241412262561504016376 0ustar rootroot gendyc Signal Generators:Models and Emulations gendyc gendyc Dynamic stochastic approach to waveform synthesis using cubic interpolation. Description Implementation with cubic interpolation of the Génération Dynamique Stochastique (GENDYN), a dynamic stochastic approach to waveform synthesis conceived by Iannis Xenakis. Syntax ares gendyc kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl [, initcps] [, knum] kres gendyc kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl [, initcps] [, knum] Initialization initcps (optional, default=12) -- max number of control points. Performance kamp -- amplitude. kampdist -- choice of probability distribution for the next perturbation of the amplitude of a control point. The valid distributions are: 0 - LINEAR 1 - CAUCHY 2 - LOGIST 3 - HYPERBCOS 4 - ARCSINE 5 - EXPON 6 - SINUS (external k-rate signal) If kampdist=6, the user can use an external k-rate signal through kadpar. kdurdist -- choice of distribution for the perturbation of the current inter control point duration. See kampdist for the valid distributions. If kdurdist=6, the user can use an external k-rate signal through kddpar. kadpar -- parameter for the kampdist distribution. Should be in the range of 0.0001 to 1. kddpar -- parameter for the kdurdist distribution. Should be in the range of 0.0001 to 1. kminfreq -- minimum allowed frequency of oscillation. kmaxfreq -- maximum allowed frequency of oscillation. kampscl -- multiplier for the distribution's delta value for amplitude (1.0 is full range). kdurscl -- multiplier for the distribution's delta value for duration. knum (optional, default=initcps) -- current number of utilized control points. The waveform is generated by knum - 1 segments and is repeated in the time. The vertexes (control points) are moved according to a stochastic action and they are limited within the boundaries of a mirror formed by an amplitude barrier and a time barrier. Examples Here is an example of the gendyc opcode. It uses the file gendyc.csd. Example of the gendyc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. References Formalized Music (1992, Stuyvesant, NY: Pendragon Press), pp. 246 - 254, 289 - 322. See Also gendy gendyx Credits Based on Nick Collins's Gendy1 and Bhob Rainey's Gendy4 (SuperCollider) Author: &nametito; January 2012 New in Csound version 5.16 manual_src~dfsg/opcodes/pconvolve.xml0000644000000000000000000001040212262561504017133 0ustar rootroot pconvolve Signal Modifiers:Convolution and Morphing pconvolve convolve Convolution based on a uniformly partitioned overlap-save algorithm Description Convolution based on a uniformly partitioned overlap-save algorithm. Compared to the convolve opcode, pconvolve has these benefits: small delay possible to run in real-time for shorter impulse files no pre-process analysis pass can often render faster than convolve Syntax ar1 [, ar2] [, ar3] [, ar4] pconvolve ain, ifilcod [, ipartitionsize, ichannel] Initialization ifilcod -- integer or character-string denoting an impulse response soundfile. Multichannel files are supported, the file must have the same sample-rate as the orc. [Note: cvanal files cannot be used!] Keep in mind that longer files require more calculation time [and probably larger partition sizes and more latency]. At current processor speeds, files longer than a few seconds may not render in real-time. ipartitionsize (optional, defaults to the output buffersize [-b]) -- the size in samples of each partition of the impulse file. This is the parameter that needs tweaking for best performance depending on the impulse file size. Generally, a small size means smaller latency but more computation time. If you specify a value that is not a power-of-2 the opcode will find the next power-of-2 greater and use that as the actual partition size. ichannel (optional) -- which channel to use from the impulse response data file. Performance ain -- input audio signal. The overall latency of the opcode can be calculated as such [assuming ipartitionsize is a power of 2] ilatency = (ksmps < ipartitionsize ? ipartitionsize + ksmps : ipartitionsize)/sr Examples Instrument 1 shows an example of real-time convolution. Instrument 2 shows how to do file-based convolution with a 'look ahead' method to remove all delay. NOTE You can download impulse response files from noisevault.com or replace the filenames with your own impulse files. Here is an example of the pconvolve opcode. It uses the file pconvolve.csd. Example of the pconvolve opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also convolve, dconv. Credits Author: Matt Ingalls 2004 manual_src~dfsg/opcodes/timout.xml0000644000000000000000000000455112262561504016451 0ustar rootroot timout Instrument Control:Program Flow Control timout timout Conditional branch during p-time depending on elapsed note time. Description Conditional branch during p-time depending on elapsed note time. istrt and idur specify time in seconds. The branch to label will become effective at time istrt, and will remain so for just idur seconds. Note that timout can be reinitialized for multiple activation within a single note (see example under reinit). Syntax timout istrt, idur, label where label is in the same instrument block and is not an expression. Examples Here is an example of the timout opcode. It uses the file timout.csd. Example of the timout opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also goto, if, igoto, kgoto, tigoto manual_src~dfsg/opcodes/pvbufread.xml0000644000000000000000000001447312262561504017112 0ustar rootroot pvbufread Spectral Processing:STFT pvbufread pvbufread Reads from a phase vocoder analysis file and makes the retrieved data available. Description pvbufread reads from a pvoc file and makes the retrieved data available to any following pvinterp and pvcross units that appear in an instrument before a subsequent pvbufread (just as lpread and lpreson work together). The data is passed internally and the unit has no output of its own. Syntax pvbufread ktimpnt, ifile Initialization ifile -- the pvoc number (n in pvoc.n) or the name in quotes of the analysis file made using pvanal. (See pvoc.) Performance ktimpnt -- the passage of time, in seconds, through this file. ktimpnt must always be positive, but can move forwards or backwards in time, be stationary or discontinuous, as a pointer into the analysis file. Examples The example below shows an example using pvbufread with pvinterp to interpolate between the sound of an oboe and the sound of a clarinet. The value of kinterp returned by a linseg is used to determine the timing of the transitions between the two sounds. The interpolation of frequencies and amplitudes are controlled by the same factor in this example, but for other effects it might be interesting to not have them synchronized in this way. In this example the sound will begin as a clarinet, transform into the oboe and then return again to the clarinet sound. The value of kfreqscale2 is 1.065 because the oboe in this case is a semitone higher in pitch than the clarinet and this brings them approximately to the same pitch. The value of kampscale2 is 0.75 because the analyzed clarinet was somewhat louder than the analyzed oboe. The setting of these two parameters make the transition quite smooth in this case, but such adjustments are by no means necessary or even advocated. ktime1 line 0, p3, 3.5 ; used as index in the "oboe.pvoc" file ktime2 line 0, p3, 4.5 ; used as index in the "clar.pvoc" file kinterp linseg 1, p3*0.15, 1, p3*0.35, 0, p3*0.25, 0, p3*0.15, 1, p3*0.1, 1 pvbufread ktime1, "oboe.pvoc" apv pvinterp ktime2,1,"clar.pvoc", 1, 1.065, 1, 0.75, 1-kinterp, 1-kinterp Below is an example using pvbufread with pvcross. In this example the amplitudes used in the resynthesis gradually change from those of the oboe to those of the clarinet. The frequencies, of course, remain those of the clarinet throughout the process since pvcross does not use the frequency data from the file read by pvbufread. ktime1 line 0, p3, 3.5 ; used as index in the "oboe.pvoc" file ktime2 line 0, p3, 4.5 ; used as index in the "clar.pvoc" file kcross expon 0.001, p3, 1 pvbufread ktime1, "oboe.pvoc" apv pvcross ktime2, 1, "clar.pvoc", 1-kcross, kcross Here is a complete example of the pvbufread opcode. It uses the file pvbufread.csd Example of the pvbufread opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvcross, pvinterp, pvread, tableseg, tablexseg Credits Author: Richard Karpen Seattle, WA USA 1997 manual_src~dfsg/opcodes/kbetarand.xml0000644000000000000000000000100512262561504017052 0ustar rootroot kbetarand kbetarand kbetarand Deprecated. Description Deprecated as of version 3.49. Use the betarand opcode instead. manual_src~dfsg/opcodes/tab2pvs.xml0000644000000000000000000000420512262561504016505 0ustar rootroot tab2pvs Spectral Processing:Streaming tab2pvs tab2pvs Copies spectral data from t-variables. Description Copies a pvs frame from a t-variable. Currently only AMP+FREQ is produced. This opcode requires the t-type to be defined, which means it only works in the new bison/flex-based parser. Syntax fsig tab2pvs tvar[,ihopsize, iwinsize, iwintype] Performance tvar -- t-variable containing the input. It is produced at every k-period, but may not contain a new frame, pvs frames are produced at their own frame rate that is independent of kr. The size of this vector will determine the fftsize, N = size - 2. fsig -- output fsig to be copied. iolap -- size of the analysis overlap, defaults to isize/4. iwinsize -- size of the analysis window, defaults to isize. iwintype -- type of analysis window, defaults to 1, Hanning. Examples Example t1 init 1026 ... fsig1 tab2pvst1 aout pvsynth fsig1 Credits Author: &namevictor; October 2011 New plugin in version 5 October 2011. manual_src~dfsg/opcodes/flslidbnkset.xml0000644000000000000000000001030112262561504017602 0ustar rootroot FLslidBnkSet FLTK:Other FLslidBnkSet FLslidBnkSet modify the values of a slider bank. Description FLslidBnkSet modifies the values of a slider bank according to an array of values stored in a table. Syntax FLslidBnkSet ihandle, ifn [, istartIndex, istartSlid, inumSlid] Initialization ihandle - handle of the sliderBnk (to be used to set its values). ifn - number of a table containing an array of values to set each slider to. istartIndex - (optional) starting index of the table element of to be evaluated firstly. Default value is zero istartSlid - (optional) starting slider to be evaluated. Default 0, denoting the first slider. inumSlid - (optional) number of sliders to be updated. Default 0, denoting all sliders. Performance FLslidBnkSet modifies the values of a slider bank (created with FLslidBnk or with FLvslidBnk) according to an array of values stored into table ifn. It actually allows to update an FLslidBnk (or FLvslidBnk) bank of sliders (for instance, using the slider8table opcode) to a set of values located in a table. User has to set ihandle argument to the handle got from FLslidBnkGetHandle opcode. It works at init-rate only. It is possible to reset only a range of sliders, by using the optional arguments istartIndex, istartSlid, inumSlid There is a k-rate version of this opcode called FLslidBnkSetk. See Also FLslider, FLslidBnkGetHandle, FLslidBnk, FLslidBnk2, FLvslidBnk, FLvslidBnk2 FLslidBnk2Set, FLslidBnkSetk Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/flsetvali.xml0000644000000000000000000000617012262561504017120 0ustar rootroot FLsetVal_i FLTK:Other FLsetVal_i FLsetVal_i Sets the value of a FLTK valuator to a number provided by the user. Description FLsetVal_i forces the value of a valuator to a number provided by the user. Syntax FLsetVal_i ivalue, ihandle Initialization ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. Performance ivalue -- Value to set the widget to. FLsetVal is not fully implemented yet, and may crash in certain cases (e.g. when setting the value of a widget connected to a FLvalue widget- in this case use two separate FLsetVal_i). See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/s16b14.xml0000644000000000000000000001045412262561504016047 0ustar rootroot s16b14 Real-time MIDI:Slider Banks s16b14 s16b14 Creates a bank of 16 different 14-bit MIDI control message numbers. Description Creates a bank of 16 different 14-bit MIDI control message numbers. Syntax i1,...,i16 s16b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, \ initvalue1, ifn1,..., ictlno_msb16, ictlno_lsb16, imin16, imax16, initvalue16, ifn16 k1,...,k16 s16b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, \ initvalue1, ifn1,..., ictlno_msb16, ictlno_lsb16, imin16, imax16, initvalue16, ifn16 Initialization i1 ... i64 -- output values ichan -- MIDI channel (1-16) ictlno_msb1 .... ictlno_msb32 -- MIDI control number, most significant byte (0-127) ictlno_lsb1 .... ictlno_lsb32 -- MIDI control number, least significant byte (0-127) imin1 ... imin64 -- minimum values for each controller imax1 ... imax64 -- maximum values for each controller init1 ... init64 -- initial value for each controller ifn1 ... ifn64 -- function table for conversion for each controller Performance k1 ... k64 -- output values s16b14 is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. s16b14 allows a bank of 16 different MIDI control message numbers. It uses 14-bit values instead of MIDI's normal 7-bit values. As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. In the i-rate version of s16b14, there is not an initial value input argument. The output is taken directly from the current status of internal controller array of Csound. Credits Author: &namegabriel; Italy December 1998 New in Csound version 3.50 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/exitnow.xml0000644000000000000000000000276712262561504016634 0ustar rootroot exitnow Instrument Control:Realtime Performance Control exitnow exitnow Exit Csound as fast as possible, with no cleaning up. Description In Csound4 calls an exit function to leave Csound as fast as possible. On Csound5 exits back to the driving code. Syntax exitnow Performance Stops Csound on the initialisation cycle. Examples Here is an example of the exitnow opcode. It uses the file exitnow.csd. Example of the exitnow opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. manual_src~dfsg/opcodes/flsavesnap.xml0000644000000000000000000001556712262561504017303 0ustar rootroot FLsavesnap FLTK:Other FLsavesnap FLsavesnap Saves all snapshots currently created into a file. Description FLsavesnap saves all snapshots currently created (i.e. the entire memory bank) into a file. Syntax FLsavesnap "filename" [, igroup] Initialization filename -- a double-quoted string corresponding to a file to store a bank of snapshots. igroup -- (optional) an integer number referring to a snapshot-related group of widget. It allows to get/set, or to load/save the state of a subset of valuators. Default value is zero that refers to the first group. The group number is determined by the opcode FLsetSnapGroup. The igroup parameter has not been yet fully implemented in the current version of csound. Please do not rely on it yet. Performance FLsavesnap saves all snapshots currently created (i.e. the entire memory bank) into a file whose name is filename. Since the file is a text file, snapshot values can also be edited manually by means of a text editor. The format of the data stored in the file is the following (at present time, this could be changed in next Csound version): ----------- 0 ----------- FLvalue 0 0 1 0 "" FLvalue 0 0 1 0 "" FLvalue 0 0 1 0 "" FLslider 331.946 80 5000 -1 "frequency of the first oscillator" FLslider 385.923 80 5000 -1 "frequency of the second oscillator" FLslider 80 80 5000 -1 "frequency of the third oscillator" FLcount 0 0 10 0 "this index must point to the location number where snapshot is stored" FLbutton 0 0 1 0 "Store snapshot to current index" FLbutton 0 0 1 0 "Save snapshot bank to disk" FLbutton 0 0 1 0 "Load snapshot bank from disk" FLbox 0 0 1 0 "" ----------- 1 ----------- FLvalue 0 0 1 0 "" FLvalue 0 0 1 0 "" FLvalue 0 0 1 0 "" FLslider 819.72 80 5000 -1 "frequency of the first oscillator" FLslider 385.923 80 5000 -1 "frequency of the second oscillator" FLslider 80 80 5000 -1 "frequency of the third oscillator" FLcount 1 0 10 0 "this index must point to the location number where snapshot is stored" FLbutton 0 0 1 0 "Store snapshot to current index" FLbutton 0 0 1 0 "Save snapshot bank to disk" FLbutton 0 0 1 0 "Load snapshot bank from disk" FLbox 0 0 1 0 "" ----------- 2 ----------- ..... etc... ----------- 3 ----------- ..... etc... --------------------------- As you can see, each snapshot contain several lines. Each snapshot is separated from previous and next snapshot by a line of this kind: "----------- snapshot Num -----------" Then there are several lines containing data. Each of these lines corresponds to a widget. The first field of each line is an unquoted string containing opcode name corresponding to that widget. Second field is a number that expresses current value of a snapshot. In current version, this is the only field that can be modified manually. The third and fourth fields shows minimum and maximum values allowed for that valuator. The fifth field is a special number that indicates if the valuator is linear (value 0), exponential (value -1), or is indexed by a table interpolating values (negative table numbers) or non-interpolating (positive table numbers). The last field is a quoted string with the label of the widget. Last line of the file is always "---------------------------" . Note that FLvalue andFLbox are not valuators and their values are fixed, so they cannot be modified. For purposes of snapshot saving, widgets can be grouped, so that snapshots affect only a defined group of widgets. The opcode FLsetSnapGroup is used to specify the group for all widgets declared after it, until the next FLsetSnapGroup statement. Examples Here is a simple example of the FLTK snapshot saving. It uses the file FLsavesnap_simple.csd. Example of FLTK snapshot saving. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of FLTK snapshot saving using snapshot groups. It uses the file FLsavesnap.csd. Example of FLTK snapshot saving using snapshot groups. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLgetsnap, FLloadsnap, FLsetSnapGroup, FLrun, FLsetsnap, FLupdate Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/turnon.xml0000644000000000000000000000312212262561504016446 0ustar rootroot turnon Instrument Control:Duration Control turnon turnon Activate an instrument for an indefinite time. Description Activate an instrument for an indefinite time. Syntax turnon insnum [, itime] Initialization insnum -- instrument number to be activated itime (optional, default=0) -- delay, in seconds, after which instrument insnum will be activated. Default is 0. Performance turnon activates instrument insnum after a delay of itime seconds, or immediately if itime is not specified. Instrument is active until explicitly turned off. (See turnoff.) See Also turnoff, turnoff2 manual_src~dfsg/opcodes/ptablew.xml0000644000000000000000000002201312262561504016557 0ustar rootroot ptablew Table Control:Read/Write Opreations ptablew ptablew Change the contents of existing function tables of any length. Description This opcode operates on existing function tables, changing their contents. ptablew is for writing at k- or at a-rates, with the table number being specified at init time. Using ptablew with i-rate signal and index values is allowed, but the specified data will always be written to the function table at k-rate, not during the initialization pass. The valid combinations of variable types are shown by the first letter of the variable names. Syntax ptablew asig, andx, ifn [, ixmode] [, ixoff] [, iwgmode] ptablew isig, indx, ifn [, ixmode] [, ixoff] [, iwgmode] ptablew ksig, kndx, ifn [, ixmode] [, ixoff] [, iwgmode] Initialization asig, isig, ksig -- The value to be written into the table. andx, indx, kndx -- Index into table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0) ifn -- Table number. Must be >= 1. Floats are rounded down to an integer. If a table number does not point to a valid table, or the table has not yet been loaded (GEN01) then an error will result and the instrument will be de-activated. ixmode (optional, default=0) -- index mode. 0 = xndx and ixoff ranges match the length of the table. !=0 = xndx and ixoff have a 0 to 1 range. ixoff (optional, default=0) -- index offset. 0 = Total index is controlled directly by xndx, i.e. the indexing starts from the start of the table. !=0 = Start indexing from somewhere else in the table. Value must be positive and less than the table length (ixmode = 0) or less than 1 (ixmode != 0). iwgmode (optional, default=0) -- Wrap and guardpoint mode. 0 = Limit mode. 1 = Wrap mode. 2 = Guardpoint mode. Performance Limit mode (0) Limit the total index (ndx + ixoff) to between 0 and the guard point. For a table of length 5, this means that locations 0 to 3 and location 4 (the guard point) can be written. A negative total index writes to location 0. Wrap mode (1) Wrap total index value into locations 0 to E, where E is one less than the table length. For example, wrap into a 0 to 3 range - so that total index 6 writes to location 2. Guardpoint mode (2) The guardpoint is written at the same time as location 0 is written - with the same value. This facilitates writing to tables which are intended to be read with interpolation for producing smooth cyclic waveforms. In addition, before it is used, the total index is incremented by half the range between one location and the next, before being rounded down to the integer address of a table location. Normally (igwmode = 0 or 1) for a table of length 5 - which has locations 0 to 3 as the main table and location 4 as the guard point, a total index in the range of 0 to 0.999 will write to location 0. ("0.999" means just less than 1.0.) 1.0 to 1.999 will write to location 1 etc. A similar pattern holds for all total indexes 0 to 4.999 (igwmode = 0) or to 3.999 (igwmode = 1). igwmode = 0 enables locations 0 to 4 to be written - with the guardpoint (4) being written with a potentially different value from location 0. With a table of length 5 and the iwgmode = 2, then when the total index is in the range 0 to 0.499, it will write to locations 0 and 4. Range 0.5 to 1.499 will write to location 1 etc. 3.5 to 4.0 will also write to locations 0 and 4. This way, the writing operation most closely approximates the results of interpolated reading. Guard point mode should only be used with tables that have a guardpoint. Guardpoint mode is accomplished by adding 0.5 to the total index, rounding to the next lowest integer, wrapping it modulo the factor of two which is one less than the table length, writing the table (locations 0 to 3 in our example) and then writing to the guard point if index = 0. ptablew has no output value. The last three parameters are optional and have default values of 0. Caution with k-rate table numbers At k-rate or a-rate, if a table number of < 1 is given, or the table number points to a non-existent table, or to one which has a length of 0 (it is to be loaded from a file later) then an error will result and the instrument will be deactivated. kfn and afn must be initialized at the appropriate rate using init. Attempting to load an i-rate value into kfn or afn will result in an error. Note that ptablew is always a k-rate opcode. This means that even its i-rate version runs at k-rate and will write the value of the i-rate variable. For this reason, the following code will not work as expected: <CsoundSynthesizer> <CsOptions> </CsOptions> <CsInstruments> giFt ftgen 1, 0, 8, 2, 0 instr 1 indx = 0 ptablew 10, indx, giFt ival tab_i indx, giFt print ival endin </CsInstruments> <CsScore> i 1 0 1 </CsScore> </CsoundSynthesizer> Although it may seem this program should print a 10 to the console. It will print 0, because tab_i will read the value at the initialization of the note, before the first performance pass, when ptablew writes its value. See Also tableiw, tablewkt Credits Author: &namejohn; after Robin Whittle Feb 2012 manual_src~dfsg/opcodes/minabs.xml0000644000000000000000000000616412262561504016403 0ustar rootroot minabs Signal Modifiers:Comparators and Accumulators minabs minabs Produces a signal that is the minimum of the absolute values of any number of input signals. Description The minabs opcode takes any number of a-rate or k-rate signals as input (all of the same rate), and outputs a signal at the same rate that is the minimum of all of the inputs. It is identical to the min opcode except that it takes the absolute value of each input before comparing them. Therefore, the output is always non-negative. For a-rate signals, the inputs are compared one sample at a time (i.e. minabs does not scan an entire ksmps period of a signal for its local minimum as the max_k opcode does). Syntax amin minabs ain1, ain2 [, ain3] [, ain4] [...] kmin minabs kin1, kin2 [, kin3] [, kin4] [...] Performance ain1, ain2, ... -- a-rate signals to be compared. kin1, kin2, ... -- k-rate signals to be compared. Examples Here is an example of the minabs opcode. It uses the file minabs.csd. Example of the minabs opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also maxabs, max, min, maxaccum, minaccum, maxabsaccum, minabsaccum, max_k Credits Author: &nameanthony; March 2006 New in Csound version 5.01 manual_src~dfsg/opcodes/pvsinfo.xml0000644000000000000000000000537512262561504016621 0ustar rootroot pvsinfo Spectral Processing:Streaming pvsinfo pvsinfo Get information from a PVOC-EX formatted source. Description Get format information about fsrc, whether created by an opcode such as pvsanal, or obtained from a PVOCEX file by pvsfread. This information is available at init time, and can be used to set parameters for other pvs opcodes, and in particular for creating function tables (e.g. for pvsftw), or setting the number of oscillators for pvsadsyn. Syntax ioverlap, inumbins, iwinsize, iformat pvsinfo fsrc Initialization ioverlap -- The stream overlap size. inumbins -- The number of analysis bins (amplitude+frequency) in fsrc. The underlying FFT size is calculated as (inumbins -1) * 2. iwinsize -- The analysis window size. May be larger than the FFT size. iformat -- The analysis frame format. If fsrc is created by an opcode, iformat will always be 0, signifying amplitude+frequency. If fsrc is defined from a PVOC-EX file, iformat may also have the value 1 or 2 (amplitude+phase, complex). Examples Here is an example of the pvsinfo opcode. It uses the file pvsinfo.csd. Example of the pvsinfo opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example will produce the following output: instr 1: iovl = 256.000 inb = 513.000 iws = 2048.000 ifmt = 0.000 Credits Author: &namerichard; August 2001 New in version 4.13 manual_src~dfsg/opcodes/interp.xml0000644000000000000000000000735712262561504016440 0ustar rootroot interp Signal Modifiers:Sample Level Operators interp interp Converts a control signal to an audio signal using linear interpolation. Description Converts a control signal to an audio signal using linear interpolation. Syntax ares interp ksig [, iskip] [, imode] Initialization iskip (optional, default=0) -- initial disposition of internal save space (see reson). The default value is 0. imode (optional, default=0) -- sets the initial output value to the first k-rate input instead of zero. The following graphs show the output of interp with a constant input value, in the original, when skipping init, and in the new mode: iskip=0, imode=0 | ________ | / | / | / |/ -+------------ | iskip=1, imode=0 (prev) | __________ | / |/ | | -+------------ | iskip=0, imode=1 |____________ | | | | -+------------ | Performance ksig -- input k-rate signal. interp converts a control signal to an audio signal. It uses linear interpolation between successive kvals. Examples Here is an example of the interp opcode. It uses the file interp.csd. Example of the interp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also diff, downsamp, integ, samphold, upsamp More information on this opcode: http://www.csounds.com/journal/issue10/CsoundRates.html , written by Andrés Cabrera Credits Updated November 2002, thanks to a note from both &namerasmus; and &nameistvan;. manual_src~dfsg/opcodes/dam.xml0000644000000000000000000001110612262561504015663 0ustar rootroot dam Signal Modifiers:Amplitude Modifiers dam dam A dynamic compressor/expander. Description This opcode dynamically modifies a gain value applied to the input sound ain by comparing its power level to a given threshold level. The signal will be compressed/expanded with different factors regarding that it is over or under the threshold. Syntax ares dam asig, kthreshold, icomp1, icomp2, irtime, iftime Initialization icomp1 -- compression ratio for upper zone. icomp2 -- compression ratio for lower zone irtime -- gain rise time in seconds. Time over which the gain factor is allowed to raise of one unit. iftime -- gain fall time in seconds. Time over which the gain factor is allowed to decrease of one unit. Performance asig -- input signal to be modified kthreshold -- level of input signal which acts as the threshold. Can be changed at k-time (e.g. for ducking) Note on the compression factors: A compression ratio of one leaves the sound unchanged. Setting the ratio to a value smaller than one will compress the signal (reduce its volume) while setting the ratio to a value greater than one will expand the signal (augment its volume). Examples Because the results of the dam opcode can be subtle, I recommend looking at them in a graphical audio editor program like audacity. audacity is available for Linux, Windows, and the MacOS and may be downloaded from http://audacity.sourceforge.net. Here is an example of the dam opcode. It uses the file dam.csd, and beats.wav. An example of the dam opcode compressing an audio signal. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This example compresses the audio file beats.wav. You should hear a drum pattern repeat twice. The second time, the sound should be quieter (compressed) than the first. Here is another example of the dam opcode. It uses the file dam_expanded.csd, and beats.wav. An example of the dam opcode expanding an audio signal. This example expands the audio file beats.wav. You should hear a drum pattern repeat twice. The second time, the sound should be louder (expanded) than the first. To prevent distortion the volume of the signal has been lowered. See Also compress Credits Author: Marc Resibois Belgium 1997 New in version 3.47 manual_src~dfsg/opcodes/bbcuts.xml0000644000000000000000000001115412262561504016407 0ustar rootroot bbcuts Signal Generators:Sample Playback bbcuts bbcuts Generates breakbeat-style cut-ups of a stereo audio stream. Description The BreakBeat Cutter automatically generates cut-ups of a source audio stream in the style of drum and bass/jungle breakbeat manipulations. There are two versions, for mono (bbcutm) or stereo (bbcuts) sources. Whilst originally based on breakbeat cutting, the opcode can be applied to any type of source audio. The prototypical cut sequence favoured over one bar with eighth note subdivisions would be 3+ 3R + 2 where we take a 3 unit block from the source's start, repeat it, then 2 units from the 7th and 8th eighth notes of the source. We talk of rendering phrases (a sequence of cuts before reaching a new phrase at the beginning of a bar) and units (as subdivision th notes). The opcode comes most alive when multiple synchronised versions are used simultaneously. Syntax a1,a2 bbcuts asource1, asource2, ibps, isubdiv, ibarlength, iphrasebars, \ inumrepeats [, istutterspeed] [, istutterchance] [, ienvchoice] Initialization ibps -- Tempo to cut at, in beats per second. isubdiv -- Subdivisions unit, for a bar. So 8 is eighth notes (of a 4/4 bar). ibarlength -- How many beats per bar. Set to 4 for default 4/4 bar behaviour. iphrasebars -- The output cuts are generated in phrases, each phrase is up to iphrasebars long inumrepeats -- In normal use the algorithm would allow up to one additional repeat of a given cut at a time. This parameter allows that to be changed. Value 1 is normal- up to one extra repeat. 0 would avoid repeating, and you would always get back the original source except for enveloping and stuttering. istutterspeed -- (optional, default=1) The stutter can be an integer multiple of the subdivision speed. For instance, if subdiv is 8 (quavers) and stutterspeed is 2, then the stutter is in semiquavers (sixteenth notes= subdiv 16). The default is 1. istutterchance -- (optional, default=0) The tail of a phrase has this chance of becoming a single repeating one unit cell stutter (0.0 to 1.0). The default is 0. ienvchoice -- (optional, default=1) choose 1 for on (exponential envelope for cut grains) or 0 for off. Off will cause clicking, but may give good noisy results, especially for percussive sources. The default is 1, on. Performance asource -- The audio signal to be cut up. This version runs in real-time without knowledge of future audio. Examples Here is an example of the bbcuts opcode. It uses the file bbcuts.csd. Example of the bbcuts opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also bbcutm Credits Author: Nick Collins London August 2001 New in version 4.13 manual_src~dfsg/opcodes/moogladder.xml0000644000000000000000000000622512262561504017245 0ustar rootroot moogladder Signal Modifiers:Standard Filters:Resonant moogladder moogladder Moog ladder lowpass filter. Description Moogladder is an new digital implementation of the Moog ladder filter based on the work of Antti Huovilainen, described in the paper "Non-Linear Digital Implementation of the Moog Ladder Filter" (Proceedings of DaFX04, Univ of Napoli). This implementation is probably a more accurate digital representation of the original analogue filter. Syntax asig moogladder ain, kcf, kres[, istor] asig moogladder ain, acf, kres[, istor] asig moogladder ain, kcf, ares[, istor] asig moogladder ain, acf, ares[, istor] Initialization istor --initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance asig -- input signal. kcf/acf -- filter cutoff frequency kres/ares -- resonance, generally < 1, but not limited to it. Higher than 1 resonance values might cause aliasing, analogue synths generally allow resonances to be above 1. Examples Here is an example of the moogladder opcode. It uses the file moogladder.csd. Example of the moogladder opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; January 2005 New plugin in version 5 January 2005. Audio rate parameters introduced in version 6.02 October 2013. manual_src~dfsg/opcodes/midifilestatus.xml0000644000000000000000000000201212262561504020144 0ustar rootroot midifilestatus Instrument Control:Sensing and Control midifilestatus midifilestatus Returns the playback status of MIDI file input. Description Returns the current playback status at k-rate, of the input MIDI file, 1 if file is playing, 0 if the end-of-the file has been reached. Syntax ksig midifilestatus Credits Author: &namevictor; March 2006 New in Csound6 manual_src~dfsg/opcodes/vincr.xml0000644000000000000000000000624612262561504016254 0ustar rootroot vincr Mathematical Operations:Comparators and Accumulators vincr vincr Accumulates audio signals. Description vincr increments one audio variable with another signal, i.e. it accumulates output. Syntax vincr accum, aincr Performance accum -- audio-rate accumulator variable to be incremented aincr -- incrementing signal vincr (variable increment) and clear are intended to be used together. vincr stores the result of the sum of two audio variables into the first variable itself (which is intended to be used as an accumulator in polyphony). The accumulator is typically a global variable that is used to combine signals from several sources (different instruments or instrument instances) for further processing (for example, via a global effect that reads the accumulator) or for outputting the combined signal by some means other than one of the out opcodes (eg. via the fout opcode). After the accumulator is used, the accumulator variable should be set to zero by means of the clear opcode (or it will explode). Examples The following example uses the vincr opcode. It uses the file vincr.csd. Example of the vincr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This is another example uses the vincr opcode. It uses the file vincr-complex.csd. See Also clear Credits Author: &namegabriel; Italy 1999 New in Csound version 3.56 manual_src~dfsg/opcodes/tlineto.xml0000644000000000000000000000531212262561504016602 0ustar rootroot tlineto Signal Modifiers:Standard Filters:Control tlineto tlineto Generate glissandos starting from a control signal. Description Generate glissandos starting from a control signal with a trigger. Syntax kres tlineto ksig, ktime, ktrig Performance kres -- Output signal. ksig -- Input signal. ktime -- Time length of glissando in seconds. ktrig -- Trigger signal. tlineto is similar to lineto but can be applied to any kind of signal (not only stepped signals) without producing discontinuities. Last value of each segment is sampled and held from input signal each time ktrig value is set to a nonzero value. Normally ktrig signal consists of a sequence of zeroes (see trigger opcode). The effect of glissando is quite different from port. Since in these cases, the lines are straight. Also the context of usage is different. Examples Here is an example of the tlineto opcode. It uses the file tlineto.csd. Example of the tlineto opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also lineto Credits Author: &namegabriel; New in Version 4.13 manual_src~dfsg/opcodes/instance.xml0000644000000000000000000001242412262561504016732 0ustar rootroot instance Instrument Control:Invocation instance instance Schedules a new instrument instance, storing the instance handle in a variable. Description Schedules a new instrument instance, returning a handle that can be used later to refer directly to the running instance. This opcode is similar to schedule, but has the added facility of retrieving the instance handle. Syntax iHandle instance insnum, iwhen, idur [, ip4] [, ip5] [...] iHandle instance "insname", iwhen, idur [, ip4] [, ip5] [...] Initialization iHandle -- this variable will contain a handle to the event instance. It can be used, for example, to stop the given instrument instance via the kill opcode. Note that the instance handle is only valid once the instrument is initialised. insnum -- instrument number. Equivalent to p1 in a score i statement. insnum must be a number greater than the number of the calling instrument. insname -- A string (in double-quotes) representing a named instrument. iwhen -- start time of the new event. Equivalent to p2 in a score i statement. iwhen must be nonnegative. If iwhen is zero, insum must be greater than or equal to the p1 of the current instrument. idur -- duration of event. Equivalent to p3 in a score i statement. ip4, ip5, ... -- Equivalent to p4, p5, etc., in a score i statement. Performance instance adds a new score event. The arguments, including options, are the same as in a score. The iwhen time (p2) is measured from the time of this event. If the duration is zero or negative the new event is of MIDI type, and inherits the release sub-event from the scheduling instruction. Note that the instance opcode can't accept string p-fields. If you need to pass strings when instantiating an instrument, use the scoreline or scoreline_i opcode. Examples Here is an example of the instance opcode. It uses the file instance.csd. Example of the instance opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also schedwhen See also event, event_i, schedwhen, schedkwhen, schedkwhennamed, scoreline, scoreline_i More information on this opcode: http://www.csounds.com/journal/issue15/phrase_loops.html , written by Jim Aikin Credits Author: &namevictor; November 2013 New in Csound version 3.491 Based on work by &namegabriel; Thanks goes to David Gladstein, for clarifying the iwhen parameter. manual_src~dfsg/opcodes/fmmetal.xml0000644000000000000000000001056412262561504016556 0ustar rootroot fmmetal Signal Generators:FM Synthesis fmmetal fmmetal Uses FM synthesis to create a Heavy Metal sound. Description Uses FM synthesis to create a Heavy Metal sound. It comes from a family of FM sounds, all using 4 basic oscillators and various architectures, as used in the TX81Z synthesizer. Syntax ares fmmetal kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, \ ifn4, ivfn Initialization All these opcodes take 5 tables for initialization. The first 4 are the basic inputs and the last is the low frequency oscillator (LFO) used for vibrato. The last table should usually be a sine wave. The initial waves should be: ifn1 -- sine wave ifn2 -- twopeaks.aiff ifn3 -- twopeaks.aiff ifn4 -- sine wave Note The file twopeaks.aiff is also available at ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. kc1, kc2 -- Controls for the synthesizer: kc1 -- Total mod index kc2 -- Crossfade of two modulators Algorithm -- 3 kvdepth -- Vibrator depth kvrate -- Vibrator rate Examples Here is an example of the fmmetal opcode. It uses the file fmmetal.csd, and twopeaks.aiff. Example of the fmmetal opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also fmb3, fmbell, fmpercfl, fmrhode, fmwurlie Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/slider32.xml0000644000000000000000000001131012262561504016546 0ustar rootroot slider32 Real-time MIDI:Slider Banks slider32 slider32 Creates a bank of 32 different MIDI control message numbers. Description Creates a bank of 32 different MIDI control message numbers. Syntax i1,...,i32 slider32 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum32, imin32, imax32, init32, ifn32 k1,...,k32 slider32 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum32, imin32, imax32, init32, ifn32 Initialization i1 ... i32 -- output values ichan -- MIDI channel (1-16) ictlnum1 ... ictlnum32 -- MIDI control number (0-127) imin1 ... imin32 -- minimum values for each controller imax1 ... imax32 -- maximum values for each controller init1 ... init32 -- initial value for each controller ifn1 ... ifn32 -- function table for conversion for each controller Performance k1 ... k32 -- output values slider32 is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider32 allows a bank of 32 different MIDI control message numbers. As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. In the i-rate version of slider32, there is not an initial value input argument, because the output is gotten directly from current status of internal controller array of Csound. See Also s16b14, s32b14, slider16, slider16f, slider32f, slider64, slider64f, slider8, slider8f Credits Author: &namegabriel; Italy December 1998 New in Csound version 3.50 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/flscroll.xml0000644000000000000000000001136112262561504016745 0ustar rootroot FLscroll FLTK:Containers FLscroll FLscroll A FLTK opcode that adds scroll bars to an area. Description FLscroll adds scroll bars to an area. Syntax FLscroll iwidth, iheight [, ix] [, iy] Initialization iwidth -- width of widget. iheight -- height of widget. ix (optional) -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy (optional) -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). Performance Containers are useful to format the graphic appearance of the widgets. The most important container is FLpanel, that actually creates a window. It can be filled with other containers and/or valuators or other kinds of widgets. There are no k-rate arguments in containers. FLscroll adds scroll bars to an area. Normally you must set arguments iwidth and iheight equal to that of the parent window or other parent container. ix and iy are optional since they normally are set to zero. For example the following code: FLpanel "PanelPluto",400,300,100,100 FLscroll 400,300 gk1, ih1 FLslider "FLslider 1", 500, 1000, 2 ,1, -1, 300,15, 20,50 gk2, ih2 FLslider "FLslider 2", 300, 5000, 2 ,3, -1, 300,15, 20,100 gk3, ih3 FLslider "FLslider 3", 350, 1000, 2 ,5, -1, 300,15, 20,150 gk4, ih4 FLslider "FLslider 4", 250, 5000, 1 ,11,-1, 300,30, 20,200 FLscrollEnd FLpanelEnd will show scroll bars, when the main window size is reduced: FLscroll. FLscroll. Examples Here is an example of the FLscroll opcode. It uses the file FLscroll.csd. Example of the FLscroll opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLgroup, FLgroupEnd, FLpack, FLpackEnd, FLpanel, FLpanelEnd, FLscrollEnd, FLtabs, FLtabsEnd Credits Author: &namegabriel; New in version 4.22 Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/JackoFreewheel.xml0000644000000000000000000000455112262561504020006 0ustar rootroot JackoFreewheel Jacko Opcodes JackoFreewheel JackoFreewheel Turns Jack's freewheeling mode on or off. Description Turns Jack's freewheeling mode on or off. When freewheeling is on, if supported by the rest of the Jack system, Csound will run as fast as possible, which may be either faster or slower than real time. This is essential for rendering scores that are too dense for real-time performance to a soundfile, without xruns or dropouts. Syntax JackoFreewheel [ienabled] Initialization ienabled -- Turns freewheeling on (the default) or off. See Also JackoInit, JackoInfo, JackoAudioInConnect, JackoAudioOutConnect, JackoMidiInConnect, JackoMidiOutConnect, JackoOn, JackoAudioIn, JackoAudioOut, JackoMidiOut, JackoNoteOut, JackoTransport. Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/flgroup_end.xml0000644000000000000000000000152512262561504017432 0ustar rootroot FLgroup_end FLTK:Containers FLgroup_end FLgroup_end Marks the end of a group of FLTK child widgets. Description Marks the end of a group of FLTK child widgets. This is another name for FLgroupEnd provides for compatibility. See FLgroupEnd Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/outipc.xml0000644000000000000000000000740612262561504016435 0ustar rootroot outipc Real-time MIDI:Output outipc outipc Sends MIDI program change messages at i-rate Description Sends MIDI program change messages at i-rate Syntax outipc ichn, iprog, imin, imax Initialization ichn -- MIDI channel number (1-16) iprog -- program change number in floating point imin -- minimum floating point value (converted in MIDI integer value 0) imax -- maximum floating point value (converted in MIDI integer value 127 (7 bit)) Performance outipc (i-rate program change output) sends program change messages. It works only with MIDI instruments which recognize them. It can drive a different value of a parameter for each note currently active. It can scale an i-value floating-point argument according to the imin and imax values. For example, set imin = 1.0 and imax = 2.0. When the ivalue argument receives a 2.0 value, the opcode will send a 127 value to the MIDI OUT device. When the ivalue argument receives a 1.0 value, it will send a 0 value. i-rate opcodes send their message once during instrument initialization. Examples Here is an example of the outipc opcode. It uses the file outipc.csd. Example of the outipc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also outiat, outic14, outic, outipat, outipb, outkat, outkc14, outkc, outkpat, outkpb, outkpc Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/wgpluck.xml0000644000000000000000000000736112262561504016606 0ustar rootroot wgpluck Signal Generators:Waveguide Physical Modeling wgpluck wgpluck A high fidelity simulation of a plucked string. Description A high fidelity simulation of a plucked string, using interpolating delay-lines. Syntax ares wgpluck icps, iamp, kpick, iplk, idamp, ifilt, axcite Initialization icps -- frequency of plucked string iamp -- amplitude of string pluck iplk -- point along the string, where it is plucked, in the range of 0 to 1. 0 = no pluck idamp -- damping of the note. This controls the overall decay of the string. The greater the value of idamp, the faster the decay. Negative values will cause an increase in output over time. ifilt -- control the attenuation of the filter at the bridge. Higher values cause the higher harmonics to decay faster. Performance kpick -- proportion of the way along the point to sample the output. axcite -- a signal which excites the string. A string of frequency icps is plucked with amplitude iamp at point iplk. The decay of the virtual string is controlled by idamp and ifilt which simulate the bridge. The oscillation is sampled at the point kpick, and excited by the signal axcite. Examples The following example produces a moderately long note with rapidly decaying upper partials. It uses the file wgpluck.csd. An example of the wgpluck opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The following example produces a shorter, brighter note. It uses the file wgpluck_brighter.csd. An example of the wgpluck opcode with a shorter, brighter note. Credits Author: Michael A. Casey M.I.T. Cambridge, Mass. 1997 New in Version 3.47 manual_src~dfsg/opcodes/ntrpol.xml0000644000000000000000000000641112262561504016443 0ustar rootroot ntrpol Signal Modifiers:Sample Level Operators ntrpol ntrpol Calculates the weighted mean value of two input signals. Description Calculates the weighted mean value (i.e. linear interpolation) of two input signals Syntax ares ntrpol asig1, asig2, kpoint [, imin] [, imax] ires ntrpol isig1, isig2, ipoint [, imin] [, imax] kres ntrpol ksig1, ksig2, kpoint [, imin] [, imax] Initialization imin -- minimum xpoint value (optional, default 0) imax -- maximum xpoint value (optional, default 1) Performance xsig1, xsig2 -- input signals xpoint -- interpolation point between the two values ntrpol opcode outputs the linear interpolation between two input values. xpoint is the distance of evaluation point from the first value. With the default values of imin and imax, (0 and 1) a zero value indicates no distance from the first value and the maximum distance from the second one. With a 0.5 value, ntrpol will output the mean value of the two inputs, indicating the exact half point between xsig1 and xsig2. A 1 value indicates the maximum distance from the first value and no distance from the second one. The range of xpoint can be also defined with imin and imax to make its management easier. These opcodes are useful for crossfading two signals. Examples Here is an example of the ntrpol opcode. It uses the file ntrpol.csd. Example of the ntrpol opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; Italy October 1998 New in Csound version 3.49 manual_src~dfsg/opcodes/vcomb.xml0000644000000000000000000000660112262561504016234 0ustar rootroot vcomb Signal Modifiers:Reverberation vcomb vcomb Variably reverberates an input signal with a colored frequency response. Description Variably reverberates an input signal with a colored frequency response. Syntax ares vcomb asig, krvt, xlpt, imaxlpt [, iskip] [, insmps] Initialization imaxlpt -- maximum loop time for klpt iskip (optional, default=0) -- initial disposition of delay-loop data space (cf. reson). The default value is 0. insmps (optional, default=0) -- delay amount, as a number of samples. Performance krvt -- the reverberation time (defined as the time in seconds for a signal to decay to 1/1000, or 60dB down from its original amplitude). xlpt -- variable loop time in seconds, same as ilpt in comb. Loop time can be as large as imaxlpt. This filter reiterates input with an echo density determined by loop time xlpt. The attenuation rate is independent and is determined by krvt, the reverberation time (defined as the time in seconds for a signal to decay to 1/1000, or 60dB down from its original amplitude). Output will appear only after ilpt seconds. Examples Here is an example of the vcomb opcode. It uses the file vcomb.csd. Example of the vcomb opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also alpass, comb, reverb, valpass Credits Author: William Pete Moss University of Texas at Austin Austin, Texas USA January 2002 manual_src~dfsg/opcodes/undef.xml0000644000000000000000000000460212262561504016226 0ustar rootroot macrosorchestra #undeforchestra #undef Orchestra Syntax:Macros #undef #undef Un-defines a macro. Description Macros are textual replacements which are made in the orchestra as it is being read. The macro system in Csound is a very simple one, and uses the characters # and $ to define and call macros. This can save typing, and can lead to a coherent structure and consistent style. This is similar to, but independent of, the macro system in the score language. #undef NAME -- undefines a macro name. If a macro is no longer required, it can be undefined with #undef NAME. Syntax #undef NAME Performance Some care is needed with textual replacement macros, as they can sometimes do strange things. They take no notice of any meaning, so spaces are significant. This is why, unlike the C programming language, the definition has the replacement text surrounded by # characters. Used carefully, this simple macro system is a powerful concept, but it can be abused. See Also #define, $NAME Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April 1998 New in Csound version 3.48 manual_src~dfsg/opcodes/vaget.xml0000644000000000000000000000507312262561504016236 0ustar rootroot vaget Signal Modifiers:Sample Level Operators vaget vaget Access values of the current buffer of an a-rate variable by indexing. Description Access values of the current buffer of an a-rate variable by indexing. Useful for doing sample-by-sample manipulation at k-rate without using setksmps 1. Note Because this opcode does not do any bounds checking, the user must be careful not to try to read values past ksmps (the size of a buffer for an a-rate variable) by using index values greater than ksmps. Note In Csound6 this opcode does do bounds checking. Syntax kval vaget kndx, avar Performance kval - value read from avar kndx - index of the sample to read from the current buffer of the given avar variable avar - a-rate variable to read from Examples Here is an example of the vaget opcode. It uses the file vaget.csd. Example of the vaget opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vaset Credits Author: Steven Yi New in version 5.04 September 2006. manual_src~dfsg/opcodes/islider64.xml0000644000000000000000000000100512262561504016724 0ustar rootroot islider64 islider64 islider64 Deprecated. Description Deprecated as of version 3.52. Use the slider64 opcode instead. manual_src~dfsg/opcodes/outic14.xml0000644000000000000000000000736112262561504016422 0ustar rootroot outic14 Real-time MIDI:Output outic14 outic14 Sends 14-bit MIDI controller output at i-rate. Description Sends 14-bit MIDI controller output at i-rate. Syntax outic14 ichn, imsb, ilsb, ivalue, imin, imax Initialization ichn -- MIDI channel number (1-16) imsb -- most significant byte controller number when using 14-bit parameters (0-127) ilsb -- least significant byte controller number when using 14-bit parameters (0-127) ivalue -- floating point value imin -- minimum floating point value (converted in MIDI integer value 0) imax -- maximum floating point value (converted in MIDI integer value 16383 (14-bit)) Performance outic14 (i-rate MIDI 14-bit controller output) sends a pair of controller messages. This opcode can drive 14-bit parameters on MIDI instruments that recognize them. The first control message contains the most significant byte of ivalue argument while the second message contains the less significant byte. imsb and ilsb are the number of the most and less significant controller. This opcode can drive a different value of a parameter for each note currently active. It can scale an i-value floating-point argument according to the imin and imax values. For example, set imin = 1.0 and imax = 2.0. When the ivalue argument receives a 2.0 value, the opcode will send a 127 value to the MIDI OUT device. When the ivalue argument receives a 1.0 value, it will send a 0 value. i-rate opcodes send their message once during instrument initialization. See Also outiat, outic, outipat, outipb, outipc, outkat, outkc14, outkc, outkpat, outkpb, outkpc Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/jitter.xml0000644000000000000000000000530112262561504016423 0ustar rootroot jitter Signal Generators:Random (Noise) Generators jitter jitter Generates a segmented line whose segments are randomly generated. Description Generates a segmented line whose segments are randomly generated. Syntax kout jitter kamp, kcpsMin, kcpsMax Performance kamp -- Amplitude of jitter deviation kcpsMin -- Minimum speed of random frequency variations (expressed in cps) kcpsMax -- Maximum speed of random frequency variations (expressed in cps) jitter generates a segmented line whose segments are randomly generated inside the +kamp and -kamp interval. Duration of each segment is a random value generated according to kcpsmin and kcpsmax values. jitter can be used to make more natural and analog-sounding some static, dull sound. For best results, it is suggested to keep its amplitude moderate. Examples Here is an example of the jitter opcode. It uses the file jitter.csd. Example of the jitter opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also jitter2, vibr, vibrato Credits Author: &namegabriel; New in Version 4.15 manual_src~dfsg/opcodes/pset.xml0000644000000000000000000000503312262561504016077 0ustar rootroot pset Instrument Control:Initialization and Reinitialization pset pset Defines and initializes numeric arrays at orchestra load time. Description Defines and initializes numeric arrays at orchestra load time. Syntax pset icon1 [, icon2] [...] Initialization icon1, icon2, ... -- preset values for a MIDI instrument pset (optional) defines and initializes numeric arrays at orchestra load time. It may be used as an orchestra header statement (i.e. instrument 0) or within an instrument. When defined within an instrument, it is not part of its i-time or performance operation, and only one statement is allowed per instrument. These values are available as i-time defaults. When an instrument is triggered from MIDI it only gets p1 and p2 from the event, and p3, p4, etc. will receive the actual preset values. Examples Here is an example of the pset opcode. It uses the file pset.csd Example of the pset opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the pset opcode, using pset with midi. It uses the file pset-midi.csd Second example of the pset opcode. See Also strset manual_src~dfsg/opcodes/exp.xml0000644000000000000000000000550712262561504015726 0ustar rootroot exp Mathematical Operations:Mathematical Functions exp exp Returns e raised to the x-th power. Description Returns e raised to the xth power. Syntax exp(x) (no rate restriction) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Examples Here is an example of the exp opcode. It uses the file exp.csd. Example of the exp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: Generated events per second: 1 rtevent: T 0.000 TT 0.000 M: 0.00000 0.00000 new alloc for instr 10: rtevent: T 0.811 TT 0.811 M: 0.48906 0.48906 new alloc for instr 10: Generated events per second: 2 rtevent: T 1.387 TT 1.387 M: 0.48611 0.48611 rtevent: T 1.833 TT 1.833 M: 0.48421 0.48421 Generated events per second: 3 rtevent: T 2.198 TT 2.198 M: 0.47536 0.47536 rtevent: T 2.506 TT 2.506 M: 0.46530 0.46530 rtevent: T 2.773 TT 2.773 M: 0.44986 0.44986 Generated events per second: 4 rtevent: T 3.009 TT 3.009 M: 0.48096 0.48096 ......... See Also abs, frac, int, log, log10, i, sqrt New in version 4.21 manual_src~dfsg/opcodes/vbapg.xml0000644000000000000000000001315412262561504016226 0ustar rootroot vbapg Signal Modifiers:Panning and Spatialization vbapg vbapg Calculates the gains for a sound location between multiple channels. Description Calculates the gains for a sound location for up to 64. Syntax k1[, k2...] vbapg kazim [,kelev] [, kspread] [, ilayout] karray[] vbapg kazim [,kelev] [, kspread] [, ilayout] Initialization ilayout -- index of the speaker layout in the range 0-99, corresponding to a call to vbaplsinit. The default value is 0. Performance kazim -- azimuth angle of the virtual source kelev (optional) -- elevation angle of the virtual source kspread (optional) -- spreading of the virtual source (range 0 - 100). If value is zero, conventional amplitude panning is used. When kspread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. vbapg calculates the gains that an input signal would have between multiple speakers according to the controls kazim and kelev, and the configured loudspeaker placement ilayout. If idim = 2, kelev is set to zero. The distribution is performed using Vector Base Amplitude Panning (VBAP - See reference). VBAP distributes the signal using loudspeaker data configured with vbaplsinit. The signal is applied to, at most, two loudspeakers in 2-D loudspeaker configurations, and three loudspeakers in 3-D loudspeaker configurations. If the virtual source is panned outside the region spanned by loudspeakers, the nearest loudspeakers are used in panning. Please note that all vbap panning opcodes require the vbap system to be initialized using vbaplsinit. Examples Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. Examples Here is an example of the vbapg opcode. It uses the file vbapg.csd. Example of the vbapg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: spread of source = 0 spread of source = 8 spread of source = 17 spread of source = 25 spread of source = 33 spread of source = 42 spread of source = 50 spread of source = 58 spread of source = 67 spread of source = 75 spread of source = 83 spread of source = 92 spread of source = 100 See Also vbap, vbapmove, vbap4, vbap4move, vbap8, vbap8move, vbap16, vbap16move, vbaplsinit, vbapz, vbapzmove Credits Author: Ville Pulkki Sibelius Academy Computer Music Studio Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Helsinki, Finland May 2000 Author: &namejohn; July 2012, Sept 2013 New in Csound Version 5.17.13. Array version new in Csound Version 6.01. manual_src~dfsg/opcodes/fluidSetInterpMethod.xml0000644000000000000000000000675012262561504021235 0ustar rootroot fluidSetInterpMethod Signal Generators:Sample Playback fluidSetInterpMethod fluidSetInterpMethod Set interpolation method for channel in Fluid Engine Syntax fluidSetInterpMethod ienginenum, ichannelnum, iInterpMethod Description Set interpolation method for channel in Fluid Engine. Lower order interpolation methods will render faster at lower fidelity while higher order interpolation methods will render slower at higher fidelity. Default interpolation for a channel is 4th order interpolation. Initialization ienginenum -- engine number assigned from fluidEngine ichannelnum -- which channel number to use for the preset in the given fluidEngine iInterpMethod -- interpolation method, can be any of the following 0 -- No Interpolation 1 -- Linear Interpolation 4 -- 4th Order Interpolation (Default) 7 -- 7th Order Interpolation (Highest) Examples Here is an example of the fluidSetInterpMethod opcode. It uses the file fluidSetInterpMethod.csd and 07AcousticGuitar.sf2. Example of the fluidSetInterpMethod opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also fluidEngine More information on soundfonts in the Floss Manuals: http://en.flossmanuals.net/csound/ch048_d-reading-midi-files Other information on soundfonts on Wikipedia: http://en.wikipedia.org/wiki/Soundfont Credits Author: &namesteven; New in version 5.07 manual_src~dfsg/opcodes/lpread.xml0000644000000000000000000001533712262561504016403 0ustar rootroot lpread Spectral Processing:LPC lpread lpread Reads a control file of time-ordered information frames. Description Reads a control file of time-ordered information frames. Syntax krmsr, krmso, kerr, kcps lpread ktimpnt, ifilcod [, inpoles] [, ifrmrate] Initialization ifilcod -- integer or character-string denoting a control-file (reflection coefficients and four parameter values) derived from n-pole linear predictive spectral analysis of a source audio signal. An integer denotes the suffix of a file lp.m; a character-string (in double quotes) gives a filename, optionally a full pathname. If not fullpath, the file is sought first in the current directory, then in that of the environment variable SADIR (if defined). Memory usage depends on the size of the file, which is held entirely in memory during computation but shared by multiple calls (see also adsyn, pvoc). inpoles (optional, default=0) -- number of poles in the lpc analysis. It is required only when the control file does not have a header; it is ignored when a header is detected. ifrmrate (optional, default=0) -- frame rate per second in the lpc analysis. It is required only when the control file does not have a header; it is ignored when a header is detected. Performance lpread accesses a control file of time-ordered information frames, each containing n-pole filter coefficients derived from linear predictive analysis of a source signal at fixed time intervals (e.g. 1/100 of a second), plus four parameter values: krmsr -- root-mean-square (rms) of the residual of analysis krmso -- rms of the original signal kerr -- the normalized error signal kcps -- pitch in Hz ktimpnt -- The passage of time, in seconds, through the analysis file. ktimpnt must always be positive, but can move forwards or backwards in time, be stationary or discontinuous, as a pointer into the analysis file. lpread gets its values from the control file according to the input value ktimpnt (in seconds). If ktimpnt proceeds at the analysis rate, time-normal synthesis will result; proceeding at a faster, slower, or variable rate will result in time-warped synthesis. At each k-period, lpread interpolates between adjacent frames to more accurately determine the parameter values (presented as output) and the filter coefficient settings (passed internally to a subsequent lpreson). The error signal kerr (between 0 and 1) derived during predictive analysis reflects the deterministic/random nature of the analyzed source. This will emerge low for pitched (periodic) material and higher for noisy material. The transition from voiced to unvoiced speech, for example, produces an error signal value of about .3. During synthesis, the error signal value can be used to determine the nature of the lpreson driving function: for example, by arbitrating between pitched and non-pitched input, or even by determining a mix of the two. In normal speech resynthesis, the pitched input to lpreson is a wideband periodic signal or pulse train derived from a unit such as buzz, and the nonpitched source is usually derived from rand. However, any audio signal can be used as the driving function, the only assumption of the analysis being that it has a flat response. lpfreson is a formant shifted lpreson, in which kfrqratio is the (cps) ratio of shifted to original formant positions. This permits synthesis in which the source object changes its apparent acoustic size. lpfreson with kfrqratio = 1 is equivalent to lpreson. Generally, lpreson provides a means whereby the time-varying content and spectral shaping of a composite audio signal can be controlled by the dynamic spectral content of another. There can be any number of lpread/lpreson (or lpfreson) pairs in an instrument or in an orchestra; they can read from the same or different control files independently. Examples Here is an example of the lpread opcode. It uses the file lpread.csd. Example of the lpread opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The audio file fox.wav is 2.8 seconds long. So filelen's output should be a line like this: fox.wav = 2.756667 seconds See Also lpfreson, lpreson, LPANAL manual_src~dfsg/opcodes/STKWurley.xml0000644000000000000000000000735212262561504017003 0ustar rootroot STKWurley Signal Generators:STK Opcodes STKWurley STKWurley STKWurley simulates a Wurlitzer electric piano FM synthesis instrument. Description STKWurley simulates a Wurlitzer electric piano FM synthesis instrument. It uses two simple FM Pairs summed together, also referred to as algorithm 5 of the TX81Z. Syntax asignal STKWurley ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kmod -- controller 2, modulator index 1. Value range of kv1 is 0-127. kcross -- controller 4, crossfade of outputs. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kadsr -- controller 128, ADSR 2 and 4 target. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the Wurley class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKWurley opcode. It uses the file STKWurley.csd. Example of the STKWurley opcode. See Also STKRhodey. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/until.xml0000644000000000000000000000504112262561504016256 0ustar rootroot until Instrument Control:Program Flow Control until until A syntactic looping construction. Description A syntactic looping construction. Syntax until condition do ... od Performance The statements between the do and od form the body of a loop which is obeyed until the conditional becomes true. Examples Here is an example of the until construction. It uses the file until.csd. Example of the until opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: B 0.000 .. 1.000 T 1.000 TT 1.000 M: 0.0 new alloc for instr 1: instr 1: p4 = 3.000 instr 1: p4 = 2.000 instr 1: p4 = 1.000 instr 1: p4 = 0.000 instr 1: p4 = -1.000 B 1.000 .. 2.000 T 2.000 TT 2.000 M: 0.0 new alloc for instr 2: instr 2: p4 = 3.000 instr 2: p4 = 2.000 instr 2: p4 = 1.000 instr 2: p4 = 0.000 instr 2: p4 = -1.000 B 2.000 .. 3.000 T 3.000 TT 3.000 M: 0.0 See Also loop_ge, loop_gt and loop_le. loop_lt. Credits &namejohn;. New in Csound version 5.14 with new parser manual_src~dfsg/opcodes/pvsgain.xml0000644000000000000000000000466012262561504016600 0ustar rootroot pvsgain Spectral Processing:Streaming pvsgain pvsgain Scale the amplitude of a pv stream. Description Scale the amplitude of a pv stream. Syntax fsig pvsgain fsigin, kgain Performance fsig -- output pv stream fsigin -- input pv stream kgain -- amplitude scaling (defaults to 1). It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the use of the pvsgain opcode. It uses the file pvsgain.csd. Example of the <emphasis>pvsgain</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namevictor; 2011 New plugin in version 5 2011 manual_src~dfsg/opcodes/tan.xml0000644000000000000000000000434212262561504015710 0ustar rootroot tan Mathematical Operations:Trigonometric Functions tan tan Performs a tangent function. Description Returns the tangent of x (x in radians). Syntax tan(x) (no rate restriction) Examples Here is an example of the tan opcode. It uses the file tan.csd. Example of the tan opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: i1 = -0.134 See Also cos, cosh, cosinv, sin, sinh, sininv, tan, taninv Credits Written by &namejohn;. New in version 3.47 Example written by &namekevin;. manual_src~dfsg/opcodes/ckgoto.xml0000644000000000000000000000537012262561504016416 0ustar rootroot ckgoto Instrument Control:Program Flow Control ckgoto ckgoto Conditionally transfer control during the p-time passes. Description During the p-time passes only, conditionally transfer control to the statement labeled by label. Syntax ckgoto condition, label where label is in the same instrument block and is not an expression, and where condition uses one of the Relational operators (<, =, <=, ==, !=) (and = for convenience, see also under Conditional Values). Examples Here is an example of the ckgoto opcode. It uses the file ckgoto.csd. Example of the ckgoto opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: kval = 0.000000, kfreq = 440.000000 kval = 0.999732, kfreq = 440.000000 kval = 1.999639, kfreq = 880.000000 See Also cggoto, cigoto, cngoto, goto, if, igoto, tigoto, timout Credits Example written by &namekevin;. manual_src~dfsg/opcodes/inletf.xml0000644000000000000000000000456312262561504016414 0ustar rootroot inletf Signal Flow Graph Opcodes inletf inletf Receives an frate signal (fsig) into an instrument from a named port. Description Receives an frate signal (fsig) into an instrument from a named port. Syntax fsignal inletf Sname Initialization Sname -- String name of the inlet port. The name of the inlet is implicitly qualified by the instrument name or number, so it is valid to use the same inlet name in more than one instrument (but not to use the same inlet name twice in one instrument). Performance ksignal -- frate input signal During performance, the frate inlet signal is received from each instance of an instrument containing an outlet port to which this inlet has been connected using the connect opcode. The signals of all the outlets connected to an inlet are combined in the inlet. See Also outleta outletk outletkid outletf inleta inletk inletkid connect alwayson ftgenonce Credits By: &namemichael; 2009 manual_src~dfsg/opcodes/barmodel.xml0000644000000000000000000000667312262561504016724 0ustar rootroot barmodel Signal Generators:Models and Emulations barmodel barmodel Creates a tone similar to a struck metal bar. Description Audio output is a tone similar to a struck metal bar, using a physical model developed from solving the partial differential equation. There are controls over the boundary conditions as well as the bar characteristics. Syntax ares barmodel kbcL, kbcR, iK, ib, kscan, iT30, ipos, ivel, iwid Initialization iK -- dimensionless stiffness parameter. If this parameter is negative then the initialisation is skipped and the previous state of the bar is continued. ib -- high-frequency loss parameter (keep this small). iT30 -- 30 db decay time in seconds. ipos -- position along the bar that the strike occurs. ivel -- normalized strike velocity. iwid -- spatial width of strike. Performance A note is played on a metalic bar, with the arguments as below. kbcL -- Boundary condition at left end of bar (1 is clamped, 2 pivoting and 3 free). kbcR -- Boundary condition at right end of bar (1 is clamped, 2 pivoting and 3 free). kscan -- Speed of scanning the output location. Note that changing the boundary conditions during playing may lead to glitches and is made available as an experiment. The use of a non-zero kscan can give apparent re-introduction of sound due to modulation. Examples Here is an example of the barmodel opcode. It uses the file barmodel.csd. Example of the barmodel opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Stefan Bilbao University of Edinburgh, UK Author: &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 5.01 manual_src~dfsg/opcodes/koutpb.xml0000644000000000000000000000076212262561504016434 0ustar rootroot koutpb koutpb koutpb Deprecated. Description Deprecated as of version 3.52. Use the outkpb opcode instead. manual_src~dfsg/opcodes/deltap3.xml0000644000000000000000000001547412262561504016472 0ustar rootroot deltape Signal Modifiers:Delay deltap3 deltap Taps a delay line at variable offset times, uses cubic interpolation. Description Taps a delay line at variable offset times, uses cubic interpolation. Syntax ares deltap3 xdlt Performance xdlt -- specifies the tapped delay time in seconds. Each can range from 1 control period to the full delay time of the read/write pair; however, since there is no internal check for adherence to this range, the user is wholly responsible. Each argument can be a constant, a variable, or a time-varying signal; the xdlt argument in deltap3 implies that an audio-varying delay is permitted there. deltap3 is experimental, and uses cubic interpolation. (New in Csound version 3.50.) This opcode can tap into a delayr/delayw pair, extracting delayed audio from the idlt seconds of stored sound. There can be any number of deltap and/or deltapi units between a read/write pair. Each receives an audio tap with no change of original amplitude. This opcode can provide multiple delay taps for arbitrary delay path and feedback networks. They can deliver either constant-time or time-varying taps, and are useful for building chorus effects, harmonizers, and Doppler shifts. Constant-time delay taps (and some slowly changing ones) do not need interpolated readout; they are well served by deltap. Medium-paced or fast varying dlt's, however, will need the extra services of deltapi. delayr/delayw pairs may be interleaved. To associate a delay tap unit with a specific delayr unit, it not only has to be located between that delayr and the appropriate delayw unit, but must also precede any following delayr units. See Example 2. (This feature added in Csound version 3.57 by Jens Groh and &namejohn;). N.B. k-rate delay times are not internally interpolated, but rather lay down stepped time-shifts of audio samples; this will be found quite adequate for slowly changing tap times. For medium to fast-paced changes, however, one should provide a higher resolution audio-rate timeshift as input. Examples deltap example #1 asource buzz 1, 440, 20, 1 atime linseg 1, p3/2,.01, p3/2,1 ; trace a distance in secs ampfac = 1/atime/atime ; and calc an amp factor adump delayr 1 ; set maximum distance amove deltapi atime ; move sound source past delayw asource ; the listener out amove * ampfac deltap example #2 ainput1 = ..... ainput2 = ..... kdlyt1 = ..... kdlyt2 = ..... ;Read delayed signal, first delayr instance: adump delayr 4.0 adly1 deltap kdlyt1 ; associated with first delayr instance ;Read delayed signal, second delayr instance: adump delayr 4.0 adly2 deltap kdlyt2 ; associated with second delayr instance ;Do some cross-coupled manipulation: afdbk1 = 0.7 * adly1 + 0.7 * adly2 + ainput1 afdbk2 = -0.7 * adly1 + 0.7 * adly2 + ainput2 ;Feed back signal, associated with first delayr instance: delayw afdbk1 ;Feed back signal, associated with second delayr instance: delayw afdbk2 outs adly1, adly2 Here is yet another example of the deltap3 opcode. It uses the file deltap3.csd. Example of the deltap3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also deltap, deltapi, deltapn manual_src~dfsg/opcodes/flsetcolor.xml0000644000000000000000000000760412262561504017306 0ustar rootroot FLsetColor FLTK:Appearance FLsetColor FLsetColor Sets the primary color of a FLTK widget. Description FLsetColor sets the primary color of the target widget. Syntax FLsetColor ired, igreen, iblue, ihandle Initialization ired -- The red color of the target widget. The range for each RGB component is 0-255 igreen -- The green color of the target widget. The range for each RGB component is 0-255 iblue -- The blue color of the target widget. The range for each RGB component is 0-255 ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. Examples Here is an example of the FLsetcolor opcode. It uses the file FLsetcolor.csd. Example of the FLsetcolor opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/fluidProgramSelect.xml0000644000000000000000000000572712262561504020731 0ustar rootroot fluidProgramSelect Signal Generators:Sample Playback fluidProgramSelect fluidProgramSelect Assigns a preset from a SoundFont to a channel on a fluidEngine. Syntax fluidProgramSelect ienginenum, ichannelnum, isfnum, ibanknum, ipresetnum Description Assigns a preset from a SoundFont to a channel on a fluidEngine. Initialization ienginenum -- engine number assigned from fluidEngine ichannelnum -- which channel number to use for the preset in the given fluidEngine isfnum -- number of the SoundFont from which the preset is assigned ibanknum -- number of the bank in the SoundFont from which the preset is assigned ipresetnum -- number of the preset to assign Examples Here is an example of the fluidProgramSelect opcode. It uses the file fluidProgramSelect.csd. Here is another more complex example of the fluidsynth opcodes written by Istvan Varga. It uses the file fluidcomplex.csd. See Also fluidEngine, fluidNote, fluidLoad More information on soundfonts in the Floss Manuals: http://en.flossmanuals.net/csound/ch048_d-reading-midi-files Other information on soundfonts on Wikipedia: http://en.wikipedia.org/wiki/Soundfont Credits Michael Gogins (gogins at pipeline dot com), Steven Yi. Thanks to Peter Hanappe for Fluidsynth. manual_src~dfsg/opcodes/STKBrass.xml0000644000000000000000000000665412262561504016572 0ustar rootroot STKBrass Signal Generators:STK Opcodes STKBrass STKBrass STKBrass is a simple brass instrument. Description STKBrass uses a simple brass instrument waveguide model, a la Cook. Syntax asignal STKBrass ifrequency, iamplitude, [klip, kv1[, kslide, kv2[, klfo, kv3[, klfodepth, kv4[, kvol, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance klip -- controller 2, lip tension. Value range of kv1 is 0-127. kslide -- controller 4, slide length. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kvol -- controller 128, volume. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the Brass class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKBrass opcode. It uses the file STKBrass.csd. Example of the STKBrass opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/flooper2.xml0000644000000000000000000001034212262561504016653 0ustar rootroot flooper2 Signal Generators:Sample Playback flooper2 flooper2 Function-table-based crossfading looper. Description This opcode implements a crossfading looper with variable loop parameters and three looping modes, optionally using a table for its crossfade shape. It accepts non-power-of-two tables for its source sounds, such as deferred-allocation GEN01 tables. Syntax asig flooper2 kamp, kpitch, kloopstart, kloopend, kcrossfade, ifn \ [, istart, imode, ifenv, iskip] Initialization ifn -- sound source function table number, generally created using GEN01 istart -- playback start pos in seconds imode -- loop modes: 0 forward, 1 backward, 2 back-and-forth [def: 0] ifenv -- if non-zero, crossfade envelope shape table number. The default, 0, sets the crossfade to linear. iskip -- if 1, the opcode initialisation is skipped, for tied notes, performance continues from the position in the loop where the previous note stopped. The default, 0, does not skip initialisation Performance asig -- output sig kamp -- amplitude control kpitch -- pitch control (transposition ratio); negative values are not allowed. kloopstart -- loop start point (secs). Note that although k-rate, loop parameters such as this are only updated once per loop cycle. kloopend -- loop end point (secs), updated once per loop cycle. kcrossfade -- crossfade length (secs), updated once per loop cycle and limited to loop length. Mode 1 for imode will only loop backwards from the end point to the start point. Examples aout flooper2 16000, 1, 1, 5, 0.05, 1 ; loop starts at 1 sec, for 4 secs, 0.05 crossfade out aout The example above shows the basic operation of flooper2. Pitch can be controlled at the k-rate, as well as amplitude and loop parameters. The example assumes table 1 to contain at least 5.05 seconds of audio (4 secs loop duration, starting 1 sec into the table, using 0.05 secs after the loop end for the crossfade). Looping is in mode 0 (normal forward loop). Here is another example of the flooper2 opcode. It uses the file flooper2.csd and fox.wav. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; July 2006 New plugin in version 5 July 2006. manual_src~dfsg/opcodes/cell.xml0000644000000000000000000000707112262561504016047 0ustar rootroot cell Vectorial:Cellular Automata cell cell Cellular Automaton Description One-Dimensional Cellular Automaton. This opcode is the modified version of vcella by Gabriel Maldonado. Syntax cell ktrig, kreinit, ioutFunc, initStateFunc, iRuleFunc, ielements Initialization ioutFunc -- number of the table where the state of each cell is stored. initStateFunc -- number of the table containing the inital states of cells. iRuleFunc -- number of a lookup table containing the 8-bit rule. ielements -- total number of cells in a row. Performance ktri -- trigger signal. Each time it is non-zero, a new generation of cells is evaluated. kreinit -- reset signal. Each time it is non-zero, state of all cells is forced to be that of initStateFunc. cell models a classical 1D cellular automaton and stores the state of each cell in the table identified by ioutFunc. initStateFunc is an input vector containing the inital value of the row of cells, while iRuleFunc is an input vector containing the chosen rule in the binary form (least significant bit first). A new generation of cells is evaluated each time ktrig contains a non-zero value. Also the status of all cells can be forced to assume the status corresponding to the contents of initStateFunc each time kreinit contains a non-zero value. Note that each cell is supposed to be in one of two possible states (1 = "alive", 0 = "dead"), although fractional values should work too, because truncation is used. Examples Here is a simple example of the cell opcode. It uses the file cell.csd. A simple example of the cell opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Gleb Rogozinsky October 2011 New in Csound version 5.16.6 manual_src~dfsg/opcodes/tabsum.xml0000644000000000000000000000425412262561504016423 0ustar rootroot tabsum Table Control:Table Queries tabsum tabsum Adding values in a range of a table. Description Sums the values in an f-table in a consecutive range. Syntax kr tabsum ifn[[, kmin] [, kmax]] Initialization ifn -- table number Performance kr -- input signal to write. kmin, kmax -- range of the table to sum. If omitted or zero they default to 0 to length of the table. Examples Here is an example of the tabsum opcode. It uses the file tabsum.csd. Example of the tabsum opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also Vectorial opcodes Credits Author: &namejohn; Codemist Ltd 2009 New in version 5.11 manual_src~dfsg/opcodes/flshow.xml0000644000000000000000000000520712262561504016431 0ustar rootroot FLshow FLTK:Appearance FLshow FLshow Restores the visibility of a previously hidden FLTK widget. Description FLshow restores the visibility of a previously hidden widget. Syntax FLshow ihandle Initialization ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/release.xml0000644000000000000000000000500712262561504016545 0ustar rootroot release Real-time MIDI:Event Extenders release release Indicates whether a note is in its release stage. Description Provides a way of knowing when a note off message for the current note is received. Only a noteoff message with the same MIDI note number as the one which triggered the note will be reported by release. Syntax kflag release Performance kflag -- indicates whether the note is in its release stage. (1 if a note off is received, otherwise 0) release outputs current note state. If current note is in the release stage (i.e. if its duration has been extended with xtratim opcode and if it has only just deactivated), then the kflag output argument is set to 1. Otherwise (in sustain stage of current note), kflag is set to 0. This opcode is useful for implementing complex release-oriented envelopes. When used in conjunction with xtratim it can provide an alternative to the hard-coded behaviour of the "r" opcodes (linsegr, expsegr et al), where release time is set to the longest time specified in the active instrument. Examples See the examples for xtratim. See Also xtratim Credits Author: &namegabriel; Italy New in Csound version 3.47 manual_src~dfsg/opcodes/vlowres.xml0000644000000000000000000000542012262561504016625 0ustar rootroot vlowres Signal Modifiers:Standard Filters:Resonant vlowres vlowres A bank of filters in which the cutoff frequency can be separated under user control. Description A bank of filters in which the cutoff frequency can be separated under user control Syntax ares vlowres asig, kfco, kres, iord, ksep Initialization iord -- total number of filters (1 to 10) Performance asig -- input signal kfco -- frequency cutoff (not in Hz) kres -- resonance amount ksep -- frequency cutoff separation for each filter: the first filter has a kfreq cutoff, the second has a kfreq + ksep and the third kfreq + 2*ksep and so on, depending on the number of filters. vlowres (variable resonant lowpass filter) allows a variable response curve in resonant filters. It can be thought of as a bank of lowpass resonant filters, each with the same resonance, serially connected. The frequency cutoff of each filter can vary with the kcfo and ksep parameters. Examples Here is an example of the vlowres opcode. It uses the file vlowres.csd. Example of the vlowres opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; Italy New in Csound version 3.49 manual_src~dfsg/opcodes/trfilter.xml0000644000000000000000000000542412262561504016763 0ustar rootroot trfilter Spectral Processing:Streaming trfilter trfilter Streaming partial track filtering. Description The trfilter opcode takes an input containg a TRACKS pv streaming signal (as generated, for instance by partials) and filters it using an amplitude response curve stored in a function table. The function table can have any size (no restriction to powers-of-two). The table lookup is done by linear-interpolation. It is possible to create time-varying filter curves by updating the amlitude response table with a table-writing opcode. Syntax fsig trfilter fin, kamnt, ifn Performance fsig -- output pv stream in TRACKS format fin -- input pv stream in TRACKS format kamnt -- amount of filtering (0-1) ifn -- function table number. This will contain an amplitude response curve, from 0 Hz to the Nyquist (table indexes 0 to N). Any size is allowed. Larger tables will provide a smoother amplitude response curve. Table reading uses linear interpolation. Examples Here is an example of the trfilter opcode. It uses the file trfilter.csd. Example of the trfilter opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of an ifd-analysis signal and linear additive resynthesis with low-pass filtering. Credits Author: Victor Lazzarini; February 2006 New in Csound 5.01 manual_src~dfsg/opcodes/chani.xml0000644000000000000000000000475712262561504016222 0ustar rootroot chani Signal I/O:Software Bus chani chani Reads data from the software bus Description Reads data from a channel of the inward software bus. Syntax kval chani kchan aval chani kchan Performance kchan -- a positive integer that indicates which channel of the software bus to read Note that the inward and outward software busses are independent, and are not mixer buses. Also the k-rate and a-rate busses are independent. The last value remains until a new value is written. There is no imposed limit to the number of busses but they use memory so small numbers are to be preferred. Example The example shows the software bus being used as an asynchronous control signal to select a filter cutoff. It assumes that an external program that has access to the API is feeding the values sr = 44100 kr = 100 ksmps = 1 instr 1 kc chani 1 a1 oscil p4, p5, 100 a2 lowpass2 a1, kc, 200 out a2 endin Credits Author: &namejohn; 2005 New in Csound 5.00 manual_src~dfsg/opcodes/OSCinit.xml0000644000000000000000000000576312262561504016446 0ustar rootroot OSCinit OSC OSCinit OSCinit Start a listening process for OSC messages to a particular port. Description Starts a listening process, which can be used by OSClisten. Syntax ihandle OSCinit iport Initialization ihandle -- handle returned that can be passed to any number of OSClisten opcodes to receive messages on this port. iport -- the port on which to listen. Performance The listener runs in the background. See OSClisten for details. Example The example shows a pair of floating point numbers being received on port 7770. sr = 44100 ksmps = 100 nchnls = 2 gihandle OSCinit 7770 instr 1 kf1 init 0 kf2 init 0 nxtmsg: kk OSClisten gihandle, "/foo/bar", "ff", kf1, kf2 if (kk == 0) goto ex printk 0,kf1 printk 0,kf2 kgoto nxtmsg ex: endin See Also OSClisten, OSCsend More information on this opcode: http://www.youtube.com/watch?v=JX1C3TqP_9Y , made by Andrés Cabrera Credits Author: &namejohn; 2005 manual_src~dfsg/opcodes/syncphasor.xml0000644000000000000000000001140412262561504017314 0ustar rootroot syncphasor Signal Generators:Phasors syncphasor syncphasor Produces a normalized moving phase value with sync input and output. Description Produces a moving phase value between zero and one and an extra impulse output ("sync out") whenever its phase value crosses or is reset to zero. The phase can be reset at any time by an impulse on the "sync in" parameter. Syntax aphase, asyncout syncphasor xcps, asyncin, [, iphs] Initialization iphs (optional) -- initial phase, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is zero. Performance aphase -- the output phase value; always between 0 and 1. asyncout -- the sync output has a value of 1.0 for one sample whenever the phase value crosses zero or whenever the sync input is non-zero. It is zero at all other times. asyncin -- the sync input causes the phase to reset to zero whenever asyncin is non-zero. xcps -- frequency of the phasor in cycles-per-second. If xcps is negative, the phase value will decrease from 1 to 0 instead of increasing. An internal phase is successively accumulated in accordance with the xcps frequency to produce a moving phase value, normalized to lie in the range 0 <= phs < 1. When used as the index to a table unit, this phase (multiplied by the desired function table length) will cause it to behave like an oscillator. The phase of syncphasor though can be synced to another phasor (or other signal) using the asyncin parameter. Any time that asyncin is a non-zero value, the value of aphase will be reset to zero. syncphasor also outputs its own "sync" signal that consists of a one-sample impulse whenever its phase crosses zero or is reset. This makes it easy to chain together multiple syncphasor opcodes to create an oscillator "hard sync" effect. Examples Here is an example of the syncphasor opcode. It uses the file syncphasor.csd. Example of the syncphasor opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the syncphasor opcode. It uses the file syncphasor-CZresonance.csd. Another example of the syncphasor opcode. See also phasor. And the Table Access opcodes like: table, tablei, table3 and tab. Credits Adapted from the phasor opcode by &nameanthony; January 2008 New in Csound version 5.08 manual_src~dfsg/opcodes/push.xml0000644000000000000000000001037712262561504016112 0ustar rootroot push Instrument Control:Stacks push push Pushes a value into the global stack. Description Pushes a value into the global stack. Syntax push xval1, [xval2, ... , xval31] push ival1, [ival2, ... , ival31] Initialization ival1 ... ival31 - values to be pushed into the stack. Performance xval1 ... xval31 - values to be pushed into the stack. The given values are pushed into the global stack as a bundle. The global stack works in LIFO order: after multiple push calls, pop should be used in reverse order. Each push or pop operation can work on a "bundle" of multiple variables. When using pop, the number, type, and order of items must match those used by the corresponding push. That is, after a 'push Sfoo, ibar', you must call something like 'Sbar, ifoo pop', and not e.g. two separate 'pop' statements. push and pop opcodes can take variables of any type (i-, k-, a- and strings). Use of any combination of i, k, a, and S types is allowed. Variables of type 'a' and 'k' are passed at performance time only, while 'i' and 'S' are passed at init time only. push/pop for a, k, i, and S types copy data by value. By contrast, push_f only pushes a "reference" to the f-signal, and then the corresponding pop_f will copy directly from the original variable to its output signal. For this reason, changing the source f-signal of push_f before pop_f is called is not recommended, and if the instrument instance owning the variable that was passed by push_f is deactivated before pop_f is called, undefined behavior may occur. Any stack errors (trying to push when there is no more space, or pop from an empty stack, inconsistent number or type of arguments, etc.) are fatal and terminate performance. Examples Here is an example of the push opcode. It uses the file push.csd. Example of the push opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also stack, pop, pop_f and push_f. Using this opcode is somewhat hackish, as you can read here: Credits By: &nameistvan;. 2006 manual_src~dfsg/opcodes/gainslider.xml0000644000000000000000000000501512262561504017245 0ustar rootroot gainslider Signal Generators:Linear and Exponential Generators gainslider gainslider An implementation of a logarithmic gain curve which is similar to the gainslider~ object from Cycling 74 Max / MSP. Description This opcode is intended for use to multiply by an audio signal to give a console mixer like feel. There is no bounds in the source code so you can for example give higher than 127 values for extra amplitude but possibly clipped audio. Syntax kout gainslider kindex Performance kindex -- Index value. Nominal range from 0-127. For example a range of 0-152 will give you a range from -∞ to +18.0 dB. kout -- Scaled output. Examples Here is an example of the gainslider opcode. It uses the file gainslider.csd. Example of the gainslider opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also scale, logcurve, expcurve Credits Author: David Akbari October 2006 manual_src~dfsg/opcodes/pvscale.xml0000644000000000000000000001125312262561504016562 0ustar rootroot pvscale Spectral Processing:Streaming pvscale pvscale Scale the frequency components of a pv stream. Description Scale the frequency components of a pv stream, resulting in pitch shift. Output amplitudes can be optionally modified in order to attempt formant preservation. Syntax fsig pvscale fsigin, kscal[, kkeepform, kgain, kcoefs] Performance fsig -- output pv stream fsigin -- input pv stream kscal -- scaling ratio. kkeepform -- attempt to keep input signal formants; 0: do not keep formants; 1: keep formants using a liftered cepstrum method; 2: keep formants by using a true envelope method (defaults to 0). kgain -- amplitude scaling (defaults to 1). kcoefs -- number of cepstrum coefs used in formant preservation (defaults to 80). The quality of the pitch shift will be improved with the use of a Hanning window in the pvoc analysis. Formant preservation method 1 is less intensive than method 2, which might not be suited to realtime use. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Example asig in ; get the signal in fsig pvsanal asig, 1024, 256, 1024, 1 ; analyse it ftps pvscale fsig, 1.5, 1, 1 ; transpose it keeping formants atps pvsynth ftps ; synthesise it adp delayr 0.1 ; delay original signal adel deltapn 1024 ; by 1024 samples delayw asig out atps + adel ; add tranposed and original The example above shows a vocal harmoniser. The delay is necessary to time-align the signals, as the analysis-synthesis process will imply a delay of 1024 samples between the analysis input and the synthesis output. Here is an example of the use of the pvscale opcode. It uses the file pvscale.csd. Example of the <emphasis>pvscale</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namevictor; November 2004 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/vco2.xml0000644000000000000000000001716412262561504016005 0ustar rootroot vco2 Signal Generators:Dynamic Spectrum Oscillators vco2 vco2 Implementation of a band-limited oscillator using pre-calculated tables. Description vco2 is similar to vco. But the implementation uses pre-calculated tables of band-limited waveforms (see also GEN30) rather than integrating impulses. This opcode can be faster than vco (especially if a low control-rate is used) and also allows better sound quality. Additionally, there are more waveforms and oscillator phase can be modulated at k-rate. The disadvantage is increased memory usage. For more details about vco2 tables, see also vco2init and vco2ft. Syntax ares vco2 kamp, kcps [, imode] [, kpw] [, kphs] [, inyx] Initialization imode (optional, default=0) -- a sum of values representing the waveform and its control values. One may use any of the following values for imode: 16: enable k-rate phase control (if set, kphs is a required k-rate parameter that allows phase modulation) 1: skip initialization One may use exactly one of these imode values to select the waveform to be generated: 14: user defined waveform -1 (requires using the vco2init opcode) 12: triangle (no ramp, faster) 10: square wave (no PWM, faster) 8: 4 * x * (1 - x) (i.e. integrated sawtooth) 6: pulse (not normalized) 4: sawtooth / triangle / ramp 2: square / PWM 0: sawtooth The default value for imode is zero, which means a sawtooth wave with no k-rate phase control. inyx (optional, default=0.5) -- bandwidth of the generated waveform, as percentage (0 to 1) of the sample rate. The expected range is 0 to 0.5 (i.e. up to sr/2), other values are limited to the allowed range. Setting inyx to 0.25 (sr/4), or 0.3333 (sr/3) can produce a fatter sound in some cases, although it is more likely to reduce quality. Performance ares -- the output audio signal. kamp -- amplitude scale. In the case of a imode waveform value of 6 (a pulse waveform), the actual output level can be a lot higher than this value. kcps -- frequency in Hz (should be in the range -sr/2 to sr/2). kpw (optional) -- the pulse width of the square wave (imode waveform=2) or the ramp characteristics of the triangle wave (imode waveform=4). It is required only by these waveforms and ignored in all other cases. The expected range is 0 to 1, any other value is wrapped to the allowed range. Warning kpw must not be an exact integer value (e.g. 0 or 1) if a sawtooth / triangle / ramp (imode waveform=4) is generated. In this case, the recommended range is about 0.01 to 0.99. There is no such limitation for a square/PWM waveform. kphs (optional) -- oscillator phase (depending on imode, this can be either an optional i-rate parameter that defaults to zero or required k-rate). Similarly to kpw, the expected range is 0 to 1. Note When a low control-rate is used, pulse width (kpw) and phase (kphs) modulation is internally converted to frequency modulation. This allows for faster processing and reduced artifacts. But in the case of very long notes and continuous fast changes in kpw or kphs, the phase may drift away from the requested value. In most cases, the phase error is at most 0.037 per hour (assuming a sample rate of 44100 Hz). This is a problem mainly in the case of pulse width (kpw), where it may result in various artifacts. While future releases of vco2 may fix such errors, the following work-arounds may also be of some help: Use kpw values only in the range 0.05 to 0.95. (There are more artifacts around integer values) Try to avoid modulating kpw by asymmetrical waveforms like a sawtooth wave. Relatively slow (<= 20 Hz) symmetrical modulation (e.g. sine or triangle), random splines (also slow), or a fixed pulse width is a lot less likely to cause synchronization problems. In some cases, adding random jitter (for example: random splines with an amplitude of about 0.01) to kpw may also fix the problem. Examples Here is an example of the vco2 opcode. It uses the file vco2.csd. Example of the vco2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vco, vco2ft, vco2ift, and vco2init. Credits Author: &nameistvan; New in version 4.22 manual_src~dfsg/opcodes/mincer.xml0000644000000000000000000000632012262561504016401 0ustar rootroot mincer Spectral Processing:Other mincer mincer Phase-locked vocoder processing. Description mincer implements phase-locked vocoder processing using function tables containing sampled-sound sources, with GEN01, and mincer will accept deferred allocation tables. This opcode allows for time and frequency-independent scaling. Time is controlled by a time index (in seconds) to the function table position and can be moved forward and backward at any chosen speed, as well as stopped at a given position ("frozen"). The quality of the effect is generally improved with phase locking switched on. mincer will also scale pitch, independently of frequency, using a transposition factor (k-rate). Syntax asig mincer atimpt, kamp, kpitch, ktab, klock[,ifftsize,idecim] Initialization ifftsize -- FFT size (power-of-two), defaults to 2048. idecim -- decimation, defaults to 4 (meaning hopsize = fftsize/4) Performance atimpt -- time position of current audio sample in secs. Table reading wraps around the ends of the function table. kamp -- amplitude scaling kpitch -- grain pitch scaling (1=normal pitch, < 1 lower, > 1 higher; negative, backwards) klock -- 0 or 1, to switch phase-locking on/off ktab -- source signal function table. Deferred-allocation tables (see GEN01) are accepted, but the opcode expects a mono source. Tables can be switched at k-rate. Examples Here is an example of the mincer opcode. It uses the file mincer.csd Example of the mincer opcode. Credits Author: &namevictor; February 2010 New plugin in version 5.13 February 2005. manual_src~dfsg/opcodes/octave.xml0000644000000000000000000000471612262561504016414 0ustar rootroot octave Pitch Converters:Functions octave octave Calculates a factor to raise/lower a frequency by a given amount of octaves. Description Calculates a factor to raise/lower a frequency by a given amount of octaves. Syntax octave(x) This function works at a-rate, i-rate, and k-rate. Initialization x -- a value expressed in octaves. Performance The value returned by the octave function is a factor. You can multiply a frequency by this factor to raise/lower it by the given amount of octaves. Examples Here is an example of the octave opcode. It uses the file octave.csd. Example of the octave opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i1 3520.00000 i1 1760.00000 i1 880.00000 i1 440.00000 i1 7040.00000 i1 3520.00000 i1 1760.00000 i1 880.00000 i1 440.00000 ....... See Also cent, db, semitone Credits New in version 4.16 manual_src~dfsg/opcodes/raises.xml0000644000000000000000000000750412262561504016417 0ustar rootroot ˆ Mathematical Operations:Arithmetic and Logic Operations ˆ ˆ Power of operator. Description Arithmetic operators perform operations of change-sign (negate), don't-change-sign, logical AND logical OR, add, subtract, multiply and divide. Note that a value or an expression may fall between two of these operators, either of which could take it as its left or right argument, as in a + b * c. In such cases three rules apply: 1. * and / bind to their neighbors more strongly than + and −. Thus the above expression is taken as a + (b * c) with * taking b and c and then + taking a and b * c. 2. + and − bind more strongly than &&, which in turn is stronger than ||: a && b - c || d is taken as (a && (b - c)) || d 3. When both operators bind equally strongly, the operations are done left to right: a - b - c is taken as (a - b) - c Parentheses may be used as above to force particular groupings. The operator ˆ raises a to the b power. b may not be audio-rate. Use with caution as precedence may not work correctly. See pow. (New in Csound version 3.493.) Syntax a ˆ b (b not audio-rate) where the arguments a and b may be further expressions. Examples Here is an example of the ˆ operator. It uses the file raises.csd. Example of the ˆ operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i1 0.50000 i1 0.50021 i1 0.50042 ...... i1 1327.49320 i1 1327.74465 See Also -, +, &&, ||, *, /, % manual_src~dfsg/opcodes/tablefilter.xml0000644000000000000000000000673712262561504017435 0ustar rootroot tablefilter Table Control:Read/Write Opreations tablefilter tablefilter Filters a source table and writes result into a destination table. Description This opcode can be used in order to filter values from function tables following certain algorithms. The filtered output is written into a destination table and the number of elements that have passed the filter is returned. Syntax knumpassed tablefilter kouttable, kintatble, kmode, kparam Performance knumpassed -- the number of elements that have passed the filter. kouttable -- the number of the table containing the values that have passed. kintatble -- the number of the table used as filter input. kmode -- mode of the filter: 1 -- tests the weight of the denominators of the fractions in the source table. Letting pass only values from the source that are less heavy than the weight of the threshold. 2 -- tests the weight of the denominators of the fractions in the source table. Letting pass only values from the source that are heavier than or equal to the weight of the threshold. kparam -- integer threshold parameter for the filter. It means that denominators whose weights are heavier than the weight of this threshold are not passed through the filter. The weight of an integer is calculated using Clarence Barlow's function of indigestibility of a number. According to this function, higher prime numbers contribute to an increased weight of any natural integer they divide. The order of the first 16 integers according to their indigestibility is: 1, 2, 4, 3, 8, 6, 16, 12, 9, 5, 10, 15, 7, 14. Examples Here is an example of the tablefilter opcode. It uses the file tablefilter.csd. Example of the tablefilter opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Georg Boenn University of Glamorgan, UK New in Csound version 5.13 manual_src~dfsg/opcodes/socksend.xml0000644000000000000000000000565212262561504016744 0ustar rootroot socksend Network socksend socksend Sends data to other processes using the low-level UDP or TCP protocols Description Transmits data directly using the UDP (socksend and socksends) or TCP (stsend) protocol onto a network. The data is not subject to any encoding or special routing. The socksends opcode send a stereo signal interleaved. Syntax socksend asig, Sipaddr, iport, ilength socksends asigl, asigr, Sipaddr, iport, ilength stsend asig, Sipaddr, iport Initialization Sipaddr -- a string that is the IP address of the receiver in standard 4-octet dotted form. iport -- the number of the port that is used for the communication. ilength -- the length of the individual packets in UDP transmission. This number must be sufficiently small to fit a single MTU, which is set to the save value of 1456. In UDP transmissions the receiver needs to know this value Performance asig, asigl, asigr -- audio data to be transmitted. Example The example shows a simple sine wave being sent just once to a computer called "172.16.0.255", on port 7777 using UDP. Note that .255 is often used for broadcasting. sr = 44100 ksmps = 100 nchnls = 1 instr 1 a1 oscil 20000,441,1 socksend a1, "172.16.0.255",7777, 200 endin Credits Author: &namejohn; 2006 manual_src~dfsg/opcodes/hvs3.xml0000644000000000000000000002765512262561504016025 0ustar rootroot hvs3 Signal Generators:Hyper Vectorial Synthesis hvs3 hvs3 Allows three-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables. Description hvs3 allows three-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables. Syntax hvs3 kx, ky, kz, inumParms, inumPointsX, inumPointsY, inumPointsZ, iOutTab, iPositionsTab, iSnapTab [, iConfigTab] Initialization inumParms - number of parameters controlled by the HVS. Each HVS snapshot is made up of inumParms elements. inumPointsX, inumPointsY, inumPointsZ - number of points that each dimension of the HVS cube (or square in case of two-dimensional HVS; or line in case of one-dimensional HVS) is made up. iOutTab - number of the table receiving the set of output-parameter instant values of the HVS. The total amount of parameters is defined by the inumParms argument. iPositionsTab – a table filled with the individual positions of snapshots in the HVS matrix (see below for more information). iSnapTab – a table filled with all the snapshots. Each snapshot is made up of a set of parameter values. The amount of elements contained in each snapshots is specified by the inumParms argument. The set of elements of each snapshot follows (and is adjacent) to the previous one in this table. So the total size of this table should be >= to inumParms multiplied the number of snapshots you intend to store for the HVS. iConfigTab – (optional) a table containing the behavior of the HVS for each parameter. If the value of iConfigTab is zero (default), this argument is ignored, meaning that each parameter is treated with linear interpolation by the HVS. If iConfigTab is different than zero, then it must refer to an existing table whose contents are in its turn referring to a particolar kind of interpolation. In this table, a value of -1 indicates that corresponding parameter is leaved unchanged (ignored) by the HVS; a value of zero indicates that corresponding parameter is treated with linear-interpolation; each other values must be integer numbers indicating an existing table filled with a shape which will determine the kind of special interpolation to be used (table-based interpolation). Performance kx, ky, kz - these are externally-modified variables which controls the motion of the pointer in the HVS matrix cube (or square or line in case of HVS matrices made up of less than 3 dimensions). The range of these input arguments must be 0 to 1. Hyper Vectorial Synthesis is a technique that allows control of a huge set of parameters by using a simple and global approach. The key concepts of the HVS are: The set of HVS parameters, whose amount is fixed and defined by the inumParms argument. During the HVS performance, all these parameters are variant and can be applied to any sound synthesis technique, as well as to any global control for algorithmic composition and any other kind of level. The user must previously define several sets of fixed values for each HVS parameter, each set corresponding to a determinate synthesis configuration. Each set of values is called snapshot, and can be considered as the coordinates of a bound of a multi-dimensional space. The HVS consists on moving a point in this multi-dimensional space (by using a special motion pointer, see below), according and inside the bounds defined by the snapshots. You can fix any amount of HVS parameters (each parameter being a dimension of the multi-dimensional space), even a huge number, the limit only depends on the processing power (and the memory) of your computer and on the complexity of the sound-synthesis you will use. The HVS cube (or square or line). This is the matrix (of 3, 2 or 1 dimensions, according to the hvs opcode you intend to use) of “mainstays” (or pivot) points of HVS. The total amount of pivot-points depends on the value of the inumPointsX, inumPointsY and inumPointsZ arguments. In the case of a 3-dimensional HVS matrix you can define, for instance, 3 points for the X dimension, 5 for the Y dimension and 2 for the Z dimension. In this case, the total number of pivot-points is 3 * 5 * 2 = 30. With this set of pivot points, the cube Is divided into smaller cubed zones each one bounded by eight nearby points. Each point is numbered. The numeral order of these points is enstabilished in the following way: number zero is the first point, number 1 the second and so on. Assuming you are using a 3-dimensional HVS cube having the number of points above mentioned (i.e. 3, 5 and 2 respectively for the X, Y and Z axis), the first point (point zero) is the upper-left-front vertex of the cube, by facing the XY plane of the cube. The second point is the middle point of the upper front edge of the cube and so on. You can refer to the figure below in order to understand how the numeral order of the pivot-points proceeds: For the 2-dimensional HVS, it is the same, by only omitting the rear cube face, so each zone is bounded by 4 pivot-points instead of 8. For the 1-dimensional HVS, the whole thing is even simpler because it is a line with the pivot-points proceeding from left to right. Each point is coupled with a snapshot. Snapshot order, as stored into the iSnapTab, can or cannot follow the order of the pivot-points numbers. In fact it is possible to alter this order by means the iPositionsTab, a table that remaps the position of each snapshot in relation to the pivot points. The iPositionsTab is made up of the positions of the snapshots (contained in the iSnapTab) in the two-dimensional grid. Each subsequent element is actually a pointer representing the position in the iSnapTab. For example, in a 2-dimensional HVS matrix such as the following (in this case having inumPointsX = 3 and inumPointsY = 5: 571 349 620 413 827
These numbers (to be stored in the iSnapTab table by using, for instance, the GEN02 function generator) represents the snapshot position within the grid (in this case a 3x5 matrix). So, the first element 5, has index zero and represents the 6th (element zero is the first) snapshot contained in the iSnapTab, the second element 7 represents the 8th element of iSnapTab and so on. Summing up, the vertices of each zone (a cubed zone is delimited by 8 vertices; a squared zone by 4 vertices and a linear zone by 2 points) are coupled with a determinate snapshot, whose number is remapped by the iSnapTab. Output values of the HVS are influenced by the motion pointer, a point whose position, in the HVS cube (or square or segment) is determined by the kx, ky and kz arguments. The values of these arguments, which must be in the 0 to 1 range, are externally set by the user. The output values, whose amount is equal to the inumParms argument, are stored in the iOutTab, a table that must be already allocated by the user, and must be at least inumParms size. In what way the motion pointer influences the output? Well, when the motion pointer falls in a determinate cubed zone, delimited, for instance, by 8 vertices (or pivot points), we assume that each vertex has associated a different snapshot (i.e. a set of inumParms values), well, the output will be the weighted average value of the 8 vertices, calculated according on the distance of the motion pointer from each of the 8 vertices. In the case of a default behavior, when the iConfigTab argument is not set, the exact output is calculated by using linear interpolation which is applied to each different parameter of the HVS. Anyway, it is possible to influence this behavior by setting the iConfigTab argument to a number of a table whose contents can affect one or more HVS parameters. The iConfigTab table elements are associated to each HVS parameter and their values affect the HVS output in the following way: If iConfigTab is equal to -1, corresponding output is skipped, i.e. the element is not calculated, leaving corresponding element value in the iOutTab unchanged; If iConfigTab is equal to zero, then the normal HVS output is calculated (by using weighted average of the nearest vertex of current zone where it falls the motion pointer); If iConfigTab element is equal to an integer number > zero, then the contents of a table having that number is used as a shape of a table-based interpolation. See Also hvs1, hvs2, vphaseseg Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/setscorepos.xml0000644000000000000000000000365712262561504017507 0ustar rootroot setscorepos Instrument Control:Sensing and Control setscorepos setscorepos Sets the playback position of the current score performance to a given position. Description Sets the playback position of the current score performance to a given position. Syntax setscorepos ipos Initialization ipos -- playback position in seconds. Examples Here is an example of the setscorepos opcode. It uses the file setscorepos.csd. Example of the setscorepos opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also rewindscore Credits Author: &namevictor; 2008 New in Csound version 5.09 manual_src~dfsg/opcodes/nrpn.xml0000644000000000000000000000514412262561504016104 0ustar rootroot nrpn Real-time MIDI:Output nrpn nrpn Sends a Non-Registered Parameter Number to the MIDI OUT port. Description Sends a NPRN (Non-Registered Parameter Number) message to the MIDI OUT port each time one of the input arguments changes. Syntax nrpn kchan, kparmnum, kparmvalue Performance kchan -- MIDI channel (1-16) kparmnum -- number of NRPN parameter kparmvalue -- value of NRPN parameter This opcode sends new message when the MIDI translated value of one of the input arguments changes. It operates at k-rate. Useful with the MIDI instruments that recognize NRPNs (for example with the newest sound-cards with internal MIDI synthesizer such as SB AWE32, AWE64, GUS etc. in which each patch parameter can be changed during the performance via NRPN) Examples Here is an example of the nrpn opcode. It uses the file nrpn.csd. Example of the nrpn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This example generates notes for every note received on the MIDI input. It generates MIDI notes on csound's MIDI output, so be sure to connect something. Credits Author: &namegabriel; Italy 1998 New in Csound version 3.492 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/midiprogramchange.xml0000644000000000000000000001032212262561504020601 0ustar rootroot midiprogramchange Real-time MIDI:MIDI/Score Interoperability midiprogramchange midiprogramchange Gets a MIDI program change value. Description midiprogramchange is designed to simplify writing instruments that can be used interchangeably for either score or MIDI input, and to make it easier to adapt instruments originally written for score input to work with MIDI input. In general, it should be possible to write instrument definitions that work identically with both scores and MIDI, including both MIDI files and real-time MIDI input, without using any conditional statements, and that take full advantage of MIDI voice messages. Note that correlating Csound instruments with MIDI channel numbers is done using the massign opcode for real-time performance,. For file-driven performance, instrument numbers default to MIDI channel number + 1, but the defaults are overridden by any MIDI program change messages in the file. Syntax midiprogramchange xprogram Performance xprogram -- returns the MIDI program change value during MIDI activation, remains unchanged otherwise. If the instrument was activated by MIDI input, the opcode overwrites the value of xprogram with the corresponding value from MIDI input. If the instrument was NOT activated by MIDI input, the value of xprogram remains unchanged. This enables score p-fields to receive MIDI input data during MIDI activation, and score values otherwise. Adapting a score-activated Csound instrument. See the MIDI interop opcodes section for details on adapting score driven instruments for MIDI or vice-versa. Examples Here is an example of the midiprogramchange opcode. It uses the file midiprogramchange.csd. Example of the midiprogramchange opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also midichannelaftertouch, midicontrolchange, mididefault, midinoteoff, midinoteoncps, midinoteonkey, midinoteonoct, midinoteonpch, midipitchbend, midipolyaftertouch Credits Author: Michael Gogins New in version 4.20 manual_src~dfsg/opcodes/printks.xml0000644000000000000000000001565712262561504016633 0ustar rootroot printks Signal I/O:Printing and Display printks printks Prints at k-rate using a printf() style syntax. Description Prints at k-rate using a printf() style syntax. Syntax printks "string", itime [, kval1] [, kval2] [...] Initialization "string" -- the text string to be printed. Can be up to 8192 characters and must be in double quotes. itime -- time in seconds between printings. Performance kval1, kval2, ... (optional) -- The k-rate values to be printed. These are specified in string with the standard C value specifier (%f, %d, etc.) in the order given. In Csound version 4.23, you can use as many kval variables as you like. In versions prior to 4.23, you must specify 4 and only 4 kvals (using 0 for unused kvals). printks prints numbers and text which can be i-time or k-rate values. printks is highly flexible, and if used together with cursor positioning codes, could be used to write specific values to locations in the screen as the Csound processing proceeds. A special mode of operation allows this printks to convert kval1 input parameter into a 0 to 255 value and to use it as the first character to be printed. This enables a Csound program to send arbitrary characters to the console. To achieve this, make the first character of the string a # and then, if desired continue with normal text and format specifiers. This opcode can be run on every k-cycle it is run in the instrument. To every accomplish this, set itime to 0. When itime is not 0, the opcode print on the first k-cycle it is called, and subsequently when every itime period has elapsed. The time cycles start from the time the opcode is initialized - typically the initialization of the instrument. Print Output Formatting All standard C language printf() control characters may be used. For example, if kval1 = 153.26789 then some common formatting options are: %f prints with full precision: 153.26789 %5.2f prints: 153.26 %d prints integers-only: 153 %c treats kval1 as an ascii character code. In addition to all the printf() codes, printks supports these useful character codes: printks Code Character Code \\r, \\R, %r, or %R return character (\r) \\n, \\N, %n, %N newline character (\n) \\t, \\T, %t, or %T tab character (\t) %! semicolon character (;) This was needed because a ; is interpreted as an comment. ^ escape character (0x1B) ^ ^ caret character (^) ˜ ESC[ (escape+[ is the escape sequence for ANSI consoles) ˜˜ tilde (˜) For more information about printf() formatting, consult any C language documentation. Note Prior to version 4.23, only the %f format code was supported. Examples Here is an example of the printks opcode. It uses the file printks.csd. Example of the printks opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: kup = 0.000000, kdown = 30.000000 kup = 20.010843, kdown = 25.962524 kup = 40.029991, kdown = 21.925049 kup = 60.049141, kdown = 17.887573 kup = 79.933266, kdown = 13.872493 See Also printk2 and printk Credits Author: Robin Whittle Australia May 1997 Example written by &namekevin;. Thanks goes to &nameluis; for pointing out a mistake with the itime parameter. Thanks to &namematt;, updated the documentation for version 4.23. manual_src~dfsg/opcodes/tonek.xml0000644000000000000000000000634112262561504016247 0ustar rootroot tonek Signal Modifiers:Standard Filters:Control tonek tonek A first-order recursive low-pass filter with variable frequency response. Description A first-order recursive low-pass filter with variable frequency response. Syntax kres tonek ksig, khp [, iskip] Initialization iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance kres -- the output signal at control-rate. ksig -- the input signal at control-rate. khp -- the response curve's half-power point, in Hertz. Half power is defined as peak power / root 2. tonek is like tone except its output is at control-rate rather than audio rate. Examples Here is an example of the tonek opcode. It uses the file tonek.csd. Example of the tonek opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also areson, aresonk, atone, atonek, port, portk, reson, resonk, tone Credits Author: Robin Whittle Australia May 1997 manual_src~dfsg/opcodes/kgauss.xml0000644000000000000000000000076012262561504016423 0ustar rootroot kgauss kgauss kgauss Deprecated. Description Deprecated as of version 3.49. Use the gauss opcode instead. manual_src~dfsg/opcodes/convolve.xml0000644000000000000000000001543012262561504016761 0ustar rootroot convolve Signal Modifiers:Convolution and Morphing convolve convolve Convolves a signal and an impulse response. Description Output is the convolution of signal ain and the impulse response contained in ifilcod. If more than one output signal is supplied, each will be convolved with the same impulse response. Note that it is considerably more efficient to use one instance of the operator when processing a mono input to create stereo, or quad, outputs. Note: this opcode can also be written as convle. Syntax ar1 [, ar2] [, ar3] [, ar4] convolve ain, ifilcod [, ichannel] Initialization ifilcod -- integer or character-string denoting an impulse response data file. An integer denotes the suffix of a file convolve.m; a character string (in double quotes) gives a filename, optionally a full pathname. If not a fullpath, the file is sought first in the current directory, then in the one given by the environment variable SADIR (if defined). The data file contains the Fourier transform of an impulse response. Memory usage depends on the size of the data file, which is read and held entirely in memory during computation, but which is shared by multiple calls. ichannel (optional) -- which channel to use from the impulse response data file. Performance ain -- input audio signal. convolve implements Fast Convolution. The output of this operator is delayed with respect to the input. The following formulas should be used to calculate the delay: For (1/kr) <= IRdur: Delay = ceil(IRdur * kr) / kr For (1/kr) > IRdur: Delay = IRdur * ceil(1/(kr*IRdur)) Where: kr = Csound control rate IRdur = duration, in seconds, of impulse response ceil(n) = smallest integer not smaller than n One should be careful to also take into account the initial delay, if any, of the impulse response. For example, if an impulse response is created from a recording, the soundfile may not have the initial delay included. Thus, one should either ensure that the soundfile has the correct amount of zero padding at the start, or, preferably, compensate for this delay in the orchestra (the latter method is more efficient). To compensate for the delay in the orchestra, subtract the initial delay from the result calculated using the above formula(s), when calculating the required delay to introduce into the 'dry' audio path. For typical applications, such as reverb, the delay will be in the order of 0.5 to 1.5 seconds, or even longer. This renders the current implementation unsuitable for real time applications. It could conceivably be used for real time filtering however, if the number of taps is small enough. The author intends to create a higher-level operator at some stage, that would mix the wet & dry signals, using the correct amount of delay automatically. Examples Create frequency domain impulse response file using the cvanal utility: csound -Ucvanal l1_44.wav l1_44.cv Determine duration of impulse response. For high accuracy, determine the number of sample frames in the impulse response soundfile, and then compute the duration with: duration = (sample frames)/(sample rate of soundfile) This is due to the fact that the sndinfo utility only reports the duration to the nearest 10ms. If you have a utility that reports the duration to the required accuracy, then you can simply use the reported value directly. sndinfo l1_44.wav length = 60822 samples, sample rate = 44100 Duration = 60822/44100 = 1.379s. Determine initial delay, if any, of impulse response. If the impulse response has not had the initial delay removed, then you can skip this step. If it has been removed, then the only way you will know the initial delay is if the information has been provided separately. For this example, let's assume that the initial delay is 60ms (0.06s) Determine the required delay to apply to the dry signal, to align it with the convolved signal: If kr = 441: 1/kr = 0.0023, which is <= IRdur (1.379s), so: Delay1 = ceil(IRdur * kr) / kr = ceil(608.14) / 441 = 609/441 = 1.38s Accounting for the initial delay: Delay2 = 0.06s Total delay = delay1 - delay2 = 1.38 - 0.06 = 1.32s Here is similar example of the convolve opcode. It uses the file convolve.csd. Example of the convolve opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also pconvolve, dconv, cvanal. Credits Author: Greg Sullivan 1996 New in version 3.28 manual_src~dfsg/opcodes/timedseq.xml0000644000000000000000000001647212262561504016750 0ustar rootroot timedseq Instrument Control:Sensing and Control timedseq timedseq Time Variant Sequencer Description An event-sequencer in which time can be controlled by a time-pointer. Sequence data are stored into a table. Syntax ktrig timedseq ktimpnt, ifn, kp1 [,kp2, kp3, ...,kpN] Initialization ifn -- number of table containing sequence data. Performance ktri -- output trigger signal ktimpnt -- time pointer into sequence file, in seconds. kp1,...,kpN -- output p-fields of notes. kp2 meaning is relative action time and kp3 is the duration of notes in seconds. timedseq is a sequencer that allows to schedule notes starting from a user sequence, and depending from an external timing given by a time-pointer value (ktimpnt argument). User should fill table ifn with a list of notes, that can be provided in an external text file by using GEN23, or by typing it directly in the orchestra (or score) file with GEN02. The format of the text file containing the sequence is made up simply by rows containing several numbers separated by space (similarly to normal Csound score). The first value of each row must be a positive or null value, except for a special case that will be explained below. This first value is normally used to define the instrument number corresponding to that particular note (like normal score). The second value of each row must contain the action time of corresponding note and the third value its duration. This is an example: 0 0 0.25 1 93 0 0.25 0.25 2 63 0 0.5 0.25 3 91 0 0.75 0.25 4 70 0 1 0.25 5 83 0 1.25 0.25 6 75 0 1.5 0.25 7 78 0 1.75 0.25 8 78 0 2 0.25 9 83 0 2.25 0.25 10 70 0 2.5 0.25 11 54 0 2.75 0.25 12 80 -1 3 -1 -1 -1 ;; last row of the sequence In this example, the first value of each row is always zero (it is a dummy value, but this p-field can be used, for example, to express a MIDI channel or an instrument number), except the last row, that begins with -1. This value (-1) is a special value, that indicates the end of sequence. It has itself an action time, because sequences can be looped. So the previous sequence has a default duration of 3 seconds, being value 3 the last action time of the sequence. It is important that ALL lines contains the same number of values (in the example all rows contains exactly 5 values). The number of values contained by each row, MUST be the number of kpXX output arguments (notice that, even if kp1, kp2 etc. are placed at the right of the opcode, they are output arguments, not input arguments). ktimpnt argument provide the real temporization of the sequence. Actually the passage of time through sequence is specified by ktimpnt itself, which represents the time in seconds. ktimpnt must always be positive, but can move forwards or backwards in time, be stationary or discontinuous, as a pointer into the sequence file, in the same way of pvoc or lpread. When ktimpnt crosses the action time of a note, a trigger signal is sent to ktrig output argument, and kp1, kp2,...kpN arguments are updated with the values of that note. This information can then be used with schedkwhen to actually activate note events. Notice that kp1,...kpn data can be further processed (for example delayed with delayk, transposed, etc.) before feeding schedkwhen. ktimpnt can be controlled by a linear signal, for example: ktimpnt line 0, p3, 3 ; original sequence duration was 3 secs ktrig timedseq ktimpnt, 1, kp1, kp2, kp3, kp4, kp5 schedkwhen ktrig, 105, 2, 0, kp3, kp4, kp5 in this case the complete sequence (with original duration of 3 seconds) will be played in p3 seconds. You can loop a sequence by controlling it with a phasor: kphs phasor 1/3 ktimpnt = kphs * 3 ktrig timedseq ktimpnt ,1 ,kp1, kp2, kp3, kp4, kp5 schedkwhen ktrig, 105, 2, 0, kp3, kp4, kp5 Obviously you can play only a fragment of the sequence, read it backward, and non-linearly access sequence data in the same way of pvoc and lpread opcodes. With timedseq opcode you can do almost all things of a normal score, except you have the following limitations: You can't have two notes exactly starting with the same action time; actually at least a k-cycle should separate timing of two notes (otherwise the schedkwhen mechanism eats one of them). All notes of the sequence must have the same number of p-fields (even if they activate different instruments). You can remedy this limitation by filling with dummy values notes that belongs to instruments with less p-fields than other ones. Examples Here is a complete example of the timedseq opcode. It uses the file timedseq.csd. Example of the timedseq opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also GEN02, GEN23, seqtime, seqtime2, trigseq Credits Author: &namegabriel; manual_src~dfsg/opcodes/maparray.xml0000644000000000000000000000444612262561504016747 0ustar rootroot maparray Table Control:Table Queries maparray maparray Apply a function to every element of a vector. Description Apply a function of one argument to every element of a vector (one-dimensional k-rate array). Syntax karray maparray kinarray, String karray maparray_i kinarray, String Initialization String -- a string that names an opcode function, at i-rate for maparray_i or k-rate for maparray. Performance karray -- array for answers. kinarray -- array for arguments to the function. Examples Here is an example of the maparray opcode. It uses the file maparray.csd. Example of the maparray opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also Vectorial opcodes Credits Author: &namejohn; Codemist Ltd 2013 New in version 6.00 manual_src~dfsg/opcodes/fltabs.xml0000644000000000000000000001752212262561504016405 0ustar rootroot FLtabs FLTK:Containers FLtabs FLtabs Creates a tabbed FLTK interface. Description FLtabs is a file card tabs interface that is useful to display several areas containing widgets in the same windows, alternatively. It must be used together with FLgroup, another container that groups child widgets. Syntax FLtabs iwidth, iheight, ix, iy Initialization iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window. Expressed in pixels. iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window. Expressed in pixels. Performance Containers are useful to format the graphic appearance of the widgets. The most important container is FLpanel, that actually creates a window. It can be filled with other containers and/or valuators or other kinds of widgets. There are no k-rate arguments in containers. FLtabs is a file card tabs interface that is useful to display several alternate areas containing widgets in the same window. FLtabs. FLtabs. It must be used together with FLgroup, another FLTK container opcode that groups child widgets. Examples The following example code: FLpanel "Panel1", 450, 550, 100, 100 FLscroll 450, 550, 0, 0 FLtabs 400, 550, 5, 5 FLgroup "sliders", 380, 500, 10, 40, 1 gk1, ihs FLslider "FLslider 1", 500, 1000, 2 ,1, -1, 300,15, 20,50 gk2, ihs FLslider "FLslider 2", 300, 5000, 2 ,3, -1, 300,15, 20,100 gk3, ihs FLslider "FLslider 3", 350, 1000, 2 ,5, -1, 300,15, 20,150 gk4, ihs FLslider "FLslider 4", 250, 5000, 1 ,11, -1, 300,30, 20,200 gk5, ihs FLslider "FLslider 5", 220, 8000, 2 ,1, -1, 300,15, 20,250 gk6, ihs FLslider "FLslider 6", 1, 5000, 1 ,13, -1, 300,15, 20,300 gk7, ihs FLslider "FLslider 7", 870, 5000, 1 ,15, -1, 300,30, 20,350 gk8, ihs FLslider "FLslider 8", 20, 20000, 2 ,6, -1, 30,400, 350,50 FLgroupEnd FLgroup "rollers", 380, 500, 10, 30, 2 gk1, ihr FLroller "FLroller 1", 50, 1000,.1,2 ,1 ,-1, 200,22, 20,50 gk2, ihr FLroller "FLroller 2", 80, 5000,1,2 ,1 ,-1, 200,22, 20,100 gk3, ihr FLroller "FLroller 3", 50, 1000,.1,2 ,1 ,-1, 200,22, 20,150 gk4, ihr FLroller "FLroller 4", 80, 5000,1,2 ,1 ,-1, 200,22, 20,200 gk5, ihr FLroller "FLroller 5", 50, 1000,.1,2 ,1 ,-1, 200,22, 20,250 gk6, ihr FLroller "FLroller 6", 80, 5000,1,2 ,1 ,-1, 200,22, 20,300 gk7, ihr FLroller "FLroller 7",50, 5000,1,1 ,2 ,-1, 30,300, 280,50 FLgroupEnd FLgroup "joysticks", 380, 500, 10, 40, 3 gk1, gk2, ihj1, ihj2 FLjoy "FLjoy", 50, 18000, 50, 18000, 2, 2, -1, -1, 300, 300, 30, 60 FLgroupEnd FLtabsEnd FLscrollEnd FLpanelEnd ...will produce the following result: FLtabs example, sliders tab. FLtabs example, sliders tab. FLtabs example, rollers tab. FLtabs example, rollers tab. FLtabs example, joysticks tab. FLtabs example, joysticks tab. (Each picture shows a different tab selection inside the same window.) Examples Here is an example of the FLtabs opcode. It uses the file FLtabs.csd. Example of the FLtabs opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLgroup, FLgroupEnd, FLpack, FLpackEnd, FLpanel, FLpanelEnd, FLscroll, FLscrollEnd, FLtabsEnd Credits Author: &namegabriel; New in version 4.22 Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/opbitshr.xml0000644000000000000000000000313512262561504016757 0ustar rootroot ¬ Mathematical Operations:Arithmetic and Logic Operations >> >> Bitshift right operator. Description The bitshift operators shift the bits to the left or to the right the number of bits given. The priority of these operators is less binding that the arithmetic ones, but more binding that the comparisons. Parentheses may be used as above to force particular groupings. Syntax a >> b (bitshift left) where the arguments a and b may be further expressions. Examples See the entry for the << operator for an example. See Also <<, &, | # manual_src~dfsg/opcodes/scanu.xml0000644000000000000000000001160412262561504016236 0ustar rootroot scanu Signal Generators:Scanned Synthesis scanu scanu Compute the waveform and the wavetable for use in scanned synthesis. Description Compute the waveform and the wavetable for use in scanned synthesis. Syntax scanu init, irate, ifnvel, ifnmass, ifnstif, ifncentr, ifndamp, kmass, \ kstif, kcentr, kdamp, ileft, iright, kpos, kstrngth, ain, idisp, id Initialization init -- the initial position of the masses. If this is a negative number, then the absolute of init signifies the table to use as a hammer shape. If init > 0, the length of it should be the same as the intended mass number, otherwise it can be anything. ifnvel -- the ftable that contains the initial velocity for each mass. It should have the same size as the intended mass number. ifnmass -- ftable that contains the mass of each mass. It should have the same size as the intended mass number. ifnstif -- ftable that contains the spring stiffness of each connection. It should have the same size as the square of the intended mass number. The data ordering is a row after row dump of the connection matrix of the system. ifncentr -- ftable that contains the centering force of each mass. It should have the same size as the intended mass number. ifndamp -- the ftable that contains the damping factor of each mass. It should have the same size as the intended mass number. ileft -- If init < 0, the position of the left hammer (ileft = 0 is hit at leftmost, ileft = 1 is hit at rightmost). iright -- If init < 0, the position of the right hammer (iright = 0 is hit at leftmost, iright = 1 is hit at rightmost). idisp -- If 0, no display of the masses is provided. id -- If positive, the ID of the opcode. This will be used to point the scanning opcode to the proper waveform maker. If this value is negative, the absolute of this value is the wavetable on which to write the waveshape. That wavetable can be used later from an other opcode to generate sound. The initial contents of this table will be destroyed. Performance kmass -- scales the masses kstif -- scales the spring stiffness kcentr -- scales the centering force kdamp -- scales the damping kpos -- position of an active hammer along the string (kpos = 0 is leftmost, kpos = 1 is rightmost). The shape of the hammer is determined by init and the power it pushes with is kstrngth. kstrngth -- power that the active hammer uses ain -- audio input that adds to the velocity of the masses. Amplitude should not be too great. Examples For an example, see the documentation on scans. See Also More information on this opcode: http://www.csounds.com/stevenyi/scanned/yi_scannedSynthesis.html , written by Steven Yi Credits Author: &nameparis; MIT Media Lab Boston, Massachussetts USA March 2000 New in Csound version 4.05 manual_src~dfsg/opcodes/vibr.xml0000644000000000000000000000503312262561504016066 0ustar rootroot vibr Signal Generators:Basic Oscillators vibr vibr Easier-to-use user-controllable vibrato. Description Easier-to-use user-controllable vibrato. Syntax kout vibr kAverageAmp, kAverageFreq, ifn Initialization ifn -- Number of vibrato table. It normally contains a sine or a triangle wave. Performance kAverageAmp -- Average amplitude value of vibrato kAverageFreq -- Average frequency value of vibrato (in cps) vibr is an easier-to-use version of vibrato. It has the same generation-engine of vibrato, but the parameters corresponding to missing input arguments are hard-coded to default values. Examples Here is an example of the vibr opcode. It uses the file vibr.csd. Example of the vibr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also jitter, jitter2, vibrato Credits Author: &namegabriel; New in Version 4.15 manual_src~dfsg/opcodes/JackoTransport.xml0000644000000000000000000000464112262561504020074 0ustar rootroot JackoTransport Jacko Opcodes JackoTransport JackoTransport Control the Jack transport. Description Starts, stops, or repositions the Jack transport. This is useful, e.g., for starting an external sequencer playing to send MIDI messages to Csound. Syntax JackoTransport kcommand, [kposition] Performance kcommand -- 0 means "no action", 1 starts the transport, 2 stops the transport, and 3 positions the transport to kposition seconds from the beginning of performance (i.e. time 0 in the score). kposition -- Time to position to the transport, in seconds from the beginning of performance (i.e. time 0 in the score). This opcode can be used at init time or during performance. The granularity of timing is Csound's kperiod. See Also JackoInfo, JackoInfo, JackoFreewheel, JackoAudioOutConnect, JackoMidiInConnect, JackoMidiOutConnect, JackoOn, JackoAudioIn, JackoMidiOut, JackoNoteOut. Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/octmidinn.xml0000644000000000000000000001675312262561504017123 0ustar rootroot octmidinn Pitch Converters:Functions octmidinn octmidinn Converts a Midi note number value to octave-point-decimal. Description Converts a Midi note number value to octave-point-decimal. Syntax octmidinn (MidiNoteNumber) (init- or control-rate args only) where the argument within the parentheses may be a further expression. Performance octmidinn is a function that takes an i-rate or k-rate value representing a Midi note number and returns the equivalent pitch value in Csound's octave-point-decimal format. This conversion assumes that Middle C (8.000 in oct) is Midi note number 60. Midi note number values are typically integers in the range from 0 to 127 but fractional values or values outside of this range will be interpreted consistently. octmidinn vs. octmidi The octmidinn opcode may be used in any Csound instrument instance whether it is activated from a Midi event, score event, line event, or from another instrument. The input value for octmidinn might for example come from a p-field in a textual score or it may have been retrieved from the real-time Midi event that activated the current note using the notnum opcode. You must specify an i-rate or k-rate expression for the Midi note number that is to be converted. On the other hand, the octmidi opcode only produces meaningful results in a Midi-activated note (either real-time or from a Midi score with the -F flag). With octmidi, the Midi note number value is taken from the Midi event associated with the instrument instance, and no location or expression for this value may be specified. octmidinn and its related opcodes are really value converters with a special function of manipulating pitch data. Data concerning pitch and frequency can exist in any of the following forms: Pitch and Frequency Values Name Abbreviation octave point pitch-class (8ve.pc) pch octave point decimal oct cycles per second cps Midi note number (0-127) midinn
The first two forms consist of a whole number, representing octave registration, followed by a specially interpreted fractional part. For pch, the fraction is read as two decimal digits representing the 12 equal-tempered pitch classes from .00 for C to .11 for B. For oct, the fraction is interpreted as a true decimal fractional part of an octave. The two fractional forms are thus related by the factor 100/12. In both forms, the fraction is preceded by a whole number octave index such that 8.00 represents Middle C, 9.00 the C above, etc. Midi note number values range between 0 and 127 (inclusively) with 60 representing Middle C, and are usually whole numbers. Thus A440 can be represented alternatively by 440 (cps), 69 (midinn), 8.09 (pch), or 8.75 (oct). Microtonal divisions of the pch semitone can be encoded by using more than two decimal places. The mnemonics of the pitch conversion units are derived from morphemes of the forms involved, the second morpheme describing the source and the first morpheme the object (result). Thus cpspch(8.09) will convert the pitch argument 8.09 to its cps (or Hertz) equivalent, giving the value of 440. Since the argument is constant over the duration of the note, this conversion will take place at i-time, before any samples for the current note are produced. By contrast, the conversion cpsoct(8.75 + k1) which gives the value of A440 transposed by the octave interval k1. The calculation will be repeated every k-period since that is the rate at which k1 varies. Note The conversion from pch, oct, or midinn into cps is not a linear operation but involves an exponential process that could be time-consuming when executed repeatedly. Csound now uses a built-in table lookup to do this efficiently, even at audio rates. Because the table index is truncated without interpolation, pitch resolution when using one of these opcodes is limited to 8192 discrete and equal divisions of the octave, and some pitches of the standard 12-tone equally-tempered scale are very slightly mistuned (by at most 0.15 cents). Examples Here is an example of the octmidinn opcode. It uses the file cpsmidinn.csd. Example of the octmidinn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cpsmidinn, pchmidinn, octmidi, notnum, cpspch, cpsoct, octcps, octpch, pchoct Credits Derived from original value converters by Barry Vercoe. New in version 5.07 manual_src~dfsg/opcodes/tabmorphi.xml0000644000000000000000000001267212262561504017120 0ustar rootroot tabmorphi Table Control:Read/Write Opreations tabmorphi tabmorphi Allow morphing between a set of tables with interpolation. Description tabmorphi allows morphing between a set of tables of the same size, by means of a weighted average between two currently selected tables. Syntax kout tabmorphi kindex, kweightpoint, ktabnum1, ktabnum2, \ ifn1, ifn2 [, ifn3, ifn4, ..., ifnN] Initialization ifn1, ifn2 [, ifn3, ifn4, ..., ifnN] - function table numbers. This is a set of chosen tables the user want to use in the morphing. All tables must have the same length. Be aware that only two of these tables can be chosen for the morphing at one time. Since it is possible to use non-integer numbers for the ktabnum1 and ktabnum2 arguments, the morphing is the result from the interpolation between adjacent consecutive tables of the set. Performance kout - The output value for index kindex, resulting from morphing two tables (see below). kindex - main index index of the morphed resultant table. The range is 0 to table_length (not included). kweightpoint - the weight of the influence of a pair of selected tables in the morphing. The range of this argument is 0 to 1. A zero makes it output the first table unaltered, a 1 makes it output the second table of the pair unaltered. All intermediate values between 0 and 1 determine the gradual morphing between the two tables of the pair. ktabnum1 - the first table chosen for the morphing. This number doesn’t express the table number directly, but the position of the table in the set sequence (starting from 0 to N-1). If this number is an integer, the corresponding table will be chosen unaltered. If it contains fractional values, then an interpolation with the next adjacent table will result. ktabnum2 - the second table chosen for the morphing. This number doesn’t express the table number directly, but the position of the table in the set sequence (starting from 0 to N-1). If this number is an integer, corresponding table will be chosen unaltered. If it contains fractional values, then an interpolation with the next adjacent table will result. The tabmorphi family of opcodes is similar to the table family, but allows morphing between two tables chosen into a set of tables. Firstly the user has to provide a set of tables of equal length (ifn2 [, ifn3, ifn4, ..., ifnN]). Then he can choose a pair of tables in the set in order to perform the morphing: ktabnum1 and ktabnum2 are filled with numbers (zero represents the first table in the set, 1 the second, 2 the third and so on). Then determine the morphing between the two chosen tables with the kweightpoint parameter. After that the resulting table can be indexed with the kindex parameter like a normal table opcode. If the value of this parameter surpasses the length of tables (which must be the same for all tables), then it is wrapped around. tabmorphi is identical to tabmorph, but it performs linear interpolation for non-integer values of kindex, much like tablei. Examples Here is an example of the tabmorphi opcode. It uses the file tabmorphi.csd. Example of the tabmorphi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also table, tabmorph, tabmorpha, tabmorphak, ftmorf, Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/remove.xml0000644000000000000000000000273112262561504016423 0ustar rootroot remove Instrument Control:Invocation remove remove Removes the definition of an instrument. Description Removes the definition of an instrument as long as it is not in use. Syntax remove insnum Initialization insnum -- number or name of the instrument to be deleted Performance As long as the indicated instrument is not active, remove deletes the instrument and memory associated with it. It should be treated with care as it is possible that in some cases its use may lead to a crash. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK June, 2006 New in Csound version 5.04 manual_src~dfsg/opcodes/slider16f.xml0000644000000000000000000001156712262561504016734 0ustar rootroot slider16f Real-time MIDI:Slider Banks slider16f slider16f Creates a bank of 16 different MIDI control message numbers, filtered before output. Description Creates a bank of 16 different MIDI control message numbers, filtered before output. Syntax k1,...,k16 slider16f ichan, ictlnum1, imin1, imax1, init1, ifn1, \ icutoff1,..., ictlnum16, imin16, imax16, init16, ifn16, icutoff16 Initialization ichan -- MIDI channel (1-16) ictlnum1 ... ictlnum16 -- MIDI control number (0-127) imin1 ... imin16 -- minimum values for each controller imax1 ... imax16 -- maximum values for each controller init1 ... init16 -- initial value for each controller ifn1 ... ifn16 -- function table for conversion for each controller icutoff1 ... icutoff16 -- low-pass filter cutoff frequency for each controller Performance k1 ... k16 -- output values slider16f is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider16f allows a bank of 16 different MIDI control message numbers. It filters the signal before output. This eliminates discontinuities due to the low resolution of the MIDI (7 bit). The cutoff frequency can be set separately for each controller (suggested range: .1 to 5 Hz). As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. slider16f does not output the required initial value immediately, but only after some k-cycles because the filter slightly delays the output. See Also s16b14, s32b14, slider16, slider32, slider32f, slider64, slider64f, slider8, slider8f Credits Author: &namegabriel; Italy December 1998 New in Csound version 3.50 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/bamboo.xml0000644000000000000000000001014312262561504016361 0ustar rootroot bamboo Signal Generators:Models and Emulations bamboo bamboo Semi-physical model of a bamboo sound. Description bamboo is a semi-physical model of a bamboo sound. It is one of the PhISEM percussion opcodes. PhISEM (Physically Informed Stochastic Event Modeling) is an algorithmic approach for simulating collisions of multiple independent sound producing objects. Syntax ares bamboo kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ [, ifreq1] [, ifreq2] Initialization idettack -- period of time over which all sound is stopped inum (optional) -- The number of beads, teeth, bells, timbrels, etc. If zero, the default value is 1.25. idamp (optional) -- the damping factor, as part of this equation: damping_amount = 0.9999 + (idamp * 0.002) The default damping_amount is 0.9999 which means that the default value of idamp is 0. The maximum damping_amount is 1.0 (no damping). This means the maximum value for idamp is 0.05. The recommended range for idamp is usually below 75% of the maximum value. imaxshake (optional, default=0) -- amount of energy to add back into the system. The value should be in range 0 to 1. ifreq (optional) -- the main resonant frequency. The default value is 2800. ifreq1 (optional) -- the first resonant frequency. The default value is 2240. ifreq2 (optional) -- the second resonant frequency. The default value is 3360. Performance kamp -- Amplitude of output. Note: As these instruments are stochastic, this is only an approximation. Examples Here is an example of the bamboo opcode. It uses the file bamboo.csd. Example of the bamboo opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also dripwater, guiro, sleighbells, tambourine Credits Author: &nameperry;, part of the PhISEM (Physically Informed Stochastic Event Modeling) Adapted by &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.07 Added notes by &namerasmus; on May 2002. manual_src~dfsg/opcodes/flbutton.xml0000644000000000000000000002057612262561504016772 0ustar rootroot FLbutton FLTK:Other FLbutton FLbutton A FLTK widget opcode that creates a button. Description A FLTK widget opcode that creates a button. Syntax kout, ihandle FLbutton "label", ion, ioff, itype, iwidth, iheight, ix, \ iy, iopcode [, kp1] [, kp2] [, kp3] [, kp4] [, kp5] [....] [, kpN] Initialization ihandle -- a handle value (an integer number) that unequivocally references a corresponding widget. This is used by other opcodes that modify a widget's properties (see Modifying FLTK Widget Appearance). It is automatically output by FLbutton and must not be set by the user label. (The user label is a double-quoted string containing some user-provided text placed near the widget.) label -- a double-quoted string containing some user-provided text, placed near the corresponding widget. Notice that with FLbutton, it is not necessary to call the FLsetTextType opcode at all in order to use a symbol. In this case, it is sufficient to set a label starting with @ followed by the proper formatting string. The following symbols are supported: FLTK label supported symbols. FLTK label supported symbols. The @ sign may be followed by the following optional formatting characters, in this order: # forces square scaling rather than distortion to the widget's shape. +[1-9] or -[1-9] tweaks the scaling a little bigger or smaller. [1-9] rotates by a multiple of 45 degrees. 6 does nothing, the others point in the direction of that key on a numeric keypad. ion -- value output when the button is checked. ioff -- value output when the button is unchecked. itype -- an integer number denoting the appearance of the widget. Several kind of buttons are possible, according to the value of itype argument: 1 - normal button 2 - light button 3 - check button 4 - round button You can add 20 to the value to create a "plastic" type button. (Note that there is no Platic Round button. i.e. if you set type to 24 it will look exactly like type 23). This is the appearance of the buttons: FLbutton. FLbutton. iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iopcode -- score opcode type. You have to provide the ascii code of the letter corresponding to the score opcode. At present time only i (ascii code 105) score statements are supported. A zero value refers to a default value of i. So both 0 and 105 activates the i opcode. A value of -1 disables this opcode feature. Performance kout -- output value kp1, kp2, ..., kpN -- arguments of the activated instruments. Buttons of type 2, 3, and 4 also output (kout argument) the value contained in the ion argument when checked, and that contained in ioff argument when unchecked. By adding 10 to itype argument (i.e. by setting 11 for type 1, 12 for type 2, 13 for type 3 and 14 for type 4) it is possible to skip the button value when getting/setting snapshots (see later section). FLbutton not only outputs a value, but can also activate (or schedule) an instrument provided by the user each time a button is pressed. You can also add 10 to "plastic" button types (31 for type 1, 32 for type 2, etc.) If the iopcode argument is set to a negative number, no instrument is activated. So this feature is optional. In order to activate an instrument, iopcode must be set to 0 or to 105 (the ascii code of character i, referring to the i score opcode). P-fields of the activated instrument are kp1 (instrument number), kp2 (action time), kp3 (duration) and so on with user p-fields. Notice that in dual state buttons (light button, check button and round button), the instrument is activated only when button state changes from unchecked to checked (not when passing from checked to unchecked). Examples Here is an example of the FLbutton opcode. It uses the file FLbutton.csd, and beats.wav. Example of the FLbutton opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLbox, FLbutBank, FLprintk, FLprintk2, FLvalue Credits Author: &namegabriel; New in version 4.22 Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/int.xml0000644000000000000000000000511512262561504015717 0ustar rootroot int Mathematical Operations:Mathematical Functions int int Extracts an integer from a decimal number. Description Returns the integer part of x. Syntax int(x) (init-rate or control-rate; also works at audio rate in Csound5) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Examples Here is an example of the int opcode. It uses the file int.csd. Example of the int opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: integer (0.000000/3) = 0.000000 integer (1.000000/3) = 0.000000 integer (2.000000/3) = 0.000000 integer (3.000000/3) = 1.000000 integer (4.000000/3) = 1.000000 integer (5.000000/3) = 1.000000 integer (6.000000/3) = 2.000000 integer (7.000000/3) = 2.000000 integer (8.000000/3) = 2.000000 integer (9.000000/3) = 3.000000 See Also abs, exp, frac, log, log10, i, sqrt Credits manual_src~dfsg/opcodes/initc14.xml0000644000000000000000000000554012262561504016402 0ustar rootroot initc14 Real-time MIDI:Input initc14 initc14 Initializes the controllers used to create a 14-bit MIDI value. Description Initializes the controllers used to create a 14-bit MIDI value. Syntax initc14 ichan, ictlno1, ictlno2, ivalue Initialization ichan -- MIDI channel (1-16) ictlno1 -- most significant byte controller number (0-127) ictlno2 -- least significant byte controller number (0-127) ivalue -- floating point value (must be within 0 to 1) Performance initc14 can be used together with both midic14 and ctrl14 opcodes for initializing the first controller's value. ivalue argument must be set with a number within 0 to 1. An error occurs if it is not. Use the following formula to set ivalue according with midic14 and ctrl14 min and max range: ivalue = (initial_value - min) / (max - min) See Also ctrl7, ctrl14, ctrl21, ctrlinit, initc7, initc21, midic7, midic14, midic21 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/outipat.xml0000644000000000000000000000641512262561504016616 0ustar rootroot outipat Real-time MIDI:Output outipat outipat Sends polyphonic MIDI aftertouch messages at i-rate. Description Sends polyphonic MIDI aftertouch messages at i-rate. Syntax outipat ichn, inotenum, ivalue, imin, imax Initialization ichn -- MIDI channel number (1-16) inotenum -- MIDI note number (used in polyphonic aftertouch messages) ivalue -- floating point value imin -- minimum floating point value (converted in MIDI integer value 0) imax -- maximum floating point value (converted in MIDI integer value 127 (7 bit)) Performance outipat (i-rate polyphonic aftertouch output) sends polyphonic aftertouch messages. It works only with MIDI instruments which recognize them. It can drive a different value of a parameter for each note currently active. It can scale an i-value floating-point argument according to the imin and imax values. For example, set imin = 1.0 and imax = 2.0. When the ivalue argument receives a 2.0 value, the opcode will send a 127 value to the MIDI OUT device. When the ivalue argument receives a 1.0 value, it will send a 0 value. i-rate opcodes send their message once during instrument initialization. See Also outiat, outic14, outic, outipb, outipc, outkat, outkc14, outkc, outkpat, outkpb, outkpc Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/fink.xml0000644000000000000000000000466512262561504016065 0ustar rootroot fink Signal I/O:File I/O fink fink Read signals from a file at k-rate. Description Read signals from a file at k-rate. Syntax fink ifilename, iskipframes, iformat, kin1 [, kin2] [, kin3] [,...] Initialization ifilename -- input file name (can be a string or a handle number generated by fiopen) iskipframes -- number of frames to skip at the start (every frame contains a sample of each channel) iformat -- a number specifying the input file format. If a header is found, this argument is ignored. 0 - 32 bit floating points without header 1 - 16 bit integers without header Performance fink is the same as fin but operates at k-rate. Please note that since this opcode generates its output using input parameters (on the right side of the opcode), these variables must be initialized before use, otherwise a 'used before defined' error will occur. You can use the init opcode for this. See Also fin, fini Credits Author: &namegabriel; Italy 1999 New in Csound version 3.56 manual_src~dfsg/opcodes/greaterthan.xml0000644000000000000000000000742312262561504017435 0ustar rootroot : ? conditional expressions > Instrument Control:Conditional Values > > Determines if one value is greater than another. Description Determines if one value is greater than another. Syntax (a > b ? v1 : v2) where a, b, v1 and v2 may be expressions, but a, b not audio-rate. Performance In the above conditional, a and b are first compared. If the indicated relation is true (a greater than b), then the conditional expression has the value of v1; if the relation is false, the expression has the value of v2. NB.: If v1 or v2 are expressions, these will be evaluated before the conditional is determined. In terms of binding strength, all conditional operators (i.e. the relational operators (<, etc.), and ?, and : ) are weaker than the arithmetic and logical operators (+, -, *, /, && and ||). These are operators not opcodes. Therefore, they can be used within orchestra statements, but do not form complete statements themselves. Examples Here is an example of the > operator. It uses the file greaterthan.csd. Example of the > operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: k1 = 2.000000, k2 = 0.000000 k1 = 3.000000, k2 = 0.000000 k1 = 4.000000, k2 = 1.000000 See Also ==, >=, <=, <, != Credits Example written by &namekevin;. manual_src~dfsg/opcodes/vstparamset.xml0000644000000000000000000000514212262561504017476 0ustar rootroot vstparamset Plugin Hosting:VST vstparamset,vstparamget vstparamset Used for parameter comunication to and from a VST plugin. Syntax vstparamset instance, kparam, kvalue kvalue vstparamget instance, kparam Description vstparamset and vstparamget are used for parameter comunication to and from a VST plugin. Initialization instance - the number which identifies the plugin, to be passed to other vst4cs opcodes. Performance kparam - The number of the parameter to set or get. kvalue - the value to set, or the the value returned by the plugin. Parameters vary according to the plugin. To find out what parameters are available, use the verbose option when loading the plugin with vstinit. Note that the VST protocol specifies that all parameter values must lie between 0.0 and 1.0, inclusive. Examples Here is an example of the use of the vstparamset opcode. It uses the file vst4cs.csd. Example of the <emphasis>vstparamset</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits By: &nameandres; and &namemichael; Uses code from Hermann Seib's VSTHost and Thomas Grill's vst~ object. VST is a trademark of Steinberg Media Technologies GmbH. VST Plug-In Technology by Steinberg. manual_src~dfsg/opcodes/vstedit.xml0000644000000000000000000000415312262561504016610 0ustar rootroot vstedit Plugin Hosting:VST vstedit vstedit Opens the GUI editor window for a VST plugin. Syntax vstedit instance Description vstedit opens the custom GUI editor window for a VST plugin. Note that not all VST plugins have custom GUI editors. It may be necessary to use the --displays command-line option to ensure that Csound handles events from the editor window and displays it properly. Initialization instance - the number which identifies the plugin, to be passed to other vst4cs opcodes. Examples Here is an example of the use of the vstedit opcode. It uses the file vst4cs.csd. Example of the <emphasis>vstedit</emphasis> opcode See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits By: &nameandres; and &namemichael; Uses code from Hermann Seib's VSTHost and Thomas Grill's vst~ object. VST is a trademark of Steinberg Media Technologies GmbH. VST Plug-In Technology by Steinberg. manual_src~dfsg/opcodes/atonek.xml0000644000000000000000000000721212262561504016406 0ustar rootroot atonek Signal Modifiers:Standard Filters:Control atonek atonek A hi-pass filter whose transfer functions are the complements of the tonek opcode. Description A hi-pass filter whose transfer functions are the complements of the tonek opcode. Syntax kres atonek ksig, khp [, iskip] Initialization iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance kres -- the output signal at control-rate. ksig -- the input signal at control-rate. khp -- the response curve's half-power point, in Hertz. Half power is defined as peak power / root 2. atonek is a filter whose transfer functions is the complement of tonek. atonek is thus a form of high-pass filter whose transfer functions represent the filtered out aspects of their complements. However, power scaling is not normalized in atonek but remains the true complement of the corresponding unit. Examples Here is an example of the atonek opcode. It uses the file atonek.csd. Example of the atonek opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also areson, aresonk, atone, port, portk, reson, resonk, tone, tonek Credits Author: Robin Whittle Australia May 1997 manual_src~dfsg/opcodes/fareyleni.xml0000644000000000000000000001110112262561504017073 0ustar rootroot fareyleni Miscellaneous fareyleni fareyleni returns the length of a Farey Sequence. Description This opcode can be used in conjunction with GENfarey. It calculates the length of Farey Sequence Fn. Its length is given by: |Fn| = 1 + SUM over n phi(m) where phi(m) is Euler's totient function, which gives the number of integers ≤ m that are coprime to m. Some values for the length of Fn given n: n Fn 1 2 2 3 3 5 4 7 5 11 6 13 7 19 8 23 9 29 10 33 11 43 12 47 13 59 14 65 15 73 16 81 17 97 18 103 19 121 20 129 Syntax ifl fareyleni ifn Initialisation The length of the identified Farey sequence is returned. ifn -- Integer identifying the sequence. Examples Here is an example of the fareyleni opcode. It uses the file fareyleni.csd. Example of the fareyleni opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Georg Boenn University of Glamorgan, UK New in Csound version 5.13 manual_src~dfsg/opcodes/fprintks.xml0000644000000000000000000001044312262561504016765 0ustar rootroot fprintks Signal I/O:File I/O fprintks fprintks Similar to printks but prints to a file. Description Similar to printks but prints to a file. Syntax fprintks "filename", "string", [, kval1] [, kval2] [...] Initialization "filename" -- name of the output file. "string" -- the text string to be printed. Can be up to 8192 characters and must be in double quotes. Performance kval1, kval2, ... (optional) -- The k-rate values to be printed. These are specified in string with the standard C value specifier (%f, %d, etc.) in the order given. fprintks is similar to the printks opcode except it outputs to a file and doesn't have a itime parameter. For more information about output formatting, please look at printks's documentation. Examples Here is an example of the fprintks opcode. It uses the file fprintks.csd. Example of the fprintks opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This example will generate a file called my.sco. It should contain lines like this: i1 0.00 3.94 10.26 i1 0.20 3.35 6.22 i1 0.67 3.65 11.33 i1 1.31 1.42 4.13 Here is an example of the fprintks opcode, which converts a standard MIDI file to a csound score. It uses the file fprintks-2.csd. Example of the fprintks opcode to convert a MIDI file to a csound score. This example will generate a file called MIDI2cs.sco containing i-events according to the MIDI file Here is an advanced example of the fprintks opcode, which generates scores for Csound. It uses the file scogen-2.csd. Example of the fprintks opcode to create a Csound score file generator using Csound. This example will generate a file called schottstaedt.sco which can be used as a score together with schottstaedt.orc See Also printks Credits Author: &namematt; January 2003 manual_src~dfsg/opcodes/outo.xml0000644000000000000000000000432212262561504016112 0ustar rootroot outo Signal I/O:Signal Output outo outo Writes 8-channel audio data to an external device or stream. Description Writes 8-channel audio data to an external device or stream. Syntax outo asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8 Performance Sends 8-channel audio samples to an accumulating output buffer (created at the beginning of performance) which serves to collect the output of all active instruments before the sound is written to disk. There can be any number of these output units in an instrument. The type (mono, stereo, quad, hex, or oct) should agree with nchnls. But as of version 3.50, Csound will attempt to change an incorrect opcode to agree with nchnls statement. See Also out, outh, outq, outq1, outq2, outq3, outq4, outs, outs1, outs2, soundout Credits Author: &namejohn; New after 3.30 manual_src~dfsg/opcodes/chanctrl.xml0000644000000000000000000000424512262561504016726 0ustar rootroot chanctrl Real-time MIDI:Input chanctrl chanctrl Get the current value of a MIDI channel controller. Description Get the current value of a controller and optionally map it onto specified range. Syntax ival chanctrl ichnl, ictlno [, ilow] [, ihigh] kval chanctrl ichnl, ictlno [, ilow] [, ihigh] Initialization ichnl -- the MIDI channel (1-16). ictlno -- the MIDI controller number (0-127). ilow, ihigh -- low and high ranges for mapping Examples Here is an example of the chanctrl opcode. It uses the file chanctrl.csd. Example of the chanctrl opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namemike; Mills College May, 1997 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/ipcauchy.xml0000644000000000000000000000077612262561504016742 0ustar rootroot ipcauchy ipcauchy ipcauchy Deprecated. Description Deprecated as of version 3.49. Use the pcauchy opcode instead. manual_src~dfsg/opcodes/islider16.xml0000644000000000000000000000100512262561504016721 0ustar rootroot islider16 islider16 islider16 Deprecated. Description Deprecated as of version 3.52. Use the slider16 opcode instead. manual_src~dfsg/opcodes/mpulse.xml0000644000000000000000000000612212262561504016431 0ustar rootroot mpulse Signal Generators:Dynamic Spectrum Oscillators mpulse mpulse Generates a set of impulses. Description Generates a set of impulses of amplitude kamp separated by kintvl seconds (or samples if kintvl is negative). The first impulse is generated after a delay of ioffset seconds. Syntax ares mpulse kamp, kintvl [, ioffset] Initialization ioffset (optional, default=0) -- the delay before the first impulse. If it is negative, the value is taken as the number of samples, otherwise it is in seconds. Default is zero. Performance kamp -- amplitude of the impulses generated kintvl -- Interval of time in seconds (or samples if kintvl is negative) to the next pulse. After the initial delay, an impulse of kamp amplitude is generated as a single sample. Immediately after generating the impulse, the time of the next one is determined from the value of kintvl at that precise moment. This means that any changes in kintvl between impulses are discarded. If kintvl is zero, there is an infinite wait to the next impulse. If kintvl is negative, the interval is counted in number of samples rather than seconds. Examples Here is an example of the mpulse opcode. It uses the file mpulse.csd. Example of the mpulse opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also Another example of how to use mpulse can be found here: mode Credits Written by &namejohn;. New in version 4.08 manual_src~dfsg/opcodes/chano.xml0000644000000000000000000000464312262561504016222 0ustar rootroot chano Signal I/O:Software Bus chano chano Send data to the outwards software bus Description Send data to a channel of the outward software bus. Syntax chano kval, kchan chano aval, kchan Performance xval --- value to transmit kchan -- a positive integer that indicates which channel of the software bus to write Note that the inward and outward software busses are independent, and are not mixer buses. Also the k-rate and a-rate busses are independent. The last value remains until a new value is written. There is no imposed limit to the number of busses but they use memory so small numbers are to be preferred. Example The example shows the software bus being used as an output audio channel. It assumes that an external program that has access to the API is receiving the values. sr = 44100 kr = 100 ksmps = 1 instr 1 a1 oscil p4, p5, 100 chano 1, a1 endin Credits Author: &namejohn; 2005 New in Csound 5.00 manual_src~dfsg/opcodes/deltapn.xml0000644000000000000000000000761112262561504016557 0ustar rootroot deltapn Signal Modifiers:Delay deltapn deltapn Taps a delay line at variable offset times. Description Tap a delay line at variable offset times. Syntax ares deltapn xnumsamps Performance xnumsamps -- specifies the tapped delay time in number of samples. Each can range from 1 control period to the full delay time of the read/write pair; however, since there is no internal check for adherence to this range, the user is wholly responsible. Each argument can be a constant, a variable, or a time-varying signal. deltapn is identical to deltapi, except delay time is specified in number of samples, instead of seconds (&namehans;). This opcode can tap into a delayr/delayw pair, extracting delayed audio from the idlt seconds of stored sound. There can be any number of deltap and/or deltapi units between a read/write pair. Each receives an audio tap with no change of original amplitude. This opcode can provide multiple delay taps for arbitrary delay path and feedback networks. They can deliver either constant-time or time-varying taps, and are useful for building chorus effects, harmonizers, and Doppler shifts. Constant-time delay taps (and some slowly changing ones) do not need interpolated readout; they are well served by deltap. Medium-paced or fast varying dlt's, however, will need the extra services of deltapi. delayr/delayw pairs may be interleaved. To associate a delay tap unit with a specific delayr unit, it not only has to be located between that delayr and the appropriate delayw unit, but must also precede any following delayr units. See Example 2. (This feature added in Csound version 3.57 by Jens Groh and &namejohn;). N.B. k-rate delay times are not internally interpolated, but rather lay down stepped time-shifts of audio samples; this will be found quite adequate for slowly changing tap times. For medium to fast-paced changes, however, one should provide a higher resolution audio-rate timeshift as input. Examples Here is an example of the deltapn opcode. It uses the file deltapn.csd. Example of the deltapn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also deltap, deltap3, deltapi manual_src~dfsg/opcodes/zkw.xml0000644000000000000000000000477112262561504015747 0ustar rootroot zkw Zak Patch System zkw zkw Writes to a zk variable at k-rate without mixing. Description Writes to a zk variable at k-rate without mixing. Syntax zkw ksig, kndx Performance ksig -- value to be written to the zk location. kndx -- points to the zk or za location to which to write. zkw writes ksig into the zk variable specified by kndx. Examples Here is an example of the zkw opcode. It uses the file zkw.csd. Example of the zkw opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zaw, zawm, ziw, ziwm, zkr, zkwm Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/stack.xml0000644000000000000000000001116412262561504016233 0ustar rootroot stack Instrument Control:Stacks stack stack Initializes the stack. Description Initializes and sets the size of the global stack. Syntax stack iStackSize Initialization iStackSize - size of the stack in bytes. Performance Csound implements a single global stack. Initializing the stack with the stack opcode is not required - it is optional, and if not done, the first use of push or push_f will automatically create a stack of 32768 bytes. Otherwise, stack is normally called from the orchestra header, and takes a stack size parameter in bytes (there is an upper limit of about 16 MB). Once set, the stack size is fixed and cannot be changed during performance. The global stack works in LIFO order: after multiple push calls, pop should be used in reverse order. Each push or pop operation can work on a "bundle" of multiple variables. When using pop, the number, type, and order of items must match those used by the corresponding push. That is, after a 'push Sfoo, ibar', you must call something like 'Sbar, ifoo pop', and not e.g. two separate 'pop' statements. push and pop opcodes can take variables of any type (i-, k-, a- and strings). Variables of type 'a' and 'k' are passed at performance time only, while 'i' and 'S' are passed at init time only. push/pop for a, k, i, and S types copy data by value. By contrast, push_f only pushes a "reference" to the f-signal, and then the corresponding pop_f will copy directly from the original variable to its output signal. For this reason, changing the source f-signal of push_f before pop_f is called is not recommended, and if the instrument instance owning the variable that was passed by push_f is deactivated before pop_f is called, undefined behavior may occur. Any stack errors (trying to push when there is no more space, or pop from an empty stack, inconsistent number or type of arguments, etc.) are fatal and terminate performance. Examples Here is an example of the stack opcode. It uses the file stack.csd. Example of the stack opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also pop, push, pop_f and push_f. Using this opcode is somewhat hackish, as you can read here: Credits By: &nameistvan;. 2006 manual_src~dfsg/opcodes/vstbankload.xml0000644000000000000000000000466012262561504017441 0ustar rootroot vstbankload Plugin Hosting:VST vstbankload vstbankload Loads parameter banks to a VST plugin. Syntax vstbankload instance, ipath Description vstbankload is used for loading parameter banks to a VST plugin. Initialization instance - the number which identifies the plugin, to be passed to other vst4cs opcodes. ipath - the full pathname of the parameter bank (.fxb file). Examples Example for vstbankload /* orc */ sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 gihandle1 vstinit "c:/vstplugins/cheeze/cheeze machine.dll", 1 instr 4 vstbankload gihandle1,"c:/vstplugins/cheeze/chengo'scheese.fxb" vstinfo gihandle1 endin /* sco */ i 3 0 21 i 4 1 1 57 32 Credits By: &nameandres; and &namemichael; Uses code from Hermann Seib's VSTHost and Thomas Grill's vst~ object. VST is a trademark of Steinberg Media Technologies GmbH. VST Plug-In Technology by Steinberg. manual_src~dfsg/opcodes/mixerclear.xml0000644000000000000000000000664612262561504017272 0ustar rootroot MixerClear Mixer Opcodes MixerClear MixerClear Resets all channels of a buss to 0. Syntax MixerClear Description Resets all channels of a buss to 0. Performance Use of the mixer requires that instruments setting gains have smaller numbers than instruments sending signals, and that instruments sending signals have smaller numbers than instruments receiving those signals. However, an instrument may have any number of sends or receives. After the final signal is received, MixerClear must be invoked to reset the busses to 0 before the next kperiod. Examples instr 220 ; Master output ; It applies a bass enhancement, compression and fadeout ; to the whole piece, outputs signals, and clears the mixer. a1 MixerReceive 220, 0 a2 MixerReceive 220, 1 ; Bass enhancement al1 butterlp a1, 100 al2 butterlp a2, 100 a1 = al1*1.5 + a1 a2 = al2*1.5 + a2 ; Global amplitude shape kenv linseg 0., p5 / 2.0, p4, p3 - p5, p4, p5 / 2.0, 0. a1=a1*kenv a2=a2*kenv ; Compression a1 dam a1, 5000, 0.5, 1, 0.2, 0.1 a2 dam a2, 5000, 0.5, 1, 0.2, 0.1 ; Remove DC bias a1blocked dcblock a1 a2blocked dcblock a2 ; Output signals outs a1blocked, a2blocked MixerClear endin Here is a complete example of the Mixerclear opcode. It uses the file Mixer.csd Complete example of the Mixerclear opcode. Credits Author: Michael Gogins (gogins at pipeline dot com). manual_src~dfsg/opcodes/schedwhen.xml0000644000000000000000000001047212262561504017077 0ustar rootroot schedwhen Instrument Control:Invocation schedwhen schedwhen Adds a new score event. Description Adds a new score event. Syntax schedwhen ktrigger, kinsnum, kwhen, kdur [, ip4] [, ip5] [...] schedwhen ktrigger, "insname", kwhen, kdur [, ip4] [, ip5] [...] Initialization ip4, ip5, ... -- Equivalent to p4, p5, etc., in a score i statement. Performance kinsnum -- instrument number. Equivalent to p1 in a score i statement. insname -- A string (in double-quotes) representing a named instrument. ktrigger -- trigger value for new event kwhen -- start time of the new event. Equivalent to p2 in a score i statement. kdur -- duration of event. Equivalent to p3 in a score i statement. schedwhen adds a new score event. The event is only scheduled when the k-rate value ktrigger is first non-zero. The arguments, including options, are the same as in a score. The kwhen time (p2) is measured from the time of this event. If the duration is zero or negative the new event is of MIDI type, and inherits the release sub-event from the scheduling instruction. Note that the schedwhen opcode can't accept string p-fields. If you need to pass strings when instantiating an instrument, use the scoreline or scoreline_i opcode. Examples Here is an example of the schedwhen opcode. It uses the file schedwhen.csd. Example of the schedwhen opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also event, event_i, schedule, schedkwhen, schedkwhennamed, scoreline, scoreline_i Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK November 1998 New in Csound version 3.491 Based on work by &namegabriel; manual_src~dfsg/opcodes/bqrez.xml0000644000000000000000000000762112262561504016254 0ustar rootroot bqrez Signal Modifiers:Standard Filters:Resonant bqrez bqrez A second-order multi-mode filter. Description A second-order multi-mode filter. Syntax ares bqrez asig, xfco, xres [, imode] [, iskip] Initialization imode (optional, default=0) -- The mode of the filter. Choose from one of the following: 0 = low-pass (default) 1 = high-pass 2 = band-pass 3 = band-reject 4 = all-pass iskip (optional, default=0) -- if non zero skip the initialisation of the filter. (New in Csound version 4.23f13 and 5.0) Performance ares -- output audio signal. asig -- input audio signal. xfco -- filter cut-off frequency in Hz. May be i-time, k-rate, a-rate. xres -- amount of resonance. Values of 1 to 100 are typical. Resonance should be one or greater. A value of 100 gives a 20dB gain at the cutoff frequency. May be i-time, k-rate, a-rate. All filter modes can be frequency modulated as well as the resonance can also be frequency modulated. bqrez is a resonant low-pass filter created using the Laplace s-domain equations for low-pass, high-pass, and band-pass filters normalized to a frequency. The bi-linear transform was used which contains a frequency transform constant from s-domain to z-domain to exactly match the frequencies together. Alot of trigonometric identities where used to simplify the calculation. It is very stable across the working frequency range up to the Nyquist frequency. Examples Here is an example of the bqrez opcode. It uses the file bqrez.csd. Example of the bqrez opcode borrowed from the <quote>rezzy</quote> opcode in Kevin Conder's manual. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also biquad, moogvcf, rezzy Credits Author: Matt Gerassimoff New in version 4.32 Written in November 2002. manual_src~dfsg/opcodes/scoreline_i.xml0000644000000000000000000000545212262561504017424 0ustar rootroot scoreline_i Instrument Control:Invocation scoreline_i scoreline_i Issues one or more score line events from an instrument at i-time. Description scoreline_i will issue score events at i-time. It can handle strings in the same conditions as the standard score. Multi-line strings are accepted, using {{ }} to enclose the string. Syntax scoreline_i Sin Initialization Sin -- a string (in double-quotes or enclosed by {{ }}) containing one or more score events. Examples Here is an example of the scoreline_i opcode. It uses the file scoreline_i.csd. Example of the scoreline_i opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also event, event_i, schedule, schedwhen, schedkwhen, schedkwhennamed, scoreline More information on this opcode: http://www.csounds.com/journal/issue15/phrase_loops.html , written by Jim Aikin And in the Floss Manuals: http://en.flossmanuals.net/csound/ch020_e-triggering-instrument-events/. Credits Author: &namevictor;, 2007 manual_src~dfsg/opcodes/dispfft.xml0000644000000000000000000000624212262561504016566 0ustar rootroot dispfft Signal I/O:Printing and Display dispfft displayfft Displays the Fourier Transform of an audio or control signal. Description These units will print orchestra init-values, or produce graphic display of orchestra control signals and audio signals. Uses X11 windows if enabled, else (or if -g flag is set) displays are approximated in ASCII characters. Syntax dispfft xsig, iprd, iwsiz [, iwtyp] [, idbout] [, iwtflg] Initialization iprd -- the period of display in seconds. iwsiz -- size of the input window in samples. A window of iwsiz points will produce a Fourier transform of iwsiz/2 points, spread linearly in frequency from 0 to sr/2. iwsiz must be a power of 2, with a minimum of 16 and a maximum of 4096. The windows are permitted to overlap. iwtyp (optional, default=0) -- window type. 0 = rectangular, 1 = Hanning. The default value is 0 (rectangular). idbout (optional, default=0) -- units of output for the Fourier coefficients. 0 = magnitude, 1 = decibels. The default is 0 (magnitude). iwtflg (optional, default=0) -- wait flag. If non-zero, each display is held until released by the user. The default value is 0 (no wait). Performance dispfft -- displays the Fourier Transform of an audio or control signal (asig or ksig) every iprd seconds using the Fast Fourier Transform method. Examples Here is an example of the dispfft opcode. It uses the file dispfft.csd. Example of the dispfft opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also display, print manual_src~dfsg/opcodes/vbap8move.xml0000644000000000000000000001323112262561504017032 0ustar rootroot vbap8move Signal Modifiers:Panning and Spatialization vbap8move vbap8move Distributes an audio signal among 8 channels with moving virtual sources. Description Distributes an audio signal among 8 channels with moving virtual sources. Syntax ar1, ..., ar8 vbap8move asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] Initialization idur -- the duration over which the movement takes place. ispread -- spreading of the virtual source (range 0 - 100). If value is zero, conventional amplitude panning is used. When ispread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. ifldnum -- number of fields (absolute value must be 2 or larger). If ifldnum is positive, the virtual source movement is a polyline specified by given directions. Each transition is performed in an equal time interval. If ifldnum is negative, specified angular velocities are applied to the virtual source during specified relative time intervals (see below). ifld1, ifld2, ... -- azimuth angles or angular velocities, and relative durations of movement phases (see below). Performance asig -- audio signal to be panned vbap8move allows the use of moving virtual sources. If ifldnum is positive, the fields represent directions of virtual sources and equal times, iazi1, [iele1,] iazi2, [iele2,], etc. The position of the virtual source is interpolated between directions starting from the first direction and ending at the last. Each interval is interpolated in time that is fraction total_time / number_of_intervals of the duration of the sound event. If ifldnum is negative, the fields represent angular velocities and equal times. The first field is, however, the starting direction, iazi1, [iele1,] iazi_vel1, [iele_vel1,] iazi_vel2, [iele_vel2,] .... Each velocity is applied to the note that is fraction total_time / number_of_velocities of the duration of the sound event. If the elevation of the virtual source becomes greater than 90 degrees or less than 0 degrees, the polarity of angular velocity is changed. Thus the elevational angular velocity produces a virtual source that moves up and down between 0 and 90 degrees. Please note that all vbap panning opcodes require the vbap system to be initialized using vbaplsinit. Example Here is a simple example of the vbap8move opcode. It uses the file vbap8move.csd. Example of the vbap8move opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. See Also vbap16, vbap16move, vbap4, vbap4move, vbap8, vbaplsinit, vbapz, vbapzmove Credits Author: Ville Pulkki Sibelius Academy Computer Music Studio Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Helsinki, Finland May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/ephasor.xml0000644000000000000000000000433112262561504016565 0ustar rootroot ephasor ephasor ephasor Performance Ephasor has been added to Csound 5.10, but its behavior will change for 5.11. Stay tuned... Credits Author: &namevictor; 2008 New in version 5.10 manual_src~dfsg/opcodes/gendy.xml0000644000000000000000000001440412262561504016234 0ustar rootroot gendy Signal Generators:Models and Emulations gendy gendy Dynamic stochastic approach to waveform synthesis conceived by Iannis Xenakis. Description Implementation of the Génération Dynamique Stochastique (GENDYN), a dynamic stochastic approach to waveform synthesis conceived by Iannis Xenakis. Syntax ares gendy kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl [, initcps] [, knum] kres gendy kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl [, initcps] [, knum] Initialization initcps (optional, default=12) -- max number of control points. Performance kamp -- amplitude. kampdist -- choice of probability distribution for the next perturbation of the amplitude of a control point. The valid distributions are: 0 - LINEAR 1 - CAUCHY 2 - LOGIST 3 - HYPERBCOS 4 - ARCSINE 5 - EXPON 6 - SINUS (external k-rate signal) If kampdist=6, the user can use an external k-rate signal through kadpar. kdurdist -- choice of distribution for the perturbation of the current inter control point duration. See kampdist for the valid distributions. If kdurdist=6, the user can use an external k-rate signal through kddpar. kadpar -- parameter for the kampdist distribution. Should be in the range of 0.0001 to 1. kddpar -- parameter for the kdurdist distribution. Should be in the range of 0.0001 to 1. kminfreq -- minimum allowed frequency of oscillation. kmaxfreq -- maximum allowed frequency of oscillation. kampscl -- multiplier for the distribution's delta value for amplitude (1.0 is full range). kdurscl -- multiplier for the distribution's delta value for duration. knum (optional, default=initcps) -- current number of utilized control points. The waveform is generated by knum - 1 segments and is repeated in the time. The vertexes (control points) are moved according to a stochastic action and they are limited within the boundaries of a mirror formed by an amplitude barrier and a time barrier. [A repetition of the generated waveform with knum=12.] A repetition of the generated waveform with knum=12. Examples Here is an example of the gendy opcode. It uses the file gendy.csd. Example of the gendy opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an example of the gendy opcode with some modulations. It uses the file gendy-2.csd. Example of the gendy opcode with some modulations. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. References Formalized Music (1992, Stuyvesant, NY: Pendragon Press), pp. 246 - 254, 289 - 322. See Also gendyc gendyx Credits Based on Nick Collins's Gendy1 ugen (SuperCollider) Author: &nametito; January 2012 New in Csound version 5.16 manual_src~dfsg/opcodes/adds.xml0000644000000000000000000000742412262561504016045 0ustar rootroot + Mathematical Operations:Arithmetic and Logic Operations + + Addition operator Description Arithmetic operators perform operations of change-sign (negate), do-no-change-sign, logical AND logical OR, add, subtract, multiply and divide. Note that a value or an expression may fall between two of these operators, either of which could take it as its left or right argument, as in a + b * c. In such cases three rules apply: 1. * and / bind to their neighbors more strongly than + and −. Thus the above expression is taken as a + (b * c) with * taking b and c and then + taking a and b * c. 2. + and − bind more strongly than &&, which in turn is stronger than ||: a && b - c || d is taken as (a && (b - c)) || d 3. When both operators bind equally strongly, the operations are done left to right: a - b - c is taken as (a - b) - c Parentheses may be used as above to force particular groupings. Syntax +a (no rate restriction) a + b (no rate restriction) where the arguments a and b may be further expressions. Arguments The arguments of + can be scalar values or k-rate one dimensional arrays (vectors), or any combination. If one of the arguments is an array, so is the value. Examples Here is an example of the + operator. It uses the file adds.csd. Example of the + operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also -, &&, ||, *, /, ˆ, % Credits Arithmetic on vectors is new in version 6.00 manual_src~dfsg/opcodes/kread4.xml0000644000000000000000000000076212262561504016302 0ustar rootroot kread4 kread4 kread4 Deprecated. Description Deprecated as of version 3.52. Use the readk4 opcode instead. manual_src~dfsg/opcodes/alinrand.xml0000644000000000000000000000077612262561504016725 0ustar rootroot alinrand alinrand alinrand Deprecated. Description Deprecated as of version 3.49. Use the linrand opcode instead. manual_src~dfsg/opcodes/lposcil3.xml0000644000000000000000000000545012262561504016657 0ustar rootroot lposcil3 Signal Generators:Sample Playback lposcil3 lposcil3 Read sampled sound from a table with high precision and cubic interpolation. Description Read sampled sound (mono or stereo) from a table, with looping, and high precision. lposcil3 uses cubic interpolation. Syntax ares lposcil3 kamp, kfreqratio, kloop, kend, ifn [, iphs] Initialization ifn -- function table number Performance kamp -- amplitude kfreqratio -- multiply factor of table frequency (for example: 1 = original frequency, 1.5 = a fifth up , .5 = an octave down) kloop -- start loop point (in samples) kend -- end loop point (in samples) lposcil3 (looping precise oscillator) allows varying at k-rate, the starting and ending point of a sample contained in a table (GEN01). This can be useful when reading a sampled loop of a wavetable, where repeat speed can be varied during the performance. Examples Here is an example of the lposcil3 opcode. It uses the file lposcil3.csd. Example of the lposcil3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also lposcil Credits Author: &namegabriel; Italy New in Csound version 3.52 manual_src~dfsg/opcodes/fltabs_end.xml0000644000000000000000000000147412262561504017232 0ustar rootroot FLtabs)end FLTK:Containers FLtabs_end FLtabs_end Marks the end of a tabbed FLTK interface. Description Marks the end of a tabbed FLTK interface. This is another name for FLtabsEnd provided for compatibility. See FLtabsEnd Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/soundouts.xml0000644000000000000000000000742312262561504017174 0ustar rootroot soundouts Signal I/O:Signal Output soundouts soundouts Deprecated. Writes audio output to a disk file. Description The usage of soundouts is discouraged. Please use fout instead. Writes audio output to a disk file. Syntax soundouts asigl, asigr, ifilcod [, iformat] Initialization ifilcod -- integer or character-string denoting the destination soundfile name. An integer denotes the file soundout.ifilcod; a character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the named file is written relative to the directory given by the SFDIR environment variable if defined, or the current directory. See also GEN01. iformat (optional, default=0) -- specifies the audio data file format: 1 = 8-bit signed char (high-order 8 bits of a 16-bit integer) 4 = 16-bit short integers 5 = 32-bit long integers 6 = 32-bit floats If iformat = 0 it is taken from the Csound -o command-line flag. The default value is 0. Performance soundouts writes stereo audio output to a disk file in raw (headerless) format without 0dBFS scaling. The expected range of the audio signals depends on the selected sample format. Use of fout is recommended instead of soundouts See Also out, outh, outo, outq, outq1, outq2, outq3, outq4, outs, outs1, outs2 soundout Credits Author: &nameistvan; manual_src~dfsg/opcodes/strtol.xml0000644000000000000000000000442212262561504016454 0ustar rootroot strtol Strings:Conversion strtol strtol Converts a string to a signed integer (i-rate). Description Convert a string to a signed integer value. It is also possible to pass an strset index or a string p-field from the score instead of a string argument. If the string cannot be parsed as an integer number, an init error occurs and the instrument is deactivated. Syntax ir strtol Sstr ir strtol indx Initialization Sstr -- String to convert. indx -- index of string set by strset strtol can parse numbers in decimal, octal (prefixed by 0), and hexadecimal (with a prefix of 0x) format. Performance ir -- Value of string as signed integer. Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/STKPlucked.xml0000644000000000000000000000445712262561504017106 0ustar rootroot STKPlucked Signal Generators:STK Opcodes STKPlucked STKPlucked STKPlucked uses a plucked string physical model. Description STKPlucked uses a plucked string physical model based on the Karplus-Strong algorithm. Syntax asignal STKPlucked ifrequency, iamplitude Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Note The code for this opcode is taken directly from the Plucked class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html Examples Here is an example of the STKPlucked opcode. It uses the file STKPlucked.csd. Example of the STKPlucked opcode. See Also STKSitar. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/sfinstrm.xml0000644000000000000000000001160212262561504016770 0ustar rootroot sfinstrm Signal Generators:Sample Playback sfinstrm sfinstrm Plays a SoundFont2 (SF2) sample instrument, generating a mono sound. Description Plays a SoundFont2 (SF2) sample instrument, generating a mono sound. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. Syntax ares sfinstrm ivel, inotenum, xamp, xfreq, instrnum, ifilhandle \ [, iflag] [, ioffset] Initialization ivel -- velocity value inotenum -- MIDI note number value instrnum -- number of an instrument of a SF2 file. ifilhandle -- unique number generated by sfload opcode to be used as an identifier for a SF2 file. Several SF2 files can be loaded and activated at the same time. iflag (optional) -- flag regarding the behavior of xfreq and inotenum ioffset (optional) -- start playing at offset, in samples. Performance xamp -- amplitude correction factor xfreq -- frequency value or frequency multiplier, depending by iflag. When iflag = 0, xfreq is a multiplier of a the default frequency, assigned by SF2 preset to the inotenum value. When iflag = 1, xfreq is the absolute frequency of the output sound, in Hz. Default is 0. When iflag = 0, inotenum sets the frequency of the output according to the MIDI note number used, and xfreq is used as a multiplier. When iflag = 1, the frequency of the output, is set directly by xfreq. This allows the user to use any kind of micro-tuning based scales. However, this method is designed to work correctly only with presets tuned to the default equal temperament. Attempts to use this method with a preset already having non-standard tunings, or with drum-kit-based presets, could give unexpected results. Adjustment of the amplitude can be done by varying the xamp argument, which acts as a multiplier. The ioffset parameter allows the sound to start from a sample different than the first one. The user should make sure that its value is within the length of the specific sound. Otherwise, Csound will probably crash. sfinstrm is a mono version of sfinstr. This is the fastest opcode of the SF2 family. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. See Also sfilist, sfinstr, sfload, sfpassign, sfplay, sfplaym, sfplist, sfpreset Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/inletk.xml0000644000000000000000000000530012262561504016407 0ustar rootroot inletk Signal Flow Graph Opcodes inletk inletk Receives a krate signal into an instrument from a named port. Description Receives a krate signal into an instrument from a named port. Syntax ksignal inletk Sname Initialization Sname -- String name of the inlet port. The name of the inlet is implicitly qualified by the instrument name or number, so it is valid to use the same inlet name in more than one instrument (but not to use the same inlet name twice in one instrument). Performance ksignal -- krate input signal During performance, the krate inlet signal is received from each instance of an instrument containing an outlet port to which this inlet has been connected using the connect opcode. The signals of all the outlets connected to an inlet are summed in the inlet. Examples Here is an example of the inletk opcode. It uses the file inletk.csd. Example of the inletk opcode. See Also outleta outletk outletkid outletf inleta inletf connect alwayson ftgenonce Credits By: &namemichael; 2009 manual_src~dfsg/opcodes/pcauchy.xml0000644000000000000000000001055412262561504016564 0ustar rootroot x-class noise generators pcauchy Signal Generators:Random (Noise) Generators pcauchy pcauchy Cauchy distribution random number generator (positive values only). Description Cauchy distribution random number generator (positive values only). This is an x-class noise generator. Syntax ares pcauchy kalpha ires pcauchy kalpha kres pcauchy kalpha Performance pcauchy kalpha -- controls the spread from zero (big kalpha = big spread). Outputs positive numbers only. For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the pcauchy opcode. It uses the file pcauchy.csd. Example of the pcauchy opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: i 1 time 0.00033: 10.48851 i 1 time 0.20033: 0.29508 i 1 time 0.40033: 1.75214 i 1 time 0.60033: 22.88281 i 1 time 0.80033: 16.06435 i 1 time 1.00000: 0.43110 i 1 time 1.20033: 16.51694 i 1 time 1.40033: 2.98797 i 1 time 1.60033: 1.32767 i 1 time 1.80000: 17.94039 i 1 time 2.00000: 4.85994 Seeding from current time 1526147515 i 2 time 3.00033: 0.89797 i 2 time 3.20033: 9.19447 i 2 time 3.40033: 0.73053 i 2 time 3.60000: 7.43371 i 2 time 3.80033: 5.87640 i 2 time 4.00000: 0.80456 i 2 time 4.20000: 4.50637 i 2 time 4.40033: 2.08145 i 2 time 4.60033: 13.47125 i 2 time 4.80033: 3.16399 i 2 time 5.00000: 11.05428 See Also seed, betarand, bexprnd, cauchy, exprand, gauss, linrand, poisson, trirand, unirand, weibull Credits Author: &nameparis; MIT, Cambridge 1995 manual_src~dfsg/opcodes/ioutpb.xml0000644000000000000000000000076212262561504016432 0ustar rootroot ioutpb ioutpb ioutpb Deprecated. Description Deprecated as of version 3.52. Use the outipb opcode instead. manual_src~dfsg/opcodes/outq2.xml0000644000000000000000000000615112262561504016200 0ustar rootroot outq2 Signal I/O:Signal Output outq2 outq2 Writes samples to quad channel 2 of an external device or stream. Description Writes samples to quad channel 2 of an external device or stream. Syntax outq2 asig Performance Sends audio samples to an accumulating output buffer (created at the beginning of performance) which serves to collect the output of all active instruments before the sound is written to disk. There can be any number of these output units in an instrument. The type (mono, stereo, quad, hex, or oct) should agree with nchnls. But as of version 3.50, Csound will attempt to change an incorrect opcode to agree with the nchnls statement. Opcodes can be chosen to direct sound to any particular channel: outs1 sends to stereo channel 1, outq3 to quad channel 3, etc. Examples Here is an example of the outq2 opcode. It uses the file outq2.csd. Example of the outq2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also out, outh, outo, outq, outq1, outq3, outq4, outs, outs1, outs2, soundout Credits Author: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 manual_src~dfsg/opcodes/fluidNote.xml0000644000000000000000000000577112262561504017066 0ustar rootroot fluidNote Signal Generators:Sample Playback fluidNote fluidNote Plays a note on a channel in a fluidSynth engine. Syntax fluidNote ienginenum, ichannelnum, imidikey, imidivel Description Plays a note at imidikey pitch and imidivel velocity on ichannelnum channel of number ienginenum fluidEngine. Initialization ienginenum -- engine number assigned from fluidEngine ichannelnum -- which channel number to play a note on in the given fluidEngine imidikey -- MIDI key for note (0-127) imidivel -- MIDI velocity for note (0-127) Examples Here is an example of the fluidNote opcode. It uses the file fluidNote.csd and 19Trumpet.sf2. Example of the fluidNote opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output should include a line like this: SoundFont: 1 Bank: 0 Preset: 56 Trumpet metallic See Also fluidEngine, fluidLoad More information on soundfonts in the Floss Manuals: http://en.flossmanuals.net/csound/ch048_d-reading-midi-files Other information on soundfonts on Wikipedia: http://en.wikipedia.org/wiki/Soundfont Credits Michael Gogins (gogins at pipeline dot com), Steven Yi. Thanks to Peter Hanappe for Fluidsynth. manual_src~dfsg/opcodes/oscbnk.xml0000644000000000000000000002250112262561504016402 0ustar rootroot oscbnk Signal Generators:Basic Oscillators oscbnk oscbnk Mixes the output of any number of oscillators. Description This unit generator mixes the output of any number of oscillators. The frequency, phase, and amplitude of each oscillator can be modulated by two LFOs (all oscillators have a separate set of LFOs, with different phase and frequency); additionally, the output of each oscillator can be filtered through an optional parametric equalizer (also controlled by the LFOs). This opcode is most useful for rendering ensemble (strings, choir, etc.) instruments. Although the LFOs run at k-rate, amplitude, phase and filter modulation are interpolated internally, so it is possible (and recommended in most cases) to use this unit at low (˜1000 Hz) control rates without audible quality degradation. The start phase and frequency of all oscillators and LFOs can be set by a built-in seedable 31-bit random number generator, or specified manually in a function table (GEN2). Syntax ares oscbnk kcps, kamd, kfmd, kpmd, iovrlap, iseed, kl1minf, kl1maxf, \ kl2minf, kl2maxf, ilfomode, keqminf, keqmaxf, keqminl, keqmaxl, \ keqminq, keqmaxq, ieqmode, kfn [, il1fn] [, il2fn] [, ieqffn] \ [, ieqlfn] [, ieqqfn] [, itabl] [, ioutfn] Initialization iovrlap -- Number of oscillator units. iseed -- Seed value for random number generator (positive integer in the range 1 to 2147483646 (2 ˆ 31 - 2)). iseed <= 0 seeds from the current time. ieqmode -- Parametric equalizer mode -1: disable EQ (faster) 0: peak 1: low shelf 2: high shelf 3: peak (filter interpolation disabled) 4: low shelf (interpolation disabled) 5: high shelf (interpolation disabled) The non-interpolated modes are faster, and in some cases (e.g. high shelf filter at low cutoff frequencies) also more stable; however, interpolation is useful for avoiding zipper noise at low control rates. ilfomode -- LFO modulation mode, sum of: 128: LFO1 to frequency 64: LFO1 to amplitude 32: LFO1 to phase 16: LFO1 to EQ 8: LFO2 to frequency 4: LFO2 to amplitude 2: LFO2 to phase 1: LFO2 to EQ If an LFO does not modulate anything, it is not calculated, and the ftable number (il1fn or il2fn) can be omitted. il1fn (optional: default=0) -- LFO1 function table number. The waveform in this table has to be normalized (absolute value <= 1), and is read with linear interpolation. il2fn (optional: default=0) -- LFO2 function table number. The waveform in this table has to be normalized, and is read with linear interpolation. ieqffn, ieqlfn, ieqqfn (optional: default=0) -- Lookup tables for EQ frequency, level, and Q (optional if EQ is disabled). Table read position is 0 if the modulator signal is less than, or equal to -1, (table length / 2) if the modulator signal is zero, and the guard point if the modulator signal is greater than, or equal to 1. These tables have to be normalized to the range 0 - 1, and have an extended guard point (table length = power of two + 1). All tables are read with linear interpolation. itabl (optional: default=0) -- Function table storing phase and frequency values for all oscillators (optional). The values in this table are in the following order (5 for each oscillator unit): oscillator phase, lfo1 phase, lfo1 frequency, lfo2 phase, lfo2 frequency, ... All values are in the range 0 to 1; if the specified number is greater than 1, it is wrapped (phase) or limited (frequency) to the allowed range. A negative value (or end of table) will use the output of the random number generator. The random seed is always updated (even if no random number was used), so switching one value between random and fixed will not change others. ioutfn (optional: default=0) -- Function table to write phase and frequency values (optional). The format is the same as in the case of itabl. This table is useful when experimenting with random numbers to record the best values. The two optional tables (itabl and ioutfn) are accessed only at i-time. This is useful to know, as the tables can be safely overwritten after opcode initialization, which allows precalculating parameters at i-time and storing in a temporary table before oscbnk initialization. Performance ares -- Output signal. kcps -- Oscillator frequency in Hz. kamd -- AM depth (0 - 1). (AM output) = (AM input) * ((1 - (AM depth)) + (AM depth) * (modulator)) If ilfomode isn't set to modulate the amplitude, then (AM output) = (AM input) regardless of the value of kamd. That means that kamd will have no effect. Note: Amplitude modulation is applied before the parametric equalizer. kfmd -- FM depth (in Hz). kpmd -- Phase modulation depth. kl1minf, kl1maxf -- LFO1 minimum and maximum frequency in Hz. kl2minf, kl2maxf -- LFO2 minimum and maximum frequency in Hz. (Note: oscillator and LFO frequencies are allowed to be zero or negative.) keqminf, keqmaxf -- Parametric equalizer minimum and maximum frequency in Hz. keqminl, keqmaxl -- Parametric equalizer minimum and maximum level. keqminq, keqmaxq -- Parametric equalizer minimum and maximum Q. kfn -- Oscillator waveform table. Table number can be changed at k-rate (this is useful to select from a set of band-limited tables generated by GEN30, to avoid aliasing). The table is read with linear interpolation. Note oscbnk uses the same random number generator as rnd31. So reading its documentation is also recommended. Examples Here is an example of oscbnk opcode. It uses the file oscbnk.csd. Example of the oscbnk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &nameistvan; 2001 New in version 4.15 Updated April 2002 by &nameistvan; manual_src~dfsg/opcodes/dcblock.xml0000644000000000000000000000561612262561504016534 0ustar rootroot dcblock Signal Modifiers:Specialized Filters dcblock dcblock A DC blocking filter. Description Implements the DC blocking filter Y[i] = X[i] - X[i-1] + (igain * Y[i-1]) Based on work by &nameperry;. Syntax ares dcblock ain [, igain] Initialization igain -- the gain of the filter, which defaults to 0.99 Performance ain -- audio signal input The new dcblock2 opcode is an improved method of removing DC from an audio signal. Examples The result can be viewed in a graphical audio editor program like audacity. audacity is available for Linux, Windows, and the MacOS and may be downloaded from http://audacity.sourceforge.net. Here is an example of the dcblock opcode. It uses the file dcblock.csd, and beats.wav. Example of the dcblock opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also dcblock2 Credits Author: &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.49 February 2003: Thanks to a note from Anders Andersson, corrected the formula. manual_src~dfsg/opcodes/STKStifKarp.xml0000644000000000000000000000657112262561504017241 0ustar rootroot STKStifKarp Signal Generators:STK Opcodes STKStifKarp STKStifKarp STKStifKarp is a plucked stiff string instrument. Description STKStifKarp is a plucked stiff string instrument. It a simple plucked string algorithm (Karplus Strong) with enhancements, including string stiffness and pluck position controls. The stiffness is modeled with allpass filters. Syntax asignal STKStifKarp ifrequency, iamplitude, [kpos, kv1[, ksus, kv2[, kstretch, kv3]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kpos -- controller 4, pickup position. Value range of kv1 is 0-127. ksus -- controller11, string sustain. Value range of kv2 is 0-127. kstretch -- controller 1, string stretch. Value range of kv3 is 0-127. Note The code for this opcode is taken directly from the StifKarp class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKStifKarp opcode. It uses the file STKStifKarp.csd. Example of the STKStifKarp opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/pop_f.xml0000644000000000000000000000527212262561504016234 0ustar rootroot pop_f Instrument Control:Stacks pop_f pop_f Pops an f-sig frame from the global stack. Description Pops an f-sig frame from the global stack. Syntax fsig pop_f Performance fsig - f-signal to be popped from the stack. The values are popped the stack. The global stack must be initialized before used, and its size must be set. The global stack works in LIFO order: after multiple push_f calls, pop_f should be used in reverse order. push/pop for a, k, i, and S types copy data by value. By contrast, push_f only pushes a "reference" to the f-signal, and then the corresponding pop_f will copy directly from the original variable to its output signal. For this reason, changing the source f-signal of push_f before pop_f is called is not recommended, and if the instrument instance owning the variable that was passed by push_f is deactivated before pop_f is called, undefined behavior may occur. push_f and pop_f can only take a single argument, and the data is passed both at init and performance time. Any stack errors (trying to push when there is no more space, or pop from an empty stack, inconsistent number or type of arguments, etc.) are fatal and terminate performance. See also stack, push, pop and push_f. Credits By: &nameistvan;. 2006 manual_src~dfsg/opcodes/ampdbfs.xml0000644000000000000000000000421312262561504016537 0ustar rootroot ampdbfs Mathematical Operations:Amplitude Functions ampdbfs ampdbfs Returns the amplitude equivalent (in 16-bit signed integer scale) of the full scale decibel (dB FS) value x. Description Returns the amplitude equivalent of the full scale decibel (dB FS) value x. The logarithmic full scale decibel values will be converted to linear 16-bit signed integer values from −32,768 to +32,767. Syntax ampdbfs(x) (no rate restriction) Examples Here is an example of the ampdbfs opcode. It uses the file ampdbfs.csd. Example of the ampdbfs opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: instr 1: iamp = 29204.511 instr 1: iamp = 18426.801 instr 1: iamp = 16422.904 instr 1: iamp = 3276.800 See Also ampdb, dbamp, dbfsamp, 0dbfs New in Csound version 4.10 manual_src~dfsg/opcodes/spat3di.xml0000644000000000000000000002750312262561504016501 0ustar rootroot spat3di Signal Modifiers:Panning and Spatialization spat3di spat3di Positions the input sound in a 3D space with the sound source position set at i-time. Description This opcode positions the input sound in a 3D space, with optional simulation of room acoustics, in various output formats. With spat3di, sound source position is set at i-time. Syntax aW, aX, aY, aZ spat3di ain, iX, iY, iZ, idist, ift, imode [, istor] Initialization iX -- Sound source X coordinate in meters (positive: right, negative: left) iY -- Sound source Y coordinate in meters (positive: front, negative: back) iZ -- Sound source Z coordinate in meters (positive: up, negative: down) idist -- For modes 0 to 3, idist is the unit circle distance in meters. For mode 4, idist is the distance between microphones. The following formulas describe amplitude and delay as a function of sound source distance from microphone(s): amplitude = 1 / (0.1 + distance) delay = distance / 340 (in seconds) Distance can be calculated as: distance = sqrt(iXˆ2 + iYˆ2 + iZˆ2) In Mode 4, distance can be calculated as: distance from left mic = sqrt((iX + idist/2)ˆ2 + iYˆ2 + iZˆ2) distance from right mic = sqrt((iX - idist/2)ˆ2 + iYˆ2 + iZˆ2) With spat3d the distance between the sound source and any microphone should be at least (340 * 18) / sr meters. Shorter distances will work, but may produce artifacts in some cases. There is no such limitation for spat3di and spat3dt. Sudden changes or discontinuities in sound source location can result in pops or clicks. Very fast movement may also degrade quality. ift -- Function table storing room parameters (for free field spatialization, set it to zero or negative). Table size is 54. The values in the table are: Room Parameter Purpose 0 Early reflection recursion depth (0 is the sound source, 1 is the first reflection etc.) for spat3d and spat3di. The number of echoes for four walls (front, back, right, left) is: N = (2*R + 2) * R. If all six walls are enabled: N = (((4*R + 6)*R + 8)*R) / 3 1 Late reflection recursion depth (used by spat3dt only). spat3dt skips early reflections and renders echoes up to this level. If early reflection depth is negative, spat3d and spat3di will output zero, while spat3dt will start rendering from the sound source. 2 imdel for spat3d. Overrides opcode parameter if non-negative. 3 irlen for spat3dt. Overrides opcode parameter if non-negative. 4 idist value. Overrides opcode parameter if >= 0. 5 Random seed (0 - 65535) -1 seeds from current time. 6 - 53 wall parameters (w = 6: ceil, w = 14: floor, w = 22: front, w = 30: back, w = 38: right, w = 46: left) w + 0 Enable reflections from this wall (0: no, 1: yes) w + 1 Wall distance from listener (in meters) w + 2 Randomization of wall distance (0 - 1) (in units of 1 / (wall distance)) w + 3 Reflection level (-1 - 1) w + 4 Parametric equalizer frequency in Hz. w + 5 Parametric equalizer level (1.0: no filtering) w + 6 Parametric equalizer Q (0.7071: no resonance) w + 7 Parametric equalizer mode (0: peak EQ, 1: low shelf, 2: high shelf) imode -- Output mode 0: B format with W output only (mono) aout = aW 1: B format with W and Y output (stereo) aleft = aW + 0.7071*aY aright = aW - 0.7071*aY 2: B format with W, X, and Y output (2D). This can be converted to UHJ: aWre, aWim hilbert aW aXre, aXim hilbert aX aYre, aYim hilbert aY aWXr = 0.0928*aXre + 0.4699*aWre aWXiYr = 0.2550*aXim - 0.1710*aWim + 0.3277*aYre aleft = aWXr + aWXiYr aright = aWXr - aWXiYr 3: B format with all outputs (3D) 4: Simulates a pair of microphones (stereo output) aW butterlp aW, ifreq ; recommended values for ifreq aY butterlp aY, ifreq ; are around 1000 Hz aleft = aW + aX aright = aY + aZ Mode 0 is the cheapest to calculate, while mode 4 is the most expensive. In Mode 4, The optional lowpass filters can change the frequency response depending on direction. For example, if the sound source is located left to the listener then the high frequencies are attenuated in the right channel and slightly increased in the left. This effect can be disabled by not using filters. You can also experiment with other filters (tone etc.) for better effect. Note that mode 4 is most useful for listening with headphones, and is also more expensive to calculate than the B-format (0 to 3) modes. The idist parameter in this case sets the distance between left and right microphone; for headphones, values between 0.2 - 0.25 are recommended, although higher settings up to 0.4 may be used for wide stereo effects. More information about B format can be found here: http://www.york.ac.uk/inst/mustech/3d_audio/ambis2.htm istor (optional, default=0) -- Skip initialization if non-zero (default: 0). Performance ain -- Input signal aW, aX, aY, aZ -- Output signals   mode 0 mode 1 mode 2 mode 3 mode 4 aW W out W out W out W out left chn / low freq. aX 0 0 X out X out left chn / high frq. aY 0 Y out Y out Y out right chn / low frq. aZ 0 0 0 Z out right chn / high fr. If you encounter very slow performance (up to 100 times slower), it may be caused by denormals (this is also true of many other IIR opcodes, including butterlp, pareq, hilbert, and many others). Underflows can be avoided by: Using the denorm opcode on ain before spat3di. mixing low level DC or noise to the input signal, e.g. atmp rnd31 1/1e24, 0, 0 aW, aX, aY, aZ spat3di ain + atmp, ... or aW, aX, aY, aZ spa3di ain + 1/1e24, ... reducing irlen in the case of spat3dt (which does not have an input signal). A value of about 0.005 is suitable for most uses, although it also depends on EQ settings. If the equalizer is not used, irlen can be set to 0. Examples See the examples for spat3d. See Also spat3d, spat3dt Credits Author: &nameistvan; 2001 New in version 4.12 Updated April 2002 by &nameistvan; manual_src~dfsg/opcodes/locsig.xml0000644000000000000000000001446212262561504016412 0ustar rootroot locsig Signal Modifiers:Panning and Spatialization locsig locsig Takes an input signal and distributes between 2 or 4 channels. Description locsig takes an input signal and distributes it among 2 or 4 channels using values in degrees to calculate the balance between adjacent channels. It also takes arguments for distance (used to attenuate signals that are to sound as if they are some distance further than the loudspeaker itself), and for the amount the signal that will be sent to reverberators. This unit is based upon the example in the Charles Dodge/Thomas Jerse book, Computer Music, page 320. Syntax a1, a2 locsig asig, kdegree, kdistance, kreverbsend a1, a2, a3, a4 locsig asig, kdegree, kdistance, kreverbsend Performance kdegree -- value between 0 and 360 for placement of the signal in a 2 or 4 channel space configured as: a1=0, a2=90, a3=180, a4=270 (kdegree=45 would balanced the signal equally between a1 and a2). locsig maps kdegree to sin and cos functions to derive the signal balances (e.g.: asig=1, kdegree=45, a1=a2=.707). kdistance -- value >= 1 used to attenuate the signal and to calculate reverb level to simulate distance cues. As kdistance gets larger the sound should get softer and somewhat more reverberant (assuming the use of locsend in this case). kreverbsend -- the percentage of the direct signal that will be factored along with the distance and degree values to derive signal amounts that can be sent to a reverb unit such as reverb, or reverb2. Examples Here is an example of the locsig opcode. It uses the file locsig_quad.csd. Example of the locsig opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. In the above example, the signal, aout, is sent around a complete circle once during the duration of a note while at the same time it becomes more and more "distant" from the listeners' location. locsig sends the appropriate amount of the signal internally to locsend. The outputs of the locsend are added to global accumulators in a common Csound style and the global signals are used as inputs to the reverb units in a separate instrument. locsig is useful for quad and stereo panning as well as fixed placed of sounds anywhere between two loudspeakers. Below is an example of the fixed placement of sounds in a stereo field. instr 1 a1, a2 locsig asig, p4, p5, .1 ar1, ar2 locsend ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a endin instr 99 ; reverb.... endin A few notes: ;place the sound in the left speaker and near: i1 0 1 0 1 ;place the sound in the right speaker and far: i1 1 1 90 25 ;place the sound equally between left and right and in the middle ground distance: i1 2 1 45 12 e The next example shows a simple intuitive use of the distance value to simulate Doppler shift. The same value is used to scale the frequency as is used as the distance input to locsig. kdistance line 1, p3, 10 kfreq = (ifreq * 340) / (340 + kdistance) asig oscili iamp, kfreq, 1 kdegree line 0, p3, 360 a1, a2, a3, a4 locsig asig, kdegree, kdistance, .1 ar1, ar2, ar3, ar4 locsend See Also locsend Credits Author: Richard Karpen Seattle, WA USA 1998 New in Csound version 3.48 manual_src~dfsg/opcodes/butlp.xml0000644000000000000000000000137612262561504016260 0ustar rootroot butlp Signal Modifiers:Standard Filters butlp butlp Same as the butterlp opcode. Description Same as the butterlp opcode. Syntax ares butlp asig, kfreq [, iskip] ares butlp asig, afreq [, iskip] manual_src~dfsg/opcodes/sekere.xml0000644000000000000000000000727712262561504016416 0ustar rootroot sekere Signal Generators:Models and Emulations sekere sekere Semi-physical model of a sekere sound. Description sekere is a semi-physical model of a sekere sound. It is one of the PhISEM percussion opcodes. PhISEM (Physically Informed Stochastic Event Modeling) is an algorithmic approach for simulating collisions of multiple independent sound producing objects. Syntax ares sekere iamp, idettack [, inum] [, idamp] [, imaxshake] Initialization iamp -- Amplitude of output. Note: As these instruments are stochastic, this is only a approximation. idettack -- period of time over which all sound is stopped inum (optional) -- The number of beads, teeth, bells, timbrels, etc. If zero, the default value is 64. idamp (optional) -- the damping factor, as part of this equation: damping_amount = 0.998 + (idamp * 0.002) The default damping_amount is 0.999 which means that the default value of idamp is 0.5. The maximum damping_amount is 1.0 (no damping). This means the maximum value for idamp is 1.0. The recommended range for idamp is usually below 75% of the maximum value. imaxshake (optional) -- amount of energy to add back into the system. The value should be in range 0 to 1. Examples Here is an example of the sekere opcode. It uses the file sekere.csd. Example of the sekere opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cabasa, crunch, sandpaper, stix Credits Author: &nameperry;, part of the PhISEM (Physically Informed Stochastic Event Modeling) Adapted by &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.07 Added notes by &namerasmus; on May 2002. manual_src~dfsg/opcodes/midic21.xml0000644000000000000000000000740212262561504016356 0ustar rootroot midic21 Real-time MIDI:Input midic21 midic21 Allows a floating-point 21-bit MIDI signal scaled with a minimum and a maximum range. Description Allows a floating-point 21-bit MIDI signal scaled with a minimum and a maximum range. Syntax idest midic21 ictlno1, ictlno2, ictlno3, imin, imax [, ifn] kdest midic21 ictlno1, ictlno2, ictlno3, kmin, kmax [, ifn] Initialization idest -- output signal ictln1o -- most-significant byte controller number (0-127) ictlno2 -- mid-significant byte controller number (0-127) ictlno3 -- least-significant byte controller number (0-127) imin -- user-defined minimum floating-point value of output imax -- user-defined maximum floating-point value of output ifn (optional) -- table to be read when indexing is required. Table must be normalized. Output is scaled according to the imin and imax values. Performance kdest -- output signal kmin -- user-defined minimum floating-point value of output kmax -- user-defined maximum floating-point value of output midic21 (i- and k-rate 21 bit MIDI control) allows a floating-point 21-bit MIDI signal scaled with a minimum and a maximum range. Minimum and maximum values can be varied at k-rate. It can use optional interpolated table indexing. It requires three MIDI controllers as input. Please note that the midic family of opcodes are designed for MIDI triggered events, and don't require a channel number since they will respond to the same channel as the one that triggered the instrument (see massign). However they will crash if called from a score driven event. See Also ctrl7, ctrl14, ctrl21, initc7, initc14, initc21, midic7, midic14 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct controller number range. manual_src~dfsg/opcodes/diskin.xml0000644000000000000000000001704212262561504016410 0ustar rootroot diskin Signal I/O:Signal Input diskin diskin Deprecated. Reads audio data from an external device or stream and can alter its pitch. Description Deprecated. Reads audio data from an external device or stream and can alter its pitch. Syntax ar1 [, ar2 [, ar3 [, ... arN]]] diskin ifilcod, kpitch [, iskiptim] \ [, iwraparound] [, iformat] [, iskipinit] Note the N was 24 in versions before 5.14, and 40 after. ar1[] diskin2 ifilcod, kpitch[, iskiptim \ [, iwrap[, iformat [, iwsize[, ibufsize[, iskipinit]]]]]] (in this version, the number of output channels is not limited.) Initialization ifilcod -- integer or character-string denoting the source soundfile name. An integer denotes the file soundin.filcod ; a character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the named file is sought first in the current directory, then in that given by the environment variable SSDIR (if defined) then by SFDIR. See also GEN01. iskptim (optional) -- time in seconds of input sound to be skipped. The default value is 0. iformat (optional) -- specifies the audio data file format: 1 = 8-bit signed char (high-order 8 bits of a 16-bit integer) 2 = 8-bit A-law bytes 3 = 8-bit U-law bytes 4 = 16-bit short integers 5 = 32-bit long integers 6 = 32-bit floats 7 = 8-bit unsigned int (not available in Csound versions older than 5.00) 8 = 24-bit int (not available in Csound versions older than 5.00) 9 = 64-bit doubles (not available in Csound versions older than 5.00) iwraparound -- 1 = on, 0 = off (wraps around to end of file either direction, enabling looping) iskipinit switches off all initialisation if non zero (default =0). This was introduced in 4_23f13 and csound5. If iformat = 0 it is taken from the soundfile header, and if no header from the Csound -o command-line flag. The default value is 0. Performance a1 ... a24 -- output signals, in the range -0dbfs to 0dbfs. Any samples before the beginning (i.e. negative location) and after the end of the file are assumed to be zero, unless iwrap is non-zero. The number of output arguments must be the same as the number of sound file channels - which can be determined with the filenchnls opcode, otherwise an init error will occur. ar1[] --- output signals, in a-rate array of size N, where N is the number of channels in the file. Arrays are automatically allocated to the correct size. kpitch -- can be any real number. A negative number signifies backwards playback. The given number is a pitch ratio, where: 1 = normal pitch 2 = 1 octave higher 3 = 12th higher, etc. .5 = 1 octave lower .25 = 2 octaves lower, etc. -1 = normal pitch backwards -2 = 1 octave higher backwards, etc. Note to Windows users Windows users typically use back-slashes, \, when specifying the paths of their files. As an example, a Windows user might use the path c:\music\samples\loop001.wav. This is problematic because back-slashes are normally used to specify special characters. To correctly specify this path in Csound, one may alternately: Use forward slashes: c:/music/samples/loop001.wav Use back-slash special characters, \\: c:\\music\\samples\\loop001.wav Examples Here is an example of the diskin opcode. It uses the file diskin.csd, beats.wav. Example of the diskin opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also in, inh, ino, inq, ins, soundin and diskin2 Credits Authors: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 New in version 3.46 Warning to Windows users added by &namekevin;, April 2002 manual_src~dfsg/opcodes/bexprnd.xml0000644000000000000000000001043012262561504016563 0ustar rootroot x-class noise generators bexprnd Signal Generators:Random (Noise) Generators bexprnd bexprnd Exponential distribution random number generator. Description Exponential distribution random number generator. This is an x-class noise generator. Syntax ares bexprnd krange ires bexprnd krange kres bexprnd krange Performance krange -- the range of the random numbers (-krange to +krange) For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the bexprnd opcode. It uses the file bexprnd.csd. Example of the bexprnd opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like: i 1 time 0.00267: 99.27598 i 1 time 0.20267: 74.97176 i 1 time 0.40267: -35.67213 i 1 time 0.60267: 1.10579 i 1 time 0.80267: -18.08816 i 1 time 1.00000: 28.93329 i 1 time 1.20267: 320.63733 i 1 time 1.40267: -332.05614 i 1 time 1.60267: -212.66361 i 1 time 1.80000: -92.57433 i 1 time 2.00000: 140.70939 Seeding from current time 4055201702 i 2 time 3.00267: 190.30495 i 2 time 3.20267: -58.30677 i 2 time 3.40267: 192.39784 i 2 time 3.60000: 12.72448 i 2 time 3.80267: 79.91503 i 2 time 4.00000: 34.44258 i 2 time 4.20000: 167.92680 i 2 time 4.40267: -117.10278 i 2 time 4.60267: -70.99155 i 2 time 4.80267: -23.24037 i 2 time 5.00000: -226.35500 See Also seed, betarand, cauchy, exprand, gauss, linrand, pcauchy, poisson, trirand, unirand, weibull Credits Author: &nameparis; MIT, Cambridge 1995 manual_src~dfsg/opcodes/pvsdemix.xml0000644000000000000000000001071712262561504016770 0ustar rootroot pvsdemix Spectral Processing:Streaming pvsdemix pvsdemix Spectral azimuth-based de-mixing of stereo sources. Description Spectral azimuth-based de-mixing of stereo sources, with a reverse-panning result. This opcode implements the Azimuth Discrimination and Resynthesis (ADRess) algorithm, developed by Dan Barry (Barry et Al. "Sound Source Separation Azimuth Discrimination and Resynthesis". DAFx'04, Univ. of Napoli). The source separation, or de-mixing, is controlled by two parameters: an azimuth position (kpos) and a subspace width (kwidth). The first one is used to locate the spectral peaks of individual sources on a stereo mix, whereas the second widens the 'search space', including/exclufing the peaks around kpos. These two parameters can be used interactively to extract source sounds from a stereo mix. The algorithm is particularly successful with studio recordings where individual instruments occupy individual panning positions; it is, in fact, a reverse-panning algorithm. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Syntax fsig pvsdemix fleft, fright, kpos, kwidth, ipoints Performance fsig -- output pv stream fleft -- left channel input pv stream. fright -- right channel pv stream. kpos -- the azimuth target centre position, which will be de-mixed, from left to right (-1 <= kpos <= 1). This is the reverse pan-pot control. kwidth -- the azimuth subspace width, which will determine the number of points around kpos which will be used in the de-mixing process. (1 <= kwidth <= ipoints) ipoints -- total number of discrete points, which will divide each pan side of the stereo image. This ultimately affects the resolution of the process. Examples The example below takes a stereo input and passes through a de-mixing process revealing a source located at ipos +/- iwidth points. These parameters can be controlled in realtime (e.g. using FLTK widgets or MIDI) for an interactive search of sound sources. Example ifftsize = 1024 iwtype = 1 /* cleaner with hanning window */ ipos = -0.8 /* to the left of the stereo image */ iwidth = 20 /* use peaks of 20 points around it */ al,ar soundin "sinput.wav" flc pvsanal al, ifftsize, ifftsize/4, ifftsize, iwtype frc pvsanal ar, ifftsize, ifftsize/4, ifftsize, iwtype fdm pvsdemix flc, frc, kpos, kwidth, 100 adm pvsynth fdm outs adm, adm Credits Author: &namevictor; January 2005 New plugin in version 5 January 2005. manual_src~dfsg/opcodes/fmb3.xml0000644000000000000000000000754712262561504015767 0ustar rootroot fmb3 Signal Generators:FM Synthesis fmb3 fmb3 Uses FM synthesis to create a Hammond B3 organ sound. Description Uses FM synthesis to create a Hammond B3 organ sound. It comes from a family of FM sounds, all using 4 basic oscillators and various architectures, as used in the TX81Z synthesizer. Syntax ares fmb3 kamp, kfreq, kc1, kc2, kvdepth, kvrate[, ifn1, ifn2, ifn3, \ ifn4, ivfn] Initialization fmb3 takes 5 tables for initialization. The first 4 are the basic inputs and the last is the low frequency oscillator (LFO) used for vibrato. The last table should usually be a sine wave. These all default to a sine wave table. The initial waves should be: ifn1 -- sine wave ifn2 -- sine wave ifn3 -- sine wave ifn4 -- sine wave Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. kc1, kc2 -- Controls for the synthesizer: kc1 -- Total mod index kc2 -- Crossfade of two modulators Algorithm -- 4 kvdepth -- Vibrator depth kvrate -- Vibrator rate Examples Here is an example of the fmb3 opcode. It uses the file fmb3.csd. Example of the fmb3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also fmbell, fmmetal, fmpercfl, fmrhode, fmwurlie Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/pvsbandr.xml0000644000000000000000000000660212262561504016746 0ustar rootroot pvsbandr Spectral Processing:Streaming pvsbandr pvsbandr A band reject filter working in the spectral domain. Description Filter the pvoc frames, rejecting bins whose frequency is within a band, and with linear interpolation for transitional bands. Syntax fsig pvsbandr fsigin, xlowcut, xlowfull, \ xhighfull, xhighcut[, ktype] Performance fsig -- output pv stream fsigin -- input pv stream. xlowcut, xlowfull, xhighfull, xhighcut -- define a trapezium shape for the band that is rejected. The a-rate versions only apply to the sliding case. ktype -- specifies the shape of the transitional band. If at the default value of zero the shape is as below, with linear transition in amplitude. Other values give an exponential curve (1 - exp( r*type )) / (1 - exp(type)) This includes a linear dB shape when ktype is log(10) or about 2.30. The opcode performs a band-reject filter with a spectral envelope shaped like klowcut khighcut ________ ______________ \ / \ / \ / \ / \ / klowfull\__________________________/ khighfull Examples Here is an example of the pvsbandr opcode. It uses the file pvsbandr.csd. Example of the pvsbandr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn, pvsbandp Credits Author: &namejohn; December 2007 manual_src~dfsg/opcodes/xtratim.xml0000644000000000000000000000764612262561504016630 0ustar rootroot xtratim Real-time MIDI:Event Extenders xtratim xtratim Extend the duration of real-time generated events. Description Extend the duration of real-time generated events and handle their extra life (Usually for usage along with release instead of linenr, linsegr, etc). Syntax xtratim iextradur Initialization iextradur -- additional duration of current instrument instance Performance xtratim extends current MIDI-activated note duration by iextradur seconds after the corresponding noteoff message has deactivated the current note itself. It is usually used in conjunction with release. This opcode has no output arguments. This opcode is useful for implementing complex release-oriented envelopes, whose duration is not known when the envelope starts (e.g. for real-time MIDI generated events). Examples Here is a simple example of the xtratim opcode. It uses the file xtratim.csd. Example of the xtratim opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This example shows how to generate a release segment for an ADSR envelope after a MIDI noteoff is received, extending the duration with xtratim and using release to check whether the note is on the release phase. Here is a more elaborate example of the xtratim opcode. It uses the file xtratim-2.csd. More complex example of the xtratim opcode. This example shows how to generate a release segment for an ADSR envelope after a MIDI noteoff is received, extending the duration with xtratim and using release to check whether the note is on the release phase. Two envelopes are generated simultaneously for the left and right channels. See Also linenr, release Credits Author: &namegabriel; Italy Examples by Gabriel Maldonado and Jonathan Murphy New in Csound version 3.47 manual_src~dfsg/opcodes/flsetval.xml0000644000000000000000000000645412262561504016754 0ustar rootroot FLsetVal FLTK:Other FLsetVal FLsetVal Sets the value of a FLTK valuator at control-rate. Description FLsetVal is almost identical to FLsetVal_i. Except it operates at k-rate and it affects the target valuator only when ktrig is set to a non-zero value. Syntax FLsetVal ktrig, kvalue, ihandle Initialization ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. Performance ktrig -- triggers the opcode when different than 0. kvalue -- Value to set the widget to. FLsetVal is not fully implemented yet, and may crash in certain cases (e.g. when setting the value of a widget connected to a FLvalue widget- in this case use two separate FLsetVal) See Also FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/strlower.xml0000644000000000000000000000326712262561504017014 0ustar rootroot strlower Strings:Conversion strlower strlower Convert a string to lower case Description Convert Ssrc to lower case, and write the result to Sdst. strlower runs at init time only. Syntax Sdst strlower Ssrc Examples Here is an example of the strlower opcode. It uses the file strlower.csd. Example of the strlower opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also strlowerk Credits Author: Istvan Varga 2006 New in version 5.02 manual_src~dfsg/opcodes/wgbowedbar.xml0000644000000000000000000000602712262561504017253 0ustar rootroot wgbowedbar Signal Generators:Waveguide Physical Modeling wgbowedbar wgbowedbar A physical model of a bowed bar. Description A physical model of a bowed bar, belonging to the &nameperry; family of waveguide instruments. Syntax ares wgbowedbar kamp, kfreq, kpos, kbowpres, kgain [, iconst] [, itvel] \ [, ibowpos] [, ilow] Initialization iconst (optional, default=0) -- an integration constant. Default is zero. itvel (optional, default=0) -- either 0 or 1. When itvel = 0, the bow velocity follows an ADSR style trajectory. When itvel = 1, the value of the bow velocity decays in an exponentially. ibowpos (optional, default=0) -- the position on the bow, which affects the bow velocity trajectory. ilow (optional, default=0) -- lowest frequency required Performance kamp -- amplitude of signal kfreq -- frequency of signal kpos -- position of the bow on the bar, in the range 0 to 1 kbowpres -- pressure of the bow (as in wgbowed) kgain -- gain of filter. A value of about 0.809 is suggested. Examples Here is an example of the wgbowedbar opcode. It uses the file wgbowedbar.csd. Example of the wgbowedbar opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.07 manual_src~dfsg/opcodes/scoreline.xml0000644000000000000000000000670112262561504017112 0ustar rootroot scoreline Instrument Control:Invocation scoreline scoreline Issues one or more score line events from an instrument. Description Scoreline will issue one or more score events, if ktrig is 1 every k-period. It can handle strings in the same conditions as the standard score. Multi-line strings are accepted, using {{ }} to enclose the string. Syntax scoreline Sin, ktrig Initialization Sin -- a string (in double-quotes or enclosed by {{ }}) containing one or more score events. Performance ktrig -- event trigger, 1 issues the score event, 0 bypasses it. Examples Here is an example of the scoreline opcode. It uses the file scoreline.csd. Example of the scoreline opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. You can use string opcodes like sprintfk to produce strings to be passed to scoreline like this: Sfil = "/Volumes/Bla/file.aif" String sprintfk {{i 2 0 %f "%s" %f %f %f %f}}, idur, Sfil, p5, p6, knorm, iskip scoreline String, ktrig See also event, event_i, schedule, schedwhen, schedkwhen, schedkwhennamed, scoreline_i More information on this opcode: http://www.csounds.com/journal/issue15/phrase_loops.html , written by Jim Aikin And in the Floss Manuals: http://en.flossmanuals.net/csound/ch020_e-triggering-instrument-events/. Credits Author: &namevictor;, 2007 manual_src~dfsg/opcodes/loscil3.xml0000644000000000000000000001304612262561504016477 0ustar rootroot loscil3 Signal Generators:Sample Playback loscil3 loscil3 Read sampled sound from a table using cubic interpolation. Description Read sampled sound (mono or stereo) from a table, with optional sustain and release looping, using cubic interpolation. Syntax ar1 [,ar2] loscil3 xamp, kcps, ifn [, ibas] [, imod1] [, ibeg1] [, iend1] \ [, imod2] [, ibeg2] [, iend2] Initialization ifn -- function table number, typically denoting an sampled sound segment with prescribed looping points loaded using GEN01. The source file may be mono or stereo. ibas (optional) -- base frequency in Hz of the recorded sound. This optionally overrides the frequency given in the audio file, but is required if the file did not contain one. The default value is 261.626 Hz, i.e. middle C. (New in Csound 4.03). If this value is not known or not present, use 1 here and in kcps. imod1, imod2 (optional, default=-1) -- play modes for the sustain and release loops. A value of 1 denotes normal looping, 2 denotes forward & backward looping, 0 denotes no looping. The default value (-1) will defer to the mode and the looping points given in the source file. Make sure you select an appropriate mode if the file does not contain this information. ibeg1, iend1, ibeg2, iend2 (optional, dependent on mod1, mod2) -- begin and end points of the sustain and release loops. These are measured in sample frames from the beginning of the file, so will look the same whether the sound segment is monaural or stereo. If no loop points are specified, and a looping mode (imod1, imod2) is given, the file will be looped for the whole length. Performance ar1, ar2 -- the output at audio-rate. There is just ar1 for mono output. However, there is both ar1 and ar2 for stereo output. xamp -- the amplitude of the output signal. kcps -- the frequency of the output signal in cycles per second. loscil3 is identical to loscil except that it uses cubic interpolation. New in Csound version 3.50. Note to Windows users Windows users typically use back-slashes, \, when specifying the paths of their files. As an example, a Windows user might use the path c:\music\samples\loop001.wav. This is problematic because back-slashes are normally used to specify special characters. To correctly specify this path in Csound, one may alternately: Use forward slashes: c:/music/samples/loop001.wav Use back-slash special characters, \\: c:\\music\\samples\\loop001.wav Note This is mono loscil3: a1 loscil3 10000, 1, 1, 1, 1 ...and this is stereo loscil3: a1, a2 loscil3 10000, 1, 1, 1, 1 Examples Here is an example of the loscil3 opcode. It uses the file loscil3.csd, mary.wav and kickroll.wav. Example of the loscil3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also loscil and GEN01 Credits Note about the mono/stereo difference was contributed by &namerasmus;. manual_src~dfsg/opcodes/moogvcf2.xml0000644000000000000000000000721512262561504016652 0ustar rootroot moogvcf2 Signal Modifiers:Standard Filters:Resonant moogvcf2 moogvcf2 A digital emulation of the Moog diode ladder filter configuration. Description A digital emulation of the Moog diode ladder filter configuration. Syntax ares moogvcf2 asig, xfco, xres [,iscale, iskip] Initialization iscale (optional, default=0dBfs) -- internal scaling factor, as the operation of the code requires the signal to be in the range +/-1. Input is first divided by iscale, then output is mutliplied by iscale. iskip (optional, default=0) -- if non zero skip the initialisation of the filter. Performance asig -- input signal xfco -- filter cut-off frequency in Hz. which may be i-,k-, or a-rate. xres -- amount of resonance. Self-oscillation occurs when xres is approximately one. May be a-rate, i-rate, or k-rate. moogvcf2 is a digital emulation of the Moog diode ladder filter configuration. This emulation is based loosely on the paper Analyzing the Moog VCF with Considerations for Digital Implementation by Stilson and Smith (CCRMA). This version was originally coded in Csound by Josep Comajuncosas. Some modifications and conversion to C were done by &namehans; and then adjusted. moogvcf2 is identical to moogvcf, except that the iscale parameter defaults to 0dbfs instead of 0, guaranteeing that amplitude will usually be OK. Examples Here is an example of the moogvcf2 opcode. It uses the file moogvcf2.csd. Example of the moogvcf2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also moogvcf, biquad, rezzy Credits Author: &namehans; and &namejohn; October 1998/ July 2006 New in Csound version 5.03 manual_src~dfsg/opcodes/genarray_i.xml0000644000000000000000000000433212262561504017245 0ustar rootroot genarray_i Table Control:Table Queries genarray_i genarray_i Generate a vector with an arithmetic sequence. Description Generate a vector (one-dimensional k-rate) with an arithmetic sequence at initialisation time. Syntax karray genarray_i istart, iend [,inc] Initialization istart -- value to place in first element. iend -- last value to place in array. inc -- amount to add to previous value (default 1). Examples Here is an example of the genarray_i opcode. It uses the file genarray_i.csd. Example of the genarray_i opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also Vectorial opcodes fillarray Credits Author: &namejohn; Codemist Ltd 2013 New in version 6.00 manual_src~dfsg/opcodes/portk.xml0000644000000000000000000000600012262561504016256 0ustar rootroot portk Signal Modifiers:Standard Filters:Control portk portk Applies portamento to a step-valued control signal. Description Applies portamento to a step-valued control signal. Syntax kres portk ksig, khtim [, isig] Initialization isig (optional, default=0) -- initial (i.e. previous) value for internal feedback. The default value is 0. Performance kres -- the output signal at control-rate. ksig -- the input signal at control-rate. khtim -- half-time of the function in seconds. portk is like port except the half-time can be varied at the control rate. Examples Here is an example of the portk opcode. It uses the file portk.csd. Example of the portk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also areson, aresonk, atone, atonek, port, reson, resonk, tone, tonek Credits Author: Robin Whittle Australia May 1997 manual_src~dfsg/opcodes/alwayson.xml0000644000000000000000000000457712262561504016775 0ustar rootroot alwayson Signal Flow Graph Opcodes alwayson alwayson Activates the indicated instrument in the orchestra header, without need for an i statement. Description Activates the indicated instrument in the orchestra header, without need for an i statement. Instruments must be activated in the same order as they are defined. The alwayson opcode is designed to simplify the definition of re-usable orchestras with signal processing or effects chains and networks. Syntax alwayson Tinstrument [p4, ..., pn] Initialization Tinstrument -- String name of the instrument definition to be turned on. [p4, ..., pn] -- Optional pfields to be passed to the instrument, in the same order and type as if this were an i statement. When the instrument is activated, p1 is the insno, p2 is 0, and p3 is -1. Pfields from p4 on may optionally be sent to the instrument. Examples Here is an example of the alwayson opcode. It uses the file alwayson.csd. Example of the alwayson opcode. Credits By: &namemichael; 2009 manual_src~dfsg/opcodes/readscore.xml0000644000000000000000000000542312262561504017076 0ustar rootroot readscore Instrument Control:Invocation readscore readscore Read, preprocess and schedule a score from an input string. Description Readscore will issue one or more score events. It can handle strings in the same conditions as the standard score, including preprocessing (carry, sort, ramp, etc). Multi-line strings are accepted, using {{ }} to enclose the string. Syntax readscore Sin Initialization Sin -- a string (in double-quotes or enclosed by {{ }}) containing one or more score events. Examples Here is an example of the readscore opcode. It uses the file readscore.csd. Example of the readscore opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. You can use string opcodes like sprintfk to produce strings to be passed to readscore like this: Sfil = "/Volumes/Bla/file.aif" String sprintfk {{i 2 0 %f "%s" %f %f %f %f}}, idur, Sfil, p5, p6, knorm, iskip readscore String See also event, event_i, schedule, schedwhen, schedkwhen, schedkwhennamed Credits Author: &namevictor;, 2013 manual_src~dfsg/opcodes/trcross.xml0000644000000000000000000000660312262561504016627 0ustar rootroot trcross Spectral Processing:Streaming trcross trcross Streaming partial track cross-synthesis. Description The trcross opcode takes two inputs containg TRACKS pv streaming signals (as generated, for instance by partials) and cross-synthesises them into a single TRACKS stream. Two different modes of operation are used: mode 0, cross-synthesis by multiplication of the amplitudes of the two inputs and mode 1, cross-synthesis by the substititution of the amplitudes of input 1 by the input 2. Frequencies and phases of input 1 are preserved in the output. The cross-synthesis is done by matching tracks between the two inputs using a 'search interval'. The matching algorithm will look for tracks in the second input that are within the search interval around each track in the first input. This interval can be changed at the control rate. Wider search intervals will find more matches. Syntax fsig trcross fin1, fin2, ksearch, kdepth [, kmode] Performance fsig -- output pv stream in TRACKS format fin1 -- first input pv stream in TRACKS format. fin2 -- second input pv stream in TRACKS format ksearch -- search interval ratio, defining a 'search area' around each track of 1st input for matching purposes. kdepth -- depth of effect (0-1). kmode -- mode of cross-synthesis. 0, multiplication of amplitudes (filtering), 1, subsitution of amplitudes of input 1 by input 2 (akin to vocoding). Defaults to 0. Examples Here is an example of the trcross opcode. It uses the file trcross.csd. Example of the trcross opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of two ifd-analysis signals, cross-synthesis, followed by the remix of the two parts of the spectrum and resynthesis. Credits Author: &namevictor; February 2006 New in Csound 5.01 manual_src~dfsg/opcodes/wgbrass.xml0000644000000000000000000000635412262561504016603 0ustar rootroot wgbrass Signal Generators:Waveguide Physical Modeling wgbrass wgbrass Creates a tone related to a brass instrument. Description Audio output is a tone related to a brass instrument, using a physical model developed from &nameperry;, but re-coded for Csound. Syntax ares wgbrass kamp, kfreq, ktens, iatt, kvibf, kvamp, ifn [, iminfreq] Initialization iatt -- time taken to reach full pressure ifn -- table of shape of vibrato, usually a sine table, created by a function iminfreq -- lowest frequency at which the instrument will play. If it is omitted it is taken to be the same as the initial kfreq. If iminfreq is negative, initialization will be skipped. Performance A note is played on a brass-like instrument, with the arguments as below. kamp -- Amplitude of note. kfreq -- Frequency of note played. ktens -- lip tension of the player. Suggested value is about 0.4 kvibf -- frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp -- amplitude of the vibrato NOTE This is rather poor, and at present uncontrolled. Needs revision, and possibly more parameters. Examples Here is an example of the wgbrass opcode. It uses the file wgbrass.csd. Example of the wgbrass opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/rbjeq.xml0000644000000000000000000002006112262561504016225 0ustar rootroot rbjeq Signal Modifiers:Specialized Filters rbjeq rbjeq Parametric equalizer and filter opcode with 7 filter types, based on algorithm by Robert Bristow-Johnson. Description Parametric equalizer and filter opcode with 7 filter types, based on algorithm by Robert Bristow-Johnson. Syntax ar rbjeq asig, kfco, klvl, kQ, kS[, imode] Initialization imode ( optional, defaults to zero) - sum of: 1: skip initialization (should be used in tied, or re-initialized notes only) and exactly one of the following values to select filter type: 0: resonant lowpass filter. kQ controls the resonance: at the cutoff frequency (kfco), the amplitude gain is kQ (e.g. 20 dB for kQ = 10), and higher kQ values result in a narrower resonance peak. If kQ is set to sqrt(0.5) (about 0.7071), there is no resonance, and the filter has a response that is very similar to that of butterlp. If kQ is less than sqrt(0.5), there is no resonance, and the filter has a -6 dB / octave response from about kfco * kQ to kfco. Above kfco, there is always a -12 dB / octave cutoff. NOTE The rbjeq lowpass filter is basically the same as ar pareq asig, kfco, 0, kQ, 2 but is faster to calculate. 2: resonant highpass filter. The parameters are the same as for the lowpass filter, but the equivalent filter is butterhp if kQ is 0.7071, and "ar pareq asig, kfco, 0, kQ, 1" in other cases. 4: bandpass filter. kQ controls the bandwidth, which is kfco / kQ, and must be always less than sr / 2. The bandwidth is measured between -3 dB points (i.e. amplitude gain = 0.7071), beyond which there is a +/- 6 dB / octave slope. This filter type is very similar to ar butterbp asig, kfco, kfco / kQ. 6: band-reject filter, with the same parameters as the bandpass filter, and a response similar to that of butterbr. 8: peaking EQ. It has an amplitude gain of 1 (0 dB) at 0 Hz and sr / 2, and klvl at the center frequency (kfco). Thus, klvl controls the amount of boost (if it is greater than 1), or cut (if it is less than 1). Setting klvl to 1 results in a flat response. Similarly to the bandpass and band-reject filters, the bandwidth is determined by kfco / kQ (which must be less than sr / 2 again); however, this time it is between sqrt(klvl) points (or, in other words, half the boost or cut in decibels). NOTE: excessively low or high values of klvl should be avoided (especially with 32-bit floats), though the opcode was tested with klvl = 0.01 and klvl = 100. klvl = 0 is always an error, unlike in the case of pareq, which does allow a zero level. 10: low shelf EQ, controlled by klvl and kS (kQ is ignored by this filter type). There is an amplitude gain of klvl at zero frequency, while the level of high frequencies (around sr / 2) is not changed. At the corner frequency (kfco), the gain is sqrt(klvl) (half the boost or cut in decibels). The kS parameter controls the steepness of the slope of the frequency response (see below). 12: high shelf EQ. Very similar to the low shelf EQ, but affects the high frequency range. The default value for imode is zero (lowpass filter, initialization not skipped). Performance ar -- the output signal. asig -- the input signal NOTE If the input contains silent sections, on Intel CPUs a significant slowdown can occur due to denormals. In such cases, it is recommended to process the input signal with "denorm" opcode before filtering it with rbjeq (and actually many other filters). kfco -- cutoff, corner, or center frequency, depending on filter type, in Hz. It must be greater than zero, and less than sr / 2 (the range of about sr * 0.0002 to sr * 0.49 should be safe). klvl -- level (amount of boost or cut), as amplitude gain (e.g. 1: flat response, 4: 12 dB boost, 0.1: 20 dB cut); zero or negative values are not allowed. It is recognized by the peaking and shelving EQ types (8, 10, 12) only, and is ignored by other filters. kQ -- resonance (also kfco / bandwidth in many filter types). Not used by the shelving EQs (imode = 10 and 12). The exact meaning of this parameter depends on the filter type (see above), but it should be always greater than zero, and usually (kfco / kQ) less than sr / 2. kS -- shelf slope parameter for shelving filters. Must be greater than zero; a higher value means a steeper slope, with resonance if kS > 1 (however, a too high kS value may make the filter unstable). If kS is set to exactly 1, the shelf slope is as steep as possible without a resonance. Note that the effect of kS - especially if it is greater than 1 - also depends on klvl, and it does not have any well defined unit. Examples Here is an example of the rbjeq opcode. It uses the file rbjeq.csd. An example of the rbjeq opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Original algorithm by Robert Bristow-Johnson Csound orchestra version by Josep M Comajuncosas, Aug 1999 Converted to C (with optimizations and bug fixes) by Istvan Varga, Dec 2002 manual_src~dfsg/opcodes/pchmidib.xml0000644000000000000000000000626412262561504016712 0ustar rootroot pchmidib Real-time MIDI:Converters pchmidib pchmidib Get the note number of the current MIDI event and modify it by the current pitch-bend value, express it in pitch-class units. Description Get the note number of the current MIDI event and modify it by the current pitch-bend value, express it in pitch-class units. Syntax ipch pchmidib [irange] kpch pchmidib [irange] Initialization irange (optional) -- the pitch bend range in semitones Performance Get the note number of the current MIDI event, modify it by the current pitch-bend value, and express the result in pitch-class units. Available as an i-time value or as a continuous k-rate value. Examples Here is an example of the pchmidib pchmidib. It uses the file pchmidib.csd. Example of the pchmidib pchmidib. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also aftouch, ampmidi, cpsmidi, cpsmidib, midictrl, notnum, octmidi, octmidib, pchbend, pchmidi, veloc Credits Author: &namebarry; - &namemike; MIT - Mills May 1997 Example written by &namekevin;. manual_src~dfsg/opcodes/STKSimple.xml0000644000000000000000000000711712262561504016744 0ustar rootroot STKSimple Signal Generators:STK Opcodes STKSimple STKSimple STKSimple is a wavetable/noise instrument. Description STKSimple is a wavetable/noise instrument. It combines a looped wave, a noise source, a biquad resonance filter, a one-pole filter, and an ADSR envelope to create some interesting sounds. Syntax asignal STKSimple ifrequency, iamplitude, [kpos, kv1[, kcross, kv2[, kenv, kv3[, kgain, kv4]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kpos -- controller 2, position of filter pole. Value range of kv1 is 0-127. kcross -- controller 4, noise/pitched cross-fade. Value range of kv2 is 0-127. kenv -- controller 11, rate of envelope. Value range of kv3 is 0-127. kgain -- controller 128, gain. Value range of kv4 is 0-127. Note The code for this opcode is taken directly from the Simple class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKSimple opcode. It uses the file STKSimple.csd. Example of the STKSimple opcode. See Also STKClarinet. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/readfi.xml0000644000000000000000000000500712262561504016357 0ustar rootroot readfi Signal I/O:File I/O readfi readfi Read a line of text from an external file. Description Read a line of text from an external file once on initialisation. Syntax Sres, iline readfi ifilname Initialization ifilname -- an integer N denoting a file named "input.N" or a character string (in double quotes, spaces permitted) denoting the external file name. For a string, it may either be a full path name with directory specified or a simple filename. In the later case, the file is sought first in the current directory, then in SSDIR, and finally in SFDIR. iline -- line number, or -1 when end of file has been reached. Sres -- output of the line read from ifilname. This opcode allows a line of text to be read from a named external file. There may be any number of readfi opcodes in an instrument or orchestra but they read separately from the same or different files. Examples Here is an example of the readfi opcode. It uses the file readfi.csd. Example of the readfi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also readf. Credits John ffitch and Joachim Heintz 2012; new in 5.17.12 manual_src~dfsg/opcodes/pvoc.xml0000644000000000000000000001360012262561504016072 0ustar rootroot pvoc Spectral Processing:STFT pvoc pvoc Implements signal reconstruction using an fft-based phase vocoder. Description Implements signal reconstruction using an fft-based phase vocoder. Syntax ares pvoc ktimpnt, kfmod, ifilcod [, ispecwp] [, iextractmode] \ [, ifreqlim] [, igatefn] Initialization ifilcod -- integer or character-string denoting a control-file derived from analysis of an audio signal. An integer denotes the suffix of a file pvoc.m; a character-string (in double quotes) gives a filename, optionally a full pathname. If not fullpath, the file is sought first in the current directory, then in the one given by the environment variable SADIR (if defined). pvoc control contains breakpoint amplitude and frequency envelope values organized for fft resynthesis. Memory usage depends on the size of the files involved, which are read and held entirely in memory during computation but are shared by multiple calls (see also lpread). ispecwp (optional) -- if non-zero, attempts to preserve the spectral envelope while its frequency content is varied by kfmod. The default value is zero. iextractmode (optional) -- determines if spectral extraction will be carried out and if so whether components that have changes in frequency below ifreqlim or above ifreqlim will be discarded. A value for iextractmode of 1 will cause pvoc to synthesize only those components where the frequency difference between analysis frames is greater than ifreqlim. A value of 2 for iextractmode will cause pvoc to synthesize only those components where the frequency difference between frames is less than ifreqlim. The default values for iextractmode and ifreqlim are 0, in which case a simple resynthesis will be done. See examples under pvadd for how to use spectral extraction. igatefn (optional) -- the number of a stored function which will be applied to the amplitudes of the analysis bins before resynthesis takes place. If igatefn is greater than 0 the amplitudes of each bin will be scaled by igatefn through a simple mapping process. First, the amplitudes of all of the bins in all of the frames in the entire analysis file are compared to determine the maximum amplitude value. This value is then used create normalized amplitudes as indeces into the stored function igatefn. The maximum amplitude will map to the last point in the function. An amplitude of 0 will map to the first point in the function. Values between 0 and 1 will map accordingly to points along the function table. See examples under pvadd for how to use amplitude gating. Performance ktimpnt -- The passage of time, in seconds, through the analysis file. ktimpnt must always be positive, but can move forwards or backwards in time, be stationary or discontinuous, as a pointer into the analysis file. kfmod -- a control-rate transposition factor: a value of 1 incurs no transposition, 1.5 transposes up a perfect fifth, and .5 down an octave. pvoc implements signal reconstruction using an fft-based phase vocoder. The control data stems from a precomputed analysis file with a known frame rate. This implementation of pvoc was orignally written by Dan Ellis. It is based in part on the system of Mark Dolson, but the pre-analysis concept is new. The spectral extraction and amplitude gating (new in Csound version 3.56) were added by Richard Karpen based on functions in SoundHack by Tom Erbe. Examples Here is an example of the pvoc opcode. It uses the file pvoc.csd. Example of the pvoc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vpvoc, PVANAL. Credits Authors: Dan Ellis and Richard Karpen Seattle, Wash 1997 manual_src~dfsg/opcodes/bformenc.xml0000644000000000000000000000601612262561504016721 0ustar rootroot bformenc Signal Modifiers:Panning and Spatialization bformenc bformenc Deprecated. Codes a signal into the ambisonic B format. Description Codes a signal into the ambisonic B format. Note that this opcode is deprecated as it is inaccurate, and is replaced by the much better opcode bformenc1 which replicates all the important features; also note that the gain arguments are not available in bformenc1. Syntax aw, ax, ay, az bformenc asig, kalpha, kbeta, kord0, kord1 aw, ax, ay, az, ar, as, at, au, av bformenc asig, kalpha, kbeta, \ kord0, kord1 , kord2 aw, ax, ay, az, ar, as, at, au, av, ak, al, am, an, ao, ap, aq bformenc \ asig, kalpha, kbeta, kord0, kord1, kord2, kord3 Performance aw, ax, ay, ... -- output cells of the B format. asig -- input signal. kalpha –- azimuth angle in degrees (clockwise). kbeta -- altitude angle in degrees. kord0 -- linear gain of the zero order B format. kord1 -- linear gain of the first order B format. kord2 -- linear gain of the second order B format. kord3 -- linear gain of the third order B format. Example Here is an example of the bformenc opcode. It uses the file bformenc.csd. Example of the bformenc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Samuel Groner 2005 New in version 5.07. Deprecated in 5.09. manual_src~dfsg/opcodes/flhide.xml0000644000000000000000000000517312262561504016364 0ustar rootroot FLhide FLTK:Appearance FLhide FLhide Hides the target FLTK widget. Description Hides the target FLTK widget, making it invisible. Syntax FLhide ihandle Initialization ihandle -- a handle value (an integer number) that unequivocally references a corresponding widget. This is used by other opcodes that modify a widget's properties (see Modifying FLTK Widget Appearance). Performance FLhide hides target widget, making it invisible. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/denorm.xml0000644000000000000000000000515412262561504016414 0ustar rootroot denorm Signal Modifiers:Sample Level Operators denorm denorm Mixes low level noise to a list of a-rate signals Description Mixes low level (~1e-20 for floats, and ~1e-56 for doubles) noise to a list of a-rate signals. Can be used before IIR filters and reverbs to avoid denormalized numbers which may otherwise result in significantly increased CPU usage. Syntax denorm a1[, a2[, a3[, ... ]]] Performance a1[, a2[, a3[, ... ]]] -- signals to mix noise with Some processor architectures (particularly Pentium IVs) are very slow at processing extremely small numbers. These small numbers can appear as a result of some decaying feedback process like reverb and IIR filters. Low level noise can be added so that very small numbers are never reached, and they are 'absorbed' by this 'noise floor'. If CPU usage goes to 100% at the end of reverb tails, or you get audio glitches in processes that shouldn't use too much CPU, using denorm before the culprit opcode or process might solve the problem. Examples Here is an example of the denorm opcode. It uses the file denorm.csd. Example of the denorm opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/bformdec.xml0000644000000000000000000000761712262561504016717 0ustar rootroot bformdec Signal Modifiers:Panning and Spatialization bformdec bformdec Deprecated. Decodes an ambisonic B format signal. Description Decodes an ambisonic B format signal into loudspeaker specific signals. Note that this opcode is deprecated as it is inaccurate, and is replaced by the much better opcode bformdec1 which replicates all the important features. Syntax ao1, ao2 bformdec isetup, aw, ax, ay, az [, ar, as, at, au, av \ [, abk, al, am, an, ao, ap, aq]] ao1, ao2, ao3, ao4 bformdec isetup, aw, ax, ay, az [, ar, as, at, \ au, av [, abk, al, am, an, ao, ap, aq]] ao1, ao2, ao3, ao4, ao5 bformdec isetup, aw, ax, ay, az [, ar, as, \ at, au, av [, abk, al, am, an, ao, ap, aq]] ao1, ao2, ao3, ao4, ao5, ao6, ao7, ao8 bformdec isetup, aw, ax, ay, az \ [, ar, as, at, au, av [, abk, al, am, an, ao, ap, aq]]] Initialization isetup –- loudspeaker setup. There are five supported setups: 1 denotes stereo setup. There must be two output cells with loudspeaker positions assumed to be (330/0, 30/0). 2 denotes quad setup. There must be four output cells. Loudspeaker positions assumed to be (45°/0), (135°/0), (225/0), (315/0). 3 is a 5.1 surround setup. There must be five output cells. LFE channel is not supported. Loudspeaker positions assumed to be (330/0), (30/0), (0/0), (250/0), (110/0). 4 denotes eight loudspeaker circle setup. There must be eight output cells. Loudspeaker positions assumed to be (22.5/0), (67.5/0), (112.5/0), (157.5/0), (202.5/0), (247.5/0), (292.5/0), (337.5/0). 5 means an eight loudspeaker cubic setup. There must be eight output cells. Loudspeaker positions assumed to be (45/0), (45/30), (135/0), (135/30), (225/0), (225/30), (315/0), (315/30). Performance aw, ax, ay, ... -- input signal in the B format. ao1 .. ao8 -– loudspeaker specific output signals. Example Here is an example of the bformdec opcode. It uses the file bformenc.csd. Example of the bformdec opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Samuel Groner 2005 New in version 5.07. Deprecated in 5.09 manual_src~dfsg/opcodes/tableimix.xml0000644000000000000000000001240312262561504017101 0ustar rootroot tableimix Table Control:Read/Write Opreations tableimix tableimix Mixes two tables. Description Mixes two tables. Syntax tableimix idft, idoff, ilen, is1ft, is1off, is1g, is2ft, is2off, is2g Initialization idft -- Destination function table. idoff -- Offset to start writing from. Can be negative. ilen -- Number of write operations to perform. Negative means work backwards. is1ft, is2ft -- Source function tables. These can be the same as the destination table, if care is exercised about direction of copying data. is1off, is2off -- Offsets to start reading from in source tables. is1g, is2g -- Gains to apply when reading from the source tables. The results are added and the sum is written to the destination table. Performance tableimix -- This opcode mixes from two tables, with separate gains into the destination table. Writing is done for ilen locations, usually stepping forward through the table - if ilen is positive. If it is negative, then the writing and reading order is backwards - towards lower indexes in the tables. This bi-directional option makes it easy to shift the contents of a table sideways by reading from it and writing back to it with a different offset. If ilen is 0, no writing occurs. Note that the internal integer value of ilen is derived from the ANSI C floor() function - which returns the next most negative integer. Hence a fractional negative ilen value of -2.3 would create an internal length of 3, and cause the copying to start from the offset locations and proceed for two locations to the left. The total index for table reading and writing is calculated from the starting offset for each table, plus the index value, which starts at 0 and then increments (or decrements) by 1 as mixing proceeds. These total indexes can potentially be very large, since there is no restriction on the offset or the ilen. However each total index for each table is ANDed with a length mask (such as 0000 0111 for a table of length 8) to form a final index which is actually used for reading or writing. So no reading or writing can occur outside the tables. This is the same as wrap mode in table read and write. These opcodes do not read or write the guardpoint. If a table has been rewritten with one of these, then if it has a guardpoint which is supposed to contain the same value as the location 0, then call tableigpw afterwards. The indexes and offsets are all in table steps - they are not normalized to 0 - 1. So for a table of length 256, ilen should be set to 256 if all the table was to be read or written. The tables do not need to be the same length - wrapping occurs individually for each table. Examples Here is an example of the tableimix opcode. It uses the file tableimix.csd. Example of the tableimix opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also tablecopy, tablegpw, tablemix, tableicopy, tableigpw Credits Author: Robin Whittle Australia May 1997 New in version 3.47 manual_src~dfsg/opcodes/midichn.xml0000644000000000000000000000577312262561504016552 0ustar rootroot midichn Real-time MIDI:MIDI/Score Interoperability midichn midichn Returns the MIDI channel number from which the note was activated. Description midichn returns the MIDI channel number (1 - 16) from which the note was activated. In the case of score notes, it returns 0. Syntax ichn midichn Initialization ichn -- channel number. If the current note was activated from score, it is set to zero. Examples Here is a simple example of the midichn opcode. It uses the file midichn.csd. Example of the midichn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an advanced example of the midichn opcode. It uses the file midichn_advanced.csd. Don't forget that you must include the -F flag when using an external MIDI file like midichn_advanced.mid. An advanced example of the midichn opcode. Its output should include lines like: note 1 (time = 0.00) was activated from channel 1 note 2 (time = 2.00) was activated from channel 4 note 3 (time = 3.00) was activated from channel 2 note 4 (time = 5.00) was activated from channel 3 See Also pgmassign Credits Author: &nameistvan; May 2002 New in version 4.20 manual_src~dfsg/opcodes/filebit.xml0000644000000000000000000000577312262561504016555 0ustar rootroot filebit Signal I/O:Soundfile Queries filebit filebit Returns the number of bits in each sample in a sound file. Description Returns the number of bits in each sample in a sound file. Syntax ir filebit ifilcod [, iallowraw] Initialization ifilcod -- sound file to be queried iallowraw -- (Optional) Allow raw sound files (default=1) Performance filebit returns the number of bits in each sample in the sound file ifilcod. In the case of floating point samples the value -1 is returned for floats and -2 for doubles. For non-PCM formats the value is negative, and based on libsndfile's format encoding. Examples Here is an example of the filebit opcode. It uses the file filebit.csd, and mary.wav. Example of the filebit opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The audio file mary.wav is in monoaural CD format, so filebit's output should include a line like this: instr 1: ibits = 16.000 See Also filelen, filenchnls, filepeak, filesr Credits Author: &namevictor; July 1999 Example written by &namejohn;. New in Csound version 5.11 manual_src~dfsg/opcodes/lposcila.xml0000644000000000000000000000556712262561504016746 0ustar rootroot lposcila Signal Generators:Sample Playback lposcila lposcila Read sampled sound from a table with looping and high precision. Description lposcila reads sampled sound from a table with looping and high precision. Syntax ar lposcila aamp, kfreqratio, kloop, kend, ift [,iphs] Initialization ift -- function table number iphs -- initial phase (in samples) Performance ar -- output signal aamp -- amplitude kfreqratio -- multiply factor of table frequency (for example: 1 = original frequency, 1.5 = a fifth up , .5 = an octave down) kloop -- start loop point (in samples) kend -- end loop point (in samples) lposcila is the same as lposcil, but has an audio-rate amplitude argument (instead of k-rate) to allow fast envelope transients. Examples Here is an example of the lposcila opcode. It uses the file lposcila.csd. Example of the lposcila opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also lposcil, lposcilsa, lposcilsa2 Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/iexprand.xml0000644000000000000000000000077612262561504016747 0ustar rootroot iexprand iexprand iexprand Deprecated. Description Deprecated as of version 3.49. Use the exprand opcode instead. manual_src~dfsg/opcodes/butterlp.xml0000644000000000000000000000565312262561504016775 0ustar rootroot butterlp Signal Modifiers:Standard Filters butterlp butterlp A low-pass Butterworth filter. Description Implementation of a second-order low-pass Butterworth filter. This opcode can also be written as butlp. Syntax ares butterlp asig, kfreq [, iskip] ares butterlp asig, afreq [, iskip] Initialization iskip (optional, default=0) -- Skip initialization if present and non-zero. Performance These filters are Butterworth second-order IIR filters. They are slightly slower than the original filters in Csound, but they offer an almost flat passband and very good precision and stopband attenuation. asig -- Input signal to be filtered. kfreq/afreq -- Cutoff or center frequency for each of the filters. Examples Here is an example of the butterlp opcode. It uses the file butterlp.csd. Example of the butterlp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also butterbp, butterbr, butterhp Credits Author: &nameparis; MIT, Cambridge 1995 Existed in 3.30 Audio rate parameters introduced in version 6.02 October 2013. manual_src~dfsg/opcodes/acauchy.xml0000644000000000000000000000076712262561504016552 0ustar rootroot acauchy acauchy acauchy Deprecated. Description Deprecated as of version 3.49. Use the cauchy opcode instead. manual_src~dfsg/opcodes/tabmorphak.xml0000644000000000000000000001303512262561504017255 0ustar rootroot tabmorphak Table Control:Read/Write Opreations tabmorphak tabmorphak Allow morphing between a set of tables at audio rate with interpolation. Description tabmorphak allows morphing between a set of tables of the same size, by means of a weighted average between two currently selected tables. Syntax aout tabmorphak aindex, kweightpoint, ktabnum1, ktabnum2, \ ifn1, ifn2 [, ifn3, ifn4, ... ifnN] Initialization ifn1, ifn2 , ifn3, ifn4, ... ifnN - function table numbers. This is a set of chosen tables the user want to use in the morphing. All tables must have the same length. Be aware that only two of these tables can be chosen for the morphing at one time. Since it is possible to use non-integer numbers for the ktabnum1 and ktabnum2 arguments, the morphing is the result from the interpolation between adjacent consecutive tables of the set. Performance aout - The output value for index aindex, resulting from morphing two tables (see below). aindex - main index index of the morphed resultant table. The range is 0 to table_length (not included). kweightpoint - the weight of the influence of a pair of selected tables in the morphing. The range of this argument is 0 to 1. A zero makes it output the first table unaltered, a 1 makes it output the second table of the pair unaltered. All intermediate values between 0 and 1 determine the gradual morphing between the two tables of the pair. ktabnum1 - the first table chosen for the morphing. This number doesn’t express the table number directly, but the position of the table in the set sequence (starting from 0 to N-1). If this number is an integer, the corresponding table will be chosen unaltered. If it contains fractional values, then an interpolation with the next adjacent table will result. ktabnum2 - the second table chosen for the morphing. This number doesn’t express the table number directly, but the position of the table in the set sequence (starting from 0 to N-1). If this number is an integer, corresponding table will be chosen unaltered. If it contains fractional values, then an interpolation with the next adjacent table will result. The tabmorphak family of opcodes is similar to the table family, but allows morphing between two tables chosen into a set of tables. Firstly the user has to provide a set of tables of equal length (ifn2 [, ifn3, ifn4, ... ifnN]). Then he can choose a pair of tables in the set in order to perform the morphing: ktabnum1 and ktabnum2 are filled with numbers (zero represents the first table in the set, 1 the second, 2 the third and so on). Then determine the morphing between the two chosen tables with the kweightpoint parameter. After that the resulting table can be indexed with the aindex parameter like a normal table opcode. If the value of this parameter surpasses the length of tables (which must be the same for all tables), then it is wrapped around. tabmorphak works at a-rate, but kweightpoint, ktabnum1 and ktabnum2 are working at k-rate, making it more efficient than tabmorpha, since there are less calculations. Except the rate of these three arguments, it is identical to tabmorpha. Examples Here is an example of the tabmorphak opcode. It uses the file tabmorphak.csd. Example of the tabmorphak opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also table, tabmorph, tabmorphi, tabmorpha, ftmorf, Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/icauchy.xml0000644000000000000000000000076712262561504016562 0ustar rootroot icauchy icauchy icauchy Deprecated. Description Deprecated as of version 3.49. Use the cauchy opcode instead. manual_src~dfsg/opcodes/bbcutm.xml0000644000000000000000000002724312262561504016407 0ustar rootroot bbcutm Signal Generators:Sample Playback bbcutm bbcutm Generates breakbeat-style cut-ups of a mono audio stream. Description The BreakBeat Cutter automatically generates cut-ups of a source audio stream in the style of drum and bass/jungle breakbeat manipulations. There are two versions, for mono (bbcutm) or stereo (bbcuts) sources. Whilst originally based on breakbeat cutting, the opcode can be applied to any type of source audio. The prototypical cut sequence favoured over one bar with eighth note subdivisions would be 3+ 3R + 2 where we take a 3 unit block from the source's start, repeat it, then 2 units from the 7th and 8th eighth notes of the source. We talk of rendering phrases (a sequence of cuts before reaching a new phrase at the beginning of a bar) and units (as subdivision th notes). The opcode comes most alive when multiple synchronised versions are used simultaneously. Syntax a1 bbcutm asource, ibps, isubdiv, ibarlength, iphrasebars, inumrepeats \ [, istutterspeed] [, istutterchance] [, ienvchoice ] Initialization ibps -- Tempo to cut at, in beats per second. isubdiv -- Subdivisions unit, for a bar. So 8 is eighth notes (of a 4/4 bar). ibarlength -- How many beats per bar. Set to 4 for default 4/4 bar behaviour. iphrasebars -- The output cuts are generated in phrases, each phrase is up to iphrasebars long inumrepeats -- In normal use the algorithm would allow up to one additional repeat of a given cut at a time. This parameter allows that to be changed. Value 1 is normal- up to one extra repeat. 0 would avoid repeating, and you would always get back the original source except for enveloping and stuttering. istutterspeed -- (optional, default=1) The stutter can be an integer multiple of the subdivision speed. For instance, if subdiv is 8 (quavers) and stutterspeed is 2, then the stutter is in semiquavers (sixteenth notes= subdiv 16). The default is 1. istutterchance -- (optional, default=0) The tail of a phrase has this chance of becoming a single repeating one unit cell stutter (0.0 to 1.0). The default is 0. ienvchoice -- (optional, default=1) choose 1 for on (exponential envelope for cut grains) or 0 for off. Off will cause clicking, but may give good noisy results, especially for percussive sources. The default is 1, on. Performance asource -- The audio signal to be cut up. This version runs in real-time without knowledge of future audio. Examples Here is a simple example of the bbcutm opcode. It uses the file bbcutm.csd, and beats.wav. A simple example of the bbcutm opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here are some more advanced examples... First steps- mono and stereo versions <CsoundSynthesizer> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 asource diskin "break7.wav",1,0,1 ; a source breakbeat sample, wraparound lest it stop! ; cuts in eighth notes per 4/4 bar, up to 4 bar phrases, up to 1 ; repeat in total (standard use) rare stuttering at 16 note speed, ; no enveloping asig bbcutm asource, 2.6937, 8, 4, 4, 1, 2, 0.1, 0 outs asig,asig endin instr 2 ;stereo version asource1,asource2 diskin "break7stereo.wav", 1, 0, 1 ; a source breakbeat sample, wraparound lest it stop! ; cuts in eighth notes per 4/4 bar, up to 4 bar phrases, up to 1 ; repeat in total (standard use) rare stuttering at 16 note speed, ; no enveloping asig1,asig2 bbcuts asource1, asource2, 2.6937, 8, 4, 4, 1, 2, 0.1, 0 outs asig1,asig2 endin </CsInstruments> <CsScore> i1 0 10 i2 11 10 e </CsScore> </CsoundSynthesizer> Multiple simultaneous synchronised breaks <CsoundSynthesizer> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 ibps = 2.6937 iplaybackspeed = ibps/p5 asource diskin p4, iplaybackspeed, 0, 1 asig bbcutm asource, 2.6937, p6, 4, 4, p7, 2, 0.1, 1 out asig endin </CsInstruments> <CsScore> ; source bps cut repeats i1 0 10 "break1.wav" 2.3 8 2 //2.3 is the source original tempo i1 0 10 "break2.wav" 2.4 8 3 i1 0 10 "break3.wav" 2.5 16 4 e </CsScore> </CsoundSynthesizer> Cutting up any old audio- much more interesting noises than this should be possible! <CsoundSynthesizer> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 asource oscil 20000, 70, 1 ; ain, bps, subdiv, barlength, phrasebars, numrepeats, ;stutterspeed, stutterchance, envelopingon asig bbcutm asource, 2, 32, 1, 1, 2, 4, 0.6, 1 outs asig endin </CsInstruments> <CsScore> f1 0 256 10 1 i1 0 10 e </CsScore> </CsoundSynthesizer> Constant stuttering- faked, not possible since can only stutter in last half bar could make extra stuttering option parameter <CsoundSynthesizer> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 asource diskin "break7.wav", 1, 0, 1 ;16th note cuts- but cut size 2 over half a beat. ;each half beat will either survive intact or be turned into ;the first sixteenth played twice in succession asig bbcutm asource, 2.6937, 2, 0.5, 1, 2, 2, 1.0, 0 outs asig endin </CsInstruments> <CsScore> i1 0 30 e </CsScore> </CsoundSynthesizer> See Also bbcuts Credits Author: Nick Collins London August 2001 New in version 4.13 manual_src~dfsg/opcodes/marimba.xml0000644000000000000000000000756212262561504016545 0ustar rootroot marimba Signal Generators:Models and Emulations marimba marimba Physical model related to the striking of a wooden block. Description Audio output is a tone related to the striking of a wooden block as found in a marimba. The method is a physical model developed from &nameperry; but re-coded for Csound. Syntax ares marimba kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec \ [, idoubles] [, itriples] Initialization ihrd -- the hardness of the stick used in the strike. A range of 0 to 1 is used. 0.5 is a suitable value. ipos -- where the block is hit, in the range 0 to 1. imp -- a table of the strike impulses. The file marmstk1.wav is a suitable function from measurements and can be loaded with a GEN01 table. It is also available at ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. ivfn -- shape of vibrato, usually a sine table, created by a function idec -- time before end of note when damping is introduced idoubles (optional) -- percentage of double strikes. Default is 40%. itriples (optional) -- percentage of triple strikes. Default is 20%. Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. kvibf -- frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp -- amplitude of the vibrato Examples Here is an example of the marimba opcode. It uses the file marimba.csd, and marmstk1.wav. Example of the marimba opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vibes Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/fillarray.xml0000644000000000000000000000376512262561504017123 0ustar rootroot fillarray Table Control:Table Queries fillarray fillarray Generate a vector with initial values. Description Generate a vector (one-dimensional k-rate array) with a sequence of numeric values. Syntax karray[] fillarray ival1, ival2,.....ivaln Initialization ival1,...ivaln -- values to place in the vector. Examples Here is an example of the fillarray opcode. It uses the file fillarray.csd. Example of the fillarray opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also Vectorial opcodes array Credits Author: &namejohn; Codemist Ltd 2013 New in version 6.00 manual_src~dfsg/opcodes/copya2ftab.xml0000644000000000000000000000401412262561504017154 0ustar rootroot copya2ftab Array Opcodes copya2ftab copya2ftab Copy data from a vector to an f-table. Description The copya2ftab opcode takes a t-var and copies the contents to an f-table. Syntax copya2ftab tab, kftbl Performance tab -- tables for source. kftbl -- f-tables for destination. Example Here is an example of the copya2ftab opcode. It uses the file copya2ftab.csd. Example of the copya2ftab opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also copyf2array Credits Author: &namejohn; October 2011 New in Csound version 5.15 Renamed in Csound version 6.00 manual_src~dfsg/opcodes/flsetalign.xml0000644000000000000000000000670012262561504017256 0ustar rootroot FLsetAlign FLTK:Appearance FLsetAlign FLsetAlign Sets the text alignment of a label of a FLTK widget. Description FLsetAlign sets the text alignment of the label of the target widget. Syntax FLsetAlign ialign, ihandle Initialization ialign -- sets the alignment of the label text of widgets. The legal values for the ialign argument are: 1 - align center 2 - align top 3 - align bottom 4 - align left 5 - align right 6 - align top-left 7 - align top-right 8 - align bottom-left 9 - align bottom-right ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/fljoy.xml0000644000000000000000000001474612262561504016262 0ustar rootroot FLjoy FLTK:Valuators FLjoy FLjoy A FLTK opcode that acts like a joystick. Description FLjoy is a squared area that allows the user to modify two output values at the same time. It acts like a joystick. Syntax koutx, kouty, ihandlex, ihandley FLjoy "label", iminx, imaxx, iminy, \ imaxy, iexpx, iexpy, idispx, idispy, iwidth, iheight, ix, iy Initialization ihandlex -- a handle value (an integer number) that unequivocally references a corresponding widget. Used by further opcodes that changes some valuator's properties. It is automatically set by the corresponding valuator. ihandley -- a handle value (an integer number) that unequivocally references a corresponding widget. Used by further opcodes that changes some valuator's properties. It is automatically set by the corresponding valuator. label -- a double-quoted string containing some user-provided text, placed near the corresponding widget. iminx -- minimum x value of output range imaxx -- maximum x value of output range iminy -- minimum y value of output range imaxy -- maximum y value of output range iwidth -- width of widget. idispx -- a handle value that was output from a previous instance of the FLvalue opcode to display the current value of the current valuator in the FLvalue widget itself. If the user doesn't want to use this feature that displays current values, it must be set to a negative number by the user. idispy -- a handle value that was output from a previous instance of the FLvalue opcode to display the current value of the current valuator in the FLvalue widget itself. If the user doesn't want to use this feature that displays current values, it must be set to a negative number by the user. iexpx -- an integer number denoting the behaviour of valuator: 0 = valuator output is linear -1 = valuator output is exponential All other positive numbers for iexpx indicate the number of an existing table that is used for indexing. Linear interpolation is provided in table indexing. A negative table number suppresses interpolation. iexpy -- an integer number denoting the behaviour of valuator: 0 = valuator output is linear -1 = valuator output is exponential All other positive numbers for iexpy indicate the number of an existing table that is used for indexing. Linear interpolation is provided in table indexing. A negative table number suppresses interpolation. IMPORTANT! Notice that the tables used by valuators must be created with the ftgen opcode and placed in the orchestra before the corresponding valuator. They can not placed in the score. In fact, tables placed in the score are created later than the initialization of the opcodes placed in the header section of the orchestra. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). Performance koutx -- x output value kouty -- y output value Examples Here is an example of the FLjoy opcode. It uses the file FLjoy.csd. Example of the FLjoy opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLcount, FLknob, FLroller, FLslider, FLtext Credits Author: &namegabriel; New in version 4.22 Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/transegr.xml0000644000000000000000000000753112262561504016756 0ustar rootroot transegr Signal Generators:Linear and Exponential Generators transegr transegr Constructs a user-definable envelope with extended release segment. Description Constructs a user-definable envelope. It is the same as transeg, with an extended release segment. Syntax ares transegr ia, idur, itype, ib [, idur2] [, itype] [, ic] ... kres transegr ia, idur, itype, ib [, idur2] [, itype] [, ic] ... Initialization ia -- starting value. ib, ic, etc. -- value after idur seconds. idur -- duration in seconds of first segment. A zero or negative value will cause all initialization to be skipped. idur2,... idurx etc. -- duration in seconds of segment itype, itype2, etc. -- if 0, a straight line is produced. If non-zero, then transegr creates the following curve, for n steps: ibeg + (ivalue - ibeg) * (1 - exp( i*itype/(n-1) )) / (1 - exp(itype)) Performance If itype > 0, there is a slowly rising (concave) or slowly decaying (convex) curve, while if itype < 0, the curve is fast rising (convex) or fast decaying (concave). See also GEN16. This opcode is the same as of transeg with an additional release segment triggered by a MIDI noteoff event, a negative p1 note event in the score or a turnoff2 opcode. Examples Here is an example of the transegr opcode. It uses the file transegr.csd. Example of the transegr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also expsega, expsegr, linseg, linsegr, transeg Credits Author: &namejohn; january 2010 New in Csound version 5.12 manual_src~dfsg/opcodes/iunirand.xml0000644000000000000000000000077612262561504016746 0ustar rootroot iunirand iunirand iunirand Deprecated. Description Deprecated as of version 3.49. Use the unirand opcode instead. manual_src~dfsg/opcodes/pvsmaska.xml0000644000000000000000000000671112262561504016755 0ustar rootroot pvsmaska Spectral Processing:Streaming pvsmaska pvsmaska Modify amplitudes using a function table, with dynamic scaling. Description Modify amplitudes of fsrc using function table, with dynamic scaling. Syntax fsig pvsmaska fsrc, ifn, kdepth Initialization ifn -- The f-table to use. Given fsrc has N analysis bins, table ifn must be of size N or larger. The table need not be normalized, but values should lie within the range 0 to 1. It can be supplied from the score in the usual way, or from within the orchestra by using pvsinfo to find the size of fsrc, (returned by pvsinfo in inbins), which can then be passed to ftgen to create the f-table. Performance kdepth -- Controls the degree of modification applied to fsrc, using simple linear scaling. 0 leaves amplitudes unchanged, 1 applies the full profile of ifn. Note that power-of-two FFT sizes are particularly convenient when using table-based processing, as the number of analysis bins (inbins) is then a power-of-two plus one, for which an exactly matching f-table can be created. In this case it is important that the f-table be created with a size of inbins, rather than as a power of two, as the latter will copy the first table value to the guard point, which is inappropriate for this opcode. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the use of the pvsmaska opcode. It uses the file pvsmaska.csd. Example of the <emphasis>pvsmaska</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namerichard; August 2001 New in version 4.13 manual_src~dfsg/opcodes/flroller.xml0000644000000000000000000001434712262561504016755 0ustar rootroot FLroller FLTK:Valuators FLroller FLroller A FLTK widget that creates a transversal knob. Description FLroller is a sort of knob, but put transversally. Syntax kout, ihandle FLroller "label", imin, imax, istep, iexp, itype, idisp, \ iwidth, iheight, ix, iy Initialization ihandle -- a handle value (an integer number) that unequivocally references a corresponding widget. This is used by other opcodes that modify a widget's properties (see Modifying FLTK Widget Appearance). It is automatically output by FLroller and must not be set by the user label. (The user label is a double-quoted string containing some user-provided text placed near the widget.) label -- a double-quoted string containing some user-provided text, placed near the corresponding widget. imin -- minimum value of output range. imax -- maximum value of output range. istep -- a floating-point number indicating the increment of valuator value corresponding to of each mouse click. The istep argument allows the user to arbitrarily slow roller's motion, enabling arbitrary precision. iexp -- an integer number denoting the behaviour of valuator: 0 = valuator output is linear -1 = valuator output is exponential All other positive numbers for iexp indicate the number of an existing table that is used for indexing. Linear interpolation is provided in table indexing. A negative table number suppresses interpolation. IMPORTANT! Notice that the tables used by valuators must be created with the ftgen opcode and placed in the orchestra before the corresponding valuator. They can not placed in the score. In fact, tables placed in the score are created later than the initialization of the opcodes placed in the header section of the orchestra. itype -- an integer number denoting the appearance of the valuator. The itype argument can be set to the following values: 1 - horizontal roller 2 - vertical roller idisp -- a handle value that was output from a previous instance of the FLvalue opcode to display the current value of the current valuator in the FLvalue widget itself. If the user doesn't want to use this feature that displays current values, it must be set to a negative number by the user. iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). Performance kout -- output value FLroller is a sort of knob, but put transversally: FLroller. FLroller. Examples Here is an example of the FLroller opcode. It uses the file FLroller.csd. Example of the FLroller opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLcount, FLjoy, FLknob, FLslider, FLtext Credits Author: &namegabriel; New in version 4.22 Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/iweibull.xml0000644000000000000000000000077612262561504016751 0ustar rootroot iweibull iweibull iweibull Deprecated. Description Deprecated as of version 3.49. Use the weibull opcode instead. manual_src~dfsg/opcodes/vdelayk.xml0000644000000000000000000000365112262561504016567 0ustar rootroot vdelayk Vectorial:Delay Paths vdelayk vdelayk k-rate variable time delay. Description Variable delay applied to a k-rate signal Syntax kout vdelayk ksig, kdel, imaxdel [, iskip, imode] Initialization imaxdel - maximum value of delay in seconds. iskip (optional) - Skip initialization if present and non zero. imode (optional) - if non-zero it suppresses linear interpolation. While, normally, interpolation increases the quality of a signal, it should be suppressed if using vdelay with discrete control signals, such as, for example, trigger signals. Performance kout - delayed output signal ksig - input signal kdel - delay time in seconds can be varied at k-rate vdelayk is similar to vdelay, but works at k-rate. It is designed to delay control signals, to be used, for example, in algorithmic composition. Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/dripwater.xml0000644000000000000000000001054512262561504017131 0ustar rootroot dripwater Signal Generators:Models and Emulations dripwater dripwater Semi-physical model of a water drop. Description dripwater is a semi-physical model of a water drop. It is one of the PhISEM percussion opcodes. PhISEM (Physically Informed Stochastic Event Modeling) is an algorithmic approach for simulating collisions of multiple independent sound producing objects. Syntax ares dripwater kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ [, ifreq1] [, ifreq2] Initialization idettack -- period of time over which all sound is stopped inum (optional) -- The number of beads, teeth, bells, timbrels, etc. If zero, the default value is 10. idamp (optional) -- the damping factor, as part of this equation: damping_amount = 0.996 + (idamp * 0.002) The default damping_amount is 0.996 which means that the default value of idamp is 0. The maximum damping_amount is 1.0 (no damping). This means the maximum value for idamp is 2.0. The recommended range for idamp is usually below 75% of the maximum value. &namerasmus; suggests a range of 1.4-1.75. He also suggests a maximum value of 1.9 instead of the theoretical limit of 2.0. imaxshake (optional, default=0) -- amount of energy to add back into the system. The value should be in range 0 to 1. ifreq (optional) -- the main resonant frequency. The default value is 450. ifreq1 (optional) -- the first resonant frequency. The default value is 600. ifreq2 (optional) -- the second resonant frequency. The default value is 750. Performance kamp -- Amplitude of output. Note: As these instruments are stochastic, this is only an approximation. Examples Here is an example of the dripwater opcode. It uses the file dripwater.csd. Example of the dripwater opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also bamboo, guiro, sleighbells, tambourine Credits Author: &nameperry;, part of the PhISEM (Physically Informed Stochastic Event Modeling) Adapted by &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.07 Added notes by &namerasmus; on May 2002. manual_src~dfsg/opcodes/p5gconnect.xml0000644000000000000000000000421412262561504017171 0ustar rootroot p5gconnect Instrument Control:Sensing and Control p5gconnect p5gconnect Reads data from a P5 Glove controller. Description Opens and at control-rate polls a P5 Glove controller. Syntax p5gconnect Initialization The opcode locates a P5 Glove attached to the computer by USB, and starts a listener thread to poll the device. Performance Every control cycle the glove is polled for its position, and finger and button states. These values are read by the p5gdata opcode. Example Here is an example of the p5g opcodes. It uses the file p5g.csd. Example of the p5g opcodes. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also p5gdata, Credits Author: &namejohn; Codemist Ltd 2009 New in version 5.12 manual_src~dfsg/opcodes/polyaft.xml0000644000000000000000000000504112262561504016601 0ustar rootroot polyaft Real-time MIDI:Input polyaft polyaft Returns the polyphonic after-touch pressure of the selected note number. Description polyaft returns the polyphonic pressure of the selected note number, optionally mapped to an user-specified range. Syntax ires polyaft inote [, ilow] [, ihigh] kres polyaft inote [, ilow] [, ihigh] Initialization inote -- note number. Normally set to the value returned by notnum ilow (optional, default: 0) -- lowest output value ihigh (optional, default: 127) -- highest output value Performance kres -- polyphonic pressure (aftertouch). Examples Here is an example of the polyaft opcode. It uses the file polyaft.csd. Don't forget that you must include the -F flag when using an external MIDI file like polyaft.mid. Example of the polyaft opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Added thanks to an email from &nameistvan; New in version 4.12 manual_src~dfsg/opcodes/tigoto.xml0000644000000000000000000000521012262561504016426 0ustar rootroot tigoto Instrument Control:Program Flow Control tigoto tigoto Transfer control at i-time when a new note is being tied onto a previously held note Description Similar to igoto but effective only during an i-time pass at which a new note is being tied onto a previously held note. (See i Statement) It does not work when a tie has not taken place. Allows an instrument to skip initialization of units according to whether a proposed tie was in fact successful. (See also tival). Syntax tigoto label where label is in the same instrument block and is not an expression. Examples Here is an example of the tigoto opcode. It uses the file tigoto.csd. Example of the tigoto opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cigoto, goto, if, igoto, kgoto, timout More on this opcode: http://www.csounds.com/journal/2005fall/tiedNotes.html , written by Steven Yi manual_src~dfsg/opcodes/clockon.xml0000644000000000000000000000445112262561504016557 0ustar rootroot clockon Instrument Control:Clock Control clockon clockon Starts one of a number of internal clocks. Description Starts one of a number of internal clocks. Syntax clockon inum Initialization inum -- the number of a clock. There are 32 clocks numbered 0 through 31. All other values are mapped to clock number 32. Performance Between a clockon and a clockoff opcode, the CPU time used is accumulated in the clock. The precision is machine dependent but is the millisecond range on UNIX and Windows systems. The readclock opcode reads the current value of a clock at initialization time. Examples Here is an example of the clockon opcode. It uses the file clockon.csd. Example of the clockon opcode. See Also clockoff, readclock Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK July, 1999 New in Csound version 3.56 manual_src~dfsg/opcodes/sumarray.xml0000644000000000000000000000425112262561504016770 0ustar rootroot sumarray Array Opcodes sumarray sumarray returns the sum of the elements in an array. Description The sumarray opcode returns the sum of all elements in a k-rate array array. Syntax ksum sumarray karray Performance ksum -- variable for result. karray -- array for reading. Examples Here is an example of the sumarray opcode. It uses the file sumarray.csd. Example of the sumarray opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also minarray, sumarray, scalearray, Credits Author: &namejohn; October 2011 New in Csound version 5.14 Revised in Csound version 6.00 to work on multidimensional arrays manual_src~dfsg/opcodes/resony.xml0000644000000000000000000001023412262561504016442 0ustar rootroot resony Signal Modifiers:Standard Filters:Resonant resony resony A bank of second-order bandpass filters, connected in parallel. Description A bank of second-order bandpass filters, connected in parallel. Syntax ares resony asig, kbf, kbw, inum, ksep [, isepmode] [, iscl] [, iskip] Initialization inum -- number of filters isepmode (optional, default=0) -- if isepmode = 0, the separation of center frequencies of each filter is generated logarithmically (using octave as unit of measure). If isepmode not equal to 0, the separation of center frequencies of each filter is generated linearly (using Hertz). Default value is 0. iscl (optional, default=0) -- coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. (This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise.) A zero value signifies no scaling of the signal, leaving that to some later adjustment (e.g. balance). The default value is 0. iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance asig -- audio input signal kbf -- base frequency, i.e. center frequency of lowest filter in Hz kbw -- bandwidth in Hz ksep -- separation of the center frequency of filters in octaves resony is a bank of second-order bandpass filters, with k-rate variant frequency separation, base frequency and bandwidth, connected in parallel (i.e. the resulting signal is a mix of the output of each filter). The center frequency of each filter depends of kbf and ksep variables. The maximum number of filters is set to 100. Examples Here is an example of the resony opcode. It uses the file resony.csd. Example of the resony opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; Italy 1999 New in Csound version 3.56 manual_src~dfsg/opcodes/sense.xml0000644000000000000000000000073712262561504016247 0ustar rootroot sense sense sense Same as the sensekey opcode. Description Same as the sensekey opcode. manual_src~dfsg/opcodes/chnsend.xml0000644000000000000000000000612312262561504016547 0ustar rootroot chnsend Signal I/O:Software Bus chnsend chnsend Sends data via the named software bus. Description Send to a channel of the named software bus. Implies declaring the channel with imode=2 (see also chn_k, chn_a, and chn_S). Note that chnsend/chnrecv (which in Csound6 work identically to invalue/outvalue) are usually used for the callback-based communication between Csound and an external host. Use the chnset/chnget opcodes for sending and receiving data inside Csound. Syntax chnsend ival, Sname chnsend kval, Sname chnsend aval, Sname chnsend Sval, Sname Initialization Sname -- a string that indicates which named channel of the software bus to send to. Performance ival -- the control value to write at i-time. kval -- the control value to write at performance time. aval -- the audio signal to write at performance time. Sval -- the string value to write at i-time. Example The example shows the software bus being used to write pitch information to a controlling program. sr = 44100 ksmps = 100 nchnls = 1 instr 1 a1 in kp,ka pitchamdf a1 chnsend kp, "pitch" endin See Also chnrecv, chnset, chnget Credits Author: &nameistvan; 2005 manual_src~dfsg/opcodes/midicontrolchange.xml0000644000000000000000000001120412262561504020612 0ustar rootroot midicontrolchange Real-time MIDI:MIDI/Score Interoperability midicontrolchange midicontrolchange Gets a MIDI control change value. Description midicontrolchange is designed to simplify writing instruments that can be used interchangeably for either score or MIDI input, and to make it easier to adapt instruments originally written for score input to work with MIDI input. In general, it should be possible to write instrument definitions that work identically with both scores and MIDI, including both MIDI files and real-time MIDI input, without using any conditional statements, and that take full advantage of MIDI voice messages. Note that correlating Csound instruments with MIDI channel numbers is done using the massign opcode for real-time performance,. For file-driven performance, instrument numbers default to MIDI channel number + 1, but the defaults are overridden by any MIDI program change messages in the file. Syntax midicontrolchange xcontroller, xcontrollervalue [, ilow] [, ihigh] Initialization ilow (optional) -- optional low value after rescaling, defaults to 0. ihigh (optional) -- optional high value after rescaling, defaults to 127. Performance xcontroller -- specifies a MIDI controller number (0-127). xcontrollervalue -- returns the value of the MIDI controller during MIDI activation, remains unchanged otherwise. If the instrument was activated by MIDI input, the opcode overwrites the values of the xcontroller and xcontrollervalue with the corresponding values from MIDI input. If the instrument was NOT activated by MIDI input, the values of xcontroller and xcontrollervalue remain unchanged. This enables score p-fields to receive MIDI input data during MIDI activation, and score values otherwise. Adapting a score-activated Csound instrument. See the MIDI interop opcodes section for details on adapting score driven instruments for MIDI or vice-versa. Examples Here is an example of the midicontrolchange opcode. It uses the file midicontrolchange.csd Example of the midicontrolchange opcode. See Also midichannelaftertouch, mididefault, midinoteoff, midinoteoncps, midinoteonkey, midinoteonoct, midinoteonpch, midipitchbend, midipolyaftertouch, midiprogramchange Credits Author: Michael Gogins New in version 4.20 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/osciln.xml0000644000000000000000000000473112262561504016417 0ustar rootroot osciln Signal Generators:Basic Oscillators osciln osciln Accesses table values at a user-defined frequency. Description Accesses table values at a user-defined frequency. This opcode can also be written as oscilx. Syntax ares osciln kamp, ifrq, ifn, itimes Initialization ifrq, itimes -- rate and number of times through the stored table. ifn -- function table number. Performance kamp -- amplitude factor osciln will sample several times through the stored table at a rate of ifrq times per second, after which it will output zeros. Generates audio signals only, with output values scaled by kamp. Examples Here is an example of the osciln opcode. It uses the file osciln.csd. Example of the osciln opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also table, tablei, table3, oscil1, oscil1i manual_src~dfsg/opcodes/expcurve.xml0000644000000000000000000000547712262561504017001 0ustar rootroot expcurve Signal Generators:Linear and Exponential Generators expcurve expcurve This opcode implements a formula for generating a normalised exponential curve in range 0 - 1. It is based on the Max / MSP work of Eric Singer (c) 1994. Description Generates an exponential curve in range 0 to 1 of arbitrary steepness. Steepness index equal to or lower than 1.0 will result in Not-a-Number errors and cause unstable behavior. The formula used to calculate the curve is: (exp(x * log(y))-1) / (y-1) where x is equal to kindex and y is equal to ksteepness. Syntax kout expcurve kindex, ksteepness Performance kindex -- Index value. Expected range 0 to 1. ksteepness -- Steepness of the generated curve. Values closer to 1.0 result in a straighter line while larger values steepen the curve. kout -- Scaled output. Examples Here is an example of the expcurve opcode. It uses the file expcurve.csd. Example of the expcurve opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also scale, gainslider, logcurve Credits Author: David Akbari October 2006 manual_src~dfsg/opcodes/sfinstr3.xml0000644000000000000000000001410712262561504016701 0ustar rootroot sfinstr3 Signal Generators:Sample Playback sfinstr3 sfinstr3 Plays a SoundFont2 (SF2) sample instrument, generating a stereo sound with cubic interpolation. Description Plays a SoundFont2 (SF2) sample instrument, generating a stereo sound with cubic interpolation. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. Syntax ar1, ar2 sfinstr3 ivel, inotenum, xamp, xfreq, instrnum, ifilhandle \ [, iflag] [, ioffset] Initialization ivel -- velocity value inotenum -- MIDI note number value instrnum -- number of an instrument of a SF2 file. ifilhandle -- unique number generated by sfload opcode to be used as an identifier for a SF2 file. Several SF2 files can be loaded and activated at the same time. iflag (optional) -- flag regarding the behavior of xfreq and inotenum ioffset (optional) -- start playing at offset, in samples. Performance xamp -- amplitude correction factor xfreq -- frequency value or frequency multiplier, depending by iflag. When iflag = 0, xfreq is a multiplier of a the default frequency, assigned by SF2 preset to the inotenum value. When iflag = 1, xfreq is the absolute frequency of the output sound, in Hz. Default is 0. When iflag = 0, inotenum sets the frequency of the output according to the MIDI note number used, and xfreq is used as a multiplier. When iflag = 1, the frequency of the output, is set directly by xfreq. This allows the user to use any kind of micro-tuning based scales. However, this method is designed to work correctly only with presets tuned to the default equal temperament. Attempts to use this method with a preset already having non-standard tunings, or with drum-kit-based presets, could give unexpected results. Adjustment of the amplitude can be done by varying the xamp argument, which acts as a multiplier. The ioffset parameter allows the sound to start from a sample different than the first one. The user should make sure that its value is within the length of the specific sound. Otherwise, Csound will probably crash. sfinstr3 is a cubic-interpolation version of sfinstr. Difference of sound-quality is noticeable specially in bass-frequency-transposed samples. In high-freq-transposed samples the difference is less noticeable, and I suggest to use linear-interpolation versions, because they are faster. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. Examples Here is an example of the sfinstr3 opcode. It uses the file sfinstr3.csd. Example of the sfinstr3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also sfilist, sfinstr3m, sfinstrm, sfinstr, sfload, sfpassign, sfplay3, sfplay3m, sfplay, sfplaym, sfplist, sfpreset Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/vstinfo.xml0000644000000000000000000000426412262561504016621 0ustar rootroot vstinfo Plugin Hosting:VST vstinfo vstinfo Displays the parameters and the programs of a VST plugin. Syntax vstinfo instance Description vstinfo displays the parameters and the programs of a VST plugin. Note: The verbose flag in vstinit gives the same information as vstinfo. vstinfo is useful after loading parameter banks, or when the plugin changes parameters dynamically. Initialization instance - the number which identifies the plugin, to be passed to other vst4cs opcodes. Examples Here is an example of the use of the vstinfo opcode. It uses the file vst4cs.csd. Example of the <emphasis>vstinfo</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits By: &nameandres; and &namemichael; Uses code from Hermann Seib's VSTHost and Thomas Grill's vst~ object. VST is a trademark of Steinberg Media Technologies GmbH. VST Plug-In Technology by Steinberg. manual_src~dfsg/opcodes/buthp.xml0000644000000000000000000000136612262561504016253 0ustar rootroot buthp Signal Modifiers:Standard Filters buthp buthp Same as the butterhp opcode. Description Same as the butterhp opcode. Syntax ares buthp asig, kfreq [, iskip] ares buthp asig, afreq [, iskip] manual_src~dfsg/opcodes/instimes.xml0000644000000000000000000000120312262561504016752 0ustar rootroot instimes instimes instimes Deprecated. Description Deprecated as of version 3.62. Use the timeinsts opcode instead. Credits &namedavid; originally pointed out this deprecated name. manual_src~dfsg/opcodes/vrandh.xml0000644000000000000000000001030012262561504016377 0ustar rootroot vrandh Vectorial:Random vrandh vrandh Generates a vector of random numbers stored into a table, holding the values for a period of time. Description Generates a vector of random numbers stored into a table, holding the values for a period of time. Generates a sort of 'vectorial band-limited noise'. Syntax vrandh ifn, krange, kcps, ielements [, idstoffset] [, iseed] \ [, isize] [, ioffset] Initialization ifn - number of the table where the vectorial signal will be generated ielements - number of elements of the vector idstoffset - (optional, default=0) -- index offset for the destination table iseed (optional, default=0.5) -- seed value for the recursive pseudo-random formula. A value between 0 and +1 will produce an initial output of kamp * iseed. A negative value will cause seed re-initialization to be skipped. A value greater than 1 will seed from system time, this is the best option to generate a different random sequence for each run. isize (optional, default=0) -- if zero, a 16 bit number is generated. If non-zero, a 31-bit random number is generated. Default is 0. ioffset - (optional, default=0) -- a base value added to the random result. Performance krange - range of random elements (from -krange to krange). kcps - rate of generated elements in cycles per seconds. This opcode is similar to randh, but operates on vectors instead of with scalar values. Though the argument isize defaults to 0, thus using a 16-bit random number generator, using the newer 31-bit algorithm is recommended, as this will produce a random sequence with a longer period (more random numbers before the sequence starts repeating). The output is a vector contained in ifn (that must be previously allocated). All these operators are designed to be used together with other opcodes that operate with vector such as bmscan, adsynt, etc. Note: bmscan not yet available on Canonical Csound Examples Here is an example of the vrandh opcode. It uses the file vrandh.csd. Example of the vrandh opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also vrandi, randh Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/pchmidinn.xml0000644000000000000000000001700212262561504017074 0ustar rootroot pchmidinn Pitch Converters:Functions pchmidinn pchmidinn Converts a Midi note number value to octave point pitch-class units. Description Converts a Midi note number value to octave point pitch-class units. Syntax pchmidinn (MidiNoteNumber) (init- or control-rate args only) where the argument within the parentheses may be a further expression. Performance pchmidinn is a function that takes an i-rate or k-rate value representing a Midi note number and returns the equivalent pitch value in Csound's octave point pitch-class format. This conversion assumes that Middle C (8.00 in pch) is Midi note number 60. Midi note number values are typically integers in the range from 0 to 127 but fractional values or values outside of this range will be interpreted consistently. pchmidinn vs. pchmidi The pchmidinn opcode may be used in any Csound instrument instance whether it is activated from a Midi event, score event, line event, or from another instrument. The input value for pchmidinn might for example come from a p-field in a textual score or it may have been retrieved from the real-time Midi event that activated the current note using the notnum opcode. You must specify an i-rate or k-rate expression for the Midi note number that is to be converted. On the other hand, the pchmidi opcode only produces meaningful results in a Midi-activated note (either real-time or from a Midi score with the -F flag). With pchmidi, the Midi note number value is taken from the Midi event associated with the instrument instance, and no location or expression for this value may be specified. pchmidinn and its related opcodes are really value converters with a special function of manipulating pitch data. Data concerning pitch and frequency can exist in any of the following forms: Pitch and Frequency Values Name Abbreviation octave point pitch-class (8ve.pc) pch octave point decimal oct cycles per second cps Midi note number (0-127) midinn
The first two forms consist of a whole number, representing octave registration, followed by a specially interpreted fractional part. For pch, the fraction is read as two decimal digits representing the 12 equal-tempered pitch classes from .00 for C to .11 for B. For oct, the fraction is interpreted as a true decimal fractional part of an octave. The two fractional forms are thus related by the factor 100/12. In both forms, the fraction is preceded by a whole number octave index such that 8.00 represents Middle C, 9.00 the C above, etc. Midi note number values range between 0 and 127 (inclusively) with 60 representing Middle C, and are usually whole numbers. Thus A440 can be represented alternatively by 440 (cps), 69 (midinn), 8.09 (pch), or 8.75 (oct). Microtonal divisions of the pch semitone can be encoded by using more than two decimal places. The mnemonics of the pitch conversion units are derived from morphemes of the forms involved, the second morpheme describing the source and the first morpheme the object (result). Thus cpspch(8.09) will convert the pitch argument 8.09 to its cps (or Hertz) equivalent, giving the value of 440. Since the argument is constant over the duration of the note, this conversion will take place at i-time, before any samples for the current note are produced. By contrast, the conversion cpsoct(8.75 + k1) which gives the value of A440 transposed by the octave interval k1. The calculation will be repeated every k-period since that is the rate at which k1 varies. Note The conversion from pch, oct, or midinn into cps is not a linear operation but involves an exponential process that could be time-consuming when executed repeatedly. Csound now uses a built-in table lookup to do this efficiently, even at audio rates. Because the table index is truncated without interpolation, pitch resolution when using one of these opcodes is limited to 8192 discrete and equal divisions of the octave, and some pitches of the standard 12-tone equally-tempered scale are very slightly mistuned (by at most 0.15 cents). Examples Here is an example of the pchmidinn opcode. It uses the file cpsmidinn.csd. Example of the pchmidinn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cpsmidinn, octmidinn, pchmidi, notnum, cpspch, cpsoct, octcps, octpch, pchoct Credits Derived from original value converters by Barry Vercoe. New in version 5.07 manual_src~dfsg/opcodes/logbtwo.xml0000644000000000000000000000536012262561504016604 0ustar rootroot logbtwo Mathematical Operations:Mathematical Functions logbtwo logbtwo Performs a logarithmic base two calculation. Description Performs a logarithmic base two calculation. Syntax logbtwo(x) (init-rate or control-rate args only) Performance logbtwo() returns the logarithm base two of x. The range of values admitted as argument is .25 to 4 (i.e. from -2 octave to +2 octave response). This function is the inverse of powoftwo(). These functions are fast, because they read values stored in tables. Also they are very useful when working with tuning ratios. They work at i- and k-rate. Examples Here is an example of the logbtwo opcode. It uses the file logbtwo.csd. Example of the logbtwo opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: i1 = 1.585 See Also powoftwo Credits Author: &namegabriel; Italy June 1998 Author: &namejohn; University of Bath, Codemist, Ltd. Bath, UK July 1999 Example written by &namekevin;. New in Csound version 3.57 manual_src~dfsg/opcodes/tabmorpha.xml0000644000000000000000000001255112262561504017104 0ustar rootroot tabmorpha Table Control:Read/Write Opreations tabmorpha tabmorpha Allow morphing between a set of tables at audio rate with interpolation. Description tabmorpha allows morphing between a set of tables of the same size, by means of a weighted average between two currently selected tables. Syntax aout tabmorpha aindex, aweightpoint, atabnum1, atabnum2, \ ifn1, ifn2 [, ifn3, ifn4, ... ifnN] Initialization ifn1, ifn2 , ifn3, ifn4, ... ifnN - function table numbers. This is a set of chosen tables the user want to use in the morphing. All tables must have the same length. Be aware that only two of these tables can be chosen for the morphing at one time. Since it is possible to use non-integer numbers for the atabnum1 and atabnum2 arguments, the morphing is the result from the interpolation between adjacent consecutive tables of the set. Performance aout - The output value for index aindex, resulting from morphing two tables (see below). aindex - main index index of the morphed resultant table. The range is 0 to table_length (not included). aweightpoint - the weight of the influence of a pair of selected tables in the morphing. The range of this argument is 0 to 1. A zero makes it output the first table unaltered, a 1 makes it output the second table of the pair unaltered. All intermediate values between 0 and 1 determine the gradual morphing between the two tables of the pair. atabnum1 - the first table chosen for the morphing. This number doesn’t express the table number directly, but the position of the table in the set sequence (starting from 0 to N-1). If this number is an integer, the corresponding table will be chosen unaltered. If it contains fractional values, then an interpolation with the next adjacent table will result. atabnum2 - the second table chosen for the morphing. This number doesn’t express the table number directly, but the position of the table in the set sequence (starting from 0 to N-1). If this number is an integer, corresponding table will be chosen unaltered. If it contains fractional values, then an interpolation with the next adjacent table will result. The tabmorpha family of opcodes is similar to the table family, but allows morphing between two tables chosen into a set of tables. Firstly the user has to provide a set of tables of equal length (ifn2 [, ifn3, ifn4,…ifnN]). Then he can choose a pair of tables in the set in order to perform the morphing: atabnum1 and atabnum2 are filled with numbers (zero represents the first table in the set, 1 the second, 2 the third and so on). Then determine the morphing between the two chosen tables with the aweightpoint parameter. After that the resulting table can be indexed with the aindex parameter like a normal table opcode. If the value of this parameter surpasses the length of tables (which must be the same for all tables), then it is wrapped around. tabmorpha is the audio-rate version of tabmorphi (it uses interpolation). All input arguments work at a-rate. Examples Here is an example of the tabmorpha opcode. It uses the file tabmorpha.csd. Example of the tabmorpha opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also table, tabmorph, tabmorphi, tabmorphak, ftmorf, Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/pvsbufread2.xml0000644000000000000000000001073612262561504017355 0ustar rootroot pvsbufread2 Spectral Processing:Streaming pvsbufread2 pvsbufread2 This opcode reads a circular buffer of f-signals (streaming PV signals), with binwise additional delays. Description This opcode reads from a circular buffer of length ilen (secs), taking a handle for the buffer and a time pointer, which holds the current read position (also in seconds). It is used in conjunction with a pvsbuffer opocde. Reading is circular, wrapping around at the end of the buffer. Extra delay times are taken from a function table, with each point on it defining a delay time in seconds affecting the corresponding bin. Syntax fsig pvsbufread2 ktime, khandle, ift1, ift2 Initialization ift1 -- function table with at least fftisze/2+1 points where delays (in secs) for bin amplitudes are set (function table positions are equivalent to bin numbers) ift2 -- function table with at least fftisze/2+1 points where delays (in secs) for bin frequencies are set (function table positions are equivalent to bin numbers) Performance fsig -- output pv stream ktime -- time position of reading pointer (in secs). khandle -- handle identifying the buffer to be read. When using k-rate handles, it is important to initialise the k-rate variable to a given existing handle. When changing buffers, fsig buffers need to be compatible (same fsig format). With this opcode and pvsbuffer, it is possible to, among other things: time-stretch/compress a fsig stream, by reading it at different rates delay a fsig or portions of it. 'brassage' two or more fsigs by switching buffers, since the reading handles are k-rate. Note that, when using k-rate handles, it is important to initialise the k-rate variable to a given handle (so that the fsig initialisation can take place) and it is only possible to switch handles between compatible fsig buffers (with the same fftsize and overlap). It is important that the handle value passed to pvsbufread2 is valid and was created by pvsbuffer. Csound will crash with invalid handles. Examples Here is an example of the pvsbufread2 opcode. It uses the file pvsbufread2.csd. Example of the pvsbufread2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsbuffer, pvsadsyn Credits Author: &namevictor; July 2011 manual_src~dfsg/opcodes/p.xml0000644000000000000000000000357512262561504015374 0ustar rootroot p Instrument Control:Initialization and Reinitialization p p Show the value in a given p-field. Description Show the value in a given p-field. Syntax p(x) This function works at i-rate and k-rate. Initialization x -- the number of the p-field. Performance The value returned by the p function is the value in a p-field. Examples Here is an example of the p opcode. It uses the file p.csd. Example of the p opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: instr 1: i1 = 50.375 Credits Example written by &namekevin;. manual_src~dfsg/opcodes/vco2ift.xml0000644000000000000000000000762612262561504016512 0ustar rootroot vco2ift Signal Generators:Dynamic Spectrum Oscillators vco2ift vco2ift Returns a table number at i-time for a given oscillator frequency and wavform. Description vco2ift is the same as vco2ft, but works at i-time. It is suitable for use with opcodes that expect an i-rate table number (for example, oscili). Syntax ifn vco2ift icps, iwave [, inyx] Initialization ifn -- the ftable number. icps -- frequency in Hz. Zero and negative values are allowed. However, if the absolute value exceeds sr/2 (or sr * inyx), the selected table will contain silence. iwave -- the waveform for which table number is to be selected. Allowed values are: 0: sawtooth 1: 4 * x * (1 - x) (integrated sawtooth) 2: pulse (not normalized) 3: square wave 4: triangle Additionally, negative iwave values select user defined waveforms (see also vco2init). inyx (optional, default=0.5) -- bandwidth of the generated waveform, as percentage (0 to 1) of the sample rate. The expected range is 0 to 0.5 (i.e. up to sr/2), other values are limited to the allowed range. Setting inyx to 0.25 (sr/4), or 0.3333 (sr/3) can produce a fatter sound in some cases, although it is more likely to reduce quality. Examples Here is an example of the vco2ift opcode. It uses the file vco2ift.csd. Example of the vco2ift opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like these: instr 1: ifn = 22.000 instr 1: ifn = 8.000 instr 1: ifn = 3.000 See the example for the vco2 opcode too. See Also vco2ft, vco2init, and vco2. Credits Author: &nameistvan; New in version 4.22 manual_src~dfsg/opcodes/lenarray.xml0000644000000000000000000000374712262561504016753 0ustar rootroot lenarray Table Control:Table Queries lenarray lenarray Calculate the length of a vector. Description Calculate the length of a vector (one-dimensionly k-rate array). Syntax ir lenarray karray kr lenarray karray Performance kr -- length of vector. karray -- array to query. Examples Here is an example of the lenarray opcode. It uses the file lenarray.csd. Example of the lenarray opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also Vectorial opcodes Credits Author: &namejohn; Codemist Ltd 2013 New in version 6.00 manual_src~dfsg/opcodes/pvswarp.xml0000644000000000000000000000671012262561504016631 0ustar rootroot pvswarp Spectral Processing:Streaming pvswarp pvswarp Warp the spectral envelope of a PVS signal Description Warp the spectral envelope of a PVS signal by means of shifting and scaling. Syntax fsig pvswarp fsigin, kscal, kshift[, klowest, kmeth, kgain, kcoefs] Performance fsig -- output pv stream fsigin -- input pv stream kscal -- spectral envelope scaling ratio. Values > 1 stretch the envelope and < 1 compress it. kshift -- spectral envelope shift, values > 0 shift the envelope linearly upwards and values < 1 shift it downwards. klowest -- lowest frequency shifted (affects only kshift, defaults to 0). kmethod -- spectral envelope extraction method 1: liftered cepstrum method; 2: true envelope method (defaults to 1). kgain -- amplitude scaling (defaults to 1). kcoefs -- number of cepstrum coefs used in formant preservation (defaults to 80). It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the pvswarp opcode. It uses the file pvswarp.csd. Example of the pvswarp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Used with vocal sounds, it will shift the formants and result in a changed vowel timbre, similar to the effect of a singer inhaling helium (the 'donald duck' effect). See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namevictor; November 2004 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/rspline.xml0000644000000000000000000000644412262561504016607 0ustar rootroot rspline Signal Generators:Linear and Exponential Generators rspline rspline Generate random spline curves. Description Generate random spline curves. Syntax ares rspline xrangeMin, xrangeMax, kcpsMin, kcpsMax kres rspline krangeMin, krangeMax, kcpsMin, kcpsMax Performance kres, ares -- Output signal xrangeMin, xrangeMax -- Range of values of random-generated points kcpsMin, kcpsMax -- Range of point-generation rate. Min and max limits are expressed in cps. rspline (random-spline-curve generator) is similar to jspline but output range is defined by means of two limit values. Also in this case, real output range could be a bit greater of range values, because of interpolating curves beetween each pair of random-points. At present time generated curves are quite smooth when cpsMin is not too different from cpsMax. When cpsMin-cpsMax interval is big, some little discontinuity could occurr, but it should not be a problem, in most cases. Maybe the algorithm will be improved in next versions. These opcodes are often better than jitter when user wants to naturalize or analogize digital sounds. They could be used also in algorithmic composition, to generate smooth random melodic lines when used together with samphold opcode. Note that the result is quite different from the one obtained by filtering white noise, and they allow the user to obtain a much more precise control. Examples Here is an example of the rspline opcode. It uses the file rspline.csd. Example of the rspline opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; New in version 4.15 manual_src~dfsg/opcodes/strupperk.xml0000644000000000000000000000206412262561504017164 0ustar rootroot strupperk Strings:Conversion strupperk strupperk Convert a string to upper case Description Convert Ssrc to upper case, and write the result to Sdst. strupperk runs both at init and performance time. Syntax Sdst strupperk Ssrc See also strupper Credits Author: Istvan Varga 2006 New in version 5.02 manual_src~dfsg/opcodes/vdelay3.xml0000644000000000000000000000572612262561504016504 0ustar rootroot vdelay3 Signal Modifiers:Delay vdelay3 vdelay3 A variable time delay with cubic interpolation. Description vdelay3 is experimental. It is the same as vdelay except that it uses cubic interpolation. (New in Version 3.50.) Syntax ares vdelay3 asig, adel, imaxdel [, iskip] Initialization imaxdel -- Maximum value of delay in milliseconds. If adel gains a value greater than imaxdel it is folded around imaxdel. This should not happen. iskip (optional) -- Skip initialization if present and non-zero. Performance With this unit generator it is possible to do Doppler effects or chorusing and flanging. asig -- Input signal. adel -- Current value of delay in milliseconds. Note that linear functions have no pitch change effects. Fast changing values of adel will cause discontinuities in the waveform resulting noise. Examples Here is an example of the vdelay3 opcode. It uses the file vdelay3.csd. Example of the vdelay3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Two important points here. First, the delay time must be always positive. And second, even though the delay time can be controlled in k-rate, it is not advised to do so, since sudden time changes will create clicks. See Also vdelay Credits Author: &nameparis; MIT, Cambridge 1995 manual_src~dfsg/opcodes/product.xml0000644000000000000000000000341612262561504016607 0ustar rootroot product Mathematical Operations:Opcode Equivalents of Functions product product Multiplies any number of a-rate signals. Description Multiplies any number of a-rate signals. Syntax ares product asig1, asig2 [, asig3] [...] Performance asig1, asig2, asig3, ... -- a-rate signals to be multiplied. Examples Here is an example of the product opcode. It uses the file product.csd. Example of the product opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; Italy April 1999 New in Csound version 3.54 manual_src~dfsg/opcodes/vibes.xml0000644000000000000000000000764312262561504016245 0ustar rootroot vibes Signal Generators:Models and Emulations vibes vibes Physical model related to the striking of a metal block. Description Audio output is a tone related to the striking of a metal block as found in a vibraphone. The method is a physical model developed from &nameperry;, but re-coded for Csound. Syntax ares vibes kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec Initialization ihrd -- the hardness of the stick used in the strike. A range of 0 to 1 is used. 0.5 is a suitable value. ipos -- where the block is hit, in the range 0 to 1. imp -- a table of the strike impulses. The file marmstk1.wav is a suitable function from measurements and can be loaded with a GEN01 table. It is also available at ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. ivfn -- shape of tremolo, usually a sine table, created by a function idec -- time before end of note when damping is introduced Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. kvibf -- frequency of tremolo in Hertz. Suggested range is 0 to 12 kvamp -- amplitude of the tremolo Examples Here is an example of the vibes opcode. It uses the file vibes.csd, and marmstk1.wav. Example of the vibes opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also marimba Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/harmon234.xml0000644000000000000000000001476112262561504016651 0ustar rootroot harmon2 Signal Modifiers:Special Effects harmon2 harmon2 Analyze an audio input and generate harmonizing voices in synchrony with formants preserved. Description Generate harmonizing voices with formants preserved. Syntax ares harmon2 asig, koct, kfrq1, kfrq2, icpsmode, ilowest[, ipolarity] ares harmon3 asig, koct, kfrq1, \ kfrq2, kfrq3, icpsmode, ilowest[, ipolarity] ares harmon4 asig, koct, kfrq1, \ kfrq2, kfrq3, kfrq4, icpsmode, ilowest[, ipolarity] Initialization icpsmode -- interpreting mode for the generating frequency inputs kfrq1, kfrq2, kfrq3 and kfrq4: 0: input values are ratios w.r.t. the cps equivalent of koct. 1: input values are the actual requested frequencies in cps. ilowest -- lowest value of the koct input for which harmonizing voices will be generated. ipolarity -- polarity of asig input, 1 = positive glottal pulses, 0 = negative. Default is 1. Performance Harmon2, harmon3 and harmon4 are high-performance harmonizers, able to provide up to four pitch-shifted copies of the input asig with spectral formants preserved. The pitch-shifting algorithm requires an accurate running estimate (koct, in decimal oct units) of the pitched content of asig, normally gained from an independent pitch tracker such as specptrk. The algorithm then isolates the most recent full pulse within asig, and uses this to generate the other voices at their required pulse rates. If the frequency (or ratio) presented to kfrq1, kfrq2, kfrq3 or kfrq4 is zero, then no signal is generated for that voice. If any of them is non-zero, but the koct input is below the value ilowest, then that voice will output a direct copy of the input asig. As a consequence, the data arriving at the k-rate inputs can variously cause the generated voices to be turned on or off, to pass a direct copy of a non-voiced fricative source, or to harmonize the source according to some constructed algorithm. The transition from one mode to another is cross-faded, giving seemless alternating between voiced (harmonized) and non-voiced fricatives during spoken or sung input. harmon2, harmon3, harmon4 are especially matched to the output of specptrk. The latter generates pitch data in decimal octave format; it also emits its base value if no pitch is identified (as in fricative noise) and emits zero if the energy falls below a threshold, so that harmon2, harmon3, harmon4 can be set to pass the direct signal in both cases. Of course, any other form of pitch estimation could also be used. Since pitch trackers usually incur a slight delay for accurate estimation (for specptrk the delay is printed by the spectrum unit), it is normal to delay the audio signal by the same amount so that harmon2, harmon3, harmon4 can work from a fully concurrent estimate. Examples Here is an example of the harmon2 opcode. It uses the file harmon.csd. Example of the harmon2 opcode. a1,a2 ins ; get mic input w1 spectrum a1, .02, 7, 24, 12, 1, 3 ; and examine it koct,kamp specptrk w1, 1, 6.5, 9.5, 7.5, 10, 7, .7, 0, 3, 1 a3 delay a1, .065 ; allow for ptrk delay a4 harmon2 a3, koct, 1.25, 0.75, 0, 6.9 ; output a fixed 6-4 harmony outs a3, a4 ; as well as the original Here is a complete example of the harmon3 opcode. It uses the file harmon3.csd. Example of the harmon3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namebarry; M.I.T., Cambridge, Mass 2006 New in version 5.04 manual_src~dfsg/opcodes/zamod.xml0000644000000000000000000000522512262561504016241 0ustar rootroot zamod Zak Patch System zamod zamod Modulates one a-rate signal by a second one. Description Modulates one a-rate signal by a second one. Syntax ares zamod asig, kzamod Performance asig -- the input signal kzamod -- controls which za variable is used for modulation. A positive value means additive modulation, a negative value means multiplicative modulation. A value of 0 means no change to asig. zamod modulates one a-rate signal by a second one, which comes from a za variable. The location of the modulating variable is controlled by the i-rate or k-rate variable kzamod. This is the a-rate version of zkmod. Examples Here is an example of the zamod opcode. It uses the file zamod.csd. Example of the zamod opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zacl, ziw, ziwm Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/cos.xml0000644000000000000000000000370312262561504015712 0ustar rootroot cos Mathematical Operations:Trigonometric Functions cos cos Performs a cosine function. Description Returns the cosine of x (x in radians). Syntax cos(x) (no rate restriction) Examples Here is an example of the cos opcode. It uses the file cos.csd. Example of the cos opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cosh, cosinv, sin, sinh, sininv, tan, tanh, taninv manual_src~dfsg/opcodes/ATSsinnoi.xml0000644000000000000000000001415312262561504016776 0ustar rootroot ATSsinnoi Spectral Processing:ATS ATSsinnoi ATSsinnoi uses the data from an ATS analysis file to perform resynthesis. Description ATSsinnoi reads data from an ATS data file and uses the information to synthesize sines and noise together. Syntax ar ATSsinnoi ktimepnt, ksinlev, knzlev, kfmod, iatsfile, ipartials \ [, ipartialoffset, ipartialincr] Initialization iatsfile – the ATS number (n in ats.n) or the name in quotes of the analysis file made using ATS. ipartials – number of partials that will be used in the resynthesis (the noise has a maximum of 25 bands) ipartialoffset (optional) – is the first partial used (defaults to 0). ipartialincr (optional) – sets an increment by which these synthesis opcodes counts up from ipartialoffset for ibins components in the re-synthesis (defaults to 1). Performance ktimepnt – The time pointer in seconds used to index the ATS file. Used for ATSsinnoi exactly the same as for pvoc. ksinlev - controls the level of the sines in the ATSsinnoi ugen. A value of 1 gives full volume sinewaves. knzlev - controls the level of the noise components in the ATSsinnoi ugen. A value of 1 gives full volume noise. kfmod – an input for performing pitch transposition or frequency modulation on all of the synthesized partials, if no fm or pitch change is desired then use a 1 for this value. ATSsinnoi reads data from an ATS data file and uses the information to synthesize sines and noise together. The noise energy for each band is distributed equally among each partial that falls in that band. Each partial is then synthesized, along with that partial's noise component. Each noise component is then modulated by the corresponding partial to be put in the correct place in the frequency spectrum. The level of the noise and the partials are individually controllable. See the ATS webpage for more info about the sinnoi synthesis. An ATS analysis differs from a pvanal in that ATS tracks the partials and computes the noise energy of the sound being analyzed. For more info about ATS analysis read Juan Pampin's description on the the ATS web-page. Examples ktime line 0, p3, 2.5 asig ATSsinnoi ktime, 1, 1, 1, "beats.ats", 42 Here we synthesize both the noise and the sinewaves (all 42 partials) contained in "beats.ats" together. The relative volumes of the noise and the partials are unaltered (each set to 1). Here is another example of the ATSsinnoi opcode. It uses the file ATSsinnoi.csd. Example of the ATSsinnoi opcode. This example here is like the other example except that we use an envelope to control knzlev (the noise level). The result of this will be the "beats.wav" sound that has its noise component fade in over the duration of the note. Here is another example of the ATSsinnoi opcode. It uses the file ATSsinnoi-2.csd. Example 2 of the ATSsinnoi opcode. See also ATSread, ATSreadnz, ATSinfo, ATSbufread, ATScross, ATSinterpread, ATSpartialtap, ATSadd, ATSaddnz Credits Author: Alex Norman Seattle,Washington 2004 manual_src~dfsg/opcodes/strchark.xml0000644000000000000000000000223512262561504016746 0ustar rootroot strchark Strings:Conversion strchark strchark Return the ASCII code of a character in a string Description Return the ASCII code of the character in Sstr at kpos (defaults to zero which means the first character), or zero if kpos is out of range. strchark runs both at init and performance time. Syntax kchr strchark Sstr[, kpos] See also strchar Credits Author: Istvan Varga 2006 New in version 5.02 manual_src~dfsg/opcodes/OSClisten.xml0000644000000000000000000001324712262561504016775 0ustar rootroot OSClisten OSC OSClisten OSClisten Listen for OSC messages to a particular path. Description On each k-cycle looks to see if an OSC message has been send to a given path of a given type. Syntax kans OSClisten ihandle, idest, itype [, xdata1, xdata2, ...] Initialization ihandle -- a handle returned by an earlier call to OSCinit, to associate OSClisten with a particular port number. idest -- a string that is the destination address. This takes the form of a file name with directories. Csound uses this address to decide if messages are meant for csound. itype -- a string that indicates the types of the optional arguments that are to be read. The string can contain the characters "cdfhis" which stand for character, double, float, 64-bit integer, 32-bit integer, and string. All types other than 's' require a k-rate variable, while 's' requires a string variable. A handler is inserted into the listener (see OSCinit) to intercept messages of this pattern. Performance kans -- set to 1 if a new message was received, or zero if not. If multiple messages are received in a single control period, the messages are buffered, and OSClisten can be called again until zero is returned. If there was a message the xdata variables are set to the incoming values, as interpretted by the itype parameter. Note that although the xdata variables are on the right of an operation they are actually outputs, and so must be variables of type k, gk, S, or gS, and may need to be declared with init, or = in the case of string variables, before calling OSClisten. Example The example shows a pair of floating point numbers being received on port 7770. sr = 44100 ksmps = 100 nchnls = 2 gihandle OSCinit 7770 instr 1 kf1 init 0 kf2 init 0 nxtmsg: kk OSClisten gihandle, "/foo/bar", "ff", kf1, kf2 if (kk == 0) goto ex printk 0,kf1 printk 0,kf2 kgoto nxtmsg ex: endin Below are two .csd files which demonstrate the usage of the OSC opcodes. They use the files OSCmidisend.csd and OSCmidircv.csd. Example of the OSC opcodes. The following two .csd files demonstrate the usage of the OSC opcodes in csound. The first file, OSCmidisend.csd, transforms received real-time MIDI messages into OSC data. The second file, OSCmidircv.csd, can take these OSC messages, and intrepret them to generate sound from note messages, and store controller values. It will use controller number 7 to control volume. Note that these files are designed to be on the same machine, but if a different host address (in the IPADDRESS macro) is used, they can be separate machines on a network, or connected through the internet. CSD file to send OSC messages: CSD file to receive OSC messages: See Also OSCsend, OSCinit More information on this opcode: http://www.youtube.com/watch?v=JX1C3TqP_9Y , made by Andrés Cabrera Credits Author: &namejohn; 2005 Examples by: David Akbari, &nameandres; and Jonathan Murphy 2007 manual_src~dfsg/opcodes/randomh.xml0000644000000000000000000001041512262561504016554 0ustar rootroot randomh Signal Generators:Random (Noise) Generators randomh randomh Generates random numbers with a user-defined limit and holds them for a period of time. Description Generates random numbers with a user-defined limit and holds them for a period of time. Syntax ares randomh kmin, kmax, xcps [,imode] [,ifirstval] kres randomh kmin, kmax, kcps [,imode] [,ifirstval] Initialization imode (optional, default=0) -- generation mode of the first output value (see below) ifirstval (optional, default=0) -- first output value Performance kmin -- minimum range limit kmax -- maximum range limit kcps, xcps -- rate of random break-point generation The randomh opcode is similar to randh but allows the user to set arbitrary minimum and maximum values. When imode = 0 (the default), the kmin argument value is outputted during 1/kcps (resp. 1/xcps) seconds at the beginning of the note. Then, the normal process takes place with a new random number generated and held every 1/kcps (resp. 1/xcps) seconds. When imode = 2, the ifirstval argument value is outputted during 1/kcps (resp. 1/xcps) seconds at the beginning of the note. Then, the normal process takes place with a new random number generated and held every 1/kcps (resp. 1/xcps) seconds. When imode = 3, the generation process begins with a random value from the initialization time. Examples Here is an example of the randomh opcode. It uses the file randomh.csd. Example of the randomh opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: Mode: 0 i1 220.00000 i1 396.26079 i1 240.75446 i1 364.24577 ... Mode: 2 i1 330.00000 i1 416.50935 i1 356.11619 i1 433.59324 ... Mode: 3 i1 261.17741 i1 402.00891 i1 393.86592 i1 307.19839 ... See Also randh, random, randomi Credits Author: &namegabriel; Arguments imode and ifirstval added by &namepinot;, Jan. 2011, after a discussion with Peiman Khosravi on the csnd list. Example written by &namekevin;, adapted for new args by &namepinot;. manual_src~dfsg/opcodes/pvsfreeze.xml0000644000000000000000000001000712262561504017132 0ustar rootroot pvsfreeze Spectral Processing:Streaming pvsfreeze pvsfreeze Freeze the amplitude and frequency time functions of a pv stream according to a control-rate trigger. Description This opcodes 'freezes' the evolution of pvs stream by locking into steady amplitude and/or frequency values for each bin. The freezing is controlled, independently for amplitudes and frequencies, by a control-rate trigger, which switches the freezing 'on' if equal to or above 1 and 'off' if below 1. Syntax fsig pvsfreeze fsigin, kfreeza, kfreezf Performance fsig -- output pv stream fsigin -- input pv stream. kfreeza -- freezing switch for amplitudes. Freezing is on if above or equal to 1 and off if below 1. kfcf -- freezing switch for frequencies. Freezing is on if above or equal to 1 and off if below 1. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Example asig in ; input ktrig oscil 1.5, 0.25, 1 ; trigger fim pvsanal asig1, 1024, 256, 1024, 0 ; pvoc analysis fou pvsfreeze fim, abs(ktrig), abs(ktrig) ; regular 'freeze' of spectra aout pvsynth fou ; pvoc synthesis In the example above the input signal will be regularly 'frozen' for a short while, as the trigger rises above 1 about every two seconds. Here is an example of the use of the pvsfreeze opcode. It uses the file pvsfreeze.csd. Example of the <emphasis>pvsfreeze</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namevictor; May 2006 New plugin in version 5 May 2006. manual_src~dfsg/opcodes/vstmidiout.xml0000644000000000000000000000676112262561504017344 0ustar rootroot vstmidiout Plugin Hosting:VST vstmidiout vstmidiout Sends MIDI information to a VST plugin. Syntax vstmidiout instance, kstatus, kchan, kdata1, kdata2 Description vstmidiout is used for sending MIDI information to a VST plugin. Initialization instance - the number which identifies the plugin, to be passed to other vst4cs opcodes. Performance kstatus - the type of midi message to be sent. Currently noteon (144), note off (128), Control Change (176), Program change (192), Aftertouch (208) and Pitch Bend (224) are supported. kchan - the MIDI channel transmitted on. kdata1, kdata2 - the MIDI data pair, which varies depending on kstatus. e.g. note/velocity for note on and note off, Controller number/value for control change. Examples Example for vstmidiout /* orc */ sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 gihandle1 vstinit "c:/vstplugins/cheeze/cheeze machine.dll", 1 instr 3 ain1 = 0 ab1, ab2 vstaudio gihandle1, ain1, ain1 outs ab1, ab2 endin instr 4 vstmidiout gihandle1, 144, 1, p4, p5 endin /* sco */ i 3 0 21 i 4 1 1 57 32 i 4 3 1 60 100 i 4 5 1 62 100 i 4 7 1 64 100 i 4 9 1 65 100 i 4 11 1 67 100 i 4 13 1 69 100 i 4 15 3 71 100 i 4 18 3 72 100 e Credits By: &nameandres; and &namemichael; Uses code from Hermann Seib's VSTHost and Thomas Grill's vst~ object. VST is a trademark of Steinberg Media Technologies GmbH. VST Plug-In Technology by Steinberg. manual_src~dfsg/opcodes/readk2.xml0000644000000000000000000001321412262561504016274 0ustar rootroot readk2 Signal I/O:File I/O readk2 readk2 Periodically reads two orchestra control-signal values from an external file. Description Periodically reads two orchestra control-signal values from an external file. Syntax kr1, kr2 readk2 ifilname, iformat, iprd Initialization ifilname -- an integer N denoting a file named "readk.N" or a character string (in double quotes, spaces permitted) denoting the external file name. For a string, it may either be a full path name with directory specified or a simple filename. In the later case, the file is sought first in the current directory, then in SSDIR, and finally in SFDIR. iformat -- specifies the input data format: 1 = 8-bit signed integers (char) 4 = 16-bit short integers 5 = 32-bit long integers 6 = 32-bit floats 7 = ASCII long integers (plain text) 8 = ASCII floats (plain text) Note that A-law and U-law formats are not available, and that all formats except the last two are binary. The input file should be a "raw", headerless data file. iprd -- the rate (period) in seconds, rounded to the nearest orchestra control period, at which the signals are read from the input file. A value of 0 implies one control period (the enforced minimum), which will read new values at the orchestra control rate. Longer periods will cause the same values to repeat for more than one control period. Performance kr1, kr2 -- output of the signals read from ifilname. This opcode allows two generated control signal values to be read from a named external file. The file should contain no header information but it should contain a regularly sampled time series of control values. For binary formats, the individual samples of each signal are interleaved. For ASCII text formats, the values are assumed to be separated by at least one whitespace character. The two "channels" in a sample frame may be on the same line or separated by newline characters, it does not matter. There may be any number of readk2 opcodes in an instrument or orchestra and they may read from the same or different files. Examples Here is an example of the readk2 opcode. It uses the file readk2.csd. Example of the readk2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output should include lines like these: kfreq = 429.202551, kdb = -20.495694 kfreq = 429.202551, kdb = -20.495694 kfreq = 407.275258, kdb = -23.123776 kfreq = 475.264472, kdb = -9.300846 kfreq = 569.979181, kdb = -7.315527 kfreq = 440.103457, kdb = -0.058331 kfreq = 429.202600, kdb = -20.495700 kfreq = 429.202600, kdb = -20.495700 kfreq = 407.275300, kdb = -23.123800 kfreq = 475.264500, kdb = -9.300800 kfreq = 569.979200, kdb = -7.315500 kfreq = 440.103500, kdb = -0.058300 See Also dumpk, dumpk2, dumpk3, dumpk4, readk, readk3, readk4 Credits By: John ffitch and Barry Vercoe 1999 or earlier manual_src~dfsg/opcodes/iondur.xml0000644000000000000000000000077012262561504016427 0ustar rootroot iondur iondur iondur Deprecated. Description Deprecated as of version 3.52. Use the noteondur opcode instead. manual_src~dfsg/opcodes/reverbsc.xml0000644000000000000000000000731712262561504016746 0ustar rootroot reverbsc Signal Modifiers:Reverberation reverbsc reverbsc 8 delay line stereo FDN reverb, based on work by Sean Costello Description 8 delay line stereo FDN reverb, with feedback matrix based upon physical modeling scattering junction of 8 lossless waveguides of equal characteristic impedance. Based on Csound orchestra version by Sean Costello. Syntax aoutL, aoutR reverbsc ainL, ainR, kfblvl, kfco[, israte[, ipitchm[, iskip]]] Initialization israte (optional, defaults to the orchestra sample rate) -- assume a sample rate of israte. This is normally set to sr, but a different setting can be useful for special effects. ipitchm (optional, defaults to 1) -- depth of random variation added to delay times, in the range 0 to 10. The default is 1, but this may be too high and may need to be reduced for held pitches such as piano tones. iskip (optional, defaults to zero) -- if non-zero, initialization of the opcode is skipped, whenever possible. Performance aoutL, aoutR -- output signals for left and right channel ainL, ainR -- left and right channel input. Note that having an input signal on either the left or right channel only will still result in having reverb output on both channels, making this unit more suitable for reverberating stereo input than the freeverb opcode. kfblvl -- feedback level, in the range 0 to 1. 0.6 gives a good small "live" room sound, 0.8 a small hall, and 0.9 a large hall. A setting of exactly 1 means infinite length, while higher values will make the opcode unstable. kfco -- cutoff frequency of simple first order lowpass filters in the feedback loop of delay lines, in Hz. Should be in the range 0 to israte/2 (not sr/2). A lower value means faster decay in the high frequency range. Examples Here is an example of the reverbsc opcode. It uses the file reverbsc.csd. An example of the reverbsc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/aftouch.xml0000644000000000000000000000561512262561504016563 0ustar rootroot aftouch Real-time MIDI:Input aftouch aftouch Get the current after-touch value for this channel. Description Get the current after-touch value for this channel. Syntax kaft aftouch [imin] [, imax] Initialization imin (optional, default=0) -- minimum limit on values obtained. imax (optional, default=127) -- maximum limit on values obtained. Performance Get the current after-touch value for this channel. Examples Here is an example of the aftouch opcode. It uses the file aftouch.csd. Example of the aftouch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also ampmidi, cpsmidi, cpsmidib, midictrl, notnum, octmidi, octmidib, pchbend, pchmidi, pchmidib, veloc Credits Author: &namebarry; - &namemike; MIT - Mills May 1997 manual_src~dfsg/opcodes/pvsosc.xml0000644000000000000000000001032612262561504016442 0ustar rootroot pvsosc Spectral Processing:Streaming pvsosc pvsosc PVS-based oscillator simulator. Description Generates periodic signal spectra in AMP-FREQ format, with the option of four wave types: sawtooth-like (harmonic weight 1/n, where n is partial number) square-like (similar to 1., but only odd partials) pulse (all harmonics with same weight) cosine Complex waveforms (ie. all types except cosine) contain all harmonics up to the Nyquist. This makes pvsosc an option for generation of band-limited periodic waves. In addition, types can be changed using a k-rate variable. Syntax fsig pvsosc kamp, kfreq, ktype, isize [,ioverlap] [, iwinsize] [, iwintype] [, iformat] Initialisation fsig -- output pv stream set to zero. isize -- size of analysis frame and window. ioverlap -- (Optional) size of overlap, defaults to isize/4. iwinsize -- (Optional) window size, defaults to isize. iwintype -- (Optional) window type, defaults to Hanning. The choices are currently: 0 = Hamming window 1 = von Hann window iformat -- (Optional) data format, defaults to 0 which produces AMP:FREQ data. That is currently the only option. Performance kamp -- signal amplitude. Note that the actual signal amplitude can, depending on wave type and frequency, vary slightly above or below this value. Generally the amplitude will tend to exceed kamp on higher frequencies (> 1000 Hz) and be reduced on lower ones. Also due to the overlap-add process, when resynthesing with pvsynth, frequency glides will cause the output amplitude to fluctuate above and below kamp. kfreq -- fundamental frequency in Hz. ktype -- wave type: 1. sawtooh-like, 2.square-like, 3.pulse and any other value for cosine. Examples Here is an example of the pvsosc opcode. It uses the file pvsosc.csd. Example of the pvsosc opcode See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namevictor; August 2006 manual_src~dfsg/opcodes/vtabwk.xml0000644000000000000000000000411012262561504016415 0ustar rootroot vtabwk Vectorial:Tables vtabwk vtabwk Write vectors (to tables -or arrays of vectors). Description This opcode writes vectors to tables at k-rate. Syntax vtabwk kndx, ifn, kinarg1 [, kinarg2, kinarg3 , .... , kinargN ] Initialization ifn - table number Performance kndx - Index into f-table, a positive number range matching the table length kinarg1...kinargN - input vector elements This opcode is useful in all cases in which one needs to write sets of values associated to unique indexes (for example, multi-channel samples, STFT bin frames, spectral formants, p-field based scores etc.) . The number of elements of each vector (length of the vector) is determined by the number of optional arguments on the right (kinarg1, kinarg2, kinarg3, .... kinargN). Notice that no wrap nor limit mode is implemented. So, if an index attempt to access to a zone not allocated by the table, Csound will probably crash. However this drawback can be easily avoided by using wrap or limit opcodes applied to indexes before using vtabwk, in order to correct eventual out-of-range values. Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/flkeyin.xml0000644000000000000000000000704012262561504016565 0ustar rootroot FLkeyIn FLTK:Other FLkeyIn FLkeyIn Reports keys pressed (on alphanumeric keyboard) when an FLTK panel has focus. Description FLkeyIn informs about the status of a key pressed by the user on the alphanumeric keyboard when an FLTK panel has got the focus. Syntax kascii FLkeyIn [ifn] Initialization ifn – (optional, default value is zero) set the behavior of FLkeyIn (see below). Performance kascii - the ascii value of last pressed key. If the key is pressed, the value is positive, when the key is released the value is negative. FLkeyIn is useful to know whether a key has been pressed on the computer keyboard. The behavior of this opcode depends on the optional ifn argument. If ifn = 0 (default), FLkeyIn outputs the ascii code of the last pressed key. If it is a special key (ctrl, shift, alt, f1-f12 etc.), a value of 256 is added to the output value in order to distinguish it from normal keys. The output will continue to output the last key value, until a new key is pressed or released. Notice that the output will be negative when a key is depressed. If ifn is set to the number of an already-allocated table having at least 512 elements, then the table element having index equal to the ascii code of the key pressed is set to 1, all other table elements are set to 0. This allows to check the state of a certain key or set of keys. Be aware that you must set the ikbdcapture parameter to something other than 0 on a designated FLpanel for FLkeyIn to capture keyboard events from that panel. FLkeyIn works internally at k-rate, so it can't be used in the header as other FLTK opcodes. It must be used inside an instrument. Examples Here is an example of the FLkeyIn opcode. It uses the file FLkeyIn.csd. Example of the FLkeyIn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/noise.xml0000644000000000000000000000633412262561504016246 0ustar rootroot noise Signal Generators:Random (Noise) Generators noise noise A white noise generator with an IIR lowpass filter. Description A white noise generator with an IIR lowpass filter. Syntax ares noise xamp, kbeta Performance xamp -- amplitude of final output kbeta -- beta of the lowpass filter. Should be in the range of -1 to 1, exclusive of the end-points. The filter equation is: [Filter equation for filter in noise opcode.] where xn is the original white noise and yn is lowpass filtered noise. The higher β is, the lower the filter's cut-off frequency. The cutoff frequency is roughly sr * ((1-kbeta)/2). Examples Here is an example of the noise opcode. It uses the file noise.csd. Example of the noise opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an example of the noise opcode controlling the kbeta parameter with a GUI interface. It uses the file noise-2.csd. Example of the noise opcode controlled with a GUI. Credits Author: &namejohn; University of Bath, Codemist. Ltd. Bath, UK December 2000 New in Csound version 4.10 manual_src~dfsg/opcodes/outz.xml0000644000000000000000000000305112262561504016123 0ustar rootroot outz Signal I/O:Signal Output outz outz Writes multi-channel audio data from a ZAK array to an external device or stream. Description Writes multi-channel audio data from a ZAK array to an external device or stream. Syntax outz ksig1 Performance outz outputs from a ZAK array for nchnls of audio. See Also out32, outc, outch, outx Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK May 2000 New in Csound Version 4.06 manual_src~dfsg/opcodes/flsetsnap.xml0000644000000000000000000001202512262561504017122 0ustar rootroot FLsetsnap FLTK:Other FLsetsnap FLsetsnap Stores the current status of all FLTK valuators into a snapshot location. Description FLsetsnap stores the current status of all valuators present in the orchestra into a snapshot location (in memory). Syntax inumsnap, inumval FLsetsnap index [, ifn, igroup] Initialization inumsnap -- current number of snapshots. inumval -- number of valuators (whose value is stored in a snapshot) present in current orchestra. index -- a number referring unequivocally to a snapshot. Several snapshots can be stored in the same bank. ifn (optional) -- optional argument referring to an already allocated table, to store values of a snapshot. igroup -- (optional) an integer number referring to a snapshot-related group of widget. It allows to get/set, or to load/save the state of a subset of valuators. Default value is zero that refers to the first group. The group number is determined by the opcode FLsetSnapGroup. The igroup parameter has not been yet fully implemented in the current version of csound. Please do not rely on it yet. Performance The FLsetsnap opcode stores current status of all valuators present in the orchestra into a snapshot location (in memory). Any number of snapshots can be stored in the current bank. Banks are structures that only exist in memory, there are no other reference to them other that they can be accessed by FLsetsnap, FLsavesnap, FLloadsnap and FLgetsnap opcodes. Only a single bank can be present in memory. If the optional ifn argument refers to an already allocated and valid table, the snapshot will be stored in the table instead of in the bank. So that table can be accessed from other Csound opcodes. The index argument unequivocally refers to a determinate snapshot. If the value of index refers to a previously stored snapshot, all its old values will be replaced with current ones. If index refers to a snapshot that doesn't exist, a new snapshot will be created. If the index value is not adjacent with that of a previously created snapshot, some empty snapshots will be created. For example, if a location with index 0 contains the only and unique snapshot present in a bank and the user stores a new snapshot using index 5, all locations between 1 and 4 will automatically contain empty snapshots. Empty snapshots don't contain any data and are neutral. FLsetsnap outputs the current number of snapshots (the inumsnap argument) and the total number of values stored in each snapshot (inumval). inumval is equal to the number of valuators present in the orchestra. For purposes of snapshot saving, widgets can be grouped, so that snapshots affect only a defined group of widgets. The opcode FLsetSnapGroup is used to specify the group for all widgets declared after it, until the next FLsetSnapGroup statement. See Also FLgetsnap, FLloadsnap, FLsetSnapGroup, FLrun, FLsavesnap, FLupdate Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/strlowerk.xml0000644000000000000000000000206412262561504017161 0ustar rootroot strlowerk Strings:Conversion strlowerk strlowerk Convert a string to lower case Description Convert Ssrc to lower case, and write the result to Sdst. strlowerk runs both at init and performance time. Syntax Sdst strlowerk Ssrc See also strlower Credits Author: Istvan Varga 2006 New in version 5.02 manual_src~dfsg/opcodes/strcpyk.xml0000644000000000000000000000337012262561504016625 0ustar rootroot strcpyk Strings:Manipulation strcpyk strcpyk Assign value to a string variable (k-rate) Description Assign to a string variable by copying the source which may be a constant or another string variable. strcpyk does the assignment both at initialization and performance time. Syntax Sdst strcpyk Ssrc Examples Here is an example of the strcpyk opcode. It uses the file strcpyk.csd. Example of the strcpyk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also strcpy Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/opcode.xml0000644000000000000000000004073512262561504016405 0ustar rootroot opcode Orchestra Syntax:Block Statements opcode opcode Defines the start of user-defined opcode block. Defining opcodes The opcode and endop statements allow defining a new opcode that can be used the same way as any of the built-in Csound opcodes. These opcode blocks are very similar to instruments (and are, in fact, implemented as special instruments), but cannot be called as a normal instrument e.g. with the i statements. A user-defined opcode block must precede the instrument (or other opcode) from which it is used. But it is possible to call the opcode from itself. This allows recursion of any depth that is limited only by available memory. Additionally, there is an experimental feature that allows running the opcode definition at a higher control rate than the kr value specified in the orchestra header. Similarly to instruments, the variables and labels of a user-defined opcode block are local and cannot be accessed from the caller instrument (and the opcode cannot access variables of the caller, either). Some parameters are automatically copied at initialization, however: all p-fields (including p1) extra time (see also xtratim, linsegr, and related opcodes). This may affect the operation of linsegr/expsegr/linenr/envlpxr in the user-defined opcode block. MIDI parameters, if there are any. Also, the release flag (see the release opcode) is copied at performance time. Modifying the note duration in the opcode definition by assigning to p3, or using ihold, turnoff, xtratim, linsegr, or similar opcodes will also affect the caller instrument. Changes to MIDI controllers (for example with ctrlinit) will also apply to the instrument from which the opcode was called. Use the setksmps opcode to set the local ksmps value. The xin and xout opcodes copy variables to and from the opcode definition, allowing communication with the calling instrument. The types of input and output variables are defined by the parameters intypes and outtypes. You can create UDOs which take no inputs or outputs by using 0 instead of a string. Notes xin and xout should be called only once, and xin should precede xout, otherwise an init error and deactivation of the current instrument may occur. These opcodes actually run only at i-time. Performance time copying is done by the user opcode call. This means that skipping xin or xout with kgoto has no effect, while skipping with igoto affects both init and performance time operation. Syntax opcode name, outtypes, intypes Initialization name -- name of the opcode. It may consist of any combination of letters, digits, and underscore but should not begin with a digit. If an opcode with the specified name already exists, it is redefined (a warning is printed in such cases). Some reserved words (like instr and endin) cannot be redefined. intypes -- list of input types, any combination of the characters: a, k, O, P, V, K, i, o, p, and j. A single 0 character can be used if there are no input arguments. Double quotes and delimiter characters (e.g. comma) are not needed. The meaning of the various intypes is shown in the following table: Type Description Variable Types Allowed Updated At a a-rate variable a-rate a-rate i i-rate variable i-rate i-time j optional i-time, defaults to -1 i-rate, constant i-time k k-rate variable k- and i-rate, constant k-rate O optional k-rate variable, defaults to 0 k- and i-rate, constant k-rate P optional k-rate variable, defaults to 1 k- and i-rate, constant k-rate V optional k-rate variable, defaults to 0.5 k- and i-rate, constant k-rate K k-rate with initialization k- and i-rate, constant i-time and k-rate o optional i-time, defaults to 0 i-rate, constant i-time p optional i-time, defaults to 1 i-rate, constant i-time S string variable i-rate string i-time The maximum allowed number of input arguments is 256. outtypes -- list of output types. The format is the same as in the case of intypes. Here are the available outtypes: Type Description Variable Types Allowed Updated At a a-rate variable a-rate a-rate i i-rate variable i-rate i-time k k-rate variable k-rate k-rate K k-rate with initialization k-rate i-time and k-rate The maximum allowed number of output arguments is 256. iksmps (optional, default=0) -- sets the local ksmps value. Must be a positive integer, and also the ksmps of the calling instrument or opcode must be an integer multiple of this value. For example, if ksmps is 10 in the instrument from which the opcode was called, the allowed values for iksmps are 1, 2, 5, and 10. If iksmps is set to zero, the ksmps of the caller instrument or opcode is used (this is the default behavior). Note The local ksmps is implemented by splitting up a control period into smaller sub-kperiods and temporarily modifying internal Csound global variables. This also requires converting the rate of k-rate input and output arguments (input variables receive the same value in all sub-kperiods, while outputs are written only in the last one). Warning about local <emphasis>ksmps</emphasis> When the local ksmps is not the same as the orchestra level ksmps value (as specified in the orchestra header), global a-rate operations must not be used in the user-defined opcode block. These include: any access to ga variables a-rate zak opcodes (zar, zaw, etc.) tablera and tablewa (these two opcodes may in fact work, but caution is needed) The in and out opcode family (these read from, and write to global a-rate buffers) In general, the local ksmps should be used with care as it is an experimental feature, although it works correctly in most cases. The setksmps statement can be used to set the local ksmps value of the user-defined opcode block. It has one i-time parameter specifying the new ksmps value (which is left unchanged if zero is used, see also the notes about iksmps above). setksmps should be used before any other opcodes (but allowed after xin), otherwise unpredictable results may occur. The input parameters can be read with xin, and the output is written by xout opcode. Only one instance of these units should be used, as xout overwrites and does not accumulate the output. The number and type of arguments for xin and xout must be the same as in the declaration of the user-defined opcode block (see tables above). The input and output arguments must agree with the definition both in number (except if the optional i-time input is used) and type. An optional i-time input parameter (iksmps) is automatically added to the intypes list, and (similarly to setksmps) sets the local ksmps value. Performance The syntax of a user-defined opcode block is as follows: opcode name, outtypes, intypes xinarg1 [, xinarg2] [, xinarg3] ... [xinargN] xin [setksmps iksmps] ... the rest of the instrument's code. xout xoutarg1 [, xoutarg2] [, xoutarg3] ... [xoutargN] endop The new opcode can then be used with the usual syntax: [xoutarg1] [, xoutarg2] ... [xoutargN] name [xinarg1] [, xinarg2] ... [xinargN] [, iksmps] Note The opcode call is always executed both at initialization and performance time, even if there are no a- or k-rate arguments. If there are many user opcode calls that are known to have no effect at performance time in an instrument, then it may save some CPU time to jump over groups of such opcodes with kgoto. Examples Here is an example of a user-defined opcode. It uses the file opcode.csd. Example of a user-defined opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also endop, setksmps, xin, xout Credits Author: Istvan Varga, 2002; based on code by Matt J. Ingalls New in version 4.22 manual_src~dfsg/opcodes/pvspitch.xml0000644000000000000000000001446412262561504016774 0ustar rootroot pvspitch Spectral Processing:Streaming pvspitch pvspitch Track the pitch and amplitude of a PVS signal. Description Track the pitch and amplitude of a PVS signal as k-rate variables. Syntax kfr, kamp pvspitch fsig, kthresh Performance kamp -- Amplitude of fundamental frequency kfr -- Fundamental frequency fsig -- an input pv stream kthresh -- analysis threshold (between 0 and 1). Higher values will eliminate low-amplitude components from the analysis. Performance The pitch detection algorithm implemented by pvspitch is based upon J. F. Schouten's hypothesis of the neural processes of the brain used to determine the pitch of a sound after the frequency analysis of the basilar membrane. Except for some further considerations, pvspitch essentially seeks out the highest common factor of an incoming sound's spectral peaks to find the pitch that may be attributed to it. In general, input sounds that exhibit pitch will also exhibit peaks in their spectrum according to where their harmonics lie. There are some the exceptions, however. Some sounds whose spectral representation is continuous can impart a sensation of pitch. Such sounds are explained by the centroid or center of gravity of the spectrum and are beyond the scope of the method of pitch detection implemented by pvspitch (Using opcodes like pvscent might be more appriopriate in these cases). pvspitch is able (using a previous analysis fsig generated by pvsanal) to locate the spectral peaks of a signal. The threshold parameter (kthresh) is of utmost importance, as adjusting it can introduce weak yet significant harmonics into the calculation of the fundamental. However, bringing kthresh too low would allow harmonically unrelated partials into the analysis algorithm and this will compromise the method's accuracy. These initial steps emulate the response of the basilar membrane by identifying physical characteristics of the input sound. The choice of kthresh depends on the actual level of the input signal, since its range (from 0 to 1) spans the whole dynamic range of an analysis bin (from -inf to 0dBFS). It is important to remember that the input to the pvspitch opcode is assumed to be characterised by strong partials within its spectrum. If this is not the case, the results outputted by the opcode may not bear any relation to the pitch of the input signal. If a spectral frame with many unrelated partials was analysed, the greatest common factor of these frequency values that allows for adjacent “harmonics” would be chosen. Thus, noisy frames can be characterised by low frequency outputs of pvspitch. This fact allows for a primitive type of instrumental transient detection, as the attack portion of some instrumental tones contain inharmonic components. Should the lowest frequency of the analysed melody be known, then all frequencies detected below this threshold are inaccurate readings, due to the presence of unrelated partials. In order to facilitate efficient testing of the pvspitch algorithm, an amplitude value proportional to the one in the observed in the signal frame is also outputted (kamp). The results of pvspitch can then be employed to drive an oscillator whose pitch can be audibly compared with that of the original signal (In the example below, an oscillator generates a signal which appears a fifth above the detected pitch). Examples Here is an example of the pvspitch opcode. It uses the file pvspitch.csd. This example uses realtime audio input but can be used for audiofile input as well. Example of the pvspitch opcode See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn, pvscent Credits Author: Alan OCinneide August 2005, added by &namevictor;, August 2006 Part of the text has been adapted from the Csound Journal winter 2006 issue's article "Introducing PVSPITCH: A pitch tracking opcode for Csound" by Alan OCinneide. The article is available at: www.csounds.com/journal/2006winter/pvspitch.html manual_src~dfsg/opcodes/ficlose.xml0000644000000000000000000000752112262561504016554 0ustar rootroot ficlose Signal I/O:File I/O ficlose ficlose Closes a previously opened file. Description ficlose can be used to close a file which was opened with fiopen. Syntax ficlose ihandle ficlose Sfilename Initialization ihandle -- a number which identifies this file (generated by a previous fiopen). Sfilename -- A string in double quotes or string variable with the filename. The full path must be given if the file directory is not in the system PATH and is not present in the current directory. Performance ficlose closes a file which was previously opened with fiopen. ficlose is only needed if you need to read a file written to during the same csound performance, since only when csound ends a performance does it close and save data in all open files. The opcode ficlose is useful for instance if you want to save presets within files which you want to be accesible without having to terminate csound. If you don't need this functionality it is safer not to call ficlose, and just let csound close the files when it exits. If a files closed with ficlose is being accessed by another opcode (like fout or foutk), it will be closed later when it is no longer being used. This opcode should be used with care, as the file handle will become invalid, and will cause an init error when an opcode tries to access the closed file. Examples Here is an example of the ficlose opcode. It uses the file ficlose.csd. Example of the ficlose opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also fiopen, fout, fouti, foutir, foutk Credits Author: &namegabriel; Italy 1999 New in Csound version 5.02 manual_src~dfsg/opcodes/specfilt.xml0000644000000000000000000000510512262561504016735 0ustar rootroot specfilt Spectral Processing:Non-Standard specfilt specfilt Filters each channel of an input spectrum. Description Filters each channel of an input spectrum. Syntax wsig specfilt wsigin, ifhtim Initialization ifhtim -- half-time constant. Performance wsigin -- the input spectrum. Filters each channel of an input spectrum. At each new frame of wsigin, each magnitude value is injected into a 1st-order lowpass recursive filter, whose half-time constant has been initially set by sampling the ftable ifhtim across the (logarithmic) frequency space of the input spectrum. This unit effectively applies a persistence factor to the data occurring in each spectral channel, and is useful for simulating the energy integration that occurs during auditory perception. It may also be used as a time-attenuated running histogram of the spectral distribution. Examples wsig2 specdiff wsig1 ; sense onsets wsig3 specfilt wsig2, 2 ; absorb slowly specdisp wsig2, 0.1 ; & display both spectra specdisp wsig3, 0.1 See Also specaddm, specdiff, spechist, specscal manual_src~dfsg/opcodes/midipolyaftertouch.xml0000644000000000000000000001171212262561504021040 0ustar rootroot midipolyaftertouch Real-time MIDI:MIDI/Score Interoperability midipolyaftertouch midipolyaftertouch Gets a MIDI polyphonic aftertouch value. Description midipolyaftertouch is designed to simplify writing instruments that can be used interchangeably for either score or MIDI input, and to make it easier to adapt instruments originally written for score input to work with MIDI input. In general, it should be possible to write instrument definitions that work identically with both scores and MIDI, including both MIDI files and real-time MIDI input, without using any conditional statements, and that take full advantage of MIDI voice messages. Note that correlating Csound instruments with MIDI channel numbers is done using the massign opcode for real-time performance,. For file-driven performance, instrument numbers default to MIDI channel number + 1, but the defaults are overridden by any MIDI program change messages in the file. Syntax midipolyaftertouch xpolyaftertouch, xcontrollervalue [, ilow] [, ihigh] Initialization ilow (optional) -- optional low value after rescaling, defaults to 0. ihigh (optional) -- optional high value after rescaling, defaults to 127. Performance xpolyaftertouch -- returns MIDI polyphonic aftertouch during MIDI activation, remains unchanged otherwise. xcontrollervalue -- returns the value of the MIDI controller during MIDI activation, remains unchanged otherwise. If the instrument was activated by MIDI input, the opcode overwrites the values of xpolyaftertouch and xcontrollervalue with the corresponding values from MIDI input. If the instrument was NOT activated by MIDI input, the values of xpolyaftertouch and xcontrollervalue remain unchanged. This enables score p-fields to receive MIDI input data during MIDI activation, and score values otherwise. Adapting a score-activated Csound instrument. See the MIDI interop opcodes section for details on adapting score driven instruments for MIDI or vice-versa. Examples Here is an example of the midipolyaftertouch opcode. It uses the file midipolyaftertouch.csd. Example of the midipolyaftertouch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i1 127.00000 i1 20.00000 i1 44.00000 See Also midichannelaftertouch, midicontrolchange, mididefault, midinoteoff, midinoteoncps, midinoteonkey, midinoteonoct, midinoteonpch, midipitchbend, midiprogramchange Credits Author: Michael Gogins New in version 4.20 manual_src~dfsg/opcodes/aresonk.xml0000644000000000000000000001016512262561504016570 0ustar rootroot aresonk Signal Modifiers:Standard Filters:Control aresonk aresonk A notch filter whose transfer functions are the complements of the reson opcode. Description A notch filter whose transfer functions are the complements of the reson opcode. Syntax kres aresonk ksig, kcf, kbw [, iscl] [, iskip] Initialization iscl (optional, default=0) -- coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. (This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise.) A zero value signifies no scaling of the signal, leaving that to some later adjustment (see balance). The default value is 0. iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance kres -- the output signal at control-rate. ksig -- the input signal at control-rate. kcf -- the center frequency of the filter, or frequency position of the peak response. kbw -- bandwidth of the filter (the Hz difference between the upper and lower half-power points). aresonk is a filter whose transfer functions is the complement of resonk. Thus aresonk is a notch filter whose transfer functions represents the filtered out aspects of their complements. However, power scaling is not normalized in aresonk but remains the true complement of the corresponding unit. Examples Here is an example of the aresonk opcode. It uses the file aresonk.csd. Example of the aresonk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also areson, atone, atonek, port, portk, reson, resonk, tone, tonek manual_src~dfsg/opcodes/hrtfreverb.xml0000644000000000000000000001161112262561504017274 0ustar rootroot hrtfreverb Signal Modifiers:Panning and Spatialization hrtfreverb hrtfreverb A binaural, dynamic FDN based diffuse-field reverberator. The opcode works independently as an efficient, flexible reverberator. Description A frequency-dependent, efficient reverberant field is created based on low and high frequency desired reverb times. The opcode is designed to work with hrtfearly, ideally using its outputs as inputs. However, hrtfreverb can be used as a standalone tool. Stability is enforced. It is, however, designed for use with hrtfearly to provide spatially accurate reverberation with user definable source trajectories. Accurate interaural coherence is also provided. Syntax aleft, aright, idel hrtfreverb asrc, ilowrt60, ihighrt60, ifilel, ifiler [,isr, imfp, iorder] Initialization ilowrt60 - low frequency reverb time. ihighrt60 - high frequency reverb time. ifilel - left HRTF spectral data file. ifiler - right HRTF spectral data file. Spectral datafiles (based on the MIT HRTF database) are available in 3 different sampling rates: 44.1, 48 and 96 kHz and are labelled accordingly. Input and processing sr should match datafile sr. Files should be in the current directory or the SADIR (see Environment Variables). HRTF Data files for use with hrtfmove, hrtfmove2, hrtfstat, hrtfreverb, hrtfreverb were updated for Csound 5.15 and later (the code was updated and is more efficient). Old datafiles are now deprecated. isr - optional, default 44.1kHz, legal values: 44100, 48000 and 96000. imfp - optional, mean free path, defaults to that of a medium room. If used with hrtfearly, the mean free path of the room can be used to calculate the appropriate delay for the later reverb. Legal range: the mean free path of the smallest room allowed by hrtfearly (0.003876) 1. iorder - optional, order of early reflection processing. If used with hrtfearly, the order of early reflections can be used to calculate the appropriate delay on the later reverb. Performance asrc - Input/source signal. Output idel - if used with hrtfearly, the appropriate delay for the later reverb, based on the room and order of processing. Example See the hrtfearly manual page for a simple example of the hrtfearly and hrtfreverb opcodes. See Also hrtfearly hrtfmove, hrtfmove2, hrtfstat, hrtfer. Credits Author: Brian Carty Maynooth 2011 manual_src~dfsg/opcodes/weibull.xml0000644000000000000000000001167312262561504016576 0ustar rootroot x-class noise generators weibull Signal Generators:Random (Noise) Generators weibull weibull Weibull distribution random number generator (positive values only). Description Weibull distribution random number generator (positive values only). This is an x-class noise generator Syntax ares weibull ksigma, ktau ires weibull ksigma, ktau kres weibull ksigma, ktau Performance ksigma -- scales the spread of the distribution. ktau -- if greater than one, numbers near ksigma are favored. If smaller than one, small values are favored. If t equals 1, the distribution is exponential. Outputs only positive numbers. For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the weibull opcode. It uses the file weibull.csd. Example of the weibull opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like these: i 1 time 0.00073: 168.59070 i 1 time 0.20027: 98.72078 i 1 time 0.40054: 48.57351 i 1 time 0.60009: 171.46941 i 1 time 0.80036: 50.20434 i 1 time 1.00063: 50.84775 i 1 time 1.20018: 18.16301 i 1 time 1.40045: 44.41001 i 1 time 1.60000: 0.98506 i 1 time 1.80027: 36.19192 WARNING: Seeding from current time 2444541554 i 2 time 3.00045: 20.81653 i 2 time 3.20000: 116.17060 i 2 time 3.40027: 9.23891 i 2 time 3.59982: 95.67111 i 2 time 3.80009: 296.52851 i 2 time 4.00036: 39.28636 i 2 time 4.19991: 13.54326 i 2 time 4.40018: 54.92388 i 2 time 4.59973: 268.05584 i 2 time 4.80000: 95.27069 i 2 time 5.00027: 91.62076 WARNING: Seeding from current time 2447542341 i 3 time 6.00091: 94.40902 i 3 time 6.20045: 111.10193 i 3 time 6.40073: 99.38797 i 3 time 6.60027: 98.54267 i 3 time 6.80054: 106.53899 i 3 time 7.00082: 106.30752 i 3 time 7.20036: 88.75486 i 3 time 7.40063: 106.45703 i 3 time 7.60091: 84.59854 i 3 time 7.80045: 106.76515 See Also seed, betarand, bexprnd, cauchy, exprand, gauss, linrand, pcauchy, poisson, trirand, unirand Credits Author: &nameparis; MIT, Cambridge 1995 manual_src~dfsg/opcodes/vport.xml0000644000000000000000000000340412262561504016276 0ustar rootroot vport Vectorial:Delay Paths vport vport Vectorial Control-rate Delay Paths Description Generate a sort of 'vectorial' portamento Syntax vport ifn, khtime, ielements [, ifnInit] Initialization ifn - number of the table containing the output vector ielements - number of elements of the two vectors ifnInit (optional) - number of the table containing a vector whose elements contain initial portamento values. Performance vport is similar to port, but operates with vectorial signals, instead of with scalar signals. Each vector element is treated as an independent control signal. Input and output vectors are placed in the same table and output vector overrides input vector. If you want to keep input vector, use vcopy opcode to copy it in another table. Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/vbap16.xml0000644000000000000000000001014212262561504016220 0ustar rootroot vbap16 Signal Modifiers:Panning and Spatialization vbap16 vbap16 Distributes an audio signal among 16 channels. Description Distributes an audio signal among 16 channels. Syntax ar1, ..., ar16 vbap16 asig, kazim [, kelev] [, kspread] Performance asig -- audio signal to be panned kazim -- azimuth angle of the virtual source kelev (optional) -- elevation angle of the virtual source kspread (optional) -- spreading of the virtual source (range 0 - 100). If value is zero, conventional amplitude panning is used. When kspread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. vbap16 takes an input signal, asig, and distribute it among 16 outputs, according to the controls kazim and kelev, and the configured loudspeaker placement. If idim = 2, kelev is set to zero. The distribution is performed using Vector Base Amplitude Panning (VBAP - See reference). VBAP distributes the signal using loudspeaker data configured with vbaplsinit. The signal is applied to, at most, two loudspeakers in 2-D loudspeaker configurations, and three loudspeakers in 3-D loudspeaker configurations. If the virtual source is panned outside the region spanned by loudspeakers, the nearest loudspeakers are used in panning. Please note that all vbap panning opcodes require the vbap system to be initialized using vbaplsinit. Examples See the entry for vbap8 for an example of usage of the vbap opcodes. Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. See Also vbap16move, vbap4, vbap4move, vbap8, vbap8move, vbaplsinit, vbapz, vbapzmove Credits Author: Ville Pulkki Sibelius Academy Computer Music Studio Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Helsinki, Finland May 2000 New in Csound Version 4.07. Input parameters accept k-rate since Csound 5.09. manual_src~dfsg/opcodes/flscroll_end.xml0000644000000000000000000000156212262561504017575 0ustar rootroot FLscrollEnd FLTK:Containers FLscroll_end FLscroll_end A FLTK opcode that marks the end of an area with scrollbars. Description A FLTK opcode that marks the end of an area with scrollbars. This is another name for FLscrollEnd provided for compatibility. See FLscrollEnd Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/abs.xml0000644000000000000000000000434412262561504015675 0ustar rootroot abs Mathematical Operations:Mathematical Functions abs abs Returns an absolute value. Description Returns the absolute value of x. Syntax abs(x) (no rate restriction) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Examples Here is an example of the abs opcode. It uses the file abs.csd. Example of the abs opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: instr 1: iabs = 0.000 instr 1: iabs = 0.150 instr 1: iabs = 13.000 See Also exp, frac, int, log, log10, i, sqrt manual_src~dfsg/opcodes/passign.xml0000644000000000000000000000454112262561504016573 0ustar rootroot passign Instrument Control:Initialization and Reinitialization passign passign Assigns a range of p-fields to ivariables. Description Assigns a range of p-fields to ivariables. Syntax ivar1, ... passign [istart] Initialisation The optional argument istart gives the index of the first p-field to assign. The default value is 1, corresponding to the instrument number. One of the variables can be a string variable, in which case it is assigned the only string parameter, if there is one, or an error otherwise. Performance passign transfers the instrument p-fields to instrument variables, starting with the one identified by the istart argument. There should not be more variables than p-fields, but there may be fewer. See Also assign, Example Here is an example of the passign opcode. It uses the file passign.csd. A variant of toot8.csd that uses passign. Credits Author: &namejohn; after an idea of Tobiah University of Bath, Codemist Ltd. Bath, UK December 2009 New in version 5.12 manual_src~dfsg/opcodes/ziw.xml0000644000000000000000000000531012262561504015733 0ustar rootroot ziw Zak Patch System ziw ziw Writes to a zk variable at i-rate without mixing. Description Writes to a zk variable at i-rate without mixing. Syntax ziw isig, indx Initialization isig -- initializes the value of the zk location. indx -- points to the zk or za location to which to write. Performance ziw writes isig into the zk variable specified by indx. These opcodes are fast, and always check that the index is within the range of zk or za space. If not, an error is reported, 0 is returned, and no writing takes place. Examples Here is an example of the ziw opcode. It uses the file ziw.csd. Example of the ziw opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zaw, zawm, ziwm, zkw, zkwm Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/xscanu.xml0000644000000000000000000001476512262561504016441 0ustar rootroot xscanu Signal Generators:Scanned Synthesis xscanu xscanu Compute the waveform and the wavetable for use in scanned synthesis. Description Experimental version of scanu. Allows much larger matrices and is faster and smaller but removes some (unused?) flexibility. If liked, it will replace the older opcode as it is syntax compatible but extended. Syntax xscanu init, irate, ifnvel, ifnmass, ifnstif, ifncentr, ifndamp, kmass, \ kstif, kcentr, kdamp, ileft, iright, kpos, kstrngth, ain, idisp, id Initialization init -- the initial position of the masses. If this is a negative number, then the absolute of init signifies the table to use as a hammer shape. If init > 0, the length of it should be the same as the intended mass number, otherwise it can be anything. irate -- update rate. ifnvel -- the ftable that contains the initial velocity for each mass. It should have the same size as the intended mass number. ifnmass -- ftable that contains the mass of each mass. It should have the same size as the intended mass number. ifnstif -- either an ftable that contains the spring stiffness of each connection. It should have the same size as the square of the intended mass number. The data ordering is a row after row dump of the connection matrix of the system. or a string giving the name of a file in the MATRIX format ifncentr -- ftable that contains the centering force of each mass. It should have the same size as the intended mass number. ifndamp -- the ftable that contains the damping factor of each mass. It should have the same size as the intended mass number. ileft -- If init < 0, the position of the left hammer (ileft = 0 is hit at leftmost, ileft = 1 is hit at rightmost). iright -- If init < 0, the position of the right hammer (iright = 0 is hit at leftmost, iright = 1 is hit at rightmost). idisp -- If 0, no display of the masses is provided. id -- If positive, the ID of the opcode. This will be used to point the scanning opcode to the proper waveform maker. If this value is negative, the absolute of this value is the wavetable on which to write the waveshape. That wavetable can be used later from an other opcode to generate sound. The initial contents of this table will be destroyed. Performance kmass -- scales the masses kstif -- scales the spring stiffness kcentr -- scales the centering force kdamp -- scales the damping kpos -- position of an active hammer along the string (kpos = 0 is leftmost, kpos = 1 is rightmost). The shape of the hammer is determined by init and the power it pushes with is kstrngth. kstrngth -- power that the active hammer uses ain -- audio input that adds to the velocity of the masses. Amplitude should not be too great. Matrix Format The new matrix format is a list of connections, one per line linking point x to point y. There is no weight given to the link; it is assumed to be unity. The list is proceeded by the line <MATRIX> and ends with a </MATRIX> line For example, a circular string of 8 would be coded as <MATRIX> 0 1 1 0 1 2 2 1 2 3 3 2 3 4 4 3 4 5 5 4 5 6 6 5 6 7 7 6 0 7 </MATRIX> Examples Here is an example of the xscanu opcode. It uses the file xscanu.csd. Example of the xscanu opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. For similar examples, see the documentation on scans. See Also More information on Scanned Synthesis (as well as several other matrices) is available on the Scanned Synthesis page at cSounds.com. Also an article on these opcodes: http://www.csounds.com/stevenyi/scanned/yi_scannedSynthesis.html , written by Steven Yi scanu, xscans Credits Written by &namejohn;. New in version 4.20 manual_src~dfsg/opcodes/planet.xml0000644000000000000000000000634012262561504016411 0ustar rootroot planet Signal Generators:Models and Emulations planet planet Simulates a planet orbiting in a binary star system. Description planet simulates a planet orbiting in a binary star system. The outputs are the x, y and z coordinates of the orbiting planet. It is possible for the planet to achieve escape velocity by a close encounter with a star. This makes this system somewhat unstable. Syntax ax, ay, az planet kmass1, kmass2, ksep, ix, iy, iz, ivx, ivy, ivz, idelta \ [, ifriction] [, iskip] Initialization ix, iy, iz -- the initial x, y and z coordinates of the planet ivx, ivy, ivz -- the initial velocity vector components for the planet. idelta -- the step size used to approximate the differential equation. ifriction (optional, default=0) -- a value for friction, which can be used to keep the system from blowing up iskip (optional, default=0) -- if non zero skip the initialisation of the filter. (New in Csound version 4.23f13 and 5.0) Performance ax, ay, az -- the output x, y, and z coodinates of the planet kmass1 -- the mass of the first star kmass2 -- the mass of the second star Examples Here is an example of the planet opcode. It uses the file planet.csd. Example of the planet opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also More information on this opcode: http://www.csounds.com/journal/issue9/FlutesInOrbit.html , written by Brian Redfern. Credits Author: &namehans; December 1998 New in Csound version 3.50 manual_src~dfsg/opcodes/chnset.xml0000644000000000000000000000572212262561504016415 0ustar rootroot chnset Signal I/O:Software Bus chnset chnset Writes data to the named software bus. Description Write to a channel of the named software bus. Implies declaring the channel with imod=2 (see also chn_k, chn_a, and chn_S). Syntax chnset ival, Sname chnset kval, Sname chnset aval, Sname chnset Sval, Sname Initialization Sname -- a string that indicates which named channel of the software bus to write. ival -- the control value to write at i-time. Sval -- the string value to write at i-time. Performance kval -- the control value to write at performance time. aval -- the audio signal to write at performance time. Example The example shows the software bus being used to write pitch information to a controlling program. sr = 44100 kr = 100 ksmps = 1 instr 1 a1 in kp,ka pitchamdf a1 chnset kp, "pitch" endin Here is another example of the chnset opcode. It uses the file chnset.csd. Example of the chnset opcode. Credits Author: &nameistvan; 2005 manual_src~dfsg/opcodes/ktableseg.xml0000644000000000000000000000130512262561504017063 0ustar rootroot ktableseg Spectral Processing:STFT ktableseg ktableseg Deprecated. Description Deprecated. Use the tableseg opcode instead. Syntax ktableseg ifn1, idur1, ifn2 [, idur2] [, ifn3] [...] manual_src~dfsg/opcodes/pdhalf.xml0000644000000000000000000001443112262561504016364 0ustar rootroot pdhalf Signal Modifiers:Waveshaping pdhalf pdhalf Distorts a phasor for reading the two halves of a table at different rates. Description The pdhalf opcode is designed to emulate the "classic" phase distortion synthesis method of the Casio CZ-series of synthesizers from the mid-1980's. This technique reads the first and second halves of a function table at different rates in order to warp the waveform. For example, pdhalf can smoothly transform a sine wave into something approximating the shape of a saw wave. Syntax aout pdhalf ain, kShapeAmount [, ibipolar [, ifullscale]] Initialization ibipolar -- an optional parameter specifying either unipolar (0) or bipolar (1) mode. Defaults to unipolar mode. ifullscale -- an optional parameter specifying the range of input and output values. The maximum will be ifullscale. The minimum depends on the mode of operation: zero for unipolar or -ifullscale for bipolar. Defaults to 1.0 -- you should set this parameter to the maximum expected input value. Performance ain -- the input signal to be distorted. aout -- the output signal. kShapeAmount -- the amount of distortion applied to the input. Must be between negative one and one (-1 to 1). An amount of zero means no distortion. Transfer function created by pdhalf and a negative kShapeAmount. Transfer function created by pdhalf and a negative kShapeAmount. The pdhalf opcode calculates a transfer function that is composed of two linear segments (see the graph). These segments meet at a "pivot point" which always lies on the same horizontal axis. (In unipolar mode, the axis is y = 0.5, and for bipolar mode it is the x axis). The kShapeAmount parameter specifies where on the horizontal axis this point falls. When kShapeAmount is zero, the pivot point is in the middle of the input range, forming a straight line for the transfer function and thus causing no change in the input signal. As kShapeAmount changes from zero (0) to negative one (-1), the pivot point moves towards the left side of the graph, producing a phase distortion pattern like the Casio CZ's "sawtooth waveform". As it changes from zero (0) to positive one (1), the pivot point moves toward the right, producing an inverted pattern. If the input to pdhalf is a phasor and the output is used to index a table, values for kShapeAmount that are less than zero will cause the first half of the table to be read more quickly than the second half. The reverse is true for values of kShapeAmount greater than zero. The rates at which the halves are read are calculated so that the frequency of the phasor is unchanged. Thus, this method of phase distortion can only produce higher partials in a harmonic series. It cannot produce inharmonic sidebands in the way that frequency modulation does. pdhalf can work in either unipolar or bipolar modes. Unipolar mode is appropriate for signals like phasors that range between zero and some maximum value (selectable with ifullscale). Bipolar mode is appropriate for signals that range above and below zero by roughly equal amounts such as most audio signals. Applying pdhalf directly to an audio signal in this way results in a crude but adjustable sort of waveshaping/distortion. A typical example of the use of pdhalf is aphase phasor ifreq apd pdhalf aphase, kamount aout tablei apd, 1, 1 Examples Here is an example of the pdhalf opcode. It uses the file pdhalf.csd. Example of the pdhalf opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pdhalfy, pdclip More information about phase distortion on Wikipedia: http://en.wikipedia.org/wiki/Phase_distortion_synthesis Credits Author: &nameanthony; January 2008 New in Csound version 5.08 manual_src~dfsg/opcodes/pvsbandp.xml0000644000000000000000000000656212262561504016751 0ustar rootroot pvsbandp Spectral Processing:Streaming pvsbandp pvsbandp A band pass filter working in the spectral domain. Description Filter the pvoc frames, passing bins whose frequency is within a band, and with linear interpolation for transitional bands. Syntax fsig pvsbandp fsigin, xlowcut, xlowfull, \ xhighfull, xhighcut[, ktype] Performance fsig -- output pv stream fsigin -- input pv stream. xlowcut, xlowfull, xhighfull, xhighcut -- define a trapezium shape for the band that is passed. The a-rate versions only apply to the sliding case. ktype -- specifies the shape of the transitional band. If at the default value of zero the shape is as below, with linear transition in amplitude. Other values yield and exponential shape: (1 - exp( r*type )) / (1 - exp(type)) This includes a linear dB shape when ktype is log(10) or about 2.30. The opcode performs a band-pass filter with a spectral envelope shaped like klowfull __________________________ khighfull / \ / \ / \ / \ / \ ________/ \______________ klowcut khighcut Examples Here is an example of the pvsbandp opcode. It uses the file pvsbandp.csd. Example of the pvsbandp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn, pvsbandr Credits Author: &namejohn; December 2007 manual_src~dfsg/opcodes/filesr.xml0000644000000000000000000000552712262561504016420 0ustar rootroot filesr Signal I/O:Soundfile Queries filesr filesr Returns the sample rate of a sound file. Description Returns the sample rate of a sound file. Syntax ir filesr ifilcod [, iallowraw] Initialization ifilcod -- sound file to be queried iallowraw -- (Optional) Allow raw sound files (default=1) Performance filesr returns the sample rate of the sound file ifilcod. filesr can return the sample rate of convolve and PVOC files if the iallowraw flag is not zero (it is non-zero by default). Examples Here is an example of the filesr opcode. It uses the file filesr.csd, and beats.wav. Example of the filesr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The audio file beats.wav was sampled at 44.1 KHz. So filesr's output should include a line like this: sample rate = 44100.000000, size = 131072.000000 See Also filebit, filelen, filenchnls, filepeak Credits Author: &namematt; July 1999 New in Csound version 3.57 manual_src~dfsg/opcodes/hrtfstat.xml0000644000000000000000000001400012262561504016755 0ustar rootroot hrtfstat Signal Modifiers:Panning and Spatialization hrtfstat hrtfstat Generates static 3d binaural audio for headphones using a Woodworth based spherical head model with improved low frequency phase accuracy. Description This opcode takes a source signal and spatialises it in the 3 dimensional space around a listener using head related transfer function (HRTF) based filters. It produces a static output (azimuth and elevation parameters are i-rate), because a static source allows much more efficient processing than hrtfmove and hrtfmove2,. Syntax aleft, aright hrtfstat asrc, iAz, iElev, ifilel, ifiler [,iradius, isr] Initialization iAz -- azimuth value in degrees. Positive values represent position on the right, negative values are positions on the left. iElev -- elevation value in degrees. Positive values represent position above horizontal, negative values are positions below horizontal (min -40). ifilel -- left HRTF spectral data file ifiler -- right HRTF spectral data file Spectral datafiles (based on the MIT HRTF database) are available in 3 different sampling rates: 44.1, 48 and 96 khz and are labelled accordingly. Input and processing sr should match datafile sr. Files should be in the current directory or the SADIR (see Environment Variables). HRTF Data files for use with hrtfmove, hrtfmove2, hrtfstat, hrtfearly, hrtfreverb were updated for Csound 5.15 and later (the code was updated and is more efficient). Old datafiles are now deprecated. iradius -- optional, head radius used for phase spectra calculation in centimeters (default 9.0) isr - optional (default 44.1kHz). Legal values are 44100, 48000 and 96000. Performance Artifact-free user-defined static spatialisation is made possible using an interpolation algorithm based on spectral magnitude interpolation and a derived phase based on the Woodworth spherical head model. Accuracy is increased for the data set provided by extracting and applying a frequency dependent scaling factor to the phase spectra, leading to a more precise low frequency interaural time difference. Users can control head radius for the phase derivation, allowing a crude level of individualisation. The static source version of the opcode uses overlap add convolution (it does not need STFT processing, see hrtfmove2), and is thus considerably more efficient than hrtfmove2 or hrtfmove, but cannot generate moving sources. Examples Here is an example of the hrtfstat opcode. It uses the file hrtfstat.csd. Example of the hrtfstat opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the hrtfstat opcode. It uses the file htrfstat-2.csd, and Church.wav, which is a looped sample. Example two of the hrtfstat opcode See Also hrtfmove, hrtfmove2, hrtfer. More information on this opcode: http://www.csounds.com/journal/issue9/newHRTFOpcodes.html , written by Brian Carty Credits Author: Brian Carty Maynooth 2008 manual_src~dfsg/opcodes/ftloadk.xml0000644000000000000000000000433312262561504016552 0ustar rootroot ftloadk Table Control:Read/Write Opreations ftloadk ftloadk Load a set of previously-allocated tables from a file. Description Load a set of previously-allocated tables from a file. Syntax ftloadk "filename", ktrig, iflag, ifn1 [, ifn2] [...] Initialization "filename" -- A quoted string containing the name of the file to load. iflag -- Type of the file to load/save. (0 = binary file, Non-zero = text file) ifn1, ifn2, ... -- Numbers of tables to load. Performance ktrig -- The trigger signal. Load the file each time it is non-zero. ftloadk loads a list of tables from a file. (The tables have to be already allocated though.) The file's format can be binary or text. Unlike ftload, the loading operation can be repeated numerous times within the same note by using a trigger signal. Warning The file's format is not compatible with a WAV-file and is not endian-safe. See Also ftload, ftsavek, ftsave Credits Author: &namegabriel; New in version 4.21 manual_src~dfsg/opcodes/massign.xml0000644000000000000000000000744212262561504016573 0ustar rootroot massign Real-time MIDI:Input massign massign Assigns a MIDI channel number to a Csound instrument. Description Assigns a MIDI channel number to a Csound instrument. Syntax massign ichnl, insnum[, ireset] massign ichnl, "insname"[, ireset] Initialization ichnl -- MIDI channel number (1-16). insnum -- Csound orchestra instrument number. If zero or negative, the channel is muted (i.e. it doesn't trigger a csound instrument, though information will still be received by opcodes like midiin). insname -- A string (in double-quotes) representing a named instrument. ireset -- If non-zero resets the controllers; default is to reset. Performance Assigns a MIDI channel number to a Csound instrument. Also useful to make sure a certain instrument (if its number is from 1 to 16) will not be triggered by midi noteon messages (if using something midiin to interpret midi information). In this case set insnum to 0 or a negative number. If ichan is set to 0, the value of insnum is used for all channels. This way you can route all MIDI channels to a single Csound instrument. You can also disable triggering of instruments from MIDI note events from all channels with the following line: massign 0,0 This can be useful if you are doing all MIDI evaluation within Csound with an always on instrument(e.g. using midiin and turnon) to avoid doubling the instrument when a note is played. Examples Here is an example of the massign opcode. It uses the file massign.csd. Example of the massign opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also ctrlinit Credits Author: &namebarry; - &namemike; MIT, Cambridge, Mass. New in Csound version 3.47 ireset parameter new in Csound5 Thanks goes to &namerasmus; for pointing out the correct MIDI channel range. manual_src~dfsg/opcodes/delayk.xml0000644000000000000000000000525612262561504016404 0ustar rootroot delayk Signal Modifiers:Delay delayk delayk Delays an input signal by some time interval. Description k-rate delay opcodes Syntax kr delayk ksig, idel[, imode] kr vdel_k ksig, kdel, imdel[, imode] Initialization idel -- delay time (in seconds) for delayk. It is rounded to the nearest integer multiple of a k-cycle (i.e. 1/kr). imode -- sum of 1 for skipping initialization (e.g. in tied notes) and 2 for holding the first input value during the initial delay, instead of outputting zero. This is mainly of use when delaying envelopes that do not start at zero. imdel -- maximum delay time for vdel_k, in seconds. Performance kr -- the output signal. Note: neither of the opcodes interpolate the output. ksig -- the input signal. kdel -- delay time (in seconds) for vdel_k. It is rounded to the nearest integer multiple of a k-cycle (i.e. 1/kr). Examples Here is an example of the delayk opcode. It uses the file delayk.csd. Example of the delayk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits &nameistvan;. manual_src~dfsg/opcodes/out.xml0000644000000000000000000000632212262561504015735 0ustar rootroot out Signal I/O:Signal Output out out Writes audio data to an external device or stream. Description Writes audio data to an external device or stream, either from audio variables or from an audio array. Syntax out asig1[, asig2,....] out aarray Performance Sends audio samples to an accumulating output buffer (created at the beginning of performance) which serves to collect the output of all active instruments before the sound is written to disk. There can be any number of these output units in an instrument. The type (mono, stereo, quad, hex, or oct) should agree with nchnls. But as of version 3.50, Csound will attempt to change an incorrect opcode to agree with nchnls statement. Examples Here is an example of the out opcode. It uses the file out.csd. Example of the out opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also outh, outo, outq, outq1, outq2, outq3, outq4, outs, outs1, outs2, soundout Credits Author: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 Author: &namejohn; NUIM, 2013 Original in Csound v1 Aray version in 6.01 manual_src~dfsg/opcodes/signum.xml0000644000000000000000000000321212262561504016423 0ustar rootroot signum Mathematical Operations:Trigonometric Functions signum signum Performs a signum function. Description Returns the signum of x returning -1, 0 or 1. Syntax signum(x) (no rate restriction) Examples Here is an example of the signum opcode. It uses the file signum.csd. Example of the signum opcode. See the sections Real-time Audio and Command Line Flags for more information on usingg command line flags. Credits Author: &namejohn; University of Bath, Codemist Ltd. Bath, UK July 2013 New in Csound version 6.01 manual_src~dfsg/opcodes/polynomial.xml0000644000000000000000000000553712262561504017320 0ustar rootroot polynomial Mathematical Operations:Opcode Equivalents of Functions polynomial polynomial Efficiently evaluates a polynomial of arbitrary order. Description The polynomial opcode calculates a polynomial with a single a-rate input variable. The polynomial is a sum of any number of terms in the form kn*x^n where kn is the nth coefficient of the expression. These coefficients are k-rate values. Syntax aout polynomial ain, k0 [, k1 [, k2 [...]]] Performance ain -- the input signal used as the independent variable of the polynomial ("x"). aout -- the output signal ("y"). k0, k1, k2, ... -- the coefficients for each term of the polynomial. If we consider the input parameter ain to be "x" and the output aout to be "y", then the polynomial opcode calculates the following equation: y = k0 + k1*x + k2*x^2 + k3*x^3 + ... Examples Here is an example of the polynomial opcode. It uses the file polynomial.csd. Example of the polynomial opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also chebyshevpoly, mac maca Credits Author: &nameanthony; January 2008 New in Csound version 5.08 manual_src~dfsg/opcodes/outh.xml0000644000000000000000000000432312262561504016104 0ustar rootroot outh Signal I/O:Signal Output outh outh Writes 6-channel audio data to an external device or stream. Description Writes 6-channel audio data to an external device or stream. Syntax outh asig1, asig2, asig3, asig4, asig5, asig6 Performance Sends 6-channel audio samples to an accumulating output buffer (created at the beginning of performance) which serves to collect the output of all active instruments before the sound is written to disk. There can be any number of these output units in an instrument. The type (mono, stereo, quad, hex, or oct) should agree with nchnls. But as of version 3.50, Csound will attempt to change an incorrect opcode to agree with nchnls statement. See Also out, outo, outq, outq1, outq2, outq3, outq4, outs, outs1, outs2, soundout Credits Author: &namejohn; Introduced before Version 3 manual_src~dfsg/opcodes/follow.xml0000644000000000000000000000454312262561504016433 0ustar rootroot follow Instrument Control:Sensing and Control follow follow Envelope follower unit generator. Description Envelope follower unit generator. Syntax ares follow asig, idt Initialization idt -- This is the period, in seconds, that the average amplitude of asig is reported. If the frequency of asig is low then idt must be large (more than half the period of asig ) Performance asig -- This is the signal from which to extract the envelope. Examples Here is an example of the follow opcode. It uses the file follow.csd, and beats.wav. Example of the follow opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. To avoid zipper noise, by discontinuities produced from complex envelope tracking, a lowpass filter could be used, to smooth the estimated envelope. Credits Author: &nameparis; MIT, Cambridge 1995 manual_src~dfsg/opcodes/genarray.xml0000644000000000000000000000427312262561504016741 0ustar rootroot genarray Table Control:Table Queries genarray genarray Generate a vector with an arithmetic sequence. Description Generate a vector (one-dimensional k-rate or i-rate array) with an arithmetic sequence. Syntax karray genarray kstart, kens[, inc] iarray genarray istart, iens[, inc] Initialization istart -- value to place in first element. iend -- last value to place in array. inc -- amount to add to previous value (default 1). Examples Here is an example of the genarray opcode. It uses the file genarray.csd. Example of the genarray opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also Vectorial opcodes Credits Author: &namejohn; Codemist Ltd 2013 New in version 6.00 manual_src~dfsg/opcodes/flsetcolor2.xml0000644000000000000000000000614512262561504017367 0ustar rootroot FLsetColor2 FLTK:Appearance FLsetColor2 FLsetColor2 Sets the secondary (or selection) color of a FLTK widget. Description FLsetColor2 sets the secondary (or selection) color of the target widget. Syntax FLsetColor2 ired, igreen, iblue, ihandle Initialization ired -- The red color of the target widget. The range for each RGB component is 0-255 igreen -- The green color of the target widget. The range for each RGB component is 0-255 iblue -- The blue color of the target widget. The range for each RGB component is 0-255 ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/slider64tablef.xml0000644000000000000000000001527012262561504017742 0ustar rootroot slider64tablef Real-time MIDI:Slider Banks slider64tablef slider64tablef Stores a bank of 64 different MIDI control messages to a table, filtered before output. Description Stores a bank of 64 different MIDI MIDI control messages to a table, filtered before output. Syntax kflag slider64tablef ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, icutoff1, .... , ictlnum64, imin64, imax64, init64, ifn64, icutoff64 Initialization ichan -- MIDI channel (1-16) ioutTable -- number of the table that will contain the output ioffset -- output table offset. A zero means that the output of the first slider will affect the first table element. A 10 means that the output of the first slider will affect the 11th table element. ictlnum1 ... ictlnum64 -- MIDI control number (0-127) imin1 ... imin64 -- minimum values for each controller imax1 ... imax64 -- maximum values for each controller init1 ... init64 -- initial value for each controller ifn1 ... ifn64 -- function table for conversion for each controller icutoff1 ... icutoff64 -- low-pass filter cutoff frequency for each controller Performance kflag -- a flag that informs if any control-change message in the bank has been received. In this case kflag is set to 1. Otherwise is set to zero. slider64tablef is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider64tablef allows a bank of 64 different MIDI control message numbers. It filters the signal before output. This eliminates discontinuities due to the low resolution of the MIDI (7 bit). The cutoff frequency can be set separately for each controller (suggested range: .1 to 5 Hz). As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. slider64tablef is very similar to slider64f and sliderN family of opcodes (see their manual for more information). The actual difference is that the output is not stored to k-rate variables, but to a table, denoted by the ioutTable argument. It is possible to define a starting index in order to use the same table for more than one slider bank (or other purposes). It is possible to use this opcode together with FLslidBnk2Setk and FLslidBnk2, so you can synchronize the position of the MIDI values to the position of the FLTK valuator widgets of FLslidBnk2. Notice that you have to specify the same min/max values as well the linear/exponential responses in both sliderNtable(f) and FLslidBnk2. The exception is when using table-indexed response instead of a lin/exp response. In this case, in order to achieve a useful result, the table-indexed response and actual min/max values must be set only in FLslidBnk2, whereas, in sliderNtable(f), you have to set a linear response and a minimum of zero and a maximum of one in all sliders. slider64tablef opcodes do not output the required initial value immediately, but only after some k-cycles because the filter slightly delays the output. See Also slider16table, slider16tablef, slider32table, slider32tablef, slider64table, slider8table, slider8tablef Credits Author: &namegabriel; New in Csound version 5.06 manual_src~dfsg/opcodes/endif.xml0000644000000000000000000000505412262561504016214 0ustar rootroot endif Instrument Control:Program Flow Control endif endif Closes a block of code that begins with an "if...then" statement. Description Closes a block of code that begins with an "if...then" statement. Syntax endif Performance Any block of code that begins with an "if...then" statement must end with an endif statement. Examples Here is an example of the endif opcode. It uses the file endif.csd. Example of the endif opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also elseif, else, goto, if, igoto, kgoto, tigoto, timout More information on this opcode: http://www.csounds.com/journal/2006spring/controlFlow.html , written by Steven Yi Credits New in version 4.21 manual_src~dfsg/opcodes/cosseg.xml0000644000000000000000000000720312262561504016410 0ustar rootroot cosseg Signal Generators:Linear and Exponential Generators cosseg cosseg Trace a series of line segments between specified points with cosine interpolation. Description Trace a series of line segments between specified points with cosine interpolation. Syntax ares cosseg ia, idur1, ib [, idur2] [, ic] [...] kres cosseg ia, idur1, ib [, idur2] [, ic] [...] Initialization ia -- starting value. ib, ic, etc. -- value after dur1 seconds, etc. idur1 -- duration in seconds of first segment. A zero or negative value will cause all initialization to be skipped. idur2, idur3, etc. -- duration in seconds of subsequent segments. A zero or negative value will terminate the initialization process with the preceding point, permitting the last-defined line or curve to be continued indefinitely in performance. The default is zero. Performance These units generate control or audio signals whose values can pass through 2 or more specified points. The sum of dur values may or may not equal the instrument's performance time: a shorter performance will truncate the specified pattern, while a longer one will cause the last value to be repeated until the end of the note. Examples Here is an example of the cosseg opcode. It uses the file cosseg.csd. Example of the cosseg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also expon, expseg, expsegb, expsegr, line, linseg, linsegr transeg transegb Credits Author: &namejohn; Aug 2012 New in version 5.18 manual_src~dfsg/opcodes/printk2.xml0000644000000000000000000000565212262561504016524 0ustar rootroot printk2 Signal I/O:Printing and Display printk2 printk2 Prints a new value every time a control variable changes. Description Prints a new value every time a control variable changes. Syntax printk2 kvar [, inumspaces] Initialization inumspaces (optional, default=0) -- number of space characters printed before the value of kvar Performance kvar -- signal to be printed Derived from Robin Whittle's printk, prints a new value of kvar each time kvar changes. Useful for monitoring MIDI control changes when using sliders. WARNING! Don't use this opcode with normal, continuously variant k-signals, because it can hang the computer, as the rate of printing is too fast. Examples Here is an example of the printk2 opcode. It uses the file printk2.csd. Example of the printk2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i1 0.00000 i1 3.00000 i1 1.00000 i1 3.00000 i1 1.00000 i1 2.00000 i1 3.00000 ..... See Also printk and printks Credits Author: &namegabriel; Italy 1998 New in Csound version 3.48 manual_src~dfsg/opcodes/times.xml0000644000000000000000000000577212262561504016257 0ustar rootroot times Instrument Control:Time Reading times times Read absolute time in seconds. Description Read absolute time, in seconds, since the start of the performance. Syntax ires times kres times Performance Time in seconds is available with times. This would return 0.5 after half a second. times can both produce a k-rate variable for output. There are no input parameters. times can also operate at the start of the instance of the instrument. It produces an i-rate variable (starting with i or gi) as its output. Examples Here is an example of the times opcode. It uses the file times_complex.csd. Example of the times opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: new alloc for instr again: instance = 1, start = 0.000000, duration = 0.650439 new alloc for instr fx_processor: instance = 2, start = 0.650884, duration = 0.411043 new alloc for instr fx_processor: instance = 3, start = 1.061587, duration = 0.231085 new alloc for instr fx_processor: instance = 4, start = 1.292336, duration = 0.543473 new alloc for instr fx_processor: instance = 5, start = 1.835828, duration = 1.777097 See Also timeinstk, timeinsts, timek Credits Author: Robin Whittle Australia May 1997 manual_src~dfsg/opcodes/kexprand.xml0000644000000000000000000000077612262561504016751 0ustar rootroot kexprand kexprand kexprand Deprecated. Description Deprecated as of version 3.49. Use the exprand opcode instead. manual_src~dfsg/opcodes/flslidbnk2setk.xml0000644000000000000000000001327712262561504020056 0ustar rootroot FLslidBnk2Setk FLTK:Other FLslidBnk2Setk FLslidBnk2Setk modify the values of a slider bank. Description FLslidBnk2Setk modifies the values of a slider bank according to an array of values stored in a table. Syntax FLslidBnk2Setk ktrig, ihandle, ifn [, istartIndex, istartSlid, inumSlid] Initialization ihandle - handle of the sliderBnk (to be used to set its values). ifn - number of a table containing an array of values to set each slider to. istartIndex - (optional) starting index of the table element of to be evaluated firstly. Default value is zero istartSlid - (optional) starting slider to be evaluated. Default 0, denoting the first slider. inumSlid - (optional) number of sliders to be updated. Default 0, denoting all sliders. Performance ktrig – the output of FLslidBnk2Setk consists of a trigger that informs if sliders have to be updated or not. A non-zero value forces the slider to be updated. FLslidBnk2Setk is similar to FLslidBnkSet but allows k-rate to modify the values of FLslidBnk2 (FLslidBnk2Setk can also be used with FLvslidBnk2, obtaining identical result). It also allows the slider bank to be joined with MIDI. If you are using MIDI (for instance, when using the slider8table opcode), FLslidBnk2Setk changes the values of FLslidBnk2 bank of sliders to a set of values located in a table. This opcode is actually able to serve as a MIDI bridge to the FLslidBnk2 widget when used together with the sliderXXtable set of opcodes (see slider8table entry for more information). Notice, that, when you want to use table indexing as a curve response, it is not possible to do it directly in the iconfigtable configuration of FLslidBnk2, when you intend to use the FLslidBnk2Setk opcode. In fact, corresponding inputTable element of FLslidBnk2Setk must be set in linear mode and respect the 0 to 1 range. Even the corresponding elements of sliderXXtable must be set in linear mode and in the normalized range. You can do table indexing later, by using the tab and tb opcodes, and rescaling output according to max and min values. By the other hand, it is possible to use linear and exponential curve response directly, by setting the actual min-max range and flag both in the iconfigtable of corresponding FLslidBnk2 and in sliderXXtable. FLslidBnk2Setk the k-rate version of FLslidBnk2Set. Examples Here is an example of the FLslidBnk2Setk opcode. It uses the file FLslidBnk2Setk.csd. Example of the FLslidBnk2Setk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLslider, FLslidBnkGetHandle, FLslidBnk, FLslidBnk2, FLvslidBnk, FLvslidBnk2 FLslidBnkSet, FLslidBnk2Set, slider8table Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/flpack_end.xml0000644000000000000000000000154212262561504017213 0ustar rootroot FLpack_end FLTK:Containers FLpack_end FLpack_End Marks the end of a group of compressed or aligned FLTK widgets. Description Marks the end of a group of compressed or aligned FLTK widgets. This is another name for FLpackEnd provided for compatibility. See FLpackEnd Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/greaterequal.xml0000644000000000000000000000752212262561504017612 0ustar rootroot : ? conditional expressions >= Instrument Control:Conditional Values >= >= Determines if one value is greater than or equal to another. Description Determines if one value is greater than or equal to another. Syntax (a >= b ? v1 : v2) where a, b, v1 and v2 may be expressions, but a, b not audio-rate. Performance In the above conditional, a and b are first compared. If the indicated relation is true (a greater than or equal to b), then the conditional expression has the value of v1; if the relation is false, the expression has the value of v2. NB.: If v1 or v2 are expressions, these will be evaluated before the conditional is determined. In terms of binding strength, all conditional operators (i.e. the relational operators (<, etc.), and ?, and : ) are weaker than the arithmetic and logical operators (+, -, *, /, && and ||). These are operators not opcodes. Therefore, they can be used within orchestra statements, but do not form complete statements themselves. Examples Here is an example of the >= operator. It uses the file greaterequal.csd. Example of the >= operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: k1 = 2.000000, k2 = 0.000000 k1 = 3.000000, k2 = 1.000000 k1 = 4.000000, k2 = 1.000000 See Also ==, >, <=, <, != Credits Example written by &namekevin;. manual_src~dfsg/opcodes/veloc.xml0000644000000000000000000000533112262561504016235 0ustar rootroot veloc Real-time MIDI:Input veloc veloc Get the velocity from a MIDI event. Description Get the velocity from a MIDI event. Syntax ival veloc [ilow] [, ihigh] Initialization ilow, ihigh -- low and hi ranges for mapping Performance Get the MIDI byte value (0 - 127) denoting the velocity of the current event. Examples Here is an example of the veloc opcode. It uses the file veloc.csd. Example of the veloc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also aftouch, ampmidi, cpsmidi, cpsmidib, midictrl, notnum, octmidi, octmidib, pchbend, pchmidi, pchmidib Credits Author: &namebarry; - &namemike; MIT - Mills May 1997 manual_src~dfsg/opcodes/tablexkt.xml0000644000000000000000000001266612262561504016754 0ustar rootroot tablexkt Table Control:Dynamic Selection tablexkt tablexkt Reads function tables with linear, cubic, or sinc interpolation. Description Reads function tables with linear, cubic, or sinc interpolation. Syntax ares tablexkt xndx, kfn, kwarp, iwsize [, ixmode] [, ixoff] [, iwrap] Initialization iwsize -- This parameter controls the type of interpolation to be used: 2: Use linear interpolation. This is the lowest quality, but also the fastest mode. 4: Cubic interpolation. Slightly better quality than iwsize = 2, at the expense of being somewhat slower. 8 and above (up to 1024): sinc interpolation with window size set to iwsize (should be an integer multiply of 4). Better quality than linear or cubic interpolation, but very slow. When transposing up, a kwarp value above 1 can be used for anti-aliasing (this is even slower). ixmode1 (optional) -- index data mode. The default value is 0. 0: raw index any non-zero value: normalized (0 to 1) Notes if tablexkt is used to play back samples with looping (e.g. table index is generated by lphasor), there must be at least iwsize / 2 extra samples after the loop end point for interpolation, otherwise audible clicking may occur (also, at least iwsize / 2 samples should be before the loop start point). ixoff (optional) -- amount by which index is to be offset. For a table with origin at center, use tablesize / 2 (raw) or 0.5 (normalized). The default value is 0. iwrap (optional) -- wraparound index flag. The default value is 0. 0: Nowrap (index < 0 treated as index = 0; index >= tablesize (or 1.0 in normalized mode) sticks at the guard point). any non-zero value: Index is wrapped to the allowed range (not including the guard point in this case). Note iwrap also applies to extra samples for interpolation. Performance ares -- audio output xndx -- table index kfn -- function table number kwarp -- if greater than 1, use sin (x / kwarp) / x function for sinc interpolation, instead of the default sin (x) / x. This is useful to avoid aliasing when transposing up (kwarp should be set to the transpose factor in this case, e.g. 2.0 for one octave), however it makes rendering up to twice as slow. Also, iwsize should be at least kwarp * 8. This feature is experimental, and may be optimized both in terms of speed and quality in new versions. Note kwarp has no effect if it is less than, or equal to 1, or linear or cubic interpolation is used. Examples Here is an example of the tablexkt opcode. It uses the file tablexkt.csd. Example of the tablexkt opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &nameistvan; January 2002 Example by: Jonathan Murphy 2006 New in version 4.18 manual_src~dfsg/opcodes/resonxk.xml0000644000000000000000000000705112262561504016617 0ustar rootroot resonxk Signal Modifiers:Standard Filters:Control resonxk resonxk Control signal resonant filter stack. Description resonxk is equivalent to a group of resonk filters, with the same arguments, serially connected. Using a stack of a larger number of filters allows a sharper cutoff. Syntax kres resonxk ksig, kcf, kbw[, inumlayer, iscl, istor] Initialization inumlayer - number of elements of filter stack. Default value is 4. Maximum value is 10 iscl (optional, default=0) - coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. (This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise.) A zero value signifies no scaling of the signal, leaving that to some later adjustment (see balance). The default value is 0. istor (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance kres - output signal ksig - input signal kcf - the center frequency of the filter, or frequency position of the peak response. kbw - bandwidth of the filter (the Hz difference between the upper and lower half-power points) resonxk is a lot faster than using individual instances in Csound orchestra of the old opcodes, because only one initialization and 'k' cycle are needed at a time, and the audio loop falls enterely inside the cache memory of processor. Examples Here is an example of the resonxk opcode. It uses the file resonxk.csd. Example of the resonxk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by &namegabriel;. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/outq4.xml0000644000000000000000000000615112262561504016202 0ustar rootroot outq4 Signal I/O:Signal Output outq4 outq4 Writes samples to quad channel 4 of an external device or stream. Description Writes samples to quad channel 4 of an external device or stream. Syntax outq4 asig Performance Sends audio samples to an accumulating output buffer (created at the beginning of performance) which serves to collect the output of all active instruments before the sound is written to disk. There can be any number of these output units in an instrument. The type (mono, stereo, quad, hex, or oct) should agree with nchnls. But as of version 3.50, Csound will attempt to change an incorrect opcode to agree with the nchnls statement. Opcodes can be chosen to direct sound to any particular channel: outs1 sends to stereo channel 1, outq3 to quad channel 3, etc. Examples Here is an example of the outq4 opcode. It uses the file outq4.csd. Example of the outq4 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also out, outh, outo, outq, outq1, outq2, outq3, outs, outs1, outs2, soundout Credits Author: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 manual_src~dfsg/opcodes/mdelay.xml0000644000000000000000000000473312262561504016405 0ustar rootroot mdelay Signal I/O:Signal Output mdelay mdelay A MIDI delay opcode. Description A MIDI delay opcode. Syntax mdelay kstatus, kchan, kd1, kd2, kdelay Performance kstatus -- status byte of MIDI message to be delayed kchan -- MIDI channel (1-16) kd1 -- first MIDI data byte kd2 -- second MIDI data byte kdelay -- delay time in seconds Each time that kstatus is other than zero, mdelay outputs a MIDI message to the MIDI out port after kdelay seconds. This opcode is useful in implementing MIDI delays. Several instances of mdelay can be present in the same instrument with different argument values, so complex and colorful MIDI echoes can be implemented. Further, the delay time can be changed at k-rate. Examples Here is an example of the mdelay opcode. It uses the file mdelay.csd. Example of the mdelay opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; Italy November 1998 New in Csound version 3.492 manual_src~dfsg/opcodes/grain3.xml0000644000000000000000000001714012262561504016311 0ustar rootroot grain3 Signal Generators:Granular Synthesis grain3 grain3 Generate granular synthesis textures with more user control. Description Generate granular synthesis textures. grain2 is simpler to use but grain3 offers more control. Syntax ares grain3 kcps, kphs, kfmd, kpmd, kgdur, kdens, imaxovr, kfn, iwfn, \ kfrpow, kprpow [, iseed] [, imode] Initialization imaxovr -- maximum number of overlapping grains. The number of overlaps can be calculated by (kdens * kgdur); however, it can be overestimated at no cost in rendering time, and a single overlap uses (depending on system) 16 to 32 bytes of memory. iwfn -- function table containing window waveform (Use GEN20 to calculate iwfn). iseed (optional, default=0) -- seed value for random number generator (positive integer in the range 1 to 2147483646 (231 - 2)). Zero or negative value seeds from current time (this is also the default). imode (optional, default=0) -- sum of the following values: 64: synchronize start phase of grains to kcps. 32: start all grains at integer sample location. This may be faster in some cases, however it also makes the timing of grain envelopes less accurate. 16: do not render grains with start time less than zero. (see the image below; this option turns off grains marked with red on the image). 8: interpolate window waveform (slower). 4: do not interpolate grain waveform (fast, but lower quality). 2: grain frequency is continuously modified by kcps and kfmd (by default, each grain keeps the frequency it was launched with). This may be slower at high control rates. It also controls phase modulation (kphs). 1: skip initialization. [A diagram showing grains with a start time less than zero in red.] A diagram showing grains with a start time less than zero in red. Performance ares -- output signal. kcps -- grain frequency in Hz. kphs -- grain phase. This is the location in the grain waveform table, expressed as a fraction (between 0 to 1) of the table length. kfmd -- random variation (bipolar) in grain frequency in Hz. kpmd -- random variation (bipolar) in start phase. kgdur -- grain duration in seconds. kgdur also controls the duration of already active grains (actually the speed at which the window function is read). This behavior does not depend on the imode flags. kdens -- number of grains per second. kfrpow -- this value controls the distribution of grain frequency variation. If kfrpow is positive, the random distribution (x is in the range -1 to 1) is abs(x) ˆ ((1 / kfrpow) - 1); for negative kfrpow values, it is (1 - abs(x)) ˆ ((-1 / kfrpow) - 1)Setting kfrpow to -1, 0, or 1 will result in uniform distribution (this is also faster to calculate). The image below shows some examples for kfrpow. The default value of kfrpow is 0. [A graph of distributions for different values of kfrpow.] A graph of distributions for different values of kfrpow. kprpow -- distribution of random phase variation (see kfrpow). Setting kphs and kpmd to 0.5, and kprpow to 0 will emulate grain2. kfn -- function table containing grain waveform. Table number can be changed at k-rate (this is useful to select from a set of band-limited tables generated by GEN30, to avoid aliasing). Note grain3 internally uses the same random number generator as rnd31. So reading its documentation is also recommended. Examples Here is an example of the grain3 opcode. It uses the file grain3.csd. Example of the grain3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also grain2 Credits Author: &nameistvan; New in version 4.15 Updated April 2002 by &nameistvan; manual_src~dfsg/opcodes/topXO.xml0000644000000000000000000005240712262561504016204 0ustar rootroot Orchestra Opcodes and Operators &opcodesnotequal; &opcodesdefine; &opcodesinclude; &opcodesundef; &opcodesifdef; &opcodesifndef; &opcodesdollar; &opcodesmodulus; &opcodesopand; &opcodesgreaterthan; &opcodesgreaterequal; &opcodeslessthan; &opcodeslessequal; &opcodesmultiplies; &opcodesadds; &opcodessubtracts; &opcodesdivides; &opcodesassign; &opcodesequals; &opcodesraises; &opcodesopor; &opcodes0dbfs; &opcodesopbitshl; &opcodesopbitshr; &opcodesopbitand; &opcodesopbitor; &opcodesopbitnot; &opcodesopnonequiv; &opcodesopa; &opcodesabs; &opcodesactive; &opcodesadsr; &opcodesadsyn; &opcodesadsynt; &opcodesadsynt2; &opcodesaftouch; &opcodesalpass; &opcodesampdb; &opcodesampdbfs; &opcodesampmidi; &opcodesareson; &opcodesaresonk; &opcodesatone; &opcodesatonek; &opcodesatonex; &opcodesbalance; &opcodesbamboo; &opcodesbarmodel; &opcodesbbcutm; &opcodesbbcuts; &opcodesbetarand; &opcodesbexprnd; &opcodesbinit; &opcodesbiquad; &opcodesbiquada; &opcodesbirnd; &opcodesbqrez; &opcodesbutbp; &opcodesbutbr; &opcodesbuthp; &opcodesbutlp; &opcodesbutterbp; &opcodesbutterbr; &opcodesbutterhp; &opcodesbutterlp; &opcodesbuzz; &opcodescabasa; &opcodescauchy; &opcodesceil; &opcodescent; &opcodescggoto; &opcodeschanctrl; &opcodeschanged; &opcodeschani; &opcodeschano; &opcodeschebyshevpoly; &opcodeschn; &opcodeschnclear; &opcodeschnexport; &opcodeschnget; &opcodeschnmix; &opcodeschnparams; &opcodeschnset; &opcodescigoto; &opcodesckgoto; &opcodesclear; &opcodesclfilt; &opcodesclip; &opcodesclockoff; &opcodesclockon; &opcodescngoto; &opcodescomb; &opcodescompress; &opcodesconvle; &opcodesconvolve; &opcodescos; &opcodescosh; &opcodescosinv; &opcodescps2pch; &opcodescpsmidi; &opcodescpsmidib; &opcodescpsmidinn; &opcodescpsoct; &opcodescpspch; &opcodescpstmid; &opcodescpstun; &opcodescpstuni; &opcodescpsxpch; &opcodescpuprc; &opcodescross2; &opcodescrunch; &opcodesctrl14; &opcodesctrl21; &opcodesctrl7; &opcodesctrlinit; &opcodescuserrnd; &opcodesdam; &opcodesdate; &opcodesdates; &opcodesdb; &opcodesdbamp; &opcodesdbfsamp; &opcodesdcblock; &opcodesdcblock2; &opcodesdconv; &opcodesdelay; &opcodesdelay1; &opcodesdelayk; &opcodesdelayr; &opcodesdelayw; &opcodesdeltap; &opcodesdeltap3; &opcodesdeltapi; &opcodesdeltapn; &opcodesdeltapx; &opcodesdeltapxw; &opcodesdenorm; &opcodesdiff; &opcodesdiskgrain; &opcodesdiskin; &opcodesdiskin2; &opcodesdispfft; &opcodesdisplay; &opcodesdistort; &opcodesdistort1; &opcodesdivz; &opcodesdownsamp; &opcodesdripwater; &opcodesdumpk; &opcodesdumpk2; &opcodesdumpk3; &opcodesdumpk4; &opcodesduserrnd; &opcodesdust; &opcodesdust2; &opcodeselse; &opcodeselseif; &opcodesendif; &opcodesendin; &opcodesendop; &opcodesenvlpx; &opcodesenvlpxr; &opcodesephasor; &opcodeseqfil; &opcodesevent; &opcodesevent_i; &opcodesexitnow; &opcodesexp; &opcodesexpcurve; &opcodesexpon; &opcodesexprand; &opcodesexpseg; &opcodesexpsega; &opcodesexpsegr; &opcodesficlose; &opcodesfilelen; &opcodesfilenchnls; &opcodesfilepeak; &opcodesfilesr; &opcodesfilter2; &opcodesfin; &opcodesfini; &opcodesfink; &opcodesfiopen; &opcodesflanger; &opcodesflashtxt; &opcodesflooper; &opcodesflooper2; &opcodesfloor; &opcodesfmb3; &opcodesfmbell; &opcodesfmmetal; &opcodesfmpercfl; &opcodesfmrhode; &opcodesfmvoice; &opcodesfmwurlie; &opcodesfof; &opcodesfof2; &opcodesfofilter; &opcodesfog; &opcodesfold; &opcodesfollow; &opcodesfollow2; &opcodesfoscil; &opcodesfoscili; &opcodesfout; &opcodesfouti; &opcodesfoutir; &opcodesfoutk; &opcodesfprintks; &opcodesfprints; &opcodesfrac; &opcodesfractalnoise; &opcodesfreeverb; &opcodesftchnls; &opcodesftconv; &opcodesftfree; &opcodesftgen; &opcodesftgentmp; &opcodesftlen; &opcodesftload; &opcodesftloadk; &opcodesftlptim; &opcodesftmorf; &opcodesftsave; &opcodesftsavek; &opcodesftsr; &opcodesgain; &opcodesgainslider; &opcodesgauss; &opcodesgausstrig; &opcodesgbuzz; &opcodesgendy &opcodesgendyc; &opcodesgendyx; &opcodesgetcfg; &opcodesgogobel; &opcodesgoto; &opcodesgrain; &opcodesgrain2; &opcodesgrain3; &opcodesgranule; &opcodesguiro; &opcodesharmon; &opcodesharmon2; &opcodeshilbert; &opcodeshrtfer; &opcodeshrtfmove; &opcodeshrtfmove2; &opcodeshrtfstat; &opcodeshsboscil; &opcodeshvs1; &opcodeshvs2; &opcodeshvs3; &opcodesopi; &opcodesif; &opcodesigoto; &opcodesihold; &opcodesimagecreate; &opcodesimagefree; &opcodesimagegetpixel; &opcodesimageload; &opcodesimagesave; &opcodesimagesetpixel; &opcodesimagesize; &opcodesin; &opcodesinch; &opcodesinit; &opcodesinitc14; &opcodesinitc21; &opcodesinitc7; &opcodesinrg; &opcodesins; &opcodesinsremot; &opcodesinsglobal; &opcodesinstr; &opcodesint; &opcodesinteg; &opcodesinterp; &opcodesinvalue; &opcodesinz; &opcodesjacktransport; &opcodesjitter; &opcodesjitter2; &opcodesjspline; &opcodesopk; &opcodeskgoto; &opcodeskr; &opcodesksmps; &opcodesktableseg; &opcodeslfo; &opcodeslimit; &opcodesline; &opcodeslinen; &opcodeslinenr; &opcodeslineto; &opcodeslinrand; &opcodeslinseg; &opcodeslinsegr; &opcodeslocsend; &opcodeslocsig; &opcodeslog; &opcodeslog10; &opcodeslogbtwo; &opcodeslogcurve; &opcodesloop_ge; &opcodesloop_gt; &opcodesloop_le; &opcodesloop_lt; &opcodesloopseg; &opcodesloopsegp; &opcodeslorenz; &opcodesloscil; &opcodesloscil3; &opcodesloscilx; &opcodeslowpass2; &opcodeslowres; &opcodeslowresx; &opcodeslpf18; &opcodeslpfreson; &opcodeslphasor; &opcodeslpinterp; &opcodeslposcil; &opcodeslposcil3; &opcodeslposcila; &opcodeslposcilsa; &opcodeslposcilsa2; &opcodeslpread; &opcodeslpreson; &opcodeslpshold; &opcodeslpsholdp; &opcodeslpslot; &opcodesmac; &opcodesmaca; &opcodesmadsr; &opcodesmandel; &opcodesmandol; &opcodesmarimba; &opcodesmassign; &opcodesmax; &opcodesmaxabs; &opcodesmaxabsaccum; &opcodesmaxaccum; &opcodesmaxalloc; &opcodesmax_k; &opcodesmclock; &opcodesmdelay; &opcodesmetro; &opcodesmidic14; &opcodesmidic21; &opcodesmidic7; &opcodesmidichannelaftertouch; &opcodesmidichn; &opcodesmidicontrolchange; &opcodesmidictrl; &opcodesmididefault; &opcodesmidiin; &opcodesmidinoteoff; &opcodesmidinoteoncps; &opcodesmidinoteonkey; &opcodesmidinoteonoct; &opcodesmidinoteonpch; &opcodesmidion; &opcodesmidion2; &opcodesmidiout; &opcodesmidipitchbend; &opcodesmidipolyaftertouch; &opcodesmidiprogramchange; &opcodesmiditempo; &opcodesmidremot; &opcodesmidglobal; &opcodesmin; &opcodesminabs; &opcodesminabsaccum; &opcodesminaccum; &opcodesmirror; &opcodesmixersetlevel; &opcodesmixergetlevel; &opcodesmixersend; &opcodesmixerreceive; &opcodesmixerclear; &opcodesmode; &opcodesmonitor; &opcodesmoog; &opcodesmoogladder; &opcodesmoogvcf; &opcodesmoogvcf2; &opcodesmoscil; &opcodesmpulse; &opcodesmrtmsg; &opcodesmultitap; &opcodesmute; &opcodesmxadsr; &opcodesnchnls; &opcodesnestedap; &opcodesnlfilt; &opcodesnoise; &opcodesnoteoff; &opcodesnoteon; &opcodesnoteondur; &opcodesnoteondur2; &opcodesnotnum; &opcodesnreverb; &opcodesnrpn; &opcodesnsamp; &opcodesnstrnum; &opcodesntrpol; &opcodesoctave; &opcodesoctcps; &opcodesoctmidi; &opcodesoctmidib; &opcodesoctmidinn; &opcodesoctpch; &opcodesopcode; &opcodesOSCsend; &opcodesOSCinit; &opcodesOSClisten; &opcodesoscbnk; &opcodesoscil; &opcodesoscil1; &opcodesoscil1i; &opcodesoscil3; &opcodesoscili; &opcodesoscilikt; &opcodesosciliktp; &opcodesoscilikts; &opcodesosciln; &opcodesoscils; &opcodesoscilx; &opcodesout; &opcodesoutc; &opcodesoutch; &opcodesoutiat; &opcodesoutic; &opcodesoutic14; &opcodesoutipat; &opcodesoutipb; &opcodesoutipc; &opcodesoutkat; &opcodesoutkc; &opcodesoutkc14; &opcodesoutkpat; &opcodesoutkpb; &opcodesoutkpc; &opcodesoutrg; &opcodesouts; &opcodesouts1; &opcodesouts2; &opcodesoutvalue; &opcodesoutz; &opcodesp; &opcodespan; &opcodespan2; &opcodespareq; &opcodespartials; &opcodespartikkel; &opcodespartikkelsync; &opcodespcauchy; &opcodespchbend; &opcodespchmidi; &opcodespchmidib; &opcodespchmidinn; &opcodespchoct; &opcodespconvolve; &opcodespcount; &opcodespdclip; &opcodespdhalf; &opcodespdhalfy; &opcodespeak; &opcodespgmassign; &opcodesphaser1; &opcodesphaser2; &opcodesphasor; &opcodesphasorbnk; &opcodespindex; &opcodespinkish; &opcodespitch; &opcodespitchamdf; &opcodesplanet; &opcodespluck; &opcodespoisson; &opcodespolyaft; &opcodespolynomial; &opcodespop; &opcodespop_f; &opcodesport; &opcodesportk; &opcodesposcil; &opcodesposcil3; &opcodespow; &opcodespowershape; &opcodespowoftwo; &opcodesprealloc; &opcodesprepiano; &opcodesprint; &opcodesprintf; &opcodesprintk; &opcodesprintk2; &opcodesprintks; &opcodesprints; &opcodesproduct; &opcodespset; &opcodesptrack; &opcodesputs; &opcodespush; &opcodespush_f; &opcodespvadd; &opcodespvbufread; &opcodespvcross; &opcodespvinterp; &opcodespvoc; &opcodespvread; &opcodespvsadsyn; &opcodespvsanal; &opcodespvsarp; &opcodespvsbandp; &opcodespvsbandr; &opcodespvsbin; &opcodespvsblur; &opcodespvsbuffer; &opcodespvsbufread; &opcodespvscale; &opcodespvscent; &opcodespvscross; &opcodespvsdemix; &opcodespvsdiskin; &opcodespvsdisp; &opcodespvsfilter; &opcodespvsfread; &opcodespvsfreeze; &opcodespvsftr; &opcodespvsftw; &opcodespvsfwrite; &opcodespvshift; &opcodespvsifd; &opcodespvsinfo; &opcodespvsinit; &opcodespvsin; &opcodespvsmaska; &opcodespvsmix; &opcodespvsmorph; &opcodespvsmooth; &opcodespvsout; &opcodespvsosc; &opcodespvspitch; &opcodespvstencil; &opcodespvsvoc; &opcodespvsynth; &opcodespyassign; &opcodespycall; &opcodespyeval; &opcodespyexec; &opcodespyinit; &opcodespyrun; &opcodesrand; &opcodesrandh; &opcodesrandi; &opcodesrandom; &opcodesrandomh; &opcodesrandomi; &opcodesrbjeq; &opcodesreadclock; &opcodesreadk; &opcodesreadk2; &opcodesreadk3; &opcodesreadk4; &opcodesreinit; &opcodesrelease; &opcodesremoteport; &opcodesremove; &opcodesrepluck; &opcodesreson; &opcodesresonk; &opcodesresonr; &opcodesresonx; &opcodesresonxk; &opcodesresony; &opcodesresonz; &opcodesresyn; &opcodesreverb; &opcodesreverb2; &opcodesreverbsc; &opcodesrewindscore; &opcodesrezzy; &opcodesrigoto; &opcodesrireturn; &opcodesrms; &opcodesrnd; &opcodesrnd31; &opcodesround; &opcodesrspline; &opcodesrtclock; &opcodess16b14; &opcodess32b14; &opcodesscale; &opcodessamphold; &opcodessandpaper; &opcodesscanhammer; &opcodesscantable; &opcodesscoreline; &opcodesscoreline_i; &opcodesschedkwhen; &opcodesschedkwhennamed; &opcodesschedule; &opcodesschedwhen; &opcodesseed; &opcodessekere; &opcodessemitone; &opcodessense; &opcodessensekey; &opcodesseqtime; &opcodesseqtime2; &opcodessetksmps; &opcodessetscorepos; &opcodessfilist; &opcodessfinstr; &opcodessfinstr3; &opcodessfinstr3m; &opcodessfinstrm; &opcodessfload; &opcodessflooper; &opcodessfpassign; &opcodessfplay; &opcodessfplay3; &opcodessfplay3m; &opcodessfplaym; &opcodessfplist; &opcodessfpreset; &opcodesshaker; &opcodessin; &opcodessinh; &opcodessininv; &opcodessinsyn; &opcodessleighbells; &opcodesslider16; &opcodesslider16f; &opcodesslider32; &opcodesslider32f; &opcodesslider64; &opcodesslider64f; &opcodesslider8; &opcodesslider8f; &opcodesslider16table; &opcodesslider16tablef; &opcodesslider32table; &opcodesslider32tablef; &opcodesslider64table; &opcodesslider64tablef; &opcodesslider8table; &opcodesslider8tablef; &opcodessliderkawai; &opcodessndload; &opcodessndloop; &opcodessndwarp; &opcodessndwarpst; &opcodessocksend; &opcodessockrecv; &opcodessoundin; &opcodessoundout; &opcodessoundouts; &opcodesspace; &opcodesspdist; &opcodessplitrig; &opcodesspsend; &opcodessprintf; &opcodessprintfk; &opcodessqrt; &opcodessr; &opcodesstack; &opcodesstatevar; &opcodesstix; &opcodesstrchar; &opcodesstrchark; &opcodesstrcpy; &opcodesstrcpyk; &opcodesstrcat; &opcodesstrcatk; &opcodesstrcmp; &opcodesstrcmpk; &opcodesstreson; &opcodesstrget; &opcodesstrindex; &opcodesstrindexk; &opcodesstrlen; &opcodesstrlenk; &opcodesstrlower; &opcodesstrlowerk; &opcodesstrrindex; &opcodesstrrindexk; &opcodesstrset; &opcodesstrsub; &opcodesstrsubk; &opcodesstrtod; &opcodesstrtodk; &opcodesstrtol; &opcodesstrtolk; &opcodesstrupper; &opcodesstrupperk; &opcodessubinstr; &opcodessubinstrinit; &opcodessum; &opcodessvfilter; &opcodessyncgrain; &opcodessyncloop; &opcodessyncphasor; &opcodessystem; &opcodestb; &opcodestab; &opcodestabrec; &opcodestable; &opcodestable3; &opcodestablecopy; &opcodestablegpw; &opcodestablei; &opcodestableicopy; &opcodestableigpw; &opcodestableikt; &opcodestableimix; &opcodestableiw; &opcodestablekt; &opcodestablemix; &opcodestableng; &opcodestablera; &opcodestableseg; &opcodestablew; &opcodestablewa; &opcodestablewkt; &opcodestablexkt; &opcodestablexseg; &opcodestabmorph; &opcodestabmorpha; &opcodestabmorphak; &opcodestabmorphi; &opcodestabplay; &opcodestambourine; &opcodestan; &opcodestanh; &opcodestaninv; &opcodestaninv2; &opcodestbvcf; &opcodestempest; &opcodestempo; &opcodestempoval; &opcodestigoto; &opcodestimedseq; &opcodestimeinstk; &opcodestimeinsts; &opcodestimek; &opcodestimes; &opcodestimout; &opcodestival; &opcodestlineto; &opcodestone; &opcodestonek; &opcodestonex; &opcodestrandom; &opcodestradsyn; &opcodestranseg; &opcodestrcross; &opcodestrfilter; &opcodestrhighest; &opcodestrigger; &opcodestrigseq; &opcodestrirand; &opcodestrlowest; &opcodestrmix; &opcodestrscale; &opcodestrshift; &opcodestrsplit; &opcodesturnoff; &opcodesturnoff2; &opcodesturnon; &opcodesunirand; &opcodesupsamp; &opcodesurd; &opcodesvadd; &opcodesvaddi; &opcodesvaddv; &opcodesvaddv_i; &opcodesvaget; &opcodesvalpass; &opcodesvaset; &opcodesvcella; &opcodesvco; &opcodesvco2; &opcodesvco2ft; &opcodesvco2ift; &opcodesvco2init; &opcodesvcomb; &opcodesvcopy; &opcodesvcopy_i; &opcodesvdelay; &opcodesvdelay3; &opcodesvdelayx; &opcodesvdelayxq; &opcodesvdelayxs; &opcodesvdelayxw; &opcodesvdelayxwq; &opcodesvdelayxws; &opcodesvdivv; &opcodesvdivv_i; &opcodesvdelayk; &opcodesvecdelay; &opcodesveloc; &opcodesvexp; &opcodesvexpi; &opcodesvexpseg; &opcodesvexpv; &opcodesvexpv_i; &opcodesvibes; &opcodesvibr; &opcodesvibrato; &opcodesvincr; &opcodesvlimit; &opcodesvlinseg; &opcodesvlowres; &opcodesvmap; &opcodesvmirror; &opcodesvmult; &opcodesvmulti; &opcodesvmultv; &opcodesvmultv_i; &opcodesvoice; &opcodesvosim; &opcodesvphaseseg; &opcodesvport; &opcodesvpow; &opcodesvpowi; &opcodesvpowv; &opcodesvpowv_i; &opcodesvpvoc; &opcodesvrandh; &opcodesvrandi; &opcodesvsubv; &opcodesvsubv_i; &opcodesvtable1k; &opcodesvtablei; &opcodesvtablek; &opcodesvtablea; &opcodesvtablewi; &opcodesvtablewk; &opcodesvtablewa; &opcodesvtabi; &opcodesvtabk; &opcodesvtaba; &opcodesvtabwi; &opcodesvtabwk; &opcodesvtabwa; &opcodesvwrap; &opcodeswaveset; &opcodesweibull; &opcodeswgbow; &opcodeswgbowedbar; &opcodeswgbrass; &opcodeswgclar; &opcodeswgflute; &opcodeswgpluck; &opcodeswgpluck2; &opcodeswguide1; &opcodeswguide2; &opcodeswrap; &opcodeswterrain; &opcodesxadsr; &opcodesxin; &opcodesxout; &opcodesxtratim; &opcodesxyin; &opcodeszacl; &opcodeszakinit; &opcodeszamod; &opcodeszar; &opcodeszarg; &opcodeszaw; &opcodeszawm; &opcodeszfilter2; &opcodeszir; &opcodesziw; &opcodesziwm; &opcodeszkcl; &opcodeszkmod; &opcodeszkr; &opcodeszkw; &opcodeszkwm; manual_src~dfsg/opcodes/mp3in.xml0000644000000000000000000000721012262561504016151 0ustar rootroot mp3in Signal I/O:Signal Input mp3in mp3in Reads stereo audio data from an external MP3 file. Description Reads stereo audio data from an external MP3 file. Syntax ar1, ar2 mp3in ifilcod[, iskptim, iformat, iskipinit, ibufsize] Initialization ifilcod -- integer or character-string denoting the source soundfile name. An integer denotes the file soundin.filcod ; a character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the named file is sought first in the current directory, then in that given by the environment variable SSDIR (if defined) then by SFDIR. iskptim (optional) -- time in seconds of input sound to be skipped. The default value is 0. iformat (optional) -- specifies the audio data file format: currently not implemented and always defaults to stereo. iskipinit (optional) -- switches off all initialisation if non zero (default =0). ibuffersize (optional) -- sets the internal buffer size for reading. If the value is omitted, zero or negative it defaults to 4096 bytes. Performance Reads stereo audio data from an external MP3 file. Examples Here is an example of the mp3in opcode. It uses the file mp3in.csd. Example of the mp3in opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also diskin, ins, in, inh, inh, ino, inq, mp3len, soundin Credits Author: &namejohn; Codemist Ltd 2009 New in version 5.11 manual_src~dfsg/opcodes/in.xml0000644000000000000000000000575012262561504015540 0ustar rootroot in Signal I/O:Signal Input in in Reads mono audio data from an external device or stream. Description Reads audio data from an external device or stream. Warning There are two versions of this opcode. The first is designed to be used only with orchestras that have inchnls=1. Doing so with orchestras with inchnls > 1 will cause incorrect audio input. The second form will read multiple channels into an array. Syntax ar1 in aarray in Performance Form 1 reads mono audio data from an external device or stream. If the command-line -i flag is set, sound is read continuously from the audio input stream (e.g. stdin or a soundfile) into an internal buffer. Any number of these opcodes can read freely from this buffer. The second format will read upto ichnls of audio into an audio array, which must be initialised. Examples Here is an example of the in opcode. It uses the file in.csd. Example of the in opcode. See Also diskin, inh, inh, ino, inq, ins, soundin Credits Authors: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 Authors: &namejohn; NUIM, 2013 Already in version 3.30 Array version in versio 6.01 manual_src~dfsg/opcodes/dssictls.xml0000644000000000000000000000501412262561504016753 0ustar rootroot dssictls Plugin Hosting:DSSI and LADSPA dssictls dssictls Send control information to a LADSPA or DSSI plugin. Syntax dssictls ihandle, iport, kvalue, ktrigger Description dssictls sends control values to a plugin's control port Initialization ihandle - handle for the plugin returned by dssiinit iport - control port number Performance kvalue - value to be assigned to the port ktrigger - determines whether the control information will be sent (ktrigger = 1) or not. This is useful for thinning control information, generating ktrigger with metro dssictls sends control information to a LADSPA or DSSI plugin's control port. The valid control ports and ranges are given by dssiinit . Using values outside the ranges may produce unspecified behaviour. Examples Here is an example of the dssictls opcode. It uses the file dssictls.csd. Example of the dssictls opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits 2005 By: &nameandres; Uses code from Richard Furse's LADSPA sdk. manual_src~dfsg/opcodes/joystick.xml0000644000000000000000000000507612262561504016772 0ustar rootroot joystick Instrument Control:Sensing and Control joystick joystick Reads data from a joystick controller. Description Reads data from a Linux joystick controller Syntax kres joystick kdevice ktab Performance Please note that this opcode is currently only supported on Linux. kdevice -- the index of the joystick device, either /dev/jsN or /dev/input/jsN. ktab -- A table to hold input results, should be at least enough elements to store one value for each stick axis and one for each button + 2. The first two elements of the table are initialized with the number of axes and the number of buttons, respectively, when a joystick is opened. If a joystick is unplugged during performance, the opcode will repeatedly attempt to reopen the device with a delay between attempts. Examples Here is an example of the joystick opcode. It uses the file joystick.csd. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the joystick opcode. It uses the file joystick-2.csd. Credits Author: Justin Glenn Smith 2010 New in version 5.17.12 manual_src~dfsg/opcodes/nestedap.xml0000644000000000000000000001102512262561504016725 0ustar rootroot nestedap Signal Modifiers:Reverberation nestedap nestedap Three different nested all-pass filters. Description Three different nested all-pass filters, useful for implementing reverbs. Syntax ares nestedap asig, imode, imaxdel, idel1, igain1 [, idel2] [, igain2] \ [, idel3] [, igain3] [, istor] Initialization imode -- operating mode of the filter: 1 = simple all-pass filter 2 = single nested all-pass filter 3 = double nested all-pass filter idel1, idel2, idel3 -- delay times of the filter stages. Delay times are in seconds and must be greater than zero. idel1 must be greater than the sum of idel2 and idel3. igain1, igain2, igain3 -- gain of the filter stages. imaxdel -- will be necessary if k-rate delays are implemented. Not currently used. istor -- Skip initialization if non-zero (default: 0). Performance asig -- input signal If imode = 1, the filter takes the form: Picture of nestedap imode 1 filter. Picture of imode 1 filter. If imode = 2, the filter takes the form: Picture of nestedap imode 2 filter. Picture of imode 2 filter. If imode = 3, the filter takes the form: Picture of nestedap imode 3 filter. Picture of imode 3 filter. Examples Here is an example of the nestedap opcode. It uses the file nestedap.csd, and beats.wav. Example of the nestedap opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namehans; February 1999 New in Csound version 3.53 The example was updated May 2002, thanks to &namehans; manual_src~dfsg/opcodes/tival.xml0000644000000000000000000000450312262561504016244 0ustar rootroot tival Instrument Control:Initialization and Reinitialization tival tival Puts the value of the instrument's internal tie-in flag into the named i-rate variable. Syntax ir tival Description Puts the value of the instrument's internal tie-in flag into the named i-rate variable. Initialization Puts the value of the instrument's internal tie-in flag into the named i-rate variable. Assigns 1 if this note has been tied onto a previously held note (see i statement); assigns 0 if no tie actually took place. (See also tigoto.) Examples Here is an example of the tival opcode. It uses the file tival.csd. Example of the tival opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also =, divz, init More on this opcode: http://www.csounds.com/journal/2005fall/tiedNotes.html , written by Steven Yi manual_src~dfsg/opcodes/xyin.xml0000644000000000000000000000630512262561504016116 0ustar rootroot xyin Instrument Control:Sensing and Control xyin xyin Sense the cursor position in an output window Description Sense the cursor position in an output window. When xyin is called the position of the mouse within the output window is used to reply to the request. This simple mechanism does mean that only one xyin can be used accurately at once. The position of the mouse is reported in the output window. Syntax kx, ky xyin iprd, ixmin, ixmax, iymin, iymax [, ixinit] [, iyinit] Initialization iprd -- period of cursor sensing (in seconds). Typically .1 seconds. xmin, xmax, ymin, ymax -- edge values for the x-y coordinates of a cursor in the input window. ixinit, iyinit (optional) -- initial x-y coordinates reported; the default values are 0,0. If these values are not within the given min-max range, they will be coerced into that range. Performance xyin samples the cursor x-y position in an input window every iprd seconds. Output values are repeated (not interpolated) at the k-rate, and remain fixed until a new change is registered in the window. There may be any number of input windows. This unit is useful for real-time control, but continuous motion should be avoided if iprd is unusually small. Depending on your platform and distribution, you might need to enable displays using the --displays command line flag. Examples Here is an example of the xyin opcode. It uses the file xyin.csd. Example of the xyin opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. As the values of kx and ky change, they will be printed out like this: kx=8.612036, ky=22.677933 kx=10.765685, ky=15.644135 Credits Example written by &namekevin;. manual_src~dfsg/opcodes/serialRead.xml0000644000000000000000000000570212262561504017202 0ustar rootroot serialRead Serial I/O serialRead serialRead Read data from a serial port. Description Read data from a serial port for arduino. Syntax kByte serialRead iPort Performance iPort -- port number optained from a serialBegin opcode. kByte -- a byte of data to read. Examples Here is an example of the serialRead opcode. It uses the file serialRead.csd. Example of the serialRead opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This is the matching Arduino code : void setup() { // enable serial communication Serial.begin(9600); // declare pin 9 to be an output: pinMode(9, OUTPUT); } void loop() { // only do something if we received something (this should be at csound's k-rate) if (Serial.available()) { // set the brightness of LED (connected to pin 9) to our input value int brightness = Serial.read(); analogWrite(9, brightness); // while we are here, get our knob value and send it to csound int sensorValue = analogRead(A0); Serial.write(sensorValue/4); // scale to 1-byte range (0-255) } } ..... See Also serialBegin, serialEnd, serialWrite_i, serialWrite, serialPrint, serialFlush. Credits Author: &namematt; 2011 New in version 5.14 manual_src~dfsg/opcodes/template.xml0000644000000000000000000001123312262561504016736 0ustar rootroot newopcodename Signal Modifiers:Delay newopcodename newopcodename Short description. Single line for opcode listing. Description Less concise description, but still short. Syntax outarg1, outarg2 newopcodename inarg1, inarg2 Initialization ivar -- i-time variable ivar2 -- i-time variable 2 Performance asig -- audio signal ksig -- control signal Then describe performace in detail. Use for internal manual links: delayr and this for external hyperlinks: newopcodename.csd or for the internet: Csounds.com page. You can also emphasize words. Title of the note (optional) An important note. You can also use "tip" or "warning" instead of note. You can also do lists. First item Second item You can also do numbered lists. First item Second item Put inline program lines like this: aout oscil 10000, 440, 1 And Csound output like this: Output text You can put images like this: [A diagram showing grains with a start time less than zero in red.] A diagram showing grains with a start time less than zero in red. Examples Here is an example of the delay opcode. It uses the file newopcodename.csd. Example of the newopcodename opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also deltap Credits By: &namejohn; 2010 Additional information manual_src~dfsg/opcodes/push_f.xml0000644000000000000000000000517512262561504016417 0ustar rootroot push_f Instrument Control:Stacks push_f push_f Pushes an f-sig frame into the global stack. Description Pushes an f-sig frame into the global stack. Syntax push_f fsig Performance fsig - f-signal to be pushed into the stack. The values are pushed into the global stack. The global stack works in LIFO order: after multiple push_f calls, pop_f should be used in reverse order. push/pop for a, k, i, and S types copy data by value. By contrast, push_f only pushes a "reference" to the f-signal, and then the corresponding pop_f will copy directly from the original variable to its output signal. For this reason, changing the source f-signal of push_f before pop_f is called is not recommended, and if the instrument instance owning the variable that was passed by push_f is deactivated before pop_f is called, undefined behavior may occur. pop_f and push_f can only take a single argument, and the data is passed both at init and performance time. Any stack errors (trying to push when there is no more space, or pop from an empty stack, inconsistent number or type of arguments, etc.) are fatal and terminate performance. See also stack, push, pop and pop_f. Credits By: &nameistvan;. 2006 manual_src~dfsg/opcodes/vcella.xml0000644000000000000000000001042412262561504016372 0ustar rootroot vcella Vectorial:Cellular Automata vcella vcella Cellular Automata Description Unidimensional Cellular Automata applied to Csound vectors Syntax vcella ktrig, kreinit, ioutFunc, initStateFunc, \ iRuleFunc, ielements, irulelen [, iradius] Initialization ioutFunc - number of the table where the state of each cell is stored initStateFunc - number of a table containig the inital states of each cell iRuleFunc - number of a lookup table containing the rules ielements - total number of cells irulelen - total number of rules iradius (optional) - radius of Cellular Automata. At present time CA radius can be 1 or 2 (1 is the default) Performance ktrig - trigger signal. Each time it is non-zero, a new generation of cells is evaluated kreinit - trigger signal. Each time it is non-zero, state of all cells is forced to be that of initStateFunc. vcella supports unidimensional cellular automata, where the state of each cell is stored in ioutFunc. So ioutFunc is a vector containing current state of each cell. This variant vector can be used together with any other vector-based opcode, such as adsynt, vmap, vpowv etc. initStateFunc is an input vector containing the inital value of the row of cells, while iRuleFunc is an input vector containing the rules in the form of a lookup table. Notice that initStateFunc and iRuleFunc can be updated during the performance by means of other vector-based opcodes (for example vcopy) in order to force to change rules and status at performance time. A new generation of cells is evaluated each time ktrig contains a non-zero value. Also the status of all cells can be forced to assume the status corresponding to the contents of initStateFunc each time kreinit contains a non-zero value. Radius of CA algorithm can be 1 or 2 (optional iradius arguement). Examples Here is an example of the vcella opcode. It uses the file vcella.csd. The following example uses vcella Example of the vcella opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by: Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) Example by: Anthony Kozar manual_src~dfsg/opcodes/endop.xml0000644000000000000000000000575312262561504016242 0ustar rootroot endop Orchestra Syntax:Block Statements endop endop Marks the end of an user-defined opcode block. Description Marks the end of an user-defined opcode block. Syntax endop Performance The syntax of a user-defined opcode block is as follows: opcode name, outtypes, intypes xinarg1 [, xinarg2] [, xinarg3] ... [xinargN] xin [setksmps iksmps] ... the rest of the instrument's code. xout xoutarg1 [, xoutarg2] [, xoutarg3] ... [xoutargN] endop The new opcode can then be used with the usual syntax: [xinarg1] [, xinarg2] ... [xinargN] name [xoutarg1] [, xoutarg2] ... [xoutargN] [, iksmps] Examples Here is an example of the endop opcode. It uses the file endop.csd. Example of the endop opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also opcode, setksmps, xin, xout More information on this opcode: http://www.csounds.com/journal/2006summer/controlFlow_part2.html , written by Steven Yi The user-defined opcode page: http://www.csounds.com/udo/ , maintained by Steven Yi Credits Author: Istvan Varga, 2002; based on code by Matt J. Ingalls New in version 4.22 manual_src~dfsg/opcodes/midipitchbend.xml0000644000000000000000000001132312262561504017726 0ustar rootroot midipitchbend Real-time MIDI:MIDI/Score Interoperability midipitchbend midipitchbend Gets a MIDI pitchbend value. Description midipitchbend is designed to simplify writing instruments that can be used interchangeably for either score or MIDI input, and to make it easier to adapt instruments originally written for score input to work with MIDI input. In general, it should be possible to write instrument definitions that work identically with both scores and MIDI, including both MIDI files and real-time MIDI input, without using any conditional statements, and that take full advantage of MIDI voice messages. Note that correlating Csound instruments with MIDI channel numbers is done using the massign opcode for real-time performance,. For file-driven performance, instrument numbers default to MIDI channel number + 1, but the defaults are overridden by any MIDI program change messages in the file. Syntax midipitchbend xpitchbend [, ilow] [, ihigh] Initialization ilow (optional) -- optional low value after rescaling, defaults to 0. ihigh (optional) -- optional high value after rescaling, defaults to 127. Performance xpitchbend -- returns the MIDI pitch bend during MIDI activation, remains unchanged otherwise. If the instrument was activated by MIDI input, the opcode overwrites the value of xpitchbend with the corresponding value from MIDI input. If the instrument was NOT activated by MIDI input, the value of xpitchbend remains unchanged. This enables score p-fields to receive MIDI input data during MIDI activation, and score values otherwise. Adapting a score-activated Csound instrument. See the MIDI interop opcodes section for details on adapting score driven instruments for MIDI or vice-versa. Examples Here is an example of the midipitchbend opcode. It uses the file midipitchbend.csd. Example of the midipitchbend opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i1 0.12695 i1 0.00000 i1 -0.01562 See Also midichannelaftertouch, midicontrolchange, mididefault, midinoteoff, midinoteoncps, midinoteonkey, midinoteonoct, midinoteonpch, midipolyaftertouch, midiprogramchange Credits Author: Michael Gogins New in version 4.20 manual_src~dfsg/opcodes/vdelayx.xml0000644000000000000000000000716012262561504016603 0ustar rootroot vdelayx Signal Modifiers:Delay vdelayx vdelayx A variable delay opcode with high quality interpolation. Description A variable delay opcode with high quality interpolation. Syntax aout vdelayx ain, adl, imd, iws [, ist] Initialization imd -- max. delay time (seconds) iws -- interpolation window size (see below) ist (optional) -- skip initialization if not zero Performance aout -- output audio signal ain -- input audio signal adl -- delay time in seconds This opcode uses high quality (and slow) interpolation, that is much more accurate than the currently available linear and cubic interpolation. The iws parameter sets the number of input samples used for calculating one output sample (allowed values are any integer multiply of 4 in the range 4 - 1024); higher values mean better quality and slower speed. Notes Delay time is measured in seconds (unlike in vdelay and vdelay3), and must be a-rate. The minimum allowed delay is iws/2 samples. Using the same variables as input and output is allowed in these opcodes. In vdelayxw*, changing the delay time has some effects on output volume: a = 1 / (1 + dt) where a is the output gain, and dt is the change of delay time per seconds. These opcodes are best used in the double-precision version of Csound. Examples Here is an example of the use of the vdelayx opcode. It uses the file vdelayx.csd. Example of the <emphasis>vdelayx</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vdelayxq, vdelayxs, vdelayxw, vdelayxwq, vdelayxws manual_src~dfsg/opcodes/pchmidi.xml0000644000000000000000000000732012262561504016542 0ustar rootroot pchmidi Real-time MIDI:Converters pchmidi pchmidi Get the note number of the current MIDI event, expressed in pitch-class units. Description Get the note number of the current MIDI event, expressed in pitch-class units. Syntax ipch pchmidi Performance Get the note number of the current MIDI event, expressed in pitch-class units for local processing. pchmidi vs. pchmidinn The pchmidi opcode only produces meaningful results in a Midi-activated note (either real-time or from a Midi score with the -F flag). With pchmidi, the Midi note number value is taken from the Midi event that is internally associated with the instrument instance. On the other hand, the pchmidinn opcode may be used in any Csound instrument instance whether it is activated from a Midi event, score event, line event, or from another instrument. The input value for pchmidinn might for example come from a p-field in a textual score or it may have been retrieved from the real-time Midi event that activated the current note using the notnum opcode. Examples Here is an example of the pchmidi opcode. It uses the file pchmidi.csd. Example of the pchmidi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also aftouch, ampmidi, cpsmidi, cpsmidib, midictrl, notnum, octmidi, octmidib, pchbend, pchmidib, veloc, cpsmidinn, octmidinn, pchmidinn Credits Author: &namebarry; - &namemike; MIT - Mills May 1997 Example written by &namekevin;. manual_src~dfsg/opcodes/STKBeeThree.xml0000644000000000000000000000726112262561504017176 0ustar rootroot STKBeeThree Signal Generators:STK Opcodes STKBeeThree STKBeeThree STK Hammond-oid organ-like FM synthesis instrument. Description STK Hammond-oid organ-like FM synthesis instrument. This opcode a simple 4 operator topology, also referred to as algorithm 8 of the TX81Z. It simulates the sound of a Hammond-oid organ, and some related sounds. Syntax asignal STKBeeThree ifrequency, iamplitude, [kop4, kv1[, kop3, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kop4 -- controller 2, gain of feedback of operator 4. Value range of kv1 is 0-127. kop3 -- controller 4, gain of operator 3. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kadsr -- controller 128, ADSR 2 and 4 target. Value range of kv5 is 0-127. Notes The code for this opcode is taken directly from the BeeThree class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKBeeThree opcode. It uses the file STKBeeThree.csd. Example of the STKBeeThree opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/outipb.xml0000644000000000000000000000735012262561504016432 0ustar rootroot outipb Real-time MIDI:Output outipb outipb Sends MIDI pitch-bend messages at i-rate. Description Sends MIDI pitch-bend messages at i-rate. Syntax outipb ichn, ivalue, imin, imax Initialization ichn -- MIDI channel number (1-16) ivalue -- floating point value imin -- minimum floating point value (converted in MIDI integer value 0) imax -- maximum floating point value (converted in MIDI integer value 127 (7 bit)) Performance outipb (i-rate pitch bend output) sends pitch bend messages. It works only with MIDI instruments which recognize them. It can drive a different value of a parameter for each note currently active. It can scale an i-value floating-point argument according to the imin and imax values. For example, set imin = 1.0 and imax = 2.0. When the ivalue argument receives a 2.0 value, the opcode will send a 127 value to the MIDI OUT device. When the ivalue argument receives a 1.0 value, it will send a 0 value. i-rate opcodes send their message once during instrument initialization. Examples Here is an example of the outipb opcode. It uses the file outipb.csd. Example of the outipb opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also outiat, outic14, outic, outipat, outipc, outkat, outkc14, outkc, outkpat, outkpb, outkpc Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/out32.xml0000644000000000000000000000330312262561504016076 0ustar rootroot out32 Signal I/O:Signal Output out32 out32 Writes 32-channel audio data to an external device or stream. Description Writes 32-channel audio data to an external device or stream. Syntax out32 asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8, asig10, \ asig11, asig12, asig13, asig14, asig15, asig16, asig17, asig18, \ asig19, asig20, asig21, asig22, asig23, asig24, asig25, asig26, \ asig27, asig28, asig29, asig30, asig31, asig32 Performance out32 outputs 32 channels of audio. See Also outc, outch, outx, outz Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/flvkeybd.xml0000644000000000000000000000574712262561504016746 0ustar rootroot FLvkeybd FLTK:Other FLvkeybd FLvkeybd An FLTK widget opcode that creates a virtual keyboard widget. Description An FLTK widget opcode that creates a virtual keyboard widget. This must be used in conjunction with the virtual midi keyboard driver for this to operate correctly. The purpose of this opcode is for making demo versions of MIDI orchestras with the virtual keyboard embedded within the main window. Note The widget version of the virtual keyboard does not include the MIDI sliders found in the full window version of the virtual keyboard. Syntax FLvkeybd "keyboard.map", iwidth, iheight, ix, iy Initialization keyboard.map -- a double-quoted string containing the keyboard map to use. An empty string ("") may be used to use the default bank/channel name values. See Virtual Midi Keyboard for more information on keyboard mappings. iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the keyboard, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the keyboard, relative to the upper left corner of corresponding window (expressed in pixels). Note The standard width and height for the virtual keyboard is 624x120 for the dialog version that is shown when FLvkeybd is not used. See Also FLbutton, FLbox, FLbutBank, FLprintk, FLprintk2, FLvalue Credits Author: &namesteven; New in version 5.05 manual_src~dfsg/opcodes/vaddv_i.xml0000644000000000000000000001037212262561504016542 0ustar rootroot vaddv_i Vectorial:Vectorial operations vaddv_i vaddv_i Performs addition between two vectorial control signals at init time. Description Performs addition between two vectorial control signals at init time. Syntax vaddv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Initialization ifn1 - number of the table hosting the first vector to be processed ifn2 - number of the table hosting the second vector to be processed ielements - number of elements of the two vectors idstoffset - index offset for the destination (ifn1) table (Default=0) isrcoffset - index offset for the source (ifn2) table (Default=0) Performance vaddv_i adds two vectorial control signals, that is, each element of the first vector is processed (only) with the corresponding element of the other vector. Each vectorial signal is hosted by a table (ifn1 and ifn2). The number of elements contained in both vectors must be the same. The result is a new vectorial control signal that overrides old values of ifn1. If you want to keep the old ifn1 vector, use vcopy_i opcode to copy it in another table. You can use idstoffset and isrcoffset to specify vectors in any location of the tables. Negative values for idstoffset and isrcoffset are acceptable. If idstoffset is negative, the out of range section of the vector will be discarded. If isrcoffset is negative, the out of range elements will be assumed to be 0 (i.e. the destination elements will not be changed). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 0 (i.e. the destination vector will not be changed for these elements). Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at init time. There's an k-rate version of this opcode called vaddv. All these operators (vaddv_i,vsubv_i,vmultv_i,vdivv_i,vpowv_i,vexpv_i, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/initc21.xml0000644000000000000000000000572312262561504016403 0ustar rootroot initc21 Real-time MIDI:Input initc21 initc21 Initializes the controllers used to create a 21-bit MIDI value. Description Initializes the controllers used to create a 21-bit MIDI value. Syntax initc21 ichan, ictlno1, ictlno2, ictlno3, ivalue Initialization ichan -- MIDI channel (1-16) ictlno1 -- most significant byte controller number (0-127) ictlno2 -- medium significant byte controller number (0-127) ictlno3 -- least significant byte controller number (0-127) ivalue -- floating point value (must be within 0 to 1) Performance initc21 can be used together with both midic21 and ctrl21 opcodes for initializing the first controller's value. ivalue argument must be set with a number within 0 to 1. An error occurs if it is not. Use the following formula to set ivalue according with midic21 and ctrl21 min and max range: ivalue = (initial_value - min) / (max - min) See Also ctrl7, ctrl14, ctrl21, ctrlinit, initc7, initc14, midic7, midic14, midic21 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/statevar.xml0000644000000000000000000000674012262561504016763 0ustar rootroot statevar Signal Modifiers:Standard Filters:Resonant statevar statevar State-variable filter. Description Statevar is a new digital implementation of the analogue state-variable filter. This filter has four simultaneous outputs: high-pass, low-pass, band-pass and band-reject. This filter uses oversampling for sharper resonance (default: 3 times oversampling). It includes a resonance limiter that prevents the filter from getting unstable. Syntax ahp,alp,abp,abr statevar ain, xcf, xq [, iosamps, istor] Initialization iosamps -- number of times of oversampling used in the filtering process. This will determine the maximum sharpness of the filter resonance (Q). More oversampling allows higher Qs, less oversampling will limit the resonance. The default is 3 times (iosamps=0). istor --initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance ahp -- high-pass output signal. alp -- low-pass output signal. abp -- band-pass signal. abr -- band-reject signal. asig -- input signal. xcf -- filter cutoff frequency (k-rate or a-rate). xq -- filter Q (k-rate or a-rate). This value is limited internally depending on the frequency and the number of times of oversampling used in the process (3-times oversampling by default). Examples Here is an example of the statevar opcode. It uses the file statevar.csd. Example of the statevar opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; January 2005 New plugin in version 5 January 2005. Audio rate parameters introduced in version 6.02 October 2013. manual_src~dfsg/opcodes/vlimit.xml0000644000000000000000000000463112262561504016433 0ustar rootroot vlimit Vectorial:Limiting and Wrapping vlimit vlimit Limiting and Wrapping Vectorial Signals Description Limits elements of vectorial control signals. Syntax vlimit ifn, kmin, kmax, ielements Initialization ifn - number of the table hosting the vector to be processed ielements - number of elements of the vector Performance kmin - minimum threshold value kmax - maximum threshold value vlimit set lower and upper limits on each element of the vector they process. These opcodes are similar to limit, wrap and mirror, but operate with a vectorial signal instead of with a scalar signal. Result overrides old values of ifn1, if these are out of min/max interval. If you want to keep input vector, use vcopy opcode to copy it in another table. All these opcodes are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Note: bmscan not yet available on Canonical Csound Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/seqtime2.xml0000644000000000000000000001432412262561504016660 0ustar rootroot seqtime2 Instrument Control:Sensing and Control seqtime2 seqtime2 Generates a trigger signal according to the values stored in a table. Description Generates a trigger signal according to the values stored in a table. Syntax ktrig_out seqtime2 ktrig_in, ktime_unit, kstart, kloop, kinitndx, kfn_times Performance ktrig_out -- output trigger signal ktime_unit -- unit of measure of time, related to seconds. ktrig_in -- input trigger signal. kstart -- start index of looped section kloop -- end index of looped section kinitndx -- initial index Note Although kinitndx is listed as k-rate, it is in fact accessed only at init-time. So if you are using a k-rate argument, it must be assigned with init. kfn_times -- number of table containing a sequence of times This opcode handles timed-sequences of groups of values stored into a table. seqtime2 generates a trigger signal (a sequence of impulses, see also trigger opcode), according to the values stored in the kfn_times table. This table should contain a series of delta-times (i.e. times beetween to adjacent events). The time units stored into table are expressed in seconds, but can be rescaled by means of ktime_unit argument. The table can be filled with GEN02 or by means of an external text-file containing numbers, with GEN23. It is possible to start the sequence from a value different than the first, by assigning to initndx an index different than zero (which corresponds to the first value of the table). Normally the sequence is looped, and the start and end of loop can be adjusted by modifying kstart and kloop arguments. User must be sure that values of these arguments (as well as initndx) correspond to valid table numbers, otherwise Csound will crash (because no range-checking is implementeted). It is possible to disable loop (one-shot mode) by assigning the same value both to kstart and kloop arguments. In this case, the last read element will be the one corresponding to the value of such arguments. Table can be read backward by assigning a negative kloop value. It is possible to trigger two events almost at the same time (actually separated by a k-cycle) by giving a zero value to the corresponding delta-time. First element contained in the table should be zero, if the user intends to send a trigger impulse, it should come immediately after the orchestra instrument containing seqtime2 opcode. seqtime2 is similar to seqtime, the difference is that when ktrig_in contains a non-zero value, current index is reset to kinitndx value. kinitndx can be varied at performance time. Examples Here is an example of the seqtime2 opcode. It uses the file seqtime2.csd. Example of the seqtime2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also GEN02, GEN23, seqtime, trigseq, timedseq Credits Author: &namegabriel; manual_src~dfsg/opcodes/port.xml0000644000000000000000000000637412262561504016121 0ustar rootroot port Signal Modifiers:Standard Filters:Control port port Applies portamento to a step-valued control signal. Description Applies portamento to a step-valued control signal. Syntax kres port ksig, ihtim [, isig] Initialization ihtim -- half-time of the function, in seconds. isig (optional, default=0) -- initial (i.e. previous) value for internal feedback. The default value is 0. Negative value will cause initialization to be skipped and last value from previous instance to be used as initial value for note. Performance kres -- the output signal at control-rate. ksig -- the input signal at control-rate. port applies portamento to a step-valued control signal. At each new step value, ksig is low-pass filtered to move towards that value at a rate determined by ihtim. ihtim is the half-time of the function (in seconds), during which the curve will traverse half the distance towards the new value, then half as much again, etc., theoretically never reaching its asymptote. With portk, the half-time can be varied at the control rate. Examples Here is an example of the port opcode. It uses the file port.csd. Example of the port opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also areson, aresonk, atone, atonek, portk, reson, resonk, tone, tonek manual_src~dfsg/opcodes/opbitshl.xml0000644000000000000000000000623112262561504016751 0ustar rootroot ¬ Mathematical Operations:Arithmetic and Logic Operations << << Bitshift left operator. Description The bitshift operators shift the bits to the left or to the right the number of bits given. The priority of these operators is less binding that the arithmetic ones, but more binding that the comparisons. Parentheses may be used as above to force particular groupings. Syntax a << b (bitshift left) where the arguments a and b may be further expressions. Examples Here is an example of the bitshift left operator. It uses the file bitshift.csd. Example of the bitshift left operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above will produce the following output: 2>>1 = 1 B 0.000 .. 0.100 T 0.100 TT 0.100 M: 0.0 0.0 3>>1 = 1 B 0.100 .. 0.200 T 0.200 TT 0.200 M: 0.0 0.0 7>>2 = 1 B 0.200 .. 0.300 T 0.300 TT 0.300 M: 0.0 0.0 16>>1 = 8 B 0.300 .. 0.400 T 0.400 TT 0.400 M: 0.0 0.0 16>>2 = 4 B 0.400 .. 0.500 T 0.500 TT 0.500 M: 0.0 0.0 16>>3 = 2 B 0.500 .. 5.000 T 5.000 TT 5.000 M: 0.0 0.0 new alloc for instr 2: 1<<1 = 2 B 5.000 .. 5.100 T 5.100 TT 5.100 M: 0.0 0.0 1<<2 = 4 B 5.100 .. 5.200 T 5.200 TT 5.200 M: 0.0 0.0 1<<3 = 8 B 5.200 .. 5.300 T 5.300 TT 5.300 M: 0.0 0.0 1<<4 = 16 B 5.300 .. 5.400 T 5.400 TT 5.400 M: 0.0 0.0 2<<1 = 4 B 5.400 .. 5.500 T 5.500 TT 5.500 M: 0.0 0.0 2<<2 = 8 B 5.500 .. 5.600 T 5.600 TT 5.600 M: 0.0 0.0 2<<3 = 16 B 5.600 .. 5.700 T 5.700 TT 5.700 M: 0.0 0.0 3<<2 = 12 See Also >>, &, | # manual_src~dfsg/opcodes/outletf.xml0000644000000000000000000000456612262561504016620 0ustar rootroot outletf Signal Flow Graph Opcodes outletf outletf Sends a frate signal (fsig) out from an instrument to a named port. Description Sends a frate signal (fsig) out from an instrument to a named port. Syntax outletf Sname, fsignal Initialization Sname -- String name of the outlet port. The name of the outlet is implicitly qualified by the instrument name or number, so it is valid to use the same outlet name in more than one instrument (but not to use the same outlet name twice in one instrument). Performance fsignal -- frate output signal (fsig) During performance, the output signal is sent to each instance of an instrument containing an inlet port to which this outlet has been connected using the connect opcode. The signals of all the outlets connected to an inlet are combined in the inlet. See Also outleta outletk outletkid inleta inletk inletkid inletf connect alwayson ftgenonce Credits By: &namemichael; 2009 manual_src~dfsg/opcodes/fmbell.xml0000644000000000000000000001007412262561504016366 0ustar rootroot fmbell Signal Generators:FM Synthesis fmbell fmbell Uses FM synthesis to create a tublar bell sound. Description Uses FM synthesis to create a tublar bell sound. It comes from a family of FM sounds, all using 4 basic oscillators and various architectures, as used in the TX81Z synthesizer. Syntax ares fmbell kamp, kfreq, kc1, kc2, kvdepth, kvrate[, ifn1, ifn2, ifn3, \ ifn4, ivfn, isus] Initialization All these opcodes take 5 tables for initialization. The first 4 are the basic inputs and the last is the low frequency oscillator (LFO) used for vibrato. These all default to a siewave table. The initial waves should be: ifn1 -- sine wave ifn2 -- sine wave ifn3 -- sine wave ifn4 -- sine wave The optional argument isus controls how long the sound lasts, or how quickly it decays. It defaults to 4. Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. kc1, kc2 -- Controls for the synthesizer: kc1 -- Mod index 1 kc2 -- Crossfade of two outputs Algorithm -- 5 kvdepth -- Vibrator depth kvrate -- Vibrator rate Examples Here is an example of the fmbell opcode. It uses the file fmbell.csd. Example of the fmbell opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also fmb3, fmmetal, fmpercfl, fmrhode, fmwurlie Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 Optional argument new in 5.16 More optional argument new in 6.0 manual_src~dfsg/opcodes/slider32tablef.xml0000644000000000000000000001522212262561504017732 0ustar rootroot slider32tablef Real-time MIDI:Slider Banks slider32tablef slider32tablef Stores a bank of 32 different MIDI control messages to a table, filtered before output. Description Stores a bank of 32 different MIDI control messages to a table, filtered before output. Syntax kflag slider32tablef ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, icutoff1, .... , ictlnum32, imin32, imax32, init32, ifn32, icutoff32 Initialization ichan -- MIDI channel (1-16) ioutTable -- number of the table that will contain the output ioffset -- output table offset. A zero means that the output of the first slider will affect the first table element. A 10 means that the output of the first slider will affect the 11th table element. ictlnum1 ... ictlnum32 -- MIDI control number (0-127) imin1 ... imin32 -- minimum values for each controller imax1 ... imax32 -- maximum values for each controller init1 ... init32 -- initial value for each controller ifn1 ... ifn32 -- function table for conversion for each controller icutoff1 ... icutoff32 -- low-pass filter cutoff frequency for each controller Performance kflag -- a flag that informs if any control-change message in the bank has been received. In this case kflag is set to 1. Otherwise is set to zero. slider32tablef is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider32tablef allows a bank of 32 different MIDI control message numbers. It filters the signal before output. This eliminates discontinuities due to the low resolution of the MIDI (7 bit). The cutoff frequency can be set separately for each controller (suggested range: .1 to 5 Hz). As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. slider32tablef is very similar to slider32f and sliderNf family of opcodes (see their manual for more information). The actual difference is that the output is not stored to k-rate variables, but to a table, denoted by the ioutTable argument. It is possible to define a starting index in order to use the same table for more than one slider bank (or other purposes). It is possible to use this opcode together with FLslidBnk2Setk and FLslidBnk2, so you can synchronize the position of the MIDI values to the position of the FLTK valuator widgets of FLslidBnk2. Notice that you have to specify the same min/max values as well the linear/exponential responses in both sliderNtable(f) and FLslidBnk2. The exception is when using table-indexed response instead of a lin/exp response. In this case, in order to achieve a useful result, the table-indexed response and actual min/max values must be set only in FLslidBnk2, whereas, in sliderNtable(f), you have to set a linear response and a minimum of zero and a maximum of one in all sliders. slider32tablef opcodes do not output the required initial value immediately, but only after some k-cycles because the filter slightly delays the output. See Also slider16, slider16f, slider32, slider64, slider64f, slider8, slider8f Credits Author: &namegabriel; New in Csound version 5.06 manual_src~dfsg/opcodes/clear.xml0000644000000000000000000000524312262561504016215 0ustar rootroot clear Mathematical Operations:Comparators and Accumulators clear clear Zeroes a list of audio signals. Description clear zeroes a list of audio signals. Syntax clear avar1 [, avar2] [, avar3] [...] Performance avar1, avar2, avar3, ... -- signals to be zeroed clear sets every sample of each of the given audio signals to zero when it is performed. This is equivalent to writing avarN = 0 in the orchestra for each of the specified variables. Typically, clear is used with global variables that combine multiple signals from different sources and change with each k-pass (performance loop) through all of the active instrument instances. After the final usage of such a variable and before the next k-pass, it is necessary to clear the variable so that it does not add the next cycle's signals to the previous result. clear is especially useful in combination with vincr (variable increment) and they are intended to be used together with file output opcodes such as fout. Examples Here is an example of the clear opcode. It uses the file clear.csd. Example of the clear opcode. See the fout opcode for another example. See Also vincr Credits Author: &namegabriel; Italy 1999 New in Csound version 3.56 manual_src~dfsg/opcodes/sin.xml0000644000000000000000000000415212262561504015716 0ustar rootroot sin Mathematical Operations:Trigonometric Functions sin sin Performs a sine function. Description Returns the sine of x (x in radians). Syntax sin(x) (no rate restriction) Examples Here is an example of the sin opcode. It uses the file sin.csd. Example of the sin opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: isin1 = 0.000 isin2 = 1.000 isin3 = 0.000 isin4 = -1.000 isin5 = -0.000 isin6 = 0.000 See Also cos, cosh, cosinv, sinh, sininv, tan, tanh, taninv manual_src~dfsg/opcodes/zkwm.xml0000644000000000000000000000706012262561504016116 0ustar rootroot zkwm Zak Patch System zkwm zkwm Writes to a zk variable at k-rate with mixing. Description Writes to a zk variable at k-rate with mixing. Syntax zkwm ksig, kndx [, imix] Initialization imix (optional, default=1) -- indicates if mixing should occur. Performance ksig -- value to be written to the zk location. kndx -- points to the zk or za location to which to write. zkwm is a mixing opcode, it adds the signal to the current value of the variable. If no imix is specified, mixing always occurs. imix = 0 will cause overwriting like ziw, zkw, and zaw. Any other value will cause mixing. Caution: When using the mixing opcodes ziwm, zkwm, and zawm, care must be taken that the variables mixed to, are zeroed at the end (or start) of each k- or a-cycle. Continuing to add signals to them, can cause their values can drift to astronomical figures. One approach would be to establish certain ranges of zk or za variables to be used for mixing, then use zkcl or zacl to clear those ranges. Examples Here is an example of the zkwm opcode. It uses the file zkwm.csd. Example of the zkwm opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zaw, zawm, ziw, ziwm, zkcl, zkw, zkr Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/cps2pch.xml0000644000000000000000000001060612262561504016470 0ustar rootroot cps2pch Pitch Converters:Tuning Opcodes cps2pch cps2pch Converts a pitch-class value into cycles-per-second (Hz) for equal divisions of the octave. Description Converts a pitch-class value into cycles-per-second (Hz) for equal divisions of the octave. Syntax icps cps2pch ipch, iequal Initialization ipch -- Input number of the form 8ve.pc, indicating an 'octave' and which note in the octave. iequal -- If positive, the number of equal intervals into which the 'octave' is divided. Must be less than or equal to 100. If negative, is the number of a table of frequency multipliers. Note The following are essentially the same ia = cpspch(8.02) ib cps2pch 8.02, 12 ic cpsxpch 8.02, 12, 2, 1.02197503906 These are opcodes not functions Negative values of ipch are allowed. Examples Here is an example of the cps2pch opcode. It uses the file cps2pch.csd. Example of the cps2pch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: icps = 293.666 Here is an example of the cps2pch opcode using a table of frequency multipliers. It uses the file cps2pch_ftable.csd. Example of the cps2pch opcode using a table of frequency multipliers. Its output should include lines like this: instr 1: icps = 313.951 Here is an example of the cps2pch opcode using a 19ET scale. It uses the file cps2pch_19et.csd. Example of the cps2pch opcode using a 19ET scale. Its output should include lines like this: instr 1: icps = 281.429 See Also cpspch, cpsxpch Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK 1997 New in Csound version 3.492 manual_src~dfsg/opcodes/vtablea.xml0000644000000000000000000000741712262561504016552 0ustar rootroot vtablea Vectorial:Tables vtablea vtablea Read vectors (from tables -or arrays of vectors). Description This opcode reads vectors from tables at a-rate. Syntax vtablea andx, kfn, kinterp, ixmode, aout1 [, aout2, aout3, .... , aoutN ] Initialization ixmode - index data mode. The default value is 0. == 0 index is treated as a raw table location, == 1 index is normalized (0 to 1). Performance andx - Index into f-table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0). kfn - table number kinterp - switch beetween interpolated or non-interpolated output. 0 -> non-interpolation , non-zero -> interpolation activated aout1...aoutN - output vector elements This opcode is useful in all cases in which one needs to access sets of values associated to unique indexes (for example, multi-channel samples, STFT bin frames, spectral formants, p-field based scores etc.). The number of elements of each vector (length of the vector) is determined by the number of optional arguments on the right (aout1 , aout2, aout3, .... aoutN). vtablea allows the user to switch beetween interpolated or non-interpolated output at k-rate by means of kinterp argument. vtablea allows also to switch the table number at k-rate (but this is possible only when vector frames of each used table have the same number of elements, otherwise unpredictable results could occurr), as well as to choose indexing style (raw or normalized, see also ixmode argument of table opcode ). Notice that no wrap nor limit mode is implemented. So, if an index attempt to access to a zone not allocated by the table, Csound will probably crash. However this drawback can be easily avoided by using wrap or limit opcodes applied to indexes before using vtablea, in order to correct eventual out-of-range values. Notice that vtablea's output arguments are placed at the right of the opcode name, differently from usual (this style is already used in other opcodes using undefined lists of output arguments such as fin or trigseq). See also vtablek, vtablei, vtaba, vtablewa, vtabwa, Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/reverb2.xml0000644000000000000000000000137312262561504016476 0ustar rootroot reverb2 Signal Modifiers:Reverberation reverb2 reverb2 Same as the nreverb opcode. Description Same as the nreverb opcode. Syntax ares reverb2 asig, ktime, khdif [, iskip] [,inumCombs] \ [, ifnCombs] [, inumAlpas] [, ifnAlpas] manual_src~dfsg/opcodes/pvsadsyn.xml0000644000000000000000000000762212262561504017001 0ustar rootroot pvsadsyn Spectral Processing:Streaming pvsadsyn pvsadsyn Resynthesize using a fast oscillator-bank. Description Resynthesize using a fast oscillator-bank. Syntax ares pvsadsyn fsrc, inoscs, kfmod [, ibinoffset] [, ibinincr] [, iinit] Initialization inoscs -- The number of analysis bins to synthesise. Cannot be larger than the size of fsrc (see pvsinfo), e.g. as created by pvsanal. Processing time is directly proportional to inoscs. ibinoffset (optional, default=0) -- The first (lowest) bin to resynthesise, counting from 0 (default = 0). ibinincr (optional) -- Starting from bin ibinoffset, resynthesize bins ibinincr apart. iinit (optional) -- Skip reinitialization. This is not currently implemented for any of these opcodes, and it remains to be seen if it is even practical. Performance kfmod -- Scale all frequencies by factor kfmod. 1.0 = no change, 2 = up one octave. pvsadsyn is experimental, and implements the oscillator bank using a fast direct calculation method, rather than a lookup table. This takes advantage of the fact, empirically arrived at, that for the analysis rates generally used, (and presuming analysis using pvsanal, where frequencies in a bin change only slightly between frames) it is not necessary to interpolate frequencies between frames, only amplitudes. Accurate resynthesis is often contingent on the use of pvsanal with iwinsize = ifftsize*2. This opcode is the most likely to change, or be much extended, according to feedback and advice from users. It is likely that a full interpolating table-based method will be added, via a further optional iarg. The parameter list to pvsadsyn mimics that for pvadd, but excludes spectral extraction. Examples Here is an example of the pvsadsyn opcode. It uses the file pvsadsyn.csd. Example of the pvsadsyn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namerichard; August 2001 New in version 4.13 manual_src~dfsg/opcodes/opnonequiv.xml0000644000000000000000000000274312262561504017334 0ustar rootroot # Mathematical Operations:Arithmetic and Logic Operations # # Bitwise NON EQUIVALENCE operator. Description The bitwise operators perform operations of bitwise AND, bitwise OR, bitwise NOT and bitwise non-equivalence. The priority of these operators is less binding that the arithmetic ones, but more binding that the comparisons. Parentheses may be used as above to force particular groupings. Syntax a # b (bitwise NON EQUIVALENCE) where the arguments a and b may be further expressions. They are converted to the nearest integer to machine precision and then the operation is performed. See Also &, | ¬ manual_src~dfsg/opcodes/pvsdiskin.xml0000644000000000000000000000514612262561504017143 0ustar rootroot pvsdiskin Spectral Processing:Streaming pvsdiskin pvsdiskin Read a selected channel from a PVOC-EX analysis file. Description Create an fsig stream by reading a selected channel from a PVOC-EX analysis file, with frame interpolation. Syntax fsig pvsdiskin SFname,ktscal,kgain[,ioffset, ichan] Initialization Sfname -- Name of the analysis file. This must have the .pvx file extension. A multi-channel PVOC-EX file can be generated using the extended pvanal utility. ichan -- (optional) The channel to read (counting from 1). Default is 1. ioff -- start offset from beginning of file (secs) (default: 0) . Performance ktscal -- time scale, ie. the read pointer speed (1 is normal speed, negative is backwards, 0 < ktscal < 1 is slower and ktscal > 1 is faster) kgain -- gain scaling. Examples Here is an example of the pvsdiskin opcode. It uses the file pvsdiskin.csd. Example of the pvsdiskin opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; May 2007 New in Csound 5.06 manual_src~dfsg/opcodes/wterrain.xml0000644000000000000000000000423712262561504016764 0ustar rootroot wterrain Signal Generators:Wave Terrain Synthesis wterrain wterrain A simple wave-terrain synthesis opcode. Description A simple wave-terrain synthesis opcode. Syntax aout wterrain kamp, kpch, k_xcenter, k_ycenter, k_xradius, k_yradius, \ itabx, itaby Initialization itabx, itaby -- The two tables that define the terrain. Performance The output is the result of drawing an ellipse with axes k_xradius and k_yradius centered at (k_xcenter, k_ycenter), and traversing it at frequency kpch. Examples Here is an example of the wterrain opcode. It uses the file wterrain.csd. Example of the wterrain opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Matthew Gillard New in version 4.19 manual_src~dfsg/opcodes/pcount.xml0000644000000000000000000000565212262561504016443 0ustar rootroot pcount Instrument Control:Sensing and Control pcount pcount Returns the number of pfields belonging to a note event. Description pcount returns the number of pfields belonging to a note event. Syntax icount pcount Initialization icount - stores the number of pfields for the current note event. Note that the reported number of pfields is not necessarily what's explicitly written in the score, but the pfields available to the instrument through mechanisms like pfield carry. Examples Here is an example of the pcount opcode. It uses the file pcount.csd. Example of the pcount opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example will produce the following output: WARNING: instr 1 uses 3 p-fields but is given 8 B 0.000 .. 2.000 T 2.000 TT 2.000 M: 0.49966 0.49966 WARNING: instr 1 uses 3 p-fields but is given 10 B 2.000 .. 4.000 T 4.000 TT 4.000 M: 0.50000 0.50000 WARNING: instr 1 uses 3 p-fields but is given 10 B 4.000 .. 6.000 T 6.000 TT 6.000 M: 0.49943 0.49943 WARNING: instr 1 uses 3 p-fields but is given 6 B 6.000 .. 8.000 T 8.000 TT 8.000 M: 0.00000 0.00000 WARNING: instr 1 uses 3 p-fields but is given 20 B 8.000 .. 18.000 T 18.000 TT 18.000 M: 0.49994 0.49994 The warnings occur because pfields are not used explicitly by the instrument. See Also pindex Credits Example by: Anthony Kozar Dec. 2006 manual_src~dfsg/opcodes/return.xml0000644000000000000000000000265512262561504016452 0ustar rootroot return Instrument Control:Compilation return return Returns a value from an instrument. Description Return will return a value from an instrument at i-time. The value of a global instrument (instrument 0) can be retrieved after compilation by the evalstr opcode. The retrieval of values returned by other instruments is not yet implemented. Syntax return ival Initialization ival -- a value to be returned by instrument. Examples Here is an example of the return opcode in conjunction with evalcode: ival evalstr "return 2 + 2" print ival Credits Author: &namevictor;, 2013 manual_src~dfsg/opcodes/loopseg.xml0000644000000000000000000001057212262561504016600 0ustar rootroot loopseg Signal Generators:Linear and Exponential Generators loopseg loopseg Generate control signal consisting of linear segments delimited by two or more specified points. Description Generate control signal consisting of linear segments delimited by two or more specified points. The entire envelope is looped at kfreq rate. Each parameter can be varied at k-rate. Syntax ksig loopseg kfreq, ktrig, iphase, kvalue0, ktime0 [, kvalue1] [, ktime1] \ [, kvalue2] [, ktime2][...] Initialization iphase -- A value between 0 and 1 to say where to start the loop. Zero, the commonest value, indicates the beginning. Performance ksig -- Output signal. kfreq -- Repeat rate in Hz or fraction of Hz. ktrig -- If non-zero, retriggers the envelope from start (see trigger opcode), before the envelope cycle is completed. kvalue0...kvalueN -- Values of points ktime0...ktimeN -- Times between points; expressed in fractions of a cycle (see below). The final time designates a ramp between the final value and the first value. loopseg opcode is similar to linseg, but the entire envelope is looped at kfreq rate. Notice that times are not expressed in seconds but in fraction of a cycle. Actually each duration represent is proportional to the other, and the entire cycle duration is proportional to the sum of all duration values. The sum of all duration is then rescaled according to kfreq argument. For example, considering an envelope made up of 3 segments, each segment having 100 as duration value, their sum will be 300. This value represents the total duration of the envelope, and is actually divided into 3 equal parts, a part for each segment. Actually, the real envelope duration in seconds is determined by kfreq. Again, if the envelope is made up of 3 segments, but this time the first and last segments have a duration of 50, whereas the central segment has a duration of 100 again, their sum will be 200. This time 200 represent the total duration of the 3 segments, so the central segment will be twice as long as the other segments. All parameters can be varied at k-rate. Negative frequency values are allowed, reading the envelope backward. ktime0 should always be set to 0, except if the user wants some special effect. Examples Here is an example of the loopseg opcode. It uses the file loopseg.csd. Example of the loopseg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also lpshold loopxseg Credits Author: &namegabriel; New in Version 4.13 manual_src~dfsg/opcodes/JackoMidiOutConnect.xml0000644000000000000000000000475412262561504020771 0ustar rootroot JackoMidiOutConnect Jacko Opcodes JackoMidiOutConnect JackoMidiOutConnect Creates a MIDI connection from Csound to a Jack port. Description In the orchestra header, creates a connection from a Jack MIDI output port inside this instance of Csound to an external Jack MIDI input port. Syntax JackoMidiOutConnect ScsoundPortName, SexternalPortName Initialization ScsoundPortName -- The short name ("portname") of the internal Jack MIDI output port. SexternalPortName -- The full name ("clientname:portname") of an external Jack MIDI input port. Performance The actual MIDI data must be written with the JackoMidiOut or JackoNoteOut opcodes. See Also JackoInfo JackoInfo JackoFreewheel JackoAudioOutConnect JackoMidiInConnect JackoMidiOutConnect JackoOn JackoAudioIn JackoAudioOut JackoMidiOut JackoNoteOut JackoTransport Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/seqtime.xml0000644000000000000000000001244412262561504016577 0ustar rootroot seqtime Instrument Control:Sensing and Control seqtime seqtime Generates a trigger signal according to the values stored in a table. Description Generates a trigger signal according to the values stored in a table. Syntax ktrig_out seqtime ktime_unit, kstart, kloop, kinitndx, kfn_times Performance ktrig_out -- output trigger signal ktime_unit -- unit of measure of time, related to seconds. kstart -- start index of looped section kloop -- end index of looped section kinitndx -- initial index Note Although kinitndx is listed as k-rate, it is in fact accessed only at init-time. So if you are using a k-rate argument, it must be assigned with init. kfn_times -- number of table containing a sequence of times This opcode handles timed-sequences of groups of values stored into a table. seqtime generates a trigger signal (a sequence of impulses, see also trigger opcode), according to the values stored in the kfn_times table. This table should contain a series of delta-times (i.e. times beetween to adjacent events). The time units stored into table are expressed in seconds, but can be rescaled by means of ktime_unit argument. The table can be filled with GEN02 or by means of an external text-file containing numbers, with GEN23. Note that the kloop index marks the loop boundary and is NOT included in the looped elements. If you want to loop the first four elements, you would set kstart to 0 and kloop to 4. It is possible to start the sequence from a value different than the first, by assigning to kinitndx an index different than zero (which corresponds to the first value of the table). Normally the sequence is looped, and the start and end of loop can be adjusted by modifying kstart and kloop arguments. User must be sure that values of these arguments (as well as kinitndx) correspond to valid table numbers, otherwise Csound will crash (because no range-checking is implementeted). It is possible to disable loop (one-shot mode) by assigning the same value both to kstart and kloop arguments. In this case, the last read element will be the one corresponding to the value of such arguments. Table can be read backward by assigning a negative kloop value. It is possible to trigger two events almost at the same time (actually separated by a k-cycle) by giving a zero value to the corresponding delta-time. First element contained in the table should be zero, if the user intends to send a trigger impulse, it should come immediately after the orchestra instrument containing seqtime opcode. Examples Here is an example of the seqtime opcode. It uses the file seqtime.csd. Example of the seqtime opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also GEN02, GEN23, trigseq seqtime2 Credits Author: &namegabriel; November 2002. Added a note about the kinitndx parameter, thanks to &namerasmus;. New in version 4.06 Example by: Tim Mortimer and Andres Cabrera 2007 manual_src~dfsg/opcodes/dbfsamp.xml0000644000000000000000000000406612262561504016545 0ustar rootroot dbfsamp Mathematical Operations:Amplitude Functions dbfsamp dbfsamp Returns the decibel equivalent of the raw amplitude x, relative to full scale amplitude. Description Returns the decibel equivalent of the raw amplitude x, relative to full scale amplitude. Full scale is assumed to be 16 bit. New is Csound version 4.10. Syntax dbfsamp(x) (init-rate or control-rate args only) Examples Here is an example of the dbfsamp opcode. It uses the file dbfsamp.csd. Example of the dbfsamp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: idb = -90.309 instr 1: idb = -50.309 instr 1: idb = -30.309 instr 1: idb = -10.309 instr 1: idb = -0.767 See Also ampdb, ampdbfs, dbamp manual_src~dfsg/opcodes/fractalnoise.xml0000644000000000000000000000603512262561504017601 0ustar rootroot fractalnoise Signal Generators:Random (Noise) Generators fractalnoise fractalnoise A fractal noise generator. Description A fractal noise generator implemented as a white noise filtered by a cascade of 15 first-order filters. Syntax ares fractalnoise kamp, kbeta Performance kamp -- amplitude. kbeta -- spectral parameter related to the fractal dimension 0 - white 1 - pink 2 - brown Examples Here is an example of the fractalnoise opcode. It uses the file fractalnoise.csd. Example of the fractalnoise opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. References R. Saletti. A comparison between two methods to generate 1/(f^gamma) noise. In Proc. IEEE, volume 74, pp. 1595-1596, November 1986. G. Corsini and R. Saletti. A 1/(f^gamma) power spectrum noise sequence generator. IEEE Trans. on Instrumentation and Measurement, 37(4):615-619, December 1988. The Sounding Object, edited by Davide Rocchesso and Federico Fontana, Edizioni di Mondo Estremo. Chapter 8 by Federico Avanzini, pp. 154-157. Credits Author: &nametito; January 2012 New in Csound version 5.16 manual_src~dfsg/opcodes/zar.xml0000644000000000000000000000435012262561504015721 0ustar rootroot zar Zak Patch System zar zir Reads from a location in za space at a-rate. Description Reads from a location in za space at a-rate. Syntax ares zar kndx Performance kndx -- points to the za location to be read. zar reads the array of floats at kndx in za space, which are ksmps number of a-rate floats to be processed in a k cycle. Examples Here is an example of the zar opcode. It uses the file zar.csd. Example of the zar opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zarg, zir, zkr Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/ftfree.xml0000644000000000000000000000414112262561504016376 0ustar rootroot ftfree Table Control ftfree ftfree Deletes function table. Description Deletes function table. Syntax ftfree ifno, iwhen Initialization ifno -- the number of the table to be deleted iwhen -- if zero the table is deleted at init time; otherwise the table number is registered for being deleted at note deactivation. Examples Here is an example of the ftfree opcode. It uses the file ftfree.csd. Example of the ftfree opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. instr 1: gitempTable = 101.000 B 0.000 .. 3.000 T 3.000 TT 3.000 M: 0.50000 0.50000 INIT ERROR in instr 1: Invalid ftable no. 101.000000 instr 1: gitempTable = 101.000 Error deleting ftable 101 B 3.000 - note deleted. i1 had 1 init errors B 3.000 .. 5.000 T 5.000 TT 5.000 M: 0.00000 0.00000 Credits Authors: &namesteven;, &nameistvan; 2005 manual_src~dfsg/opcodes/specaddm.xml0000644000000000000000000000374512262561504016714 0ustar rootroot specaddm Spectral Processing:Non-Standard specaddm specaddm Perform a weighted add of two input spectra. Description Perform a weighted add of two input spectra. Syntax wsig specaddm wsig1, wsig2 [, imul2] Initialization imul2 (optional, default=0) -- if non-zero, scale the wsig2 magnitudes before adding. The default value is 0. Performance wsig1 -- the first input spectra. wsig2 -- the second input spectra. Do a weighted add of two input spectra. For each channel of the two input spectra, the two magnitudes are combined and written to the output according to: magout = mag1in + mag2in * imul2 The operation is performed whenever the input wsig1 is sensed to be new. This unit will (at Initialization) verify the consistency of the two spectra (equal size, equal period, equal mag types). See Also specdiff, specfilt, spechist, specscal manual_src~dfsg/opcodes/fluidCCi.xml0000644000000000000000000000545012262561504016611 0ustar rootroot fluidCCi Signal Generators:Sample Playback fluidCCi fluidCCi Sends a MIDI controller data message to fluid. Syntax fluidCCi iEngineNumber, iChannelNumber, iControllerNumber, iValue Description Sends a MIDI controller data (MIDI controller number and value to use) message to a fluid engine by number on the user specified MIDI channel number. Initialization iEngineNumber -- engine number assigned from fluidEngine iChannelNumber -- MIDI channel number to which the Fluidsynth program is assigned: from 0 to 255. MIDI channels numbered 16 or higher are virtual channels. iControllerNumber -- MIDI controller number to use for this message iValue -- value to set for controller (usually 0-127) Performance This opcode is useful for setting controller values at init time. For continous changes, use fluidCCk. Examples Here is an example of the fluidCCi opcodes. It uses the file fluidCCi.csd. See Also fluidEngine, fluidNote, fluidLoad, fluidCCk More information on soundfonts in the Floss Manuals: http://en.flossmanuals.net/csound/ch048_d-reading-midi-files Other information on soundfonts on Wikipedia: http://en.wikipedia.org/wiki/Soundfont Credits Michael Gogins (gogins at pipeline dot com), Steven Yi. Thanks to Peter Hanappe for Fluidsynth. manual_src~dfsg/opcodes/frac.xml0000644000000000000000000000456112262561504016044 0ustar rootroot frac Mathematical Operations:Mathematical Functions frac frac Returns the fractional part of a decimal number. Description Returns the fractional part of x. Syntax frac(x) (init-rate or control-rate args; also works at audio rate in Csound5) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Examples Here is an example of the frac opcode. It uses the file frac.csd. Example of the frac opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: i2 = 0.200 See Also abs, exp, int, log, log10, i, sqrt Credits Example written by &namekevin;. manual_src~dfsg/opcodes/midic7.xml0000644000000000000000000000756412262561504016313 0ustar rootroot midic7 Real-time MIDI:Input midic7 midic7 Allows a floating-point 7-bit MIDI signal scaled with a minimum and a maximum range. Description Allows a floating-point 7-bit MIDI signal scaled with a minimum and a maximum range. Syntax idest midic7 ictlno, imin, imax [, ifn] kdest midic7 ictlno, kmin, kmax [, ifn] Initialization idest -- output signal ictlno -- MIDI controller number (0-127) imin -- user-defined minimum floating-point value of output imax -- user-defined maximum floating-point value of output ifn (optional) -- table to be read when indexing is required. Table must be normalized. Output is scaled according to the imin and imax values. Performance kdest -- output signal kmin -- user-defined minimum floating-point value of output kmax -- user-defined maximum floating-point value of output midic7 (i- and k-rate 7 bit MIDI control) allows a floating-point 7-bit MIDI signal scaled with a minimum and a maximum range. It also allows optional non-interpolated table indexing. In midic7 minimum and maximum values can be varied at k-rate. Please note that the midic family of opcodes are designed for MIDI triggered events, and don't require a channel number since they will respond to the same channel as the one that triggered the instrument (see massign). However they will crash if called from a score driven event. Examples Here is an example of the midic7 opcode. It uses the file midic7.csd Example of the midic7 opcode. See Also ctrl7, ctrl14, ctrl21, initc7, initc14, initc21, midic14, midic21 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct controller number range. manual_src~dfsg/opcodes/dust2.xml0000644000000000000000000000430712262561504016170 0ustar rootroot dust2 Signal Generators:Random (Noise) Generators dust2 dust2 Random impulses. Description Generates random impulses from -1 to 1. Syntax ares dust2 kamp, kdensity kres dust2 kamp, kdensity Performance kamp -- amplitude. kdensity -- average number of impulses per second. Examples Here is an example of the dust2 opcode. It uses the file dust2.csd. Example of the dust2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also dust gausstrig mpulse Credits Based on James McCartney's Dust2 (SuperCollider) Author: &nametito; January 2012 New in Csound version 5.16 manual_src~dfsg/opcodes/slider8table.xml0000644000000000000000000001416612262561504017515 0ustar rootroot slider8table Real-time MIDI:Slider Banks slider8table slider8table Stores a bank of 8 different MIDI control messages to a table. Description Stores a bank of 8 different MIDI control messages to a table. Syntax kflag slider8table ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1,..., ictlnum8, imin8, imax8, init8, ifn8 Initialization i1 ... i8 -- output values ichan -- MIDI channel (1-16) ioutTable -- number of the table that will contain the output ioffset -- output table offset. A zero means that the output of the first slider will affect the first table element. A 10 means that the output of the first slider will affect the 11th table element. ictlnum1 ... ictlnum8 -- MIDI control number (0-127) imin1 ... imin8 -- minimum values for each controller imax1 ... imax8 -- maximum values for each controller init1 ... init8 -- initial value for each controller ifn1 ... ifn8 -- function table for conversion for each controller Performance kflag -- a flag that informs if any control-change message in the bank has been received. In this case kflag is set to 1. Otherwise is set to zero. slider8table handles a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider8table allows a bank of 8 different MIDI control message numbers. As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. slider8table is very similar to slider8 and sliderN family of opcodes (see their manual for more information). The actual difference is that the output is not stored to k-rate variables, but to a table, denoted by the ioutTable argument. It is possible to define a starting index in order to use the same table for more than one slider bank (or other purposes). It is possible to use this opcode together with FLslidBnk2Setk and FLslidBnk2, so you can synchronize the position of the MIDI values to the position of the FLTK valuator widgets of FLslidBnk2. Notice that you have to specify the same min/max values as well the linear/exponential responses in both sliderNtable(f) and FLslidBnk2. The exception is when using table-indexed response instead of a lin/exp response. In this case, in order to achieve a useful result, the table-indexed response and actual min/max values must be set only in FLslidBnk2, whereas, in sliderNtable(f), you have to set a linear response and a minimum of zero and a maximum of one in all sliders. See Also slider16table, slider16tablef, slider32table, slider32tablef, slider64table, slider64tablef, slider8tabletablef Credits Author: &namegabriel; New in Csound version 5.06 manual_src~dfsg/opcodes/wguide1.xml0000644000000000000000000000626312262561504016477 0ustar rootroot wguide1 Signal Modifiers:Waveguides wguide1 wguide1 A simple waveguide model consisting of one delay-line and one first-order lowpass filter. Description A simple waveguide model consisting of one delay-line and one first-order lowpass filter. Syntax ares wguide1 asig, xfreq, kcutoff, kfeedback Performance asig -- the input of excitation noise. xfreq -- the frequency (i.e. the inverse of delay time) Changed to x-rate in Csound version 3.59. kcutoff -- the filter cutoff frequency in Hz. kfeedback -- the feedback factor. wguide1 is the most elemental waveguide model, consisting of one delay-line and one first-order lowpass filter. Implementing waveguide algorithms as opcodes, instead of orc instruments, allows the user to set kr different than sr, allowing better performance particulary when using real-time. wguide1. wguide1. Examples Here is an example of the wguide1 opcode. It uses the file wguide1.csd. Example of the wguide1 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also wguide2 Credits Author: &namegabriel; Italy October 1998 Example written by &namekevin;. New in Csound version 3.49 manual_src~dfsg/opcodes/deltapi.xml0000644000000000000000000001573712262561504016562 0ustar rootroot deltapi Signal Modifiers:Delay deltapi deltapi Taps a delay line at variable offset times, uses interpolation. Description Taps a delay line at variable offset times, uses interpolation. Syntax ares deltapi xdlt Performance xdlt -- specifies the tapped delay time in seconds. Each can range from 1 control period to the full delay time of the read/write pair; however, since there is no internal check for adherence to this range, the user is wholly responsible. Each argument can be a constant, a variable, or a time-varying signal; the xdlt argument in deltapi implies that an audio-varying delay is permitted there. deltapi extracts sound by interpolated readout. By interpolating between adjacent stored samples deltapi represents a particular delay time with more accuracy, but it will take about twice as long to run. This opcode can tap into a delayr/delayw pair, extracting delayed audio from the idlt seconds of stored sound. There can be any number of deltap and/or deltapi units between a read/write pair. Each receives an audio tap with no change of original amplitude. This opcode can provide multiple delay taps for arbitrary delay path and feedback networks. They can deliver either constant-time or time-varying taps, and are useful for building chorus effects, harmonizers, and Doppler shifts. Constant-time delay taps (and some slowly changing ones) do not need interpolated readout; they are well served by deltap. Medium-paced or fast varying dlt's, however, will need the extra services of deltapi. delayr/delayw pairs may be interleaved. To associate a delay tap unit with a specific delayr unit, it not only has to be located between that delayr and the appropriate delayw unit, but must also precede any following delayr units. See Example 2. (This feature added in Csound version 3.57 by Jens Groh and &namejohn;). N.B. k-rate delay times are not internally interpolated, but rather lay down stepped time-shifts of audio samples; this will be found quite adequate for slowly changing tap times. For medium to fast-paced changes, however, one should provide a higher resolution audio-rate timeshift as input. Examples deltap example #1 asource buzz 1, 440, 20, 1 atime linseg 1, p3/2,.01, p3/2,1 ; trace a distance in secs ampfac = 1/atime/atime ; and calc an amp factor adump delayr 1 ; set maximum distance amove deltapi atime ; move sound source past delayw asource ; the listener out amove * ampfac deltap example #2 ainput1 = ..... ainput2 = ..... kdlyt1 = ..... kdlyt2 = ..... ;Read delayed signal, first delayr instance: adump delayr 4.0 adly1 deltap kdlyt1 ; associated with first delayr instance ;Read delayed signal, second delayr instance: adump delayr 4.0 adly2 deltap kdlyt2 ; associated with second delayr instance ;Do some cross-coupled manipulation: afdbk1 = 0.7 * adly1 + 0.7 * adly2 + ainput1 afdbk2 = -0.7 * adly1 + 0.7 * adly2 + ainput2 ;Feed back signal, associated with first delayr instance: delayw afdbk1 ;Feed back signal, associated with second delayr instance: delayw afdbk2 outs adly1, adly2 Here is yet another example of the deltapi opcode. It uses the file deltapi.csd. Example of the deltapi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also deltap, deltap3, deltapn manual_src~dfsg/opcodes/ptrack.xml0000644000000000000000000000640412262561504016413 0ustar rootroot ptrack Instrument Control:Sensing and Control ptrack ptrack Tracks the pitch of a signal. Description ptrack takes an input signal, splits it into ihopsize blocks and using a STFT method, extracts an estimated pitch for its fundamental frequency as well as estimating the total amplitude of the signal in dB, relative to full-scale (0dB). The method implies an analysis window size of 2*ihopsize samples (overlaping by 1/2 window), which has to be a power-of-two, between 128 and 8192 (hopsizes between 64 and 4096). Smaller windows will give better time precision, but worse frequency accuracy (esp. in low fundamentals).This opcode is based on an original algorithm by M. Puckette. Syntax kcps, kamp ptrack asig, ihopsize[,ipeaks] Initialization ihopsize -- size of the analysis 'hop', in samples, required to be power-of-two (min 64, max 4096). This is the period between measurements. ipeaks, ihi -- number of spectral peaks to use in the analysis, defaults to 20 (optional) Performance kcps -- estimated pitch in Hz. kamp -- estimated amplitude in dB relative to full-scale (0dB) (ie. always <= 0). ptrack analyzes the input signal, asig, to give a pitch/amplitude pair of outputs, for the fundamental of a monophonic signal. The output is updated every sr/ihopsize seconds. Examples Here is an example of the ptrack opcode. This example uses the files ptrack.csd. Example of the ptrack opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; NUI, Maynooth. Maynooth, Ireland March, 2007 New in Csound version 5.05 manual_src~dfsg/opcodes/tablemix.xml0000644000000000000000000001230512262561504016731 0ustar rootroot tablemix Table Control:Read/Write Opreations tablemix tablemix Mixes two tables. Description Mixes two tables. Syntax tablemix kdft, kdoff, klen, ks1ft, ks1off, ks1g, ks2ft, ks2off, ks2g Performance kdft -- Destination function table. kdoff -- Offset to start writing from. Can be negative. klen -- Number of write operations to perform. Negative means work backwards. ks1ft, ks2ft -- Source function tables. These can be the same as the destination table, if care is exercised about direction of copying data. ks1off, ks2off -- Offsets to start reading from in source tables. ks1g, ks2g -- Gains to apply when reading from the source tables. The results are added and the sum is written to the destination table. tablemix -- This opcode mixes from two tables, with separate gains into the destination table. Writing is done for klen locations, usually stepping forward through the table - if klen is positive. If it is negative, then the writing and reading order is backwards - towards lower indexes in the tables. This bi-directional option makes it easy to shift the contents of a table sideways by reading from it and writing back to it with a different offset. If klen is 0, no writing occurs. Note that the internal integer value of klen is derived from the ANSI C floor() function - which returns the next most negative integer. Hence a fractional negative klen value of -2.3 would create an internal length of 3, and cause the copying to start from the offset locations and proceed for two locations to the left. The total index for table reading and writing is calculated from the starting offset for each table, plus the index value, which starts at 0 and then increments (or decrements) by 1 as mixing proceeds. These total indexes can potentially be very large, since there is no restriction on the offset or the klen. However each total index for each table is ANDed with a length mask (such as 0000 0111 for a table of length 8) to form a final index which is actually used for reading or writing. So no reading or writing can occur outside the tables. This is the same as wrap mode in table read and write. These opcodes do not read or write the guardpoint. If a table has been rewritten with one of these, then if it has a guardpoint which is supposed to contain the same value as the location 0, then call tablegpw afterwards. The indexes and offsets are all in table steps - they are not normalized to 0 - 1. So for a table of length 256, klen should be set to 256 if all the table was to be read or written. The tables do not need to be the same length - wrapping occurs individually for each table. Examples Here is an example of the tablemix opcode. It uses the file tablemix.csd. Example of the tablemix opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also tablecopy, tablegpw, tableicopy, tableigpw, tableimix Credits Author: Robin Whittle Australia May 1997 New in version 3.47 manual_src~dfsg/opcodes/ftload.xml0000644000000000000000000000412212262561504016373 0ustar rootroot ftload Table Control:Read/Write Opreations ftload ftload Load a set of previously-allocated tables from a file. Description Load a set of previously-allocated tables from a file. Syntax ftload "filename", iflag, ifn1 [, ifn2] [...] Initialization "filename" -- A quoted string containing the name of the file to load. iflag -- Type of the file to load/save. (0 = binary file, Non-zero = text file) ifn1, ifn2, ... -- Numbers of tables to load. Performance ftload loads a list of tables from a file. (The tables have to be already allocated though.) The file's format can be binary or text. Warning The file's format is not compatible with a WAV-file and is not endian-safe. Examples See the example for ftsave. See Also ftloadk, ftsavek, ftsave Credits Author: &namegabriel; New in version 4.21 manual_src~dfsg/opcodes/lposcilsa2.xml0000644000000000000000000000636612262561504017211 0ustar rootroot lposcilsa2 Signal Generators:Sample Playback lposcilsa2 lposcilsa2 Read stereo sampled sound from a table with looping and high precision. Description lposcilsa2 reads stereo sampled sound from a table with looping and high precision. Syntax ar1, ar2 lposcilsa2 aamp, kfreqratio, kloop, kend, ift [,iphs] Initialization ift -- function table number iphs -- initial phase (in samples) Performance ar1, ar2 -- output signal aamp -- amplitude kfreqratio -- multiply factor of table frequency (for example: 1 = original frequency, 2 = an octave up). Only integers are allowed kloop -- start loop point (in samples) kend -- end loop point (in samples) lposcilsa2 is the same as lposcilsa, but no interpolation is implemented and only works with integer kfreqratio values. Much faster than lposcilsa, it is mainly intended to be used with kfreqratio = 1, being in this case a fast substitute of soundin, since the soundfile must be entirely loaded in memory. Examples Here is an example of the lposcilsa2 opcode. It uses the file lposcilsa2.csd. Example of the lposcilsa2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also lposcil, lposcila, lposcilsa Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/floor.xml0000644000000000000000000000577012262561504016255 0ustar rootroot floor Mathematical Operations:Mathematical Functions floor floor Returns the largest integer not greater than x Description Returns the largest integer not greater than x Syntax floor(x) (init-, control-, or audio-rate arg allowed) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Examples Here is an example of the floor opcode. It uses the file floor.csd. Example of the floor opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output should include lines like these: instr 1: inumber = 9.000 idiv = 1.000 ifl = 9.000 instr 1: inumber = 9.000 idiv = 2.000 ifl = 4.000 instr 1: inumber = 9.000 idiv = 3.000 ifl = 3.000 instr 1: inumber = 9.000 idiv = 4.000 ifl = 2.000 instr 1: inumber = 9.000 idiv = 5.000 ifl = 1.000 instr 1: inumber = 9.000 idiv = 6.000 ifl = 1.000 instr 1: inumber = 9.000 idiv = 7.000 ifl = 1.000 instr 1: inumber = 9.000 idiv = 8.000 ifl = 1.000 instr 1: inumber = 9.000 idiv = 9.000 ifl = 1.000 instr 1: inumber = 9.000 idiv = 10.000 ifl = 0.000 See Also abs, exp, int, log, log10, i, sqrt Credits Author: Istvan Varga New in Csound 5 2005 manual_src~dfsg/opcodes/envlpxr.xml0000644000000000000000000001460212262561504016624 0ustar rootroot envlpxr Signal Generators:Envelope Generators envlpxr envlpxr The envlpx opcode with a final release segment. Description envlpxr is the same as envlpx except that the final segment is entered only on sensing a MIDI note release. The note is then extended by the decay time. Syntax ares envlpxr xamp, irise, idec, ifn, iatss, iatdec [, ixmod] [,irind] kres envlpxr kamp, irise, idec, ifn, iatss, iatdec [, ixmod] [,irind] Initialization irise -- rise time in seconds. A zero or negative value signifies no rise modification. idec -- decay time in seconds. Zero means no decay. ifn -- function table number of stored rise shape with extended guard point. iatss -- attenuation factor, by which the last value of the envlpxr rise is modified during the note's pseudo steady state. A factor greater than 1 causes an exponential growth and a factor less than 1 creates an exponential decay. A factor of 1 will maintain a true steady state at the last rise value. Note that this attenuation is not by fixed rate (as in a piano), but is sensitive to a note's duration. However, if iatss is negative (or if steady state < 4 k-periods) a fixed attenuation rate of abs(iatss) per second will be used. 0 is illegal. iatdec -- attenuation factor by which the closing steady state value is reduced exponentially over the decay period. This value must be positive and is normally of the order of .01. A large or excessively small value is apt to produce a cutoff which is audible. A zero or negative value is illegal. ixmod (optional, between +- .9 or so) -- exponential curve modifier, influencing the steepness of the exponential trajectory during the steady state. Values less than zero will cause an accelerated growth or decay towards the target (e.g. subito piano). Values greater than zero will cause a retarded growth or decay. The default value is zero (unmodified exponential). irind (optional) -- independence flag. If left zero, the release time (idec) will influence the extended life of the current note following a note-off. If non-zero, the idec time is quite independent of the note extension (see below). The default value is 0. Performance kamp, xamp -- input amplitude signal. envlpxr is an example of the special Csound r units that contain a note-off sensor and release time extender. When each senses a score event termination or a MIDI noteoff, it will immediately extend the performance time of the current instrument by idec seconds unless it is made independent by irind. Then it will begin a decay from wherever it was at the time. You can use other pre-made envelopes which start a release segment upon recieving a note off message, like linsegr and expsegr, or you can construct more complex envelopes using xtratim and release. Note that you don't need to use xtratim if you are using envlpxr, since the time is extended automatically. These r units can also be modified by MIDI noteoff velocities (see veloffs). If the irind flag is on (non-zero), the overall performance time is unaffected by note-off and veloff data. Multiple <quote>r</quote> units When two or more r units occur in the same instrument it is usual to have only one of them influence the overall note duration. This is normally the master amplitude unit. Other units controlling, say, filter motion can still be sensitive to note-off commands while not affecting the duration by making them independent (irind non-zero). Depending on their own idec (release time) values, independent r units may or may not reach their final destinations before the instrument terminates. If they do, they will simply hold their target values until termination. If two or more r units are simultaneously master, note extension is by the greatest idec. Examples Here is an example of the envlpxr opcode. It uses the file envlpxr.csd. Example of the envlpxr opcode. See Also envlpx, linen, linenr Credits Thanks goes to &nameluis; for pointing out a mistake with iatss. manual_src~dfsg/opcodes/gogobel.xml0000644000000000000000000000667312262561504016555 0ustar rootroot gogobel Signal Generators:Models and Emulations gogobel gogobel Audio output is a tone related to the striking of a cow bell or similar. Description Audio output is a tone related to the striking of a cow bell or similar. The method is a physical model developed from &nameperry;, but re-coded for Csound. Syntax ares gogobel kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivfn Initialization ihrd -- the hardness of the stick used in the strike. A range of 0 to 1 is used. 0.5 is a suitable value. ipos -- where the block is hit, in the range 0 to 1. imp -- a table of the strike impulses. The file marmstk1.wav is a suitable function from measurements and can be loaded with a GEN01 table. It is also available at ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. ivfn -- shape of vibrato, usually a sine table, created by a function. Performance A note is played on a cowbell-like instrument, with the arguments as below. kamp -- Amplitude of note. kfreq -- Frequency of note played. kvibf -- frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp -- amplitude of the vibrato Examples Here is an example of the gogobel opcode. It uses the file gogobel.csd, and marmstk1.wav, Example of the gogobel opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/schedkwhen.xml0000644000000000000000000001257212262561504017255 0ustar rootroot schedkwhen Instrument Control:Invocation schedkwhen schedkwhen Adds a new score event generated by a k-rate trigger. Description Adds a new score event generated by a k-rate trigger. Syntax schedkwhen ktrigger, kmintim, kmaxnum, kinsnum, kwhen, kdur \ [, ip4] [, ip5] [...] schedkwhen ktrigger, kmintim, kmaxnum, "insname", kwhen, kdur \ [, ip4] [, ip5] [...] Initialization insname -- A string (in double-quotes) representing a named instrument. ip4, ip5, ... -- Equivalent to p4, p5, etc., in a score i statement Performance ktrigger -- triggers a new score event. If ktrigger = 0, no new event is triggered. kmintim -- minimum time between generated events, in seconds. If kmintim <= 0, no time limit exists. If the kinsnum is negative (to turn off an instrument), this test is bypassed. kmaxnum -- maximum number of simultaneous instances of instrument kinsnum allowed. If the number of existant instances of kinsnum is >= kmaxnum, no new event is generated. If kmaxnum is <= 0, it is not used to limit event generation. If the kinsnum is negative (to turn off an instrument), this test is bypassed. kinsnum -- instrument number. Equivalent to p1 in a score i statement. kwhen -- start time of the new event. Equivalent to p2 in a score i statement. Measured from the time of the triggering event. kwhen must be >= 0. If kwhen > 0, the instrument will not be initialized until the actual time when it should start performing. kdur -- duration of event. Equivalent to p3 in a score i statement. If kdur = 0, the instrument will only do an initialization pass, with no performance. If kdur is negative, a held note is initiated. (See ihold and i statement.) While waiting for events to be triggered by schedkwhen, the performance must be kept going, or Csound may quit if no score events are expected. To guarantee continued performance, an f0 statement may be used in the score. Note that the schedkwhen opcode can't accept string p-fields. If you need to pass strings when instantiating an instrument, use the scoreline or scoreline_i opcode. Examples Here is an example of the schedkwhen opcode. It uses the file schedkwhen.csd. Example of the schedkwhen opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also event, event_i, schedule, schedwhen, schedkwhennamed, scoreline, scoreline_i Credits Author: Rasmus Ekman EMS, Stockholm, Sweden Example written by &namekevin;. New in Csound version 3.59 manual_src~dfsg/opcodes/birnd.xml0000644000000000000000000000546612262561504016234 0ustar rootroot birnd Mathematical Operations:Random Functions birnd birnd Returns a random number in a bi-polar range. Description Returns a random number in a bi-polar range. Syntax birnd(x) (init- or control-rate only) Where the argument within the parentheses may be an expression. These value converters sample a global random sequence, but do not reference seed. The result can be a term in a further expression. Performance Returns a random number in the bipolar range -x to x. rnd and birnd obtain values from a global pseudo-random number generator, then scale them into the requested range. The single global generator will thus distribute its sequence to these units throughout the performance, in whatever order the requests arrive. Examples Here is an example of the birnd opcode. It uses the file birnd.csd. Example of the birnd opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: instr 1: i1 = 0.94700 instr 1: i1 = -0.72119 instr 1: i1 = 0.53898 instr 1: i1 = 0.05001 instr 1: i1 = 0.24533 instr 1: i1 = 0.93902 instr 1: i1 = 0.43364 See Also rnd Credits Author: &namebarry; MIT Cambridge, Massachussetts 1997 Extended in 3.47 to x-rate by &namejohn;. manual_src~dfsg/opcodes/poscil3.xml0000644000000000000000000000744612262561504016512 0ustar rootroot poscil3 Signal Generators:Basic Oscillators poscil3 poscil3 High precision oscillator with cubic interpolation. Description High precision oscillator with cubic interpolation. Syntax ares poscil3 aamp, acps [, ifn, iphs] ares poscil3 aamp, kcps [, ifn, iphs] ares poscil3 kamp, acps [, ifn, iphs] ares poscil3 kamp, kcps [, ifn, iphs] ires poscil3 kamp, kcps [, ifn, iphs] kres poscil3 kamp, kcps [, ifn, iphs] Initialization ifn -- (optional) function table number. This defaults to -1 which indicates a sinewave. iphs (optional, default=0) -- initial phase (normalized table index 0-1) Performance ares -- output signal kamp, aamp -- the amplitude of the output signal. kcps, acps -- the frequency of the output signal in cycles per second. poscil3 works like poscil, but uses cubic interpolation. Note that poscil3 can use defered (non-power of two) length tables. Examples Here is an example of the poscil3 opcode. It uses the file poscil3.csd. Example of the poscil3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the poscil3 opcode, which uses a table filled from a sound file. It uses the file poscil3-file.csd . Another example of the poscil3 opcode. See Also poscil Credits Authors: &namejohn;, &namegabriel; Italy New in Csound version 3.52 variants with a-rate amplitude or frequency new in 5.16 manual_src~dfsg/opcodes/pchoct.xml0000644000000000000000000001344412262561504016411 0ustar rootroot pchoct Pitch Converters:Functions pchoct pchoct Converts an octave-point-decimal value to pitch-class. Description Converts an octave-point-decimal value to pitch-class. Syntax pchoct (oct) (init- or control-rate args only) where the argument within the parentheses may be a further expression. Performance pchoct and its related opcodes are really value converters with a special function of manipulating pitch data. Data concerning pitch and frequency can exist in any of the following forms: Pitch and Frequency Values Name Abbreviation octave point pitch-class (8ve.pc) pch octave point decimal oct cycles per second cps Midi note number (0-127) midinn
The first two forms consist of a whole number, representing octave registration, followed by a specially interpreted fractional part. For pch, the fraction is read as two decimal digits representing the 12 equal-tempered pitch classes from .00 for C to .11 for B. For oct, the fraction is interpreted as a true decimal fractional part of an octave. The two fractional forms are thus related by the factor 100/12. In both forms, the fraction is preceded by a whole number octave index such that 8.00 represents Middle C, 9.00 the C above, etc. Midi note number values range between 0 and 127 (inclusively) with 60 representing Middle C, and are usually whole numbers. Thus A440 can be represented alternatively by 440 (cps), 69 (midinn), 8.09 (pch), or 8.75 (oct). Microtonal divisions of the pch semitone can be encoded by using more than two decimal places. The mnemonics of the pitch conversion units are derived from morphemes of the forms involved, the second morpheme describing the source and the first morpheme the object (result). Thus cpspch(8.09) will convert the pitch argument 8.09 to its cps (or Hertz) equivalent, giving the value of 440. Since the argument is constant over the duration of the note, this conversion will take place at i-time, before any samples for the current note are produced. By contrast, the conversion cpsoct(8.75 + k1) which gives the value of A440 transposed by the octave interval k1. The calculation will be repeated every k-period since that is the rate at which k1 varies. Note The conversion from pch, oct, or midinn into cps is not a linear operation but involves an exponential process that could be time-consuming when executed repeatedly. Csound now uses a built-in table lookup to do this efficiently, even at audio rates. Because the table index is truncated without interpolation, pitch resolution when using one of these opcodes is limited to 8192 discrete and equal divisions of the octave, and some pitches of the standard 12-tone equally-tempered scale are very slightly mistuned (by at most 0.15 cents). Examples Here is an example of the pchoct opcode. It uses the file pchoct.csd. Example of the pchoct opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: ipch = 8.090 See Also cpsoct, cpspch, octcps, octpch, cpsmidinn, octmidinn, pchmidinn Credits Example written by &namekevin;. manual_src~dfsg/opcodes/fout.xml0000644000000000000000000001622212262561504016103 0ustar rootroot fout Signal I/O:File I/O fout fout Outputs a-rate signals to an arbitrary number of channels. Description fout outputs N a-rate signals to a specified file of N channels. Syntax fout ifilename, iformat, aout1 [, aout2, aout3,...,aoutN] fout ifilename, iformat, array[] Initialization ifilename -- the output file's name (in double-quotes). iformat -- a flag to choose output file format (note: Csound versions older than 5.0 may only support formats 0, 1, and 2): 0 - 32-bit floating point samples without header (binary PCM multichannel file) 1 - 16-bit integers without header (binary PCM multichannel file) 2 - 16-bit integers with a header. The header type depends on the render (-o) format. For example, if the user chooses the AIFF format (using the -A flag), the header format will be AIFF type. 3 - u-law samples with a header (see iformat=2). 4 - 16-bit integers with a header (see iformat=2). 5 - 32-bit integers with a header (see iformat=2). 6 - 32-bit floats with a header (see iformat=2). 7 - 8-bit unsigned integers with a header (see iformat=2). 8 - 24-bit integers with a header (see iformat=2). 9 - 64-bit floats with a header (see iformat=2). In addition, Csound versions 5.0 and later allow for explicitly selecting a particular header type by specifying the format as 10 * fileType + sampleFormat, where fileType may be 1 for WAV, 2 for AIFF, 3 for raw (headerless) files, and 4 for IRCAM; sampleFormat is one of the above values in the range 0 to 9, except sample format 0 is taken from the command line (-o), format 1 is 8-bit signed integers, and format 2 is a-law. So, for example, iformat=25 means 32-bit integers with AIFF header. Performance aout1,... aoutN -- signals to be written to the file. In the case of raw files, the expected range of audio signals is determined by the selected sample format; for sound files with a header like WAV and AIFF, the audio signals should be in the range -0dbfs to 0dbfs. fout (file output) writes samples of audio signals to a file with any number of channels. Channel number depends by the number of aoutN variables (i.e. a mono signal with only an a-rate argument, a stereo signal with two a-rate arguments etc.) Maximum number of channels is fixed to 64. Multiple fout opcodes can be present in the same instrument, referring to different files. Notice that, unlike out, outs and outq, fout does not zero the audio variable so you must zero it after calling it. If polyphony is to be used, you can use vincr and clear opcodes for this task. Notice that fout and foutk can use either a string containing a file pathname, or a handle-number generated by fiopen. Whereas, with fouti and foutir, the target file can be only specified by means of a handle-number. If you are using fout to generate an audio file for the global output of csound, you might want to use the monitor opcode, which can tap the output buffer, to avoid having to route Examples Here is a simple example of the fout opcode. It uses the file fout.csd. Example of the fout opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of fout, using it to save the contents of a table to an audio file. It uses the file fout_ftable.csd and beats.wav. Example of the fout opcode to save the contents of an f-table. See Also fiopen, fouti, foutir, foutk, monitor Credits Author: &namegabriel; Italy 1999 Author: &namejohn; NUIM, 2013 New in Csound version 3.56 October 2002. Added a note from Richard Dobson. Array version new in 6.01 manual_src~dfsg/opcodes/JackoAudioOut.xml0000644000000000000000000000511412262561504017625 0ustar rootroot JackoAudioOut Jacko Opcodes JackoAudioOut JackoAudioOut Sends an audio signal to a Jack port. Description Sends an audio signal to an internal Jack audio output port, and in turn to its connected external Jack audio input port. Note that it is possible to send audio out via Jack to the system audio interface, while at the same time rendering to a regular Csound output soundfile. Syntax JackoAudioOut ScsoundPortName, asignal Initialization ScsoundPortName -- The short name ("portname") of the internal Jack audio output port. Performance asignal -- Audio to be sent to the external Jack audio input port to which CsoundPortName is connected. Audio from multiple instances of the opcode sending to the same Jack port is summed before sending. See Also JackoInfo, JackoInfo, JackoFreewheel, JackoAudioOutConnect, JackoMidiInConnect, JackoMidiOutConnect, JackoOn, JackoAudioIn, JackoMidiOut, JackoMidiOut, JackoTransport. Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/oscil3.xml0000644000000000000000000000630012262561504016316 0ustar rootroot oscil3 Signal Generators:Basic Oscillators oscil3 oscil3 A simple oscillator with cubic interpolation. Description oscil3 reads table ifn sequentially and repeatedly at a frequency xcps. The amplitude is scaled by xamp. Cubic interpolation is applied for table look up from internal phase values. Syntax ares oscil3 xamp, xcps [, ifn, iphs] kres oscil3 kamp, kcps [, ifn, iphs] Initialization ifn (optional) -- function table number. Requires a wrap-around guard point. Defaults to -1 indicating a sinewave. iphs (optional) -- initial phase of sampling, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is 0. Performance kamp, xamp -- amplitude kcps, xcps -- frequency in cycles per second. oscil3 is identical to oscili, except that it uses cubic interpolation. Table ifn is incrementally sampled modulo the table length and the value obtained is multiplied by amp. If you need to change the oscillator table with a k-rate signal, you can use oscilikt. Examples Here is an example of the oscil3 opcode. It uses the file oscil3.csd. Example of the oscil3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also oscil, oscili, oscilikt Credits Author: &namejohn; New in Csound version 3.50 manual_src~dfsg/opcodes/vmult.xml0000644000000000000000000001373312262561504016301 0ustar rootroot vmult Vectorial:Scalar operations vmult vmult Multiplies a vector in a table by a scalar value. Description Multiplies a vector in a table by a scalar value. Syntax vmult ifn, kval, kelements [, kdstoffset] [, kverbose] Initialization ifn - number of the table hosting the vectorial signal to be processed Performance kval - scalar value to be multiplied kelements - number of elements of the vector kdstoffset - index offset for the destination table (Optional, default = 0) kverbose - Selects whether or not warnings are printed (Default=0) vmult multiplies each element of the vector contained in the table ifn by kval, starting from table index kdstoffset. This enables you to process a specific section of a table by specifying the offset and the number of elements to be processed. Offset is counted starting from 0, so if no offset is specified (or set to 0), the table will be modified from the beginning. Note that this opcode runs at k-rate so the value of kval is multiplied every control period. Use with care or you will end up with very large numbers (or use vmult_i). These opcodes (vadd, vmult, vpow and vexp) perform numeric operations between a vectorial control signal (hosted by the table ifn), and a scalar signal (kval). Result is a new vector that overrides old values of ifn. All these opcodes work at k-rate. Negative values for kdstoffset are valid. Elements from the vector that are outside the table, will be discarded, and they will not wrap around the table. If the optional kverbose argument is different to 0, the opcode will print warning messages every k-pass if table lengths are exceeded. In all these opcodes, the resulting vectors are stored in ifn, overriding the initial vectors. If you want to keep initial vector, use vcopy or vcopy_i to copy it in another table. All these operators are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2, etc. They can also be useful in conjunction with the spectral opcodes pvsftw and pvsftr. Please note that the elements argument has changed in version 5.03 from i-rate to k-rate. This will change the opcode's behavior in the unusual cases where the i-rate variable ielements is changed inside the instrument, for example in: instr 1 ielements = 10 vadd 1, 1, ielements ielements = 20 vadd 2, 1, ielements turnoff endin Example Here is an example of the vmult opcode. It uses the file vmult-2.csd. Example of the vmult opcode. Here is another example of the vmult opcode. It uses the file vmult.csd. Example of the vmult opcode. See also vadd_i, vadd_i, vmult, vpow and vexp. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. Example by Andres Cabrera. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/ihold.xml0000644000000000000000000000447012262561504016227 0ustar rootroot ihold Instrument Control:Duration Control ihold ihold Creates a held note. Description Causes a finite-duration note to become a held note Syntax ihold Performance ihold -- this i-time statement causes a finite-duration note to become a held note. It thus has the same effect as a negative p3 ( see score i Statement), except that p3 here remains positive and the instrument reclassifies itself to being held indefinitely. The note can be turned off explicitly with turnoff, or its space taken over by another note of the same instrument number (i.e. it is tied into that note). Effective at i-time only; no-op during a reinit pass. Examples Here is an example of the ihold opcode. It uses the file ihold.csd. Example of the ihold opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also i Statement, turnoff Credits Example written by &namekevin;. manual_src~dfsg/opcodes/inz.xml0000644000000000000000000000336112262561504015726 0ustar rootroot inz Signal I/O:Signal Input inz inz Reads multi-channel audio samples into a ZAK array from an external device or stream. Description Reads multi-channel audio samples into a ZAK array from an external device or stream. Syntax inz ksig1 Performance inz reads audio samples in nchnls into a ZAK array starting at ksig1. If the command-line -i flag is set, sound is read continuously from the audio input stream (e.g. stdin or a soundfile) into an internal buffer. See Also in32, inch, inx Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/flslidbnkgethandle.xml0000644000000000000000000000617612262561504020761 0ustar rootroot FLslidBnkGetHandle FLTK:Other FLslidBnkGetHandle FLslidBnkGetHandle gets the handle of last slider bank created. Description FLslidBnkGetHandle gets the handle of last slider bank created. Syntax ihandle FLslidBnkGetHandle Initialization ihandle - handle of the sliderBnk (to be used to set its values). Performance There are no k-rate arguments, even if cells of the output table (or the zak space) are updated at k-rate. FLslidBnkGetHandle gets the handle of last slider bank created. This opcode must follow corresponding FLslidBnk (or FLvslidBnk, FLslidBnk2 and FLvslidBnk2) immediately, in order to get its handle. See the entry for FLslidBnk2Setk to see an example of usage. See Also FLslider, FLslidBnk, FLslidBnk2, FLvslidBnk, FLvslidBnk2, FLslidBnk2Set, FLslidBnk2Setk Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/kread.xml0000644000000000000000000000075312262561504016216 0ustar rootroot kread kread kread Deprecated. Description Deprecated as of version 3.52. Use the readk opcode instead. manual_src~dfsg/opcodes/vtablei.xml0000644000000000000000000000771712262561504016565 0ustar rootroot vtablei Vectorial:Tables vtablei vtablei Read vectors (from tables -or arrays of vectors). Description This opcode reads vectors from tables. Syntax vtablei indx, ifn, interp, ixmode, iout1 [, iout2, iout3, .... , ioutN ] Initialization indx - Index into f-table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0). ifn - table number iout1...ioutN - output vector elements ixmode - index data mode. The default value is 0. == 0 index is treated as a raw table location, == 1 index is normalized (0 to 1). interp - vtable (vector table) family of opcodes allows the user to switch beetween interpolated or non-interpolated output by means of the interp argument. Performance This opcode is useful in all cases in which one needs to access sets of values associated to unique indexes (for example, multi-channel samples, STFT bin frames, spectral formants, p-field based scores etc.). The number of elements of each vector (length of the vector) is determined by the number of optional arguments on the right (iout1 , iout2, iout3, .... ioutN). vtable (vector table) family of opcodes allows the user to switch beetween interpolated or non-interpolated output by means of the interp argument. Notice that no wrap nor limit mode is implemented. So, if an index attempt to access to a zone not allocated by the table, Csound will probably crash. However this drawback can be easily avoided by using wrap or limit opcodes applied to indexes before using vtable, in order to correct eventual out-of-range values. Notice that vtablei's output arguments are placed at the right of the opcode name, differently from usual (this style is already used in other opcodes using undefined lists of output arguments such as fin or trigseq). Examples Here is an example of the vtablei opcode. It uses the files vtablei.csd Example of the vtablei opcode. See also vtablea, vtablek, vtabi, vtablewi, vtabwi, Credits Example written by Andres Cabrera. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/outkpc.xml0000644000000000000000000001107612262561504016435 0ustar rootroot outkpc Real-time MIDI:Output outkpc outkpc Sends MIDI program change messages at k-rate. Description Sends MIDI program change messages at k-rate. Syntax outkpc kchn, kprog, kmin, kmax Performance kchn -- MIDI channel number (1-16) kprog -- program change number in floating point kmin -- minimum floating point value (converted in MIDI integer value 0) kmax -- maximum floating point value (converted in MIDI integer value 127 (7 bit)) outkpc (k-rate program change output) sends program change messages. It works only with MIDI instruments which recognize them. These opcodes can drive a different value of a parameter for each note currently active. It can scale the k-value floating-point argument according to the kmin and kmax values. For example: set kmin = 1.0 and kmax = 2.0. When the kvalue argument receives a 2.0 value, the opcode will send a 127 value to the MIDI OUT device. When the kvalue argument receives a 1.0 value, it will send a 0 value. k-rate opcodes send a message each time the MIDI converted value of argument kvalue changes. Examples Here is an example of the outkpc opcode. It uses the file outkpc.csd. Example of the outkpc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This example generates a program change and a note on Csound's MIDI output port whenever a note is received on channel 1. Be sure to have something connected to Csound's MIDI out port to hear the result. Here is another example of the outkpc opcode. It uses the file outkpc_flkt.csd. Example of the outkpc opcode using FLTK. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also outiat, outic14, outic, outipat, outipb, outipc, outkat, outkc14, outkc, outkpat, outkpb Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/ftchnls.xml0000644000000000000000000000522012262561504016563 0ustar rootroot ftchnls Table Control:Table Queries ftchnls ftchnls Returns the number of channels in a stored function table. Description Returns the number of channels in a stored function table. Syntax ftchnls(x) (init-rate args only) Performance Returns the number of channels of a GEN01 table, determined from the header of the original file. If the original file has no header or the table was not created by these GEN01, ftchnls returns -1. Examples Here is an example of the ftchnls opcode. It uses the files ftchnls.csd, mary.wav, and kickroll.wav. Example of the ftchnls opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The audio file mary.wav is monophonic, the audio file kickroll.wav is stereophonic, so its output should include lines like these: instr 1: ichnls = 1.000 instr 1: ichnls = 2.000 See Also ftlen, ftlptim, ftsr, nsamp Credits Author: Chris McCormick Perth, Australia December 2001 manual_src~dfsg/opcodes/getcfg.xml0000644000000000000000000000606712262561504016373 0ustar rootroot getcfg Instrument Control:Sensing and Control getcfg getcfg Return Csound settings. Description Return various configuration settings in Svalue as a string at init time. Syntax Svalue getcfg iopt Initialization iopt -- The parameter to be returned, can be one of: 1: the maximum length of string variables in characters; this is at least the value of the -+max_str_len command line option - 1 2: the input sound file name (-i), or empty if there is no input file 3: the output sound file name (-o), or empty if there is no output file 4: return "1" if real time audio input or output is being used, and "0" otherwise 5: return "1" if running in beat mode (-t command line option), and "0" otherwise 6: the host operating system name 7: return "1" if a callback function for the chnrecv and chnsend opcodes has been set, and "0" otherwise (which means these opcodes do nothing) Examples Here is an example of the getcfg opcode. It uses the file getcfg.csd. Example of the getcfg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output should include lines like these: ------------------------------ Max string len : 255 Input file name (-i) : adc Output file name (-o) : dac RTaudio (-odac) : 1 Beat mode (-t)? : 0 Host Op. Sys. : Linux Callback ? : 0 ------------------------------ Credits Author: Istvan Varga 2006 New in version 5.02 manual_src~dfsg/opcodes/loscil.xml0000644000000000000000000001634612262561504016422 0ustar rootroot loscil Signal Generators:Sample Playback loscil loscil Read sampled sound from a table. Description Read sampled sound (mono or stereo) from a table, with optional sustain and release looping. Syntax ar1 [,ar2] loscil xamp, kcps, ifn [, ibas] [, imod1] [, ibeg1] [, iend1] \ [, imod2] [, ibeg2] [, iend2] Initialization ifn -- function table number, typically denoting an sampled sound segment with prescribed looping points loaded using GEN01. The source file may be mono or stereo. ibas (optional) -- base frequency in Hz of the recorded sound. This optionally overrides the frequency given in the audio file, but is required if the file did not contain one. The default value is 261.626 Hz, i.e. middle C. (New in Csound 4.03). If this value is not known or not present, use 1 here and in kcps. imod1, imod2 (optional, default=-1) -- play modes for the sustain and release loops. A value of 1 denotes normal looping, 2 denotes forward & backward looping, 0 denotes no looping. The default value (-1) will defer to the mode and the looping points given in the source file. Make sure you select an appropriate mode if the file does not contain this information. ibeg1, iend1, ibeg2, iend2 (optional, dependent on mod1, mod2) -- begin and end points of the sustain and release loops. These are measured in sample frames from the beginning of the file, so will look the same whether the sound segment is monaural or stereo. If no loop points are specified, and a looping mode (imod1, imod2) is given, the file will be looped for the whole length. Performance ar1, ar2 -- the output at audio-rate. There is just ar1 for mono output. However, there is both ar1 and ar2 for stereo output. xamp -- the amplitude of the output signal. kcps -- the frequency of the output signal in cycles per second. loscil samples the ftable audio at a rate determined by kcps, then multiplies the result by xamp. The sampling increment for kcps is dependent on the table's base-note frequency ibas, and is automatically adjusted if the orchestra sr value differs from that at which the source was recorded. In this unit, ftable is always sampled with interpolation. If sampling reaches the sustain loop endpoint and looping is in effect, the point of sampling will be modified and loscil will continue reading from within that loop segment. Once the instrument has received a turnoff signal (from the score or from a MIDI noteoff event), the next sustain endpoint encountered will be ignored and sampling will continue towards the release loop end-point, or towards the last sample (henceforth to zeros). loscil is the basic unit for building a sampling synthesizer. Given a sufficient set of recorded piano tones, for example, this unit can resample them to simulate the missing tones. Locating the sound source nearest a desired pitch can be done via table lookup. Once a sampling instrument has begun, its turnoff point may be unpredictable and require an external release envelope; this is often done by gating the sampled audio with linenr, which will extend the duration of a turned-off instrument by a specific period while it implements a decay. If you want to loop the whole file, specify a looping mode in imod1 and do not enter any values for ibeg and iend. Note to Windows users Windows users typically use back-slashes, \, when specifying the paths of their files. As an example, a Windows user might use the path c:\music\samples\loop001.wav. This is problematic because back-slashes are normally used to specify special characters. To correctly specify this path in Csound, one may alternately: Use forward slashes: c:/music/samples/loop001.wav Use back-slash special characters, \\: c:\\music\\samples\\loop001.wav Note This is mono loscil: a1 loscil 10000, 1, 1, 1 ,1 ...and this is stereo loscil: a1, a2 loscil 10000, 1, 1, 1 ,1 Examples Here is an example of the loscil opcode. It uses the file loscil.csd, mary.wav and kickroll.wav. Example of the loscil opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also loscil3 and GEN01 Credits Note about the mono/stereo difference was contributed by &namerasmus;. manual_src~dfsg/opcodes/in32.xml0000644000000000000000000000417612262561504015706 0ustar rootroot in32 Signal I/O:Signal Input in32 in32 Reads a 32-channel audio signal from an external device or stream. Description Reads a 32-channel audio signal from an external device or stream. Warning This opcode is designed to be used only with orchestras that have nchnls_i=32. Doing so with orchestras with nchnls_i > 32 will cause incorrect audio input. Syntax ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, ar13, ar14, \ ar15, ar16, ar17, ar18, ar19, ar20, ar21, ar22, ar23, ar24, ar25, ar26, \ ar27, ar28, ar29, ar30, ar31, ar32 in32 Performance in32 reads a 32-channel audio signal from an external device or stream. If the command-line -i flag is set, sound is read continuously from the audio input stream (e.g. stdin or a soundfile) into an internal buffer. See Also inch, inx, inz Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/sfilist.xml0000644000000000000000000000732312262561504016605 0ustar rootroot sfilist Signal Generators:Sample Playback sfilist sfilist Prints a list of all instruments of a previously loaded SoundFont2 (SF2) file. Description Prints a list of all instruments of a previously loaded SoundFont2 (SF2) sample file. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. Syntax sfilist ifilhandle Initialization ifilhandle -- unique number generated by sfload opcode to be used as an identifier for a SF2 file. Several SF2 files can be loaded and activated at the same time. Performance sfilist prints a list of all instruments of a previously loaded SF2 file to the console. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. Examples Here is an example of the sfilist opcode. It uses the file sfilist.csd. Example of the sfilist opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: Instrument list of "sf_GMbank.sf2" 0) Piano 1 1) Piano 2 2) CP 70 3) EP 1 layer 1 4) EP 1 layer 2 5) E.Piano 2 6) Harpsichord ....... 100) Halo Pad ....... See Also sfinstr, sfinstrm, sfload, sfpassign, sfplay, sfplaym, sfplist, sfpreset Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/ceil.xml0000644000000000000000000000507312262561504016044 0ustar rootroot ceil Mathematical Operations:Mathematical Functions ceil ceil Returns the smallest integer not less than x Description Returns the smallest integer not less than x Syntax ceil(x) (init-, control-, or audio-rate arg allowed) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Examples Here is an example of the ceil opcode. It uses the file ceil.csd. Example of the ceil opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: instr 1: iceil = -1.000 instr 1: iceil = 1.000 instr 1: iceil = 1.000 instr 1: iceil = 1.000 instr 1: iceil = 0.000 instr 1: iceil = 0.000 instr 1: iceil = 0.000 See Also abs, exp, int, log, log10, i, sqrt Credits Author: Istvan Varga New in Csound 5 2005 manual_src~dfsg/opcodes/JackoAudioIn.xml0000644000000000000000000000461212262561504017426 0ustar rootroot JackoAudioIn Jacko Opcodes JackoAudioIn JackoAudioIn Receives an audio signal from a Jack port. Description Receives an audio signal from a Jack audio input port inside this instance of Csound, which in turn has received the signal from its connected external Jack audio output port. Syntax asignal JackoAudioIn ScsoundPortName Initialization ScsoundPortName -- The short name ("portname") of the internal Jack audio input port. Performance asignal -- Audio received from the external Jack output port to which ScsoundPortName is connected. See Also JackoInfo, JackoInfo, JackoFreewheel, JackoAudioOutConnect, JackoAudioOutConnect, JackoMidiInConnect, JackoMidiOutConnect, JackoOn, JackoAudioOut, JackoMidiOut, JackoNoteOut, JackoTransport. Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/slider8.xml0000644000000000000000000001125112262561504016475 0ustar rootroot slider8 Real-time MIDI:Slider Banks slider8 slider8 Creates a bank of 8 different MIDI control message numbers. Description Creates a bank of 8 different MIDI control message numbers. Syntax i1,...,i8 slider8 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum8, imin8, imax8, init8, ifn8 k1,...,k8 slider8 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum8, imin8, imax8, init8, ifn8 Initialization i1 ... i8 -- output values ichan -- MIDI channel (1-16) ictlnum1 ... ictlnum8 -- MIDI control number (0-127) imin1 ... imin8 -- minimum values for each controller imax1 ... imax8 -- maximum values for each controller init1 ... init8 -- initial value for each controller ifn1 ... ifn8 -- function table for conversion for each controller Performance k1 ... k8 -- output values slider8 is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider8 allows a bank of 8 different MIDI control message numbers. As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. In the i-rate version of slider8, there is not an initial value input argument, because the output is gotten directly from current status of internal controller array of Csound. See Also s16b14, s32b14, slider16, slider16f, slider32, slider32f, slider64, slider64f, slider8f Credits Author: &namegabriel; Italy December 1998 New in Csound version 3.50 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/flrun.xml0000644000000000000000000000267112262561504016257 0ustar rootroot FLrun FLTK:Other FLrun FLrun Starts the FLTK widget thread. Description Starts the FLTK widget thread. Syntax FLrun Performance This opcode must be located at the end of all widget declarations. It has no arguments, and its purpose is to start the thread related to widgets. Widgets would not operate if FLrun is missing. See Also FLgetsnap, FLloadsnap, FLsavesnap, FLsetsnap, FLupdate Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/fouti.xml0000644000000000000000000001102012262561504016243 0ustar rootroot fouti Signal I/O:File I/O fouti fouti Outputs i-rate signals of an arbitrary number of channels to a specified file. Description fouti output N i-rate signals to a specified file of N channels. Syntax fouti ihandle, iformat, iflag, iout1 [, iout2, iout3,....,ioutN] Initialization ihandle -- a number which specifies this file. iformat -- a flag to choose output file format: 0 - floating point in text format 1 - 32-bit floating point in binary format iflag -- choose the mode of writing to the ASCII file (valid only in ASCII mode; in binary mode iflag has no meaning, but it must be present anyway). iflag can be a value chosen among the following: 0 - line of text without instrument prefix 1 - line of text with instrument prefix (see below) 2 - reset the time of instrument prefixes to zero (to be used only in some particular cases. See below) iout,..., ioutN -- values to be written to the file Performance fouti and foutir write i-rate values to a file. The main use of these opcodes is to generate a score file during a realtime session. For this purpose, the user should set iformat to 0 (text file output) and iflag to 1, which enable the output of a prefix consisting of the strings inum, actiontime, and duration, before the values of iout1...ioutN arguments. The arguments in the prefix refer to instrument number, action time and duration of current note. Notice that fout and foutk can use either a string containing a file pathname, or a handle-number generated by fiopen. Whereas, with fouti and foutir, the target file can be only specified by means of a handle-number. Examples Here is an example of the fouti opcode. It uses the file fouti.csd. Example of the fouti opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also fiopen, fout, foutir, foutk Credits Author: &namegabriel; Italy 1999 New in Csound version 3.56 manual_src~dfsg/opcodes/vosim.xml0000644000000000000000000002146512262561504016270 0ustar rootroot vosim Signal Generators:Granular Synthesis vosim vosim Simple vocal simulation based on glottal pulses with formant characteristics. Description This opcode produces a simple vocal simulation based on glottal pulses with formant characteristics. Output is a series of sound events, where each event is composed of a burst of squared sine pulses followed by silence. The VOSIM (VOcal SIMulation) synthesis method was developed by Kaegi and Tempelaars in the 1970's. Syntax ar vosim kamp, kFund, kForm, kDecay, kPulseCount, kPulseFactor, ifn [, iskip] Intialization ifn - a sound table, normally containing half a period of a sinewave, squared (see notes below). iskip - (optional) Skip initialization, for tied notes. Performance ar - output signal. Note that the output is usually unipolar - positive only. kamp - output amplitude, the peak amplitude of the first pulse in each burst. kFund - fundamental pitch, in Herz. Each event is 1/kFund seconds long. kForm - formant center frequency. Length of each pulse in the burst is 1/kForm seconds. kDecay - a dampening factor from pulse to pulse. This is subtracted from amplitude on each new pulse. kPulseCount - number of pulses in the burst part of each event. kPulseFactor - the pulse width is multiplied by this value at each new pulse. This results in formant sweeping. If factor is < 1.0, the formant sweeps up, if > 1.0 each new pulse is longer, so the formant sweeps down. The final pitch of the formant is kForm * pow(kPulseFactor, kPulseCount) The output of vosim is a series of sound events, where each each event is composed of a burst of squared sine pulses followed by silence. The total duration of the events determines fundamental frequency. The length of each single pulse in the squared-sine bursts produce a formant frequency band. The width of the formant is determined by rate of silence to pulses (see below). The final result is also shaped by the dampening factor from pulse to pulse. A small practical problem in using this opcode is that no GEN function will create a squared sine wave out of the box. Something like the following can be used to create the appropriate table from the score. ; use GEN09 to create half a sine in table 17 f 17 time size 9 0.5 1 0 ; run instr 101 on table 17 for a single init-pass i 101 0 0 17 It can also be done with an instrument writing to an f-table in the orchestra: ; square each point in table #p4. This should be run as init-only, just once in the performance. instr 101 index tableng p4 index = index - 1 ; start from last point loop: ival table index, p4 ival = ival * ival tableiw ival, index, p4 index = index - 1 if index < 0 igoto endloop igoto loop endloop: endin Parameter LimitsThe count of pulses multiplied by pulse width should fit in the event length (1/kFund). If this is not fulfilled, the algorithm does not break, we just do not start any pulses that would outlast the event. This might introduce a silence at end of event even if none was intended. In consequence, kForm should be higher than kFund, otherwise only silence is output. Vosim was created to emulate voice sounds using a model of glottal pulse. Rich sounds can be created by combining several instances of vosim with different parameters. One drawback is that the signal is not band-limited. But as the authors point out, attenuation of high-pitch components is -60 dB at 6 times the fundamental frequency. The signal can also be changed by changing the source signal in the lookup table. The technique has historical interest, and can produce rich sound very cheaply (each sample requires only a table lookup and a single multiplication for attenuation). As stated, formant bandwidth depends on the ratio between pulse burst and silence in an event. But this is not an independent parameter: The fundamental decides event length, and formant center defines the pulse length. It is therefore impossible to guarantee a specific burst/silence ratio, since the burst length has to be an integer multiple of pulse length. The decay of pulses can be used to smooth the transition from N to N+/-1 pulses, but there will still be steps in the spectral profile of output. The example code below shows one approach to this. All input parameters are k-rate. The input parameters are only used to set up each new event (or grain). Event amplitude is fixed for each event at initialization. In normal parameter ranges, when ksmps <500, the k-rate parameters are updated more often than events are created. In any case, no wide-band noise will be injected in the system due to k-rate inputs being updated less often than they are read, but some other artefacts could be created. The opcode should behave reasonably in the face of all user inputs. Some details: kFund < 0: This is forced to positive - no point in "reversed" events. kFund == 0: This leads to "infinite" length event, ie a pulse burst followed by very long indefinite silence. kForm == 0: This leads to infinite length pulse, so no pulses are generated (i.e. silence). kForm < 0: Table is read backward. If table is symmetric, kform and -kform should give bit-identical outputs. kPulseFactor == 0: Second pulse onwards is zero. See (c). kPulseFactor < 0: Pulses alternately read table forward and reversed. With asymmetric pulse table there may be some use for negative kForm or negative kPulseFactor. Examples Here is an example of the vosim opcode. It uses the file vosim.csd. Example of the vosim opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also fof, fof2 Credits Author: &namerasmus; March 2008 manual_src~dfsg/opcodes/opi.xml0000644000000000000000000000405512262561504015716 0ustar rootroot i Signal Modifiers:Sample Level Operators i i Returns an init-type equivalent of a k-rate argument, or directly returns an i-rate argument. Description Returns an init-type equivalent of a k-rate argument, or directly returns an i-rate argument. Syntax i(x) (control-rate or init-rate arg) Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Note Using i() with a k-rate expression argument is not recommended, and can produce unexpected results. See Also a, k, abs, exp, frac, int, log, log10, sqrt More information on this opcode: http://www.csounds.com/journal/issue10/CsoundRates.html , written by Andrés Cabrera Credits i-rate arg version of function new since 5.14 manual_src~dfsg/opcodes/flupdate.xml0000644000000000000000000000120612262561504016726 0ustar rootroot FLupdate FLTK:Other FLupdate FLupdate Same as the FLrun opcode. Description Same as the FLrun opcode. Syntax FLupdate manual_src~dfsg/opcodes/cggoto.xml0000644000000000000000000000523112262561504016406 0ustar rootroot cggoto Instrument Control:Program Flow Control cggoto cggoto Conditionally transfer control on every pass. Description Transfer control to label on every pass. (Combination of cigoto and ckgoto) Syntax cggoto condition, label where label is in the same instrument block and is not an expression, and where condition uses one of the Relational operators (<, =, <=, ==, !=) (and = for convenience, see also under Conditional Values). Examples Here is an example of the cggoto opcode. It uses the file cggoto.csd. Example of the cggoto opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cigoto, ckgoto, cngoto, if, igoto, kgoto, tigoto, timout Credits Example written by &namekevin;. manual_src~dfsg/opcodes/imagefree.xml0000644000000000000000000000404312262561504017050 0ustar rootroot imagefree Image Processing Opcodes imagefree imagecreate Frees memory allocated for a previously loaded or created image. Description Frees memory allocated for a previously loaded or created image. Syntax imagefree iimagenum Initialization iimagenum -- reference of the image to free. Examples Here is an example of the imagefree opcode. It uses the file imageopcodes.csd. Example of the imagefree opcode. See Also imageload, imagecreate, imagesize, imagesave, imagegetpixel, imagesetpixel Credits Author: Cesare Marilungo New in version 5.08 manual_src~dfsg/opcodes/streson.xml0000644000000000000000000000643512262561504016630 0ustar rootroot streson Signal Generators:Waveguide Physical Modeling streson streson A string resonator with variable fundamental frequency. Description An audio signal is modified by a string resonator with variable fundamental frequency. Syntax ares streson asig, kfr, ifdbgain Initialization ifdbgain -- feedback gain, between 0 and 1, of the internal delay line. A value close to 1 creates a slower decay and a more pronounced resonance. Small values may leave the input signal unaffected. Depending on the filter frequency, typical values are > .9. Performance asig -- the input audio signal. kfr -- the fundamental frequency of the string. streson passes the input asig through a network composed of comb, low-pass and all-pass filters, similar to the one used in some versions of the Karplus-Strong algorithm, creating a string resonator effect. The fundamental frequency of the string is controlled by the k-rate variable kfr.This opcode can be used to simulate sympathetic resonances to an input signal. See Modal Frequency Ratios for frequency ratios of real intruments which can be used to determine the values of kfrq. streson is an adaptation of the StringFlt object of the SndObj Sound Object Library developed by the author. Examples Here is an example of the streson opcode. It uses the file streson.csd. Example of the streson opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; Music Department National University of Ireland, Maynooth Maynooth, Co. Kildare 1998 New in Csound version 3.494 manual_src~dfsg/opcodes/s32b14.xml0000644000000000000000000001045412262561504016045 0ustar rootroot s32b14 Real-time MIDI:Slider Banks s32b14 s32b14 Creates a bank of 32 different 14-bit MIDI control message numbers. Description Creates a bank of 32 different 14-bit MIDI control message numbers. Syntax i1,...,i32 s32b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, \ initvalue1, ifn1,..., ictlno_msb32, ictlno_lsb32, imin32, imax32, initvalue32, ifn32 k1,...,k32 s32b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, \ initvalue1, ifn1,..., ictlno_msb32, ictlno_lsb32, imin32, imax32, initvalue32, ifn32 Initialization i1 ... i64 -- output values ichan -- MIDI channel (1-16) ictlno_msb1 .... ictlno_msb32 -- MIDI control number, most significant byte (0-127) ictlno_lsb1 .... ictlno_lsb32 -- MIDI control number, least significant byte (0-127) imin1 ... imin64 -- minimum values for each controller imax1 ... imax64 -- maximum values for each controller init1 ... init64 -- initial value for each controller ifn1 ... ifn64 -- function table for conversion for each controller Performance k1 ... k64 -- output values s32b14 is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. s32b14 allows a bank of 32 different MIDI control message numbers. It uses 14-bit values instead of MIDI's normal 7-bit values. As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. In the i-rate version of s32b14, there is not an initial value input argument. The output is taken directly from the current status of internal controller array of Csound. Credits Author: &namegabriel; Italy December 1998 New in Csound version 3.50 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/partikkelsync.xml0000644000000000000000000000664112262561504020015 0ustar rootroot partikkelsync Signal Generators:Granular Synthesis partikkelsync partikkelsync Outputs partikkel's grain scheduler clock pulse and phase to synchronize several instances of the partikkel opcode to the same clock source. Description partikkelsync is an opcode for outputting partikkel's grain scheduler clock pulse and phase. partikkelsync's output can be used to synchronize other instances of the partikkel opcode to the same clock. Syntax async [,aphase] partikkelsync iopcode_id Initialization iopcode_id -- the opcode id, linking an instance of partikkel to an instance of partikkelsync. Performance async -- trigger pulse signal. Outputs trigger pulses synchronized to a partikkel opcode's grain scheduler clock. One trigger pulse is generated for each grain started in the partikkel opcode with the same opcode_id. The normal usage would be to send this signal to another partikkel opcode's async input to synchronize several instances of partikkel. aphase -- clock phase. Outputs a linear ramping phase signal. Can be used e.g. for softsynchronization, or just as a phase generator ala phasor. Example Here is an example of the partikkelsync opcodes. It uses the file partikkelsync.csd. Example with soft sync of two partikkel generators. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also partikkel Credits Author: Thom Johansen Author: Torgeir Strand Henriksen Author: &nameoeyvind; April 2007 New in version 5.06 manual_src~dfsg/opcodes/include.xml0000644000000000000000000001027712262561504016555 0ustar rootroot multiple file orchestras #includeorchestra Orchestra Syntax:Macros #include #include Includes an external file for processing. Description Includes an external file for processing. Syntax #include filename Performance It is sometimes convenient to have the orchestra arranged in a number of files, for example with each instrument in a separate file. This style is supported by the #include facility which is part of the macro system. A line containing the text #include "filename" where the character " can be replaced by any suitable character. For most uses the double quote symbol will probably be the most convenient. The file name can include a full path. This takes input from the named file until it ends, when input reverts to the previous input. Note: Csound versions prior to 4.19 had a limit of 20 on the depth of included files and macros. Another suggested use of #include would be to define a set of macros which are part of the composer's style. An extreme form would be to have each instrument defines as a macro, with the instrument number as a parameter. Then an entire orchestra could be constructed from a number of #include statements followed by macro calls. #include "clarinet" #include "flute" #include "bassoon" $CLARINET(1) $FLUTE(2) $BASSOON(3) It must be stressed that these changes are at the textual level and so take no cognizance of any meaning. If the version of Csound is built with the CURL library the filename in an #include statement can be an URL, recognised by including the substring "://" in the name. This will include text via protocols such as http, https, and ftp. Examples Here is an example of the include opcode. It uses the file include.csd, and table1.inc. Example of the include opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April 1998 Example written by &namekevin;. New in Csound version 3.48 URLs available from Csound 6.02 manual_src~dfsg/opcodes/goto.xml0000644000000000000000000000531712262561504016101 0ustar rootroot goto Instrument Control:Program Flow Control goto goto Transfer control on every pass. Description Transfer control to label on every pass. (Combination of igoto and kgoto) Syntax goto label where label is in the same instrument block and is not an expression. Using goto not as part of an if statement (as in: goto end), will cause initialization to be skipped on all the code the goto jumps over. In performance, leaving some opcodes uninitialized will cause deletion of the note/event. In these cases, using kgoto (as in: kgoto end) might be preferred." Examples Here is an example of the goto opcode. It uses the file goto.csd. Example of the goto opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cggoto, cigoto, ckgoto, if, igoto, kgoto, tigoto, timout Credits Example written by &namekevin;. Added a note by Jim Aikin. manual_src~dfsg/opcodes/divides.xml0000644000000000000000000001000212262561504016543 0ustar rootroot / Mathematical Operations:Arithmetic and Logic Operations / / Division operator. Description Arithmetic operators perform operations of change-sign (negate), don't-change-sign, logical AND logical OR, add, subtract, multiply and divide. Note that a value or an expression may fall between two of these operators, either of which could take it as its left or right argument, as in a + b * c. In such cases three rules apply: 1. * and / bind to their neighbors more strongly than + and −. Thus the above expression is taken as a + (b * c) with * taking b and c and then + taking a and b * c. 2. + and − bind more strongly than &&, which in turn is stronger than ||: a && b - c || d is taken as (a && (b - c)) || d 3. When both operators bind equally strongly, the operations are done left to right: a - b - c is taken as (a - b) - c Parentheses may be used as above to force particular groupings. Syntax a / b (no rate restriction) where the arguments a and b may be further expressions. Arguments The arguments of / can be scalar values or k-rate one dimensional arrays (vectors), or any combination. If one of the arguments is an array, so is the value. Examples Here is an example of the / operator. It uses the file divides.csd. Example of the / operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: retrigger rate per note duration = 3.333333 retrigger rate per note duration = 5.000000 retrigger rate per note duration = 0.666667 See Also -, +, &&, ||, *, ˆ, % Credits Arithmetic on vectors is new in version 6.00 manual_src~dfsg/opcodes/jacktransport.xml0000644000000000000000000000371112262561504020012 0ustar rootroot jacktransport Instrument Control:Realtime Performance Control jacktransport jacktransport Start/stop jack_transport and can optionally relocate the playback head. Description Start/stop jack_transport and can optionally relocate the playback head. Syntax jacktransport icommand [, ilocation] Initialization icommand -- 1 to start playing, 0 to stop. ilocation -- optional location in seconds to specify where the playback head should be moved. If omitted, the transport is started from current location. Since jacktransport depends on jack audio connection kit, it will work only on Linux or Mac OS X systems which have the jack server running. Examples Here is a simple example of the jacktransport opcode. It uses the file jacktransport.csd. Simple example of the jacktransport opcode. Credits Author: Cesare Marilungo New in version 5.08 manual_src~dfsg/opcodes/strindexk.xml0000644000000000000000000000347212262561504017144 0ustar rootroot strindexk Strings:Manipulation strindexk strindexk Return the position of the first occurence of a string in another string Description Return the position of the first occurence of S2 in S1, or -1 if not found. If S2 is empty, 0 is returned. strindexk runs both at init and performance time. Syntax kpos strindexk S1, S2 Examples Here is an example of the strindexk opcode. It uses the file strindexk.csd. Example of the strindexk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also strindex Credits Author: Istvan Varga 2006 New in version 5.02 manual_src~dfsg/opcodes/loop_lt.xml0000644000000000000000000000741712262561504016604 0ustar rootroot loop_lt Instrument Control:Program Flow Control loop_lt loop_lt Looping constructions. Description Construction of looping operations. Syntax loop_lt indx, incr, imax, label loop_lt kndx, kncr, kmax, label Initialization indx -- i-rate variable to count loop. incr -- value to increment the loop. imax -- maximum value of loop index. Performance kndx -- k-rate variable to count loop. kncr -- value to increment the loop. kmax -- maximum value of loop index. The actions of loop_lt are equivalent to the code indx = indx + incr if (indx < imax) igoto label or kndx = kndx + kncr if (kndx < kmax) kgoto label Examples Here is an example of the loop_lt opcode. It uses the file loop_lt.csd. Example of the loop_lt opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: play instance 0 play instance 1 play instance 2 play instance 3 play instance 4 See Also loop_ge, loop_gt and loop_le. More information on this opcode: http://www.csounds.com/journal/2006summer/controlFlow_part2.html , written by Steven Yi. And in the Floss Manuals: http://en.flossmanuals.net/csound/ch018_c-control-structures Credits &nameistvan;. New in Csound version 5.01 manual_src~dfsg/opcodes/hvs1.xml0000644000000000000000000002743712262561504016021 0ustar rootroot hvs1 Signal Generators:Hyper Vectorial Synthesis hvs1 hvs1 Allows one-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables. Description hvs1 allows one-dimensional Hyper Vectorial Synthesis (HVS) controlled by externally-updated k-variables. Syntax hvs1 kx, inumParms, inumPointsX, iOutTab, iPositionsTab, iSnapTab [, iConfigTab] Initialization inumParms - number of parameters controlled by the HVS. Each HVS snapshot is made up of inumParms elements. inumPointsX - number of points that each dimension of the HVS cube (or square in case of two-dimensional HVS; or line in case of one-dimensional HVS) is made up. iOutTab - number of the table receiving the set of output-parameter instant values of the HVS. The total amount of parameters is defined by the inumParms argument. iPositionsTab – a table filled with the individual positions of snapshots in the HVS matrix (see below for more information). iSnapTab – a table filled with all the snapshots. Each snapshot is made up of a set of parameter values. The amount of elements contained in each snapshots is specified by the inumParms argument. The set of elements of each snapshot follows (and is adjacent) to the previous one in this table. So the total size of this table should be >= to inumParms multiplied the number of snapshots you intend to store for the HVS. iConfigTab – (optional) a table containing the behavior of the HVS for each parameter. If the value of iConfigTab is zero (default), this argument is ignored, meaning that each parameter is treated with linear interpolation by the HVS. If iConfigTab is different than zero, then it must refer to an existing table whose contents are in its turn referring to a particolar kind of interpolation. In this table, a value of -1 indicates that corresponding parameter is leaved unchanged (ignored) by the HVS; a value of zero indicates that corresponding parameter is treated with linear-interpolation; each other values must be integer numbers indicating an existing table filled with a shape which will determine the kind of special interpolation to be used (table-based interpolation). Performance kx - these are externally-modified variables which controls the motion of the pointer in the HVS matrix cube (or square or line in case of HVS matrices made up of less than 3 dimensions). The range of these input arguments must be 0 to 1. Hyper Vectorial Synthesis is a technique that allows control of a huge set of parameters by using a simple and global approach. The key concepts of the HVS are: The set of HVS parameters, whose amount is fixed and defined by the inumParms argument. During the HVS performance, all these parameters are variant and can be applied to any sound synthesis technique, as well as to any global control for algorithmic composition and any other kind of level. The user must previously define several sets of fixed values for each HVS parameter, each set corresponding to a determinate synthesis configuration. Each set of values is called snapshot, and can be considered as the coordinates of a bound of a multi-dimensional space. The HVS consists on moving a point in this multi-dimensional space (by using a special motion pointer, see below), according and inside the bounds defined by the snapshots. You can fix any amount of HVS parameters (each parameter being a dimension of the multi-dimensional space), even a huge number, the limit only depends on the processing power (and the memory) of your computer and on the complexity of the sound-synthesis you will use. The HVS cube (or square or line). This is the matrix (of 3, 2 or 1 dimensions, according to the hvs opcode you intend to use) of “mainstays” (or pivot) points of HVS. The total amount of pivot-points depends on the value of the inumPointsX, inumPointsY and inumPointsZ arguments. In the case of a 3-dimensional HVS matrix you can define, for instance, 3 points for the X dimension, 5 for the Y dimension and 2 for the Z dimension. In this case, the total number of pivot-points is 3 * 5 * 2 = 30. With this set of pivot points, the cube Is divided into smaller cubed zones each one bounded by eight nearby points. Each point is numbered. The numeral order of these points is enstabilished in the following way: number zero is the first point, number 1 the second and so on. Assuming you are using a 3-dimensional HVS cube having the number of points above mentioned (i.e. 3, 5 and 2 respectively for the X, Y and Z axis), the first point (point zero) is the upper-left-front vertex of the cube, by facing the XY plane of the cube. The second point is the middle point of the upper front edge of the cube and so on. You can refer to the figure below in order to understand how the numeral order of the pivot-points proceeds: For the 2-dimensional HVS, it is the same, by only omitting the rear cube face, so each zone is bounded by 4 pivot-points instead of 8. For the 1-dimensional HVS, the whole thing is even simpler because it is a line with the pivot-points proceeding from left to right. Each point is coupled with a snapshot. Snapshot order, as stored into the iSnapTab, can or cannot follow the order of the pivot-points numbers. In fact it is possible to alter this order by means the iPositionsTab, a table that remaps the position of each snapshot in relation to the pivot points. The iPositionsTab is made up of the positions of the snapshots (contained in the iSnapTab) in the two-dimensional grid. Each subsequent element is actually a pointer representing the position in the iSnapTab. For example, in a 2-dimensional HVS matrix such as the following (in this case having inumPointsX = 3 and inumPointsY = 5: 571 349 620 413 827
These numbers (to be stored in the iSnapTab table by using, for instance, the GEN02 function generator) represents the snapshot position within the grid (in this case a 3x5 matrix). So, the first element 5, has index zero and represents the 6th (element zero is the first) snapshot contained in the iSnapTab, the second element 7 represents the 8th element of iSnapTab and so on. Summing up, the vertices of each zone (a cubed zone is delimited by 8 vertices; a squared zone by 4 vertices and a linear zone by 2 points) are coupled with a determinate snapshot, whose number is remapped by the iSnapTab. Output values of the HVS are influenced by the motion pointer, a point whose position, in the HVS cube (or square or segment) is determined by the kx, ky and kz arguments. The values of these arguments, which must be in the 0 to 1 range, are externally set by the user. The output values, whose amount is equal to the inumParms argument, are stored in the iOutTab, a table that must be already allocated by the user, and must be at least inumParms size. In what way the motion pointer influences the output? Well, when the motion pointer falls in a determinate cubed zone, delimited, for instance, by 8 vertices (or pivot points), we assume that each vertex has associated a different snapshot (i.e. a set of inumParms values), well, the output will be the weighted average value of the 8 vertices, calculated according on the distance of the motion pointer from each of the 8 vertices. In the case of a default behavior, when the iConfigTab argument is not set, the exact output is calculated by using linear interpolation which is applied to each different parameter of the HVS. Anyway, it is possible to influence this behavior by setting the iConfigTab argument to a number of a table whose contents can affect one or more HVS parameters. The iConfigTab table elements are associated to each HVS parameter and their values affect the HVS output in the following way: If iConfigTab is equal to -1, corresponding output is skipped, i.e. the element is not calculated, leaving corresponding element value in the iOutTab unchanged; If iConfigTab is equal to zero, then the normal HVS output is calculated (by using weighted average of the nearest vertex of current zone where it falls the motion pointer); If iConfigTab element is equal to an integer number > zero, then the contents of a table having that number is used as a shape of a table-based interpolation. Examples Here is an example of the hvs1 opcode. It uses the file hvs1.csd. Example of the hvs1 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also hvs2, hvs3, vphaseseg Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/integ.xml0000644000000000000000000000550312262561504016234 0ustar rootroot integ Signal Modifiers:Sample Level Operators integ integ Modify a signal by integration. Description Modify a signal by integration. Syntax ares integ asig [, iskip] kres integ ksig [, iskip] Initialization iskip (optional) -- initial disposition of internal save space (see reson). The default value is 0. Performance integ and diff perform integration and differentiation on an input control signal or audio signal. Each is the converse of the other, and applying both will reconstruct the original signal. Since these units are special cases of low-pass and high-pass filters, they produce a scaled (and phase shifted) output that is frequency-dependent. Thus diff of a sine produces a cosine, with amplitude 2 * pi * Hz / sr that of the original (for each component partial); integ will inversely affect the magnitudes of its component inputs. With this understanding, these units can provide useful signal modification. Examples Here is an example of the integ opcode. It uses the file integ.csd. Example of the integ opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also diff, downsamp, interp, samphold, upsamp manual_src~dfsg/opcodes/strtod.xml0000644000000000000000000000435412262561504016450 0ustar rootroot strtod Strings:Conversion strtod strtod Converts a string to a float (i-rate). Description Convert a string to a floating point value. It is also possible to pass an strset index or a string p-field from the score instead of a string argument. If the string cannot be parsed as a floating point or integer number, an init or perf error occurs and the instrument is deactivated. Syntax ir strtod Sstr ir strtod indx Initialization Sstr -- String to convert. indx -- index of string set by strset Performance ir -- Value of string as float. Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/changed.xml0000644000000000000000000000500112262561504016510 0ustar rootroot changed Instrument Control:Sensing and Control changed changed k-rate signal change detector. Description This opcode outputs a trigger signal that informs when any one of its k-rate arguments has changed. Useful with valuator widgets or MIDI controllers. Syntax ktrig changed kvar1 [, kvar2,..., kvarN] Performance ktrig - Outputs a value of 1 when any of the k-rate signals has changed, otherwise outputs 0. kvar1 [, kvar2,..., kvarN] - k-rate variables to watch for changes. Examples Here is an example of the changed opcode. It uses the file changed.csd. Example of the changed opcode. Its output should include lines like: i 1 time 0.00267: 0.00000 i1 0.00000 i1 1.00000 i1 0.00000 i 1 time 0.20267: 1.00000 i 1 time 0.40267: 1.00000 i1 1.00000 i1 0.00000 i 1 time 0.60267: 1.00000 i 1 time 0.80267: 1.00000 i1 1.00000 i1 0.00000 i 1 time 1.00000: 0.00000 i1 1.00000 i1 0.00000 i 1 time 1.20267: -1.00000 i 1 time 1.40267: -1.00000 i1 1.00000 i1 0.00000 i 1 time 1.60267: -1.00000 i 1 time 1.80000: -1.00000 i1 1.00000 i1 0.00000 i 1 time 2.00000: -0.00000 i1 1.00000 i1 0.00000 ......... Credits Written by &namegabriel;. Example written by &nameandres;. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/cigoto.xml0000644000000000000000000000546012262561504016414 0ustar rootroot cigoto Instrument Control:Program Flow Control cigoto cigoto Conditionally transfer control during the i-time pass. Description During the i-time pass only, conditionally transfer control to the statement labeled by label. Syntax cigoto condition, label where label is in the same instrument block and is not an expression, and where condition uses one of the Relational operators (<, =, <=, ==, !=) (and = for convenience, see also under Conditional Values). Examples Here is an example of the cigoto opcode. It uses the file cigoto.csd. Example of the cigoto opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: instr 1: iparam = 0.000 instr 1: ifreq = 440.000 instr 1: iparam = 1.000 instr 1: ifreq = 880.000 See Also cggoto, ckgoto, cngoto, goto, if, kgoto, rigoto, tigoto, timout Credits Example written by &namekevin;. manual_src~dfsg/opcodes/noteoff.xml0000644000000000000000000000510512262561504016564 0ustar rootroot noteoff Real-time MIDI:Note Output noteoff noteoff Send a noteoff message to the MIDI OUT port. Description Send a noteoff message to the MIDI OUT port. Syntax noteoff ichn, inum, ivel Initialization ichn -- MIDI channel number (1-16) inum -- note number (0-127) ivel -- velocity (0-127) Performance noteon (i-rate note on) and noteoff (i-rate note off) are the simplest MIDI OUT opcodes. noteon sends a MIDI noteon message to MIDI OUT port, and noteoff sends a noteoff message. A noteon opcode must always be followed by an noteoff with the same channel and number inside the same instrument, otherwise the note will play endlessly. These noteon and noteoff opcodes are useful only when introducing a timout statement to play a non-zero duration MIDI note. For most purposes, it is better to use noteondur and noteondur2. See Also noteon, noteondur, noteondur2 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/dumpk2.xml0000644000000000000000000001112712262561504016327 0ustar rootroot dumpk2 Signal I/O:File I/O dumpk2 dumpk2 Periodically writes two orchestra control-signal values to an external file. Description Periodically writes two orchestra control-signal values to a named external file in a specific format. Syntax dumpk2 ksig1, ksig2, ifilname, iformat, iprd Initialization ifilname -- character string (in double quotes, spaces permitted) denoting the external file name. May either be a full path name with target directory specified or a simple filename to be created within the current directory iformat -- specifies the output data format: 1 = 8-bit signed char(high order 8 bits of a 16-bit integer 4 = 16-bit short integers 5 = 32-bit long integers 6 = 32-bit floats 7 = ASCII long integers 8 = ASCII floats (2 decimal places) Note that A-law and U-law output are not available, and that all formats except the last two are binary. The output file contains no header information. iprd -- the period of ksig output in seconds, rounded to the nearest orchestra control period. A value of 0 implies one control period (the enforced minimum), which will create an output file sampled at the orchestra control rate. Performance ksig1, ksig2 -- control-rate signals. This opcode allows two generated control signal values to be saved in a named external file. The file contains no self-defining header information. But it contains a regularly sampled time series, suitable for later input or analysis. There may be any number of dumpk2 opcodes in an instrument or orchestra but each must write to a different file. Examples Here is an example of the dumpk2 opcode. It uses the file dumpk2.csd. Example of the dumpk2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output should include lines like these: kfreq = 429.202551, kdb = -20.495694 kfreq = 429.202551, kdb = -20.495694 kfreq = 407.275258, kdb = -23.123776 kfreq = 475.264472, kdb = -9.300846 kfreq = 569.979181, kdb = -7.315527 kfreq = 440.103457, kdb = -0.058331 kfreq = 429.202600, kdb = -20.495700 kfreq = 429.202600, kdb = -20.495700 kfreq = 407.275300, kdb = -23.123800 kfreq = 475.264500, kdb = -9.300800 kfreq = 569.979200, kdb = -7.315500 kfreq = 440.103500, kdb = -0.058300 See Also dumpk, dumpk3, dumpk4, readk, readk2, readk3, readk4 Credits By: John ffitch and Barry Vercoe 1999 or earlier manual_src~dfsg/opcodes/fareylen.xml0000644000000000000000000000762012262561504016735 0ustar rootroot fareylen Miscellaneous fareylen fareylen returns the length of a Farey Sequence. Description This opcode can be used in conjunction with GENfarey. It calculates the length of Farey Sequence Fn. Its length is given by: |Fn| = 1 + SUM over n phi(m) where phi(m) is Euler's totient function, which gives the number of integers ≤ m that are coprime to m. Some values for the length of Fn given n: n Fn 1 2 2 3 3 5 4 7 5 11 6 13 7 19 8 23 9 29 10 33 11 43 12 47 13 59 14 65 15 73 16 81 17 97 18 103 19 121 20 129 Syntax kfl fareylen kfn Performance The length of the identified Farey sequence is returned. kfn -- Integer identifying the sequence. Credits Author: Georg Boenn University of Glamorgan, UK New in Csound version 5.13 manual_src~dfsg/opcodes/hrtfer.xml0000644000000000000000000001013612262561504016416 0ustar rootroot hrtfer Signal Modifiers:Panning and Spatialization hrtfer hrtfer Creates 3D audio for two speakers. Description Output is binaural (headphone) 3D audio. Syntax aleft, aright hrtfer asig, kaz, kelev, HRTFcompact Initialization kAz -- azimuth value in degrees. Positive values represent position on the right, negative values are positions on the left. kElev -- elevation value in degrees. Positive values represent position above horizontal, negative values are positions under horizontal. At present, the only file which can be used with hrtfer is HRTFcompact. It must be passed to the opcode as the last argument within quotes as shown above. HRTFcompact may also be obtained via anonymous ftp from: ftp://ftp.cs.bath.ac.uk/pub/dream/utilities/Analysis/HRTFcompact Performance These unit generators place a mono input signal in a virtual 3D space around the listener by convolving the input with the appropriate HRTF data specified by the opcode's azimuth and elevation values. hrtfer allows these values to be k-values, allowing for dynamic spatialization. hrtfer can only place the input at the requested position because the HRTF is loaded in at i-time (remember that currently, CSound has a limit of 20 files it can hold in memory, otherwise it causes a segmentation fault). The output will need to be scaled either by using balance or by multiplying the output by some scaling constant. Note The sampling rate of the orchestra must be 44.1kHz. This is because 44.1kHz is the sampling rate at which the HRTFs were measured. In order to be used at a different rate, the HRTFs would need to be re-sampled at the desired rate. Examples Here is an example of the hrtfer opcode. It uses the file hrtfer.csd, HRTFcompact, and beats.wav. Example of the hrtfer opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also hrtfmove, hrtfmove2, hrtfstat. Credits Authors: Eli Breder and David MacIntyre Montreal 1996 Fixed the example thanks to a message from &nameistvan;. manual_src~dfsg/opcodes/lpinterp.xml0000644000000000000000000000617712262561504016773 0ustar rootroot lpinterp Spectral Processing:LPC lpinterp lpslot lpinterp Computes a new set of poles from the interpolation between two analysis. Description Computes a new set of poles from the interpolation between two analysis. Syntax lpinterp islot1, islot2, kmix Initialization islot1 -- slot where the first analysis was stored islot2 -- slot where the second analysis was stored kmix -- mix value between the two analysis. Should be between 0 and 1. 0 means analysis 1 only. 1 means analysis 2 only. Any value in between will produce interpolation between the filters. lpinterp computes a new set of poles from the interpolation between two analysis. The poles will be stored in the current lpslot and used by the next lpreson opcode. Examples Here is a typical orc using the opcodes: ipower init 50000 ; Define sound generator ifreq init 440 asrc buzz ipower, ifreq, 10, 1 ktime line 0, p3, p3 ; Define time lin lpslot 0 ; Read square data poles krmsr, krmso, kerr, kcps lpread ktime,"square.pol" lpslot 1 ; Read triangle data poles krmsr, krmso, kerr, kcps lpread ktime,"triangle.pol" kmix line 0, p3, 1 ; Compute result of mixing lpinterp 0, 1, kmix ; and balance power ares lpreson asrc aout balance ares, asrc out aout See Also lpslot Credits Author: &namegabriel; manual_src~dfsg/opcodes/loopsegp.xml0000644000000000000000000000542412262561504016760 0ustar rootroot loopsegp Signal Generators:Linear and Exponential Generators loopsegp loopsegp Control signals based on linear segments. Description Generate control signal consisiting of linear segments delimited by two or more specified points. The entire envelope can be looped at time-variant rate. Each segment coordinate can also be varied at k-rate. Syntax ksig loopsegp kphase, kvalue0, kdur0, kvalue1 \ [, kdur1, ... , kdurN-1, kvalueN] Performance ksig - output signal kphase - point of the sequence read, expressed as a fraction of a cycle (0 to 1) kvalue0 ...kvalueN - values of points kdur0 ...kdurN-1 - duration of points expessed in fraction of a cycle loopsegp opcode is similar to loopseg; the only difference is that, instead of frequency, a time-variant phase is required. If you use phasor to get the phase value, you will have a behaviour identical to loopseg, but interesting results can be achieved when using phases having non-linear motions, making loopsegp more powerful and general than loopseg. Examples Here is an example of the loopsegp opcode. It uses the file loopsegp.csd. Example of the loopsegp opcode. Credits Written by &namegabriel;. New in Csound 5. (Previously available only on CsoundAV) manual_src~dfsg/opcodes/readclock.xml0000644000000000000000000000570212262561504017056 0ustar rootroot readclock Instrument Control:Time Reading readclock readclock Reads the value of an internal clock. Description Reads the value of an internal clock. Syntax ir readclock inum Initialization inum -- the number of a clock. There are 32 clocks numbered 0 through 31. All other values are mapped to clock number 32. ir -- value at i-time, of the clock specified by inum Performance Between a clockon and a clockoff opcode, the CPU time used is accumulated in the clock. The precision is machine dependent but is the millisecond range on UNIX and Windows systems. The readclock opcde reads the current value of a clock at initialization time. Examples Here is an example of the readclock opcode. It uses the file readclock.csd. Example of the readclock opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: i1 = 0.000 instr 1: i1 = 90.000 instr 1: i1 = 180.000 See Also clockoff, clockon Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK July, 1999 Example written by &namekevin;. New in Csound version 3.56 manual_src~dfsg/opcodes/tonex.xml0000644000000000000000000000636312262561504016270 0ustar rootroot tonex Signal Modifiers:Standard Filters tonex tonex Emulates a stack of filters using the tone opcode. Description tonex is equivalent to a filter consisting of more layers of tone with the same arguments, serially connected. Using a stack of a larger number of filters allows a sharper cutoff. They are faster than using a larger number instances in a Csound orchestra of the old opcodes, because only one initialization and k- cycle are needed at time and the audio loop falls entirely inside the cache memory of processor. Syntax ares tonex asig, khp [, inumlayer] [, iskip] ares tonex asig, ahp [, inumlayer] [, iskip] Initialization inumlayer (optional) -- number of elements in the filter stack. Default value is 4. iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance asig -- input signal khp/ahp -- the response curve's half-power point. Half power is defined as peak power / root 2. Examples Here is an example of the tonex opcode. It uses the file tonex.csd. Example of the tonex opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also atonex, resonx Credits Author: &namegabriel; (adapted by &namejohn;) Italy New in Csound version 3.49 Audio rate parameters introduced in version 6.02 October 2013. manual_src~dfsg/opcodes/strupper.xml0000644000000000000000000000203712262561504017011 0ustar rootroot strupper Strings:Conversion strupper strupper Convert a string to upper case Description Convert Ssrc to upper case, and write the result to Sdst. strupper runs at init time only. Syntax Sdst strupper Ssrc See also strupperk Credits Author: Istvan Varga 2006 New in version 5.02 manual_src~dfsg/opcodes/slider64f.xml0000644000000000000000000001157112262561504016732 0ustar rootroot slider64f Real-time MIDI:Slider Banks slider64f slider64f Creates a bank of 64 different MIDI control message numbers, filtered before output. Description Creates a bank of 64 different MIDI control message numbers, filtered before output. Syntax k1,...,k64 slider64f ichan, ictlnum1, imin1, imax1, init1, ifn1, \ icutoff1,..., ictlnum64, imin64, imax64, init64, ifn64, icutoff64 Initialization ichan -- MIDI channel (1-16) ictlnum1 ... ictlnum64 -- MIDI control number (0-127) imin1 ... imin64 -- minimum values for each controller imax1 ... imax64 -- maximum values for each controller init1 ... init64 -- initial value for each controller ifn1 ... ifn64 -- function table for conversion for each controller icutoff1 ... icutoff64 -- low-pass filter cutoff frequency for each controller Performance k1 ... k64 -- output values slider64f is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider64f allows a bank of 64 different MIDI control message numbers. It filters the signal before output. This eliminates discontinuities due to the low resolution of the MIDI (7 bit). The cutoff frequency can be set separately for each controller (suggested range: .1 to 5 Hz). As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. slider64f opcodes do not output the required initial value immediately, but only after some k-cycles because the filter slightly delays the output. See Also s16b14, s32b14, slider16, slider16f, slider32, slider32f, slider64, slider8, slider8f Credits Author: &namegabriel; Italy December 1998 New in Csound version 3.50 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/faustctl.xml0000644000000000000000000000322612262561504016753 0ustar rootroot faustctl Faust Opcodes faustctl faustctl Adjusts a given control in a Faust DSP instance. Description Faustctl will set a given control in a running faust program Syntax faustctl idsp,Scontrol,kval Initialization Scontrol -- a string containing the control name idsp -- a handle to an existing Faust DSP instance Performance kval -- value to which the control will be set. Examples Here is an example of the faustctl opcode, with a simple program that adjusts the gain of an input: idsp,a1 faustgen {{ gain = hslider("vol",1,0,1,0.01); process = (_ * gain); }}, ain1 faustctl idsp, "vol", 0.5 Credits Author: &namevictor;, 2013 manual_src~dfsg/opcodes/initc7.xml0000644000000000000000000000655212262561504016330 0ustar rootroot initc7 Real-time MIDI:Input initc7 initc7 Initializes the controller used to create a 7-bit MIDI value. Description Initializes MIDI controller ictlno with ivalue Syntax initc7 ichan, ictlno, ivalue Initialization ichan -- MIDI channel (1-16) ictlno -- controller number (0-127) ivalue -- floating point value (must be within 0 to 1) Performance initc7 can be used together with both midic7 and ctrl7 opcodes for initializing the first controller's value. ivalue argument must be set with a number within 0 to 1. An error occurs if it is not. Use the following formula to set ivalue according with midic7 and ctrl7 min and max range: ivalue = (initial_value - min) / (max - min) Examples Here is an example of the initc7 opcode. It uses the file initc7.csd. Example of the initc7 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also ctrl7, ctrl14, ctrl21, ctrlinit, initc14, initc21, midic7, midic14, midic21 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/lorismorph.xml0000644000000000000000000001153412262561504017325 0ustar rootroot lorismorph Spectral Processing:Loris lorismorph lorismorph Morphs two stored sets of bandwidth-enhanced partials and stores a new set of partials representing the morphed sound. The morph is performed by linearly interpolating the parameter envelopes (frequency, amplitude, and bandwidth, or noisiness) of the bandwidth-enhanced partials according to control-rate frequency, amplitude, and bandwidth morphing functions. Syntax lorismorph isrcidx, itgtidx, istoreidx, kfreqmorphenv, kampmorphenv, kbwmorphenv Description lorismorph morphs two stored sets of bandwidth-enhanced partials and stores a new set of partials representing the morphed sound. The morph is performed by linearly interpolating the parameter envelopes (frequency, amplitude, and bandwidth, or noisiness) of the bandwidth-enhanced partials according to control-rate frequency, amplitude, and bandwidth morphing functions. Initialization istoreidx, ireadidx, isrcidx, itgtidx are labels that identify a stored set of bandwidth-enhanced partials. lorisread imports partials from a SDIF file and stores them with the integer label istoreidx. lorismorph morphs sets of partials labeled isrcidx and itgtidx, and stores the resulting partials with the integer label istoreidx. lorisplay renders the partials stored with the label ireadidx. The labels are used only at initialization time, and may be reused without any cost or benefit in efficiency, and without introducing any interaction between instruments or instances. Performance lorismorph generates a set of bandwidth-enhanced partials by morphing two stored sets of partials, the source and target partials, which may have been imported using lorisread, or generated by another unit generator, including another instance of lorismorph. The morph is performed by interpolating the parameters of corresponding (labeled) partials in the two source sounds. The sound morph is described by three control-rate morphing envelopes. kfreqmorphenv describes the interpolation of partial frequency values in the two source sounds. When kfreqmorphenv is 0, partial frequencies are obtained from the partials stored at isrcidx. When kfreqmorphenv is 1, partial frequencies are obtained from the partials at itgtidx. When kfreqmorphenv is between 0 and 1, the partial frequencies are interpolated between corresponding source and target partials. Interpolation of partial amplitudes and bandwidth (noisiness) coefficients are similarly described by kampmorphenv and kbwmorphenv. Examples Here is an example of the lorismorph opcode. It uses the file lorismorph.csd. Example of the lorismorph opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits This implementation of the Loris unit generators was written by Kelly Fitz (loris@cerlsoundgroup.org). It is patterned after a prototype implementation of the lorisplay unit generator written by Corbin Champion, and based on the method of Bandwidth-Enhanced Additive Synthesis and on the sound morphing algorithms implemented in the Loris library for sound modeling and manipulation. The opcodes were further adapted as a plugin for Csound 5 by Michael gogins. manual_src~dfsg/opcodes/lua_opcall.xml0000644000000000000000000001255612262561504017247 0ustar rootroot lua_opcall Lua Opcodes lua_opcall lua_opcall Calls a Lua opcode at i-rate only. Any number of output and/or input arguments may be passed. All arguments must be passed on the right-hand side. Outputs are returned in the argument. Syntax lua_iopcall Sname, ... lua_ikopcall Sname, ... lua_iaopcall Sname, ... lua_iopcall_off Sname, ... lua_ikopcall_off Sname, ... lua_iaopcall_off Sname, ... Initialization and Performance Sname -- The name of the opcode. ... -- An arbitrary list of any number of output and input arguments, of any type. The number, sequence, and types of these arguments must agree with the cdef of the arguments structure that is declared in the corresponding lua_opdef opcode. lua_iopcall calls a Lua opcode at i-rate only. Requires opname_init to be defined in Lua. lua_ikopcall calls a Lua opcode at i-rate and k-rate. Requires opname_init and opname_kontrol to be defined in Lua. lua_iaopcall calls a Lua opcode at i-rate and a-rate. Requires opname_init and opname_audio to be defined in Lua. lua_iopcall_off calls a Lua opcode at i-rate only. Requires opname_init and opname_noteoff to be defined in Lua. lua_ikopcall_off calls a Lua opcode at i-rate and k-rate. Requires opname_init, opname_kontrol, and opname_noteoff to be defined in Lua. lua_iaopcall_off calls a Lua opcode at i-rate and a-rate. Requires opname_init, opname_audio, and opname_noteoff to be defined in Lua. Lua code accesses elements of arguments as follows (pointers to both scalars and arrays are dereferenced by the Lua array access operator): ffi.cdef(' struct arguments_t { double *a_out, double *i_in, double *i_txt, double *f_sig };'); local arguments = ffi.cast("struct arguments_t *", carguments_lightuserdata) for i = 0, ksmps -1 do begin carguments.a_out[i] = carguments.i_in[0] * 3 end end The "off" variants of the opcode always schedule a "note off" event that is called when the intrument instance is removed from the active list, and which can be used to release unneeded resources, reschedule the instrument to render a reverb tail, and so on. Example Here is an example of a Lua opcode, showing how to pass strings back and forth between Lua opcodes and Csound orchestra code. The example uses the file luaopcode.csd. Example of a Lua opcode. See Also lua_exec, lua_opdef. Credits By: &namemichael; 2011 New in Csound version 5.13.2 manual_src~dfsg/opcodes/slider64table.xml0000644000000000000000000001404312262561504017571 0ustar rootroot slider64table Real-time MIDI:Slider Banks slider64table slider64table Stores a bank of 64 different MIDI control messages to a table. Description Creates a bank of 64 different MIDI control messages to a table. Syntax kflag slider64table ichan, ioutTable, ioffset, ictlnum1, imin1, \ imax1, init1, ifn1, .... , ictlnum64, imin64, imax64, init64, ifn64 Initialization ichan -- MIDI channel (1-16) ioutTable -- number of the table that will contain the output ioffset -- output table offset. A zero means that the output of the first slider will affect the first table element. A 10 means that the output of the first slider will affect the 11th table element. ictlnum1 ... ictlnum64 -- MIDI control number (0-127) imin1 ... imin64 -- minimum values for each controller imax1 ... imax64 -- maximum values for each controller init1 ... init64 -- initial value for each controller ifn1 ... ifn64 -- function table for conversion for each controller Performance kflag -- a flag that informs if any control-change message in the bank has been received. In this case kflag is set to 1. Otherwise is set to zero. slider64table is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider64table allows a bank of 64 different MIDI control message numbers. As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. slider64table is very similar to slider64 and sliderN family of opcodes (see their manual for more information). The actual difference is that the output is not stored to k-rate variables, but to a table, denoted by the ioutTable argument. It is possible to define a starting index in order to use the same table for more than one slider bank (or other purposes). It is possible to use this opcode together with FLslidBnk2Setk and FLslidBnk2, so you can synchronize the position of the MIDI values to the position of the FLTK valuator widgets of FLslidBnk2. Notice that you have to specify the same min/max values as well the linear/exponential responses in both sliderNtable(f) and FLslidBnk2. The exception is when using table-indexed response instead of a lin/exp response. In this case, in order to achieve a useful result, the table-indexed response and actual min/max values must be set only in FLslidBnk2, whereas, in sliderNtable(f), you have to set a linear response and a minimum of zero and a maximum of one in all sliders. See Also slider16table, slider16tablef, slider32table, slider32tablef, slider64tablef slider8table, slider8tablef Credits Author: &namegabriel; New in Csound version 5.06 manual_src~dfsg/opcodes/ifdef.xml0000644000000000000000000000516412262561504016206 0ustar rootroot #ifdeforchestra #ifdef Orchestra Syntax:Macros #ifdef #ifdef Conditional reading of code. Description If a macro is defined then #ifdef can incorporate text into an orchestra upto the next #end. This is similar to, but independent of, the macro system in the score language. Syntax #ifdef NAME .... #else  .... #end  Performance Note that the #ifdef can be nested, like in the C preprocessor language. Examples Here is a simple example of the conditional. Simple example of the #ifdef form. #define debug instr 1 #ifdef debug print "calling oscil" #end a1 oscil 32000,440,1 out a1 endin See Also $NAME, #define, #ifndef. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April 2005 New in Csound5 (and 4.23f13) manual_src~dfsg/opcodes/maxalloc.xml0000644000000000000000000000547312262561504016734 0ustar rootroot maxalloc Instrument Control:Realtime Performance Control maxalloc maxalloc Limits the number of allocations of an instrument. Description Limits the number of allocations of an instrument. Syntax maxalloc insnum, icount maxalloc Sinsname, icount Initialization insnum -- instrument number Sinsname -- instrument name icount -- number of instrument allocations Performance maxalloc limits the number of simultaneous instances (notes) of an instrument. Any score events after the maximum has been reached, are ignored. All instances of maxalloc must be defined in the header section, not in the instrument body. Examples Here is an example of the maxalloc opcode. It uses the file maxalloc.csd. Example of the maxalloc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should contain messages like these: WARNING: cannot allocate last note because it exceeds instr maxalloc See Also cpuprc, prealloc Credits Author: &namegabriel; Italy July 1999 New in Csound version 3.57 manual_src~dfsg/opcodes/convle.xml0000644000000000000000000000107512262561504016414 0ustar rootroot convle Signal Modifiers:Convolution and Morphing convle convle Same as the convolve opcode. Description Same as the convolve opcode. manual_src~dfsg/opcodes/sndwarpst.xml0000644000000000000000000002547612262561504017166 0ustar rootroot sndwarpst Signal Generators:Granular Synthesis sndwarpst sndwarpst Reads a stereo sound sample from a table and applies time-stretching and/or pitch modification. Description sndwarpst reads stereo sound samples from a table and applies time-stretching and/or pitch modification. Time and frequency modification are independent from one another. For example, a sound can be stretched in time while raising the pitch! The window size and overlap arguments are important to the result and should be experimented with. In general they should be as small as possible. For example, start with iwsize=sr/10 and ioverlap=15. Try irandw=iwsize*.2. If you can get away with less overlaps, the program will be faster. But too few may cause an audible flutter in the amplitude. The algorithm reacts differently depending upon the input sound and there are no fixed rules for the best use in all circumstances. But with proper tuning, excellent results can be achieved. Syntax ar1, ar2 [,ac1] [, ac2] sndwarpst xamp, xtimewarp, xresample, ifn1, \ ibeg, iwsize, irandw, ioverlap, ifn2, itimemode Initialization ifn1 -- the number of the table holding the sound samples which will be subjected to the sndwarpst processing. GEN01 is the appropriate function generator to use to store the sound samples from a pre-existing soundfile. ibeg -- the time in seconds to begin reading in the table (or soundfile). When itimemode is non-zero, the value of xtimewarp is offset by ibeg. iwsize -- the window size in samples used in the time scaling algorithm. irandw -- the bandwidth of a random number generator. The random numbers will be added to iwsize. ioverlap -- determines the density of overlapping windows. ifn2 -- a function used to shape the window. It is usually used to create a ramp of some kind from zero at the beginning and back down to zero at the end of each window. Try using a half a sine (i.e.: f1 0 16384 9 .5 1 0) which works quite well. Other shapes can also be used. Performance ar1, ar2 -- ar1 and ar2 are the stereo (left and right) outputs from sndwarpst. sndwarpst assumes that the function table holding the sampled signal is a stereo one. sndwarpst will index the table by a two-sample frame increment. The user must be aware then that if a mono signal is used with sndwarpst, time and pitch will be altered accordingly. ac1, ac2 -- ac1 and ac2 are single-layer (no overlaps), unwindowed versions of the time and/or pitch altered signal. They are supplied in order to be able to balance the amplitude of the signal output, which typically contains many overlapping and windowed versions of the signal, with a clean version of the time-scaled and pitch-shifted signal. The sndwarpst process can cause noticeable changes in amplitude, (up and down), due to a time differential between the overlaps when time-shifting is being done. When used with a balance unit, ac1 and ac2 can greatly enhance the quality of sound. They are optional, but note that they must both be present in the syntax (use both or neither). An example of how to use this is given below. xamp -- the value by which to scale the amplitude (see note on the use of this when using ac1 and ac2). xtimewarp -- determines how the input signal will be stretched or shrunk in time. There are two ways to use this argument depending upon the value given for itimemode. When the value of itimemode is 0, xtimewarp will scale the time of the sound. For example, a value of 2 will stretch the sound by 2 times. When itimemode is any non-zero value then xtimewarp is used as a time pointer in a similar way in which the time pointer works in lpread and pvoc. An example below illustrates this. In both cases, the pitch will not be altered by this process. Pitch shifting is done independently using xresample. xresample -- the factor by which to change the pitch of the sound. For example, a value of 2 will produce a sound one octave higher than the original. The timing of the sound, however, will not be altered. Example Here is an example of the sndwarpst opcode. It uses the file sndwarpst.csd. Example of the sndwarpst opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Other examples The below example shows a slowing down or stretching of the sound stored in the stored table (ifn1). Over the duration of the note, the stretching will grow from no change from the original to a sound which is ten times slower than the original. At the same time the overall pitch will move upward over the duration by an octave. iwindfun = 1 isampfun = 2 ibeg = 0 iwindsize = 2000 iwindrand = 400 ioverlap = 10 awarp line 1, p3, 1 aresamp line 1, p3, 2 kenv line 1, p3, .1 asig sndwarp kenv, awarp, aresamp, isampfun, ibeg, iwindsize, iwindrand, ioverlap, iwindfun, 0 Now, here's an example using xtimewarp as a time pointer and using stereo: itimemode = 1 atime line 0, p3, 10 ar1, ar2 sndwarpst kenv, atime, aresamp, sampfun, ibeg, iwindsize, iwindrand, ioverlap, \ iwindfun, itimemode In the above, atime advances the time pointer used in the sndwarpst from 0 to 10 over the duration of the note. If p3 is 20 then the sound will be two times slower than the original. Of course you can use a more complex function than just a single straight line to control the time factor. Now the same as above but using the balance function with the optional outputs: asig,acmp sndwarp 1, awarp, aresamp, isampfun, ibeg, iwindsize, iwindrand, ioverlap, iwindfun, itimemode abal balance asig, acmp asig1,asig2,acmp1,acmp2 sndwarpst 1, atime, aresamp, sampfun, ibeg, iwindsize, iwindrand, ioverlap, \ iwindfun, itimemode abal1 balance asig1, acmp1 abal2 balance asig2, acmp2 In the above two examples notice the use of the balance unit. The output of balance can then be scaled, enveloped, sent to an out or outs, and so on. Notice that the amplitude arguments to sndwarp and sndwarpst are 1 in these examples. By scaling the signal after the sndwarp process, abal, abal1, and abal2 should contain signals that have nearly the same amplitude as the original input signal to the sndwarp process. This makes it much easier to predict the levels and avoid samples out of range or sample values that are too small. More Advice Only use the stereo version when you really need to be processing a stereo file. It is somewhat slower than the mono version and if you use the balance function it is slower again. There is nothing wrong with using a mono sndwarp in a stereo orchestra and sending the result to one or both channels of the stereo output! See Also sndwarp Credits Author: Richard Karpen Seattle, WA USA 1997 manual_src~dfsg/opcodes/pchbend.xml0000644000000000000000000000574412262561504016540 0ustar rootroot pchbend Real-time MIDI:Input pchbend pchbend Get the current pitch-bend value for this channel. Description Get the current pitch-bend value for this channel. Syntax ibend pchbend [imin] [, imax] kbend pchbend [imin] [, imax] Initialization imin, imax (optional) -- set minimum and maximum limits on values obtained Performance Get the current pitch-bend value for this channel. Note that this access to pitch-bend data is independent of the MIDI pitch, enabling the value here to be used for any arbitrary purpose. Examples Here is an example of the pchbend opcode. It uses the file pchbend.csd. Example of the pchbend opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also aftouch, ampmidi, cpsmidi, cpsmidib, midictrl, notnum, octmidi, octmidib, pchmidi, pchmidib, veloc Credits Author: &namebarry; - &namemike; MIT - Mills May 1997 manual_src~dfsg/opcodes/babo.xml0000644000000000000000000001145412262561504016033 0ustar rootroot babo Signal Modifiers:Reverberation babo babo A physical model reverberator. Description babo stands for ball-within-the-box. It is a physical model reverberator based on the paper by Davide Rocchesso "The Ball within the Box: a sound-processing metaphor", Computer Music Journal, Vol 19, N.4, pp.45-47, Winter 1995. The resonator geometry can be defined, along with some response characteristics, the position of the listener within the resonator, and the position of the sound source. Syntax a1, a2 babo asig, ksrcx, ksrcy, ksrcz, irx, iry, irz [, idiff] [, ifno] Initialization irx, iry, irz -- the coordinates of the geometry of the resonator (length of the edges in meters) idiff -- is the coefficient of diffusion at the walls, which regulates the amount of diffusion (0-1, where 0 = no diffusion, 1 = maximum diffusion - default: 1) ifno -- expert values function: a function number that holds all the additional parameters of the resonator. This is typically a GEN2--type function used in non-rescaling mode. They are as follows: decay -- main decay of the resonator (default: 0.99) hydecay -- high frequency decay of the resonator (default: 0.1) rcvx, rcvy, rcvz -- the coordinates of the position of the receiver (the listener) (in meters; 0,0,0 is the resonator center) rdistance -- the distance in meters between the two pickups (your ears, for example - default: 0.3) direct -- the attenuation of the direct signal (0-1, default: 0.5) early_diff -- the attenuation coefficient of the early reflections (0-1, default: 0.8) Performance asig -- the input signal ksrcx, ksrcy, ksrcz -- the virtual coordinates of the source of sound (the input signal). These are allowed to move at k-rate and provide all the necessary variations in terms of response of the resonator. Examples Here is a simple example of the babo opcode. It uses the file babo.csd, and beats.wav. A simple example of the babo opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an advanced example of the babo opcode. It uses the file babo_expert.csd, and beats.wav. An advanced example of the babo opcode. Credits Author: Paolo Filippi Padova, Italy 1999 Nicola Bernardini Rome, Italy 2000 New in Csound version 4.09 manual_src~dfsg/opcodes/atone.xml0000644000000000000000000001027512262561504016236 0ustar rootroot atone Signal Modifiers:Standard Filters atone atone A hi-pass filter whose transfer functions are the complements of the tone opcode. Description A hi-pass filter whose transfer functions are the complements of the tone opcode. Syntax ares atone asig, khp [, iskip] Initialization iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance ares -- the output signal at audio rate. asig -- the input signal at audio rate. khp -- the response curve's half-power point, in Hertz. Half power is defined as peak power / root 2. atone is a filter whose transfer functions is the complement of tone. atone is thus a form of high-pass filter whose transfer functions represent the filtered out aspects of their complements. However, power scaling is not normalized in atone but remains the true complement of the corresponding unit. Thus an audio signal, filtered by parallel matching tone and atone units, would under addition simply reconstruct the original spectrum. This property is particularly useful for controlled mixing of different sources (see lpreson). Complex response curves such as those with multiple peaks can be obtained by using a bank of suitable filters in series. (The resultant response is the product of the component responses.) In such cases, the combined attenuation may result in a serious loss of signal power, but this can be regained by the use of balance. Examples Here is an example of the atone opcode. It uses the file atone.csd. Example of the atone opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also areson, aresonk, atonek, port, portk, reson, resonk, tone, tonek manual_src~dfsg/opcodes/STKVoicForm.xml0000644000000000000000000000757612262561504017250 0ustar rootroot STKVoicForm Signal Generators:STK Opcodes STKVoicForm STKVoicForm STKVoicForm is a four formant synthesis instrument. Description STKVoicForm is a four formant synthesis instrument. This instrument contains an excitation singing wavetable (looping wave with random and periodic vibrato, smoothing on frequency, etc.), excitation noise, and four sweepable complex resonances. Measured formant data is included, and enough data is there to support either parallel or cascade synthesis. In the floating point case cascade synthesis is the most natural so that's what you'll find here. Syntax asignal STKVoicForm ifrequency, iamplitude, [kmix, kv1[, ksel, kv2[, klfo, kv3[, klfodepth, kv4[, kloud, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kmix -- controller 2, voiced/unvoiced mix. Value range of kv1 is 0-127. ksel -- controller 4, vowel/phoneme selection. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kloud -- controller 128, loudness (spectral tilt). Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the VoicForm class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKVoicForm opcode. It uses the file STKVoicForm.csd. Example of the STKVoicForm opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/butterbr.xml0000644000000000000000000000570412262561504016762 0ustar rootroot butterbr Signal Modifiers:Standard Filters butterbr butterbr A band-reject Butterworth filter. Description Implementation of a second-order band-reject Butterworth filter. This opcode can also be written as butbr. Syntax ares butterbr asig, xfreq, xband [, iskip] Initialization iskip (optional, default=0) -- Skip initialization if present and non-zero. Performance These filters are Butterworth second-order IIR filters. They are slightly slower than the original filters in Csound, but they offer an almost flat passband and very good precision and stopband attenuation. asig -- Input signal to be filtered. xfreq -- Cutoff or center frequency for each of the filters. xband -- Bandwidth of the bandpass and bandreject filters. Examples Here is an example of the butterbr opcode. It uses the file butterbr.csd. Example of the butterbr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also butterbp, butterhp, butterlp Credits Author: &nameparis; MIT, Cambridge 1995 Existed in 3.30 Audio rate parameters introduced in version 6.02 October 2013. manual_src~dfsg/opcodes/strget.xml0000644000000000000000000000401312262561504016431 0ustar rootroot strget Strings:Definition strget strget Set string variable to value from strset table or string p-field Description strget sets a string variable at initialization time to the value stored in strset table at the specified index, or a string p-field from the score. If there is no string defined for the index, the variable is set to an empty string. Syntax Sdst strget indx Initialization indx -- strset index, or score p-field Sdst -- destination string variable Examples Here is an example of the strget opcode. It uses the file strget.csd. Example of the strget opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also strset Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/slider32table.xml0000644000000000000000000001404612262561504017567 0ustar rootroot slider32table Real-time MIDI:Slider Banks slider32table slider32table Stores a bank of 32 different MIDI control messages to a table. Description Creates a bank of 32 different MIDI control messages to a table. Syntax kflag slider32table ichan, ioutTable, ioffset, ictlnum1, imin1, \ imax1, init1, ifn1, .... , ictlnum32, imin32, imax32, init32, ifn32 Initialization ichan -- MIDI channel (1-16) ioutTable -- number of the table that will contain the output ioffset -- output table offset. A zero means that the output of the first slider will affect the first table element. A 10 means that the output of the first slider will affect the 11th table element. ictlnum1 ... ictlnum32 -- MIDI control number (0-127) imin1 ... imin32 -- minimum values for each controller imax1 ... imax32 -- maximum values for each controller init1 ... init32 -- initial value for each controller ifn1 ... ifn32 -- function table for conversion for each controller Performance kflag -- a flag that informs if any control-change message in the bank has been received. In this case kflag is set to 1. Otherwise is set to zero. slider32table is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider32table allows a bank of 32 different MIDI control message numbers. As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. slider32table is very similar to slider32 and sliderN family of opcodes (see their manual for more information). The actual difference is that the output is not stored to k-rate variables, but to a table, denoted by the ioutTable argument. It is possible to define a starting index in order to use the same table for more than one slider bank (or other purposes). It is possible to use this opcode together with FLslidBnk2Setk and FLslidBnk2, so you can synchronize the position of the MIDI values to the position of the FLTK valuator widgets of FLslidBnk2. Notice that you have to specify the same min/max values as well the linear/exponential responses in both sliderNtable(f) and FLslidBnk2. The exception is when using table-indexed response instead of a lin/exp response. In this case, in order to achieve a useful result, the table-indexed response and actual min/max values must be set only in FLslidBnk2, whereas, in sliderNtable(f), you have to set a linear response and a minimum of zero and a maximum of one in all sliders. See Also slider16table, slider16tablef, slider32tablef, slider64table, slider64tablef, slider8table, slider8tablef Credits Author: &namegabriel; New in Csound version 5.06 manual_src~dfsg/opcodes/pvsvoc.xml0000644000000000000000000000627512262561504016455 0ustar rootroot pvsvoc Spectral Processing:Streaming pvsvoc pvsvoc Combine the spectral envelope of one fsig with the excitation (frequencies) of another. Description This opcode provides support for cross-synthesis of amplitudes and frequencies. It takes the amplitudes of one input fsig and combines with frequencies from another. It is a spectral version of the well-known channel vocoder. Syntax fsig pvsvoc famp, fexc, kdepth, kgain [,kcoefs] Performance fsig -- output pv stream famp -- input pv stream from which the amplitudes will be extracted fexc -- input pv stream from which the frequencies will be taken kdepth -- depth of effect, affecting how much of the frequencies will be taken from the second fsig: 0, the output is the famp signal, 1 the output is the famp amplitudes and fexc frequencies. kgain -- gain boost/attenuation applied to the output. kcoefs -- number of cepstrum coefs used in spectral envelope estimation (defaults to 80). It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the pvsvoc opcode. It uses the file pvsvoc.csd. Example of the pvsvoc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows a typical cross-synthesis operation. The input signal (say a vocal sound) is used for its amplitude spectrum. An oscillator with an arbitrary complex waveform produces the excitation signal, giving the vocal sound its pitch. Credits Author: &namevictor; April 2005 New plugin in version 5 April 2005. manual_src~dfsg/opcodes/flhvsboxsetvalue.xml0000644000000000000000000000532612262561504020535 0ustar rootroot FLhvsBoxSetValue FLTK:Other FLhvsBoxSetValue FLhvsBoxSetValue Sets the cursor position of a previously-declared FLhvsBox widget. Description FLhvsBoxSetValue sets the cursor position of a previously-declared FLhvsBox widget. Syntax FLhvsBox kx, ky, ihandle Initialization ihandle – an integer number used a univocally-defined handle for identifying a specific HVS box (see below). Performance kx, ky– the coordinates of the HVS cursor position to be set. FLhvsBoxSetValue sets the cursor position of a previously-declared FLhvsBox widget. The kx and ky arguments, denoting the cursor position, have to be expressed in normalized values (0 to 1 range). See the entry for hvs2 for an example of usage of FLhvsBoxSetValue. See Also hvs2, FLhvsBox Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/xscansmap.xml0000644000000000000000000000452712262561504017130 0ustar rootroot xscansmap Signal Generators:Scanned Synthesis xscansmap xscansmap Allows the position and velocity of a node in a scanned process to be read. Description Allows the position and velocity of a node in a scanned process to be read. Syntax xscansmap kpos, kvel, iscan, kamp, kvamp [, iwhich] Initialization iscan -- which scan process to read iwhich (optional) -- which node to sense. The default is 0. Performance kpos -- the node's position. kvel -- the node's velocity. kamp -- amount to amplify the kpos value. kvamp -- amount to amplify the kvel value. The internal state of a node is read. This includes its position and velocity. They are amplified by the kamp and kvamp values. See Also More information on Scanned Synthesis (as well as several other matrices) is available on the Scanned Synthesis page at cSounds.com. Also an article on these opcodes: http://www.csounds.com/stevenyi/scanned/yi_scannedSynthesis.html , written by Steven Yi Credits New in version 4.21 November 2002. Thanks to &namerasmus; for pointing this opcode out. manual_src~dfsg/opcodes/ioutc.xml0000644000000000000000000000075312262561504016253 0ustar rootroot ioutc ioutc ioutc Deprecated. Description Deprecated as of version 3.52. Use the outic opcode instead. manual_src~dfsg/opcodes/pvsanal.xml0000644000000000000000000002125112262561504016570 0ustar rootroot pvsanal Spectral Processing:Streaming pvsanal pvsanal Generate an fsig from a mono audio source ain, using phase vocoder overlap-add analysis. Description Generate an fsig from a mono audio source ain, using phase vocoder overlap-add analysis. Syntax fsig pvsanal ain, ifftsize, ioverlap, iwinsize, iwintype [, iformat] [, iinit] Initialization ifftsize -- The FFT size in samples. Need not be a power of two (though these are especially efficient), but must be even. Odd numbers are rounded up internally. ifftsize determines the number of analysis bins in fsig, as ifftsize/2 + 1. For example, where ifftsize = 1024, fsig will contain 513 analysis bins, ordered linearly from the fundamental to Nyquist. The fundamental of analysis (which in principle gives the lowest resolvable frequency) is determined as sr/ifftsize. Thus, for the example just given and assuming sr = 44100, the fundamental of analysis is 43.07Hz. In practice, due to the phase-preserving nature of the phase vocoder, the frequency of any bin can deviate bilaterally, so that DC components are recorded. Given a strongly pitched signal, frequencies in adjacent bins can bunch very closely together, around partials in the source, and the lowest bins may even have negative frequencies. As a rule, the only reason to use a non power-of-two value for ifftsize would be to match the known fundamental frequency of a strongly pitched source. Values with many small factors can be almost as efficient as power-of-two sizes; for example: 384, for a source pitched at around low A=110Hz. ioverlap -- The distance in samples (hop size) between overlapping analysis frames. As a rule, this needs to be at least ifftsize/4, e.g. 256 for the example above. ioverlap determines the underlying analysis rate, as sr/ioverlap. ioverlap does not require to be a simple factor of ifftsize; for example a value of 160 would be legal. The choice of ioverlap may be dictated by the degree of pitch modification applied to the fsig, if any. As a rule of thumb, the more extreme the pitch shift, the higher the analysis rate needs to be, and hence the smaller the value for ioverlap. A higher analysis rate can also be advantageous with broadband transient sounds, such as drums (where a small analysis window gives less smearing, but more frequency-related errors). Note that it is possible, and reasonable, to have distinct fsigs in an orchestra (even in the same instrument), running at different analysis rates. Interactions between such fsigs is currently unsupported, and the fsig assignment opcode does not allow copying between fsigs with different properties, even if the only difference is in ioverlap. However, this is not a closed issue, as it is possible in theory to achieve crude rate conversion (especially with regard to in-memory analysis files) in ways analogous to time-domain techniques. iwinsize -- The size in samples of the analysis window filter (as set by iwintype). This must be at least ifftsize, and can usefully be larger. Though other proportions are permitted, it is recommended that iwinsize always be an integral multiple of ifftsize, e.g. 2048 for the example above. Internally, the analysis window (Hamming, von Hann) is multiplied by a sinc function, so that amplitudes are zero at the boundaries between frames. The larger analysis window size has been found to be especially important for oscillator bank resynthesis (e.g. using pvsadsyn), as it has the effect of increasing the frequency resolution of the analysis, and hence the accuracy of the resynthesis. As noted above, iwinsize determines the overall latency of the analysis/resynthesis system. In many cases, and especially in the absence of pitch modifications, it will be found that setting iwinsize=ifftsize works very well, and offers the lowest latency. iwintype -- The shape of the analysis window. Currently only two choices are implemented: 0 = Hamming window 1 = von Hann window Both are also supported by the PVOC-EX file format. The window type is stored as an internal attribute of the fsig, together with the other parameters (see pvsinfo). Other types may be implemented later on (e.g. the Kaiser window, also supported by PVOC-EX), though an obvious alternative is to enable windows to be defined via a function table. The main issue here is the constraint of f-tables to power-of-two sizes, so this method does not offer a complete solution. Most users will find the Hamming window meets all normal needs, and can be regarded as the default choice. iformat -- (optional) The analysis format. Currently only one format is implemented by this opcode: 0 = amplitude + frequency This is the classic phase vocoder format; easy to process, and a natural format for oscillator-bank resynthesis. It would be very easy (tempting, one might say) to treat an fsig frame not purely as a phase vocoder frame but as a generic additive synthesis frame. It is indeed possible to use an fsig this way, but it is important to bear in mind that the two are not, strictly speaking, directly equivalent. Other important formats (supported by PVOC-EX) are: 1 = amplitude + phase 2 = complex (real + imaginary) iformat is provided in case it proves useful later to add support for these other formats. Formats 0 and 1 are very closely related (as the phase is wrapped in both cases - it is a trivial matter to convert from one to the other), but the complex format might warrant a second explicit signal type (a csig) specifically for convolution-based processes, and other processes where the full complement of arithmetic operators may be useful. iinit -- (optional) Skip reinitialization. This is not currently implemented for any of these opcodes, and it remains to be seen if it is even practical. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the pvsanal opcode. It uses the file pvsanal.csd. Example of the pvsanal opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namerichard; August 2001 New in version 4.13 manual_src~dfsg/opcodes/chebyshevpoly.xml0000644000000000000000000001165212262561504020014 0ustar rootroot chebyshevpoly Signal Modifiers:Waveshaping chebyshevpoly chebyshevpoly Efficiently evaluates the sum of Chebyshev polynomials of arbitrary order. Description The chebyshevpoly opcode calculates the value of a polynomial expression with a single a-rate input variable that is made up of a linear combination of the first N Chebyshev polynomials of the first kind. Each Chebyshev polynomial, Tn(x), is weighted by a k-rate coefficient, kn, so that the opcode is calculating a sum of any number of terms in the form kn*Tn(x). Thus, the chebyshevpoly opcode allows for the waveshaping of an audio signal with a dynamic transfer function that gives precise control over the harmonic content of the output. Syntax aout chebyshevpoly ain, k0 [, k1 [, k2 [...]]] Performance ain -- the input signal used as the independent variable of the Chebyshev polynomials ("x"). aout -- the output signal ("y"). k0, k1, k2, ... -- k-rate multipliers for each Chebyshev polynomial. This opcode is very useful for dynamic waveshaping of an audio signal. Traditional waveshaping techniques utilize a lookup table for the transfer function -- usually a sum of Chebyshev polynomials. When a sine wave at full-scale amplitude is used as an index to read the table, the precise harmonic spectrum as defined by the weights of the Chebyshev polynomials is produced. A dynamic spectrum is acheived by varying the amplitude of the input sine wave, but this produces a non-linear change in the spectrum. By directly calculating the Chebyshev polynomials, the chebyshevpoly opcode allows more control over the spectrum and the number of harmonic partials added to the input can be varied with time. The value of each kn coefficient directly controls the amplitude of the nth harmonic partial if the input ain is a sine wave with amplitude = 1.0. This makes chebyshevpoly an efficient additive synthesis engine for N partials that requires only one oscillator instead of N oscillators. The amplitude or waveform of the input signal can also be changed for different waveshaping effects. If we consider the input parameter ain to be "x" and the output aout to be "y", then the chebyshevpoly opcode calculates the following equation: y = k0*T0(x) + k1*T1(x) + k2*T2(x) + k3*T3(x) + ... where the Tn(x) are defined by the recurrence relation T0(x) = 1, T1(x) = x, Tn(x) = 2x*T[n-1](x) - T[n-2](x) More information about Chebyshev polynomials can be found on Wikipedia at http://en.wikipedia.org/wiki/Chebyshev_polynomial See Also polynomial, mac maca Examples Here is an example of the chebyshevpoly opcode. It uses the file chebyshevpoly.csd. Example of the chebyshevpoly opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &nameanthony; January 2008 New in Csound version 5.08 manual_src~dfsg/opcodes/flsetfont.xml0000644000000000000000000000745212262561504017137 0ustar rootroot FLsetFont FLTK:Appearance FLsetFont FLsetFont Sets the font type of a FLTK widget. Description FLsetFont sets the font type of the target widget. Syntax FLsetFont ifont, ihandle Initialization ifont -- sets the the font type of the label of a widget. Legal values for ifont argument are: 1 - Helvetica (same as Arial under Windows) 2 - Helvetica Bold 3 - Helvetica Italic 4 - Helvetica Bold Italic 5 - Courier 6 - Courier Bold 7 - Courier Italic 8 - Courier Bold Italic 9 - Times 10 - Times Bold 11 - Times Italic 12 - Times Bold Italic 13 - Symbol 14 - Screen 15 - Screen Bold 16 - Dingbats ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/oscils.xml0000644000000000000000000000543512262561504016426 0ustar rootroot oscils Signal Generators:Basic Oscillators oscils oscils A simple, fast sine oscillator Description Simple, fast sine oscillator, that uses only one multiply, and two add operations to generate one sample of output, and does not require a function table. Syntax ares oscils iamp, icps, iphs [, iflg] Initialization iamp -- output amplitude. icps -- frequency in Hz (may be zero or negative, however the absolute value must be less than sr/2). iphs -- start phase between 0 and 1. iflg -- sum of the following values: 2: use double precision even if Csound was compiled to use floats. This improves quality (especially in the case of long performance time), but may be up to twice as slow. 1: skip initialization. Performance ares -- audio output Examples Here is an example of the oscils opcode. It uses the file oscils.csd. Example of the oscils opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &nameistvan; January 2002 New in version 4.18 manual_src~dfsg/opcodes/spdist.xml0000644000000000000000000001747512262561504016447 0ustar rootroot spdist Signal Modifiers:Panning and Spatialization spdist spdist Calculates distance values from xy coordinates. Description spdist uses the same xy data as space, also either from a text file using Gen28 or from x and y arguments given to the unit directly. The purpose of this unit is to make available the values for distance that are calculated from the xy coordinates. In the case of space, the xy values are used to determine a distance which is used to attenuate the signal and prepare it for use in spsend. But it is also useful to have these values for distance available to scale the frequency of the signal before it is sent to the space unit. Syntax k1 spdist ifn, ktime, kx, ky Initialization ifn -- number of the stored function created using Gen28. This function generator reads a text file which contains sets of three values representing the xy coordinates and a time-tag for when the signal should be placed at that location. The file should look like: 0 -1 1 1 1 1 2 4 4 2.1 -4 -4 3 10 -10 5 -40 0 If that file were named "move" then the Gen28 call in the score would like: f1 0 0 28 "move" Gen28 takes 0 as the size and automatically allocates memory. It creates values to 10 milliseconds of resolution. So in this case there will be 500 values created by interpolating X1 to X2 to X3 and so on, and Y1 to Y2 to Y3 and so on, over the appropriate number of values that are stored in the function table. In the above example, the sound will begin in the left front, over 1 second it will move to the right front, over another second it move further into the distance but still in the right front, then in just 1/10th of a second it moves to the left rear, a bit distant. Finally over the last .9 seconds the sound will move to the right rear, moderately distant, and it comes to rest between the two left channels (due west!), quite distant. Since the values in the table are accessed through the use of a time-pointer in the space unit, the actual timing can be made to follow the file's timing exactly or it can be made to go faster or slower through the same trajectory. If you have access to the GUI that allows one to draw and edit the files, there is no need to create the text files manually. But as long as the file is ASCII and in the format shown above, it doesn't matter how it is made! IMPORTANT: If ifn is 0 then spdist will take its values for the xy coordinates from kx and ky. Performance The configuration of the xy coordinates in space places the signal in the following way: a1 is -1, 1 a2 is 1, 1 a3 is -1, -1 a4 is 1, -1 This assumes a loudspeaker set up as a1 is left front, a2 is right front, a3 is left back, a4 is right back. Values greater than 1 will result in sounds being attenuated, as if in the distance. space considers the speakers to be at a distance of 1; smaller values of xy can be used, but space will not amplify the signal in this case. It will, however balance the signal so that it can sound as if it were within the 4 speaker space. x=0, y=1, will place the signal equally balanced between left and right front channels, x=y=0 will place the signal equally in all 4 channels, and so on. Although there must be 4 output signals from space, it can be used in a 2 channel orchestra. If the xy's are kept so that Y>=1, it should work well to do panning and fixed localization in a stereo field. ktime -- index into the table containing the xy coordinates. If used like: ktime line 0, 5, 5 a1, a2, a3, a4 space asig, 1, ktime, ... with the file "move" described above, the speed of the signal's movement will be exactly as described in that file. However: ktime line 0, 10, 5 the signal will move at half the speed specified. Or in the case of: ktime line 5, 15, 0 the signal will move in the reverse direction as specified and 3 times slower! Finally: ktime line 2, 10, 3 will cause the signal to move only from the place specified in line 3 of the text file to the place specified in line 5 of the text file, and it will take 10 seconds to do it. kx, ky -- when ifn is 0, space and spdist will use these values as the XY coordinates to localize the signal. Examples Here is an example of the spdist opcode. It uses the file spdist.csd. Example of the spdist opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The same function and time values are used for both spdist and space. This insures that the distance values used internally in the space unit will be the same as those returned by spdist to give the impression of a Doppler shift! See Also space, spsend Credits Author: Richard Karpen Seattle, WA USA 1998 New in Csound version 3.48 manual_src~dfsg/opcodes/ATSadd.xml0000644000000000000000000001617712262561504016237 0ustar rootroot ATSadd Spectral Processing:ATS ATSadd ATSadd uses the data from an ATS analysis file to perform additive synthesis. Description ATSadd reads from an ATS analysis file and uses the data to perform additive synthesis using an internal array of interpolating oscillators. Syntax ar ATSadd ktimepnt, kfmod, iatsfile, ifn, ipartials[, ipartialoffset, \ ipartialincr, igatefn] Initialization iatsfile – the ATS number (n in ats.n) or the name in quotes of the analysis file made using ATS. ifn – table number of a stored function containing a sine wave for ATSadd and a cosine for ATSaddnz (see examples below for more info) ipartials – number of partials that will be used in the resynthesis (the noise has a maximum of 25 bands) ipartialoffset (optional) – is the first partial used (defaults to 0). ipartialincr (optional) – sets an increment by which these synthesis opcodes counts up from ipartialoffset for ibins components in the re-synthesis (defaults to 1). igatefn (optional) – is the number of a stored function which will be applied to the amplitudes of the analysis bins before resynthesis takes place. If igatefn is greater than 0 the amplitudes of each bin will be scaled by igatefn through a simple mapping process. First, the amplitudes of all of the bins in all of the frames in the entire analysis file are compared to determine the maximum amplitude value. This value is then used to create normalized amplitudes as indices into the stored function igatefn. The maximum amplitude will map to the last point in the function. An amplitude of 0 will map to the first point in the function. Values between 0 and 1 will map accordingly to points along the function table. See the examples below. Performance ktimepnt – The time pointer in seconds used to index the ATS file. Used for ATSadd exactly the same as for pvoc. ATSadd and ATSaddnz are based on pvadd by Richard Karpen and use files created by Juan Pampin's ATS (Analysis - Transformation - Synthesis). kfmod – A control-rate transposition factor: a value of 1 incurs no transposition, 1.5 transposes up a perfect fifth, and .5 down an octave. Used for ATSadd exactly the same as for pvoc. ATSadd reads from an ATS analysis file and uses the data to perform additive synthesis using an internal array of interpolating oscillators. The user supplies the wave table (usually one period of a sine wave), and can choose which analysis partials will be used in the re-synthesis. Examples ktime line 0, p3, 2.5 asig ATSadd ktime, 1, "clarinet.ats", 1, 20, 2 In the example above, ipartials is 20 and ipartialoffset is 2. This will synthesize the 3rd thru 22nd partials in the "clarinet.ats" analysis file. kfmod is 1 so there will be no pitch transformation. Since the ktimepnt envelope moves from 0 to 2.5 over the duration of the note, the analysis file will be read from 0 to 2.5 seconds of the original duration of the analysis over the duration of the csound note, this way we can change the duration independent of the pitch. Examples Here is a complete example of the ATSadd opcode. It uses the file ATSadd.csd. Example of the ATSadd opcode. In the above example we synthesize 20 partials as in example 1 except this time we're using a ipartialoffset of 0 and ipartialincr of 2, which means that we'll start from the first partial and synthesize 20 partials total, skipping every other one (ie. partial 1, 3, 5,..). Here is another example of the ATSadd opcode. It uses the file ATSadd-2.csd. Example 2 of the ATSadd opcode. See also ATSread, ATSreadnz, ATSinfo, ATSbufread, ATScross, ATSinterpread, ATSpartialtap, ATSaddnz, ATSsinnoi Credits Author: Alex Norman Seattle,Washington 2004 manual_src~dfsg/opcodes/ATSinterpread.xml0000644000000000000000000001045712262561504017637 0ustar rootroot ATSinterpread Spectral Processing:ATS ATSinterpread ATSinterpread allows a user to determine the frequency envelope of any ATSbufread. Description ATSinterpread allows a user to determine the frequency envelope of any ATSbufread. Syntax kamp ATSinterpread kfreq Performance kfreq - a frequency value (given in Hertz) used by ATSinterpread as in index into the table produced by an ATSbufread. ATSinterpread takes a frequency value (kfreq in Hz). This frequency is used to index the data of an ATSbufread. The return value is an amplitude gained from the ATSbufread after interpolation. ATSinterpread allows a user to determine the frequency envelope of any ATSbufread. This data could be useful for an number of reasons, one might be performing cross synthesis of data from an ATS file and non ATS data. Examples Here is an example of the ATSinterpread opcode. It uses the file ATSinterpread.csd. Example of the ATSinterpread opcode. This example shows how to use ATSinterpread. Here a frequency is given by the score (p4) and this frequency is given to an ATSinterpread (with a corresponding ATSbufread). The ATSinterpread uses this frequency to output a corresponding amplitude value, based on the atsfile given by the ATSbufread (beats.ats in this case). We then use that amplitude to scale a sine-wave that is synthesized with the same frequency (p4). You could extend this to include multiple sine-waves. This way you could synthesize any reasonable frequency (within the low and high frequencies of the indicated ATS file), and maintain the shape (in frequency) of the indicated atsfile (given by the ATSbufread). See also ATSread, ATSreadnz, ATSinfo, ATSsinnoi, ATSbufread, ATScross, ATSpartialtap, ATSadd, ATSaddnz Credits Author: Alex Norman Seattle,Washington 2004 manual_src~dfsg/opcodes/poisson.xml0000644000000000000000000001641312262561504016622 0ustar rootroot x-class noise generators poisson Signal Generators:Random (Noise) Generators poisson poisson Poisson distribution random number generator (positive values only). Description Poisson distribution random number generator (positive values only). This is an x-class noise generator. Syntax ares poisson klambda ires poisson klambda kres poisson klambda Performance ares, kres, ires - number of events occuring (always an integer). klambda - the expected number of occurrences that occur during the rate interval. Adapted from Wikipedia: In probability theory and statistics, the Poisson distribution is a discrete probability distribution. It expresses the probability of a number of events occurring in a fixed period of time if these events occur with a known average rate, and are independent of the time since the last event. The Poisson distribution describing the probability that there are exactly k occurrences (k being a non-negative integer, k = 0, 1, 2, ...) is: [The Poisson distribution equation.] where: λ is a positive real number, equal to the expected number of occurrences that occur during the given interval. For instance, if the events occur on average every 4 minutes, and you are interested in the number of events occurring in a 10 minute interval, you would use as model a Poisson distribution with λ = 10/4 = 2.5. This parameter is called klambda on the poisson opcodes. k refers to the number of i- , k- or a- periods elapsed. The Poisson distribution arises in connection with Poisson processes. It applies to various phenomena of discrete nature (that is, those that may happen 0, 1, 2, 3, ... times during a given period of time or in a given area) whenever the probability of the phenomenon happening is constant in time or space. Examples of events that can be modelled as Poisson distributions include: The number of cars that pass through a certain point on a road (sufficiently distant from traffic lights) during a given period of time. The number of spelling mistakes one makes while typing a single page. The number of phone calls at a call center per minute. The number of times a web server is accessed per minute. The number of roadkill (animals killed) found per unit length of road. The number of mutations in a given stretch of DNA after a certain amount of radiation. The number of unstable nuclei that decayed within a given period of time in a piece of radioactive substance. The radioactivity of the substance will weaken with time, so the total time interval used in the model should be significantly less than the mean lifetime of the substance. The number of pine trees per unit area of mixed forest. The number of stars in a given volume of space. The distribution of visual receptor cells in the retina of the human eye. The number of viruses that can infect a cell in cell culture. [A diagram showing the Poisson distribution.] A diagram showing the Poisson distribution. For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the poisson opcode. It uses the file poisson.csd. It is written for *NIX systems, and will generate errors on Windows. Example of the poisson opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also seed, betarand, bexprnd, cauchy, exprand, gauss, linrand, pcauchy, trirand, unirand, weibull Credits Author: &nameparis; MIT, Cambridge 1995 manual_src~dfsg/opcodes/vphaseseg.xml0000644000000000000000000001070512262561504017113 0ustar rootroot vphaseseg FLTK:Other vphaseseg vphaseseg Allows one-dimensional HVS (Hyper-Vectorial Synthesis). Description vphaseseg allows one-dimensional HVS (Hyper-Vectorial Synthesis). Syntax vphaseseg kphase, ioutab, ielems, itab1,idist1,itab2 \ [,idist2,itab3, ... ,idistN-1,itabN] Initialization ioutab - number of output table. ielem - number of elements to process itab1,...,itabN - breakpoint table numbers idist1,...,idistN-1 - distances between breakpoints in percentage values Performance kphase - phase pointer vphaseseg returns the coordinates of section points of an N-dimensional space path. The coordinates of section points are stored into an output table. The number of dimensions of the N-dimensional space is determined by the ielem argument that is equal to N and can be set to any number. To define the path, user have to provide a set of points of the N-dimensional space, called break-points. Coordinates of each break-point must be contained by a different table. The number of coordinates to insert in each break-point table must obviously equal to ielem argument. There can be any number of break-point tables filled by the user. Hyper-Vectorial Synthesis actually deals with two kinds of spaces. The first space is the N-dimensional space in which the path is defined, this space is called time-variant parameter space (or SPACE A). The path belonging to this space is covered by moving a point into the second space that normally has a number of dimensions smaller than the first. Actually, the point in motion is the projection of corresponding point of the N-dimensional space (could also be considered a section of the path). The second space is called user-pointer-motion space (or SPACE B) and, in the case of vphaseseg opcode, has only ONE DIMENSION. Space B is covered by means of kphase argument (that is a sort of path pointer), and its range is 0 to 1. The output corresponding to current pointer value is stored in ioutab table, whose data can be afterwards used to control any syntesis parameters. In vphaseseg, each break-point is separated from the other by a distance expressed in percentage, where all the path length is equal to the sum of all distances. So distances between breakpoints can be different, differently from kinds of HVS in which space B has more than one dimension, in these cases distance between break-points MUST be THE SAME for all intervals. See Also hvs1, hvs2, hvs3 Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/writescratch.xml0000644000000000000000000000413112262561504017624 0ustar rootroot writescratch Instrument Control:Sensing and Control writescratch writescratch writes a value into the scratchpad of the instance of an instrument. Description The writescratch opcode writes one of four scalar values to be stored in the instance of an instrument. Syntax writescratchival[, index] Initialisation ival -- variable to write. index -- which value to write, defaulting to zero. Examples Here is an example of the writescratch opcode. It uses the file readscratch.csd. Example of the writescratch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also readscratch, Credits Author: &namejohn; March 2013 New in Csound version 6.00 manual_src~dfsg/opcodes/log.xml0000644000000000000000000000515612262561504015713 0ustar rootroot log Mathematical Operations:Mathematical Functions log log Returns a natural log. Description Returns the natural log of x (x positive only). The argument value is restricted for log, log10, and sqrt. Syntax log(x) (no rate restriction) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Examples Here is an example of the log opcode. It uses the file log.csd. Example of the log opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: i1 = 2.079 See Also abs, exp, frac, int, log10, i, sqrt Credits Written by &namejohn;. New in version 3.47 Example written by &namekevin;. manual_src~dfsg/opcodes/plltrack.xml0000644000000000000000000000634712262561504016751 0ustar rootroot plltrack Instrument Control:Sensing and Control plltrack plltrack Tracks the pitch of a signal. Description plltrack, a pitch tracker based on a phase-locked loop algorithm, described in Zolzer, U, Sankarababu, S.V. and Moller, S, "PLL-based Pitch Detection and Tracking for Audio Signals. Proc. of IIH-MSP 2012". Syntax acps, alock plltrack asig, kd [, kloopf, kloopq, klf, khf, kthresh] Performance acps -- estimated pitch in Hz. alock -- phase lock indicator, a phase error indicating the quality of the tracking, with values between 0 and 1. Higher values indicate good tracking kd -- PLL feedback gain, controls frequency range of PLL (between 0 and 1). Higher values increase the range of the tracking. kloopf -- PLL LP filter cf, controls frequency range of PLL (opt, defaults to 20Hz). kloopq -- PLL LP filter Q, controls rise time of FO step (opt, defaults to 1/3) klf -- lowest tracking freq (opt, defaults to 20Hz) khf -- highest tracking freq (opt, defaults to 1500Hz) kthresh -- tracking signal level threshold (optional, defaults to 0.001, equiv to -60dBfs) plltrack analyzes the input signal, asig, estimating the fundamental of a monophonic signal. Its output is updated every sample. Examples Here is an example of the plltrack operator. It uses the file plltrack.csd. Example of the plltrack operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; NUI, Maynooth. Maynooth, Ireland September, 2012 New in Csound version 5.18.031 manual_src~dfsg/opcodes/reverb.xml0000644000000000000000000001022412262561504016407 0ustar rootroot reverb Signal Modifiers:Reverberation reverb reverb Reverberates an input signal with a natural room frequency response. Description Reverberates an input signal with a natural room frequency response. Syntax ares reverb asig, krvt [, iskip] Initialization iskip (optional, default=0) -- initial disposition of delay-loop data space (cf. reson). The default value is 0. Performance krvt -- the reverberation time (defined as the time in seconds for a signal to decay to 1/1000, or 60dB down from its original amplitude). A standard reverb unit is composed of four comb filters in parallel followed by two alpass units in series. Loop times are set for optimal natural room response. Core storage requirements for this unit are proportional only to the sampling rate, each unit requiring approximately 3K words for every 10KC. The comb, alpass, delay, tone and other Csound units provide the means for experimenting with alternate reverberator designs. Since output from the standard reverb will begin to appear only after 1/20 second or so of delay, and often with less than three-fourths of the original power, it is normal to output both the source and the reverberated signal. If krvt is inadvertently set to a non-positive number, krvt will be reset automatically to 0.01. (New in Csound version 4.07.) Also, since the reverberated sound will persist long after the cessation of source events, it is normal to put reverb in a separate instrument to which sound is passed via a global variable, and to leave that instrument running throughout the performance. Examples Here is an example of the reverb opcode. It uses the file reverb.csd. Example of the reverb opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also alpass, comb, valpass, vcomb Credits Author: William Pete Moss University of Texas at Austin Austin, Texas USA January 2002 manual_src~dfsg/opcodes/expsegr.xml0000644000000000000000000001206312262561504016602 0ustar rootroot expsegr Signal Generators:Linear and Exponential Generators expsegr expsegr Trace a series of exponential segments between specified points including a release segment. Description Trace a series of exponential segments between specified points including a release segment. Syntax ares expsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz kres expsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz Initialization ia -- starting value. Zero is illegal for exponentials. ib, ic, etc. -- value after dur1 seconds, etc. For exponentials, must be non-zero and must agree in sign with ia. idur1 -- duration in seconds of first segment. A zero or negative value will cause all initialization to be skipped. idur2, idur3, etc. -- duration in seconds of subsequent segments. A zero or negative value will terminate the initialization process with the preceding point, permitting the last-defined line or curve to be continued indefinitely in performance. The default is zero. irel, iz -- duration in seconds and final value of a note releasing segment. Performance These units generate control or audio signals whose values can pass through 2 or more specified points. The sum of dur values may or may not equal the instrument's performance time: a shorter performance will truncate the specified pattern, while a longer one will cause the last-defined segment to continue on in the same direction. expsegr is amongst the Csound r units that contain a note-off sensor and release time extender. When each senses an event termination or MIDI noteoff, it immediately extends the performance time of the current instrument by irel seconds, and sets out to reach the value iz by the end of that period (no matter which segment the unit is in). r units can also be modified by MIDI noteoff velocities. For two or more extenders in an instrument, extension is by the greatest period. You can use other pre-made envelopes which start a release segment upon recieving a note off message, like linsegr and madsr, or you can construct more complex envelopes using xtratim and release. Note that you don't need to use xtratim if you are using expsegr, since the time is extended automatically. Examples Here is an example of the expsegr opcode. It uses the file expsegr.csd. Example of the expsegr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also linsegr, expsegr, envlpxr, mxadsr, madsr expon, expseg, expsega, xtratim Credits Author: &namebarry; New in Csound 3.47 manual_src~dfsg/opcodes/itablemix.xml0000644000000000000000000000100712262561504017077 0ustar rootroot itablemix itablemix itablemix Deprecated. Description Deprecated as of version 3.52. Use the tableimix opcode instead. manual_src~dfsg/opcodes/elseif.xml0000644000000000000000000000637212262561504016402 0ustar rootroot elseif Instrument Control:Program Flow Control elseif elseif Defines another "if...then" condition when a "if...then" condition is false. Description Defines another "if...then" condition when a "if...then" condition is false. Syntax elseif xa R xb then where label is in the same instrument block and is not an expression, and where R is one of the Relational operators (<, =, <=, ==, !=) (and = for convenience, see also under Conditional Values). Performance elseif is used inside of a block of code between the "if...then" and endif opcodes. When a "if...then" condition is false, it defines another "if...then" condition to be met. Any number of elseif statements are allowed. Examples Here is an example of the elseif opcode. It uses the file elseif.csd. Example of the elseif opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also else, endif, goto, if, igoto, kgoto, tigoto, timout More information on this opcode: http://www.csounds.com/journal/2006spring/controlFlow.html , written by Steven Yi Credits New in version 4.21 manual_src~dfsg/opcodes/octmidi.xml0000644000000000000000000000733212262561504016560 0ustar rootroot octmidi Real-time MIDI:Converters octmidi octmidi Get the note number, in octave-point-decimal units, of the current MIDI event. Description Get the note number, in octave-point-decimal units, of the current MIDI event. Syntax ioct octmidi Performance Get the note number of the current MIDI event, expressed in octave-point-decimal units, for local processing. octmidi vs. octmidinn The octmidi opcode only produces meaningful results in a Midi-activated note (either real-time or from a Midi score with the -F flag). With octmidi, the Midi note number value is taken from the Midi event that is internally associated with the instrument instance. On the other hand, the octmidinn opcode may be used in any Csound instrument instance whether it is activated from a Midi event, score event, line event, or from another instrument. The input value for octmidinn might for example come from a p-field in a textual score or it may have been retrieved from the real-time Midi event that activated the current note using the notnum opcode. Examples Here is an example of the octmidi opcode. It uses the file octmidi.csd. Example of the octmidi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also aftouch, ampmidi, cpsmidi, cpsmidib, midictrl, notnum, octmidib, pchbend, pchmidi, pchmidib, veloc, cpsmidinn, octmidinn, pchmidinn Credits Author: &namebarry; - &namemike; MIT - Mills May 1997 Example written by &namekevin;. manual_src~dfsg/opcodes/round.xml0000644000000000000000000000616412262561504016261 0ustar rootroot round Mathematical Operations:Mathematical Functions round round Returns the integer value nearest to x ; if the fractional part of x is exactly 0.5, the direction of rounding is undefined. Description The integer value nearest to x ; if the fractional part of x is exactly 0.5, the direction of rounding is undefined. Syntax round(x) (init-, control-, or audio-rate arg allowed) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Examples Here is an example of the round opcode. It uses the file round.csd. Example of the round opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like these: instr 1: inumber = 9.000 idiv = 1.000 ifl = 9.000 instr 1: inumber = 9.000 idiv = 2.000 ifl = 5.000 instr 1: inumber = 9.000 idiv = 3.000 ifl = 3.000 instr 1: inumber = 9.000 idiv = 4.000 ifl = 2.000 instr 1: inumber = 9.000 idiv = 5.000 ifl = 2.000 instr 1: inumber = 9.000 idiv = 6.000 ifl = 2.000 instr 1: inumber = 9.000 idiv = 7.000 ifl = 1.000 instr 1: inumber = 9.000 idiv = 8.000 ifl = 1.000 instr 1: inumber = 9.000 idiv = 9.000 ifl = 1.000 instr 1: inumber = 9.000 idiv = 10.000 ifl = 1.000 See Also abs, exp, int, log, log10, i, sqrt Credits Author: Istvan Varga New in Csound 5 2005 manual_src~dfsg/opcodes/tablefilteri.xml0000644000000000000000000000675212262561504017603 0ustar rootroot tablefilteri Table Control:Read/Write Opreations tablefilteri tablefilteri Filters a source table and writes result into a destination table. Description This opcode can be used in order to filter values from function tables following certain algorithms. The filtered output is written into a destination table and the number of elements that have passed the filter is returned. Syntax inumpassed tablefilteri iouttable, iintatble, imode, iparam Initialization inumpassed -- the number of elements that have passed the filter. iouttable -- the number of the table containing the values that have passed. iintatble -- the number of the table used as filter input. imode -- mode of the filter: 1 -- tests the weight of the denominators of the fractions in the source table. Letting pass only values from the source that are less heavy than the weight of the threshold. 2 -- tests the weight of the denominators of the fractions in the source table. Letting pass only values from the source that are heavier than or equal to the weight of the threshold. iparam -- integer threshold parameter for the filter. It means that denominators whose weights are heavier than the weight of this threshold are not passed through the filter. The weight of an integer is calculated using Clarence Barlow's function of indigestibility of a number. According to this function, higher prime numbers contribute to an increased weight of any natural integer they divide. The order of the first 16 integers according to their indigestibility is: 1, 2, 4, 3, 8, 6, 16, 12, 9, 5, 10, 15, 7, 14. Examples Here is an example of the tablefilteri opcode. It uses the file tablefilter.csd. Example of the tablefilteri opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Georg Boenn University of Glamorgan, UK New in Csound version 5.13 manual_src~dfsg/opcodes/upsamp.xml0000644000000000000000000000340312262561504016430 0ustar rootroot upsamp Signal Modifiers:Sample Level Operators upsamp upsamp Modify a signal by up-sampling. Description Modify a signal by up-sampling. Syntax ares upsamp ksig Performance upsamp converts a control signal to an audio signal. It does it by simple repetition of the kval. upsamp is a slightly more efficient form of the assignment, asig = ksig. Examples Here is an example of the upsamp opcode. It uses the file upsamp.csd. Example of the upsamp opcode. See Also diff, downsamp, integ, interp, samphold manual_src~dfsg/opcodes/flmouse.xml0000644000000000000000000000607612262561504016606 0ustar rootroot FLmouse FLTK:Other FLmouse FLmouse Returns the mouse position and the state of the three mouse buttons. Description FLmouse returns the coordinates of the mouse position within an FLTK panel and the state of the three mouse buttons. Syntax kx, ky, kb1, kb2, kb3 FLmouse [imode] Initialization imode – (optional, default = 0) Determines the mode for mouse location reporting. 0 - Absolute position normalized to range 0-1 1 - Absolute raw pixel position 2 - Raw pixel position, relative to FLTK panel Performance kx, ky – the mouse coordinates, whose range depends on the imode argument (see above). kb1, kb2, kb3 – the states of the mouse buttons, 1 when corresponding button is pressed, 0 when the button is not pressed. FLmouse returns the coordinates of the mouse position and the state of the three mouse buttons. The coordinates can be retrieved in three modes depending on the imode argument value (see above). Modes 0 and 1 report mouse position in relation to the complete screen (Absolute mode), while mode 2, reports the pixel position within an FLTK panel. Notice that FLmouse is only active when the mouse cursor passes on an FLpanel area. Examples Here is an example of the FLmouse opcode. It uses the file FLmouse.csd. Example of the FLmouse opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/flloadsnap.xml0000644000000000000000000000541212262561504017250 0ustar rootroot FLloadsnap FLTK:Other FLloadsnap FLloadsnap Loads all snapshots into the memory bank of the current orchestra. Description FLloadsnap loads all the snapshots contained in a file into the memory bank of the current orchestra. Syntax FLloadsnap "filename" [, igroup] Initialization "filename" -- a double-quoted string corresponding to a file to load a bank of snapshots. igroup -- (optional) an integer number referring to a snapshot-related group of widget. It allows to get/set, or to load/save the state of a subset of valuators. Default value is zero that refers to the first group. The group number is determined by the opcode FLsetSnapGroup. The igroup parameter has not been yet fully implemented in the current version of csound. Please do not rely on it yet. Performance FLloadsnap loads all snapshots contained in filename into the memory bank of current orchestra. For purposes of snapshot saving, widgets can be grouped, so that snapshots affect only a defined group of widgets. The opcode FLsetSnapGroup is used to specify the group for all widgets declared after it, until the next FLsetSnapGroup statement. See Also FLgetsnap, FLrun, FLsetSnapGroup, FLsavesnap, FLsetsnap, FLupdate Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/lessequal.xml0000644000000000000000000000767612262561504017141 0ustar rootroot : ? conditional expressions <= Instrument Control:Conditional Values <= <= Determines if one value is less than or equal to another. Description Determines if one value is less than or equal to another. Syntax (a <= b ? v1 : v2) where a, b, v1 and v2 may be expressions, but a, b not audio-rate. Performance In the above conditional, a and b are first compared. If the indicated relation is true (a less than or equal to b), then the conditional expression has the value of v1; if the relation is false, the expression has the value of v2. NB.: If v1 or v2 are expressions, these will be evaluated before the conditional is determined. In terms of binding strength, all conditional operators (i.e. the relational operators (<, etc.), and ?, and : ) are weaker than the arithmetic and logical operators (+, -, *, /, && and ||). These are operators not opcodes. Therefore, they can be used within orchestra statements, but do not form complete statements themselves. Examples Here is an example of the <= operator. It uses the file lessequal.csd. Example of the <= operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: random number = 1.035781, result = 3.000000 random number = 0.134037, result = 1.000000 random number = 0.130742, result = 1.000000 random number = 1.002550, result = 3.000000 random number = 0.370565, result = 1.000000 random number = 0.655759, result = 2.000000 random number = 0.676154, result = 2.000000 ...... See Also ==, >=, >, <, != manual_src~dfsg/opcodes/tablew.xml0000644000000000000000000002211212262561504016377 0ustar rootroot tablew Table Control:Read/Write Opreations tablew tablew Change the contents of existing function tables. Description This opcode operates on existing function tables, changing their contents. tablew is for writing at k- or at a-rates, with the table number being specified at init time. Using tablew with i-rate signal and index values is allowed, but the specified data will always be written to the function table at k-rate, not during the initialization pass. The valid combinations of variable types are shown by the first letter of the variable names. Syntax tablew asig, andx, ifn [, ixmode] [, ixoff] [, iwgmode] tablew isig, indx, ifn [, ixmode] [, ixoff] [, iwgmode] tablew ksig, kndx, ifn [, ixmode] [, ixoff] [, iwgmode] Initialization asig, isig, ksig -- The value to be written into the table. andx, indx, kndx -- Index into table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0) ifn -- Table number. Must be >= 1. Floats are rounded down to an integer. If a table number does not point to a valid table, or the table has not yet been loaded (GEN01) then an error will result and the instrument will be de-activated. ixmode (optional, default=0) -- index mode. 0 = xndx and ixoff ranges match the length of the table. !=0 = xndx and ixoff have a 0 to 1 range. ixoff (optional, default=0) -- index offset. 0 = Total index is controlled directly by xndx, i.e. the indexing starts from the start of the table. !=0 = Start indexing from somewhere else in the table. Value must be positive and less than the table length (ixmode = 0) or less than 1 (ixmode != 0). iwgmode (optional, default=0) -- Wrap and guardpoint mode. 0 = Limit mode. 1 = Wrap mode. 2 = Guardpoint mode. Performance Limit mode (0) Limit the total index (ndx + ixoff) to between 0 and the guard point. For a table of length 5, this means that locations 0 to 3 and location 4 (the guard point) can be written. A negative total index writes to location 0. Wrap mode (1) Wrap total index value into locations 0 to E, where E is either one less than the table length or the factor of 2 number which is one less than the table length. For example, wrap into a 0 to 3 range - so that total index 6 writes to location 2. Guardpoint mode (2) The guardpoint is written at the same time as location 0 is written - with the same value. This facilitates writing to tables which are intended to be read with interpolation for producing smooth cyclic waveforms. In addition, before it is used, the total index is incremented by half the range between one location and the next, before being rounded down to the integer address of a table location. Normally (igwmode = 0 or 1) for a table of length 5 - which has locations 0 to 3 as the main table and location 4 as the guard point, a total index in the range of 0 to 0.999 will write to location 0. ("0.999" means just less than 1.0.) 1.0 to 1.999 will write to location 1 etc. A similar pattern holds for all total indexes 0 to 4.999 (igwmode = 0) or to 3.999 (igwmode = 1). igwmode = 0 enables locations 0 to 4 to be written - with the guardpoint (4) being written with a potentially different value from location 0. With a table of length 5 and the iwgmode = 2, then when the total index is in the range 0 to 0.499, it will write to locations 0 and 4. Range 0.5 to 1.499 will write to location 1 etc. 3.5 to 4.0 will also write to locations 0 and 4. This way, the writing operation most closely approximates the results of interpolated reading. Guard point mode should only be used with tables that have a guardpoint. Guardpoint mode is accomplished by adding 0.5 to the total index, rounding to the next lowest integer, wrapping it modulo the factor of two which is one less than the table length, writing the table (locations 0 to 3 in our example) and then writing to the guard point if index = 0. tablew has no output value. The last three parameters are optional and have default values of 0. Caution with k-rate table numbers At k-rate or a-rate, if a table number of < 1 is given, or the table number points to a non-existent table, or to one which has a length of 0 (it is to be loaded from a file later) then an error will result and the instrument will be deactivated. kfn and afn must be initialized at the appropriate rate using init. Attempting to load an i-rate value into kfn or afn will result in an error. Note that tablew is always a k-rate opcode. This means that even its i-rate version runs at k-rate and will write the value of the i-rate variable. For this reason, the following code will not work as expected: <CsoundSynthesizer> <CsOptions> </CsOptions> <CsInstruments> giFt ftgen 1, 0, 8, 2, 0 instr 1 indx = 0 tablew 10, indx, giFt ival tab_i indx, giFt print ival endin </CsInstruments> <CsScore> i 1 0 1 </CsScore> </CsoundSynthesizer> Although it may seem this program should print a 10 to the console. It will print 0, because tab_i will read the value at the initialization of the note, before the first performance pass, when tablew writes its value. See Also tableiw, tablewkt Credits Author: Robin Whittle Australia May 1997 manual_src~dfsg/opcodes/repluck.xml0000644000000000000000000000623112262561504016572 0ustar rootroot repluck Signal Generators:Waveguide Physical Modeling repluck repluck Physical model of the plucked string. Description repluck is an implementation of the physical model of the plucked string. A user can control the pluck point, the pickup point, the filter, and an additional audio signal, axcite. axcite is used to excite the 'string'. Based on the Karplus-Strong algorithm. Syntax ares repluck iplk, kamp, icps, kpick, krefl, axcite Initialization iplk -- The point of pluck is iplk, which is a fraction of the way up the string (0 to 1). A pluck point of zero means no initial pluck. icps -- The string plays at icps pitch. Performance kamp -- Amplitude of note. kpick -- Proportion of the way along the string to sample the output. krefl -- the coefficient of reflection, indicating the lossiness and the rate of decay. It must be strictly between 0 and 1 (it will complain about both 0 and 1). Performance axcite -- A signal which excites the string. Examples Here is an example of the repluck opcode. It uses the file repluck.csd. Example of the repluck opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also wgpluck2 Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK 1997 New in version 3.47 manual_src~dfsg/opcodes/kon.xml0000644000000000000000000000074312262561504015716 0ustar rootroot kon kon kon Deprecated. Description Deprecated as of version 3.49. Use the midion opcode instead. manual_src~dfsg/opcodes/rezzy.xml0000644000000000000000000000545112262561504016313 0ustar rootroot rezzy Signal Modifiers:Standard Filters:Resonant rezzy rezzy A resonant low-pass filter. Description A resonant low-pass filter. Syntax ares rezzy asig, xfco, xres [, imode, iskip] Initialization imode (optional, default=0) -- high-pass or low-pass mode. If zero, rezzy is low-pass. If not zero, rezzy is high-pass. Default value is 0. (New in Csound version 3.50) iskip (optional, default=0) -- if non zero skip the initialisation of the filter. (New in Csound version 4.23f13 and 5.0) Performance asig -- input signal xfco -- filter cut-off frequency in Hz. As of version 3.50, may i-,k-, or a-rate. xres -- amount of resonance. Values of 1 to 100 are typical. Resonance should be one or greater. As of version 3.50, may a-rate, i-rate, or k-rate. rezzy is a resonant low-pass filter created empirically by &namehans;. Examples Here is an example of the rezzy opcode. It uses the file rezzy.csd. Example of the rezzy opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also biquad, moogvcf Credits Author: &namehans; October 1998 New in Csound version 3.49 manual_src~dfsg/opcodes/phaser2.xml0000644000000000000000000002040712262561504016472 0ustar rootroot phaser2 Signal Modifiers:Special Effects phaser2 phaser2 Second-order allpass filters arranged in a series. Description An implementation of iord number of second-order allpass filters in series. Syntax ares phaser2 asig, kfreq, kq, kord, kmode, ksep, kfeedback Performance kfreq -- frequency (in Hz) of the filter(s). This is the center frequency of the notch of the first allpass filter in the series. This frequency is used as the base frequency from which the frequencies of the other notches are derived. kq -- Q of each notch. Higher Q values result in narrow notches. A Q between 0.5 and 1 results in the strongest "phasing" effect, but higher Q values can be used for special effects. kord -- the number of allpass stages in series. These are second-order filters, and iord can range from 1 to 2499. With higher orders, the computation time increases. kfeedback -- amount of the output which is fed back into the input of the allpass chain. With larger amounts of feedback, more prominent notches appear in the spectrum of the output. kfeedback must be between -1 and +1. for stability. kmode -- used in calculation of notch frequencies. Note Although kord and kmode are listed as k-rate, they are in fact accessed only at init-time. So if you are using k-rate arguments, they must be assigned with init. ksep -- scaling factor used, in conjunction with imode, to determine the frequencies of the additional notches in the output spectrum. phaser2 implements iord number of second-order allpass sections, connected in series. The use of second-order allpass sections allows for the precise placement of the frequency, width, and depth of notches in the frequency spectrum. iord is used to directly determine the number of notches in the spectrum; e.g. for iord = 6, there will be 6 notches in the output spectrum. There are two possible modes for determining the notch frequencies. When imode = 1, the notch frequencies are determined by the following function: frequency of notch N = kbf + (ksep * kbf * N-1) For example, with imode = 1 and ksep = 1, the notches will be in harmonic relationship with the notch frequency determined by kfreq (i.e. if there are 8 notches, with the first at 100 Hz, the next notches will be at 200, 300, 400, 500, 600, 700, and 800 Hz). This is useful for generating a "comb filtering" effect, with the number of notches determined by iord. Different values of ksep allow for inharmonic notch frequencies and other special effects. ksep can be swept to create an expansion or contraction of the notch frequencies. A useful visual analogy for the effect of sweeping ksep would be the bellows of an accordion as it is being played - the notches will be seperated, then compressed together, as ksep changes. When imode = 2, the subsequent notches are powers of the input parameter ksep times the initial notch frequency specified by kfreq. This can be used to set the notch frequencies to octaves and other musical intervals. For example, the following lines will generate 8 notches in the output spectrum, with the notches spaced at octaves of kfreq: aphs phaser2 ain, kfreq, 0.5, 8, 2, 2, 0 aout = ain + aphs When imode = 2, the value of ksep must be greater than 0. ksep can be swept to create a compression and expansion of notch frequencies (with more dramatic effects than when imode = 1). Examples Here is an example of the phaser2 opcode. It uses the file phaser2.csd. Example of the phaser2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Technical History A general description of the differences between flanging and phasing can be found in Hartmann [1]. An early implementation of first-order allpass filters connected in series can be found in Beigel [2], where the bilinear z-transform is used for determining the phase shift frequency of each stage. Cronin [3] presents a similar implementation for a four-stage phase shifting network. Chamberlin [4] and Smith [5] both discuss using second-order allpass sections for greater control over notch depth, width, and frequency. References Hartmann, W.M. "Flanging and Phasers." Journal of the Audio Engineering Society, Vol. 26, No. 6, pp. 439-443, June 1978. Beigel, Michael I. "A Digital 'Phase Shifter' for Musical Applications, Using the Bell Labs (Alles-Fischer) Digital Filter Module." Journal of the Audio Engineering Society, Vol. 27, No. 9, pp. 673-676,September 1979. Cronin, Dennis. "Examining Audio DSP Algorithms." Dr. Dobb's Journal, July 1994, p. 78-83. Chamberlin, Hal. Musical Applications of Microprocessors. Second edition. Indianapolis, Indiana: Hayden Books, 1985. Smith, Julius O. "An Allpass Approach to Digital Phasing and Flanging." Proceedings of the 1984 ICMC, p. 103-108. See Also phaser1 Other information about phasers on Wikipedia: http://en.wikipedia.org/wiki/Phaser_(effect) Credits Author: &namesean; Seattle, Washington 1999 November 2002. Added a note about the kord and kmode parameters, thanks to &namerasmus;. New in Csound version 4.0 manual_src~dfsg/opcodes/vbapmove.xml0000644000000000000000000001437512262561504016754 0ustar rootroot vbapmove Signal Modifiers:Panning and Spatialization vbapmove vbapmove Distributes an audio signal among many channels with moving virtual sources. Description Distributes an audio signal among upto 64 channels with moving virtual sources. Syntax ar1[, ar2...] vbapmove asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] aarray[] vbapmove asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] Initialization idur -- the duration over which the movement takes place. ispread -- spreading of the virtual source (range 0 - 100). If value is zero, conventional amplitude panning is used. When ispread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. ifldnum -- number of fields (absolute value must be 2 or larger). If ifldnum is positive, the virtual source movement is a polyline specified by given directions. Each transition is performed in an equal time interval. If ifldnum is negative, specified angular velocities are applied to the virtual source during specified relative time intervals (see below). ifld1, ifld2, ... -- azimuth angles or angular velocities, and relative durations of movement phases (see below). Performance asig -- audio signal to be panned vbapmove allows the use of moving virtual sources. If ifldnum is positive, the fields represent directions of virtual sources and equal times, iazi1, [iele1,] iazi2, [iele2,], etc. The position of the virtual source is interpolated between directions starting from the first direction and ending at the last. Each interval is interpolated in time that is fraction total_time / number_of_intervals of the duration of the sound event. If ifldnum is negative, the fields represent angular velocities and equal times. The first field is, however, the starting direction, iazi1, [iele1,] iazi_vel1, [iele_vel1,] iazi_vel2, [iele_vel2,] .... Each velocity is applied to the note that is fraction total_time / number_of_velocities of the duration of the sound event. If the elevation of the virtual source becomes greater than 90 degrees or less than 0 degrees, the polarity of angular velocity is changed. Thus the elevational angular velocity produces a virtual source that moves up and down between 0 and 90 degrees. Please note that all vbap panning opcodes require the vbap system to be initialized using vbaplsinit. Examples See the entry for vbap8move for an example of usage of the vbapXmove opcodes. Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. Examples Here is an example of the vbapmove opcode. It uses the file vbapmove.csd. Example of the vbapmove opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vbap, vbap16, vbap16move, vbap4, vbap4move, vbap8, vbap8move, vbaplsinit, vbapz, vbapzmove Credits Author: Ville Pulkki Sibelius Academy Computer Music Studio Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Helsinki, Finland May 2000 Author: &namejohn; July 2012, Sept 2013 New in Csound Version 5.17.13 Array outputs new in Csound Version 6.01 manual_src~dfsg/opcodes/printf.xml0000644000000000000000000000715512262561504016435 0ustar rootroot printf_i printf Signal I/O:Printing and Display printf printf printf-style formatted output Description printf and printf_i write formatted output, similarly to the C function printf(). printf_i runs at i-time only, while printf runs both at initialization and performance time. Syntax printf_i Sfmt, itrig, [iarg1[, iarg2[, ... ]]] printf Sfmt, ktrig, [xarg1[, xarg2[, ... ]]] Initialization Sfmt -- format string, has the same format as in printf() and other similar C functions, except length modifiers (l, ll, h, etc.) are not supported. The following conversion specifiers are allowed: d, i, o, u, x, X, e, E, f, F, g, G, c, s iarg1, iarg2, ... -- input arguments (max. 30) for format. Integer formats like %d round the input values to the nearest integer. itrig -- if greater than zero the opcode performs the printing; otherwise it is an null operation. Performance ktrig -- if greater than zero and different from the value on the previous control cycle the opcode performs the requested printing. Initially this previous value is taken as zero. xarg1, xarg2, ... -- input arguments (max. 30) for format. Integer formats like %d round the input values to the nearest integer. Note that only k-rate and i-rate arguments are valid (no a-rate printing) Examples Here is an example of the printf opcode. It uses the file printf.csd. Example of the printf opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example will produce the following output: Audiofile 'frox.wav' does not exist! See Also More information about printf: http://www.cplusplus.com/reference/clibrary/cstdio/printf/ Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/fofilter.xml0000644000000000000000000000567112262561504016746 0ustar rootroot fofilter Signal Modifiers:Specialized Filters fofilter fofilter Formant filter. Description Fofilter generates a stream of overlapping sinewave grains, when fed with a pulse train. Each grain is the impulse response of a combination of two BP filters. The grains are defined by their attack time (determining the skirtwidth of the formant region at -60dB) and decay time (-6dB bandwidth). Overlapping will occur when 1/freq < decay, but, unlike FOF, there is no upper limit on the number of overlaps. The original idea for this opcode came from J McCartney's formlet class in SuperCollider, but this is possibly implemented differently(?). Syntax asig fofilter ain, xcf, xris, xdec[, istor] Initialization istor --initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance asig -- input signal. xcf -- filter centre frequency xris -- impulse response attack time (secs). xdec -- impulse response decay time (secs). Examples Here is an example of the fofilter opcode. It uses the file fofilter.csd. Example of the fofilter opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; January 2005 New plugin in version 5 Audio rate parameters introduced in version 6.02 November 2013. manual_src~dfsg/opcodes/flexecbutton.xml0000644000000000000000000000754312262561504017636 0ustar rootroot FLexecButton FLTK:Other FLexecButton FLexecButton A FLTK widget opcode that creates a button that executes a command. Description A FLTK widget opcode that creates a button that executes a command. Useful for opening up HTML documentation as About text or to start a separate program from an FLTK widget interface. Warning Because any command can be executed, the user is advised to be very careful when using this opcode and when running orchestras by others using this opcode. Syntax ihandle FLexecButton "command", iwidth, iheight, ix, iy Initialization ihandle -- a handle value (an integer number) that unequivocally references a corresponding widget. This is used by other opcodes that modify a widget's properties (see Modifying FLTK Widget Appearance). It is automatically output by FLexecButton. command -- a double-quoted string containing a command to execute. Notice that with FLexecButton, the default text for the button is "About" and it is necessary to call the FLsetText opcode to change the text of the button. iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). Examples Here is an example of the FLexecButton opcode. It uses the file FLexecButton.csd. Example of the FLexecButton opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLbutton, FLbox, FLbutBank, FLprintk, FLprintk2, FLvalue Credits Author: &namesteven; Example by: Jonathan Murphy New in version 5.05 manual_src~dfsg/opcodes/platerev.xml0000644000000000000000000000712712262561504016754 0ustar rootroot platerev Signal Modifiers:Reverberation platerev platerev Models the reverberation of a metal plate. Description Models the reverberation of a rectangular metal plate with settable physical characteristics when excited by audio signal(s). Syntax a1[, a2, ...] platerev itabexcite. itabouts, kbndry, iaspect, istiff, idecay, iloss, aexcite1[, aexcite2, ...] Initialization itabexcite -- number of a table that contains triplets for each excitation signal, (frequency, radius, initial phase in radians). The radius should be less than 1. These control where the excitation happens. The values in the table for frequency and radius may be changed in performance, with the proviso about clicking if the changes are too large. itabouts -- number of a table that contains triplets for each output signal, (frequency, radius, initial phase in radians). See itabexcite description. kbndry -- boundary condition of the plate; 0 = free, 1 = clamped, 2 = pivoting. Other values are undefined. Thus parameter may be changed at k-rate but that might give rise to clicks. iaspect -- plate aspect ratio which should be less than or equal to 1. istiff -- plate stiffness parameter (set around 1 or less for plate reverb). idecay -- time taken for 30 dB decay iloss -- loss parameter for high-frequency damping (value about 0.001 suggested). Performance A metal plate is excited by a stereo signal and the resulting reverberation captured. aexciten -- excitation signal to be inserted into the plate. Examples Here is an example of the platerev opcode. It uses the file plate.csd. Example of the platerev opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Stefan Bilbao University of Edinburgh, UK Author: &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 5.17.12 manual_src~dfsg/opcodes/flslidbnk.xml0000644000000000000000000002335312262561504017101 0ustar rootroot FLslidBnk FLTK:Other FLslidBnk FLslidBnk A FLTK widget containing a bank of horizontal sliders. Description FLslidBnk is a widget containing a bank of horizontal sliders. Syntax FLslidBnk "names", inumsliders [, ioutable] [, iwidth] [, iheight] [, ix] \ [, iy] [, itypetable] [, iexptable] [, istart_index] [, iminmaxtable] Initialization names -- a double-quoted string containing the names of each slider. Each slider can have a different name. Separate each name with @ character, for example: frequency@amplitude@cutoff. It is possible to not provide any name by giving a single space . In this case, the opcode will automatically assign a progressive number as a label for each slider. inumsliders -- the number of sliders. ioutable (optional, default=0) -- number of a previously-allocated table in which to store output values of each slider. The user must be sure that table size is large enough to contain all output cells, otherwise a segfault will crash Csound. By assigning zero to this argument, the output will be directed to the zak space in the k-rate zone. In this case, the zak space must be previously allocated with the zakinit opcode and the user must be sure that the allocation size is big enough to cover all sliders. The default value is zero (i.e. store output in zak space). istart_index (optional, default=0) -- an integer number referring to a starting offset of output cell locations. It can be positive to allow multiple banks of sliders to output in the same table or in the zak space. The default value is zero (no offset). iminmaxtable (optional, default=0) -- number of a previously-defined table containing a list of min-max pairs, referred to each slider. A zero value defaults to the 0 to 1 range for all sliders without necessity to provide a table. The default value is zero. iexptable (optional, default=0) -- number of a previously-defined table containing a list of identifiers (i.e. integer numbers) provided to modify the behaviour of each slider independently. Identifiers can assume the following values: -1 -- exponential curve response 0 -- linear response number > than 0 -- follow the curve of a previously-defined table to shape the response of the corresponding slider. In this case, the number corresponds to table number. You can assume that all sliders of the bank have the same response curve (exponential or linear). In this case, you can assign -1 or 0 to iexptable without worrying about previously defining any table. The default value is zero (all sliders have a linear response, without having to provide a table). itypetable (optional, default=0) -- number of a previously-defined table containing a list of identifiers (i.e. integer numbers) provided to modify the aspect of each individual slider independently. Identifiers can assume the following values: 0 = Nice slider 1 = Fill slider 3 = Normal slider 5 = Nice slider 7 = Nice slider with down-box You can assume that all sliders of the bank have the same aspect. In this case, you can assign a negative number to itypetable without worrying about previously defining any table. Negative numbers have the same meaning of the corresponding positive identifiers with the difference that the same aspect is assigned to all sliders. You can also assign a random aspect to each slider by setting itypetable to a negative number lower than -7. The default value is zero (all sliders have the aspect of nice sliders, without having to provide a table). You can add 20 to a value inside the table to make the slider "plastic", or subtract 20 if you want to set the value for all widgets without defining a table (e.g. -21 to set all sliders types to Plastic Fill slider). iwidth (optional) -- width of the rectangular area containing all sliders of the bank, excluding text labels, that are placed to the left of that area. iheight (optional) -- height of the rectangular area containing all sliders of the bank, excluding text labels, that are placed to the left of that area. ix (optional) -- horizontal position of the upper left corner of the rectangular area containing all sliders belonging to the bank. You have to leave enough space, at the left of that rectangle, in order to make sure labels of sliders to be visible. This is because the labels themselves are external to the rectangular area. iy (optional) -- vertical position of the upper left corner of the rectangular area containing all sliders belonging to the bank. You have to leave enough space, at the left of that rectangle, in order to make sure labels of sliders to be visible. This is because the labels themselves are external to the rectangular area. Performance There are no k-rate arguments, even if cells of the output table (or the zak space) are updated at k-rate. FLslidBnk is a widget containing a bank of horizontal sliders. Any number of sliders can be placed into the bank (inumsliders argument). The output of all sliders is stored into a previously allocated table or into the zak space (ioutable argument). It is possible to determine the first location of the table (or of the zak space) in which to store the output of the first slider by means of istart_index argument. Each slider can have an individual label that is placed to the left of it. Labels are defined by the names argument. The output range of each slider can be individually set by means of an external table (iminmaxtable argument). The curve response of each slider can be set individually, by means of a list of identifiers placed in a table (iexptable argument). It is possible to define the aspect of each slider independently or to make all sliders have the same aspect (itypetable argument). The iwidth, iheight, ix, and iy arguments determine width, height, horizontal and vertical position of the rectangular area containing sliders. Notice that the label of each slider is placed to the left of them and is not included in the rectangular area containing sliders. So the user should leave enough space to the left of the bank by assigning a proper ix value in order to leave labels visible. IMPORTANT! Notice that the tables used by FLslidBnk must be created with the ftgen opcode and placed in the orchestra before the corresponding valuator. They can not placed in the score. This is because tables placed in the score are created later than the initialization of the opcodes placed in the header section of the orchestra. Examples Here is an example of the FLslidBnk opcode. It uses the file FLslidBnk.csd. Example of the FLslidBnk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLslider, FLslidBnk2, FLvslidBnk, FLvslidBnk2 Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/median.xml0000644000000000000000000000515412262561504016365 0ustar rootroot median Signal Modifiers:Standard Filters median median A median filter, a variant FIR lowpass filter. Description Implementation of a median filter. Syntax ares median asig, ksize, imaxsize [, iskip] Initialization imaxsize -- the maximun size of the window used to select the data. iskip -- initial disposition of internal data space. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance asig -- input signal to be filtered ksize -- size of the window over which the input is to be filtered. It must not exceed the maximum window size; if it does it is truncated. median is a simple filter that retuns the median value of the last ksize values. It has a lowpass action. The efficiency decreases as the window size increases. Examples Here is an example of the median opcode. It uses the file median.csd. Example of the median opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; University of Bath May 2010 New in Csound version 5.13 manual_src~dfsg/opcodes/opbitnot.xml0000644000000000000000000000265012262561504016764 0ustar rootroot ¬ Mathematical Operations:Arithmetic and Logic Operations ¬ ¬ Bitwise NOT operator. Description The bitwise operators perform operations of bitwise AND, bitwise OR, bitwise NOT and bitwise non-equivalence. The priority of these operators is less binding that the arithmetic ones, but more binding that the comparisons. Parentheses may be used as above to force particular groupings. Syntax ~ a (bitwise NOT) where the argument a may be a further expression. It is converted to the nearest integer to machine precision and then the operation is performed. See Also &, | # manual_src~dfsg/opcodes/phaser1.xml0000644000000000000000000001534312262561504016474 0ustar rootroot phaser1 Signal Modifiers:Special Effects phaser1 phaser1 First-order allpass filters arranged in a series. Description An implementation of iord number of first-order allpass filters in series. Syntax ares phaser1 asig, kfreq, kord, kfeedback [, iskip] Initialization iskip (optional, default=0) -- used to control initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance kfreq -- frequency (in Hz) of the filter(s). This is the frequency at which each filter in the series shifts its input by 90 degrees. kord -- the number of allpass stages in series. These are first-order filters and can range from 1 to 4999. Note Although kord is listed as k-rate, it is in fact accessed only at init-time. So if you are using a k-rate argument, it must be assigned with init. kfeedback -- amount of the output which is fed back into the input of the allpass chain. With larger amounts of feedback, more prominent notches appear in the spectrum of the output. kfeedback must be between -1 and +1. for stability. phaser1 implements iord number of first-order allpass sections, serially connected, all sharing the same coefficient. Each allpass section can be represented by the following difference equation: y(n) = C * x(n) + x(n-1) - C * y(n-1) where x(n) is the input, x(n-1) is the previous input, y(n) is the output, y(n-1) is the previous output, and C is a coefficient which is calculated from the value of kfreq, using the bilinear z-transform. By slowly varying kfreq, and mixing the output of the allpass chain with the input, the classic "phase shifter" effect is created, with notches moving up and down in frequency. This works best with iord between 4 and 16. When the input to the allpass chain is mixed with the output, 1 notch is generated for every 2 allpass stages, so that with iord = 6, there will be 3 notches in the output. With higher values for iord, modulating kfreq will result in a form of nonlinear pitch modulation. Examples Here is an example of the phaser1 opcode. It uses the file phaser1.csd. Example of the phaser1 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Technical History A general description of the differences between flanging and phasing can be found in Hartmann [1]. An early implementation of first-order allpass filters connected in series can be found in Beigel [2], where the bilinear z-transform is used for determining the phase shift frequency of each stage. Cronin [3] presents a similar implementation for a four-stage phase shifting network. Chamberlin [4] and Smith [5] both discuss using second-order allpass sections for greater control over notch depth, width, and frequency. References Hartmann, W.M. "Flanging and Phasers." Journal of the Audio Engineering Society, Vol. 26, No. 6, pp. 439-443, June 1978. Beigel, Michael I. "A Digital 'Phase Shifter' for Musical Applications, Using the Bell Labs (Alles-Fischer) Digital Filter Module." Journal of the Audio Engineering Society, Vol. 27, No. 9, pp. 673-676,September 1979. Cronin, Dennis. "Examining Audio DSP Algorithms." Dr. Dobb's Journal, July 1994, p. 78-83. Chamberlin, Hal. Musical Applications of Microprocessors. Second edition. Indianapolis, Indiana: Hayden Books, 1985. Smith, Julius O. "An Allpass Approach to Digital Phasing and Flanging." Proceedings of the 1984 ICMC, p. 103-108. See Also phaser2 Other information about phasers on Wikipedia: http://en.wikipedia.org/wiki/Phaser_(effect) Credits Author: &namesean; Seattle, Washington 1999 November 2002. Added a note about the kord parameter, thanks to &namerasmus;. New in Csound version 4.0 manual_src~dfsg/opcodes/flanger.xml0000644000000000000000000000567512262561504016556 0ustar rootroot flanger Signal Modifiers:Special Effects flanger flanger A user controlled flanger. Description A user controlled flanger. Syntax ares flanger asig, adel, kfeedback [, imaxd] Initialization imaxd(optional) -- maximum delay in seconds (needed for inital memory allocation) Performance asig -- input signal adel -- delay in seconds kfeedback -- feedback amount (in normal tasks this should not exceed 1, even if bigger values are allowed) This unit is useful for generating choruses and flangers. The delay must be varied at a-rate, for example by connecting adel to an oscillator output. The feedback can vary at k-rate. This opcode is implemented to allow kr different than sr (else delay could not be lower than ksmps) enhancing realtime performance. This unit is very similar to wguide1, the only difference is flanger does not have the lowpass filter or the requirement that the delay be varied at a-rate. Examples Here is an example of the flanger opcode. It uses the file flanger.csd. Example of the flanger opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also More information on flanging on Wikipedia: http://en.wikipedia.org/wiki/Flanger Credits Author: &namegabriel; Italy New in Csound version 3.49 manual_src~dfsg/opcodes/setctrl.xml0000644000000000000000000000676712262561504016623 0ustar rootroot setctrl Instrument Control:Sensing and Control setctrl setctrl Configurable slider controls for realtime user input. Description Configurable slider controls for realtime user input. Requires Winsound or TCL/TK. setctrl sets a slider to a specific value, or sets a minimum or maximum range. Syntax setctrl inum, ival, itype Initialization Note that this opcode is not available on Windows due to the implimentation of pipes on that system inum -- number of the slider to set ival -- value to be sent to the slider itype -- type of value sent to the slider as follows: 1 -- set the current value. Initial value is 0. 2 -- set the minimum value. Default is 0. 3 -- set the maximum value. Default is 127. 4 -- set the label. (New in Csound version 4.09) Performance Calling setctrl will create a new slider on the screen. There is no theoretical limit to the number of sliders. Windows and TCL/TK use only integers for slider values, so the values may need rescaling. GUIs usually pass values at a fairly slow rate, so it may be advisable to pass the output of control through port. Examples Here is an example of the setctrl opcode. It uses the file setctrl.csd. Example of the setctrl opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: i1 38.00000 i1 40.00000 i1 43.00000 See Also control Credits Author: &namejohn; University of Bath, Codemist. Ltd. Bath, UK May 2000 Example written by &namekevin;. New in Csound version 4.06 manual_src~dfsg/opcodes/vibrato.xml0000644000000000000000000001043012262561504016567 0ustar rootroot vibrato Signal Generators:Basic Oscillators vibrato vibrato Generates a natural-sounding user-controllable vibrato. Description Generates a natural-sounding user-controllable vibrato. Syntax kout vibrato kAverageAmp, kAverageFreq, kRandAmountAmp, kRandAmountFreq, kAmpMinRate, kAmpMaxRate, kcpsMinRate, kcpsMaxRate, ifn [, iphs Initialization ifn -- Number of vibrato table. It normally contains a sine or a triangle wave. iphs -- (optional) Initial phase of table, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is 0. Performance kAverageAmp -- Average amplitude value of vibrato kAverageFreq -- Average frequency value of vibrato (in cps) kRandAmountAmp -- Amount of random amplitude deviation kRandAmountFreq -- Amount of random frequency deviation kAmpMinRate -- Minimum frequency of random amplitude deviation segments (in cps) kAmpMaxRate -- Maximum frequency of random amplitude deviation segments (in cps) kcpsMinRate -- Minimum frequency of random frequency deviation segments (in cps) kcpsMaxRate -- Maximum frequency of random frequency deviation segments (in cps) vibrato outputs a natural-sounding user-controllable vibrato. The concept is to randomly vary both frequency and amplitude of the oscillator generating the vibrato, in order to simulate the irregularities of a real vibrato. In order to have a total control of these random variations, several input arguments are present. Random variations are obtained by two separated segmented lines, the first controlling amplitude deviations, the second the frequency deviations. Average duration of each segment of each line can be shortened or enlarged by the arguments kAmpMinRate, kAmpMaxRate, kcpsMinRate, kcpsMaxRate, and the deviation from the average amplitude and frequency values can be independently adjusted by means of kRandAmountAmp and kRandAmountFreq. Examples Here is an example of the vibrato opcode. It uses the file vibrato.csd. Example of the vibrato opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also jitter, jitter2, vibr Credits Author: &namegabriel; New in Version 4.15 manual_src~dfsg/opcodes/dollar.xml0000644000000000000000000001106212262561504016400 0ustar rootroot macrosorchestra $NAMEorchestra $NAME Orchestra Syntax:Macros $NAME $NAME Calls a defined macro. Description Macros are textual replacements which are made in the orchestra as it is being read. The macro system in Csound is a very simple one, and uses the characters # and $ to define and call macros. This can save typing, and can lead to a coherent structure and consistent style. This is similar to, but independent of, the macro system in the score language. $NAME -- calls a defined macro. To use a macro, the name is used following a $ character. The name is terminated by the first character which is neither a letter nor a number. If it is necessary for the name not to terminate with a space, a period, which will be ignored, can be used to terminate the name. The string, $NAME., is replaced by the replacement text from the definition. The replacement text can also include macro calls. Syntax $NAME  Initialization # replacement text # -- The replacement text is any character string (not containing a #) and can extend over mutliple lines. The replacement text is enclosed within the # characters, which ensure that additional characters are not inadvertently captured. Performance Some care is needed with textual replacement macros, as they can sometimes do strange things. They take no notice of any meaning, so spaces are significant. This is why, unlike the C programming language, the definition has the replacement text surrounded by # characters. Used carefully, this simple macro system is a powerful concept, but it can be abused. Examples Here is an example of the calling a macro. It uses the file define.csd. An example of the calling a macro. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: Macro definition for VOLUME Macro definition for CPS Macro definition for TABLE Here is an example of the calling a macro with arguments. It uses the file define_args.csd. An example of the calling a macro with arguments. Its output should include a line like this: Macro definition for OSCMACRO See Also #define, #undef Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April, 1998 Examples written by &namekevin;. New in Csound version 3.48 manual_src~dfsg/opcodes/kpcauchy.xml0000644000000000000000000000077612262561504016744 0ustar rootroot kpcauchy kpcauchy kpcauchy Deprecated. Description Deprecated as of version 3.49. Use the pcauchy opcode instead. manual_src~dfsg/opcodes/strcatk.xml0000644000000000000000000000433312262561504016601 0ustar rootroot strcatk Strings:Manipulation strcatk strcatk Concatenate strings (k-rate) Description Concatenate two strings and store the result in a variable. strcatk does the concatenation both at initialization and performance time. It is allowed for any of the input arguments to be the same as the output variable. Syntax Sdst strcatk Ssrc1, Ssrc2 Examples Here is an example of the strcatk opcode. It uses the file strcatk.csd. Example of the strcatk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: 1 + 1 = 2 1 + 1 + 1 = 3 1 + 1 + 1 + 1 = 4 1 + 1 + 1 + 1 + 1 = 5 1 + 1 + 1 + 1 + 1 + 1 = 6 1 + 1 + 1 + 1 + 1 + 1 + 1 = 7 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 = 8 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 = 9 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 = 10 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 = 11 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 = 12 See also strcat Credits Author: Istvan Varga 2005 New in version 5.02 manual_src~dfsg/opcodes/fof.xml0000644000000000000000000001465212262561504015705 0ustar rootroot fof Signal Generators:Granular Synthesis fof fof Produces sinusoid bursts useful for formant and granular synthesis. Description Audio output is a succession of sinusoid bursts initiated at frequency xfund with a spectral peak at xform. For xfund above 25 Hz these bursts produce a speech-like formant with spectral characteristics determined by the k-input parameters. For lower fundamentals this generator provides a special form of granular synthesis. Syntax ares fof xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, \ ifna, ifnb, itotdur [, iphs] [, ifmode] [, iskip] Initialization iolaps -- number of preallocated spaces needed to hold overlapping burst data. Overlaps are frequency dependent, and the space required depends on the maximum value of xfund * kdur. Can be over-estimated at no computation cost. Uses less than 50 bytes of memory per iolap. ifna, ifnb -- table numbers of two stored functions. The first is a sine table for sineburst synthesis (size of at least 4096 recommended). The second is a rise shape, used forwards and backwards to shape the sineburst rise and decay; this may be linear (GEN07) or perhaps a sigmoid (GEN19). itotdur -- total time during which this fof will be active. Normally set to p3. No new sineburst is created if it cannot complete its kdur within the remaining itotdur. iphs (optional, default=0) -- initial phase of the fundamental, expressed as a fraction of a cycle (0 to 1). The default value is 0. ifmode (optional, default=0) -- formant frequency mode. If zero, each sineburst keeps the xform frequency it was launched with. If non-zero, each is influenced by xform continuously. The default value is 0. iskip (optional, default=0) -- If non-zero, skip initialisation (allows legato use). Performance xamp -- peak amplitude of each sineburst, observed at the true end of its rise pattern. The rise may exceed this value given a large bandwidth (say, Q < 10) and/or when the bursts are overlapping. xfund -- the fundamental frequency (in Hertz) of the impulses that create new sinebursts. xform -- the formant frequency, i.e. freq of the sinusoid burst induced by each xfund impulse. This frequency can be fixed for each burst or can vary continuously (see ifmode). koct -- octaviation index, normally zero. If greater than zero, lowers the effective xfund frequency by attenuating odd-numbered sinebursts. Whole numbers are full octaves, fractions transitional. kband -- the formant bandwidth (at -6dB), expressed in Hz. The bandwidth determines the rate of exponential decay throughout the sineburst, before the enveloping described below is applied. kris, kdur, kdec -- rise, overall duration, and decay times (in seconds) of the sinusoid burst. These values apply an enveloped duration to each burst, in similar fashion to a Csound linen generator but with rise and decay shapes derived from the ifnb input. kris inversely determines the skirtwidth (at -40 dB) of the induced formant region. kdur affects the density of sineburst overlaps, and thus the speed of computation. Typical values for vocal imitation are .003,.02,.007. Csound's fof generator is loosely based on Michael Clarke's C-coding of IRCAM's CHANT program (Xavier Rodet et al.). Each fof produces a single formant, and the output of four or more of these can be summed to produce a rich vocal imitation. fof synthesis is a special form of granular synthesis, and this implementation aids transformation between vocal imitation and granular textures. Computation speed depends on kdur, xfund, and the density of any overlaps. Examples Here is an example of the fof opcode. It uses the file fof.csd. Example of the fof opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The formant values for the alto-"a" sound were taken from the Formant Values Appendix. See Also fof2, Formant Values Appendix Credits Added in version 1 (1990) manual_src~dfsg/opcodes/STKBandedWG.xml0000644000000000000000000000754012262561504017126 0ustar rootroot STKBandedWG Signal Generators:STK Opcodes STKBandedWG STKBandedWG STKBandedWG uses banded waveguide techniques to model a variety of sounds. Description This opcode uses banded waveguide techniques to model a variety of sounds, including bowed bars, glasses, and bowls. Syntax asignal STKBandedWG ifrequency, iamplitude, [kpress, kv1[, kmot, kv2[, klfo, kv3[, klfodepth, kv4[, kvel, kv5[, kstrk, kv6[, kinstr, kv7]]]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kpress -- controller 2, pressure of bow. Value range of kv1 is 0-127. kmot -- controller 4, motion of bow. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kvel -- controller 128, velocity of bow. Value range of kv5 is 0-127. kstrk -- controller 64, striking of bow. Value range of kv6 is 0-127. kinstr -- controller 16, instrument presets (0 = uniform bar, 1 = tuned bar, 2 = glass harmonica, 3 = Tibetan bowl). Value range of kv7 is 0-3. Notes The code for this opcode is taken directly from the BandedWG class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKBandedWG opcode. It uses the file STKBandedWG.csd. Example of the STKBandedWG opcode. Credits Author: &namemichael; (after Georg Essl) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/seed.xml0000644000000000000000000000517212262561504016050 0ustar rootroot seed Signal Generators:Random (Noise) Generators seed seed Sets the global seed value. Description Sets the global seed value for all x-class noise generators, as well as other opcodes that use a random call, such as grain. Please Note rand, randh, randi, rnd(x) and birnd(x) are not affected by seed. Syntax seed ival Performance Use of seed will provide predictable results from an orchestra using with random generators, when required from multiple performances. When specifying a seed value, ival should be an integer between 0 and 232. If ival = 0, the value of ival will be derived from the system clock. Examples Here is an example of the seed opcode. It uses the file seed.csd. Example of the seed opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i 1 time 0.00067: 100.00000 i 1 time 0.50000: 175.78677 i 1 time 1.00000: 170.89579 WARNING: Seeding from current time 834128659 i 2 time 2.00067: 100.00000 i 2 time 2.50000: 197.58517 i 2 time 3.00000: 188.69525 manual_src~dfsg/opcodes/tablecopy.xml0000644000000000000000000000653712262561504017120 0ustar rootroot tablecopy Table Control:Read/Write Opreations tablecopy tablecopy Simple, fast table copy opcode. Description Simple, fast table copy opcode. Syntax tablecopy kdft, ksft Performance kdft -- Destination function table. ksft -- Number of source function table. tablecopy -- Simple, fast table copy opcode. Takes the table length from the destination table, and reads from the start of the source table. For speed reasons, does not check the source length - just copies regardless - in wrap mode. This may read through the source table several times. A source table with length 1 will cause all values in the destination table to be written to its value. tablecopy cannot read or write the guardpoint. To read it use table, with ndx = the table length. Likewise use table write to write it. To write the guardpoint to the value in location 0, use tablegpw. This is primarily to change function tables quickly in a real-time situation. Examples Here is an example of the tablecopy opcode. It uses the file tablecopy.csd. Example of the tablecopy opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a lines like these: i1 5.00000 i1 3.00000 i1 2.00000 i1 4.00000 i1 7.00000 i1 6.00000 i1 7.00000 See Also tablegpw, tablemix, tableicopy, tableigpw, tableimix Credits Author: Robin Whittle Australia May 1997 New in version 3.47 manual_src~dfsg/opcodes/chnexport.xml0000644000000000000000000001232612262561504017141 0ustar rootroot chnexport Signal I/O:Software Bus chnexport chnexport Export a global variable as a channel of the bus. Description Export a global variable as a channel of the bus; the channel should not already exist, otherwise an init error occurs. This opcode is normally called from the orchestra header, and allows the host application to read or write orchestra variables directly, without having to use chnget or chnset to copy data. Syntax gival chnexport Sname, imode[, itype, idflt, imin, imax] gkval chnexport Sname, imode[, itype, idflt, imin, imax] gaval chnexport Sname, imode gSval chnexport Sname, imode Initialization imode -- sum of at least one of 1 for input and 2 for output. itype (optional, defaults to 0) -- channel subtype for control channels only. Possible values are: 0: default/unspecified (idflt, imin, and imax are ignored) 1: integer values only 2: linear scale 3: exponential scale idflt (optional, defaults to 0) -- default value, for control channels with non-zero itype only. Must be greater than or equal to imin, and less than or equal to imax. imin (optional, defaults to 0) -- minimum value, for control channels with non-zero itype only. Must be non-zero for exponential scale (itype = 3). imax (optional, defaults to 0) -- maximum value, for control channels with non-zero itype only. Must be greater than imin. In the case of exponential scale, it should also match the sign of imin. Notes The channel parameters (imode, itype, idflt, imin, and imax) are only hints for the host application or external software accessing the bus through the API, and do not actually restrict reading from or writing to the channel in any way. While the global variable is used as output argument, chnexport does not actually change it, and always runs at i-time only. If the variable is not previously declared, it is created by Csound with an initial value of zero or empty string. Example The example shows the software bus being used as an asynchronous control signal to select a filter cutoff. It assumes that an external program that has access to the API is feeding the values. sr = 44100 kr = 100 ksmps = 1 gkc init 1000 ; set default value gkc chnexport "cutoff", 1, 3, i(gkc), 500, 2000 instr 1 a1 oscil p4, p5, 100 a2 lowpass2 a1, gkc, 200 out a2 endin Credits Author: &nameistvan; 2005 manual_src~dfsg/opcodes/vbap8.xml0000644000000000000000000001103212262561504016140 0ustar rootroot vbap8 Signal Modifiers:Panning and Spatialization vbap8 vbap8 Distributes an audio signal among 8 channels. Description Distributes an audio signal among 8 channels. Syntax ar1, ..., ar8 vbap8 asig, kazim [, kelev] [, kspread] Performance asig -- audio signal to be panned kazim -- azimuth angle of the virtual source kelev (optional) -- elevation angle of the virtual source kspread (optional) -- spreading of the virtual source (range 0 - 100). If value is zero, conventional amplitude panning is used. When kspread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. vbap8 takes an input signal, asig, and distributes it among 8 outputs, according to the controls kazim and kelev, and the configured loudspeaker placement. If idim = 2, kelev is set to zero. The distribution is performed using Vector Base Amplitude Panning (VBAP - See reference). VBAP distributes the signal using loudspeaker data configured with vbaplsinit. The signal is applied to, at most, two loudspeakers in 2-D loudspeaker configurations, and three loudspeakers in 3-D loudspeaker configurations. If the virtual source is panned outside the region spanned by loudspeakers, the nearest loudspeakers are used in panning. Please note that all vbap panning opcodes require the vbap system to be initialized using vbaplsinit. Example Here is a simple example of the vbap8 opcode. It uses the file vbap8.csd. Example of the vbap8 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. See Also vbap16, vbap16move, vbap4, vbap4move, vbap8move, vbaplsinit, vbapz, vbapzmove Credits Author: Ville Pulkki Sibelius Academy Computer Music Studio Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Helsinki, Finland May 2000 New in Csound Version 4.07. Input parameters accept k-rate since Csund 5.09. manual_src~dfsg/opcodes/wgclar.xml0000644000000000000000000000673712262561504016417 0ustar rootroot wgclar Signal Generators:Waveguide Physical Modeling wgclar wgclar Creates a tone similar to a clarinet. Description Audio output is a tone similar to a clarinet, using a physical model developed from &nameperry;, but re-coded for Csound. Syntax ares wgclar kamp, kfreq, kstiff, iatt, idetk, kngain, kvibf, kvamp, ifn \ [, iminfreq] Initialization iatt -- time in seconds to reach full blowing pressure. 0.1 seems to correspond to reasonable playing. A longer time gives a definite initial wind sound. idetk -- time in seconds taken to stop blowing. 0.1 is a smooth ending ifn -- table of shape of vibrato, usually a sine table, created by a function iminfreq (optional) -- lowest frequency at which the instrument will play. If it is omitted it is taken to be the same as the initial kfreq. If iminfreq is negative, initialization will be skipped. Performance A note is played on a clarinet-like instrument, with the arguments as below. kamp -- Amplitude of note. kfreq -- Frequency of note played. kstiff -- a stiffness parameter for the reed. Values should be negative, and about -0.3. The useful range is approximately -0.44 to -0.18. kngain -- amplitude of the noise component, about 0 to 0.5 kvibf -- frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp -- amplitude of the vibrato Examples Here is an example of the wgclar opcode. It uses the file wgclar.csd. Example of the wgclar opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/define.xml0000644000000000000000000001731212262561504016361 0ustar rootroot #define #defineorchestra macrosorchestra Orchestra Syntax:Macros #define #define Defines a macro. Description Macros are textual replacements which are made in the orchestra as it is being read. The macro system in Csound is a very simple one, and uses the characters # and $ to define and call macros. This can save typing, and can lead to a coherent structure and consistent style. This is similar to, but independent of, the macro system in the score language. #define NAME -- defines a simple macro. The name of the macro must begin with a letter and can consist of any combination of letters and numbers. Case is significant. This form is limiting, in that the variable names are fixed. More flexibility can be obtained by using a macro with arguments, described below. #define NAME(a' b' c') -- defines a macro with arguments. This can be used in more complex situations. The name of the macro must begin with a letter and can consist of any combination of letters and numbers. Within the replacement text, the arguments can be substituted by the form: $A. In fact, the implementation defines the arguments as simple macros. There may be up to 5 arguments, and the names may be any choice of letters. Remember that case is significant in macro names. Syntax #define NAME # replacement text # #define NAME(a' b' c') # replacement text # Initialization # replacement text # -- The replacement text is any character string (not containing a #) and can extend over mutliple lines. The replacement text is enclosed within the # characters, which ensure that additional characters are not inadvertently captured. Performance Some care is needed with textual replacement macros, as they can sometimes do strange things. They take no notice of any meaning, so spaces are significant. This is why, unlike the C programming language, the definition has the replacement text surrounded by # characters. Used carefully, this simple macro system is a powerful concept, but it can be abused. Examples Here is a simple example of the defining a macro. It uses the file define.csd. Simple example of the define macro. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: Macro definition for VOLUME Macro definition for CPS Macro definition for TABLE Here is an example of the defining a macro with arguments. It uses the file define_args.csd. Example of the define macro with arguments. Its output should include lines like this: Macro definition for OSCMACRO Predefined Math Constant Macros New in Csound 5.04 are predefined Math Constant Macros. The values defined are those found in the C header math.h, and are automatically defined when Csound starts and available for use in orchestras. Macro Value Equivalent to $M_E 2.7182818284590452354 e $M_LOG2E 1.4426950408889634074 log_2(e) $M_LOG10E 0.43429448190325182765 log_10(e) $M_LN2 0.69314718055994530942 log_e(2) $M_LN10 2.30258509299404568402 log_e(10) $M_PI 3.14159265358979323846 pi $M_PI_2 1.57079632679489661923 pi/2 $M_PI_4 0.78539816339744830962 pi/4 $M_1_PI 0.31830988618379067154 1/pi $M_2_PI 0.63661977236758134308 2/pi $M_2_SQRTPI 1.12837916709551257390 2/sqrt(pi) $M_SQRT2 1.41421356237309504880 sqrt(2) $M_SQRT1_2 0.70710678118654752440 1/sqrt(2) See Also $NAME, #undef Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April 1998 Examples written by &namekevin;. New in Csound version 3.48 manual_src~dfsg/opcodes/outleta.xml0000644000000000000000000000640412262561504016604 0ustar rootroot outleta Signal Flow Graph Opcodes outleta outleta Sends an arate signal out from an instrument to a named port. Description Sends an arate signal out from an instrument to a named port. Syntax outleta Sname, asignal Initialization Sname -- String name of the outlet port. The name of the outlet is implicitly qualified by the instrument name or number, so it is valid to use the same outlet name in more than one instrument (but not to use the same outlet name twice in one instrument). Performance asignal -- audio output signal During performance, the audio output signal is sent to each instance of an instrument containing an inlet port to which this outlet has been connected using the connect opcode. The signals of all the outlets connected to an inlet are summed in the inlet. Examples Here is an example of the outleta opcode. It uses the file outleta.csd. Example of the outleta opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also outletk outletkid outletf inleta inletk inletkid inletf connect alwayson ftgenonce More information on this opcode: http://www.csounds.com/journal/issue13/signalFlowGraphOpcodes.html , written by Michael Gogins Credits By: &namemichael; 2009 manual_src~dfsg/opcodes/inh.xml0000644000000000000000000000363212262561504015705 0ustar rootroot inh Signal I/O:Signal Input inh inh Reads six-channel audio data from an external device or stream. Description Reads six-channel audio data from an external device or stream. Warning This opcode is designed to be used only with orchestras that have nchnls_i=6. Doing so with orchestras with nchnls_i > 6 will cause incorrect audio input. Syntax ar1, ar2, ar3, ar4, ar5, ar6 inh Performance Reads six-channel audio data from an external device or stream. If the command-line -i flag is set, sound is read continuously from the audio input stream (e.g. stdin or a soundfile) into an internal buffer. Any number of these opcodes can read freely from this buffer. See Also diskin, in, ino, inq, ins, soundin Credits Author: &namejohn; manual_src~dfsg/opcodes/fluidCCk.xml0000644000000000000000000000524412262561504016614 0ustar rootroot fluidCCk Signal Generators:Sample Playback fluidCCk fluidCCk Sends a MIDI controller data message to fluid. Syntax fluidCCk iEngineNumber, iChannelNumber, iControllerNumber, kValue Description Sends a MIDI controller data (MIDI controller number and value to use) message to a fluid engine by number on the user specified MIDI channel number. Initialization iEngineNumber -- engine number assigned from fluidEngine iChannelNumber -- MIDI channel number to which the Fluidsynth program is assigned: from 0 to 255. MIDI channels numbered 16 or higher are virtual channels. iControllerNumber -- MIDI controller number to use for this message Performance kValue -- value to set for controller (usually 0-127) Examples Here is an example of the fluidCCk opcodes. It uses the file fluidCCk.csd. See Also fluidEngine, fluidNote, fluidLoad, fluidCCi More information on soundfonts in the Floss Manuals: http://en.flossmanuals.net/csound/ch048_d-reading-midi-files Other information on soundfonts on Wikipedia: http://en.wikipedia.org/wiki/Soundfont Credits Michael Gogins (gogins at pipeline dot com), Steven Yi. Thanks to Peter Hanappe for Fluidsynth. manual_src~dfsg/opcodes/outvalue.xml0000644000000000000000000000423512262561504016773 0ustar rootroot outvalue Signal I/O:Signal Output outvalue outvalue Sends a k-rate signal or string to a user-defined channel. Description Sends a k-rate signal or string to a user-defined channel. Syntax outvalue "channel name", kvalue outvalue "channel name", "string" Performance "channel name" -- An integer or string (in double-quotes) representing channel. kvalue -- The k-rate value that is sent to the channel. string -- The string or string variable that is sent to the channel. Examples Here is an example of the outvalue opcode. It uses the file outvalue.csd. Example of the outvalue opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also invalue Credits Author: Matt Ingalls manual_src~dfsg/opcodes/wgpluck2.xml0000644000000000000000000000560112262561504016663 0ustar rootroot wgpluck2 Signal Generators:Waveguide Physical Modeling wgpluck2 wgpluck2 Physical model of the plucked string. Description wgpluck2 is an implementation of the physical model of the plucked string, with control over the pluck point, the pickup point and the filter. Based on the Karplus-Strong algorithm. Syntax ares wgpluck2 iplk, kamp, icps, kpick, krefl Initialization iplk -- The point of pluck is iplk, which is a fraction of the way up the string (0 to 1). A pluck point of zero means no initial pluck. icps -- The string plays at icps pitch. Performance kamp -- Amplitude of note. kpick -- Proportion of the way along the string to sample the output. krefl -- the coefficient of reflection, indicating the lossiness and the rate of decay. It must be strictly between 0 and 1 (it will complain about both 0 and 1). Examples Here is an example of the wgpluck2 opcode. It uses the file wgpluck2.csd. Example of the wgpluck2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also repluck Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/strsubk.xml0000644000000000000000000000264312262561504016625 0ustar rootroot strsubk Strings:Manipulation strsubk strsubk Extract a substring Description Return a substring of the source string. strsubk runs both at init and performance time. Syntax Sdst strsubk Ssrc, kstart, kend Performance kstart -- start position in Ssrc, counting from 0. A negative value means the end of the string. kend -- end position in Ssrc, counting from 0. A negative value means the end of the string. If kend is less than kstart, the output is reversed. See also strsub Credits Author: Istvan Varga 2006 manual_src~dfsg/opcodes/deltap.xml0000644000000000000000000001521212262561504016375 0ustar rootroot deltap Signal Modifiers:Delay deltap deltap Taps a delay line at variable offset times. Description Tap a delay line at variable offset times. Syntax ares deltap kdlt Performance kdlt -- specifies the tapped delay time in seconds. Each can range from 1 control period to the full delay time of the read/write pair; however, since there is no internal check for adherence to this range, the user is wholly responsible. Each argument can be a constant, a variable, or a time-varying signal. deltap extracts sound by reading the stored samples directly. This opcode can tap into a delayr/delayw pair, extracting delayed audio from the idlt seconds of stored sound. There can be any number of deltap and/or deltapi units between a read/write pair. Each receives an audio tap with no change of original amplitude. This opcode can provide multiple delay taps for arbitrary delay path and feedback networks. They can deliver either constant-time or time-varying taps, and are useful for building chorus effects, harmonizers, and Doppler shifts. Constant-time delay taps (and some slowly changing ones) do not need interpolated readout; they are well served by deltap. Medium-paced or fast varying dlt's, however, will need the extra services of deltapi. delayr/delayw pairs may be interleaved. To associate a delay tap unit with a specific delayr unit, it not only has to be located between that delayr and the appropriate delayw unit, but must also precede any following delayr units. See Example 2. (This feature added in Csound version 3.57 by Jens Groh and &namejohn;). N.B. k-rate delay times are not internally interpolated, but rather lay down stepped time-shifts of audio samples; this will be found quite adequate for slowly changing tap times. For medium to fast-paced changes, however, one should provide a higher resolution audio-rate timeshift as input. Examples deltap example #1 asource buzz 1, 440, 20, 1 atime linseg 1, p3/2,.01, p3/2,1 ; trace a distance in secs ampfac = 1/atime/atime ; and calc an amp factor adump delayr 1 ; set maximum distance amove deltapi atime ; move sound source past delayw asource ; the listener out amove * ampfac deltap example #2 ainput1 = ..... ainput2 = ..... kdlyt1 = ..... kdlyt2 = ..... ;Read delayed signal, first delayr instance: adump delayr 4.0 adly1 deltap kdlyt1 ; associated with first delayr instance ;Read delayed signal, second delayr instance: adump delayr 4.0 adly2 deltap kdlyt2 ; associated with second delayr instance ;Do some cross-coupled manipulation: afdbk1 = 0.7 * adly1 + 0.7 * adly2 + ainput1 afdbk2 = -0.7 * adly1 + 0.7 * adly2 + ainput2 ;Feed back signal, associated with first delayr instance: delayw afdbk1 ;Feed back signal, associated with second delayr instance: delayw afdbk2 outs adly1, adly2 Here is yet another example of the deltap opcode. It uses the file deltap.csd. Example of the deltap opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also deltap3, deltapi, deltapn manual_src~dfsg/opcodes/resonr.xml0000644000000000000000000001727112262561504016443 0ustar rootroot resonr Signal Modifiers:Standard Filters:Resonant resonr resonr A bandpass filter with variable frequency response. Description Implementations of a second-order, two-pole two-zero bandpass filter with variable frequency response. Syntax ares resonr asig, xcf, xbw [, iscl] [, iskip] Initialization The optional initialization variables for resonr are identical to the i-time variables for reson. iscl (optional, default=0) -- coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise. A zero value signifies no scaling of the signal, leaving that to some later adjustment (see balance). The default value is 0. iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance asig -- input signal to be filtered xcf -- cutoff or resonant frequency of the filter, measured in Hz xbw -- bandwidth of the filter (the Hz difference between the upper and lower half-power points) resonr and resonz are variations of the classic two-pole bandpass resonator (reson). Both filters have two zeroes in their transfer functions, in addition to the two poles. resonz has its zeroes located at z = 1 and z = -1. resonr has its zeroes located at +sqrt(R) and -sqrt(R), where R is the radius of the poles in the complex z-plane. The addition of zeroes to resonr and resonz results in the improved selectivity of the magnitude response of these filters at cutoff frequencies close to 0, at the expense of less selectivity of frequencies above the cutoff peak. resonr and resonz are very close to constant-gain as the center frequency is swept, resulting in a more efficient control of the magnitude response than with traditional two-pole resonators such as reson. resonr and resonz produce a sound that is considerably different from reson, especially for lower center frequencies; trial and error is the best way of determining which resonator is best suited for a particular application. Examples Here is an example of the resonr and resonz opcodes. It uses the file resonr.csd. Example of the resonr and resonz opcodes. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Technical History resonr and resonz were originally described in an article by Julius O. Smith and James B. Angell.1 Smith and Angell recommended the resonz form (zeros at +1 and -1) when computational efficiency was the main concern, as it has one less multiply per sample, while resonr (zeroes at + and - the square root of the pole radius R) was recommended for situations when a perfectly constant-gain center peak was required. Ken Steiglitz, in a later article 2, demonstrated that resonz had constant gain at the true peak of the filter, as opposed to resonr, which displayed constant gain at the pole angle. Steiglitz also recommended resonz for its sharper notches in the gain curve at zero and Nyquist frequency. Steiglitz's recent book 3 features a thorough technical discussion of reson and resonz, while Dodge and Jerse's textbook 4 illustrates the differences in the response curves of reson and resonz. References Smith, Julius O. and Angell, James B., "A Constant-Gain Resonator Tuned by a Single Coefficient," Computer Music Journal, vol. 6, no. 4, pp. 36-39, Winter 1982. Steiglitz, Ken, "A Note on Constant-Gain Digital Resonators," Computer Music Journal, vol. 18, no. 4, pp. 8-10, Winter 1994. Ken Steiglitz, A Digital Signal Processing Primer, with Applications to Digital Audio and Computer Music. Addison-Wesley Publishing Company, Menlo Park, CA, 1996. Dodge, Charles and Jerse, Thomas A., Computer Music: Synthesis, Composition, and Performance. New York: Schirmer Books, 1997, 2nd edition, pp. 211-214. See Also resonz Credits Author: &namesean; Seattle, Washington 1999 New in Csound version 3.55 Audio rate parameters introduced in version 6.02 November 2013. manual_src~dfsg/opcodes/oscilikt.xml0000644000000000000000000000645712262561504016760 0ustar rootroot oscilikt Signal Generators:Basic Oscillators oscilikt oscilikt A linearly interpolated oscillator that allows changing the table number at k-rate. Description oscilikt is very similar to oscili, but allows changing the table number at k-rate. It is slightly slower than oscili (especially with high control rate), although also more accurate as it uses a 31-bit phase accumulator, as opposed to the 24-bit one used by oscili. Syntax ares oscilikt xamp, xcps, kfn [, iphs] [, istor] kres oscilikt kamp, kcps, kfn [, iphs] [, istor] Initialization iphs (optional, defaults to 0) -- initial phase in the range 0 to 1. Other values are wrapped to the allowed range. istor (optional, defaults to 0) -- skip initialization. Performance kamp, xamp -- amplitude. kcps, xcps -- frequency in Hz. Zero and negative values are allowed. However, the absolute value must be less than sr (and recommended to be less than sr/2). kfn -- function table number. Can be varied at control rate (useful to morph waveforms, or select from a set of band-limited tables generated by GEN30). Examples Here is an example of the oscilikt opcode. It uses the file oscilikt.csd. Example of the oscilikt opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also osciliktp and oscilikts. Credits Author: &nameistvan; Example written by &namekevin;. New in version 4.22 manual_src~dfsg/opcodes/array.xml0000644000000000000000000000367512262561504016254 0ustar rootroot array Table Control:Table Queries array array Generate a vector with initial values. Description Generate a vector (one-dimensional k-rate array) with initial values. Syntax karray[] array ival1, ival2,.....ivaln Initialization ival1,...ivaln -- values to place in the vector. Examples Here is an example of the array opcode. It uses the file array.csd. Example of the array opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also Vectorial opcodes fillarray Credits Author: &namejohn; Codemist Ltd 2013 New in version 6.00 manual_src~dfsg/opcodes/chnrecv.xml0000644000000000000000000000601012262561504016550 0ustar rootroot chnrecv Signal I/O:Software Bus chnrecv chnrecv Recieves data from the software bus. Description Receives data from a channel of the inward named software bus. Implies declaring the channel with imode=1 (see also chn_k, chn_a, and chn_S). Note that chnsend/chnrecv (which in Csound6 work identically to invalue/outvalue) are usually used for the callback-based communication between Csound and an external host. Use the chnset/chnget opcodes for sending and receiving data inside Csound. Syntax ival chnrecv Sname kval chnrecv Sname aval chnrecv Sname Sval chnrecv Sname Initialization Sname -- a string that identifies a channel of the named software bus to read. Performance ival -- the control value read at i-time. kval -- the control value read at performance time. aval -- the audio signal read at performance time. Sval -- the string value read at i-time. Example The example shows the software bus being used as an asynchronous control signal to select a filter cutoff. It assumes that an external program that has access to the API is feeding the values. sr = 44100 ksmps = 100 nchnls = 1 instr 1 kc chnrecv "cutoff" a1 oscil p4, p5, 100 a2 lowpass2 a1, kc, 200 out a2 endin Credits Author: &nameistvan; 2005 manual_src~dfsg/opcodes/sfplay3.xml0000644000000000000000000001617112262561504016512 0ustar rootroot sfplay3 Signal Generators:Sample Playback sfplay3 sfplay3 Plays a SoundFont2 (SF2) sample preset, generating a stereo sound with cubic interpolation. Description Plays a SoundFont2 (SF2) sample preset, generating a stereo sound with cubic interpolation. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. Syntax ar1, ar2 sfplay3 ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset] [, ienv] Initialization ivel -- velocity value inotenum -- MIDI note number value ipreindex -- preset index iflag (optional) -- flag regarding the behavior of xfreq and inotenum ioffset (optional) -- start playing at offset, in samples. ienv (optional) -- enables and determines amplitude envelope. 0 = no envelope, 1 = linear attack and decay, 2 = linear attack, exponential decay (see below). Default = 0. Performance xamp -- amplitude correction factor xfreq -- frequency value or frequency multiplier, depending by iflag. When iflag = 0, xfreq is a multiplier of a the default frequency, assigned by SF2 preset to the inotenum value. When iflag = 1, xfreq is the absolute frequency of the output sound, in Hz. Default is 0. When iflag = 0, inotenum sets the frequency of the output according to the MIDI note number used, and xfreq is used as a multiplier. When iflag = 1, the frequency of the output, is set directly by xfreq. This allows the user to use any kind of micro-tuning based scales. However, this method is designed to work correctly only with presets tuned to the default equal temperament. Attempts to use this method with a preset already having non-standard tunings, or with drum-kit-based presets, could give unexpected results. Adjustment of the amplitude can be done by varying the xamp argument, which acts as a multiplier. Notice that both xamp and xfreq can use k-rate as well as a-rate signals. Both arguments must use variables of the same rate, or sfplay3 will not work correctly. ipreindex must contain the number of a previously assigned preset, or Csound will crash. The ioffset parameter allows the sound to start from a sample different than the first one. The user should make sure that its value is within the length of the specific sound. Otherwise, Csound will probably crash. The ienv parameter enables and determines the type of amplitude envelope used. The default value is 0, or no envelope. If ienv is set to 1, the attack and decay portions are linear. If set to 2, the attack is linear and the decay is exponential. The release portion of the envelope has not yet been implemented. sfplay3 plays a preset, generating a stereo sound with cubic interpolation. ivel does not directly affect the amplitude of the output, but informs sfplay3 about which sample should be chosen in multi-sample, velocity-split presets. sfplay3 is a cubic-interpolation version of sfplay. Difference of sound-quality is noticeable specially in bass-frequency-transposed samples. In high-freq-transposed samples the difference is less noticeable, and I suggest to use linear-interpolation versions, because they are faster. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. Examples Here is an example of the sfplay3 opcode. It uses the file sfplay3.csd. Example of the sfplay3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also sfilist, sfinstr3, sfinstr3m, sfinstr, sfinstrm, sfload, sfpassign, sfplay3m, sfplaym, sfplay, sfplist, sfpreset Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 New optional parameter ienv in version 5.09 manual_src~dfsg/opcodes/strtodk.xml0000644000000000000000000000457312262561504016626 0ustar rootroot strtodk Strings:Conversion strtodk strtodk Converts a string to a float (k-rate). Description Convert a string to a floating point value at i- or k-rate. It is also possible to pass an strset index or a string p-field from the score instead of a string argument. If the string cannot be parsed as a floating point or integer number, an init or perf error occurs and the instrument is deactivated. Note If a k-rate index variable is used, it should be valid at i-time as well. Syntax kr strtodk Sstr kr strtodk kndx Performance kr -- Value of string as float. Sstr -- String to convert. indx -- index of string set by strset Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/timeinsts.xml0000644000000000000000000000465312262561504017152 0ustar rootroot timeinsts Instrument Control:Time Reading timeinsts timeinsts Read absolute time in seconds. Description Read absolute time, in seconds, since the start of an instance of an instrument. Syntax kres timeinsts Performance Time in seconds is available with timeinsts. This would return 0.5 after half a second. timeinsts produces a k-rate variable for output. There are no input parameters. timeinsts is similar to times except it returns the time since the start of this instance of the instrument. Examples Here is an example of the timeinsts opcode. It uses the file timeinsts.csd. Example of the timeinsts opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also timeinstk, timek, times Credits Author: Robin Whittle Australia May 1997 manual_src~dfsg/opcodes/pdhalfy.xml0000644000000000000000000001521412262561504016555 0ustar rootroot pdhalfy Signal Modifiers:Waveshaping pdhalfy pdhalfy Distorts a phasor for reading two unequal portions of a table in equal periods. Description The pdhalfy opcode is a variation on the phase distortion synthesis method of the pdhalf opcode. It is useful for distorting a phasor in order to read two unequal portions of a table in the same number of samples. Syntax aout pdhalfy ain, kShapeAmount [, ibipolar [, ifullscale]] Initialization ibipolar -- an optional parameter specifying either unipolar (0) or bipolar (1) mode. Defaults to unipolar mode. ifullscale -- an optional parameter specifying the range of input and output values. The maximum will be ifullscale. The minimum depends on the mode of operation: zero for unipolar or -ifullscale for bipolar. Defaults to 1.0 -- you should set this parameter to the maximum expected input value. Performance ain -- the input signal to be distorted. aout -- the output signal. kShapeAmount -- the amount of distortion applied to the input. Must be between negative one and one (-1 to 1). An amount of zero means no distortion. Transfer function created by pdhalfy and a negative kShapeAmount. Transfer function created by pdhalfy and a negative kShapeAmount. The pdhalfy opcode calculates a transfer function that is composed of two linear segments (see the graph). These segments meet at a "pivot point" which always lies on the same vertical axis. (In unipolar mode, the axis is x = 0.5, and for bipolar mode it is the y axis). So, pdhalfy is a variation of the pdhalf opcode that places the pivot point of the phase distortion pattern on a vertical axis instead of a horizontal axis. The kShapeAmount parameter specifies where on the vertical axis this point falls. When kShapeAmount is zero, the pivot point is in the middle of the output range, forming a straight line for the transfer function and thus causing no change in the input signal. As kShapeAmount changes from zero (0) to negative one (-1), the pivot point downward towards the bottom of the graph. As it changes from zero (0) to positive one (1), the pivot point moves upward, producing an inverted pattern. If the input to pdhalfy is a phasor and the output is used to index a table, the use of pdhalfy will divide the table into two segments of different sizes with each segment being mapped to half of the oscillator period. Values for kShapeAmount that are less than zero will cause less than half of the table to be read in the first half of the period of oscillation. The rest of the table will be read in the second half of the period. The reverse is true for values of kShapeAmount greater than zero. Note that the frequency of the phasor is always unchanged. Thus, this method of phase distortion can only produce higher partials in a harmonic series. It cannot produce inharmonic sidebands in the way that frequency modulation does. pdhalfy tends to have a milder quality to its distortion than pdhalf. pdhalfy can work in either unipolar or bipolar modes. Unipolar mode is appropriate for signals like phasors that range between zero and some maximum value (selectable with ifullscale). Bipolar mode is appropriate for signals that range above and below zero by roughly equal amounts such as most audio signals. Applying pdhalfy directly to an audio signal in this way results in a crude but adjustable sort of waveshaping/distortion. A typical example of the use of pdhalfy is aphase phasor ifreq apd pdhalfy aphase, kamount aout tablei apd, 1, 1 Examples Here is an example of the pdhalfy opcode. It uses the file pdhalfy.csd. Example of the pdhalfy opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pdhalf, pdclip More information about phase distortion on Wikipedia: http://en.wikipedia.org/wiki/Phase_distortion_synthesis Credits Author: &nameanthony; January 2008 New in Csound version 5.08 manual_src~dfsg/opcodes/outq3.xml0000644000000000000000000000615112262561504016201 0ustar rootroot outq3 Signal I/O:Signal Output outq3 outq3 Writes samples to quad channel 3 of an external device or stream. Description Writes samples to quad channel 3 of an external device or stream. Syntax outq3 asig Performance Sends audio samples to an accumulating output buffer (created at the beginning of performance) which serves to collect the output of all active instruments before the sound is written to disk. There can be any number of these output units in an instrument. The type (mono, stereo, quad, hex, or oct) should agree with nchnls. But as of version 3.50, Csound will attempt to change an incorrect opcode to agree with the nchnls statement. Opcodes can be chosen to direct sound to any particular channel: outs1 sends to stereo channel 1, outq3 to quad channel 3, etc. Examples Here is an example of the outq3 opcode. It uses the file outq3.csd. Example of the outq3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also out, outh, outo, outq, outq1, outq2, outq4, outs, outs1, outs2, soundout Credits Author: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 manual_src~dfsg/opcodes/pvsifd.xml0000644000000000000000000000617412262561504016426 0ustar rootroot pvsifd Spectral Processing:Streaming pvsifd pvsifd Instantaneous Frequency Distribution, magnitude and phase analysis. Description The pvsifd opcode takes an input a-rate signal and performs an Instantaneous Frequency, magnitude and phase analysis, using the STFT and pvsifd (Instantaneous Frequency Distribution), as described in Lazzarini et al, "Time-stretching using the Instantaneous Frequency Distribution and Partial Tracking", Proc.of ICMC05, Barcelona. It generates two PV streaming signals, one containing the amplitudes and frequencies (a similar output to pvsanal) and another containing amplitudes and unwrapped phases. Syntax ffr,fphs pvsifd ain, ifftsize, ihopsize, iwintype[,iscal] Performance ffr -- output pv stream in AMP_FREQ format fphs -- output pv stream in AMP_PHASE format ifftsize -- FFT analysis size, must be power-of-two and integer multiple of the hopsize. ihopsize -- hopsize in samples iwintype -- window type (O: Hamming, 1: Hanning) iscal -- amplitude scaling (defaults to 1). It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the pvsifd opcode. It uses the file pvsifd.csd. Example of the pvsifd opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows the pvsifd analysis feeding into partial tracking andcubic-phase additive resynthesis with pitch shifting. Credits Author: &namevictor; June 2005 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/nchnls_i.xml0000644000000000000000000000527712262561504016733 0ustar rootroot nchnls_i Orchestra Syntax:Header nchnls_i nchnls_i Sets the number of channels of audio input. Description These statements are global value assignments, made at the beginning of an orchestra, before any instrument block is defined. Their function is to set certain reserved symbol variables that are required for performance. Once set, these reserved symbols can be used in expressions anywhere in the orchestra. Syntax nchnls_i = iarg Initialization nchnls_i = (optional) -- set number of channels of audio input to iarg. (1 = mono, 2 = stereo, 4 = quadraphonic.) The default value is the valus of nchnls. In addition, any global variable can be initialized by an init-time assignment anywhere before the first instr statement. All of the above assignments are run as instrument 0 (i-pass only) at the start of real performance. Examples Here is an example of the nchnls_i opcode. It uses the file nchnls_i.csd. Example of the nchnls_i opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also kr, ksmps, nchnls, sr manual_src~dfsg/opcodes/pvscent.xml0000644000000000000000000000430012262561504016602 0ustar rootroot pvscent Spectral Processing:Streaming pvscent pvscent Calculate the spectral centroid of a signal. Description Calculate the spectral centroid of a signal from its discrete Fourier transform. Syntax kcent pvscent fsig Performance kcent -- the spectral centroid fsig -- an input pv stream Examples Here is an example of the use of the pvscent opcode. It uses the file pvscent.csd. Example of the <emphasis>pvscent</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn, pvspitch Credits Author: John ffitch; March 2005 New plugin in version 5 March 2005. manual_src~dfsg/opcodes/ampdb.xml0000644000000000000000000000451212262561504016210 0ustar rootroot ampdb Mathematical Operations:Amplitude Functions ampdb ampdb Returns the amplitude equivalent of the decibel value x. Description Returns the amplitude equivalent of the decibel value x. Thus: 60 dB = 1000 66 dB = 1995.262 72 dB = 3891.07 78 dB = 7943.279 84 dB = 15848.926 90 dB = 31622.764 Syntax ampdb(x) (no rate restriction) Examples Here is an example of the ampdb opcode. It uses the file ampdb.csd. Example of the ampdb opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: instr 1: iamp = 316.228 instr 1: iamp = 31622.763 instr 1: iamp = 2511.886 instr 1: iamp = 9999.996 See Also ampdbfs, db, dbamp, dbfsamp manual_src~dfsg/opcodes/control.xml0000644000000000000000000000406512262561504016610 0ustar rootroot control Instrument Control:Sensing and Control control control Configurable slider controls for realtime user input. Description Configurable slider controls for realtime user input. Requires Winsound or TCL/TK. control reads a slider's value. Syntax kres control knum Performance Note that this opcode is not available on Windows due to the implimentation of pipes on that system knum -- number of the slider to be read. Calling control will create a new slider on the screen. There is no theoretical limit to the number of sliders. Windows and TCL/TK use only integers for slider values, so the values may need rescaling. GUIs usually pass values at a fairly slow rate, so it may be advisable to pass the output of control through port. Examples See the setctrl opcode for an example. See Also setctrl Credits Author: &namejohn; University of Bath, Codemist. Ltd. Bath, UK May, 2000 New in Csound version 4.06 manual_src~dfsg/opcodes/phasor.xml0000644000000000000000000000551212262561504016422 0ustar rootroot phasor Signal Generators:Phasors phasor phasor Produce a normalized moving phase value. Description Produce a normalized moving phase value. Syntax ares phasor xcps [, iphs] kres phasor kcps [, iphs] Initialization iphs (optional) -- initial phase, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is zero. Performance An internal phase is successively accumulated in accordance with the kcps or xcps frequency to produce a moving phase value, normalized to lie in the range 0 <= phs < 1. When used as the index to a table unit, this phase (multiplied by the desired function table length) will cause it to behave like an oscillator. Note that phasor is a special kind of integrator, accumulating phase increments that represent frequency settings. Examples Here is an example of the phasor opcode. It uses the file phasor.csd. Example of the phasor opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also The Table Access opcodes like: table, tablei, table3 and tab. Also: phasorbnk. manual_src~dfsg/opcodes/follow2.xml0000644000000000000000000000533512262561504016515 0ustar rootroot follow2 Instrument Control:Sensing and Control follow2 follow2 Another controllable envelope extractor. Description A controllable envelope extractor using the algorithm attributed to Jean-Marc Jot. Syntax ares follow2 asig, katt, krel Performance asig -- the input signal whose envelope is followed katt -- the attack rate (60dB attack time in seconds) krel -- the decay rate (60dB decay time in seconds) The output tracks the amplitude envelope of the input signal. The rate at which the output grows to follow the signal is controlled by the katt, and the rate at which it decreases in response to a lower amplitude, is controlled by the krel. This gives a smoother envelope than follow. Examples Here is an example of the follow2 opcode. It uses the file follow2.csd, and beats.wav. Example of the follow2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; The algorithm for the follow2 is attributed to Jean-Marc Jot. University of Bath, Codemist Ltd. Bath, UK February 2000 New in Csound version 4.03 Added notes by &namerasmus; on September 2002. manual_src~dfsg/opcodes/sndloop.xml0000644000000000000000000000560512262561504016607 0ustar rootroot sndloop Signal Generators:Sample Playback sndloop sndloop A sound looper with pitch control. Description This opcode records input audio and plays it back in a loop with user-defined duration and crossfade time. It also allows the pitch of the loop to be controlled, including reversed playback. Syntax asig, krec sndloop ain, kpitch, ktrig, idur, ifad Initialization idur -- loop duration in seconds ifad -- crossfade duration in seconds Performance asig -- output sig krec -- 'rec on' signal, 1 when recording, 0 otherwise kpitch -- pitch control (transposition ratio); negative values play the loop back in reverse ktrig -- trigger signal: when 0, processing is bypassed. When switched on (ktrig >= 1), the opcode starts recording until the loop memory is full. It then plays the looped sound until it is switched off again (ktrig = 0). Another recording can start again with ktrig >= 1. Examples Here is an example of the sndloop opcode. It uses the file sndloop.csd. Example of the sndloop opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows the basic operation of sndloop. Pitch can be controlled at the k-rate, recording is started as soon as the trigger value is >= 1. Recording can be restarted by making the trigger 0 and then 1 again. Credits Author: &namevictor; April 2005 New in Version 5.00 manual_src~dfsg/opcodes/flpanelend.xml0000644000000000000000000000374512262561504017244 0ustar rootroot FLpanelEnd FLTK:Containers FLpanelEnd FLpanelEnd Marks the end of a group of FLTK widgets contained inside of a window (panel). Description Marks the end of a group of FLTK widgets contained inside of a window (panel). Syntax FLpanelEnd Performance Containers are useful to format the graphic appearance of the widgets. The most important container is FLpanel, that actually creates a window. It can be filled with other containers and/or valuators or other kinds of widgets. There are no k-rate arguments in containers. See Also FLgroup, FLgroupEnd, FLpack, FLpackEnd, FLpanel, FLscroll, FLscrollEnd, FLtabs, FLtabsEnd Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/lessthan.xml0000644000000000000000000000713712262561504016754 0ustar rootroot : ? conditional expressions < Instrument Control:Conditional Values < < Determines if one value is less than another. Description Determines if one value is less than another. Syntax (a < b ? v1 : v2) where a, b, v1 and v2 may be expressions, but a, b not audio-rate. Performance In the above conditional, a and b are first compared. If the indicated relation is true (a less than b), then the conditional expression has the value of v1; if the relation is false, the expression has the value of v2. NB.: If v1 or v2 are expressions, these will be evaluated before the conditional is determined. In terms of binding strength, all conditional operators (i.e. the relational operators (<, etc.), and ?, and : ) are weaker than the arithmetic and logical operators (+, -, *, /, && and ||). These are operators not opcodes. Therefore, they can be used within orchestra statements, but do not form complete statements themselves. Examples Here is an example of the < operator. It uses the file lessthan.csd. Example of the < operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: ipitch = 261.626 ipitch = 800.00 See Also ==, >=, >, <=, != manual_src~dfsg/opcodes/strcpy.xml0000644000000000000000000000245112262561504016451 0ustar rootroot strcpy Strings:Manipulation strcpy strcpy Assign value to a string variable Description Assign to a string variable by copying the source which may be a constant or another string variable. strcpy and = copy the string at i-time only. Syntax Sdst strcpy Ssrc Sdst = Ssrc Example Sfoo strcpy "Hello, world !" puts Sfoo, 1 See also strcpyk Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/mandel.xml0000644000000000000000000000616112262561504016367 0ustar rootroot mandel Signal Generators:Models and Emulations mandel mandel Mandelbrot set Description Returns the number of iterations corresponding to a given point of complex plane by applying the Mandelbrot set formula. Syntax kiter, koutrig mandel ktrig, kx, ky, kmaxIter Performance kiter - number of iterations koutrig - output trigger signal ktrig - input trigger signal kx, ky - coordinates of a given point belonging to the complex plane kmaxIter - maximum iterations allowed mandel is an opcode that allows the use of the Mandelbrot set formula to generate an output that can be applied to any musical (or non-musical) parameter. It has two output arguments: kiter, that contains the iteration number of a given point, and koutrig, that generates a trigger 'bang' each time kiter changes. A new number of iterations is evaluated only when ktrig is set to a non-zero value. The coordinates of the complex plane are set in kx and ky, while kmaxIter contains the maximum number of iterations. Output values, which are integer numbers, can be mapped in any sorts of ways by the composer. Examples Here is an example of the mandel opcode. It uses the file mandel.csd. Example of the mandel opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also More information on this opcode: Composing Fractal Music with Csound, by Brian Evans Credits Written by &namegabriel;. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/STKPercFlut.xml0000644000000000000000000000710012262561504017227 0ustar rootroot STKPercFlut Signal Generators:STK Opcodes STKPercFlut STKPercFlut STKPercFlut is a percussive flute FM synthesis instrument. Description STKPercFlut is a percussive flute FM synthesis instrument. The instrument uses an algorithm like the algorithm 4 of the TX81Z. Syntax asignal STKPercFlut ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kmod -- controller 2, total modulator index. Value range of kv1 is 0-127. kcross -- controller 4, crossfade of modulator. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kadsr -- controller 128, ADSR 2 and 4 target. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the PercFlut class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKPercFlut opcode. It uses the file STKPercFlut.csd. Example of the STKPercFlut opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/pvsbuffer.xml0000644000000000000000000000531012262561504017124 0ustar rootroot pvsbuffer Spectral Processing:Streaming pvsbuffer pvsbuffer This opcode creates and writes to a circular buffer for f-signals (streaming PV signals). Description This opcode sets up and writes to a circular buffer of length ilen (secs), giving a handle for the buffer and a time pointer, which holds the current write position (also in seconds). It can be used with one or more pvsbufread opcodes. Writing is circular, wrapping around at the end of the buffer. Syntax ihandle, ktime pvsbuffer fsig, ilen Initialisation ihandle -- handle identifying this particular buffer, which should be passed to a reader opcode. ilen -- buffer length in seconds. Performance fsig -- an input pv stream ktime -- the current time of writing in the buffer pvsbuffer stores fsig in a buffer which can be read by pvsbufread using the handle ihandle. Different buffers will have different handles so they can be read independently by different pvsbufread opcodes. pvsbuffer gives in its output the current time (ktime) inside the ring buffer which has just been written. Examples See pvsbufread for examples of the pvsbuffer opcode. See also pvsbufread Credits Author: &namevictor; July 2007 manual_src~dfsg/opcodes/checkbox.xml0000644000000000000000000000465112262561504016717 0ustar rootroot checkbox Instrument Control:Sensing and Control checkbox checkbox Sense on-screen controls. Description Sense on-screen controls. Requires Winsound or TCL/TK. Syntax kres checkbox knum Performance Note that this opcode is not available on Windows due to the implimentation of pipes on that system kres -- value of the checkbox control. If the checkbox is set (pushed) then return 1, if not, return 0. knum -- the number of the checkbox. If it does not exist, it is made on-screen at initialization. Examples Here is a simple example of the checkbox opcode. It uses the file checkbox.csd. Simple example of the checkbox opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also button Credits Author: &namejohn; University of Bath, Codemist. Ltd. Bath, UK September, 2000 Example written by &namekevin;. New in Csound version 4.08 manual_src~dfsg/opcodes/maxarray.xml0000644000000000000000000000422712262561504016754 0ustar rootroot maxarray Array Opcodes maxarray maxarray returns the maximum value in an array. Description The maxarray opcode returns the maximum value in a k-rate array. Syntax kmax maxarray karray Performance kmax -- variable for result. karray -- array for reading. Examples Here is an example of the maxarray opcode. It uses the file maxarray.csd. Example of the maxarray opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also minarray, sumarray, scalearray, Credits Author: &namejohn; October 2011 New in Csound version 5.14 Revised in Csound version 6.00 to work on multidimensional arrays manual_src~dfsg/opcodes/tabmorph.xml0000644000000000000000000001310212262561504016734 0ustar rootroot tabmorph Table Control:Read/Write Opreations tabmorph tabmorph Allow morphing between a set of tables. Description tabmorph allows morphing between a set of tables of the same size, by means of a weighted average between two currently selected tables. Syntax kout tabmorph kindex, kweightpoint, ktabnum1, ktabnum2, \ ifn1, ifn2 [, ifn3, ifn4, ...,ifnN] Initialization ifn1, ifn2 [, ifn3, ifn4, ..., ifnN] - function table numbers. This is a set of chosen tables the user want to use in the morphing. All tables must have the same length. Be aware that only two of these tables can be chosen for the morphing at one time. Since it is possible to use non-integer numbers for the ktabnum1 and ktabnum2 arguments, the morphing is the result from the interpolation between adjacent consecutive tables of the set. Performance kout - The output value for index kindex, resulting from morphing two tables (see below). kindex - main index of the morphed resultant table. The range is 0 to table_length (not included). kweightpoint - the weight of the influence of a pair of selected tables in the morphing. The range of this argument is 0 to 1. A zero makes it output the first table unaltered, a 1 makes it output the second table of the pair unaltered. All intermediate values between 0 and 1 determine the gradual morphing between the two tables of the pair. ktabnum1 - the first table chosen for the morphing. This number doesn’t express the table number directly, but the position of the table in the set sequence (starting from 0 to N-1). If this number is an integer, the corresponding table will be chosen unaltered. If it contains fractional values, then an interpolation with the next adjacent table will result. ktabnum2 - the second table chosen for the morphing. This number doesn’t express the table number directly, but the position of the table in the set sequence (starting from 0 to N-1). If this number is an integer, corresponding table will be chosen unaltered. If it contains fractional values, then an interpolation with the next adjacent table will result. The tabmorph family of opcodes is similar to the table family, but allows morphing between two tables chosen into a set of tables. Firstly the user has to provide a set of tables of equal length (ifn2 [, ifn3, ifn4, ..., ifnN]). Then he can choose a pair of tables in the set in order to perform the morphing: ktabnum1 and ktabnum2 are filled with numbers (zero represents the first table in the set, 1 the second, 2 the third and so on). Then determine the morphing between the two chosen tables with the kweightpoint parameter. After that the resulting table can be indexed with the kindex parameter like a normal table opcode. If the value of this parameter surpasses the length of tables (which must be the same for all tables), then it is wrapped around. tabmorph acts similarly to the table opcode, that is, without using interpolation. This means that it truncates the fractional part of the kindex argument. Anyway, fractional parts of ktabnum1 and ktabnum2 are significant, resulting in linear interpolation between the same element of two adjacent subsequent tables. Examples Here is an example of the tabmorph opcode. It uses the file tabmorph.csd. Example of the tabmorph opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also table, tabmorphi, tabmorpha, tabmorphak, ftmorf, Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/specdisp.xml0000644000000000000000000000477512262561504016752 0ustar rootroot specdisp Spectral Processing:Non-Standard specdisp specdisp Displays the magnitude values of the spectrum. Description Displays the magnitude values of the spectrum. Syntax specdisp wsig, iprd [, iwtflg] Initialization iprd -- the period, in seconds, of each new display. iwtflg (optional, default=0) -- wait flag. If non-zero, hold each display until released by the user. The default value is 0 (no wait). Performance wsig -- the input spectrum. Displays the magnitude values of spectrum wsig every iprd seconds (rounded to some integral number of wsig's originating iprd). Examples ksum specsum wsig, 1 ; sum the spec bins, and ksmooth if ksum < 2000 kgoto zero ; if sufficient amplitude koct specptrk wsig ; pitch-track the signal kgoto contin zero: koct = 0 ; else output zero contin: See Also specsum manual_src~dfsg/opcodes/flcount.xml0000644000000000000000000001331412262561504016577 0ustar rootroot FLcount FLTK:Valuators FLcount FLcount A FLTK widget opcode that creates a counter. Description Allows the user to increase/decrease a value with mouse clicks on a corresponding arrow button. Syntax kout, ihandle FLcount "label", imin, imax, istep1, istep2, itype, \ iwidth, iheight, ix, iy, iopcode [, kp1] [, kp2] [, kp3] [...] [, kpN] Initialization ihandle -- a handle value (an integer number) that unequivocally references a corresponding widget. Used by further opcodes that changes some valuator's properties. It is automatically set by the corresponding valuator. label -- a double-quoted string containing some user-provided text, placed near the corresponding widget. imin -- minimum value of output range imax -- maximum value of output range istep1 -- a floating-point number indicating the increment of valuator value corresponding to of each mouse click. istep1 is for fine adjustments. istep2 -- a floating-point number indicating the increment of valuator value corresponding to of each mouse click. istep2 is for coarse adjustments. itype -- an integer number denoting the appearance of the valuator. iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iopcode -- score opcode type. You have to provide the ascii code of the letter corresponding to the score opcode. At present time only i (ascii code 105) score statements are supported. A zero value refers to a default value of i. So both 0 and 105 activates the i opcode. A value of -1 disables this opcode feature. Performance kout -- output value kp1, kp2, ..., kpN -- arguments of the activated instruments. FLcount allows the user to increase/decrease a value with mouse clicks on corresponding arrow buttons: FLcount. FLcount. There are two kind of arrow buttons, for larger and smaller steps. Notice that FLcount not only outputs a value and a handle, but can also activate (schedule) an instrument provided by the user each time a button is pressed. P-fields of the activated instrument are kp1 (instrument number), kp2 (action time), kp3 (duration) and so on with user p-fields. If the iopcode argument is set to a negative number, no instrument is activated. So this feature is optional. Examples Here is an example of the FLcount opcode. It uses the file FLcount.csd. Example of the FLcount opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLjoy, FLknob, FLroller, FLslider, FLtext Credits Author: &namegabriel; New in version 4.22 Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/vtablewk.xml0000644000000000000000000000641512262561504016750 0ustar rootroot vtablewk Vectorial:Tables vtablewk vtablewk Write vectors (to tables -or arrays of vectors). Description This opcode writes vectors to tables at k-rate. Syntax vtablewk kndx, kfn, ixmode, kinarg1 [, kinarg2, kinarg3 , .... , kinargN ] Initialization ixmode - index data mode. The default value is 0. == 0 index is treated as a raw table location, == 1 index is normalized (0 to 1). Performance kndx - Index into f-table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0). kfn - table number kinarg1...kinargN - input vector elements This opcode is useful in all cases in which one needs to write sets of values associated to unique indexes (for example, multi-channel samples, STFT bin frames, spectral formants, p-field based scores etc.). The number of elements of each vector (length of the vector) is determined by the number of optional arguments on the right (kinarg1, kinarg2, kinarg3, .... kinargN). vtablewk allows also to switch the table number at k-rate (but this is possible only when vector frames of each used table have the same number of elements, otherwise unpredictable results could occurr), as well as to choose indexing style (raw or normalized, see also ixmode argument of table opcode ). Notice that no wrap nor limit mode is implemented. So, if an index attempt to access to a zone not allocated by the table, Csound will probably crash. However this drawback can be easily avoided by using wrap or limit opcodes applied to indexes before using vtablewk, in order to correct eventual out-of-range values. Examples Here is an example of the vtablewk opcode. It uses the files vtablewk.csd. Example of the vtablewk opcode. Credits Written by Gabriel Maldonado. Example written by Andres Cabrera. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/pvs2tab.xml0000644000000000000000000000431712262561504016511 0ustar rootroot pvs2tab Spectral Processing:Streaming pvs2tab pvs2tab Copies spectral data to t-variables. Description Copies a pvs frame to a t-variable. Currently only AMP+FREQ and AMP+PHASE formats allowed. This opcode requires the t-type to be defined, which means it only works in the new bison/flex-based parser. Syntax kframe pvs2tab tvar, fsig Performance kframe -- current copied frame number. It can be used to detect when a new frame has been copied. tvar -- t-variable containing the output. It is produced at every k-period, but may not contain a new frame, pvs frames are produced at their own frame rate that is independent of kr. Generally, this vector needs to be big enough to contain the frame samples, i.e. N+2 (N is the dft size). If smaller, only a portion of the frame will be copied; if bigger, unused points will exist at higher indexes. fsig -- input fsig to be copied. iwinsize -- size of the analysis window, defaults to isize. Examples Example t1 init 1026 a1 inch 1 fsig1 pvsanal a1, 1024,256,1024, 1 kframe pvs2tabt1, fsig1 Credits Author: &namevictor; October 2011 New plugin in version 5 October 2011. manual_src~dfsg/opcodes/lua_opdef.xml0000644000000000000000000001340612262561504017065 0ustar rootroot lua_opdef Lua Opcodes lua_opdef lua_opdef Define an opcode in Lua at i-rate. The opcode can take any number of output and/or input arguments of any type. Description Define an opcode in Lua at i-rate. The opcode can take any number of output and/or input arguments of any type. The code is executed at initialization time, typically from the orchestra header. Global and local variables, functions, tables, and classes may be declared and defined. Objects defined at global Lua scope remain in scope throughout the performance, and are visible to any other Lua code in the same Csound thread. By default, all objects defined in Lua are defined at global scope. In order to ensure that objects are confined to their own block of code, that is to ensure that the object is visible only in lexical scope, the object must be declared as local. This is the feature of Lua that beginners tend to find the most troublesome. Another thing to look out for is that Lua arrays use 1-based indexing, not the 0-based indexing used in C and many other programming languages. Syntax lua_opdef Sname, Sluacode Initialization Sname -- The name of the opcode. Sluacode -- A block of Lua code, of any length. Multi-line blocks may be enclosed in double braces (i.e. {{ }}). This code is evaluated once at initialization time. The Lua code must define all functions that will be called from Csound, using the following naming convention, where opcodename stands for the actual opcode name: opcodename_init for the i-rate opcode subroutine. opcodename_kontrol for the k-rate opcode subroutine. opcodename_audio for the a-rate opcode subroutine. opcodename_noteoff for the note-off subroutine. Each of these Lua functions will receive three lightuserdata (i.e. pointer) arguments: the CSOUND object, the opcode instance, and a pointer to the opcode arguments, which the Lua code must be type cast to a LuaJIT FFI ctype structure containing the opcode output arguments, input arguments, and state variables. Using LuaJIT FFI, the elements of this structure will be accessible as though they were Lua types. Each of these Lua functions must return 0 for success or 1 for failure. The Lua functions may do absolutely anything, although of course if real-time performance is expected, care must be taken to disable Lua garbage collection and observe other recommendations for real-time code. Example Here is an example of a Lua opcode, implementing a Moog ladder filter. For purposes of comparison, a user-defined opcode and the native Csound opcode that compute the same sound using the same algorithm also are shown, and timed. The example uses the file luamoog.csd. Example of a Lua opcode. See Also lua_exec, lua_opcall. Credits By: &namemichael; 2011 New in Csound version 5.13.2 manual_src~dfsg/opcodes/moogvcf.xml0000644000000000000000000000751512262561504016573 0ustar rootroot moogvcf Signal Modifiers:Standard Filters:Resonant moogvcf moogvcf A digital emulation of the Moog diode ladder filter configuration. Description A digital emulation of the Moog diode ladder filter configuration. Syntax ares moogvcf asig, xfco, xres [,iscale, iskip] Initialization iscale (optional, default=1) -- internal scaling factor. Use if asig is not in the range +/-1. Input is first divided by iscale, then output is mutliplied iscale. Default value is 1. (New in Csound version 3.50) iskip (optional, default=0) -- if non zero skip the initialisation of the filter. (New in Csound version 4.23f13 and 5.0) Performance asig -- input signal xfco -- filter cut-off frequency in Hz. As of version 3.50, may i-,k-, or a-rate. xres -- amount of resonance. Self-oscillation occurs when xres is approximately one. As of version 3.50, may a-rate, i-rate, or k-rate. moogvcf is a digital emulation of the Moog diode ladder filter configuration. This emulation is based loosely on the paper Analyzing the Moog VCF with Considerations for Digital Implementation by Stilson and Smith (CCRMA). This version was originally coded in Csound by Josep Comajuncosas. Some modifications and conversion to C were done by &namehans;. Before version 6.02 this filter required that the input signal be normalized to one. This can be easily achieved using 0dbfs, like this: ares moogvcf asig, kfco, kres, 0dbfs You can also use moogvcf2 which defaults scaling to 0dbfs. Examples Here is an example of the moogvcf opcode. It uses the file moogvcf.csd. Example of the moogvcf opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also moogvcf2, biquad, rezzy Credits Author: &namehans; October 1998 New in Csound version 3.49 manual_src~dfsg/opcodes/zaw.xml0000644000000000000000000000521312262561504015725 0ustar rootroot zaw Zak Patch System zaw zaw Writes to a za variable at a-rate without mixing. Description Writes to a za variable at a-rate without mixing. Syntax zaw asig, kndx Performance asig -- value to be written to the za location. kndx -- points to the zk or za location to which to write. zaw writes asig into the za variable specified by kndx. These opcodes are fast, and always check that the index is within the range of zk or za space. If not, an error is reported, 0 is returned, and no writing takes place. Examples Here is an example of the zaw opcode. It uses the file zaw.csd. Example of the zaw opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zawm, ziw, ziwm, zkw, zkwm Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/strsub.xml0000644000000000000000000000412212262561504016444 0ustar rootroot strsub Strings:Manipulation strsub strsub Extract a substring Description Return a substring of the source string. strsub runs at init time only. Syntax Sdst strsub Ssrc[, istart[, iend]] Initialization istart (optional, defaults to 0) -- start position in Ssrc, counting from 0. A negative value means the end of the string. iend (optional, defaults to -1) -- end position in Ssrc, counting from 0. A negative value means the end of the string. If iend is less than istart, the output is reversed. Examples Here is an example of the strsub opcode. It uses the file strsub.csd. Example of the strsub opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also strsubk Credits Author: Istvan Varga 2006 manual_src~dfsg/opcodes/sinsyn.xml0000644000000000000000000000560112262561504016450 0ustar rootroot sinsyn Spectral Processing:Streaming sinsyn sinsyn Streaming partial track additive synthesis with cubic phase interpolation Description The sinsyn opcode takes an input containg a TRACKS pv streaming signal (as generated, for instance by the partials opcode). It resynthesises the signal using linear amplitude and cubic phase interpolation to drive a bank of interpolating oscillators with amplitude scaling control. sinsyn attempts to preserve the phase of the partials in the original signal and in so doing it does not allow for pitch or timescale modifications of the signal. Syntax asig sinsyn fin, kscal, kmaxtracks, ifn Performance asig -- output audio rate signal fin -- input pv stream in TRACKS format kscal -- amplitude scaling kmaxtracks -- max number of tracks in sinsynthesis. Limiting this will cause a non-linear filtering effect, by discarding newer and higher-frequency tracks (tracks are ordered by start time and ascending frequency, respectively) ifn -- function table containing one cycle of a sinusoid (sine or cosine). Examples Here is an example of the sinsyn opcode. It uses the file sinsyn.csd. Example of the sinsyn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of an ifd-analysis signal and cubic-phase additive resynthesis. Credits Author: &namevictor; June 2005 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/turnoff.xml0000644000000000000000000000450212262561504016607 0ustar rootroot turnoff Instrument Control:Duration Control turnoff turnoff Enables an instrument to turn itself off. Description Enables an instrument to turn itself off. Syntax turnoff Performance turnoff -- this p-time statement enables an instrument to turn itself off. Whether of finite duration or held, the note currently being performed by this instrument is immediately removed from the active note list. No other notes are affected. As a rule of thumb, you should turn off instruments with a higher instrument number than the one where turnoff is called, as doing otherwise might cause initialization issues. Examples The following example uses the turnoff opcode. It will cause a note to terminate when a control signal passes a certain threshold (here the Nyquist frequency). It uses the file turnoff.csd. Example of the turnoff opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also ihold, turnoff2, turnon manual_src~dfsg/opcodes/ioutc14.xml0000644000000000000000000000077112262561504016420 0ustar rootroot ioutc14 ioutc14 ioutc14 Deprecated. Description Deprecated as of version 3.52. Use the outic14 opcode instead. manual_src~dfsg/opcodes/trshift.xml0000644000000000000000000000464612262561504016620 0ustar rootroot trshift Spectral Processing:Streaming trshift trshift Streaming partial track frequency scaling. Description The trshift opcode takes an input containg a TRACKS pv streaming signal (as generated, for instance by partials) and shifts all frequencies by a k-rate frequency. It can also, optionally, scale the gain of the signal by a k-rate amount (default 1). The result is frequency shifting of the input tracks. Syntax fsig trshift fin, kpshift[, kgain] Performance fsig -- output pv stream in TRACKS format fin -- input pv stream in TRACKS format kshift -- frequency shift in Hz kgain -- amplitude scaling (default 1) Examples Here is an example of the trshift opcode. It uses the file trshift.csd. Example of the trshift opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of an ifd-analysis signal and linear additive resynthesis with frequency shifting. Credits Author: &namevictor; February 2006 New in Csound 5.01 manual_src~dfsg/opcodes/wiidata.xml0000644000000000000000000001260412262561504016550 0ustar rootroot wiidata Instrument Control:Sensing and Control wiidata wiidata Reads data fields from a number of external Nintendo Wiimote controllers. Description Reads data fields from upto four external Nintendo Wiimote controllers. Syntax kres wiidata kcontrol[, knum] Initialization This opcode must be used in conjuction with a running wiiconnect opcode. Performance Please note that these opcodes are currently only supported on Linux. kcontrol -- the code for which control to read knum -- the number of the which Wiimote to access, which defaults to the first one. On each access a particular data item of the Wiimote is read. The currently implemented controls are given below, together with the macro name defined in the file wii_mac: 0 (WII_BUTTONS): returns a bit pattern for all buttons that were pressed. 1 (WII_TWO): returns 1 if the button has just been pressed, or 0 otherwise. 2 (WII_ONE): as above. 3 (WII_B): as above. 4 (WII_A): as above. 5 (WII_MINUS): as above. 8 (WII_HOME): as above. 9 (WII_LEFT): as above. 10 (WII_RIGHT): as above. 11 (WII_DOWN): as above. 12 (WII_UP): as above. 13 (WII_PLUS): as above. If the control number is 100 more than one of these button codes then the current state of the button is returned. Macros with names like WII_S_TWO etc are defined for this. If the control number is 200 more than one of these button codes then the return value is 1 if the button i held and 0 otherwise. Macros with names like WII_H_TWO etc are defined for this. If the control number is 300 more than one of these button codes then the value is 1 if the button has just been released, and 0 otherwise. Macros with names like WII_R_TWO etc are defined for this. 20 (WII_PITCH): The pitch of the Wiimote. The value is in degrees between -90 and +90, unless modified by a wiirange call. 21 (WII_ROLL): The roll of the Wiimote. The value is in degrees between -90 and +90, unless modified by a wiirange call. 23 (WII_FORCE_X): The force applied to the Wiimote in the three axes. 24 (WII_FORCE_Y): 25 (WII_FORCE_Z): 26 (WII_FORCE_TOTAL): The total magnitude of the force applied to the Wiimote. 27 (WII_BATTERY): The percent of the battery that remains. 28 (WII_NUNCHUK_ANG): The angle of the nunchuk joystick in degrees. 29 (WII_NUNCHUK_MAG): The magnitude of the nunchuk joystick from neutral. 30 (WII_NUNCHUK_PITCH): The pitch of the nunchuk in degrees, in range -90 to +90 unless modified by a wiirange call. 31 (WII_NUNCHUK_ROLL): The roll of the nunchuk in degrees, in range -90 to +90 unless modified by a wiirange call. 33 (WII_NUNCHUK_Z): The state of the nunchuk Z button. 34 (WII_NUNCHUK_C): The state of the nunchuk C button. 35 (WII_IR1_X): The infrared pointing of the Wiimote. 36 (WII_IR1_Y): 37 (WII_IR1_Z): Examples See the example for wiiconnect. See Also wiiconnect, wiirange, wiisend, Credits Author: &namejohn; Codemist Ltd 2009 New in version 5.11 manual_src~dfsg/opcodes/ctrl7.xml0000644000000000000000000001160612262561504016162 0ustar rootroot ctrl7 Real-time MIDI:Input ctrl7 ctrl7 Allows a floating-point 7-bit MIDI signal scaled with a minimum and a maximum range. Description Allows a floating-point 7-bit MIDI signal scaled with a minimum and a maximum range. Syntax idest ctrl7 ichan, ictlno, imin, imax [, ifn] kdest ctrl7 ichan, ictlno, kmin, kmax [, ifn] adest ctrl7 ichan, ictlno, kmin, kmax [, ifn] [, icutoff] Initialization idest -- output signal ichan -- MIDI channel (1-16) ictlno -- MIDI controller number (0-127) imin -- user-defined minimum floating-point value of output imax -- user-defined maximum floating-point value of output ifn (optional) -- table to be read when indexing is required. Table must be normalized. Output is scaled according to imax and imin val. icutoff (optional) -- low pass filter cut-off frequency for smoothing a-rate output. Performance kdest, adest -- output signal kmin -- user-defined minimum floating-point value of output kmax -- user-defined maximum floating-point value of output ctrl7 (i- and k-rate 7 bit MIDI control) allows a floating-point 7-bit MIDI signal scaled with a minimum and a maximum range. It also allows optional non-interpolated table indexing. Minimum and maximum values can be varied at k-rate. ctrl7 differs from midic7 because it can be included in score-oriented instruments without Csound crashes. It also needs the additional parameter ichan containing the MIDI channel of the controller. The a-rate version of ctrl7 outputs an a-rate variable, which is low-pass filtered (smoothed). It contains an optional icutoff parameter, to set the cutoff frecuency for the low-pass filter. The default is 5. Examples Here is an example of the ctrl7 opcode. It uses the file ctrl7.csd. Example of the ctrl7 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. ctrl7 only outputs k values once the controller is first moved. To set an initial k-value, a call to initc7 is required. Ctrlinit does not work for this purpose. See Also ctrl14, ctrl21, initc7, initc14, initc21, midic7, midic14, midic21 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. The a-rate version of ctrl7 was added in version 5.06 manual_src~dfsg/opcodes/alpass.xml0000644000000000000000000001000612262561504016403 0ustar rootroot alpass Signal Modifiers:Reverberation alpass alpass Reverberates an input signal with a flat frequency response. Description Reverberates an input signal with a flat frequency response. Syntax ares alpass asig, xrvt, ilpt [, iskip] [, insmps] Initialization ilpt -- loop time in seconds, which determines the echo density of the reverberation. This in turn characterizes the color of the filter whose frequency response curve will contain ilpt * sr/2 peaks spaced evenly between 0 and sr/2 (the Nyquist frequency). Loop time can be as large as available memory will permit. The space required for an n second loop is 4n*sr bytes. The delay space is allocated and returned as in delay. iskip (optional, default=0) -- initial disposition of delay-loop data space (cf. reson). The default value is 0. insmps (optional, default=0) -- delay amount, as a number of samples. Performance xrvt -- the reverberation time (defined as the time in seconds for a signal to decay to 1/1000, or 60dB down from its original amplitude). This filter reiterates the input with an echo density determined by loop time ilpt. The attenuation rate is independent and is determined by xrvt, the reverberation time (defined as the time in seconds for a signal to decay to 1/1000, or 60dB down from its original amplitude). Output will begin to appear immediately. Examples Here is an example of the alpass opcode. It uses the file alpass.csd. Example of the alpass opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also comb, reverb, valpass, vcomb Credits Audio rate parameter introduced in version 6.02 November 2013. manual_src~dfsg/opcodes/xout.xml0000644000000000000000000000702312262561504016124 0ustar rootroot xout Signal I/O:Software Bus xout xout Retrieves variables from a user-defined opcode block, Description The xin and xout opcodes copy variables to and from the opcode definition, allowing communication with the calling instrument. The types of input and output variables are defined by the parameters intypes and outtypes. Notes xin and xout should be called only once, and xin should precede xout, otherwise an init error and deactivation of the current instrument may occur. These opcodes actually run only at i-time. Performance time copying is done by the user opcode call. This means that skipping xin or xout with kgoto has no effect, while skipping with igoto affects both init and performance time operation. Syntax xout xoutarg1 [, xoutarg2] ... [, xoutargN] Performance xoutarg1, xoutarg2, ... - output arguments. The number and type of variables must agree with the user-defined opcode's outtypes declaration. However, xout does not check for incorrect use of init-time and control-rate variables. The syntax of a user-defined opcode block is as follows: opcode name, outtypes, intypes xinarg1 [, xinarg2] [, xinarg3] ... [xinargN] xin [setksmps iksmps] ... the rest of the instrument's code. xout xoutarg1 [, xoutarg2] [, xoutarg3] ... [xoutargN] endop The new opcode can then be used with the usual syntax: [xinarg1] [, xinarg2] ... [xinargN] name [xoutarg1] [, xoutarg2] ... [xoutargN] [, iksmps] Examples See the example for the opcode opcode. See Also endop, opcode, setksmps, xin Credits Author: Istvan Varga, 2002; based on code by Matt J. Ingalls New in version 4.22 manual_src~dfsg/opcodes/delayr.xml0000644000000000000000000000517012262561504016406 0ustar rootroot delayr Signal Modifiers:Delay delayr delayr Reads from an automatically established digital delay line. Description Reads from an automatically established digital delay line. Syntax ares delayr idlt [, iskip] Initialization idlt -- requested delay time in seconds. This can be as large as available memory will permit. The space required for n seconds of delay is 4n * sr bytes. It is allocated at the time the instrument is first initialized, and returned to the pool at the end of a score section. iskip (optional, default=0) -- initial disposition of delay-loop data space (see reson). The default value is 0. Performance delayr reads from an automatically established digital delay line, in which the signal retrieved has been resident for idlt seconds. This unit must be paired with and precede an accompanying delayw unit. Any other Csound statements can intervene. Examples Here is an example of the delayr opcode. It uses the file delayr.csd. Example of the delayr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also delay, delay1, delayw manual_src~dfsg/opcodes/delayw.xml0000644000000000000000000000414712262561504016416 0ustar rootroot delayw Signal Modifiers:Delay delayw delayw Writes the audio signal to a digital delay line. Description Writes the audio signal to a digital delay line. Syntax delayw asig Performance delayw writes asig into the delay area established by the preceding delayr unit. Viewed as a pair, these two units permit the formation of modified feedback loops, etc. However, there is a lower bound on the value of idlt, which must be at least 1 control period (or 1/kr). Examples Here is an example of the delayw opcode. It uses the file delayw.csd. Example of the delayw opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also delay, delay1, delayr manual_src~dfsg/opcodes/scans.xml0000644000000000000000000001027112262561504016233 0ustar rootroot scans Signal Generators:Scanned Synthesis scans scans Generate audio output using scanned synthesis. Description Generate audio output using scanned synthesis. Syntax ares scans kamp, kfreq, ifn, id [, iorder] Initialization ifn -- ftable containing the scanning trajectory. This is a series of numbers that contains addresses of masses. The order of these addresses is used as the scan path. It should not contain values greater than the number of masses, or negative numbers. See the introduction to the scanned synthesis section. id -- ID number of the scanu opcode's waveform to use iorder (optional, default=0) -- order of interpolation used internally. It can take any value in the range 1 to 4, and defaults to 4, which is quartic interpolation. The setting of 2 is quadratic and 1 is linear. The higher numbers are slower, but not necessarily better. Performance kamp -- output amplitude. Note that the resulting amplitude is also dependent on instantaneous value in the wavetable. This number is effectively the scaling factor of the wavetable. kfreq -- frequency of the scan rate Examples Here is an example of the scanned synthesis. It uses the file scans.csd, and string-128.matrix. Example of the scans opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the scanned synthesis, using samples as excitation signal. It uses the file scans-2.csd. Second example of the scans opcode. See Also The matrix file string-128.matrix, as well as several other matrices, is also available in a zipped file from the Scanned Synthesis page at cSounds.com. More information on this opcode: http://www.csounds.com/stevenyi/scanned/yi_scannedSynthesis.html , written by Steven Yi Credits Author: &nameparis; MIT Media Lab Boston, Massachussetts USA New in Csound version 4.05 manual_src~dfsg/opcodes/clock.xml0000644000000000000000000000073412262561504016222 0ustar rootroot clock clock clock Deprecated. Description Deprecated. Use the rtclock opcode instead. manual_src~dfsg/opcodes/cabasa.xml0000644000000000000000000000730412262561504016341 0ustar rootroot cabasa Signal Generators:Models and Emulations cabasa cabasa Semi-physical model of a cabasa sound. Description cabasa is a semi-physical model of a cabasa sound. It is one of the PhISEM percussion opcodes. PhISEM (Physically Informed Stochastic Event Modeling) is an algorithmic approach for simulating collisions of multiple independent sound producing objects. Syntax ares cabasa iamp, idettack [, inum] [, idamp] [, imaxshake] Initialization iamp -- Amplitude of output. Note: As these instruments are stochastic, this is only a approximation. idettack -- period of time over which all sound is stopped inum (optional) -- The number of beads, teeth, bells, timbrels, etc. If zero, the default value is 512. idamp (optional) -- the damping factor, as part of this equation: damping_amount = 0.998 + (idamp * 0.002) The default damping_amount is 0.997 which means that the default value of idamp is -0.5. The maximum damping_amount is 1.0 (no damping). This means the maximum value for idamp is 1.0. The recommended range for idamp is usually below 75% of the maximum value. imaxshake (optional) -- amount of energy to add back into the system. The value should be in range 0 to 1. Examples Here is an example of the cabasa opcode. It uses the file cabasa.csd. Example of the cabasa opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also crunch, sandpaper, sekere, stix Credits Author: &nameperry;, part of the PhISEM (Physically Informed Stochastic Event Modeling) Adapted by &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.07 Added notes by &namerasmus; on May 2002. manual_src~dfsg/opcodes/linseg.xml0000644000000000000000000000665712262561504016422 0ustar rootroot linseg Signal Generators:Linear and Exponential Generators linseg linseg Trace a series of line segments between specified points. Description Trace a series of line segments between specified points. Syntax ares linseg ia, idur1, ib [, idur2] [, ic] [...] kres linseg ia, idur1, ib [, idur2] [, ic] [...] Initialization ia -- starting value. ib, ic, etc. -- value after dur1 seconds, etc. idur1 -- duration in seconds of first segment. A zero or negative value will cause all initialization to be skipped. idur2, idur3, etc. -- duration in seconds of subsequent segments. A zero or negative value will terminate the initialization process with the preceding point, permitting the last-defined line or curve to be continued indefinitely in performance. The default is zero. Performance These units generate control or audio signals whose values can pass through 2 or more specified points. The sum of dur values may or may not equal the instrument's performance time: a shorter performance will truncate the specified pattern, while a longer one will cause the last value to be repeated until the end of the note. Examples Here is an example of the linseg opcode. It uses the file linseg.csd. Example of the linseg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also expon, expseg, expsegb, expsegr, cosseg, line, linsegr transeg transegb Credits Author: &namebarry; manual_src~dfsg/opcodes/STKClarinet.xml0000644000000000000000000000717012262561504017253 0ustar rootroot STKClarinet Signal Generators:STK Opcodes STKClarinet STKClarinet STKClarinet uses a simple clarinet physical model. Description STKClarinet uses a simple clarinet physical model. Syntax asignal STKClarinet ifrequency, iamplitude, [kstiff, kv1[, knoise, kv2[, klfo, kv3[, klfodepth, kv4[, kbreath, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kstiff -- controller 2, reed stiffness. Value range of kv1 is 0-127. knoise -- controller 4, gain of noise. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kbreath -- controller 128, breath pressure. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the Clarinet class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKClarinet opcode. It uses the file STKClarinet.csd. Example of the STKClarinet opcode. See Also STKFlute. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/mxadsr.xml0000644000000000000000000001252512262561504016426 0ustar rootroot mxadsr Signal Generators:Envelope Generators mxadsr mxadsr Calculates the classical ADSR envelope using the expsegr mechanism. Description Calculates the classical ADSR envelope using the expsegr mechanism. Syntax ares mxadsr iatt, idec, islev, irel [, idel] [, ireltim] kres mxadsr iatt, idec, islev, irel [, idel] [, ireltim] Initialization iatt -- duration of attack phase idec -- duration of decay islev -- level for sustain phase irel -- duration of release phase idel (optional, default=0) -- period of zero before the envelope starts ireltim (optional, default=-1) -- Control release time after receiving a MIDI noteoff event. If less than zero, the longest release time given in the current instrument is used. If zero or more, the given value will be used for release time. Its default value is -1. (New in Csound 3.59 - not yet properly tested) Performance The envelope is in the range 0 to 1 and may need to be scaled further. The envelope may be described as: Picture of an ADSR envelope. Picture of an ADSR envelope. The length of the sustain is calculated from the length of the note. This means adsr is not suitable for use with MIDI events. The opcode madsr uses the linsegr mechanism, and so can be used in MIDI applications. The opcode mxadsr is identical to madsr except it uses exponential, rather than linear, line segments. You can use other pre-made envelopes which start a release segment upon recieving a note off message, like linsegr and expsegr, or you can construct more complex envelopes using xtratim and release. Note that you don't need to use xtratim if you are using mxadsr, since the time is extended automatically. mxadsr is new in Csound version 3.51. Examples Here is an example of the mxadsr opcode. It uses the file mxadsr.csd. Example of the mxadsr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also linsegr, expsegr, envlpxr, mxadsr, madsr, adsr, expon, expseg, expsega line, linseg, xtratim Credits Author: &namejohn; November 2002. Thanks to &namerasmus;, added documentation for the ireltim parameter. November 2003. Thanks to &namekanata;, fixed the link to the linsegr opcode. manual_src~dfsg/opcodes/dumpk3.xml0000644000000000000000000001150512262561504016330 0ustar rootroot dumpk3 Signal I/O:File I/O dumpk3 dumpk3 Periodically writes three orchestra control-signal values to an external file. Description Periodically writes three orchestra control-signal values to a named external file in a specific format. Syntax dumpk3 ksig1, ksig2, ksig3, ifilname, iformat, iprd Initialization ifilname -- character string (in double quotes, spaces permitted) denoting the external file name. May either be a full path name with target directory specified or a simple filename to be created within the current directory iformat -- specifies the output data format: 1 = 8-bit signed char(high order 8 bits of a 16-bit integer 4 = 16-bit short integers 5 = 32-bit long integers 6 = 32-bit floats 7 = ASCII long integers 8 = ASCII floats (2 decimal places) Note that A-law and U-law output are not available, and that all formats except the last two are binary. The output file contains no header information. iprd -- the period of ksig output in seconds, rounded to the nearest orchestra control period. A value of 0 implies one control period (the enforced minimum), which will create an output file sampled at the orchestra control rate. Performance ksig1, ksig2, ksig3 -- control-rate signals This opcode allows three generated control signal values to be saved in a named external file. The file contains no self-defining header information. But it contains a regularly sampled time series, suitable for later input or analysis. There may be any number of dumpk3 opcodes in an instrument or orchestra but each must write to a different file. Examples Here is an example of the dumpk3 opcode. It uses the file dumpk3.csd. Example of the dumpk3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output should include lines like these: WRITING: kfreq = 473.352855, kdb = -15.197657, kpan = 0.366764 kfreq = 473.352855, kdb = -15.197657, kpan = 0.366764 kfreq = 441.426368, kdb = -19.026206, kpan = 0.207327 kfreq = 452.965140, kdb = -21.447486, kpan = 0.553270 kfreq = 585.106328, kdb = -11.903852, kpan = 0.815665 kfreq = 482.056760, kdb = -4.046744, kpan = 0.876537 READING: kfreq = 473.352900, kdb = -15.197700, kpan = 0.366800 kfreq = 473.352900, kdb = -15.197700, kpan = 0.366800 kfreq = 441.426400, kdb = -19.026200, kpan = 0.207300 kfreq = 452.965100, kdb = -21.447500, kpan = 0.553300 kfreq = 585.106300, kdb = -11.903900, kpan = 0.815700 kfreq = 482.056800, kdb = -4.046700, kpan = 0.876500 See Also dumpk, dumpk2, dumpk4, readk, readk2, readk3, readk4 Credits By: John ffitch and Barry Vercoe 1999 or earlier manual_src~dfsg/opcodes/wiisend.xml0000644000000000000000000000524512262561504016573 0ustar rootroot wiisend Instrument Control:Sensing and Control wiisend wiisend Sends data to one of a number of external Nintendo Wiimote controllers. Description Sends data to one of a number of external Nintendo Wiimote controllers. Syntax kres wiisend kcontrol, kvalue[, knum] Initialization This opcode must be used in conjuction with a running wiiconnect opcode. Performance Please note that these opcodes are currently only supported on Linux. kcontrol -- the code for which control to write. kvalue -- the value to write to the control. knum -- the number of the which Wiimote to access, which defaults to the first one (zero). On each access a particular data item of the Wiimote is written. The currently implemented controls are given below, together with the macro name defined in the file wii_mac: 3 (WII_RUMBLE): starts or stops the Wiimote rumbling, depending on the value of kvalue (0 to stop, 1 to start). 4 (WII_SET_LEDS): set the four LED lights on the Wiimote to the binary representation of kvalue. Examples See the example for wiiconnect. See Also wiiconnect, wiidata, wiirange, Credits Author: &namejohn; Codemist Ltd 2009 New in version 5.11 manual_src~dfsg/opcodes/butbp.xml0000644000000000000000000000127012262561504016237 0ustar rootroot butbp Signal Modifiers:Standard Filters butbp butbp Same as the butterbp opcode. Description Same as the butterbp opcode. Syntax ares butbp asig, kfreq, kband [, iskip] manual_src~dfsg/opcodes/reson.xml0000644000000000000000000001015512262561504016253 0ustar rootroot reson Signal Modifiers:Standard Filters:Resonant reson reson A second-order resonant filter. Description A second-order resonant filter. Syntax ares reson asig, xcf, xbw [, iscl] [, iskip] Initialization iscl (optional, default=0) -- coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. (This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise.) A zero value signifies no scaling of the signal, leaving that to some later adjustment (see balance). The default value is 0. iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance ares -- the output signal at audio rate. asig -- the input signal at audio rate. xcf -- the center frequency of the filter, or frequency position of the peak response. xbw -- bandwidth of the filter (the Hz difference between the upper and lower half-power points). reson is a second-order filter in which kcf controls the center frequency, or frequency position of the peak response, and kbw controls its bandwidth (the frequency difference between the upper and lower half-power points). Examples Here is an example of the reson opcode. It uses the file reson.csd. Example of the reson opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also areson, aresonk, atone, atonek, port, portk, resonk, tone, tonek Credits Audio rate parameters introduced in version 6.02 November 2013. manual_src~dfsg/opcodes/date.xml0000644000000000000000000000457012262561504016046 0ustar rootroot date Instrument Control:Time Reading date date Returns the number seconds since a base date. Description Returns the number seconds since a base date, using the operating system's clock. The base is 1 January 1970 for Csound using doubles, and 1 January 2010 for versions using floats. Syntax ir date Initialization ir -- value at i-time, of the system clock in seconds since the start of the epoch. Note that the base date was originally 1970 but since verson 5.14 it has been changed as single precission floating point numbers are insufficient to indicate changes. Examples Here is an example of the date opcode. It uses the file date.csd. Example of the date opcode. Its output should include lines like this: instr 1: ii = 1165665152.000 Sat Dec 9 11:52:32 2006 Sat Dec 9 11:51:46 2006 Thu Jan 1 01:00:01 1970 See Also dates Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK December 2006 New in Csound version 5.05 Changed in Csound version 5.14 manual_src~dfsg/opcodes/outx.xml0000644000000000000000000000306412262561504016125 0ustar rootroot outx Signal I/O:Signal Output outx outx Writes 16-channel audio data to an external device or stream. Description Writes 16-channel audio data to an external device or stream. Syntax outx asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8, \ asig9, asig10, asig11, asig12, asig13, asig14, asig15, asig16 Performance outx outputs 16 channels of audio. See Also out32, outc, outch, outz Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/vstnote.xml0000644000000000000000000000505612262561504016633 0ustar rootroot vstnote Plugin Hosting:VST vstnote vstnote Sends a MIDI note with definite duration to a VST plugin. Syntax vstnote instance, kchan, knote, kveloc, kdur Description vstnote sends a MIDI note with definite duration to a VST plugin. Initialization instance - the number which identifies the plugin generated by vstinit. Performance kchan - The MIDI channel to trasnmit the note on. Note that MIDI channels are numbered starting from 0. knote - The MIDI note number to send. kveloc - The MIDI note's velocity. kdur - The MIDI note's duration in seconds. Note: Be sure the instrument containing vstnote is not finished before the duration of the note, otherwise you'll have a 'hung' note. Examples Here is an example of the use of the vstnote opcode. It uses the file vst4cs.csd. Example of the <emphasis>vstnote</emphasis> opcode See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits By: &nameandres; and &namemichael; Uses code from Hermann Seib's VSTHost and Thomas Grill's vst~ object. VST is a trademark of Steinberg Media Technologies GmbH. VST Plug-In Technology by Steinberg. manual_src~dfsg/opcodes/linsegb.xml0000644000000000000000000000632512262561504016554 0ustar rootroot linsegb Signal Generators:Linear and Exponential Generators linsegb linsegb Trace a series of line segments between specified absolute points. Description Trace a series of line segments between specified absolute points. Syntax ares linsegb ia, itim1, ib [, itim2] [, ic] [...] kres linsegb ia, itim1, ib [, itim2] [, ic] [...] Initialization ia -- starting value. ib, ic, etc. -- value at tim1 seconds, etc. itim1 -- time in seconds of end of first segment. A zero or negative value will cause all initialization to be skipped. itim2, itim3, etc. -- time in seconds at the end of subsequent segments. Performance These units generate control or audio signals whose values can pass through 2 or more specified points. The last tim value may or may not equal the instrument's performance time: a shorter performance will truncate the specified pattern, while a longer one will cause the last value to be repeated until the end of the note. Examples Here is an example of the linsegb opcode. It uses the file linsegb.csd. Example of the linsegb opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also expon, expseg, expsegr, line, linseg linsegr transeg Credits Author: &namevictor; June 2011 New in version 5.14 manual_src~dfsg/opcodes/chnparams.xml0000644000000000000000000000357712262561504017113 0ustar rootroot chnparams Signal I/O:Software Bus chnparams chnparams Query parameters of a channel. Description Query parameters of a channel (if it does not exist, all returned values are zero). Syntax itype, imode, ictltype, idflt, imin, imax chnparams Initialization itype -- channel data type (1: control, 2: audio, 3: string) imode -- sum of 1 for input and 2 for output ictltype -- special parameter for control channel only; if not available, set to zero. idflt -- special parameter for control channel only; if not available, set to zero. imin -- special parameter for control channel only; if not available, set to zero. imax -- special parameter for control channel only; if not available, set to zero. Credits Author: &nameistvan; 2005 manual_src~dfsg/opcodes/fluidEngine.xml0000644000000000000000000000627412262561504017365 0ustar rootroot fluidEngine Signal Generators:Sample Playback fluidEngine fluidEngine Instantiates a fluidsynth engine. Syntax ienginenum fluidEngine [iReverbEnabled] [, iChorusEnabled] [,iNumChannels] [, iPolyphony] Description Instantiates a fluidsynth engine, and returns ienginenum to identify the engine. ienginenum is passed to other other opcodes for loading and playing SoundFonts and gathering the generated sound. Initialization ienginenum -- engine number assigned from fluidEngine. iReverbEnabled -- optionally set to 0 to disable any reverb effect in the loaded SoundFonts. iChorusEnabled -- optionally set to 0 to disable any chorus effect in the loaded SoundFonts. iNumChannels -- number of channels to use; range is 16-256 and Csound default is 256 (Fluidsynth's default is 16). iPolyphony -- number of voices to be played in parallel; range is 16-4096 and Csound default is 4096 (Fluidsynth's default is 256). Note: this is not the number of notes played at the same time as a single note may use create multiple voices depending on instrument zones and velocity/key of played note. Examples Here is example of the fluidsynth opcodes using 2 engines. It uses the file fluidEngine.csd and midichn_advanced.mid. See Also fluidNote, fluidLoad More information on soundfonts in the Floss Manuals: http://en.flossmanuals.net/csound/ch048_d-reading-midi-files Other information on soundfonts on Wikipedia: http://en.wikipedia.org/wiki/Soundfont Credits Michael Gogins (gogins at pipeline dot com), Steven Yi. Thanks to Peter Hanappe for Fluidsynth. Optional iNumChannels and iPolyphony parameters added in 5.07 manual_src~dfsg/opcodes/slider8f.xml0000644000000000000000000001155312262561504016650 0ustar rootroot slider8f Real-time MIDI:Slider Banks slider8f slider8f Creates a bank of 8 different MIDI control message numbers, filtered before output. Description Creates a bank of 8 different MIDI control message numbers, filtered before output. Syntax k1,...,k8 slider8f ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1, \ ..., ictlnum8, imin8, imax8, init8, ifn8, icutoff8 Initialization ichan -- MIDI channel (1-16) ictlnum1 ... ictlnum8 -- MIDI control number (0-127) imin1 ... imin8 -- minimum values for each controller imax1 ... imax8 -- maximum values for each controller init1 ... init8 -- initial value for each controller ifn1 ... ifn8 -- function table for conversion for each controller icutoff1 ... icutoff8 -- low-pass filter cutoff frequency for each controller Performance k1 ... k8 -- output values slider8f is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider8f allows a bank of 8 different MIDI control message numbers. It filters the signal before output. This eliminates discontinuities due to the low resolution of the MIDI (7 bit). The cutoff frequency can be set separately for each controller (suggested range: .1 to 5 Hz). As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. slider8f opcodes do not output the required initial value immediately, but only after some k-cycles because the filter slightly delays the output. See Also s16b14, s32b14, slider16, slider16f, slider32, slider32f, slider64, slider64f, slider8 Credits Author: &namegabriel; Italy December 1998 New in Csound version 3.50 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/combinv.xml0000644000000000000000000001001612262561504016556 0ustar rootroot combinv Signal Modifiers:Reverberation combinv combinv Reverberates an input signal with a colored frequency response. Description Reverberates an input signal with a colored frequency response with a FIR filter. Syntax ares combinv asig, krvt, ilpt [, iskip] [, insmps] Initialization ilpt -- loop time in seconds, which determines the echo density of the reverberation. This in turn characterizes the color of the combinv filter whose frequency response curve will contain ilpt * sr/2 peaks spaced evenly between 0 and sr/2 (the Nyquist frequency). Loop time can be as large as available memory will permit. The space required for an n second loop is n*sr floating or double numbers (usually 4 or 8 bytes). Delay space is allocated and returned as in delay. iskip (optional, default=0) -- initial disposition of delay-loop data space (cf. reson). The default value is 0. insmps (optional, default=0) -- delay amount, as a number of samples. Performance krvt -- the reverberation time (defined as the time in seconds for a signal to decay to 1/1000, or 60dB down from its original amplitude). This filter reiterates input with an echo density determined by loop time ilpt. The attenuation rate is independent and is determined by krvt, the reverberation time (defined as the time in seconds for a signal to decay to 1/1000, or 60dB down from its original amplitude). Affect from a combinv filter will appear only after ilpt seconds. Examples Here is an example of the combinv opcode. It uses the file combinv.csd. Example of the combinv opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also alpass, comb reverb, valpass, vcomb Credits Author: &namejohn; October 2012 manual_src~dfsg/opcodes/ziwm.xml0000644000000000000000000000672012262561504016116 0ustar rootroot ziwm Zak Patch System ziwm ziwm Writes to a zk variable to an i-rate variable with mixing. Description Writes to a zk variable to an i-rate variable with mixing. Syntax ziwm isig, indx [, imix] Initialization isig -- initializes the value of the zk location. indx -- points to the zk location location to which to write. imix (optional, default=1) -- indicates if mixing should occur. Performance ziwm is a mixing opcode, it adds the signal to the current value of the variable. If no imix is specified, mixing always occurs. imix = 0 will cause overwriting like ziw, zkw, and zaw. Any other value will cause mixing. Caution: When using the mixing opcodes ziwm, zkwm, and zawm, care must be taken that the variables mixed to, are zeroed at the end (or start) of each k- or a-cycle. Continuing to add signals to them, can cause their values can drift to astronomical figures. One approach would be to establish certain ranges of zk or za variables to be used for mixing, then use zkcl or zacl to clear those ranges. Examples Here is an example of the ziwm opcode. It uses the file ziwm.csd. Example of the ziwm opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zaw, zawm, ziw, zkw, zkwm Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/hsboscil.xml0000644000000000000000000001166212262561504016737 0ustar rootroot hsboscil Signal Generators:Additive Synthesis/Resynthesis hsboscil hsboscil An oscillator which takes tonality and brightness as arguments. Description An oscillator which takes tonality and brightness as arguments, relative to a base frequency. Syntax ares hsboscil kamp, ktone, kbrite, ibasfreq, iwfn, ioctfn \ [, ioctcnt] [, iphs] Initialization ibasfreq -- base frequency to which tonality and brighness are relative iwfn -- function table of the waveform, usually a sine ioctfn -- function table used for weighting the octaves, usually something like: f1 0 1024 -19 1 0.5 270 0.5 ioctcnt (optional) -- number of octaves used for brightness blending. Must be in the range 2 to 10. Default is 3. iphs (optional, default=0) -- initial phase of the oscillator. If iphs = -1, initialization is skipped. Performance kamp -- amplitude of note ktone -- cyclic tonality parameter relative to ibasfreq in logarithmic octave, range 0 to 1, values > 1 can be used, and are internally reduced to frac(ktone). kbrite -- brightness parameter relative to ibasfreq, achieved by weighting ioctcnt octaves. It is scaled in such a way, that a value of 0 corresponds to the orignal value of ibasfreq, 1 corresponds to one octave above ibasfreq, -2 corresponds to two octaves below ibasfreq, etc. kbrite may be fractional. hsboscil takes tonality and brightness as arguments, relative to a base frequency (ibasfreq). Tonality is a cyclic parameter in the logarithmic octave, brightness is realized by mixing multiple weighted octaves. It is useful when tone space is understood in a concept of polar coordinates. Making ktone a line, and kbrite a constant, produces Risset's glissando. Oscillator table iwfn is always read interpolated. Performance time requires about ioctcnt * oscili. Examples Here is an example of the hsboscil opcode. It uses the file hsboscil.csd. Example of the hsboscil opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an example of the hsboscil opcode in a MIDI instrument. It uses the file hsboscil_midi.csd. Example of the hsboscil opcode in a MIDI instrument. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Peter Neubäcker Munich, Germany August, 1999 New in Csound version 3.58 manual_src~dfsg/opcodes/subinstrinit.xml0000644000000000000000000000461512262561504017666 0ustar rootroot subinstrinit Instrument Control:Subinstrument Control subinstrinit subinstrinit Creates and runs a numbered instrument instance at init-time. Description Same as subinstr, but init-time only and has no output arguments. Syntax subinstrinit instrnum [, p4] [, p5] [...] subinstrinit "insname" [, p4] [, p5] [...] Initialization instrnum -- Number of the instrument to be called. insname -- A string (in double-quotes) representing a named instrument. For more information about specifying input and output interfaces, see Calling an Instrument within an Instrument. Performance p4, p5, ... -- Additional input values the are mapped to the called instrument p-fields, starting with p4. The called instrument's p2 and p3 values will be identical to the host instrument's values. While the host instrument can control its own duration, any such attempts inside the called instrument will most likely have no effect. See Also Calling an Instrument within an Instrument, event, schedule, subinstr Credits New in version 4.23 manual_src~dfsg/opcodes/ictrl7.xml0000644000000000000000000000076012262561504016332 0ustar rootroot ictrl7 ictrl7 ictrl7 Deprecated. Description Deprecated as of version 3.52. Use the ctrl7 opcode instead. manual_src~dfsg/opcodes/vwrap.xml0000644000000000000000000000464212262561504016270 0ustar rootroot vwrap Vectorial:Limiting and Wrapping vwrap vwrap Limiting and Wrapping Vectorial Signals Description Wraps elements of vectorial control signals. Syntax vwrap ifn, kmin, kmax, ielements Initialization ifn - number of the table hosting the vector to be processed ielements - number of elements of the vector Performance kmin - minimum threshold value kmax - maximum threshold value vwrap wraps around each element of corresponding vector if it exceeds low or high thresholds. These opcodes are similar to limit, wrap and mirror, but operate with a vectorial signal instead of with a scalar signal. Result overrides old values of ifn1, if these are out of min/max interval. If you want to keep input vector, use vcopy opcode to copy it in another table. All these opcodes are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Note: bmscan not yet available on Canonical Csound Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/pvsfilter.xml0000644000000000000000000001034212262561504017141 0ustar rootroot pvsfilter Spectral Processing:Streaming pvsfilter pvsfilter Multiply amplitudes of a pvoc stream by those of a second pvoc stream, with dynamic scaling. Description Multiply amplitudes of a pvoc stream by those of a second pvoc stream, with dynamic scaling. Syntax fsig pvsfilter fsigin, fsigfil, kdepth[, igain] Performance fsig -- output pv stream fsigin -- input pv stream. fsigfil -- filtering pvoc stream. kdepth -- controls the depth of filtering of fsigin by fsigfil . igain -- amplitude scaling (optional, defaults to 1). Here the input pvoc stream amplitudes are modified by the filtering stream, keeping its frequencies intact. As usual, both signals have to be in the same format. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Example kfreq expon 500, p3, 4000 ; 3-octave sweep kdepth linseg 1, p3/2, 0.5, p3/2, 1 ; varying filter depth asig in ; input afil oscili 1, kfreq, 1 ; filter t-domain signal fim pvsanal asig,1024,256,1024,0 ; pvoc analysis fil pvsanal afil,1024,256,1024,0 fou pvsfilter fim, fil, kdepth ; filter signal aout pvsynth fou ; pvoc synthesis In the example above the filter curve will depend on the spectral envelope of afil; in the simple case of a sinusoid, it will be equivalent to a narrowband band-pass filter. Here is an example of the use of the pvsfilter opcode. It uses the file pvsfilter.csd. Example of the <emphasis>pvsfilter</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namevictor; November 2004 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/notequal.xml0000644000000000000000000000710212262561504016753 0ustar rootroot : ? conditional expressions != Instrument Control:Conditional Values != != Determines if one value is not equal to another. Description Determines if one value is not equal to another. Syntax (a != b ? v1 : v2) where a, b, v1 and v2 may be expressions, but a, b not audio-rate. Performance In the above conditional, a and b are first compared. If the indicated relation is true (a not equal to b), then the conditional expression has the value of v1; if the relation is false, the expression has the value of v2. NB.: If v1 or v2 are expressions, these will be evaluated before the conditional is determined. In terms of binding strength, all conditional operators (i.e. the relational operators (<, etc.), and ?, and : ) are weaker than the arithmetic and logical operators (+, -, *, /, && and ||). These are operators not opcodes. Therefore, they can be used within orchestra statements, but do not form complete statements themselves. Examples Here is an example of the != operator. It uses the file notequal.csd. Example of the != operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: ipch = 277.167 See Also ==, >=, >, <=, < manual_src~dfsg/opcodes/gausstrig.xml0000644000000000000000000000651412262561504017141 0ustar rootroot gausstrig Signal Generators:Random (Noise) Generators gausstrig gausstrig Random impulses around a certain frequency. Description Generates random impulses around a certain frequency. Syntax ares gausstrig kamp, kcps, kdev [, imode] kres gausstrig kamp, kcps, kdev [, imode] Initialization imode (optional, default=0) -- imode > 0 means better frequency modulation. If the frequency changes, the delay before the next impulse is calculed again. With the default mode we have the classic behavior of the GaussTrig ugen in SuperCollider, where the frequency modulation is bypassed during the delay time that precedes the next impulse. Performance kamp -- amplitude. kcps -- the mean frequency over which random impulses are distributed. kdev -- random deviation from mean (0 <= dev < 1). Examples Here is an example of the gausstrig opcode. It uses the file gausstrig.csd. Example of the gausstrig opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an example of the gausstrig opcode with imode = 1. It uses the file gausstrig-2.csd. Example of the gausstrig opcode with imode = 1. See Also dust dust2 mpulse Credits Based on Bhob Rainey's GaussTrig ugen (SuperCollider) Author: &nametito; January 2012 New in Csound version 5.16 manual_src~dfsg/opcodes/wgbow.xml0000644000000000000000000000626112262561504016255 0ustar rootroot wgbow Signal Generators:Waveguide Physical Modeling wgbow wgbow Creates a tone similar to a bowed string. Description Audio output is a tone similar to a bowed string, using a physical model developed from &nameperry;, but re-coded for Csound. Syntax ares wgbow kamp, kfreq, kpres, krat, kvibf, kvamp, ifn [, iminfreq] Initialization ifn -- table of shape of vibrato, usually a sine table, created by a function iminfreq (optional) -- lowest frequency at which the instrument will play. If it is omitted it is taken to be the same as the initial kfreq. If iminfreq is negative, initialization will be skipped. Performance A note is played on a string-like instrument, with the arguments as below. kamp -- amplitude of note. kfreq -- frequency of note played. kpres -- a parameter controlling the pressure of the bow on the string. Values should be about 3. The useful range is approximately 1 to 5. krat -- the position of the bow along the string. Usual playing is about 0.127236. The suggested range is 0.025 to 0.23. kvibf -- frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp -- amplitude of the vibrato Examples Here is an example of the wgbow opcode. It uses the file wgbow.csd. Example of the wgbow opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/iondur2.xml0000644000000000000000000000077712262561504016520 0ustar rootroot iondur2 iondur2 iondur2 Deprecated. Description Deprecated as of version 3.52. Use the noteondur2 opcode instead. manual_src~dfsg/opcodes/insglobal.xml0000644000000000000000000000714212262561504017101 0ustar rootroot insglobal Remote Opcodes insglobal insglobal An opcode which can be used to implement a remote orchestra. This opcode will send note events from a source machine to many destinations. Description With the insremot and insglobal opcodes you are able to perform instruments on remote machines and control them from a master machine. The remote opcodes are implemented using the master/client model. All the machines involved contain the same orchestra but only the master machine contains the information of the score. During the performance the master machine sends the note events to the clients. The insglobal opcode sends the events to all the machines involved in the remote concert. These machines are determined by the insremot definitions made above the insglobal command. To send events to only one machine use insremot. Syntax insglobal isource, instrnum [,instrnum...] Initialization isource -- a string that is the intended server computer (e.g. 192.168.0.100). This is the source host which generates the events of the given instrument(s) and sends it to all the machines involved in the remote concert. instrnum -- a list of instrument numbers which will be played on the destination machines Performance If is essential the computers using this opcode have the same byte-order, data size (double or float) and pointer size. One cannot use it with mixed 32 and 64 computers for instance. Internally this opcode makes use of the gethostname and gethostbyname functions to determine the client's and server's IP address for checking which messages are for which machine. If a computer has more than one IP address there is no way to control which IP address is found. The remote operation does not allow the sending of strings at all. Examples See the entry for insremot for an example of usage. See also insremot, midglobal, midremot, remoteport Credits Author: Simon Schampijer 2006 New in version 5.03 manual_src~dfsg/opcodes/imagegetpixel.xml0000644000000000000000000000660112262561504017752 0ustar rootroot imagegetpixel Image Processing Opcodes imagegetpixel imagegetpixel Return the RGB pixel values of a previously opened or created image. Description Return the RGB pixel values of a previously opened or created image. An image can be loaded with imageload. An empty image can be created with imagecreate. Syntax ared, agreen, ablue imagegetpixel iimagenum, ax, ay kred, kgreen, kblue imagegetpixel iimagenum, kx, ky Initialization iimagenum -- the reference of the image.. It should be a value returned by imageload or imagecreate. Performance ax (kx) -- horizontal pixel position (must be a float from 0 to 1). ay (ky) -- vertical pixel position (must be a float from 0 to 1). ared (kred) -- red value of the pixel (mapped to a float from 0 to 1). agreen (kgreen) -- green value of the pixel (mapped to a float from 0 to 1). ablue (kblue) -- blue value of the pixel (mapped to a float from 0 to 1). Examples Here is an example of the imagegetpixel opcode. It uses the files imageopcodesdemo2.csd test1.png and test2.png. Example of the imagegetpixel opcode. See Also imageload, imagecreate, imagesize, imagesave, imagesetpixel, imagefree Credits Author: Cesare Marilungo New in version 5.08 manual_src~dfsg/opcodes/tab.xml0000644000000000000000000000667112262561504015703 0ustar rootroot tabw_i tabw tab_i tab Signal Generators:Table Access tab tab Fast table opcodes. Description Fast table opcodes. Faster than table and tablew because don't allow wrap-around and limit and don't check index validity. Have been implemented in order to provide fast access to arrays. Support non-power of two tables (can be generated by any GEN function by giving a negative length value). Syntax ir tab_i indx, ifn[, ixmode] kr tab kndx, ifn[, ixmode] ar tab xndx, ifn[, ixmode] tabw_i isig, indx, ifn [,ixmode] tabw ksig, kndx, ifn [,ixmode] tabw asig, andx, ifn [,ixmode] Initialization ifn -- table number ixmode -- defaults to zero. If zero xndx and ixoff ranges match the length of the table; if non zero xndx and ixoff have a 0 to 1 range. isig -- input value to write. indx -- table index Performance asig, ksig -- input signal to write. andx, kndx -- table index. tab and tabw opcodes are similar to table and tablew, but are faster and support tables having non-power-of-two length. Special care of index value must be taken into account. Index values out of the table allocated space will crash Csound. Examples Here is an example of the tab opcode. It uses the file tab.csd. Example of the tab opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by &namegabriel;. manual_src~dfsg/opcodes/cpsmidi.xml0000644000000000000000000000732312262561504016560 0ustar rootroot cpsmidi Real-time MIDI:Converters cpsmidi cpsmidi Get the note number of the current MIDI event, expressed in cycles-per-second. Description Get the note number of the current MIDI event, expressed in cycles-per-second. Syntax icps cpsmidi Performance Get the note number of the current MIDI event, expressed in cycles-per-second units, for local processing. cpsmidi vs. cpsmidinn The cpsmidi opcode only produces meaningful results in a Midi-activated note (either real-time or from a Midi score with the -F flag). With cpsmidi, the Midi note number value is taken from the Midi event that is internally associated with the instrument instance. On the other hand, the cpsmidinn opcode may be used in any Csound instrument instance whether it is activated from a Midi event, score event, line event, or from another instrument. The input value for cpsmidinn might for example come from a p-field in a textual score or it may have been retrieved from the real-time Midi event that activated the current note using the notnum opcode. Examples Here is an example of the cpsmidi opcode. It uses the file cpsmidi.csd. Example of the cpsmidi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also aftouch, ampmidi, cpsmidib, cpstmid, midictrl, notnum, octmidi, octmidib, pchbend, pchmidi, pchmidib, veloc, cpsmidinn, octmidinn, pchmidinn Credits Author: &namebarry; - &namemike; MIT - Mills May 1997 manual_src~dfsg/opcodes/sflooper.xml0000644000000000000000000001244312262561504016760 0ustar rootroot sflooper Signal Generators:Sample Playback sflooper sflooper Plays a SoundFont2 (SF2) sample preset, generating a stereo sound, with user-defined time-varying crossfade looping. Description Plays a SoundFont2 (SF2) sample preset, generating a stereo sound, similarly to sfplay. Unlike that opcode, though, it ignores the looping points set in the SF2 file and substitutes them for a user-defined crossfade loop. It is a cross between sfplay and flooper2. Syntax ar1, ar2 sflooper ivel, inotenum, kamp, kpitch, ipreindex, kloopstart, kloopend, kcrossfade \ [, istart, imode, ifenv, iskip] Initialization ivel -- velocity value inotenum -- MIDI note number value ipreindex -- preset index istart -- playback start pos in seconds imode -- loop modes: 0 forward, 1 backward, 2 back-and-forth [def: 0] ifenv -- if non-zero, crossfade envelope shape table number. The default, 0, sets the crossfade to linear. iskip -- if 1, the opcode initialisation is skipped, for tied notes, performance continues from the position in the loop where the previous note stopped. The default, 0, does not skip initialisation Performance kamp -- amplitude scaling kpitch -- pitch control (transposition ratio); negative values are not allowed. kloopstart -- loop start point (secs). Note that although k-rate, loop parameters such as this are only updated once per loop cycle. If loop start is set beyond the end of the sample, no looping will result. kloopend -- loop end point (secs), updated once per loop cycle. kcrossfade -- crossfade length (secs), updated once per loop cycle and limited to loop length. sflooper plays a preset, generating a stereo sound. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. Note: The looping points are set on the root key of every sample that is part of the preset of the soundfont. For instance, a soundfont can have a single sample for the whole keyboard. In that case, sflooper will work like flooper and flooper2, because as the sample is transposed, played back at different rates, the loop will get short or longer. If however the soundfont has a sample for each key, than there will be no transposition and the loop will stay the same length (unless you change kpitch). Examples Here is an example of the sflooper opcode. It uses the file sflooper.csd. Example of the sflooper opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also sfilist, sfinstr, sfinstrm, sfload, sfpassign, sfplaym, sfplist, sfpreset Credits Author: &namevictor; August 2007 New in Csound Version 5.07 manual_src~dfsg/opcodes/lposcil.xml0000644000000000000000000000567012262561504016600 0ustar rootroot lposcil Signal Generators:Sample Playback lposcil lposcil Read sampled sound from a table with looping and high precision. Description Read sampled sound (mono or stereo) from a table, with looping, and high precision. Syntax ares lposcil kamp, kfreqratio, kloop, kend, ifn [, iphs] Initialization ifn -- function table number Performance kamp -- amplitude kfreqratio -- multiply factor of table frequency (for example: 1 = original frequency, 1.5 = a fifth up , .5 = an octave down) kloop -- start loop point (in samples) kend -- end loop point (in samples) lposcil (looping precise oscillator) allows varying at k-rate, the starting and ending point of a sample contained in a table (GEN01). This can be useful when reading a sampled loop of a wavetable, where repeat speed can be varied during the performance. Examples Here is an example of the lposcil opcode. It uses the file lposcil.csd. Example of the lposcil opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also lposcil3, lposcila, lposcilsa, lposcilsa2 Credits Author: &namegabriel; Italy New in Csound version 3.52 manual_src~dfsg/opcodes/JackoOn.xml0000644000000000000000000000404212262561504016447 0ustar rootroot JackoOn Jacko Opcodes JackoOn JackoOn Enables or disables all Jack ports. Description In the orchestra header, after all Jack connections have been created, enables or disables all Jack input and output opcodes inside this instance of Csound to read or write data. Syntax JackoOn [iactive] Initialization iactive -- A flag that turns the ports on (the default) or off. See Also JackoInit, JackoFreewheel, JackoAudioInConnect, JackoAudioOutConnect, JackoMidiInConnect, JackoMidiOutConnect, JackoAudioIn, JackoAudioOut, JackoMidiOut, JackoNoteOut, JackoTransport. Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/resonk.xml0000644000000000000000000000772312262561504016435 0ustar rootroot resonk Signal Modifiers:Standard Filters:Control resonk resonk A second-order resonant filter. Description A second-order resonant filter. Syntax kres resonk ksig, kcf, kbw [, iscl] [, iskip] Initialization iscl (optional, default=0) -- coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. (This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise.) A zero value signifies no scaling of the signal, leaving that to some later adjustment (see balance). The default value is 0. iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance kres -- the output signal at control-rate. ksig -- the input signal at control-rate. kcf -- the center frequency of the filter, or frequency position of the peak response. kbw -- bandwidth of the filter (the Hz difference between the upper and lower half-power points). resonk is like reson except its output is at control-rate rather than audio rate. Examples Here is an example of the resonk opcode. It uses the file resonk.csd. Example of the resonk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also areson, aresonk, atone, atonek, port, portk, reson, tone, tonek Credits Author: Robin Whittle Australia May 1997 manual_src~dfsg/opcodes/spat3dt.xml0000644000000000000000000002401012262561504016502 0ustar rootroot spat3dt Signal Modifiers:Panning and Spatialization spat3dt spat3dt Can be used to render an impulse response for a 3D space at i-time. Description This opcode positions the input sound in a 3D space, with optional simulation of room acoustics, in various output formats. spat3dt can be used to render the impulse response at i-time, storing output in a function table, suitable for convolution. Syntax spat3dt ioutft, iX, iY, iZ, idist, ift, imode, irlen [, iftnocl] Initialization ioutft -- Output ftable number for spat3dt. W, X, Y, and Z outputs are written interleaved to this table. If the table is too short, output will be truncated. iX -- Sound source X coordinate in meters (positive: right, negative: left) iY -- Sound source Y coordinate in meters (positive: front, negative: back) iZ -- Sound source Z coordinate in meters (positive: up, negative: down) idist -- For modes 0 to 3, idist is the unit circle distance in meters. For mode 4, idist is the distance between microphones. The following formulas describe amplitude and delay as a function of sound source distance from microphone(s): amplitude = 1 / (0.1 + distance) delay = distance / 340 (in seconds) Distance can be calculated as: distance = sqrt(iXˆ2 + iYˆ2 + iZˆ2) In Mode 4, distance can be calculated as: distance from left mic = sqrt((iX + idist/2)ˆ2 + iYˆ2 + iZˆ2) distance from right mic = sqrt((iX - idist/2)ˆ2 + iYˆ2 + iZˆ2) With spat3d the distance between the sound source and any microphone should be at least (340 * 18) / sr meters. Shorter distances will work, but may produce artifacts in some cases. There is no such limitation for spat3di and spat3dt. Sudden changes or discontinuities in sound source location can result in pops or clicks. Very fast movement may also degrade quality. ift -- Function table storing room parameters (for free field spatialization, set it to zero or negative). Table size is 54. The values in the table are: Room Parameter Purpose 0 Early reflection recursion depth (0 is the sound source, 1 is the first reflection etc.) for spat3d and spat3di. The number of echoes for four walls (front, back, right, left) is: N = (2*R + 2) * R. If all six walls are enabled: N = (((4*R + 6)*R + 8)*R) / 3 1 Late reflection recursion depth (used by spat3dt only). spat3dt skips early reflections and renders echoes up to this level. If early reflection depth is negative, spat3d and spat3di will output zero, while spat3dt will start rendering from the sound source. 2 imdel for spat3d. Overrides opcode parameter if non-negative. 3 irlen for spat3dt. Overrides opcode parameter if non-negative. 4 idist value. Overrides opcode parameter if >= 0. 5 Random seed (0 - 65535) -1 seeds from current time. 6 - 53 wall parameters (w = 6: ceil, w = 14: floor, w = 22: front, w = 30: back, w = 38: right, w = 46: left) w + 0 Enable reflections from this wall (0: no, 1: yes) w + 1 Wall distance from listener (in meters) w + 2 Randomization of wall distance (0 - 1) (in units of 1 / (wall distance)) w + 3 Reflection level (-1 - 1) w + 4 Parametric equalizer frequency in Hz. w + 5 Parametric equalizer level (1.0: no filtering) w + 6 Parametric equalizer Q (0.7071: no resonance) w + 7 Parametric equalizer mode (0: peak EQ, 1: low shelf, 2: high shelf) imode -- Output mode 0: B format with W output only (mono) aout = aW 1: B format with W and Y output (stereo) aleft = aW + 0.7071*aY aright = aW - 0.7071*aY 2: B format with W, X, and Y output (2D). This can be converted to UHJ: aWre, aWim hilbert aW aXre, aXim hilbert aX aYre, aYim hilbert aY aWXr = 0.0928*aXre + 0.4699*aWre aWXiYr = 0.2550*aXim - 0.1710*aWim + 0.3277*aYre aleft = aWXr + aWXiYr aright = aWXr - aWXiYr 3: B format with all outputs (3D) 4: Simulates a pair of microphones (stereo output) aW butterlp aW, ifreq ; recommended values for ifreq aY butterlp aY, ifreq ; are around 1000 Hz aleft = aW + aX aright = aY + aZ Mode 0 is the cheapest to calculate, while mode 4 is the most expensive. In Mode 4, The optional lowpass filters can change the frequency response depending on direction. For example, if the sound source is located left to the listener then the high frequencies are attenuated in the right channel and slightly increased in the left. This effect can be disabled by not using filters. You can also experiment with other filters (tone etc.) for better effect. Note that mode 4 is most useful for listening with headphones, and is also more expensive to calculate than the B-format (0 to 3) modes. The idist parameter in this case sets the distance between left and right microphone; for headphones, values between 0.2 - 0.25 are recommended, although higher settings up to 0.4 may be used for wide stereo effects. More information about B format can be found here: http://www.york.ac.uk/inst/mustech/3d_audio/ambis2.htm irlen -- Impulse response length of echoes (in seconds). Depending on filter parameters, values around 0.005-0.01 are suitable for most uses (higher values result in more accurate output, but slower rendering) iftnocl (optional, default=0) -- Do not clear output ftable (mix to existing data) if set to 1, clear table before writing if set to 0 (default: 0). Examples Here is an example of the spat3dt opcode. It uses the file spat3dt.csd. Example of the spat3dt opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also spat3d, spat3di Credits Author: &nameistvan; 2001 New in version 4.12 Updated April 2002 by &nameistvan; manual_src~dfsg/opcodes/outletkid.xml0000644000000000000000000000624412262561504017135 0ustar rootroot outletkid Signal Flow Graph Opcodes outletkid outletkid Sends a krate signal out from an instrument to a named port. Description Sends a krate signal out from an instrument to a named port. Syntax outletkid Sname, SinstanceID, ksignal Initialization Sname -- String name of the outlet port. The name of the outlet is implicitly qualified by the instrument name or number, so it is valid to use the same outlet name in more than one instrument (but not to use the same outlet name twice in one instrument). SinstanceID -- String name of the outlet port's instance ID. This enables the inlet to discriminate between different instances of the outlet, e,g. one instance of the outlet might be created by a note specifying one instance ID, and another instance might be created by a note specifying another ID. This might be used, e.g., to situate difference instances of an instrument at different points in an Ambisonic space in a spatializing effects processor. Performance ksignal -- krate output signal During performance, the krate output signal is sent to each instance of an instrument containing an inlet port to which this outlet has been connected using the connect opcode. The signals of all the outlets connected to an inlet, but only those that have a matching instance ID, are summed in the inlet. See Also outleta outletf inleta inletk inletkid inletf connect alwayson ftgenonce More information on this opcode: http://www.csounds.com/journal/issue13/signalFlowGraphOpcodes.html , written by Michael Gogins Credits By: &namemichael; 2009 manual_src~dfsg/opcodes/vco2ft.xml0000644000000000000000000000673312262561504016337 0ustar rootroot vco2ft Signal Generators:Dynamic Spectrum Oscillators vco2ft vco2ft Returns a table number at k-time for a given oscillator frequency and wavform. Description vco2ft returns the function table number to be used for generating the specified waveform at a given frequency. This function table number can be used by any Csound opcode that generates a signal by reading function tables (like oscilikt). The tables must be calculated by vco2init before vco2ft is called and shared as Csound ftables (ibasfn). Syntax kfn vco2ft kcps, iwave [, inyx] Initialization iwave -- the waveform for which table number is to be selected. Allowed values are: 0: sawtooth 1: 4 * x * (1 - x) (integrated sawtooth) 2: pulse (not normalized) 3: square wave 4: triangle Additionally, negative iwave values select user defined waveforms (see also vco2init). inyx (optional, default=0.5) -- bandwidth of the generated waveform, as percentage (0 to 1) of the sample rate. The expected range is 0 to 0.5 (i.e. up to sr/2), other values are limited to the allowed range. Setting inyx to 0.25 (sr/4), or 0.3333 (sr/3) can produce a fatter sound in some cases, although it is more likely to reduce quality. Performance kfn -- the ftable number, returned at k-rate. kcps -- frequency in Hz, returned at k-rate. Zero and negative values are allowed. However, if the absolute value exceeds sr/2 (or sr * inyx), the selected table will contain silence. Examples See the example for the vco2 opcode. See Also vco2ift, vco2init, and vco2. Credits Author: &nameistvan; New in version 4.22 manual_src~dfsg/opcodes/zarg.xml0000644000000000000000000000504712262561504016074 0ustar rootroot zarg Zak Patch System zarg zarg Reads from a location in za space at a-rate, adds some gain. Description Reads from a location in za space at a-rate, adds some gain. Syntax ares zarg kndx, kgain Initialization kndx -- points to the za location to be read. kgain -- multiplier for the a-rate signal. Performance zarg reads the array of floats at kndx in za space, which are ksmps number of a-rate floats to be processed in a k cycle. zarg also multiplies the a-rate signal by a k-rate value kgain. Examples Here is an example of the zarg opcode. It uses the file zarg.csd. Example of the zarg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zar, zir, zkr Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/koutpc.xml0000644000000000000000000000076212262561504016435 0ustar rootroot koutpc koutpc koutpc Deprecated. Description Deprecated as of version 3.52. Use the outkpc opcode instead. manual_src~dfsg/opcodes/miditempo.xml0000644000000000000000000000336412262561504017120 0ustar rootroot miditempo Instrument Control:Sensing and Control miditempo miditempo Returns the current tempo at k-rate, of either the MIDI file (if available) or the score Description Returns the current tempo at k-rate, of either the MIDI file (if available) or the score Syntax ksig miditempo Examples Here is an example of the miditempo opcode. It uses the files miditempo.csd. Example of the miditempo opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: miditempo = 96 Credits Author: &nameistvan; March 2005 New in Csound5 manual_src~dfsg/opcodes/cpstmid.xml0000644000000000000000000001047212262561504016572 0ustar rootroot cpstmid Real-time MIDI:Converters cpstmid cpstmid Get a MIDI note number (allows customized micro-tuning scales). Description This unit is similar to cpsmidi, but allows fully customized micro-tuning scales. Syntax icps cpstmid ifn Initialization ifn -- function table containing the parameters (numgrades, interval, basefreq, basekeymidi) and the tuning ratios. Performance Init-rate only cpsmid requires five parameters, the first, ifn, is the function table number of the tuning ratios, and the other parameters must be stored in the function table itself. The function table ifn should be generated by GEN02, with normalization inhibited. The first four values stored in this function are: numgrades -- the number of grades of the micro-tuning scale interval -- the frequency range covered before repeating the grade ratios, for example 2 for one octave, 1.5 for a fifth etc. basefreq -- the base frequency of the scale in Hz basekeymidi -- the MIDI note number to which basefreq is assigned unmodified After these four values, the user can begin to insert the tuning ratios. For example, for a standard 12 note scale with the base frequency of 261 Hz assigned to the key number 60, the corresponding f-statement in the score to generate the table should be: ; numgrades interval basefreq basekeymidi tuning ratios (equal temp) f1 0 64 -2 12 2 261 60 1 1.059463094359 1.122462048309 1.189207115003 ..etc... Another example with a 24 note scale with a base frequency of 440 assigned to the key number 48, and a repetition interval of 1.5: ; numgrades interval basefreq basekeymidi tuning-ratios (equal temp) f1 0 64 -2 24 1.5 440 48 1 1.01 1.02 1.03 ..etc... Examples Here is an example of the cpstmid opcode. It uses the file cpstmid.csd. Example of the cpstmid opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cpsmidi, GEN02 Credits Author: &namegabriel; Italy 1998 New in Csound version 3.492 manual_src~dfsg/opcodes/is32b14.xml0000644000000000000000000000076712262561504016224 0ustar rootroot is32b14 is32b14 is32b14 Deprecated. Description Deprecated as of version 3.52. Use the s32b14 opcode instead. manual_src~dfsg/opcodes/assign.xml0000644000000000000000000000627412262561504016420 0ustar rootroot = Instrument Control:Initialization and Reinitialization = = Performs a simple assignment. Syntax ares = xarg ires = iarg kres = karg ires, ... = iarg, ... kres, ... = karg, ... table [ kval] = karg Description Performs a simple assignment. Initialization = (simple assignment) - Put the value of the expression iarg (karg, xarg) into the named result. This provides a means of saving an evaluated result for later use. From version 5.13 onwards the i- and k-rate versions of assignment can take a number of outputs, and an equal or less number of inputs. If there are less the last value is repeated as necessary. From version 5.14 values can be assigned to elements of a vector with the square bracket form. Examples Here is an example of the assign opcode. It uses the file assign.csd. Example of the assign opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: i1 = 1234.000 See Also divz, init, passign, tival Credits Example written by &namekevin;. The extension to multiple values is by Author: &namejohn; University of Bath, and Codemist Ltd. Bath, UK February 2010 New in version 5.13 manual_src~dfsg/opcodes/granule.xml0000644000000000000000000002332312262561504016563 0ustar rootroot granule Signal Generators:Granular Synthesis granule granule A more complex granular synthesis texture generator. Description The granule unit generator is more complex than grain, but does add new possibilities. granule is a Csound unit generator which employs a wavetable as input to produce granularly synthesized audio output. Wavetable data may be generated by any of the GEN subroutines such as GEN01 which reads an audio data file into a wavetable. This enable a sampled sound to be used as the source for the grains. Up to 128 voices are implemented internally. The maximum number of voices can be increased by redefining the variable MAXVOICE in the grain4.h file. granule has a build-in random number generator to handle all the random offset parameters. Thresholding is also implemented to scan the source function table at initialization stage. This facilitates features such as skipping silence passage between sentences. The characteristics of the synthesis are controlled by 22 parameters. xamp is the amplitude of the output and it can be either audio rate or control rate variable. Syntax ares granule xamp, ivoice, iratio, imode, ithd, ifn, ipshift, igskip, \ igskip_os, ilength, kgap, igap_os, kgsize, igsize_os, iatt, idec \ [, iseed] [, ipitch1] [, ipitch2] [, ipitch3] [, ipitch4] [, ifnenv] Initialization ivoice -- number of voices. iratio -- ratio of the speed of the gskip pointer relative to output audio sample rate. eg. 0.5 will be half speed. imode -- +1 grain pointer move forward (same direction of the gskip pointer), -1 backward (oppose direction to the gskip pointer) or 0 for random. ithd -- threshold, if the sampled signal in the wavetable is smaller then ithd, it will be skipped. ifn -- function table number of sound source. ipshift -- pitch shift control. If ipshift is 0, pitch will be set randomly up and down an octave. If ipshift is 1, 2, 3 or 4, up to four different pitches can be set amount the number of voices defined in ivoice. The optional parameters ipitch1, ipitch2, ipitch3 and ipitch4 are used to quantify the pitch shifts. igskip -- initial skip from the beginning of the function table in sec. igskip_os -- gskip pointer random offset in sec, 0 will be no offset. ilength -- length of the table to be used starting from igskip in sec. igap_os -- gap random offset in % of the gap size, 0 gives no offset. igsize_os -- grain size random offset in % of grain size, 0 gives no offset. iatt -- attack of the grain envelope in % of grain size. idec -- decade of the grain envelope in % of grain size. iseed (optional, default=0.5) -- seed for the random number generator. ipitch1, ipitch2, ipitch3, ipitch4 (optional, default=1) -- pitch shift parameter, used when ipshift is set to 1, 2, 3 or 4. Time scaling technique is used in pitch shift with linear interpolation between data points. Default value is 1, the original pitch. ifnenv (optional, default=0) -- function table number to be used to generate the shape of the envelope. Performance xamp -- amplitude. kgap -- gap between grains in sec. kgsize -- grain size in sec. Examples Here is an example of the granule opcode. It uses the file granule.csd, and mary.wav. Example of the granule opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The above example reads a sound file called mary.wav into wavetable number 1 with 262,144 samples. It generates 10 seconds of stereo audio output using the wavetable. In the orchestra file, all parameters required to control the synthesis are passed from the score file. A linseg function generator is used to generate an envelope with 0.5 second of linear attack and decay. Stereo effect is generated by using different seeds for the two granule function calls. In the example, 0.17 is added to p20 before passing into the second granule call to ensure that all of the random offset events are different from the first one. In the score file, the parameters are interpreted as: Parameter Interpreted As p5 (ivoice) the number of voices is set to 64 p6 (iratio) set to 0.5, it scans the wavetable at half of the speed of the audio output rate p7 (imode) set to 0, the grain pointer only move forward p8 (ithd) set to 0, skipping the thresholding process p9 (ifn) set to 1, function table number 1 is used p10 (ipshift) set to 4, four different pitches are going to be generated p11 (igskip) set to 0 and p12 (igskip_os) is set to 0.005, no skipping into the wavetable and a 5 mSec random offset is used p13 (ilength) set to 5, 5 seconds of the wavetable is to be used p14 (kgap) set to 0.01 and p15 (igap_os) is set to 50, 10 mSec gap with 50% random offset is to be used p16 (kgsize) set to 0.02 and p17 (igsize_os) is set to 50, 20 mSec grain with 50% random offset is used p18 (iatt) and p19 (idec) set to 30, 30% of linear attack and decade is applied to the grain p20 (iseed) seed for the random number generator is set to 0.39 p21 - p24 pitches set to 1 which is the original pitch, 1.42 which is a 5th up, 0.29 which is a 7th down and finally 2 which is an octave up. Credits Author: Allan Lee Belfast 1996 New in version 3.35 manual_src~dfsg/opcodes/pvsmooth.xml0000644000000000000000000001004712262561504017004 0ustar rootroot pvsmooth Spectral Processing:Streaming pvsmooth pvsmooth Smooth the amplitude and frequency time functions of a pv stream using parallel 1st order lowpass IIR filters with time-varying cutoff frequency. Description Smooth the amplitude and frequency time functions of a pv stream using a 1st order lowpass IIR with time-varying cutoff frequency. This opcode uses the same filter as the tone opcode, but this time acting separately on the amplitude and frequency time functions that make up a pv stream. The cutoff frequency parameter runs at the control-rate, but unlike tone and tonek, it is not specified in Hz, but as fractions of 1/2 frame-rate (actually the pv stream sampling rate), which is easier to understand. This means that the highest cutoff frequency is 1 and the lowest 0; the lower the frequency the smoother the functions and more pronounced the effect will be. These are filters applied to control signals so the effect is basically blurring the spectral evolution. The effects produced are more or less similar to pvsblur, but with two important differences: 1.smoothing of amplitudes and frequencies use separate sets of filters; and 2. there is no increase in computational cost when higher amounts of 'blurring' (smoothing) are desired. Syntax fsig pvsmooth fsigin, kacf, kfcf Performance fsig -- output pv stream fsigin -- input pv stream. kacf -- amount of cutoff frequency for amplitude function filtering, between 0 and 1, in fractions of 1/2 frame-rate. kfcf -- amount of cutoff frequency for frequency function filtering, between 0 and 1, in fractions of 1/2 frame-rate. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the pvsmooth opcode. It uses the file pvsmooth.csd. Example of the pvsmooth opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The formula for calculating the cutoff frequency of the filter: frame rate / hopsize = new frame starts per second (in Hz), then the percentage of half the framerate. For example, looking at the first note in the example, the frame rate is 44100 / 256 = 172,265625 Hz (= 172 new frame starts per second). half of the frame rate is about 86 Hz, and one percent of this is 0.86 Hz. Credits Author: &namevictor; May 2006 New plugin in version 5 May 2006. manual_src~dfsg/opcodes/outs1.xml0000644000000000000000000000615512262561504016205 0ustar rootroot outs1 Signal I/O:Signal Output outs1 outs1 Writes samples to stereo channel 1 of an external device or stream. Description Writes samples to stereo channel 1 of an external device or stream. Syntax outs1 asig Performance Sends audio samples to an accumulating output buffer (created at the beginning of performance) which serves to collect the output of all active instruments before the sound is written to disk. There can be any number of these output units in an instrument. The type (mono, stereo, quad, hex, or oct) should agree with nchnls. But as of version 3.50, Csound will attempt to change an incorrect opcode to agree with the nchnls statement. Opcodes can be chosen to direct sound to any particular channel: outs1 sends to stereo channel 1, outq3 to quad channel 3, etc. Examples Here is an example of the outs1 opcode. It uses the file outs1.csd. Example of the outs1 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also out, outh, outo, outq, outq1, outq2, outq3, outq4, outs, outs2, soundout Credits Author: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 manual_src~dfsg/opcodes/ictrl21.xml0000644000000000000000000000076712262561504016415 0ustar rootroot ictrl21 ictrl21 ictrl21 Deprecated. Description Deprecated as of version 3.52. Use the ctrl21 opcode instead. manual_src~dfsg/opcodes/imidic21.xml0000644000000000000000000000077612262561504016536 0ustar rootroot imidic21 imidic21 imidic21 Deprecated. Description Deprecated as of version 3.52. Use the midic21 opcode instead. manual_src~dfsg/opcodes/pvstencil.xml0000644000000000000000000000706212262561504017137 0ustar rootroot pvstencil Spectral Processing:Streaming pvstencil pvstencil Transforms a pvoc stream according to a masking function table. Description Transforms a pvoc stream according to a masking function table; if the pvoc stream amplitude falls below the value of the function for a specific pvoc channel, it applies a gain to that channel. The pvoc stream amplitudes are compared to a masking table, if the fall below the table values, they are scaled by kgain. Prior to the operation, table values are scaled by klevel, which can be used as masking depth control. Tables have to be at least fftsize/2 in size; for most GENS it is important to use an extended-guard point (size power-of-two plus one), however this is not necessary with GEN43. One of the typical uses of pvstencil would be in noise reduction. A noise print can be analysed with pvanal into a PVOC-EX file and loaded in a table with GEN43. This then can be used as the masking table for pvstencil and the amount of reduction would be controlled by kgain. Skipping post-normalisation will keep the original noise print average amplitudes. This would provide a good starting point for a successful noise reduction (so that klevel can be generally set to close to 1). Other possible transformation effects are possible, such as filtering and `inverse-masking'. Syntax fsig pvstencil fsigin, kgain, klevel, iftable Performance fsig -- output pv stream fsigin -- input pv stream. kgain -- `stencil' gain. klevel -- masking function level (scales the ftable prior to `stenciling') . iftable -- masking function table. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Example fsig pvsanal asig, 1024, 256, 1024, 1 fclean pvstencil fsig, 0, 1, 1 aclean pvsynth fclean Credits Author: &namevictor; November 2004 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/fllabel.xml0000644000000000000000000001325012262561504016525 0ustar rootroot FLlabel FLTK:Appearance FLlabel FLlabel A FLTK opcode that modifies the appearance of a text label. Description Modifies a set of parameters related to the text label appearence of a widget (i.e. size, font, alignment and color of corresponding text). Syntax FLlabel isize, ifont, ialign, ired, igreen, iblue Initialization isize -- size of the font of the target widget. Normal values are in the order of 15. Greater numbers enlarge font size, while smaller numbers reduce it. ifont -- sets the the font type of the label of a widget. Legal values for ifont argument are: 1 - Helvetica (same as Arial under Windows) 2 - Helvetica Bold 3 - Helvetica Italic 4 - Helvetica Bold Italic 5 - Courier 6 - Courier Bold 7 - Courier Italic 8 - Courier Bold Italic 9 - Times 10 - Times Bold 11 - Times Italic 12 - Times Bold Italic 13 - Symbol 14 - Screen 15 - Screen Bold 16 - Dingbats ialign -- sets the alignment of the label text of the widget. Legal values for ialign argument are: 1 - align center 2 - align top 3 - align bottom 4 - align left 5 - align right 6 - align top-left 7 - align top-right 8 - align bottom-left 9 - align bottom-right ired -- The red color of the target widget. The range for each RGB component is 0-255 igreen -- The green color of the target widget. The range for each RGB component is 0-255 iblue -- The blue color of the target widget. The range for each RGB component is 0-255 Performance FLlabel modifies a set of parameters related to the text label appearance of a widget, i.e. size, font, alignment and color of corresponding text. This opcode affects (almost) all widgets defined next its location. A user can put several instances of FLlabel in front of each widget he intends to modify. However, to modify a particular widget, it is better to use the opcode belonging to the second type (i.e. those containing the ihandle argument). The influence of FLlabel on the next widget can be turned off by using -1 as its only argument. FLlabel is designed to modify text attributes of a group of related widgets. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/flsettextsize.xml0000644000000000000000000000560112262561504020042 0ustar rootroot FLsetTextSize FLTK:Appearance FLsetTextSize FLsetTextSize Sets the size of the text label of a FLTK widget. Description FLsetTextSize sets the size of the text label of the target widget. Syntax FLsetTextSize isize, ihandle Initialization isize -- size of the font of the target widget. Normal values are in the order of 15. Greater numbers enlarge font size, while smaller numbers reduce it. ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/transegb.xml0000644000000000000000000001003412262561504016726 0ustar rootroot transegb Signal Generators:Linear and Exponential Generators transegb transegb Constructs a user-definable envelope in absolute time. Description Constructs a user-definable envelope in absolute time. Syntax ares transegb ia, itim, itype, ib [, itim2] [, itype] [, ic] ... kres transegb ia, itim, itype, ib [, itim2] [, itype] [, ic] ... Initialization ia -- starting value. ib, ic, etc. -- value after itim seconds. itim -- time in seconds of end of first segment. itim2,... itimx etc. -- time in seconds at the end of the segment. itype, itype2, etc. -- if 0, a straight line is produced. If non-zero, then transegb creates the following curve, for n steps: ibeg + (ivalue - ibeg) * (1 - exp( i*itype/(n-1) )) / (1 - exp(itype)) Performance If itype > 0, there is a slowly rising (concave) or slowly decaying (convex) curve, while if itype < 0, the curve is fast rising (convex) or fast decaying (concave). See also GEN16. Examples Here is an example of the transegb opcode. It uses the file transegb.csd. The example produces the following output: Output of the transegb example. Output of the transegb example. Example of the transegb opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also expseg, expsega, expsega, expsegr, linseg, linsegb, linsegr, transeg transegr Credits Author: &namejohn; University of Bath, Codemist. Ltd. Bath, UK June 2011 New in Csound version 5.14 manual_src~dfsg/opcodes/lowpass2.xml0000644000000000000000000000632612262561504016704 0ustar rootroot lowpass2 Signal Modifiers:Standard Filters:Resonant lowpass2 lowpass2 A resonant lowpass filter. Description Implementation of a resonant second-order lowpass filter. Syntax ares lowpass2 asig, kcf, kq [, iskip] Initialization iskip -- initial disposition of internal data space. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance asig -- input signal to be filtered kcf -- cutoff or resonant frequency of the filter, measured in Hz kq -- Q of the filter, defined, for bandpass filters, as bandwidth/cutoff. kq should be between 1 and 500 lowpass2 is a second order IIR lowpass filter, with k-rate controls for cutoff frequency (kcf) and Q (kq). As kq is increased, a resonant peak forms around the cutoff frequency, transforming the lowpass filter response into a response that is similar to a bandpass filter, but with more low frequency energy. This corresponds to an increase in the magnitude and "sharpness" of the resonant peak. For high values of kq, a scaling function such as balance may be required. In practice, this allows for the simulation of the voltage-controlled filters of analog synthesizers, or for the creation of a pitch of constant amplitude while filtering white noise. Examples Here is an example of the lowpass2 opcode. It uses the file lowpass2.csd. Example of the lowpass2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namesean; Seattle, Washington August 1999 New in Csound version 4.0 manual_src~dfsg/opcodes/flcolor.xml0000644000000000000000000001120512262561504016562 0ustar rootroot FLcolor FLTK:Appearance FLcolor FLcolor A FLTK opcode that sets the primary colors. Description Sets the primary colors to RGB values given by the user. Syntax FLcolor ired, igreen, iblue [, ired2, igreen2, iblue2] Initialization ired -- The red color of the target widget. The range for each RGB component is 0-255 igreen -- The green color of the target widget. The range for each RGB component is 0-255 iblue -- The blue color of the target widget. The range for each RGB component is 0-255 ired2 -- The red component for the secondary color of the target widget. The range for each RGB component is 0-255 igreen2 -- The green component for the secondary color of the target widget. The range for each RGB component is 0-255 iblue2 -- The blue component for the secondary color of the target widget. The range for each RGB component is 0-255 Performance These opcodes modify the appearance of other widgets. There are two types of such opcodes, those that don't contain the ihandle argument which affect all subsequently declared widgets, and those with ihandle which affect only a target widget previously defined. FLcolor sets the primary colors to RGB values given by the user. This opcode affects the primary color of (almost) all widgets defined next its location. User can put several instances of FLcolor in front of each widget he intend to modify. However, to modify a single widget, it would be better to use the opcode belonging to the second type (i.e. those containing ihandle argument). FLcolor is designed to modify the colors of a group of related widgets that assume the same color. The influence of FLcolor on subsequent widgets can be turned off by using -1 as the only argument of the opcode. Also, using -2 (or -3) as the only value of FLcolor makes all next widget colors randomly selected. The difference is that -2 selects a light random color, while -3 selects a dark random color. Using ired2, igreen2, iblue2 is equivalent to using a separate FLcolor2. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/octmidib.xml0000644000000000000000000000627712262561504016731 0ustar rootroot octmidib Real-time MIDI:Converters octmidib octmidib Get the note number of the current MIDI event and modify it by the current pitch-bend value, express it in octave-point-decimal. Description Get the note number of the current MIDI event and modify it by the current pitch-bend value, express it in octave-point-decimal. Syntax ioct octmidib [irange] koct octmidib [irange] Initialization irange (optional) -- the pitch bend range in semitones Performance Get the note number of the current MIDI event, modify it by the current pitch-bend value, and express the result in octave-point-decimal units. Available as an i-time value or as a continuous k-rate value. Examples Here is an example of the octmidib opcode. It uses the file octmidib.csd. Example of the octmidib opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also aftouch, ampmidi, cpsmidi, cpsmidib, midictrl, notnum, octmidi, pchbend, pchmidi, pchmidib, veloc Credits Author: &namebarry; - &namemike; MIT - Mills May 1997 Example written by &namekevin;. manual_src~dfsg/opcodes/pvsdisp.xml0000644000000000000000000000541112262561504016614 0ustar rootroot pvsdisp Spectral Processing:Streaming pvsdisp pvsdisp Displays a PVS signal as an amplitude vs. freq graph. Description This opcode will display a PVS signal fsig. Uses X11 or FLTK windows if enabled, else (or if -g flag is set) displays are approximated in ASCII characters. Syntax pvsdisp fsig[, ibins, iwtflg] Initialization iprd -- the period of pvsdisp in seconds. ibins (optional, default=all bins) -- optionally, display only ibins bins. iwtflg (optional, default=0) -- wait flag. If non-zero, each pvsdisp is held until released by the user. The default value is 0 (no wait). Performance pvsdisp -- displays the PVS signal frame-by-frame. Examples Here is an example of the pvsdisp opcode. It uses the file pvsdisp.csd. This example uses realtime input, but you can also use it for soundfile input. Example of the pvsdisp opcode See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, dispfft, print, pvsadsyn Credits Author: &namevictor;, 2006 manual_src~dfsg/opcodes/guiro.xml0000644000000000000000000000711712262561504016256 0ustar rootroot guiro Signal Generators:Models and Emulations guiro guiro Semi-physical model of a guiro sound. Description guiro is a semi-physical model of a guiro sound. It is one of the PhISEM percussion opcodes. PhISEM (Physically Informed Stochastic Event Modeling) is an algorithmic approach for simulating collisions of multiple independent sound producing objects. Syntax ares guiro kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] [, ifreq1] Initialization idettack -- period of time over which all sound is stopped inum (optional) -- The number of beads, teeth, bells, timbrels, etc. If zero, the default value is 128. idamp (optional) -- the damping factor of the instrument. Not used. imaxshake (optional, default=0) -- amount of energy to add back into the system. The value should be in range 0 to 1. ifreq (optional) -- the main resonant frequency. The default value is 2500 Hz. ifreq1 (optional) -- the first resonant frequency. Default value here is 4000 Hz Performance kamp -- Amplitude of output. Note: As these instruments are stochastic, this is only an approximation. Examples Here is an example of the guiro opcode. It uses the file guiro.csd. Example of the guiro opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also bamboo, dripwater, sleighbells, tambourine Credits Author: &nameperry;, part of the PhISEM (Physically Informed Stochastic Event Modeling) Adapted by &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.07 Added notes by &namerasmus; on May 2002. manual_src~dfsg/opcodes/trmix.xml0000644000000000000000000000475712262561504016303 0ustar rootroot trmix Spectral Processing:Streaming trmix trmix Streaming partial track mixing. Description The trmix opcode takes two inputs containg TRACKS pv streaming signals (as generated, for instance by partials) and mixes them into a single TRACKS stream. Tracks will be mixed up to the available space (defined by the original number of FFT bins in the analysed signals). If the sum of the input tracks exceeds this space, the higher-ordered tracks in the second input will be pruned. Syntax fsig trmix fin1, fin2 Performance fsig -- output pv stream in TRACKS format fin1 -- first input pv stream in TRACKS format. fin2 -- second input pv stream in TRACKS format Examples Here is an example of the trmix opcode. It uses the file trmix.csd. Example of the trmix opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of an ifd-analysis signal, frequency splitting and pitch shifting of the upper part of the spectrum, followed by the remix of the two parts of the spectrum and resynthesis. Credits Author: &namevictor; February 2006 New in Csound 5.01 manual_src~dfsg/opcodes/outkc14.xml0000644000000000000000000000742212262561504016422 0ustar rootroot outkc14 Real-time MIDI:Output outkc14 outkc14 Sends 14-bit MIDI controller output at k-rate. Description Sends 14-bit MIDI controller output at k-rate. Syntax outkc14 kchn, kmsb, klsb, kvalue, kmin, kmax Performance kchn -- MIDI channel number (1-16) kmsb -- most significant byte controller number when using 14-bit parameters (0-127) klsb -- least significant byte controller number when using 14-bit parameters (0-127) kvalue -- floating point value kmin -- minimum floating point value (converted in MIDI integer value 0) kmax -- maximum floating point value (converted in MIDI integer value 16383 (14-bit)) outkc14 (k-rate MIDI 14-bit controller output) sends a pair of controller messages. It works only with MIDI instruments which recognize them. These opcodes can drive 14-bit parameters on MIDI instruments that recognize them. The first control message contains the most significant byte of kvalue argument while the second message contains the less significant byte. kmsb and klsb are the number of the most and less significant controller. It can drive a different value of a parameter for each note currently active. It can scale the k-value floating-point argument according to the kmin and kmax values. For example: set kmin = 1.0 and kmax = 2.0. When the kvalue argument receives a 2.0 value, the opcode will send a 127 value to the MIDI OUT device. When the kvalue argument receives a 1.0 value, it will send a 0 value. k-rate opcodes send a message each time the MIDI converted value of argument kvalue changes. See Also outiat, outic14, outic, outipat, outipb, outipc, outkat, outkc, outkpat, outkpb, outkpc Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/trhighest.xml0000644000000000000000000000506212262561504017127 0ustar rootroot trhighest Spectral Processing:Streaming trhighest trhighest Extracts the highest-frequency track from a streaming track input signal. Description The trhighest opcode takes an input containg TRACKS pv streaming signals (as generated, for instance by partials) and outputs only the highest track. In addition it outputs two k-rate signals, corresponding to the frequency and amplitude of the highest track signal. Syntax fsig, kfr, kamp trhighest fin1, kscal Performance fsig -- output pv stream in TRACKS format kfr -- frequency (in Hz) of the highest-frequency track kamp -- amplitude of the highest-frequency track fin -- input pv stream in TRACKS format. kscal -- amplitude scaling of output. Examples Here is an example of the trhighest opcode. It uses the file trhighest.csd. Example of the trhighest opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of an ifd-analysis signal, extraction of the highest frequency and resynthesis. Credits Author: &namevictor; February 2006 New in Csound 5.01 manual_src~dfsg/opcodes/flpackend.xml0000644000000000000000000000370312262561504017055 0ustar rootroot FLpackEnd FLTK:Containers FLpackEnd FLpackEnd Marks the end of a group of compressed or aligned FLTK widgets. Description Marks the end of a group of compressed or aligned FLTK widgets. Syntax FLpackEnd Performance Containers are useful to format the graphic appearance of the widgets. The most important container is FLpanel, that actually creates a window. It can be filled with other containers and/or valuators or other kinds of widgets. There are no k-rate arguments in containers. See Also FLgroup, FLgroupEnd, FLpack, FLpanel, FLpanelEnd, FLscroll, FLscrollEnd, FLtabs, FLtabsEnd Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/itablecopy.xml0000644000000000000000000000101612262561504017254 0ustar rootroot itablecopy itablecopy itablecopy Deprecated. Description Deprecated as of version 3.52. Use the tableicopy opcode instead. manual_src~dfsg/opcodes/STKBowed.xml0000644000000000000000000000665312262561504016557 0ustar rootroot STKBowed Signal Generators:STK Opcodes STKBowed STKBowed STKBowed is a bowed string instrument. Description STKBowed is a bowed string instrument, using a waveguide model. Syntax asignal STKBowed ifrequency, iamplitude, [kpress, kv1[, kpos, kv2[, klfo, kv3[, klfodepth, kv4[, kvol, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kpress -- controller 2, bow pressure. Value range of kv1 is 0-127. kpos -- controller 4, position on bow. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kvol -- controller 128, volume. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the Bowed class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKBowed opcode. It uses the file STKBowed.csd. Example of the STKBowed opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/sfinstr3m.xml0000644000000000000000000001411312262561504017053 0ustar rootroot sfinstr3m Signal Generators:Sample Playback sfinstr3m sfinstr3m Plays a SoundFont2 (SF2) sample instrument, generating a mono sound with cubic interpolation. Description Plays a SoundFont2 (SF2) sample instrument, generating a mono sound with cubic interpolation. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. Syntax ares sfinstr3m ivel, inotenum, xamp, xfreq, instrnum, ifilhandle \ [, iflag] [, ioffset] Initialization ivel -- velocity value inotenum -- MIDI note number value instrnum -- number of an instrument of a SF2 file. ifilhandle -- unique number generated by sfload opcode to be used as an identifier for a SF2 file. Several SF2 files can be loaded and activated at the same time. iflag (optional) -- flag regarding the behavior of xfreq and inotenum ioffset (optional) -- start playing at offset, in samples. Performance xamp -- amplitude correction factor xfreq -- frequency value or frequency multiplier, depending by iflag. When iflag = 0, xfreq is a multiplier of a the default frequency, assigned by SF2 preset to the inotenum value. When iflag = 1, xfreq is the absolute frequency of the output sound, in Hz. Default is 0. When iflag = 0, inotenum sets the frequency of the output according to the MIDI note number used, and xfreq is used as a multiplier. When iflag = 1, the frequency of the output, is set directly by xfreq. This allows the user to use any kind of micro-tuning based scales. However, this method is designed to work correctly only with presets tuned to the default equal temperament. Attempts to use this method with a preset already having non-standard tunings, or with drum-kit-based presets, could give unexpected results. Adjustment of the amplitude can be done by varying the xamp argument, which acts as a multiplier. The ioffset parameter allows the sound to start from a sample different than the first one. The user should make sure that its value is within the length of the specific sound. Otherwise, Csound will probably crash. sfinstr3m is a cubic-interpolation version of sfinstrm. Difference of sound-quality is noticeable specially in bass-frequency-transposed samples. In high-freq-transposed samples the difference is less noticeable, and I suggest to use linear-interpolation versions, because they are faster. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. Examples Here is an example of the sfinstr3m opcode. It uses the file sfinstr3m.csd. Example of the sfinstr3m opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also sfilist, sfinstr3, sfinstr, sfinstrm, sfload, sfpassign, sfplay3, sfplay3m, sfplay, sfplaym, sfplist, sfpreset Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/vexp.xml0000644000000000000000000001267012262561504016113 0ustar rootroot vexp Vectorial:Scalar operations vexp vexp Performs power-of operations between a vector and a scalar Description Performs power-of operations between a vector and a scalar Syntax vexp ifn, kval, kelements [, kdstoffset] [, kverbose] Initialization ifn - number of the table hosting the vectorial signal to be processed Performance kval - scalar operand to be processed kelements - number of elements of the vector kdstoffset - index offset for the destination table (Optional, default = 0) kverbose - Selects whether or not warnings are printed (Default=0) vexp rises kval to each element contained in a vector from table ifn,starting from table index kdstoffset. This enables you to process a specific section of a table by specifying the offset and the number of elements to be processed. Offset is counted starting from 0, so if no offset is specified (or set to 0), the table will be modified from the beginning. Note that this opcode runs at k-rate so the value of kval is processed every control period. Use with care or you will end up with very large (or small) numbers (or use vexp_i). These opcodes (vadd, vmult, vpow and vexp) perform numeric operations between a vectorial control signal (hosted by the table ifn), and a scalar signal (kval). Result is a new vector that overrides old values of ifn. All these opcodes work at k-rate. Negative values for kdstoffset are valid. Elements from the vector that are outside the table, will be discarded, and they will not wrap around the table. If the optional kverbose argument is different to 0, the opcode will print warning messages every k-pass if table lengths are exceeded. In all these opcodes, the resulting vectors are stored in ifn, overriding the intial vectors. If you want to keep initial vector, use vcopy or vcopy_i to copy it in another table. All these operators are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2, etc. They can also be useful in conjunction with the spectral opcodes pvsftw and pvsftr. Please note that the elements argument has changed in version 5.03 from i-rate to k-rate. This will change the opcode's behavior in the unusual cases where the i-rate variable ielements is changed inside the instrument, for example in: instr 1 ielements = 10 vadd 1, 1, ielements ielements = 20 vadd 2, 1, ielements turnoff endin Examples Here is an example of the vexp opcode. It uses the file vexp.csd. Example of the vexp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/instr.xml0000644000000000000000000001211412262561504016261 0ustar rootroot instr Orchestra Syntax:Block Statements instr instr Starts an instrument block. Description Starts an instrument block. Syntax instr i, j, ... Initialization Starts an instrument block defining instruments i, j, ... i, j, ... must be numbers, not expressions. Any positive integer is legal, and in any order, but excessively high numbers are best avoided. Note There may be any number of instrument blocks in an orchestra. Instruments can be defined in any order (but they will always be both initialized and performed in ascending instrument number order, with the exception of notes triggered by real time events that are initialized in the order of being received but still performed in ascending instrument number order). Instrument blocks cannot be nested (i.e. one block cannot contain another). Performance Calling an Instrument within an Instrument This behavior is not fully available in Csound 5. In Csound 5, you must use the subinstr opcode. You can call an instrument within an instrument as if it were an opcode either with the subinstr opcode or by specifying an instrument with a text name: instr MyOscil ... endin By default, all output parameters correspond to the called instrument's output with the signal output opcodes. All input parameters are mapped to the called instrument's p-fields starting with the fourth one, p4. The values of the called instrument's second and third p-fields, p2 and p3, are automatically set to those of the calling instrument's. A named instrument must be defined before any instrument that calls it. Hint If you use the outc opcode, you can create an instrument that will compile and function in any orchestra of any number of channels greater than or equal to the output channels of the instrument. A nice feature to use with named instruments is the #include feature. You can then define your named instruments in separate files, using #include when you need to use one. Examples Here is an example of the instr opcode. It uses the file instr.csd. Example of the instr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also endin, in, out, opcode, endop, setksmps, xin, xout, subinstr, subinstrinit Credits Example written by &namekevin;. manual_src~dfsg/opcodes/sininv.xml0000644000000000000000000000437312262561504016440 0ustar rootroot sininv Mathematical Operations:Trigonometric Functions sininv sininv Performs an arcsine function. Description Returns the arcsine of x (x in radians). Syntax sininv(x) (no rate restriction) Examples Here is an example of the sininv opcode. It uses the file sininv.csd. Example of the sininv opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: i1 = 0.524 See Also cos, cosh, cosinv, sin, sinh, tan, tanh, taninv Credits Author: &namejohn; New in version 3.48 Example written by &namekevin;. manual_src~dfsg/opcodes/adsyn.xml0000644000000000000000000001233012262561504016240 0ustar rootroot adsyn Signal Generators:Additive Synthesis/Resynthesis adsyn adsyn Output is an additive set of individually controlled sinusoids, using an oscillator bank. Description Output is an additive set of individually controlled sinusoids, using an oscillator bank. Syntax ares adsyn kamod, kfmod, ksmod, ifilcod Initialization ifilcod -- integer or character-string denoting a control-file derived from analysis of an audio signal. An integer denotes the suffix of a file adsyn.m or pvoc.m; a character-string (in double quotes) gives a filename, optionally a full pathname. If not fullpath, the file is sought first in the current directory, then in the one given by the environment variable SADIR (if defined). adsyn control contains breakpoint amplitude- and frequency-envelope values organized for oscillator resynthesis, while pvoc control contains similar data organized for fft resynthesis. Memory usage depends on the size of the files involved, which are read and held entirely in memory during computation but are shared by multiple calls (see also lpread). Performance kamod -- amplitude factor of the contributing partials. kfmod -- frequency factor of the contributing partials. It is a control-rate transposition factor: a value of 1 incurs no transposition, 1.5 transposes up a perfect fifth, and .5 down an octave. ksmod -- speed factor of the contributing partials. adsyn synthesizes complex time-varying timbres through the method of additive synthesis. Any number of sinusoids, each individually controlled in frequency and amplitude, can be summed by high-speed arithmetic to produce a high-fidelity result. Component sinusoids are described by a control file describing amplitude and frequency tracks in millisecond breakpoint fashion. Tracks are defined by sequences of 16-bit binary integers: -1, time, amp, time, amp,... -2, time, freq, time, freq,... such as from hetrodyne filter analysis of an audio file. (For details see hetro.) The instantaneous amplitude and frequency values are used by an internal fixed-point oscillator that adds each active partial into an accumulated output signal. While there is a practical limit (limit removed in version 3.47) on the number of contributing partials, there is no restriction on their behavior over time. Any sound that can be described in terms of the behavior of sinusoids can be synthesized by adsyn alone. Sound described by an adsyn control file can also be modified during re-synthesis. The signals kamod, kfmod, ksmod will modify the amplitude, frequency, and speed of contributing partials. These are multiplying factors, with kfmod modifying the frequency and ksmod modifying the speed with which the millisecond breakpoint line-segments are traversed. Thus .7, 1.5, and 2 will give rise to a softer sound, a perfect fifth higher, but only half as long. The values 1,1,1 will leave the sound unmodified. Each of these inputs can be a control signal. Examples Here is an example of the adsyn opcode. It uses the file adsyn.csd, and kickroll.het. The file kickroll.het was created by using the hetro utility with the audio file kickroll.wav. Example of the adsyn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. manual_src~dfsg/opcodes/midion.xml0000644000000000000000000001027612262561504016410 0ustar rootroot midion Real-time MIDI:Note Output midion midion Generates MIDI note messages at k-rate. Description Generates MIDI note messages at k-rate. Syntax midion kchn, knum, kvel Performance kchn -- MIDI channel number (1-16) knum -- note number (0-127) kvel -- velocity (0-127) midion (k-rate note on) plays MIDI notes with current kchn, knum and kvel. These arguments can be varied at k-rate. Each time the MIDI converted value of any of these arguments changes, last MIDI note played by current instance of midion is immediately turned off and a new note with the new argument values is activated. This opcode, as well as moscil, can generate very complex melodic textures if controlled by complex k-rate signals. Any number of midion opcodes can appear in the same Csound instrument, allowing a counterpoint-style polyphony within a single instrument. Examples Here is a simple example of the midion opcode. It uses the file midion_simple.csd. Simple Example of the midion opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This example generates a minor chord over every note received on the MIDI input. It generates MIDI notes on csound's MIDI output, so be sure to connect something. Here is another example of the midion opcode. It uses the file midion_scale.csd. Example of the midion opcode to generate random notes from a scale. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This example generates random notes from a given scale for every note received on the MIDI input. It generates MIDI notes on csound's MIDI output, so be sure to connect something. See Also moscil, midion2, noteon, noteoff, noteondur, noteondur2 Credits Author: &namegabriel; Italy May 1997 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/mac.xml0000644000000000000000000000450612262561504015670 0ustar rootroot mac Mathematical Operations:Opcode Equivalents of Functions mac mac Multiplies and accumulates a- and k-rate signals. Description Multiplies and accumulates a- and k-rate signals. Syntax ares mac ksig1, asig1 [, ksig2] [, asig2] [, ksig3] [, asig3] [...] Performance ksig1, etc. -- k-rate input signals asig1, etc. -- a-rate input signals mac multiplies and accumulates a- and k-rate signals. It is equivalent to: ares = asig1*ksig1 + asig2*ksig2 + asig3*ksig3 + ... Examples Here is an example of the mac opcode. It uses the file mac.csd. It is written for *NIX systems, and will generate errors on Windows. Example of the mac opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also maca Credits Author: &namejohn; University of Bath, Codemist, Ltd. Bath, UK May 1999 New in Csound version 3.54 manual_src~dfsg/opcodes/pan.xml0000644000000000000000000001025212262561504015701 0ustar rootroot pan Signal Modifiers:Panning and Spatialization pan pan Distribute an audio signal amongst four channels. Description Distribute an audio signal amongst four channels with localization control. Syntax a1, a2, a3, a4 pan asig, kx, ky, ifn [, imode] [, ioffset] Initialization ifn -- function table number of a stored pattern describing the amplitude growth in a speaker channel as sound moves towards it from an adjacent speaker. Requires extended guard-point. imode (optional) -- mode of the kx, ky position values. 0 signifies raw index mode, 1 means the inputs are normalized (0 - 1). The default value is 0. ioffset (optional) -- offset indicator for kx, ky. 0 infers the origin to be at channel 3 (left rear); 1 requests an axis shift to the quadraphonic center. The default value is 0. Performance pan takes an input signal asig and distributes it amongst four outputs (essentially quad speakers) according to the controls kx and ky. For normalized input (mode=1) and no offset, the four output locations are in order: left-front at (0,1), right-front at (1,1), left-rear at the origin (0,0), and right-rear at (1,0). In the notation (kx, ky), the coordinates kx and ky, each ranging 0 - 1, thus control the 'rightness' and 'forwardness' of a sound location. Movement between speakers is by amplitude variation, controlled by the stored function table ifn. As kx goes from 0 to 1, the strength of the right-hand signals will grow from the left-most table value to the right-most, while that of the left-hand signals will progress from the right-most table value to the left-most. For a simple linear pan, the table might contain the linear function 0 - 1. A more correct pan that maintains constant power would be obtained by storing the first quadrant of a sinusoid. Since pan will scale and truncate kx and ky in simple table lookup, a medium-large table (say 8193) should be used. kx, ky values are not restricted to 0 - 1. A circular motion passing through all four speakers (inscribed) would have a diameter of root 2, and might be defined by a circle of radius R = root 1/2 with center at (.5,.5). kx, ky would then come from Rcos(angle), Rsin(angle), with an implicit origin at (.5,.5) (i.e. ioffset = 1). Unscaled raw values operate similarly. Sounds can thus be located anywhere in the polar or Cartesian plane; points lying outside the speaker square are projected correctly onto the square's perimeter as for a listener at the center. Example Here is an example of the pan opcodes. It uses the file pan.csd. Example of the pan opcodes. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. manual_src~dfsg/opcodes/koutc14.xml0000644000000000000000000000077112262561504016422 0ustar rootroot koutc14 koutc14 koutc14 Deprecated. Description Deprecated as of version 3.52. Use the outkc14 opcode instead. manual_src~dfsg/opcodes/scantable.xml0000644000000000000000000000602312262561504017060 0ustar rootroot scantable Signal Generators:Scanned Synthesis scantable scantable A simpler scanned synthesis implementation. Description A simpler scanned synthesis implementation. This is an implementation of a circular string scanned using external tables. This opcode will allow direct modification and reading of values with the table opcodes. Syntax aout scantable kamp, kpch, ipos, imass, istiff, idamp, ivel Initialization ipos -- table containing position array. imass -- table containing the mass of the string. istiff -- table containing the stiffness of the string. idamp -- table containing the damping factors of the string. ivel -- table containing the velocities. Performance kamp -- amplitude (gain) of the string. kpch -- the string's scanned frequency. Examples Here is an example of the scantable opcode. It uses the file scantable.csd. Example of the scantable opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also scanhammer More information on this opcode: http://www.csounds.com/stevenyi/scanned/yi_scannedSynthesis.html , written by Steven Yi Credits Author: Matt Gilliard April 2002 New in version 4.20 manual_src~dfsg/opcodes/ftmorf.xml0000644000000000000000000000642012262561504016422 0ustar rootroot ftmorf Signal Modifiers:Convolution and Morphing ftmorf ftmorf Morphs between multiple ftables as specified in a list. Description Uses an index into a table of ftable numbers to morph between adjacent tables in the list.This morphed function is written into the table referenced by iresfn on every k-cycle. Syntax ftmorf kftndx, iftfn, iresfn Initialization iftfn -- The table containing the numbers of any existing tables which are used for the morphing. iresfn -- Table number of the morphed function The length of all the tables in iftfn must equal the length of iresfn. Performance kftndx -- the index into the iftfn table. If iftfn contains (6, 4, 6, 8, 7, 4): kftndx=4 will write the contents of f7 into iresfn. kftndx=4.5 will write the average of the contents of f7 and f4 into iresfn. iresfn is only updated if the morfing index changes it's value, if kftindx is static, no writing to iresfn will occur. Examples Here is an example of the ftmorf opcode. It uses the file ftmorf.csd. Example of the ftmorf opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: William Pete Moss University of Texas at Austin Austin, Texas USA Jan. 2002 New in version 4.18 manual_src~dfsg/opcodes/lphasor.xml0000644000000000000000000000730312262561504016576 0ustar rootroot lphasor Signal Generators:Sample Playback lphasor lphasor Generates a table index for sample playback Description This opcode can be used to generate table index for sample playback (e.g. tablexkt). Syntax ares lphasor xtrns [, ilps] [, ilpe] [, imode] [, istrt] [, istor] Initialization ilps -- loop start. ilpe -- loop end (must be greater than ilps to enable looping). The default value of ilps and ilpe is zero. imode (optional: default = 0) -- loop mode. Allowed values are: 0: no loop 1: forward loop 2: backward loop 3: forward-backward loop istrt (optional: default = 0) -- The initial output value (phase). It must be less than ilpe if looping is enabled, but is allowed to be greater than ilps (i.e. you can start playback in the middle of the loop). istor (optional: default = 0) -- skip initialization if set to any non-zero value. Performance ares -- a raw table index in samples (same unit for loop points). Can be used as index with the table opcodes. xtrns -- transpose factor, expressed as a playback ratio. ares is incremented by this value, and wraps around loop points. For example, 1.5 means a fifth above, 0.75 means fourth below. It is not allowed to be negative. Examples Here is an example of the lphasor opcode. It uses the file lphasor.csd. Example of the lphasor opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &nameistvan; January 2002 Example by: Jonathan Murphy New in version 4.18 Updated April 2002 and November 2002 by &nameistvan; manual_src~dfsg/opcodes/bformdec1.xml0000644000000000000000000001020512262561504016763 0ustar rootroot bformdec1 Signal Modifiers:Panning and Spatialization bformdec1 bformdec1 Decodes an ambisonic B format signal Description Decodes an ambisonic B format signal into loudspeaker specific signals. Syntax ao1, ao2 bformdec1 isetup, aw, ax, ay, az [, ar, as, at, au, av \ [, abk, al, am, an, ao, ap, aq]] ao1, ao2, ao3, ao4 bformdec1 isetup, aw, ax, ay, az [, ar, as, at, \ au, av [, abk, al, am, an, ao, ap, aq]] ao1, ao2, ao3, ao4, ao5 bformdec1 isetup, aw, ax, ay, az [, ar, as, \ at, au, av [, abk, al, am, an, ao, ap, aq]] ao1, ao2, ao3, ao4, ao5, ao6, ao7, ao8 bformdec1 isetup, aw, ax, ay, az \ [, ar, as, at, au, av [, abk, al, am, an, ao, ap, aq]]] aout[] bformdec1 isetup, abform[] Initialization Note that horizontal angles are measured anticlockwise in this description. isetup –- loudspeaker setup. There are five supported setups: 1. Stereo - L(90), R(-90); this is an M+S style stereo decode. 2. Quad - FL(45), BL(135), BR(-135), FR(-45). This is a first-order `in-phase' decode. 3. 5.0 - L(30),R(-30),C(0),BL(110),BR(-110). Note that many people do not actually use the angles above for their speaker arrays and a good decode for DVD etc can be achieved using the Quad configuration to feed L, R, BL and BR (leaving C silent). 4. Octagon - FFL(22.5),FLL(67.5),BLL(112.5),BBL(157.5),BBR(-157.5),BRR(-112.5),FRR(-67.5),FFR(-22.5). This is a first-, second- or third-order `in-phase' decode, depending on the number of input channels. 5. Cube - FLD(45,-35.26),FLU(45,35.26),BLD(135,-35.26),BLU(135,35.26),BRD(-135,-35.26),BRU(-135,35.26),FRD(-45,-35.26),FRU(-45,35.26). This is a first-order `in-phase' decode. Performance aw, ax, ay, ... -- input signal in the B format. ao1 .. ao8 -– loudspeaker specific output signals. Example Here is an example of the bformdec1 opcode. It uses the file bformenc1.csd. Example of the bformdec1 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also bformenc1 Credits Author: Richard Furse, Bruce Wiggins and Fons Adriaensen, following code by Samuel Groner 2008 New in version 5.09 manual_src~dfsg/opcodes/minabsaccum.xml0000644000000000000000000000754512262561504017420 0ustar rootroot minabsaccum Signal Modifiers:Comparators and Accumulators minabsaccum minabsaccum Accumulates the minimum of the absolute values of audio signals. Description minabsaccum compares two audio-rate variables and stores the minimum of their absolute values into the first. Syntax minabsaccum aAccumulator, aInput Performance aAccumulator -- audio variable to store the minimum value aInput -- signal that aAccumulator is compared to The minabsaccum opcode is designed to accumulate the minimum value from among many audio signals that may be in different note instances, different channels, or otherwise cannot all be compared at once using the minabs opcode. minabsaccum is identical to minaccum except that it takes the absolute value of aInput before the comparison. Its semantics are similar to vincr since aAccumulator is used as both an input and an output variable, except that minabsaccum keeps the minimum absolute value instead of adding the signals together. minabsaccum performs the following operation on each pair of samples: if (abs(aInput) < aAccumulator) aAccumulator = abs(aInput) aAccumulator will usually be a global audio variable. At the end of any given computation cycle (k-period), after its value is read and used in some way, the accumulator variable should usually be reset to some large enough positive value that will always be greater than the input signals to which it is compared. Examples Here is an example of the minabsaccum opcode. It uses the file minabsaccum.csd. Example of the minabsaccum opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also maxabsaccum, maxaccum, minaccum, max, min, maxabs, minabs, vincr Credits Author: &nameanthony; March 2006 New in Csound version 5.01 manual_src~dfsg/opcodes/readf.xml0000644000000000000000000000513112262561504016204 0ustar rootroot readf Signal I/O:File I/O readf readf Read a line of text from an external file. Description Read a line of text from an external file once each k-cycle. Syntax Sres, kline readf ifilname Initialization ifilname -- an integer N denoting a file named "input.N" or a character string (in double quotes, spaces permitted) denoting the external file name. For a string, it may either be a full path name with directory specified or a simple filename. In the later case, the file is sought first in the current directory, then in SSDIR, and finally in SFDIR. Performance Sres -- output of the line read from ifilname. kline -- line number, or -1 when end of file has been reached. This opcode allows a line of text to be read from a named external file. There may be any number of readf opcodes in an instrument or orchestra but they read separately from the same or different files. Examples Here is an example of the readf opcode. It uses the file readf.csd. Example of the readf opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also readfi. Credits John ffitch and Joachim Heintz 2012; new in 5.17.12 manual_src~dfsg/opcodes/modulus.xml0000644000000000000000000001014212262561504016611 0ustar rootroot % Mathematical Operations:Arithmetic and Logic Operations % % Modulus operator. Description Arithmetic operators perform operations of change-sign (negate), don't-change-sign, logical AND logical OR, add, subtract, multiply and divide. Note that a value or an expression may fall between two of these operators, either of which could take it as its left or right argument, as in a + b * c. In such cases three rules apply: 1. * and / bind to their neighbors more strongly than + and −. Thus the above expression is taken as a + (b * c) with * taking b and c and then + taking a and b * c. 2. + and − bind more strongly than &&, which in turn is stronger than ||: a && b - c || d is taken as (a && (b - c)) || d 3. When both operators bind equally strongly, the operations are done left to right: a - b - c is taken as (a - b) - c Parentheses may be used as above to force particular groupings. The operator % returns the value of a reduced by b, so that the result, in absolute value, is less than the absolute value of b, by repeated subtraction. This is the same as modulus function in integers. New in Csound version 3.50. Syntax a % b (no rate restriction) where the arguments a and b may be further expressions. Arguments The arguments of % can be scalar values or k-rate one dimensional arrays (vectors), or any combination. If one of the arguments is an array, so is the value. Examples Here is an example of the % operator. It uses the file modulus.csd. Example of the % operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also -, +, &&, ||, *, /, ˆ Credits Arithmetic on vectors is new in version 6.00 manual_src~dfsg/opcodes/limit.xml0000644000000000000000000000564412262561504016252 0ustar rootroot limit Signal Modifiers:Signal Limiters limit limit Sets the lower and upper limits of the value it processes. Description Sets the lower and upper limits of the value it processes. Syntax ares limit asig, klow, khigh ires limit isig, ilow, ihigh kres limit ksig, klow, khigh Initialization isig -- input signal ilow -- low threshold ihigh -- high threshold Performance xsig -- input signal klow -- low threshold khigh -- high threshold limit sets the lower and upper limits on the xsig value it processes. If xhigh is lower than xlow, then the output will be the average of the two - it will not be affected by xsig. This opcode is useful in several situations, such as table indexing or for clipping and modeling a-rate, i-rate or k-rate signals. Examples Here is an example of the limit opcode. It uses the file limit.csd. Example of the limit opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also mirror, wrap Credits Author: Robin Whittle Australia New in Csound version 3.46 manual_src~dfsg/opcodes/tablewkt.xml0000644000000000000000000001613312262561504016744 0ustar rootroot tablewkt Table Control:Read/Write Opreations tablewkt tablewkt Change the contents of existing function tables. Description This opcode operates on existing function tables, changing their contents. tablewkt uses a k-rate variable for selecting the table number. The valid combinations of variable types are shown by the first letter of the variable names. Syntax tablewkt asig, andx, kfn [, ixmode] [, ixoff] [, iwgmode] tablewkt ksig, kndx, kfn [, ixmode] [, ixoff] [, iwgmode] Initialization asig, ksig -- The value to be written into the table. andx, kndx -- Index into table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0) kfn -- Table number. Must be >= 1. Floats are rounded down to an integer. If a table number does not point to a valid table, or the table has not yet been loaded (GEN01) then an error will result and the instrument will be de-activated. ixmode -- index mode. Default is zero. 0 = xndx and ixoff ranges match the length of the table. Not equal to 0 = xndx and ixoff have a 0 to 1 range. ixoff -- index offset. Default is 0. 0 = Total index is controlled directly by xndx, i.e. the indexing starts from the start of the table. Not equal to 0 = Start indexing from somewhere else in the table. Value must be positive and less than the table length (ixmode = 0) or less than 1 (ixmode != 0). iwgmode -- table writing mode. Default is 0. 0 = Limit mode. 1 = Wrap mode. 2 = Guardpoint mode. Performance Limit mode (0) Limit the total index (ndx + ixoff) to between 0 and the guard point. For a table of length 5, this means that locations 0 to 3 and location 4 (the guard point) can be written. A negative total index writes to location 0. Wrap mode (1) Wrap total index value into locations 0 to E, where E is one less than either the table length or the factor of 2 number which is one less than the table length. For example, wrap into a 0 to 3 range - so that total index 6 writes to location 2. Guardpoint mode (2) The guardpoint is written at the same time as location 0 is written - with the same value. This facilitates writing to tables which are intended to be read with interpolation for producing smooth cyclic waveforms. In addition, before it is used, the total index is incremented by half the range between one location and the next, before being rounded down to the integer address of a table location. Normally (igwmode = 0 or 1) for a table of length 5 - which has locations 0 to 3 as the main table and location 4 as the guard point, a total index in the range of 0 to 0.999 will write to location 0. ("0.999" means just less than 1.0.) 1.0 to 1.999 will write to location 1 etc. A similar pattern holds for all total indexes 0 to 4.999 (igwmode = 0) or to 3.999 (igwmode = 1). igwmode = 0 enables locations 0 to 4 to be written - with the guardpoint (4) being written with a potentially different value from location 0. With a table of length 5 and the iwgmode = 2, then when the total index is in the range 0 to 0.499, it will write to locations 0 and 4. Range 0.5 to 1.499 will write to location 1 etc. 3.5 to 4.0 will also write to locations 0 and 4. This way, the writing operation most closely approximates the results of interpolated reading. Guard point mode should only be used with tables that have a guardpoint. Guardpoint mode is accomplished by adding 0.5 to the total index, rounding to the next lowest integer, wrapping it modulo the factor of two which is one less than the table length, writing the table (locations 0 to 3 in our example) and then writing to the guard point if index = 0. Caution with k-rate table numbers At k-rate or a-rate, if a table number of < 1 is given, or the table number points to a non-existent table, or to one which has a length of 0 (it is to be loaded from a file later) then an error will result and the instrument will be deactivated. kfn and afn must be initialized at the appropriate rate using init. Attempting to load an i-rate value into kfn or afn will result in an error. See Also tableiw, tablew Credits Author: Robin Whittle Australia May 1997 manual_src~dfsg/opcodes/multitap.xml0000644000000000000000000000327112262561504016765 0ustar rootroot multitap Signal Modifiers:Delay multitap multitap Multitap delay line implementation. Description Multitap delay line implementation. Syntax ares multitap asig [, itime1, igain1] [, itime2, igain2] [...] Initialization The arguments itime and igain set the position and gain of each tap. The delay line is fed by asig. Examples Here is an example of the multitap opcode. It uses the file multitap.csd Example of the multitap opcode. This results in two delays, one with length of 1.2 and gain of .5, and one with length of 1.4 and gain of .2. Credits Author: &nameparis; MIT, Cambridge 1996 manual_src~dfsg/opcodes/JackoAudioOutConnect.xml0000644000000000000000000000512612262561504021142 0ustar rootroot JackoAudioOutConnect Jacko Opcodes JackoAudioOutConnect JackoAudioOutConnect Creates an audio connection from Csound to a Jack port. Description In the orchestra header, creates an audio connection from a Jack audio output port inside this instance of Csound to an external Jack audio input port. Syntax JackoAudioOutConnect ScsoundPortName, SexternalPortName Initialization ScsoundPortName -- The short name ("portname") of the internal Jack audio output port. SexternalPortName -- The full name ("clientname:portname") of an external Jack audio input port. Performance The actual audio must be written with the JackoAudioOut opcode. See Also The Jacko Opcodes, JackoInfo, JackoInfo, JackoFreewheel, JackoAudioOutConnect, JackoMidiInConnect, JackoMidiOutConnect, JackoOn, JackoAudioIn, JackoAudioOut, JackoMidiOut, JackoNoteOut, JackoTransport. Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/cosh.xml0000644000000000000000000000437312262561504016066 0ustar rootroot cosh Mathematical Operations:Trigonometric Functions cosh cosh Performs a hyperbolic cosine function. Description Returns the hyperbolic cosine of x (x in radians). Syntax cosh(x) (no rate restriction) Examples Here is an example of the cosh opcode. It uses the file cosh.csd. Example of the cosh opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: i1 = 1.543 See Also cos, cosinv, sin, sinh, sininv, tan, tanh, taninv Credits Author: &namejohn; New in version 3.47 Example written by &namekevin;. manual_src~dfsg/opcodes/freeverb.xml0000644000000000000000000000753212262561504016732 0ustar rootroot freeverb Signal Modifiers:Reverberation freeverb freeverb Opcode version of Jezar's Freeverb Description freeverb is a stereo reverb unit based on Jezar's public domain C++ sources, composed of eight parallel comb filters on both channels, followed by four allpass units in series. The filters on the right channel are slightly detuned compared to the left channel in order to create a stereo effect. Syntax aoutL, aoutR freeverb ainL, ainR, kRoomSize, kHFDamp[, iSRate[, iSkip]] Initialization iSRate (optional, defaults to 44100): adjusts the reverb parameters for use with the specified sample rate (this will affect the length of the delay lines in samples, and, as of the latest CVS version, the high frequency attenuation). Only integer multiples of 44100 will reproduce the original character of the reverb exactly, so it may be useful to set this to 44100 or 88200 for an orchestra sample rate of 48000 or 96000 Hz, respectively. While iSRate is normally expected to be close to the orchestra sample rate, different settings may be useful for special effects. iSkip (optional, defaults to zero): if non-zero, initialization of the opcode will be skipped, whenever possible. Performance ainL, ainR -- input signals; usually both are the same, but different inputs can be used for special effect Note It is recommended to process the input signal(s) with the denorm opcode in order to avoid denormalized numbers which could significantly increase CPU usage in some cases aoutL, aoutR -- output signals for left and right channel kRoomSize (range: 0 to 1) -- controls the length of the reverb, a higher value means longer reverb. Settings above 1 may make the opcode unstable. kHFDamp (range: 0 to 1): high frequency attenuation; a value of zero means all frequencies decay at the same rate, while higher settings will result in a faster decay of the high frequency range. Examples Here is an example of the freeverb opcode. It uses the file freeverb.csd. An example of the freeverb opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/vdelayxw.xml0000644000000000000000000000766212262561504017001 0ustar rootroot vdelayxw Signal Modifiers:Delay vdelayxw vdelayxw Variable delay opcodes with high quality interpolation. Description Variable delay opcodes with high quality interpolation. Syntax aout vdelayxw ain, adl, imd, iws [, ist] Initialization imd -- max. delay time (seconds) iws -- interpolation window size (see below) ist (optional) -- skip initialization if not zero Performance aout -- output audio signal ain -- input audio signal adl -- delay time in seconds These opcodes use high quality (and slow) interpolation, that is much more accurate than the currently available linear and cubic interpolation. The iws parameter sets the number of input samples used for calculating one output sample (allowed values are any integer multiply of 4 in the range 4 - 1024); higher values mean better quality and slower speed. The vdelayxw opcodes change the position of the write tap in the delay line (unlike all other delay ugens that move the read tap), and are most useful for implementing Doppler effects where the position of the listener is fixed, and the sound source is moving. Notes Delay time is measured in seconds (unlike in vdelay and vdelay3), and must be a-rate. The minimum allowed delay is iws/2 samples. Using the same variables as input and output is allowed in these opcodes. In vdelayxw*, changing the delay time has some effects on output volume: a = 1 / (1 + dt) where a is the output gain, and dt is the change of delay time per seconds. These opcodes are best used in the double-precision version of Csound. Examples Here is an example of the use of the vdelayxw opcode. It uses the file vdelayxw.csd. Example of the <emphasis>vdelayxw</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vdelayx, vdelayxq, vdelayxs, vdelayxwq, vdelayxws manual_src~dfsg/opcodes/eqfil.xml0000644000000000000000000000611112262561504016222 0ustar rootroot eqfil Signal Modifiers:Specialized Filters eqfil eqfil Equalizer filter Description The opcode eqfil is a 2nd order tunable equalisation filter based on Regalia and Mitra design ("Tunable Digital Frequency Response Equalization Filters", IEEE Trans. on Ac., Sp. and Sig Proc., 35 (1), 1987). It provides a peak/notch filter for building parametric/graphic equalisers. The amplitude response for this filter will be flat (=1) for kgain=1. With kgain bigger than 1, there will be a peak at the centre frequency, whose width is given by the kbw parameter, but outside this band, the response will tend towards 1. Conversely, if kgain is smaller than 1, a notch will be created around the CF. Syntax asig eqfil ain, kcf, kbw, kgain[, istor] Initialization istor --initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance asig -- filtered output signal. ain -- input signal. kcf -- filter centre frequency. kbw -- peak/notch bandwidth (Hz). kgain -- peak/notch gain. Examples Here is an example of the eqfil opcode. It uses the file eqfil.csd. Example of the eqfil opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; April 2007 New in version 5.06 manual_src~dfsg/opcodes/flslidbnk2set.xml0000644000000000000000000001124212262561504017671 0ustar rootroot FLslidBnk2Set FLTK:Other FLslidBnk2Set FLslidBnk2Set modify the values of a slider bank. Description FLslidBnk2Set modifies the values of a slider bank according to an array of values stored in a table. Syntax FLslidBnk2Set ihandle, ifn [, istartIndex, istartSlid, inumSlid] Initialization ihandle - handle of the sliderBnk (to be used to set its values). ifn - number of a table containing an array of values to set each slider to. istartIndex - (optional) starting index of the table element of to be evaluated firstly. Default value is zero istartSlid - (optional) starting slider to be evaluated. Default 0, denoting the first slider. inumSlid - (optional) number of sliders to be updated. Default 0, denoting all sliders. Performance FLslidBnk2Set modifies the values of a slider bank (created with FLslidBnk2 or with FLvslidBnk2) according to an array of values stored into table ifn. It actually allows to update an FLslidBnk2 (or FLvslidBnk2) bank of sliders (for instance, using the slider8table opcode) to a set of values located in a table. User has to set ihandle argument to the handle got from FLslidBnkGetHandle opcode. It works at init-rate only. It is possible to reset only a range of sliders, by using the optional arguments istartIndex, istartSlid, inumSlid FLslidBnk2Set is identical to FLslidBnkSet, but works on FLslidBnk2 and FLvslidBnk2 instead of FLslidBnk and FLvslidBnk. There is a k-rate version of this opcode called FLslidBnk2Setk. See Also FLslider, FLslidBnkGetHandle, FLslidBnk, FLslidBnk2, FLvslidBnk, FLvslidBnk2 FLslidBnkSet, FLslidBnk2Setk Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/vstinit.xml0000644000000000000000000000446612262561504016635 0ustar rootroot vstinit Plugin Hosting:VST vstinit vstinit Load a VST plugin into memory for use with the other vst4cs opcodes. Syntax instance vstinit ilibrarypath [,iverbose] Description vstinit is used to load a VST plugin into memory for use with the other vst4cs opcodes. Both VST effects and instruments (synthesizers) can be used. Initialization instance - the number which identifies the plugin, to be passed to other vst4cs opcodes. ilibrarypath - the full path to the vst plugin shared library (DLL, on Windows). Remember to use '/' instead of '\' as separator. iverbose - show plugin information and parameters when loading. Examples Here is an example of the use of the vstinit opcode. It uses the file vst4cs.csd. Example of the <emphasis>vstinit</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits By: &nameandres; and &namemichael; Uses code from Hermann Seib's VSTHost and Thomas Grill's vst~ object. VST is a trademark of Steinberg Media Technologies GmbH. VST Plug-In Technology by Steinberg. manual_src~dfsg/opcodes/imagesave.xml0000644000000000000000000000471612262561504017074 0ustar rootroot imagesave Image Processing Opcodes imagesave imagesave Save a previously created image. Description Save a previously created image. An empty image can be created with imagecreate and its pixel RGB values can be set with imagesetpixel. The image will be saved in PNG format. Syntax imagesave iimagenum, filename Initialization iimagenum -- the reference of the image to be save. It should be a value returned by imagecreate. filename -- The filename to use to save the image. Examples Here is an example of the imagesave opcode. It uses the file imageopcodes.csd. Example of the imagesave opcode. See Also imageload, imagecreate, imagesize, imagegetpixel, imagesetpixel, imagefree Credits Author: Cesare Marilungo New in version 5.08 manual_src~dfsg/opcodes/wgflute.xml0000644000000000000000000000747212262561504016612 0ustar rootroot wgflute Signal Generators:Waveguide Physical Modeling wgflute wgflute Creates a tone similar to a flute. Description Audio output is a tone similar to a flute, using a physical model developed from &nameperry;, but re-coded for Csound. Syntax ares wgflute kamp, kfreq, kjet, iatt, idetk, kngain, kvibf, kvamp, ifn \ [, iminfreq] [, ijetrf] [, iendrf] Initialization iatt -- time in seconds to reach full blowing pressure. 0.1 seems to correspond to reasonable playing. idetk -- time in seconds taken to stop blowing. 0.1 is a smooth ending ifn -- table of shape of vibrato, usually a sine table, created by a function iminfreq (optional) -- lowest frequency at which the instrument will play. If it is omitted it is taken to be the same as the initial kfreq. If iminfreq is negative, initialization will be skipped. ijetrf (optional, default=0.5) -- amount of reflection in the breath jet that powers the flute. Default value is 0.5. iendrf (optional, default=0.5) -- reflection coefficient of the breath jet. Default value is 0.5. Both ijetrf and iendrf are used in the calculation of the pressure differential. Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. While it can be varied in performance, I have not tried it. kjet -- a parameter controlling the air jet. Values should be positive, and about 0.3. The useful range is approximately 0.08 to 0.56. kngain -- amplitude of the noise component, about 0 to 0.5 kvibf -- frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp -- amplitude of the vibrato Examples Here is an example of the wgflute opcode. It uses the file wgflute.csd. Example of the wgflute opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/endin.xml0000644000000000000000000000410412262561504016217 0ustar rootroot endin Orchestra Syntax:Block Statements endin endin Ends the current instrument block. Description Ends the current instrument block. Syntax endin Initialization Ends the current instrument block. Instruments can be defined in any order (but they will always be both initialized and performed in ascending instrument number order). Instrument blocks cannot be nested (i.e. one block cannot contain another). Note There may be any number of instrument blocks in an orchestra. Examples Here is an example of the endin opcode. It uses the file endin.csd. Example of the endin opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also instr Credits Example written by &namekevin;. manual_src~dfsg/opcodes/centroid.xml0000644000000000000000000000434112262561504016734 0ustar rootroot centroid Spectral Processing:Other centroid centroid Calculate the spectral centroid of a signal. Description Calculate the spectral centroid of an audio signal on a given trigger. Syntax kcent centroid asig, ktrig, ifftsize Initialization ifftsize -- fftsize. Non pow-of-two values are converted to the next pow-of-two not smaller than ifftsize. Performance kcent -- the spectral centroid in Hz asig -- an input audio signal ktrig -- 1 to calculate a new centroid, 0 to skip the process (and output previous value). Examples Here is an example of the centroid opcode. It uses the file centroid.csd. Example of the centroid opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvscent Credits Author: V Lazzarini; August 2012 manual_src~dfsg/opcodes/outkpat.xml0000644000000000000000000000637112262561504016621 0ustar rootroot outkpat Real-time MIDI:Output outkpat outkpat Sends polyphonic MIDI aftertouch messages at k-rate. Description Sends polyphonic MIDI aftertouch messages at k-rate. Syntax outkpat kchn, knotenum, kvalue, kmin, kmax Performance kchn -- MIDI channel number (1-16) knotenum -- MIDI note number (used in polyphonic aftertouch messages) kvalue -- floating point value kmin -- minimum floating point value (converted in MIDI integer value 0) kmax -- maximum floating point value (converted in MIDI integer value 127 (7 bit)) outkpat (k-rate polyphonic aftertouch output) sends polyphonic aftertouch messages. It works only with MIDI instruments which recognize them. It can drive a different value of a parameter for each note currently active. It can scale the k-value floating-point argument according to the kmin and kmax values. For example: set kmin = 1.0 and kmax = 2.0. When the kvalue argument receives a 2.0 value, the opcode will send a 127 value to the MIDI OUT device. When the kvalue argument receives a 1.0 value, it will send a 0 value. k-rate opcodes send a message each time the MIDI converted value of argument kvalue changes. See Also outiat, outic14, outic, outipat, outipb, outipc, outkat, outkc14, outkc, outkpb, outkpc Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/cpuprc.xml0000644000000000000000000001065612262561504016427 0ustar rootroot cpuprc Instrument Control:Realtime Performance Control cpuprc cpuprc Control allocation of cpu resources on a per-instrument basis, to optimize realtime output. Description Control allocation of cpu resources on a per-instrument basis, to optimize realtime output. Syntax cpuprc insnum, ipercent cpuprc Sinsname, ipercent Initialization insnum -- instrument number or string Sinsname -- instrument number or string ipercent -- percent of cpu processing-time to assign. Can also be expressed as a fractional value. Performance cpuprc sets the cpu processing-time percent usage of an instrument, in order to avoid buffer underrun in realtime performances, enabling a sort of polyphony theshold. The user must set ipercent value for each instrument to be activated in realtime. Assuming that the total theoretical processing time of the cpu of the computer is 100%, this percent value can only be defined empirically, because there are too many factors that contribute to limiting realtime polyphony in different computers. For example, if ipercent is set to 5% for instrument 1, the maximum number of voices that can be allocated in realtime, is 20 (5% * 20 = 100%). If the user attempts to play a further note while the 20 previous notes are still playing, Csound inhibits the allocation of that note and will display the following warning message: can't allocate last note because it exceeds 100% of cpu time In order to avoid audio buffer underruns, it is suggested to set the maximum number of voices slightly lower than the real processing power of the computer. Sometimes an instrument can require more processing time than normal. If, for example, the instrument contains an oscillator which reads a table that doesn't fit in cache memory, it will be slower than normal. In addition, any program running concurrently in multitasking, can subtract processing power to varying degrees. At the start, all instruments are set to a default value of ipercent = 0.0% (i.e. zero processing time or rather infinite cpu processing-speed). This setting is OK for deferred-time sessions. All instances of cpuprc must be defined in the header section, not in the instrument body. Examples Here is an example of the cpuprc opcode. It uses the file cpuprc.csd. Example of the cpuprc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also maxalloc, prealloc Credits Author: &namegabriel; Italy July, 1999 New in Csound version 3.57; named instruments added version 5.13 manual_src~dfsg/opcodes/lorenz.xml0000644000000000000000000001017612262561504016441 0ustar rootroot lorenz Signal Generators:Models and Emulations lorenz lorenz Implements the Lorenz system of equations. Description Implements the Lorenz system of equations. The Lorenz system is a chaotic-dynamic system which was originally used to simulate the motion of a particle in convection currents and simplified weather systems. Small differences in initial conditions rapidly lead to diverging values. This is sometimes expressed as the butterfly effect. If a butterfly flaps its wings in Australia, it will have an effect on the weather in Alaska. This system is one of the milestones in the development of chaos theory. It is useful as a chaotic audio source or as a low frequency modulation source. Syntax ax, ay, az lorenz ksv, krv, kbv, kh, ix, iy, iz, iskip [, iskipinit] Initialization ix, iy, iz -- the initial coordinates of the particle. iskip -- used to skip generated values. If iskip is set to 5, only every fifth value generated is output. This is useful in generating higher pitched tones. iskipinit (optional, default=0) -- if non zero skip the initialisation of the filter. (New in Csound version 4.23f13 and 5.0) Performance ksv -- the Prandtl number or sigma krv -- the Rayleigh number kbv -- the ratio of the length and width of the box in which the convection currents are generated kh -- the step size used in approximating the differential equation. This can be used to control the pitch of the systems. Values of .1-.001 are typical. The equations are approximated as follows: x = x + h*(s*(y - x)) y = y + h*(-x*z + r*x - y) z = z + h*(x*y - b*z) The historical values of these parameters are: ks = 10 kr = 28 kb = 8/3 This algorithm uses internal non linear feedback loops which causes audio result to depend on the orchestra sampling rate. For example, if you develop a project with sr=48000Hz and if you want to produce an audio CD from it, you should record a file with sr=48000Hz and then downsample the file to 44100Hz using the srconv utility. Examples Here is an example of the lorenz opcode. It uses the file lorenz.csd. Example of the lorenz opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namehans; February 1999 New in Csound version 3.53 Note added by &namepinot;, August 2009 manual_src~dfsg/opcodes/chnget.xml0000644000000000000000000000616212262561504016400 0ustar rootroot chnget Signal I/O:Software Bus chnget chnget Reads data from the software bus. Description Reads data from a channel of the inward named software bus. Implies declaring the channel with imode=1 (see also chn_k, chn_a, and chn_S). Syntax ival chnget Sname kval chnget Sname aval chnget Sname Sval chnget Sname Initialization Sname -- a string that identifies a channel of the named software bus to read. ival -- the control value read at i-time. Sval -- the string value read at i-time. Performance kval -- the control value read at performance time. aval -- the audio signal read at performance time. Example The example shows the software bus being used as an asynchronous control signal to select a filter cutoff. It assumes that an external program that has access to the API is feeding the values. sr = 44100 kr = 100 ksmps = 1 instr 1 kc chnget "cutoff" a1 oscil p4, p5, 100 a2 lowpass2 a1, kc, 200 out a2 endin Here is another example of the chnget opcode. It uses the file chnget.csd. Example of the chnget opcode. Credits Author: &nameistvan; 2005 manual_src~dfsg/opcodes/linenr.xml0000644000000000000000000001047312262561504016417 0ustar rootroot linenr Signal Generators:Envelope Generators linenr linenr The linen opcode extended with a final release segment. Description linenr -- same as linen except that the final segment is entered only on sensing a MIDI note release. The note is then extended by the decay time. Syntax ares linenr xamp, irise, idec, iatdec kres linenr kamp, irise, idec, iatdec Initialization irise -- rise time in seconds. A zero or negative value signifies no rise modification. idec -- decay time in seconds. Zero means no decay. An idec > idur will cause a truncated decay. iatdec -- attenuation factor by which the closing steady state value is reduced exponentially over the decay period. This value must be positive and is normally of the order of .01. A large or excessively small value is apt to produce a cutoff which is audible. A zero or negative value is illegal. Performance kamp, xamp -- input amplitude signal. linenr is unique within Csound in containing a note-off sensor and release time extender. When it senses either a score event termination or a MIDI noteoff, it will immediately extend the performance time of the current instrument by idec seconds, then execute an exponential decay towards the factor iatdec. For two or more units in an instrument, extension is by the greatest idec. You can use other pre-made envelopes which start a release segment upon recieving a note off message, like linsegr and expsegr, or you can construct more complex envelopes using xtratim and release. Note that you don't need to use xtratim if you are using linenr, since the time is extended automatically. These r units can also be modified by MIDI noteoff velocities (see veloffs). Examples Here is an example of the linenr opcode. It uses the file linenr.csd. Example of the linenr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also linsegr, expsegr, envlpxr, mxadsr, madsr, envlpx, linen, xtratim manual_src~dfsg/opcodes/tableshuffle.xml0000644000000000000000000000506612262561504017576 0ustar rootroot tableshuffle Miscellaneous tableshuffle tableshuffle shuffles the content of a function table so that each element of the source table is put into a different random position. Description This opcode can be used in order to shuffle the content of function tables into a random order but without loosing any of the elements. Imagine shuffling a deck of cards. Each element of the table is copied to a different random position. If that position was already chosen before then the next free position is chosen. The length of the table remains the same. Syntax tableshuffle ktablenum tableshufflei itablenum Performance ktablenum or itablenum -- the number of the table to shuffle. Examples Here is an example of the tableshuffle opcode. It uses the file farey7shuffled.csd. Example of the tableshuffle opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also GEN farey, tablefilter, tablecopy Credits Author: Georg Boenn University of Glamorgan, UK New in Csound version 5.13 manual_src~dfsg/opcodes/vbap.xml0000644000000000000000000001342012262561504016053 0ustar rootroot vbap Signal Modifiers:Panning and Spatialization vbap vbap Distributes an audio signal among many channels. Description Distributes an audio signal amongmany channels, up to 64 in the first form, arbitrary in the second. Syntax ar1[, ar2...] vbap asig, kazim [, kelev] [, kspread] [, ilayout] array[] vbap asig, kazim [, kelev] [, kspread] [, ilayout] Initialization ilayout -- index of the speaker layout in the range 0-99, corresponding to a call to vbaplsinit. Performance asig -- audio signal to be panned kazim -- azimuth angle of the virtual source kelev (optional) -- elevation angle of the virtual source kspread (optional) -- spreading of the virtual source (range 0 - 100). If value is zero, conventional amplitude panning is used. When kspread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. vbap takes an input signal, asig and distributes it among the outputs, according to the controls kazim and kelev, and the configured loudspeaker placement. If idim = 2, kelev is set to zero. The distribution is performed using Vector Base Amplitude Panning (VBAP - See reference). VBAP distributes the signal using loudspeaker data configured with vbaplsinit. The signal is applied to, at most, two loudspeakers in 2-D loudspeaker configurations, and three loudspeakers in 3-D loudspeaker configurations. If the virtual source is panned outside the region spanned by loudspeakers, the nearest loudspeakers are used in panning. Please note that all vbap panning opcodes require the vbap system to be initialized using vbaplsinit. Examples See the entry for vbap8 for an example of usage of the vbap opcodes. Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. Examples Here is an example of the vbap opcode. It uses the file vbap.csd. Example of the vbap opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: spread of source = 0 spread of source = 8 spread of source = 17 spread of source = 25 spread of source = 33 spread of source = 42 spread of source = 50 spread of source = 58 spread of source = 67 spread of source = 75 spread of source = 83 spread of source = 92 spread of source = 100 See Also vbapmove, vbap16, vbap16move, vbap4, vbap4move, vbap8, vbap8move, vbaplsinit, vbapz, vbapzmove Credits Author: Ville Pulkki Sibelius Academy Computer Music Studio Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Helsinki, Finland May 2000 Author: &namejohn; July 2012, Sept 2013 New in Csound Version 5.17.13. Array version new in Csound Version 6.01. manual_src~dfsg/opcodes/trandom.xml0000644000000000000000000000570412262561504016575 0ustar rootroot trandom Signal Generators:Random (Noise) Generators trandom trandom Generates a controlled pseudo-random number series between min and max values according to a trigger. Description Generates a controlled pseudo-random number series between min and max values at k-rate whenever the trigger parameter is different to 0. Syntax kout trandom ktrig, kmin, kmax Performance ktrig -- trigger (opcode produces a new random number whenever this value is not 0. kmin -- minimum range limit kmax -- maximum range limit trandom is almost identical to random opcode, except trandom updates its output with a new random value only when the ktrig argument is triggered (i.e. whenever it is not zero). Examples Here is an example of the trandom opcode. It uses the file trandom.csd. Example of the trandom opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also random Credits Written by Gabriel Maldonado. New in Csound 5.06 manual_src~dfsg/opcodes/mididefault.xml0000644000000000000000000001045312262561504017415 0ustar rootroot mididefault Real-time MIDI:MIDI/Score Interoperability mididefault mididefault Changes values, depending on MIDI activation. Description mididefault is designed to simplify writing instruments that can be used interchangeably for either score or MIDI input, and to make it easier to adapt instruments originally written for score input to work with MIDI input. In general, it should be possible to write instrument definitions that work identically with both scores and MIDI, including both MIDI files and real-time MIDI input, without using any conditional statements, and that take full advantage of MIDI voice messages. Note that correlating Csound instruments with MIDI channel numbers is done using the massign opcode for real-time performance,. For file-driven performance, instrument numbers default to MIDI channel number + 1, but the defaults are overridden by any MIDI program change messages in the file. Syntax mididefault xdefault, xvalue Performance xdefault -- specifies a default value that will be used during MIDI activation. xvalue -- overwritten by xdefault during MIDI activation, remains unchanged otherwise. If the instrument was activated by MIDI input, the opcode will overwrite the value of xvalue with the value of xdefault. If the instrument was NOT activated by MIDI input, xvalue will remain unchanged. This enables score pfields to receive a default value during MIDI activation, and score values otherwise. Adapting a score-activated Csound instrument. See the MIDI interop opcodes section for details on adapting score driven instruments for MIDI or vice-versa. Examples Here is an example of the mididefault opcode. It uses the file mididefault.csd. Example of the mididefault opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also midichannelaftertouch, midicontrolchange, midinoteoff, midinoteoncps, midinoteonkey, midinoteonoct, midinoteonpch, midipitchbend, midipolyaftertouch, midiprogramchange Credits Author: Michael Gogins New in version 4.20 manual_src~dfsg/opcodes/ino.xml0000644000000000000000000000374612262561504015722 0ustar rootroot ino Signal I/O:Signal Input ino ino Reads eight-channel audio data from an external device or stream. Description Reads eight-channel audio data from an external device or stream. Warning This opcode is designed to be used only with orchestras that have nchnls_i=8. Doing so with orchestras with nchnls_i > 8 will cause incorrect audio input. Syntax ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8 ino Performance Reads eight-channel audio data from an external device or stream. If the command-line -i flag is set, sound is read continuously from the audio input stream (e.g. stdin or a soundfile) into an internal buffer. Any number of these opcodes can read freely from this buffer. See Also diskin, in, inh, inh, inq, ins, soundin Credits Author: &namejohn; manual_src~dfsg/opcodes/agauss.xml0000644000000000000000000000076012262561504016411 0ustar rootroot agauss agauss agauss Deprecated. Description Deprecated as of version 3.49. Use the gauss opcode instead. manual_src~dfsg/opcodes/vtablewa.xml0000644000000000000000000000632212262561504016733 0ustar rootroot vtablewa Vectorial:Tables vtablewa vtablewa Write vectors (to tables -or arrays of vectors). Description This opcode writes vectors to tables at a-rate. Syntax vtablewa andx, kfn, ixmode, ainarg1 [, ainarg2, ainarg3 , .... , ainargN ] Initialization ixmode - index data mode. The default value is 0. == 0 index is treated as a raw table location, == 1 index is normalized (0 to 1). Performance andx - Index into f-table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0). kfn - table number ainarg1...ainargN - input vector elements This opcode is useful in all cases in which one needs to write sets of values associated to unique indexes (for example, multi-channel samples, STFT bin frames, spectral formants, p-field based scores etc.). The number of elements of each vector (length of the vector) is determined by the number of optional arguments on the right (ainarg1 , ainarg2, ainarg3, .... ainargN). vtablewa allows also to switch the table number at k-rate (but this is possible only when vector frames of each used table have the same number of elements, otherwise unpredictable results could occurr), as well as to choose indexing style (raw or normalized, see also ixmode argument of table opcode ). Notice that no wrap nor limit mode is implemented. So, if an index attempt to access to a zone not allocated by the table, Csound will probably crash. However this drawback can be easily avoided by using wrap or limit opcodes applied to indexes before using vtablewa, in order to correct eventual out-of-range values. Examples Here is an example of the vtablewa opcode. It uses the files vtablewa.csd. Example of the vtablewa opcode. Credits Written by Gabriel Maldonado. Example written by Andres Cabrera. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/expsegb.xml0000644000000000000000000000714612262561504016570 0ustar rootroot expsegb Signal Generators:Linear and Exponential Generators expsegb expsegb Trace a series of exponential segments between specified absolute points. Description Trace a series of exponential segments between specified absolute points. Syntax ares expsegb ia, itim1, ib [, itim2] [, ic] [...] kres expsegb ia, itim1, ib [, itim2] [, ic] [...] Initialization ia -- starting value. Zero is illegal for exponentials. ib, ic, etc. -- value at tim1 seconds, etc. For exponentials, must be non-zero and must agree in sign with ia. itim1 -- time in seconds of end of first segment. itim2, itim3, etc. -- time in seconds of subsequent ends of segments. Performance These units generate control or audio signals whose values can pass through 2 or more specified points. The last tim value may or may not equal the instrument's performance time: a shorter performance will truncate the specified pattern, while a longer one will cause the last-defined segment to continue on in the same direction. Note that the expsegb opcode does not operate correctly at audio rate when segments are shorter than a k-period. Try the expsegba opcode instead. Examples Here is an example of the expsegb opcode. It uses the file expsegb.csd. Example of the expsegb opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also expon, expseg, expsega, expsegr, line, linseg, linsegr transeg Credits Author: &namevictor; June 2011 New in version 5.14 manual_src~dfsg/opcodes/tradsyn.xml0000644000000000000000000000570412262561504016615 0ustar rootroot tradsyn Spectral Processing:Streaming tradsyn tradsyn Streaming partial track additive synthesis Description The tradsyn opcode takes an input containg a TRACKS pv streaming signal (as generated, for instance by partials),as described in Lazzarini et al, "Time-stretching using the Instantaneous Frequency Distribution and Partial Tracking", Proc.of ICMC05, Barcelona. It resynthesises the signal using linear amplitude and frequency interpolation to drive a bank of interpolating oscillators with amplitude and pitch scaling controls. Syntax asig tradsyn fin, kscal, kpitch, kmaxtracks, ifn Performance asig -- output audio rate signal fin -- input pv stream in TRACKS format kscal -- amplitude scaling kpitch -- pitch scaling kmaxtracks -- max number of tracks in resynthesis. Limiting this will cause a non-linear filtering effect, by discarding newer and higher-frequency tracks (tracks are ordered by start time and ascending frequency, respectively) ifn -- function table containing one cycle of a sinusoid (sine or cosine). Examples Here is an example of the tradsyn opcode. It uses the file tradsyn.csd. Example of the tradsyn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of an ifd-analysis signal and linear additive resynthesis with pitch shifting. Credits Author: &namevictor; June 2005 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/maxabsaccum.xml0000644000000000000000000000777212262561504017424 0ustar rootroot maxabsaccum Signal Modifiers:Comparators and Accumulators maxabsaccum maxabsaccum Accumulates the maximum of the absolute values of audio signals. Description maxabsaccum compares two audio-rate variables and stores the maximum of their absolute values into the first. Syntax maxabsaccum aAccumulator, aInput Performance aAccumulator -- audio variable to store the maximum value aInput -- signal that aAccumulator is compared to The maxabsaccum opcode is designed to accumulate the maximum value from among many audio signals that may be in different note instances, different channels, or otherwise cannot all be compared at once using the maxabs opcode. maxabsaccum is identical to maxaccum except that it takes the absolute value of aInput before the comparison. Its semantics are similar to vincr since aAccumulator is used as both an input and an output variable, except that maxabsaccum keeps the maximum absolute value instead of adding the signals together. maxabsaccum performs the following operation on each pair of samples: if (abs(aInput) > aAccumulator) aAccumulator = abs(aInput) aAccumulator will usually be a global audio variable. At the end of any given computation cycle (k-period), after its value is read and used in some way, the accumulator variable should usually be reset to zero (perhaps by using the clear opcode). Clearing to zero is sufficient for maxabsaccum, unlike the maxaccum opcode. Examples Here is an example of the maxabsaccum opcode. It uses the file maxabsaccum.csd. Example of the maxabsaccum opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also minabsaccum, maxaccum, minaccum, max, min, maxabs, minabs, vincr, clear Credits Author: &nameanthony; March 2006 New in Csound version 5.01 manual_src~dfsg/opcodes/ctrl21.xml0000644000000000000000000000760612262561504016243 0ustar rootroot ctrl21 Real-time MIDI:Input ctrl21 ctrl21 Allows a floating-point 21-bit MIDI signal scaled with a minimum and a maximum range. Description Allows a floating-point 21-bit MIDI signal scaled with a minimum and a maximum range. Syntax idest ctrl21 ichan, ictlno1, ictlno2, ictlno3, imin, imax [, ifn] kdest ctrl21 ichan, ictlno1, ictlno2, ictlno3, kmin, kmax [, ifn] Initialization idest -- output signal ichan -- MIDI channel number (1-16) ictlno1 -- most-significant byte controller number (0-127) ictlno2 -- mid-significant byte controller number (0-127) ictlno3 -- least-significant byte controller number (0-127) imin -- user-defined minimum floating-point value of output imax -- user-defined maximum floating-point value of output ifn (optional) -- table to be read when indexing is required. Table must be normalized. Output is scaled according to imax and imin val. Performance kdest -- output signal kmin -- user-defined minimum floating-point value of output kmax -- user-defined maximum floating-point value of output ctrl21 (i- and k-rate 21 bit MIDI control) allows a floating-point 21-bit MIDI signal scaled with a minimum and a maximum range. Minimum and maximum values can be varied at k-rate. It can use optional interpolated table indexing. It requires three MIDI controllers as input. ctrl21 differs from midic21 because it can be included in score oriented instruments without Csound crashes. It needs the additional parameter ichan containing the MIDI channel of the controller. MIDI channel is the same for all the controllers used in a single ctrl21 opcode. See Also ctrl7, ctrl14, initc7, initc14, initc21, midic7, midic14, midic21 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/lorisread.xml0000644000000000000000000001455712262561504017123 0ustar rootroot lorisread Spectral Processing:Loris lorisread lorisread Imports a set of bandwidth-enhanced partials from a SDIF-format data file, applying control-rate frequency, amplitude, and bandwidth scaling envelopes, and stores the modified partials in memory. Syntax lorisread ktimpnt, ifilcod, istoreidx, kfreqenv, kampenv, kbwenv[, ifadetime] Description lorisread imports a set of bandwidth-enhanced partials from a SDIF-format data file, applying control-rate frequency, amplitude, and bandwidth scaling envelopes, and stores the modified partials in memory. Initialization ifilcod - integer or character-string denoting a control-file derived from reassigned bandwidth-enhanced analysis of an audio signal. An integer denotes the suffix of a file loris.sdif (e.g. loris.sdif.1); a character-string (in double quotes) gives a filename, optionally a full pathname. If not a full pathname, the file is sought first in the current directory, then in the one given by the environment variable SADIR (if defined). The reassigned bandwidth-enhanced data file contains breakpoint frequency, amplitude, noisiness, and phase envelope values organized for bandwidth-enhanced additive resynthesis. The control data must conform to one of the SDIF formats that can be Loris stores partials in SDIF RBEP frames. Each RBEP frame contains one RBEP matrix, and each row in a RBEP matrix describes one breakpoint in a Loris partial. A RBEL frame containing one RBEL matrix describing the labeling of the partials may precede the first RBEP frame in the SDIF file. The RBEP and RBEL frame and matrix definitions are included in the SDIF file's header. In addition to RBEP frames, Loris can also read and write SDIF 1TRC frames. Since 1TRC frames do not represent bandwidth-enhancement or the exact timing of Loris breakpoints, their use is not recommended. 1TRC capabilities are provided to allow interchange with programs that are unable to handle RBEP frames. istoreidx, ireadidx, isrcidx, itgtidx are labels that identify a stored set of bandwidth-enhanced partials. lorisread imports partials from a SDIF file and stores them with the integer label istoreidx. lorismorph morphs sets of partials labeled isrcidx and itgtidx, and stores the resulting partials with the integer label istoreidx. lorisplay renders the partials stored with the label ireadidx. The labels are used only at initialization time, and may be reused without any cost or benefit in efficiency, and without introducing any interaction between instruments or instances. ifadetime (optional) - In general, partials exported from Loris begin and end at non-zero amplitude. In order to prevent artifacts, it is very often necessary to fade the partials in and out, instead of turning them abruptly on and off. Specification of a non-zero ifadetime causes partials to fade in at their onsets and to fade out at their terminations. This is achieved by adding two more breakpoints to each partial: one ifadetime seconds before the start time and another ifadetime seconds after the end time. (However, no breakpoint will be introduced at a time less than zero. If necessary, the onset fade time will be shortened.) The additional breakpoints at the partial onset and termination will have the same frequency and bandwidth as the first and last breakpoints in the partial, respectively, but their amplitudes will be zero. The phase of the new breakpoints will be extrapolated to preserve phase correctness. If no value is specified, ifadetime defaults to zero. Note that the fadetime may not be exact, since the partial parameter envelopes are sampled at the control rate (krate) and indexed by ktimpnt (see below), and not by real time. Performance lorisread reads pre-computed Reassigned Bandwidth-Enhanced analysis data from a file stored in SDIF format, as described above. The passage of time through this file is specified by ktimpnt, which represents the time in seconds. ktimpnt must always be positive, but can move forwards or backwards in time, be stationary or discontinuous, as a pointer into the analysis file. kfreqenv is a control-rate transposition factor: a value of 1 incurs no transposition, 1.5 transposes up a perfect fifth, and .5 down an octave. kampenv is a control-rate scale factor that is applied to all partial amplitude envelopes. kbwenv is a control-rate scale factor that is applied to all partial bandwidth or noisiness envelopes. The bandwidth-enhanced partial data is stored in memory with a specified label for future access by another generator. Examples Here is an example of the lorisread opcode. It uses the file lorisread.csd. Example of the lorisread opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits This implementation of the Loris unit generators was written by Kelly Fitz (loris@cerlsoundgroup.org). It is patterned after a prototype implementation of the lorisplay unit generator written by Corbin Champion, and based on the method of Bandwidth-Enhanced Additive Synthesis and on the sound morphing algorithms implemented in the Loris library for sound modeling and manipulation. The opcodes were further adapted as a plugin for Csound 5 by Michael Gogins. manual_src~dfsg/opcodes/top.xml0000644000000000000000000005524412262561504015737 0ustar rootroot Orchestra Opcodes and Operators &opcodesnotequal; &opcodesdefine; &opcodesinclude; &opcodesundef; &opcodesifdef; &opcodesifndef; &opcodesdollar; &opcodesmodulus; &opcodesopand; &opcodesgreaterthan; &opcodesgreaterequal; &opcodeslessthan; &opcodeslessequal; &opcodesmultiplies; &opcodesadds; &opcodessubtracts; &opcodesdivides; &opcodesassign; &opcodesplusbecomes; &opcodesequals; &opcodesraises; &opcodesopor; &opcodes0dbfs; &opcodesopbitshl; &opcodesopbitshr; &opcodesopbitand; &opcodesopbitor; &opcodesopbitnot; &opcodesopnonequiv; &opcodesopa; &opcodesabs; &opcodesactive; &opcodesadsr; &opcodesadsyn; &opcodesadsynt; &opcodesadsynt2; &opcodesaftouch; &opcodesalpass; &opcodesalwayson; &opcodesampdb; &opcodesampdbfs; &opcodesampmidi; &opcodesampmidid; &opcodesareson; &opcodesaresonk; &opcodesatone; &opcodesatonek; &opcodesatonex; &opcodesATSadd; &opcodesATSaddnz; &opcodesATSbufread; &opcodesATScross; &opcodesATSinfo; &opcodesATSinterpread; &opcodesATSread; &opcodesATSreadnz; &opcodesATSpartialtap; &opcodesATSsinnoi; &opcodesbabo; &opcodesbalance; &opcodesbamboo; &opcodesbarmodel; &opcodesbbcutm; &opcodesbbcuts; &opcodesbetarand; &opcodesbexprnd; &opcodesbformenc1; &opcodesbformdec1; &opcodesbinit; &opcodesbiquad; &opcodesbiquada; &opcodesbirnd; &opcodesbqrez; &opcodesbutbp; &opcodesbutbr; &opcodesbuthp; &opcodesbutlp; &opcodesbutterbp; &opcodesbutterbr; &opcodesbutterhp; &opcodesbutterlp; &opcodesbutton; &opcodesbuzz; &opcodescabasa; &opcodescauchy; &opcodescauchyi; &opcodesceil; &opcodescell; &opcodescent; &opcodescentroid; &opcodescggoto; &opcodeschanctrl; &opcodeschanged; &opcodeschani; &opcodeschano; &opcodeschebyshevpoly; &opcodescheckbox; &opcodeschn; &opcodeschnclear; &opcodeschnexport; &opcodeschnget; &opcodeschnmix; &opcodeschnparams; &opcodeschnrecv; &opcodeschnsend; &opcodeschnset; &opcodeschuap; &opcodescigoto; &opcodesckgoto; &opcodesclear; &opcodesclfilt; &opcodesclip; &opcodesclockoff; &opcodesclockon; &opcodescngoto; &opcodescomb; &opcodescombinv; &opcodescompress; &opcodescompileorc; &opcodescompilestr; &opcodesconnect; &opcodescontrol; &opcodesconvle; &opcodesconvolve; &opcodescopya2ftab; &opcodescopyf2array; &opcodescos; &opcodescosseg; &opcodescossegb; &opcodescossegr; &opcodescosh; &opcodescosinv; &opcodescps2pch; &opcodescpsmidi; &opcodescpsmidib; &opcodescpsmidinn; &opcodescpsoct; &opcodescpspch; &opcodescpstmid; &opcodescpstun; &opcodescpstuni; &opcodescpsxpch; &opcodescpumeter; &opcodescpuprc; &opcodescross2; &opcodescrossfm; &opcodescrunch; &opcodesctrl14; &opcodesctrl21; &opcodesctrl7; &opcodesctrlinit; &opcodescuserrnd; &opcodesdam; &opcodesdate; &opcodesdates; &opcodesdb; &opcodesdbamp; &opcodesdbfsamp; &opcodesdcblock; &opcodesdcblock2; &opcodesdconv; &opcodesdelay; &opcodesdelay1; &opcodesdelayk; &opcodesdelayr; &opcodesdelayw; &opcodesdeltap; &opcodesdeltap3; &opcodesdeltapi; &opcodesdeltapn; &opcodesdeltapx; &opcodesdeltapxw; &opcodesdenorm; &opcodesdiff; &opcodesdiskgrain; &opcodesdiskin2; &opcodesdispfft; &opcodesdisplay; &opcodesdistort; &opcodesdistort1; &opcodesdivz; &opcodesdoppler; &opcodesdownsamp; &opcodesdripwater; &opcodesdssiactivate; &opcodesdssiaudio; &opcodesdssictls; &opcodesdssiinit; &opcodesdssilist; &opcodesdumpk; &opcodesdumpk2; &opcodesdumpk3; &opcodesdumpk4; &opcodesduserrnd; &opcodesdust; &opcodesdust2; &opcodeselse; &opcodeselseif; &opcodesendif; &opcodesendin; &opcodesendop; &opcodesenvlpx; &opcodesenvlpxr; &opcodesephasor; &opcodeseqfil; &opcodesevalstr; &opcodesevent; &opcodesevent_i; &opcodesexitnow; &opcodesexp; &opcodesexpcurve; &opcodesexpon; &opcodesexprand; &opcodesexprandi; &opcodesexpseg; &opcodesexpsega; &opcodesexpsegb; &opcodesexpsegba; &opcodesexpsegr; &opcodesfaustaudio; &opcodesfaustcompile; &opcodesfaustctl; &opcodesfaustgen; &opcodesfareylen; &opcodesfareyleni; &opcodesficlose; &opcodesfilebit; &opcodesfilelen; &opcodesfilenchnls; &opcodesfilepeak; &opcodesfilesr; &opcodesfilevalid; &opcodesfillarray; &opcodesarray; &opcodesfilter2; &opcodesfin; &opcodesfini; &opcodesfink; &opcodesfiopen; &opcodesflanger; &opcodesflashtxt; &opcodesflbox; &opcodesflbutbank; &opcodesflbutton; &opcodesflclosebutton; &opcodesflcolor; &opcodesflcolor2; &opcodesflcount; &opcodesflexecbutton; &opcodesflgetsnap; &opcodesflgroup; &opcodesflgroupend; &opcodesflgroup_end; &opcodesflhide; &opcodesflhvsbox; &opcodesflhvsboxsetvalue; &opcodesfljoy; &opcodesflkeyin; &opcodesflknob; &opcodesfllabel; &opcodesflloadsnap; &opcodesflmouse; &opcodesflooper; &opcodesflooper2; &opcodesfloor; &opcodesflpack; &opcodesflpackend; &opcodesflpack_end; &opcodesflpanel; &opcodesflpanelend; &opcodesflpanel_end; &opcodesflprintk; &opcodesflprintk2; &opcodesflroller; &opcodesflrun; &opcodesflsavesnap; &opcodesflscroll; &opcodesflscrollend; &opcodesflscroll_end; &opcodesflsetalign; &opcodesflsetbox; &opcodesflsetcolor; &opcodesflsetcolor2; &opcodesflsetfont; &opcodesflsetposition; &opcodesflsetsize; &opcodesflsetsnap; &opcodesflsetsnapgroup; &opcodesflsettext; &opcodesflsettextcolor; &opcodesflsettextsize; &opcodesflsettexttype; &opcodesflsetvali; &opcodesflsetval; &opcodesflshow; &opcodesflslidbnk; &opcodesflslidbnk2; &opcodesflslidbnkgethandle; &opcodesflslidbnkset; &opcodesflslidbnksetk; &opcodesflslidbnk2set; &opcodesflslidbnk2setk; &opcodesflslider; &opcodesfltabs; &opcodesfltabsend; &opcodesfltabs_end; &opcodesfltext; &opcodesflupdate; &opcodesfluidAllOut; &opcodesfluidCCi; &opcodesfluidCCk; &opcodesfluidControl; &opcodesfluidEngine; &opcodesfluidLoad; &opcodesfluidNote; &opcodesfluidOut; &opcodesfluidProgramSelect; &opcodesfluidSetInterpMethod; &opcodesflvalue; &opcodesflvkeybd; &opcodesflvslidbnk; &opcodesflvslidbnk2; &opcodesflxyin; &opcodesfmb3; &opcodesfmbell; &opcodesfmmetal; &opcodesfmpercfl; &opcodesfmrhode; &opcodesfmvoice; &opcodesfmwurlie; &opcodesfof; &opcodesfof2; &opcodesfofilter; &opcodesfog; &opcodesfold; &opcodesfollow; &opcodesfollow2; &opcodesfoscil; &opcodesfoscili; &opcodesfout; &opcodesfouti; &opcodesfoutir; &opcodesfoutk; &opcodesfprintks; &opcodesfprints; &opcodesfrac; &opcodesfractalnoise; &opcodesfreeverb; &opcodesftchnls; &opcodesftconv; &opcodesftcps; &opcodesftfree; &opcodesftgen; &opcodesftgenonce; &opcodesftgentmp; &opcodesftlen; &opcodesftload; &opcodesftloadk; &opcodesftlptim; &opcodesftmorf; &opcodesftsave; &opcodesftsavek; &opcodesftsr; &opcodesgain; &opcodesgainslider; &opcodesgauss; &opcodesgaussi; &opcodesgausstrig; &opcodesgbuzz; &opcodesgenarray; &opcodesgenarray_i; &opcodesgendy; &opcodesgendyc; &opcodesgendyx; &opcodesgetcfg; &opcodesgogobel; &opcodesgoto; &opcodesgrain; &opcodesgrain2; &opcodesgrain3; &opcodesgranule; &opcodesguiro; &opcodesharmon; &opcodesharmon2; &opcodeshilbert; &opcodeshrtfer; &opcodeshrtfearly; &opcodeshrtfmove; &opcodeshrtfmove2; &opcodeshrtfreverb; &opcodeshrtfstat; &opcodeshsboscil; &opcodeshvs1; &opcodeshvs2; &opcodeshvs3; &opcodesopi; &opcodesif; &opcodesigoto; &opcodesihold; &opcodesimagecreate; &opcodesimagefree; &opcodesimagegetpixel; &opcodesimageload; &opcodesimagesave; &opcodesimagesetpixel; &opcodesimagesize; &opcodesin; &opcodesin32; &opcodesinch; &opcodesinh; &opcodesinit; &opcodesinitc14; &opcodesinitc21; &opcodesinitc7; &opcodesinleta; &opcodesinletk; &opcodesinletkid; &opcodesinletf; &opcodesino; &opcodesinq; &opcodesinrg; &opcodesins; &opcodesinsremot; &opcodesinsglobal; &opcodesinstr; &opcodesinstance; &opcodesint; &opcodesinteg; &opcodesinterp; &opcodesinvalue; &opcodesinx; &opcodesinz; &opcodesjackoaudioin; &opcodesjackoaudioinconnect; &opcodesjackoaudioout; &opcodesjackoaudiooutconnect; &opcodesjackofreewheel; &opcodesjackoinfo; &opcodesjackoinit; &opcodesjackomidiinconnect; &opcodesjackomidioutconnect; &opcodesjackomidiout; &opcodesjackonoteout; &opcodesjackoon; &opcodesjackotransport; &opcodesjacktransport; &opcodesjitter; &opcodesjitter2; &opcodesjoystick; &opcodesjspline; &opcodesopk; &opcodeskgoto; &opcodeskill; &opcodeskr; &opcodesksmps; &opcodeslenarray; &opcodeslfo; &opcodeslimit; &opcodesline; &opcodeslinen; &opcodeslinenr; &opcodeslineto; &opcodeslinrand; &opcodeslinseg; &opcodeslinsegb; &opcodeslinsegr; &opcodeslocsend; &opcodeslocsig; &opcodeslog; &opcodeslog10; &opcodeslog2; &opcodeslogbtwo; &opcodeslogcurve; &opcodesloop_ge; &opcodesloop_gt; &opcodesloop_le; &opcodesloop_lt; &opcodesloopseg; &opcodesloopsegp; &opcodeslooptseg; &opcodesloopxseg; &opcodeslorenz; &opcodeslorisread; &opcodeslorismorph; &opcodeslorisplay; &opcodesloscil; &opcodesloscil3; &opcodesloscilx; &opcodeslowpass2; &opcodeslowres; &opcodeslowresx; &opcodeslpf18; &opcodeslpfreson; &opcodeslphasor; &opcodeslpinterp; &opcodeslposcil; &opcodeslposcil3; &opcodeslposcila; &opcodeslposcilsa; &opcodeslposcilsa2; &opcodeslpread; &opcodeslpreson; &opcodeslpshold; &opcodeslpsholdp; &opcodeslpslot; &opcodeslua_exec; &opcodeslua_opdef; &opcodeslua_opcall; &opcodesmac; &opcodesmaca; &opcodesmadsr; &opcodesmandel; &opcodesmandol; &opcodesmaparray; &opcodesmarimba; &opcodesmassign; &opcodesmax; &opcodesmaxabs; &opcodesmaxabsaccum; &opcodesmaxaccum; &opcodesmaxalloc; &opcodesmax_k; &opcodesmaxarray; &opcodesmclock; &opcodesmdelay; &opcodesmedian; &opcodesmediank; &opcodesmetro; &opcodesmidglobal; &opcodesmidic14; &opcodesmidic21; &opcodesmidic7; &opcodesmidichannelaftertouch; &opcodesmidichn; &opcodesmidicontrolchange; &opcodesmidictrl; &opcodesmididefault; &opcodesmidiin; &opcodesmidifilestatus; &opcodesmidinoteoff; &opcodesmidinoteoncps; &opcodesmidinoteonkey; &opcodesmidinoteonoct; &opcodesmidinoteonpch; &opcodesmidion2; &opcodesmidion; &opcodesmidiout; &opcodesmidipitchbend; &opcodesmidipolyaftertouch; &opcodesmidiprogramchange; &opcodesmiditempo; &opcodesmidremot; &opcodesmin; &opcodesminabs; &opcodesminabsaccum; &opcodesminaccum; &opcodesmincer; &opcodesminarray; &opcodesmirror; &opcodesmixersetlevel; &opcodesmixersetleveli; &opcodesmixergetlevel; &opcodesmixersend; &opcodesmixerreceive; &opcodesmixerclear; &opcodesmode; &opcodesmodmatrix; &opcodesmonitor; &opcodesmoog; &opcodesmoogladder; &opcodesmoogvcf; &opcodesmoogvcf2; &opcodesmoscil; &opcodesmp3in; &opcodesmp3len; &opcodesmpulse; &opcodesmrtmsg; &opcodesmultitap; &opcodesmute; &opcodesmxadsr; &opcodesnchnls; &opcodesnchnlsi; &opcodesnestedap; &opcodesnlfilt; &opcodesnlfilt2; &opcodesnoise; &opcodesnoteoff; &opcodesnoteon; &opcodesnoteondur2; &opcodesnoteondur; &opcodesnotnum; &opcodesnreverb; &opcodesnrpn; &opcodesnsamp; &opcodesnstrnum; &opcodesntrpol; &opcodesoctave; &opcodesoctcps; &opcodesoctmidi; &opcodesoctmidib; &opcodesoctmidinn; &opcodesoctpch; &opcodesopcode; &opcodesoscbnk; &opcodesoscil1; &opcodesoscil1i; &opcodesoscil3; &opcodesoscil; &opcodesoscili; &opcodesoscilikt; &opcodesosciliktp; &opcodesoscilikts; &opcodesosciln; &opcodesoscils; &opcodesoscilx; &opcodesOSCinit; &opcodesOSClisten; &opcodesOSCsend; &opcodesout32; &opcodesout; &opcodesoutc; &opcodesoutch; &opcodesouth; &opcodesoutiat; &opcodesoutic14; &opcodesoutic; &opcodesoutipat; &opcodesoutipb; &opcodesoutipc; &opcodesoutkat; &opcodesoutkc14; &opcodesoutkc; &opcodesoutkpat; &opcodesoutkpb; &opcodesoutkpc; &opcodesoutleta; &opcodesoutletf; &opcodesoutletk; &opcodesoutletkid; &opcodesouto; &opcodesoutq1; &opcodesoutq2; &opcodesoutq3; &opcodesoutq4; &opcodesoutq; &opcodesoutrg; &opcodesouts1; &opcodesouts2; &opcodesouts; &opcodesoutvalue; &opcodesoutx; &opcodesoutz; &opcodesp5gconnect; &opcodesp5gdata; &opcodesp; &opcodespan2; &opcodespan; &opcodespareq; &opcodespartials; &opcodespartikkel; &opcodespartikkelsync; &opcodespassign; &opcodespcauchy; &opcodespchbend; &opcodespchmidi; &opcodespchmidib; &opcodespchmidinn; &opcodespchoct; &opcodespconvolve; &opcodespcount; &opcodespdclip; &opcodespdhalf; &opcodespdhalfy; &opcodespeak; &opcodespgmassign; &opcodesphaser1; &opcodesphaser2; &opcodesphasor; &opcodesphasorbnk; &opcodespindex; &opcodespinkish; &opcodespitch; &opcodespitchamdf; &opcodesplanet; &opcodesplaterev; &opcodesplltrack; &opcodespluck; &opcodespoisson; &opcodespolyaft; &opcodespolynomial; &opcodespop; &opcodespop_f; &opcodesport; &opcodesportk; &opcodesposcil3; &opcodesposcil; &opcodespow; &opcodespowershape; &opcodespowoftwo; &opcodesprealloc; &opcodesprepiano; &opcodesprint; &opcodesprintf; &opcodesprintk2; &opcodesprintk; &opcodesprintks; &opcodesprints; &opcodesproduct; &opcodespset; &opcodesptable; &opcodesptablei; &opcodesptable3; &opcodesptablew; &opcodesptrack; &opcodespush; &opcodespush_f; &opcodesputs; &opcodespvadd; &opcodespvbufread; &opcodespvcross; &opcodespvinterp; &opcodespvoc; &opcodespvread; &opcodespvsadsyn; &opcodespvsanal; &opcodespvsarp; &opcodespvsbandp; &opcodespvsbandr; &opcodespvsbin; &opcodespvsblur; &opcodespvsbuffer; &opcodespvsbufread; &opcodespvsbufread2; &opcodespvscale; &opcodespvscent; &opcodespvscross; &opcodespvsdemix; &opcodespvsdiskin; &opcodespvsdisp; &opcodespvsfilter; &opcodespvsfread; &opcodespvsfreeze; &opcodespvsftr; &opcodespvsftw; &opcodespvsfwrite; &opcodespvsgain; &opcodespvshift; &opcodespvsifd; &opcodespvsinfo; &opcodespvsinit; &opcodespvsin; &opcodespvslock; &opcodespvsmaska; &opcodespvsmix; &opcodespvsmorph; &opcodespvsmooth; &opcodespvsout; &opcodespvsosc; &opcodespvspitch; &opcodespvstanal; &opcodespvstencil; &opcodespvsvoc; &opcodespvsynth; &opcodespvswarp; &opcodespvs2tab; &opcodespyassign; &opcodespycall; &opcodespyeval; &opcodespyexec; &opcodespyinit; &opcodespyrun; &opcodespwd; &opcodesqinf; &opcodesqnan; &opcodesrand; &opcodesrandh; &opcodesrandi; &opcodesrandom; &opcodesrandomh; &opcodesrandomi; &opcodesrbjeq; &opcodesreadclock; &opcodesreadf; &opcodesreadfi; &opcodesreadk; &opcodesreadk2; &opcodesreadk3; &opcodesreadk4; &opcodesreadscore; &opcodesreadscratch; &opcodesreinit; &opcodesrelease; &opcodesremoteport; &opcodesremove; &opcodesrepluck; &opcodesreson; &opcodesresonk; &opcodesresonr; &opcodesresonx; &opcodesresonxk; &opcodesresony; &opcodesresonz; &opcodesresyn; &opcodesreturn; &opcodesreverb; &opcodesreverb2; &opcodesreverbsc; &opcodesrewindscore; &opcodesrezzy; &opcodesrigoto; &opcodesrireturn; &opcodesrms; &opcodesrnd; &opcodesrnd31; &opcodesround; &opcodesrspline; &opcodesrtclock; &opcodess16b14; &opcodess32b14; &opcodessamphold; &opcodessandpaper; &opcodesscale; &opcodesscalearray; &opcodesscanhammer; &opcodesscans; &opcodesscantable; &opcodesscanu; &opcodesschedkwhen; &opcodesschedkwhennamed; &opcodesschedule; &opcodesschedwhen; &opcodesscoreline; &opcodesscoreline_i; &opcodesseed; &opcodessekere; &opcodessemitone; &opcodessense; &opcodessensekey; &opcodesserialBegin; &opcodesserialEnd; &opcodesserialFlush; &opcodesserialPrint; &opcodesserialRead; &opcodesserialWrite_i; &opcodesserialWrite; &opcodesseqtime2; &opcodesseqtime; &opcodessetctrl; &opcodessetksmps; &opcodessetscorepos; &opcodessfilist; &opcodessfinstr3; &opcodessfinstr3m; &opcodessfinstr; &opcodessfinstrm; &opcodessfload; &opcodessflooper; &opcodessfpassign; &opcodessfplay3; &opcodessfplay3m; &opcodessfplay; &opcodessfplaym; &opcodessfplist; &opcodessfpreset; &opcodesshaker; &opcodessignum; &opcodessin; &opcodessinh; &opcodessininv; &opcodessinsyn; &opcodessleighbells; &opcodesslider16; &opcodesslider16f; &opcodesslider16table; &opcodesslider16tablef; &opcodesslider32; &opcodesslider32f; &opcodesslider32table; &opcodesslider32tablef; &opcodesslider64; &opcodesslider64f; &opcodesslider64table; &opcodesslider64tablef; &opcodesslider8; &opcodesslider8f; &opcodesslider8table; &opcodesslider8tablef; &opcodessliderkawai; &opcodessndload; &opcodessndloop; &opcodessndwarp; &opcodessndwarpst; &opcodessockrecv; &opcodessocksend; &opcodessoundin; &opcodesspace; &opcodesspat3d; &opcodesspat3di; &opcodesspat3dt; &opcodesspdist; &opcodesspecaddm; &opcodesspecdiff; &opcodesspecdisp; &opcodesspecfilt; &opcodesspechist; &opcodesspecptrk; &opcodesspecscal; &opcodesspecsum; &opcodesspectrum; &opcodessplitrig; &opcodessprintf; &opcodessprintfk; &opcodesspsend; &opcodessqrt; &opcodessr; &opcodesstack; &opcodesstatevar; &opcodesstix; &opcodesSTKbandedwg; &opcodesSTKbeethree; &opcodesSTKblowbotl; &opcodesSTKblowhole; &opcodesSTKbowed; &opcodesSTKbrass; &opcodesSTKclarinet; &opcodesSTKdrummer; &opcodesSTKflute; &opcodesSTKfmvoices; &opcodesSTKhevymetl; &opcodesSTKmandolin; &opcodesSTKmodalbar; &opcodesSTKmoog; &opcodesSTKpercflut; &opcodesSTKplucked; &opcodesSTKresonate; &opcodesSTKrhodey; &opcodesSTKsaxofony; &opcodesSTKshakers; &opcodesSTKsimple; &opcodesSTKsitar; &opcodesSTKstifkarp; &opcodesSTKtubebell; &opcodesSTKvoiceform; &opcodesSTKwhistle; &opcodesSTKwurley; &opcodesstrchar; &opcodesstrchark; &opcodesstrcpy; &opcodesstrcpyk; &opcodesstrcat; &opcodesstrcatk; &opcodesstrcmp; &opcodesstrcmpk; &opcodesstreson; &opcodesstrfromurl; &opcodesstrget; &opcodesstrindex; &opcodesstrindexk; &opcodesstrlen; &opcodesstrlenk; &opcodesstrlower; &opcodesstrlowerk; &opcodesstrrindex; &opcodesstrrindexk; &opcodesstrset; &opcodesstrsub; &opcodesstrsubk; &opcodesstrtod; &opcodesstrtodk; &opcodesstrtol; &opcodesstrtolk; &opcodesstrupper; &opcodesstrupperk; &opcodessubinstr; &opcodessubinstrinit; &opcodessum; &opcodessumarray; &opcodessvfilter; &opcodessyncgrain; &opcodessyncloop; &opcodessyncphasor; &opcodessystem; &opcodestb; &opcodestab; &opcodestabrec; &opcodestable; &opcodestable3; &opcodestablecopy; &opcodestablefilter; &opcodestablefilteri; &opcodestablegpw; &opcodestablei; &opcodestableicopy; &opcodestableigpw; &opcodestableikt; &opcodestableimix; &opcodestableiw; &opcodestablekt; &opcodestablemix; &opcodestableng; &opcodestablera; &opcodestableseg; &opcodestableshuffle; &opcodestablew; &opcodestablewa; &opcodestablewkt; &opcodestablexkt; &opcodestablexseg; &opcodestabmorph; &opcodestabmorpha; &opcodestabmorphak; &opcodestabmorphi; &opcodestabplay; &opcodesslicearray; &opcodestabsum; &opcodestab2pvs; &opcodestambourine; &opcodestan; &opcodestanh; &opcodestaninv; &opcodestaninv2; &opcodestbvcf; &opcodestempest; &opcodestempo; &opcodestemposcal; &opcodestempoval; &opcodestigoto; &opcodestimedseq; &opcodestimeinstk; &opcodestimeinsts; &opcodestimek; &opcodestimes; &opcodestimout; &opcodestival; &opcodestlineto; &opcodestone; &opcodestonek; &opcodestonex; &opcodestrandom; &opcodestradsyn; &opcodestranseg; &opcodestransegb; &opcodestransegr; &opcodestrcross; &opcodestrfilter; &opcodestrhighest; &opcodestrigger; &opcodestrigseq; &opcodestrirand; &opcodestrlowest; &opcodestrmix; &opcodestrscale; &opcodestrshift; &opcodestrsplit; &opcodesturnoff; &opcodesturnoff2; &opcodesturnon; &opcodesunirand; &opcodesuntil; &opcodesupsamp; &opcodesurandom; &opcodesurd; &opcodesvadd; &opcodesvaddi; &opcodesvaddv; &opcodesvaddv_i; &opcodesvaget; &opcodesvalpass; &opcodesvaset; &opcodesvbap; &opcodesvbapmove; &opcodesvbapg; &opcodesvbapgmove; &opcodesvbap16; &opcodesvbap16move; &opcodesvbap4; &opcodesvbap4move; &opcodesvbap8; &opcodesvbap8move; &opcodesvbaplsinit; &opcodesvbapz; &opcodesvbapzmove; &opcodesvcella; &opcodesvco; &opcodesvco2; &opcodesvco2ft; &opcodesvco2ift; &opcodesvco2init; &opcodesvcomb; &opcodesvcopy; &opcodesvcopy_i; &opcodesvdelay; &opcodesvdelay3; &opcodesvdelayx; &opcodesvdelayxq; &opcodesvdelayxs; &opcodesvdelayxw; &opcodesvdelayxwq; &opcodesvdelayxws; &opcodesvdivv; &opcodesvdivv_i; &opcodesvdelayk; &opcodesvecdelay; &opcodesveloc; &opcodesvexp; &opcodesvexpi; &opcodesvexpseg; &opcodesvexpv; &opcodesvexpv_i; &opcodesvibes; &opcodesvibr; &opcodesvibrato; &opcodesvincr; &opcodesvlimit; &opcodesvlinseg; &opcodesvlowres; &opcodesvmap; &opcodesvmirror; &opcodesvmult; &opcodesvmulti; &opcodesvmultv; &opcodesvmultv_i; &opcodesvoice; &opcodesvosim; &opcodesvphaseseg; &opcodesvport; &opcodesvpow; &opcodesvpowi; &opcodesvpowv; &opcodesvpowv_i; &opcodesvpvoc; &opcodesvrandh; &opcodesvrandi; &opcodesvstaudio; &opcodesvstbankload; &opcodesvstedit; &opcodesvstinit; &opcodesvstinfo; &opcodesvstmidiout; &opcodesvstnote; &opcodesvstparamset; &opcodesvstprogset; &opcodesvsubv; &opcodesvsubv_i; &opcodesvtable1k; &opcodesvtablei; &opcodesvtablek; &opcodesvtablea; &opcodesvtablewi; &opcodesvtablewk; &opcodesvtablewa; &opcodesvtabi; &opcodesvtabk; &opcodesvtaba; &opcodesvtabwi; &opcodesvtabwk; &opcodesvtabwa; &opcodesvwrap; &opcodeswaveset; &opcodesweibull; &opcodeswgbow; &opcodeswgbowedbar; &opcodeswgbrass; &opcodeswgclar; &opcodeswgflute; &opcodeswgpluck; &opcodeswgpluck2; &opcodeswguide1; &opcodeswguide2; &opcodeswiiconnect; &opcodeswiidata; &opcodeswiirange; &opcodeswiisend; &opcodeswrap; &opcodeswritescratch; &opcodeswterrain; &opcodesxadsr; &opcodesxin; &opcodesxout; &opcodesxscanmap; &opcodesxscansmap; &opcodesxscans; &opcodesxscanu; &opcodesxtratim; &opcodesxyin; &opcodeszacl; &opcodeszakinit; &opcodeszamod; &opcodeszar; &opcodeszarg; &opcodeszaw; &opcodeszawm; &opcodeszfilter2; &opcodeszir; &opcodesziw; &opcodesziwm; &opcodeszkcl; &opcodeszkmod; &opcodeszkr; &opcodeszkw; &opcodeszkwm; manual_src~dfsg/opcodes/pvsbin.xml0000644000000000000000000000442312262561504016427 0ustar rootroot pvsbin Spectral Processing:Streaming pvsbin pvsbin Obtain the amp and freq values off a PVS signal bin. Description Obtain the amp and freq values off a PVS signal bin as k-rate variables. Syntax kamp, kfr pvsbin fsig, kbin Performance kamp -- bin amplitude kfr -- bin frequency fsig -- an input pv stream kbin -- bin number Examples Here is an example of the pvsbin opcode. It uses the file pvsbin.csd. This example uses realtime input, but you can also use it for soundfile input. Example of the pvsbin opcode See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namevictor; August 2006 manual_src~dfsg/opcodes/bformenc1.xml0000644000000000000000000000556312262561504017010 0ustar rootroot bformenc1 Signal Modifiers:Panning and Spatialization bformenc1 bformenc1 Codes a signal into the ambisonic B format. Description Codes a signal into the ambisonic B format Syntax aw, ax, ay, az bformenc1 asig, kalpha, kbeta aw, ax, ay, az, ar, as, at, au, av bformenc1 asig, kalpha, kbeta aw, ax, ay, az, ar, as, at, au, av, ak, al, am, an, ao, ap, aq bformenc1 \ asig, kalpha, kbeta aarray[] bformenc1 asig, kalpha, kbeta Performance aw, ax, ay, ... -- output cells of the B format. aarray -- output array to hold cells of the B format. asig -- input signal. kalpha –- azimuth angle in degrees (anticlockwise). kbeta -- altitude angle in degrees. Example Here is an example of the bformenc1 opcode. It uses the file bformenc1.csd. Example of the bformenc1 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also bformdec1 Credits Author: Richard Furse, Bruce Wiggins and Fons Adriaensen, following code by Samuel Groner 2008 New in version 5.09 Array output new in version 6.01 manual_src~dfsg/opcodes/apoisson.xml0000644000000000000000000000077612262561504016770 0ustar rootroot apoisson apoisson apoisson Deprecated. Description Deprecated as of version 3.49. Use the poisson opcode instead. manual_src~dfsg/opcodes/gauss.xml0000644000000000000000000001043512262561504016250 0ustar rootroot x-class noise generators gauss Signal Generators:Random (Noise) Generators gauss gauss Gaussian distribution random number generator. Description Gaussian distribution random number generator. This is an x-class noise generator. Syntax ares gauss krange ires gauss krange kres gauss krange Performance krange -- the range of the random numbers (-krange to +krange). Outputs both positive and negative numbers. gauss returns random numbers following a normal distribution centered around 0.0 (mu = 0.0) with a variance (sigma) of krange / 3.83. Thus more than 99.99% of the random values generated are in the range -krange to +krange. If a mean value different of 0.0 is desired, this mean value has to be added to the generated numbers (see example below). For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the gauss opcode. It uses the file gauss.csd. Example of the gauss opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: mean = 0.000, std = 0.260, samples inside the given range: 99.993% mean = 0.005, std = 0.999, samples inside the given range: 99.998% mean = 2.700, std = 1.497, samples inside the given range: 100.000% See Also seed, betarand, bexprnd, cauchy, exprand, linrand, pcauchy, poisson, trirand, unirand, weibull Credits Author: &nameparis; MIT, Cambridge 1995 Precisions about mu and sigma added by &namepinot; after a discussion with Joachim Heintz on the Csound List, December 2010. Example written by &namepinot;, adapted from a csd file by Joachim Heintz, December 2010. Existed in 3.30 manual_src~dfsg/opcodes/flvalue.xml0000644000000000000000000000764212262561504016572 0ustar rootroot FLvalue FLTK:Other FLvalue FLvalue Shows the current value of a FLTK valuator. Description FLvalue shows current the value of a valuator in a text field. Syntax ihandle FLvalue "label", iwidth, iheight, ix, iy Initialization ihandle -- handle value (an integer number) that unequivocally references the corresponding valuator. It can be used for the idisp argument of a valuator. label -- a double-quoted string containing some user-provided text, placed near the corresponding widget. iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). Performance FLvalue shows the current values of a valuator in a text field. It outputs ihandle that can then be used for the idisp argument of a valuator (see the FLTK Valuators section). In this way, the values of that valuator will be dynamically be shown in a text field. Note that FLvalue is not a valuator and its value cannot be modified.The value for an FLvalue widget should be set only by other widgets, and NOT from FLsetVal or FLsetVal_i since this can cause Csound to crash. Examples Here is an example of the FLvalue opcode. It uses the file FLvalue.csd. Example of the FLvalue opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLbox, FLbutBank, FLbutton, FLprintk, FLprintk2 Credits Author: &namegabriel; New in version 4.22 Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/strcmpk.xml0000644000000000000000000000215312262561504016607 0ustar rootroot strcmpk Strings:Manipulation strcmpk strcmp Compare strings Description Compare strings and set the result to -1, 0, or 1 if the first string is less than, equal to, or greater than the second, respectively. strcmpk does the comparison both at initialization and performance time. Syntax kres strcmpk S1, S2 See also strcmp Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/vbaplsinit.xml0000644000000000000000000001160212262561504017276 0ustar rootroot vbaplsinit Signal Modifiers:Panning and Spatialization vbaplsinit vbaplsinit Configures VBAP output according to loudspeaker parameters. Description Configures VBAP output according to loudspeaker parameters. Syntax vbaplsinit idim, ilsnum [, idir1] [, idir2] [...] [, idir32] Initialization idim -- dimensionality of loudspeaker array. Either 2 or 3. If the dimension has a fractional part then that is used as the index of the layout created (used in vbap, vbapz and vbapg only). The factional part should be between .00 and .99. ilsnum -- number of loudspeakers. In two dimensions, the number can vary from 2 to 64. In three dimensions, the number can vary from 3 and 64. idir1, idir2, ..., idir32 -- directions of loudspeakers. Number of directions must be less than or equal to 16. In two-dimensional loudspeaker positioning, idirn is the azimuth angle respective to nth channel. In three-dimensional loudspeaker positioning, fields are the azimuth and elevation angles of each loudspeaker consequently (azi1, ele1, azi2, ele2, etc.). In 2 dimensions the angle between two adjacent speakers must be less than 179 degrees (170 degrees in earlier versions). This is a restriction of the algorithm. Performance VBAP distributes the signal using loudspeaker data configured with vbaplsinit. The signal is applied to, at most, two loudspeakers in 2-D loudspeaker configurations, and three loudspeakers in 3-D loudspeaker configurations. If the virtual source is panned outside the region spanned by loudspeakers, the nearest loudspeakers are used in panning. Examples Here is an example of the vbaplsinit opcode. It uses the file vbaplsinit.csd. Example of the vbaplsinit opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See the other entries of the vbap opcodes for different examples. Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. See Also vbap, vbapg, vbap4, vbap4move, vbap8, vbap8move, vbap16, vbap16move, vbapz, vbapzmove Credits Author: Ville Pulkki Sibelius Academy Computer Music Studio Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Helsinki, Finland May 2000 New in Csound Version 4.07 Multiple layouts new ib version 5.17.14 manual_src~dfsg/opcodes/pdclip.xml0000644000000000000000000001253112262561504016400 0ustar rootroot pdclip Signal Modifiers:Waveshaping pdclip pdclip Performs linear clipping on an audio signal or a phasor. Description The pdclip opcode allows a percentage of the input range of a signal to be clipped to fullscale. It is similar to simply multiplying the signal and limiting the range of the result, but pdclip allows you to think about how much of the signal range is being distorted instead of the scalar factor and has a offset parameter for assymetric clipping of the signal range. pdclip is also useful for remapping phasors for phase distortion synthesis. Syntax aout pdclip ain, kWidth, kCenter [, ibipolar [, ifullscale]] Initialization ibipolar -- an optional parameter specifying either unipolar (0) or bipolar (1) mode. Defaults to unipolar mode. ifullscale -- an optional parameter specifying the range of input and output values. The maximum will be ifullscale. The minimum depends on the mode of operation: zero for unipolar or -ifullscale for bipolar. Defaults to 1.0 -- you should set this parameter to the maximum expected input value. Performance ain -- the input signal to be clipped. aout -- the output signal. kWidth -- the percentage of the signal range that is clipped (must be between 0 and 1). kCenter -- an offset for shifting the unclipped window of the signal higher or lower in the range (essentially a DC offset). Values should be in the range [-1, 1] with a value of zero representing no shift (regardless of whether bipolar or unipolar mode is used). The pdclip opcode performs linear clipping on the input signal ain. kWidth specifies the percentage of the signal range that is clipped. The rest of the input range is mapped linearly from zero to ifullscale in unipolar mode and from -ifullscale to ifullscale in bipolar mode. When kCenter is zero, equal amounts of the top and bottom of the signal range are clipped. A negative value shifts the unclipped range more towards the bottom of the input range and a positive value shifts it more towards the top. ibipolar should be 1 for bipolar operation and 0 for unipolar mode. The default is unipolar mode (ibipolar = 0). ifullscale sets the maximum amplitude of the input and output signals (defaults to 1.0). This amounts to waveshaping the input with the following transfer function (normalized to ifullscale=1.0 in bipolar mode): 1| _______ x-axis is input range, y-axis is output | / | / width of clipped region is 2*kWidth -1 |/ 1 width of unclipped region is 2*(1 - kWidth) -------------------- kCenter shifts the unclipped region /| left or right (up to kWidth) / | / | ------ |-1 Bipolar mode can be used for direct, linear distortion of an audio signal. Alternatively, unipolar mode is useful for modifying the output of a phasor before it is used to index a function table, effectively making this a phase distortion technique. See Also pdhalf, pdhalfy, limit, clip, distort1 Examples Here is an example of the pdclip opcode. It uses the file pdclip.csd. Example of the pdclip opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &nameanthony; January 2008 New in Csound version 5.08 manual_src~dfsg/opcodes/reinit.xml0000644000000000000000000000411412262561504016415 0ustar rootroot reinit Instrument Control:Initialization and Reinitialization reinit reinit Suspends a performance while a special initialization pass is executed. Description Suspends a performance while a special initialization pass is executed. Whenever this statement is encountered during a p-time pass, performance is temporarily suspended while a special Initialization pass, beginning at label and continuing to rireturn or endin, is executed. Performance will then be resumed from where it left off. Syntax reinit label Examples The following statements will generate an exponential control signal whose value moves from 440 to 880 exactly ten times over the duration p3. They use the file reinit.csd. Example of the reinit opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also rigoto, rireturn manual_src~dfsg/opcodes/pindex.xml0000644000000000000000000000752212262561504016420 0ustar rootroot pindex Instrument Control:Sensing and Control pindex pindex Returns the value of a specified pfield. Description pindex returns the value of a specified pfield. Syntax ivalue pindex ipfieldIndex Initialization ipfieldIndex - pfield number to query. ivalue - value of the pfield. Examples Here is an example of the pindex opcode. It uses the file pindex.csd. Example of the pindex opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example will produce the following output: new alloc for instr 1: WARNING: instr 1 uses 3 p-fields but is given 5 p1 = 1.000000 p2 = 0.000000 p3 = 3.000000 p4 = 40.000000 p5 = 50.000000 instr 1: inum = 5.000 B 0.000 .. 1.000 T 1.000 TT 1.000 M: 0.0 new alloc for instr 1: WARNING: instr 1 uses 3 p-fields but is given 5 p1 = 1.000000 p2 = 1.000000 p3 = 2.000000 p4 = 80.000000 p5 = 50.000000 instr 1: inum = 5.000 B 1.000 .. 2.000 T 2.000 TT 2.000 M: 0.0 new alloc for instr 1: WARNING: instr 1 uses 3 p-fields but is given 7 p1 = 1.000000 p2 = 2.000000 p3 = 1.000000 p4 = 40.000000 p5 = 50.000000 p6 = 60.000000 p7 = 70.000000 instr 1: inum = 7.000 The warnings can be ignored, because the pfields are used indirectly through pindex instead of explicitly through p4, p5, etc. Here is another example of the pindex opcode. It uses the file pindex-2.csd. Second example of the pindex opcode. The example will produce the following output: WARNING: instr 1 uses 3 p-fields but is given 8 B 0.000 .. 2.000 T 2.000 TT 2.000 M: 0.49966 0.49966 WARNING: instr 1 uses 3 p-fields but is given 10 B 2.000 .. 4.000 T 4.000 TT 4.000 M: 0.50000 0.50000 WARNING: instr 1 uses 3 p-fields but is given 10 B 4.000 .. 6.000 T 6.000 TT 6.000 M: 0.49943 0.49943 WARNING: instr 1 uses 3 p-fields but is given 6 B 6.000 .. 8.000 T 8.000 TT 8.000 M: 0.00000 0.00000 WARNING: instr 1 uses 3 p-fields but is given 20 B 8.000 .. 18.000 T 18.000 TT 18.000 M: 0.49994 0.49994 See Also pcount Credits Example by: Anthony Kozar Dec. 2006 manual_src~dfsg/opcodes/STKBlowHole.xml0000644000000000000000000000761112262561504017225 0ustar rootroot STKBlowHole Signal Generators:STK Opcodes STKBlowHole STKBlowHole STK clarinet physical model with one register hole and one tonehole. Description This opcode is based on the clarinet model, with the addition of a two-port register hole and a three-port dynamic tonehole implementation. In this implementation, the distances between the reed/register hole and tonehole/bell are fixed. As a result, both the tonehole and register hole will have variable influence on the playing frequency, which is dependent on the length of the air column. In addition, the highest playing freqeuency is limited by these fixed lengths. Syntax asignal STKBlowHole ifrequency, iamplitude, [kreed, kv1[, knoise, kv2[, khole, kv3[, kreg, kv4[, kbreath, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kreed -- controller 2, stiffness of reed. Value range of kv1 is 0-127. knoise -- controller 4, gain of noise. Value range of kv2 is 0-127. khole -- controller 11, state of tonehole. Value range of kv3 is 0-127. kreg -- controller 1, state of register. Value range of kv4 is 0-127. kbreath -- controller 128, breath pressure. Value range of kv5 is 0-127. Notes The code for this opcode is taken directly from the BlowHole class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKBlowHole opcode. It uses the file STKBlowHole.csd. Example of the STKBlowHole opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/rnd31.xml0000644000000000000000000001600112262561504016050 0ustar rootroot rnd31 Signal Generators:Random (Noise) Generators rnd31 rnd31 31-bit bipolar random opcodes with controllable distribution. Description 31-bit bipolar random opcodes with controllable distribution. These units are portable, i.e. using the same seed value will generate the same random sequence on all systems. The distribution of generated random numbers can be varied at k-rate. Syntax ax rnd31 kscl, krpow [, iseed] ix rnd31 iscl, irpow [, iseed] kx rnd31 kscl, krpow [, iseed] Initialization ix -- i-rate output value. iscl -- output scale. The generated random numbers are in the range -iscl to iscl. irpow -- controls the distribution of random numbers. If irpow is positive, the random distribution (x is in the range -1 to 1) is abs(x) ˆ ((1 / irpow) - 1); for negative irpow values, it is (1 - abs(x)) ˆ ((-1 / irpow) - 1). Setting irpow to -1, 0, or 1 will result in uniform distribution (this is also faster to calculate). [A graph of distributions for different values of irpow.] A graph of distributions for different values of irpow. iseed (optional, default=0) -- seed value for random number generator (positive integer in the range 1 to 2147483646 (2 ˆ 31 - 2)). Zero or negative value seeds from current time (this is also the default). Seeding from current time is guaranteed to generate different random sequences, even if multiple random opcodes are called in a very short time. In the a- and k-rate version the seed is set at opcode initialization. With i-rate output, if iseed is zero or negative, it will seed from current time in the first call, and return the next value from the random sequence in successive calls; positive seed values are set at all i-rate calls. The seed is local for a- and k-rate, and global for i-rate units. Notes although seed values up to 2147483646 are allowed, it is recommended to use smaller numbers (< 1000000) for portability, as large integers may be rounded to a different value if 32-bit floats are used. i-rate rnd31 with a positive seed will always produce the same output value (this is not a bug). To get different values, set seed to 0 in successive calls, which will return the next value from the random sequence. Performance ax -- a-rate output value. kx -- k-rate output value. kscl -- output scale. The generated random numbers are in the range -kscl to kscl. It is the same as iscl, but can be varied at k-rate. krpow -- controls the distribution of random numbers. It is the same as irpow, but can be varied at k-rate. Examples Here is an example of the rnd31 opcode at a-rate. It uses the file rnd31.csd. An example of the rnd31 opcode at a-rate. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an example of the rnd31 opcode at k-rate. It uses the file rnd31_krate.csd. An example of the rnd31 opcode at k-rate. Its output should include lines like this: k1=0.112106 k1=-0.274665 k1=0.403933 Here is an example of the rnd31 opcode that uses the number 7 as a seed value. It uses the file rnd31_seed7.csd. An example of the rnd31 opcode that uses the number 7 as a seed value. Its output should include lines like this: instr 1: i1 = -0.649 instr 1: i2 = -0.761 instr 1: i3 = 0.677 Here is an example of the rnd31 opcode that uses the current time as a seed value. It uses the file rnd31_time.csd. An example of the rnd31 opcode that uses the current time as a seed value. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: i1 = -0.691 instr 1: i2 = -0.686 instr 1: i3 = -0.358 Credits Author: &nameistvan; New in version 4.16 manual_src~dfsg/opcodes/minarray.xml0000644000000000000000000000422712262561504016752 0ustar rootroot minarray Array Opcodes minarray minarray returns the minimum value in an array. Description The minarray opcode returns the minimum value in a k-rate array. Syntax kmin minarray karray Performance kmin -- variable for result. karray -- array for reading. Examples Here is an example of the minarray opcode. It uses the file minarray.csd. Example of the minarray opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also maxarray, sumarray, scalearray, Credits Author: &namejohn; October 2011 New in Csound version 5.14 Revised in Csound version 6.00 to work on multidimensional arrays manual_src~dfsg/opcodes/fog.xml0000644000000000000000000001453112262561504015702 0ustar rootroot fog Signal Generators:Granular Synthesis fog fog Audio output is a succession of grains derived from data in a stored function table Description Audio output is a succession of grains derived from data in a stored function table ifna. The local envelope of these grains and their timing is based on the model of fof synthesis and permits detailed control of the granular synthesis. Syntax ares fog xamp, xdens, xtrans, aspd, koct, kband, kris, kdur, kdec, \ iolaps, ifna, ifnb, itotdur [, iphs] [, itmode] [, iskip] Initialization iolaps -- number of pre-located spaces needed to hold overlapping grain data. Overlaps are density dependent, and the space required depends on the maximum value of xdens * kdur. Can be over-estimated at no computation cost. Uses less than 50 bytes of memory per iolap. ifna, ifnb -- table numbers of two stored functions. The first is the data used for granulation, usually from a soundfile (GEN01). The second is a rise shape, used forwards and backwards to shape the grain rise and decay; this is normally a sigmoid (GEN19) but may be linear (GEN05). itotdur -- total time during which this fog will be active. Normally set to p3. No new grain is created if it cannot complete its kdur within the remaining itotdur. iphs (optional) -- initial phase of the fundamental, expressed as a fraction of a cycle (0 to 1). The default value is 0. itmode (optional) -- transposition mode. If zero, each grain keeps the xtrans value it was launched with. If non-zero, each is influenced by xtrans continuously. The default value is 0. iskip (optional, default=0) -- If non-zero, skip initialization (allows legato use). Performance xamp -- amplitude factor. Amplitude is also dependent on the number of overlapping grains, the interaction of the rise shape (ifnb) and the exponential decay (kband), and the scaling of the grain waveform (ifna). The actual amplitude may therefore exceed xamp. xdens -- density. The frequency of grains per second. xtrans -- transposition factor. The rate at which data from the stored function table ifna is read within each grain. This has the effect of transposing the original material. A value of 1 produces the original pitch. Higher values transpose upwards, lower values downwards. Negative values result in the function table being read backwards. aspd -- Starting index pointer. aspd is the normalized index (0 to 1) to table ifna that determines the movement of a pointer used as the starting point for reading data within each grain. (xtrans determines the rate at which data is read starting from this pointer.) koct -- octaviation index. The operation of this parameter is identical to that in fof. kband, kris, kdur, kdec -- grain envelope shape. These parameters determine the exponential decay (kband), and the rise (kris), overall duration (kdur,) and decay (kdec ) times of the grain envelope. Their operation is identical to that of the local envelope parameters in fof. Examples Here is an example of the fog opcode. It uses the file fog.csd. Example of the fog opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Michael Clark Huddersfield May 1997 New in version 3.46 The Csound fog generator is by Michael Clarke, extending his earlier work based on IRCAM's FOF algorithm. Added notes by &namerasmus; on September 2002. manual_src~dfsg/opcodes/pitch.xml0000644000000000000000000001077512262561504016244 0ustar rootroot pitch Instrument Control:Sensing and Control pitch pitch Tracks the pitch of a signal. Description Using the same techniques as spectrum and specptrk, pitch tracks the pitch of the signal in octave point decimal form, and amplitude in dB. Syntax koct, kamp pitch asig, iupdte, ilo, ihi, idbthresh [, ifrqs] [, iconf] \ [, istrt] [, iocts] [, iq] [, inptls] [, irolloff] [, iskip] Initialization iupdte -- length of period, in seconds, that outputs are updated ilo, ihi -- range in which pitch is detected, expressed in octave point decimal idbthresh -- amplitude, expressed in decibels, necessary for the pitch to be detected. Once started it continues until it is 6 dB down. ifrqs (optional) -- number of divisons of an octave. Default is 12 and is limited to 120. iconf (optional) -- the number of conformations needed for an octave jump. Default is 10. istrt (optional) -- starting pitch for tracker. Default value is (ilo + ihi)/2. iocts (optional) -- number of octave decimations in spectrum. Default is 6. iq (optional) -- Q of analysis filters. Default is 10. inptls (optional) -- number of harmonics, used in matching. Computation time increases with the number of harmonics. Default is 4. irolloff (optional) -- amplitude rolloff for the set of filters expressed as fraction per octave. Values must be positive. Default is 0.6. iskip (optional) -- if non-zero, skips initialization. Default is 0. Performance koct -- The pitch output, given in the octave point decimal format. kamp -- The amplitude output. pitch analyzes the input signal, asig, to give a pitch/amplitude pair of outputs, for the strongest frequency in the signal. The value is updated every iupdte seconds. The number of partials and rolloff fraction can effect the pitch tracking, so some experimentation may be necessary. Suggested values are 4 or 5 harmonics, with rolloff 0.6, up to 10 or 12 harmonics with rolloff 0.75 for complex timbres, with a weak fundamental. Examples Here is an example of the pitch opcode. It uses the file pitch.csd. Example of the pitch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; University of Bath, Codemist Ltd. Bath, UK April 1999 New in Csound version 3.54 manual_src~dfsg/opcodes/sfplay.xml0000644000000000000000000001357412262561504016433 0ustar rootroot sfplay Signal Generators:Sample Playback sfplay sfplay Plays a SoundFont2 (SF2) sample preset, generating a stereo sound. Description Plays a SoundFont2 (SF2) sample preset, generating a stereo sound. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. Syntax ar1, ar2 sfplay ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset] [, ienv] Initialization ivel -- velocity value inotenum -- MIDI note number value ipreindex -- preset index iflag (optional) -- flag regarding the behavior of xfreq and inotenum ioffset (optional) -- start playing at offset, in samples. ienv (optional) -- enables and determines amplitude envelope. 0 = no envelope, 1 = linear attack and decay, 2 = linear attack, exponential decay (see below). Default = 0. Performance xamp -- amplitude correction factor xfreq -- frequency value or frequency multiplier, depending by iflag. When iflag = 0, xfreq is a multiplier of a the default frequency, assigned by SF2 preset to the inotenum value. When iflag = 1, xfreq is the absolute frequency of the output sound, in Hz. Default is 0. When iflag = 0, inotenum sets the frequency of the output according to the MIDI note number used, and xfreq is used as a multiplier. When iflag = 1, the frequency of the output, is set directly by xfreq. This allows the user to use any kind of micro-tuning based scales. However, this method is designed to work correctly only with presets tuned to the default equal temperament. Attempts to use this method with a preset already having non-standard tunings, or with drum-kit-based presets, could give unexpected results. Adjustment of the amplitude can be done by varying the xamp argument, which acts as a multiplier. Notice that both xamp and xfreq can use k-rate as well as a-rate signals. Both arguments must use variables of the same rate, or sfplay will not work correctly. ipreindex must contain the number of a previously assigned preset, or Csound will crash. The ioffset parameter allows the sound to start from a sample different than the first one. The user should make sure that its value is within the length of the specific sound. Otherwise, Csound will probably crash. The ienv parameter enables and determines the type of amplitude envelope used. The default value is 0, or no envelope. If ienv is set to 1, the attack and decay portions are linear. If set to 2, the attack is linear and the decay is exponential. The release portion of the envelope has not yet been implemented. sfplay plays a preset, generating a stereo sound. ivel does not directly affect the amplitude of the output, but informs sfplay about which sample should be chosen in multi-sample, velocity-split presets. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. See Also sfilist, sfinstr, sfinstrm, sfload, sfpassign, sfplay3, sfplaym, sfplay3m, sfplist, sfpreset Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 New optional parameter ienv in version 5.09 manual_src~dfsg/opcodes/loop_le.xml0000644000000000000000000000744412262561504016565 0ustar rootroot loop_le Instrument Control:Program Flow Control loop_le loop_le Looping constructions. Description Construction of looping operations. Syntax loop_le indx, incr, imax, label loop_le kndx, kncr, kmax, label Initialization indx -- i-rate variable to count loop. incr -- value to increment the loop. imax -- maximum value of loop index. Performance kndx -- k-rate variable to count loop. kncr -- value to increment the loop. kmax -- maximum value of loop index. The actions of loop_le are equivalent to the code indx = indx + incr if (indx <= imax) igoto label or kndx = kndx + kncr if (kndx <= kmax) kgoto label Examples Here is an example of the loop_le opcode. It uses the file loop_le.csd. Example of the loop_le opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: play instance 0 play instance 1 play instance 2 play instance 3 play instance 4 play instance 5 See Also loop_ge, loop_gt and loop_lt. More information on this opcode: http://www.csounds.com/journal/2006summer/controlFlow_part2.html , written by Steven Yi. And in the Floss Manuals: http://en.flossmanuals.net/csound/ch018_c-control-structures Credits &nameistvan;. New in Csound version 5.01 manual_src~dfsg/opcodes/is16b14.xml0000644000000000000000000000076712262561504016226 0ustar rootroot is16b14 is16b14 is16b14 Deprecated. Description Deprecated as of version 3.52. Use the s16b14 opcode instead. manual_src~dfsg/opcodes/slider16.xml0000644000000000000000000001131012262561504016550 0ustar rootroot slider16 Real-time MIDI:Slider Banks slider16 slider16 Creates a bank of 16 different MIDI control message numbers. Description Creates a bank of 16 different MIDI control message numbers. Syntax i1,...,i16 slider16 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum16, imin16, imax16, init16, ifn16 k1,...,k16 slider16 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum16, imin16, imax16, init16, ifn16 Initialization i1 ... i16 -- output values ichan -- MIDI channel (1-16) ictlnum1 ... ictlnum16 -- MIDI control number (0-127) imin1 ... imin16 -- minimum values for each controller imax1 ... imax16 -- maximum values for each controller init1 ... init16 -- initial value for each controller ifn1 ... ifn16 -- function table for conversion for each controller Performance k1 ... k16 -- output values slider16 is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider16 allows a bank of 16 different MIDI control message numbers. As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. In the i-rate version of slider16, there is not an initial value input argument, because the output is gotten directly from current status of internal controller array of Csound. See Also s16b14, s32b14, slider16f, slider32, slider32f, slider64, slider64f, slider8, slider8f Credits Author: &namegabriel; Italy December 1998 New in Csound version 3.50 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/STKWhistle.xml0000644000000000000000000000675512262561504017141 0ustar rootroot STKWhistle Signal Generators:STK Opcodes STKWhistle STKWhistle STKWhistle produces whistle sounds. Description STKWhistle produces (police) whistle sounds. It uses a hybrid physical/spectral model of a police whistle (a la Cook). Syntax asignal STKWhistle ifrequency, iamplitude, [kmod, kv1[, knoise, kv2[, kfipfreq, kv3[, kfipgain, kv4[, kvol, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kmod -- controller 2, blowing frequency modulation. Value range of kv1 is 0-127. knoise -- controller 4, noise gain. Value range of kv2 is 0-127. kfipfreq -- controller 11, fipple modulation frequency. Value range of kv3 is 0-127. kfipgain -- controller 1, fipple modulation gain. Value range of kv4 is 0-127. kvol -- controller 128, volume. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the Whistle class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKWhistle opcode. It uses the file STKWhistle.csd. Example of the STKWhistle opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/wguide2.xml0000644000000000000000000000670512262561504016501 0ustar rootroot wguide2 Signal Modifiers:Waveguides wguide2 wguide2 A model of beaten plate consisting of two parallel delay-lines and two first-order lowpass filters. Description A model of beaten plate consisting of two parallel delay-lines and two first-order lowpass filters. Syntax ares wguide2 asig, xfreq1, xfreq2, kcutoff1, kcutoff2, \ kfeedback1, kfeedback2 Performance asig -- the input of excitation noise xfreq1, xfreq2 -- the frequency (i.e. the inverse of delay time) Changed to x-rate in Csound version 3.59. kcutoff1, kcutoff2 -- the filter cutoff frequency in Hz. kfeedback1, kfeedback2 -- the feedback factor wguide2 is a model of beaten plate consisting of two parallel delay-lines and two first-order lowpass filters. The two feedback lines are mixed and sent to the delay again each cycle. Implementing waveguide algorithms as opcodes, instead of orc instruments, allows the user to set kr different than sr, allowing better performance particulary when using real-time. wguide2. wguide2. As a rule of thumb, to avoid making wguide2 unstable, the sum of the two feedback values should be below 0.5. Examples Here is an example of the wguide2 opcode. It uses the file wguide2.csd. Example of the wguide2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also wguide1 Credits Author: &namegabriel; Italy October 1998 New in Csound version 3.49 manual_src~dfsg/opcodes/zkr.xml0000644000000000000000000000474312262561504015741 0ustar rootroot zkr Zak Patch System zkr zkr Reads from a location in zk space at k-rate. Description Reads from a location in zk space at k-rate. Syntax kres zkr kndx Initialization kndx -- points to the zk location to be read. Performance zkr reads the array of floats at kndx in zk space. Examples Here is an example of the zkr opcode. It uses the file zkr.csd. Example of the zkr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zar, zarg, zir, zkcl, zkmod, zkwm, zkw Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/monitor.xml0000644000000000000000000000411512262561504016613 0ustar rootroot monitor Signal I/O:Signal Output monitor monitor Returns the audio spout frame. Description Returns the audio spout frame (if active), otherwise it returns zero. Syntax aout1 [,aout2 ... aoutX] monitor Performance This opcode can be used for monitoring the output signal from csound. It should not be used for processing the signal further. See the entry for the fout opcode for an example of usage of monitor. Examples Here is an example of the monitor opcode. It uses the file monitor.csd. Example of the monitor opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also fout, the Mixer opcodes and the Zak Patching System. Credits Istvan Varga 2006 manual_src~dfsg/opcodes/sandpaper.xml0000644000000000000000000000736412262561504017112 0ustar rootroot sandpaper Signal Generators:Models and Emulations sandpaper sandpaper Semi-physical model of a sandpaper sound. Description sandpaper is a semi-physical model of a sandpaper sound. It is one of the PhISEM percussion opcodes. PhISEM (Physically Informed Stochastic Event Modeling) is an algorithmic approach for simulating collisions of multiple independent sound producing objects. Syntax ares sandpaper iamp, idettack [, inum] [, idamp] [, imaxshake] Initialization iamp -- Amplitude of output. Note: As these instruments are stochastic, this is only a approximation. idettack -- period of time over which all sound is stopped inum (optional) -- The number of beads, teeth, bells, timbrels, etc. If zero, the default value is 128. idamp (optional) -- the damping factor, as part of this equation: damping_amount = 0.998 + (idamp * 0.002) The default damping_amount is 0.999 which means that the default value of idamp is 0.5. The maximum damping_amount is 1.0 (no damping). This means the maximum value for idamp is 1.0. The recommended range for idamp is usually below 75% of the maximum value. imaxshake (optional) -- amount of energy to add back into the system. The value should be in range 0 to 1. Examples Here is an example of the sandpaper opcode. It uses the file sandpaper.csd. Example of the sandpaper opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cabasa, crunch, sekere, stix Credits Author: &nameperry;, part of the PhOLIES (Physically-Oriented Library of Imitated Environmental Sounds) Adapted by &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.07 Added notes by &namerasmus; on May 2002. manual_src~dfsg/opcodes/nstrnum.xml0000644000000000000000000000256512262561504016641 0ustar rootroot nstrnum Instrument Control:Initialization and Reinitialization nstrnum nstrnum Returns the number of a named instrument. Description Returns the number of a named instrument. Syntax insno nstrnum "name" Initialization insno -- the instrument number of the named instrument. Performance "name" -- the named instrument's name. If an instrument with the specified name does not exist, an init error occurs, and -1 is returned. Credits Author: &nameistvan; New in version 4.23 Written in the year 2002. manual_src~dfsg/opcodes/outs.xml0000644000000000000000000000613712262561504016124 0ustar rootroot outs Signal I/O:Signal Output outs outs Writes stereo audio data to an external device or stream. Description Writes stereo audio data to an external device or stream. Syntax outs asig1, asig2 Performance Sends stereo audio samples to an accumulating output buffer (created at the beginning of performance) which serves to collect the output of all active instruments before the sound is written to disk. There can be any number of these output units in an instrument. The type (mono, stereo, quad, hex, or oct) should agree with nchnls. But as of version 3.50, Csound will attempt to change an incorrect opcode to agree with the nchnls statement. Opcodes can be chosen to direct sound to any particular channel: outs1 sends to stereo channel 1, outq3 to quad channel 3, etc. Examples Here is an example of the outs opcode. It uses the file outs.csd. Example of the outs opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also out, outh, outo, outq, outq1, outq2, outq3, outq4, outs1, outs2, soundout Credits Author: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 manual_src~dfsg/opcodes/flbutbank.xml0000644000000000000000000001545612262561504017106 0ustar rootroot FLbutBank FLTK:Other FLbutBank FLbutBank A FLTK widget opcode that creates a bank of buttons. Description A FLTK widget opcode that creates a bank of buttons. Syntax kout, ihandle FLbutBank itype, inumx, inumy, iwidth, iheight, ix, iy, \ iopcode [, kp1] [, kp2] [, kp3] [, kp4] [, kp5] [....] [, kpN] Initialization ihandle -- a handle value (an integer number) that unequivocally references a corresponding widget. This is used by other opcodes that modify a widget's properties (see Modifying FLTK Widget Appearance). It is automatically output by FLbutBank and must not be set by the user label. (The user label is a double-quoted string containing some user-provided text placed near the widget.) itype -- an integer number denoting the appearance of the widget. The valid numbers are: 1 - normal button 2 - light button 3 - check button 4 - round button You can add 20 to the value to create a "plastic" type button. (Note that there is no Platic Round button. i.e. if you set type to 24 it will look exactly like type 23). inumx -- number of buttons in each row of the bank. inumy -- number of buttons in each column of the bank ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window, expressed in pixels iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window, expressed in pixels iopcode -- score opcode type. You have to provide the ascii code of the letter corresponding to the score opcode. At present time only i (ascii code 105) score statements are supported. A zero value refers to a default value of i. So both 0 and 105 activates the i opcode. A value of -1 disables this opcode feature. Performance kout -- output value kp1, kp2, ..., kpN -- arguments of the activated instruments. The FLbutBank opcode creates a bank of buttons. For example, the following line: gkButton,ihb1 FLbutBank 22, 8, 8, 380, 180, 50, 350, 0, 7, 0, 0, 5000, 6000 will create the this bank: FLbutBank. FLbutBank. A click to a button checks that button. It may also uncheck a previous checked button belonging to the same bank. So the behaviour is always that of radio-buttons. Notice that each button is labeled with a progressive number. The kout argument is filled with that number when corresponding button is checked. FLbutBank not only outputs a value but can also activate (or schedule) an instrument provided by the user each time a button is pressed. If the iopcode argument is set to a negative number, no instrument is activated so this feature is optional. In order to activate an instrument, iopcode must be set to 0 or to 105 (the ascii code of character i, referring to the i score opcode). P-fields of the activated instrument are kp1 (instrument number), kp2 (action time), kp3 (duration) and so on with user p-fields. The itype argument sets the type of buttons identically to the FLbutton opcode. By adding 10 to the itype argument (i.e. by setting 11 for type 1, 12 for type 2, 13 for type 3 and 14 for type 4), it is possible to skip the current FLbutBank value when getting/setting snapshots (see General FLTK Widget-related Opcodes). You can also add 10 to "plastic" button types (31 for type 1, 32 for type 2, etc.) FLbutBank is very useful to retrieve snapshots. Examples Here is an example of the FLbutBank opcode. It uses the file FLbutBank.csd. Example of the FLbutBank opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLbox, FLbutton, FLprintk, FLprintk2, FLvalue Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/vpvoc.xml0000644000000000000000000001432012262561504016260 0ustar rootroot vpvoc Spectral Processing:STFT vpvoc vpvoc Implements signal reconstruction using an fft-based phase vocoder and an extra envelope. Description Implements signal reconstruction using an fft-based phase vocoder and an extra envelope. Syntax ares vpvoc ktimpnt, kfmod, ifile [, ispecwp] [, ifn] Initialization ifile -- the pvoc number (n in pvoc.n) or the name in quotes of the analysis file made using pvanal. (See pvoc.) ispecwp (optional, default=0) -- if non-zero, attempts to preserve the spectral envelope while its frequency content is varied by kfmod. The default value is zero. ifn (optional, default=0) -- optional function table containing control information for vpvoc. If ifn = 0, control is derived internally from a previous tableseg or tablexseg unit. Default is 0. (New in Csound version 3.59) Performance ktimpnt -- The passage of time, in seconds, through the analysis file. ktimpnt must always be positive, but can move forwards or backwards in time, be stationary or discontinuous, as a pointer into the analysis file. kfmod -- a control-rate transposition factor: a value of 1 incurs no transposition, 1.5 transposes up a perfect fifth, and .5 down an octave. This implementation of pvoc was orignally written by Dan Ellis. It is based in part on the system of Mark Dolson, but the pre-analysis concept is new. The spectral extraction and amplitude gating (new in Csound version 3.56) were added by Richard Karpen based on functions in SoundHack by Tom Erbe. vpvoc is identical to pvoc except that it takes the result of a previous tableseg or tablexseg and uses the resulting function table (passed internally to the vpvoc), as an envelope over the magnitudes of the analysis data channels. Optionally, a table specified by ifn may be used. The result is spectral enveloping. The function size used in the tableseg should be framesize/2, where framesize is the number of bins in the phase vocoder analysis file that is being used by the vpvoc. Each location in the table will be used to scale a single analysis bin. By using different functions for ifn1, ifn2, etc.. in the tableseg, the spectral envelope becomes a dynamically changing one. See also tableseg and tablexseg. Examples The following example, using vpvoc, shows the use of functions such as f 1 0 256 5 .001 128 1 128 .001 f 2 0 256 5 1 128 .001 128 1 f 3 0 256 7 1 256 1 to scale the amplitudes of the separate analysis bins. ktime line 0, p3, 3 ; time pointer, in seconds, into file tablexseg 1, p3*0.5, 2, p3*0.5, 3 apv vpvoc ktime, 1, "pvoc.file" The result would be a time-varying spectral envelope applied to the phase vocoder analysis data. Since this amplifies or attenuates the amount of signal at the frequencies that are paired with the amplitudes which are scaled by these functions, it has the effect of applying very accurate filters to the signal. In this example the first table would have the effect of a band-pass filter, gradually be band-rejected over half the note's duration, and then go towards no modification of the magnitudes over the second half. Here is a complete example of the vpvoc opcode. It uses the file vpvoc.csd. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvoc Credits Authors: Dan Ellis and Richard Karpen Seattle, WA USA 1997 New in version 3.44 manual_src~dfsg/opcodes/tablera.xml0000644000000000000000000002713012262561504016540 0ustar rootroot tablera Table Control:Read/Write Opreations tablera tablera Reads tables in sequential locations. Description These opcode reads tables in sequential locations to an a-rate variable. Some thought is required before using it. It has at least two major, and quite different, applications which are discussed below. Syntax ares tablera kfn, kstart, koff Performance ares -- a-rate destination for reading ksmps values from a table. kfn -- i- or k-rate number of the table to read or write. kstart -- Where in table to read or write. koff -- i- or k-rate offset into table. Range unlimited - see explanation at end of this section. In one application, tablera is intended to be used in pair with tablewa, or with several tablera opcodes before a tablewa -- all sharing the same kstart variable. These read from and write to sequential locations in a table at audio rates, with ksmps floats being written and read each cycle. tablera starts reading from location kstart. tablewa starts writing to location kstart, and then writes to kstart with the number of the location one more than the one it last wrote. (Note that for tablewa, kstart is both an input and output variable.) If the writing index reaches the end of the table, then no further writing occurs and zero is written to kstart. For instance, if the table's length was 16 (locations 0 to 15), and ksmps was 5. Then the following steps would occur with repetitive runs of the tablewa opcode, assuming that kstart started at 0. Run Number Initial kstart Final kstart Locations Written 1 0 5 0 1 2 3 4 2 5 10 5 6 7 8 9 3 10 15 10 11 12 13 14 4 15 0 15 This is to facilitate processing table data using standard a-rate orchestra code between the tablera and tablewaopcodes. They allow all Csound k-rate operators to be used (with caution) on a-rate variables - something that would only be possible otherwise by ksmps = 1, downsamp and upsamp. Several cautions The k-rate code in the processing loop is really running at a-rate, so time dependent functions like port and oscil work faster than normal - their code is expecting to be running at k-rate. This system will produce undesirable results unless the ksmps fits within the table length. For instance a table of length 16 will accommodate 1 to 16 samples, so this example will work with ksmps = 1 to 16. Both these opcodes generate an error and deactivate the instrument if a table with length < ksmps is selected. Likewise an error occurs if kstart is below 0 or greater than the highest entry in the table - if kstart = table length. kstart is intended to contain integer values between 0 and (table length - 1). Fractional values above this should not affect operation but do not achieve anything useful. These opcodes are not interpolating, and the kstart and koff parameters always have a range of 0 to (table length - 1) - not 0 to 1 as is available in other table read/write opcodes. koff can be outside this range but it is wrapped around by the final AND operation. These opcodes are permanently in wrap mode. When koff is 0, no wrapping needs to occur, since the kstart++ index will always be within the table's normal range. koff not equal to 0 can lead to wrapping. The offset does not affect the number of read/write cycles performed, or the value written to kstart by tablewa. These opcodes cannot read or write the guardpoint. Use tablegpw to write the guardpoint after manipulations have been done with tablewa. Examples kstart = 0 lab1: atemp tablera ktabsource, kstart, 0 ; Read 5 values from table into an ; a-rate variable. atemp = log(atemp) ; Process the values using a-rate ; code. kstart tablewa ktabdest, atemp, 0 ; Write it back to the table if ktemp 0 goto lab1 ; Loop until all table locations ; have been processed. The above example shows a processing loop, which runs every k-cycle, reading each location in the table ktabsource, and writing the log of those values into the same locations of table ktabdest. This enables whole tables, parts of tables (with offsets and different control loops) and data from several tables at once to be manipulated with a-rate code and written back to another (or to the same) table. This is a bit of a fudge, but it is faster than doing it with k-rate table read and write code. Another application is: kzero = 0 kloop = 0 kzero tablewa 23, asignal, 0 ; ksmps a-rate samples written ; into locations 0 to (ksmps -1) of table 23. lab1: ktemp table kloop, 23 ; Start a loop which runs ksmps times, ; in which each cycle processes one of [ Some code to manipulate ] ; table 23's values with k-rate orchestra [ the value of ktemp. ] ; code. tablew ktemp, kloop, 23 ; Write the processed value to the table. kloop = kloop + 1 ; Increment the kloop, which is both the ; pointer into the table and the loop if kloop < ksmps goto lab1 ; counter. Keep looping until all values ; in the table have been processed. asignal tablera 23, 0, 0 ; Copy the table contents back ; to an a-rate variable. koff -- This is an offset which is added to the sum of kstart and the internal index variable which steps through the table. The result is then ANDed with the lengthmask (000 0111 for a table of length 8 - or 9 with guardpoint) and that final index is used to read or write to the table. koff can be any value. It is converted into a long using the ANSI floor() function so that -4.3 becomes -5. This is what we would want when using offsets which range above and below zero. Ideally this would be an optional variable, defaulting to 0, however with the existing Csound orchestra read code, such default parameters must be init time only. We want k-rate here, so we cannot have a default. See Also tablewa manual_src~dfsg/opcodes/opand.xml0000644000000000000000000000675012262561504016234 0ustar rootroot && Mathematical Operations:Arithmetic and Logic Operations && && Logical AND operator. Description Arithmetic operators perform operations of change-sign (negate), don't-change-sign, logical AND logical OR, add, subtract, multiply and divide. Note that a value or an expression may fall between two of these operators, either of which could take it as its left or right argument, as in a + b * c. In such cases three rules apply: 1. * and / bind to their neighbors more strongly than + and −. Thus the above expression is taken as a + (b * c) with * taking b and c and then + taking a and b * c. 2. + and − bind more strongly than &&, which in turn is stronger than ||: a && b - c || d is taken as (a && (b - c)) || d 3. When both operators bind equally strongly, the operations are done left to right: a - b - c is taken as (a - b) - c Parentheses may be used as above to force particular groupings. Syntax a && b (logical AND; not audio-rate) where the arguments a and b may be further expressions. Examples Here is an example of the Logical AND operator. It uses the file opand.csd. Example of the opand opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i1 0.00000 i1 3.00000 i1 1.00000 i1 3.00000 i1 1.00000 i1 2.00000 i1 3.00000 ..... See Also -, +, ||, *, /, ˆ, % manual_src~dfsg/opcodes/foscili.xml0000644000000000000000000000670512262561504016563 0ustar rootroot foscili Signal Generators:FM Synthesis foscili foscili Basic frequency modulated oscillator with linear interpolation. Description Basic frequency modulated oscillator with linear interpolation. Syntax ares foscili xamp, kcps, xcar, xmod, kndx, ifn [, iphs] Initialization ifn -- function table number. Requires a wrap-around guard point. iphs (optional, default=0) -- initial phase of waveform in table ifn, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is 0. Performance xamp -- the amplitude of the output signal. kcps -- a common denominator, in cycles per second, for the carrier and modulating frequencies. xcar -- a factor that, when multiplied by the kcps parameter, gives the carrier frequency. xmod -- a factor that, when multiplied by the kcps parameter, gives the modulating frequency. kndx -- the modulation index. foscili differs from foscil in that the standard procedure of using a truncated phase as a sampling index is here replaced by a process that interpolates between two successive lookups. Interpolating generators will produce a noticeably cleaner output signal, but they may take as much as twice as long to run. Adequate accuracy can also be gained without the time cost of interpolation by using large stored function tables of 2K, 4K or 8K points if the space is available. Examples Here is an example of the foscili opcode. It uses the file foscili.csd. Example of the foscili opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also More information about frequency modulation on Wikipedia: http://en.wikipedia.org/wiki/Frequency_modulation_synthesis manual_src~dfsg/opcodes/trscale.xml0000644000000000000000000000462412262561504016566 0ustar rootroot trscale Spectral Processing:Streaming trscale trscale Streaming partial track frequency scaling. Description The trscale opcode takes an input containg a TRACKS pv streaming signal (as generated, for instance by partials) and scales all frequencies by a k-rate amount. It can also, optionally, scale the gain of the signal by a k-rate amount (default 1). The result is pitch shifting of the input tracks. Syntax fsig trscale fin, kpitch[, kgain] Performance fsig -- output pv stream in TRACKS format fin -- input pv stream in TRACKS format kpitch -- frequency scaling kgain -- amplitude scaling (default 1) Examples Here is an example of the trscale opcode. It uses the file trscale.csd. Example of the trscale opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of an ifd-analysis signal and linear additive resynthesis with pitch shifting. Credits Author: &namevictor; February 2006 New in Csound 5.01 manual_src~dfsg/opcodes/pitchamdf.xml0000644000000000000000000001163712262561504017072 0ustar rootroot pitchamdf Instrument Control:Sensing and Control pitchamdf pitchamdf Follows the pitch of a signal based on the AMDF method. Description Follows the pitch of a signal based on the AMDF method (Average Magnitude Difference Function). Outputs pitch and amplitude tracking signals. The method is quite fast and should run in realtime. This technique usually works best for monophonic signals. Syntax kcps, krms pitchamdf asig, imincps, imaxcps [, icps] [, imedi] \ [, idowns] [, iexcps] [, irmsmedi] Initialization imincps -- estimated minimum frequency (expressed in Hz) present in the signal imaxcps -- estimated maximum frequency present in the signal icps (optional, default=0) -- estimated initial frequency of the signal. If 0, icps = (imincps+imaxcps) / 2. The default is 0. imedi (optional, default=1) -- size of median filter applied to the output kcps. The size of the filter will be imedi*2+1. If 0, no median filtering will be applied. The default is 1. idowns (optional, default=1) -- downsampling factor for asig. Must be an integer. A factor of idowns > 1 results in faster performance, but may result in worse pitch detection. Useful range is 1 - 4. The default is 1. iexcps (optional, default=0) -- how frequently pitch analysis is executed, expressed in Hz. If 0, iexcps is set to imincps. This is usually reasonable, but experimentation with other values may lead to better results. Default is 0. irmsmedi (optional, default=0) -- size of median filter applied to the output krms. The size of the filter will be irmsmedi*2+1. If 0, no median filtering will be applied. The default is 0. Performance kcps -- pitch tracking output krms -- amplitude tracking output pitchamdf usually works best for monophonic signals, and is quite reliable if appropriate initial values are chosen. Setting imincps and imaxcps as narrow as possible to the range of the signal's pitch, results in better detection and performance. Because this process can only detect pitch after an initial delay, setting icps close to the signal's real initial pitch prevents spurious data at the beginning. The median filter prevents kcps from jumping. Experiment to determine the optimum value for imedi for a given signal. Other initial values can usually be left at the default settings. Lowpass filtering of asig before passing it to pitchamdf, can improve performance, especially with complex waveforms. Examples Here is an example of the pitchamdf opcode. It uses the file pitchamdf.csd. Example of the pitchamdf opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Peter Neubäcker Munich, Germany August 1999 New in Csound version 3.59 manual_src~dfsg/opcodes/madsr.xml0000644000000000000000000001405512262561504016236 0ustar rootroot madsr Signal Generators:Envelope Generators madsr madsr Calculates the classical ADSR envelope using the linsegr mechanism. Description Calculates the classical ADSR envelope using the linsegr mechanism. Syntax ares madsr iatt, idec, islev, irel [, idel] [, ireltim] kres madsr iatt, idec, islev, irel [, idel] [, ireltim] Initialization iatt -- duration of attack phase idec -- duration of decay islev -- level for sustain phase irel -- duration of release phase. idel -- period of zero before the envelope starts ireltim (optional, default=-1) -- Control release time after receiving a MIDI noteoff event. If less than zero, the longest release time given in the current instrument is used. If zero or more, the given value will be used for release time. Its default value is -1. (New in Csound 3.59 - not yet properly tested) Please note that the release time cannot be longer than 32767/kr seconds. Performance The envelope is in the range 0 to 1 and may need to be scaled further. The envelope may be described as: Picture of an ADSR envelope. Picture of an ADSR envelope. The length of the sustain is calculated from the length of the note. This means adsr is not suitable for use with MIDI events. The opcode madsr uses the linsegr mechanism, and so can be used in MIDI applications. You can use other pre-made envelopes which start a release segment upon recieving a note off message, like linsegr and expsegr, or you can construct more complex envelopes using xtratim and release. Note that you don't need to use xtratim if you are using madsr, since the time is extended automatically. Times for iatt, idec and irel cannot be 0. If 0 is used, no envelope is generated. Use a very small value like 0.0001 if you need an instantaneous attack, decay or release. Examples Here is an example of the madsr opcode. It uses the file madsr.csd. Example of the madsr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the madsr opcode, using midi input. It uses the file madsr-2.csd. second example of the madsr opcode. See Also linsegr, expsegr, envlpxr, mxadsr, madsr, xadsr expon, expseg, expsega line, linseg, xtratim Credits Author: &namejohn; November 2002. Thanks to &namerasmus;, added documentation for the ireltim parameter. December 2002. Thanks to Iain McCurdy, added an example. December 2002. Thanks to &nameistvan;, added documentation about the maximum release time. New in Csound version 3.49. manual_src~dfsg/opcodes/cpsmidib.xml0000644000000000000000000000620712262561504016722 0ustar rootroot cpsmidib Real-time MIDI:Converters cpsmidib cpsmidib Get the note number of the current MIDI event and modify it by the current pitch-bend value, express it in cycles-per-second. Description Get the note number of the current MIDI event and modify it by the current pitch-bend value, express it in cycles-per-second. Syntax icps cpsmidib [irange] kcps cpsmidib [irange] Initialization irange (optional) -- the pitch bend range in semitones. Performance Get the note number of the current MIDI event, modify it by the current pitch-bend value, and express the result in cycles-per-second units. Available as an i-time value or as a continuous k-rate value. Examples Here is an example of the cpsmidib opcode. It uses the file cpsmidib.csd. Example of the cpsmidib opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also aftouch, ampmidi, cpsmidi, midictrl, notnum, octmidi, octmidib, pchbend, pchmidi, pchmidib, veloc Credits Author: &namebarry; - &namemike; MIT - Mills May 1997 manual_src~dfsg/opcodes/lpfreson.xml0000644000000000000000000001150512262561504016755 0ustar rootroot lpfreson Spectral Processing:LPC lpfreson lpfreson Resynthesises a signal from the data passed internally by a previous lpread, applying formant shifting. Description Resynthesises a signal from the data passed internally by a previous lpread, applying formant shifting. Syntax ares lpfreson asig, kfrqratio Performance asig -- an audio driving function for resynthesis. kfrqratio -- frequency ratio. Must be greater than 0. lpfreson receives values internally produced by a leading lpread.lpread gets its values from the control file according to the input value ktimpnt (in seconds). If ktimpnt proceeds at the analysis rate, time-normal synthesis will result; proceeding at a faster, slower, or variable rate will result in time-warped synthesis. At each k-period, lpread interpolates between adjacent frames to more accurately determine the parameter values (presented as output) and the filter coefficient settings (passed internally to a subsequent lpreson). The error signal kerr (between 0 and 1) derived during predictive analysis reflects the deterministic/random nature of the analyzed source. This will emerge low for pitched (periodic) material and higher for noisy material. The transition from voiced to unvoiced speech, for example, produces an error signal value of about .3. During synthesis, the error signal value can be used to determine the nature of the lpreson driving function: for example, by arbitrating between pitched and non-pitched input, or even by determining a mix of the two. In normal speech resynthesis, the pitched input to lpreson is a wideband periodic signal or pulse train derived from a unit such as buzz, and the nonpitched source is usually derived from rand. However, any audio signal can be used as the driving function, the only assumption of the analysis being that it has a flat response. lpfreson is a formant shifted lpreson, in which kfrqratio is the (cps) ratio of shifted to original formant positions. This permits synthesis in which the source object changes its apparent acoustic size. lpfreson with kfrqratio = 1 is equivalent to lpreson. Generally, lpreson provides a means whereby the time-varying content and spectral shaping of a composite audio signal can be controlled by the dynamic spectral content of another. There can be any number of lpread/lpreson (or lpfreson) pairs in an instrument or in an orchestra; they can read from the same or different control files independently. Examples Here is an example of the lpfreson opcode. It uses the file lpfreson.csd. Example of the lpfreson opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The audio file fox.wav is 2.8 seconds long. So filelen's output should be a line like this: fox.wav = 2.756667 seconds See Also lpread, lpreson manual_src~dfsg/opcodes/ctrlinit.xml0000644000000000000000000000350212262561504016753 0ustar rootroot ctrlinit Real-time MIDI:Input ctrlinit ctrlinit Sets the initial values for a set of MIDI controllers. Description Sets the initial values for a set of MIDI controllers. Syntax ctrlinit ichnl, ictlno1, ival1 [, ictlno2] [, ival2] [, ictlno3] \ [, ival3] [,...ival32] Initialization ichnl -- MIDI channel number (1-16) ictlno1, ictlno1, etc. -- MIDI controller numbers (0-127) ival1, ival2, etc. -- initial value for corresponding MIDI controller number Performance Sets the initial values for a set of MIDI controllers. See Also massign Credits Author: &namebarry; - &namemike; MIT, Cambridge, Mass. New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/tablexseg.xml0000644000000000000000000000601412262561504017102 0ustar rootroot tablexseg Spectral Processing:STFT tablexseg tablexseg Creates a new function table by making exponential segments between values in stored function tables. Description tablexseg is like expseg but interpolate between values in a stored function tables. The result is a new function table passed internally to any following vpvoc which occurs before a subsequent tablexseg (much like lpread/lpreson pairs work). The uses of these are described below under vpvoc. Syntax tablexseg ifn1, idur1, ifn2 [, idur2] [, ifn3] [...] Initialization ifn1, ifn2, ifn3, etc. -- function table numbers. ifn1, ifn2, and so on, must be the same size. idur1, idur2, etc. -- durations during which interpolation from one table to the next will take place. Examples Here is an example of the tablexseg opcode. It uses the file tablexseg.csd. Example of the tablexseg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvbufread, pvcross, pvinterp, pvread, tableseg Credits Author: Richard Karpen Seattle, WA USA 1997 manual_src~dfsg/opcodes/kread2.xml0000644000000000000000000000076212262561504016300 0ustar rootroot kread2 kread2 kread2 Deprecated. Description Deprecated as of version 3.52. Use the readk2 opcode instead. manual_src~dfsg/opcodes/tone.xml0000644000000000000000000000744412262561504016101 0ustar rootroot tone Signal Modifiers:Standard Filters tone tone A first-order recursive low-pass filter with variable frequency response. Description A first-order recursive low-pass filter with variable frequency response. tone is a 1 term IIR filter. Its formula is: yn = c1 * xn + c2 * yn-1 where b = 2 - cos(2 π hp/sr); c2 = b - sqrt(b2 - 1.0) c1 = 1 - c2 Syntax ares tone asig, khp [, iskip] Initialization iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance ares -- the output audio signal. asig -- the input audio signal. khp -- the response curve's half-power point, in Hertz. Half power is defined as peak power / root 2. tone implements a first-order recursive low-pass filter in which the variable khp (in Hz) determines the response curve's half-power point. Half power is defined as peak power / root 2. Examples Here is an example of the tone opcode. It uses the file tone.csd. Example of the tone opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also areson, aresonk, atone, atonek, port, portk, reson, resonk, tonek manual_src~dfsg/opcodes/vrandi.xml0000644000000000000000000001004512262561504016406 0ustar rootroot vrandi Vectorial:Random vrandi vrandi Generate a sort of 'vectorial band-limited noise' Description Generate a sort of 'vectorial band-limited noise' Syntax vrandi ifn, krange, kcps, ielements [, idstoffset] [, iseed] \ [, isize] [, ioffset] Initialization ifn - number of the table where the vectorial signal will be generated ielements - number of elements to process idstoffset - (optional, default=0) -- index offset for the destination table iseed (optional, default=0.5) -- seed value for the recursive pseudo-random formula. A value between 0 and +1 will produce an initial output of kamp * iseed. A negative value will cause seed re-initialization to be skipped. A value greater than 1 will seed from system time, this is the best option to generate a different random sequence for each run. isize (optional, default=0) -- if zero, a 16 bit number is generated. If non-zero, a 31-bit random number is generated. Default is 0. ioffset - (optional, default=0) -- a base value added to the random result. Performance krange - range of random elements (from -krange to krange) kcps - rate of generated elements in cycles per seconds This opcode is similar to randi, but operates on vectors instead of with scalar values. Though argument isize defaults to 0, thus using a 16-bit random number generator, using the newer 31-bit algorithm is recommended, as this will produce a random sequence with a longer period (more random numbers before the sequence starts repeating). The output is a vector contained in ifn (that must be previously allocated). All these operators are designed to be used together with other opcodes that operate with vector such as bmscan, adsynt, etc. Note: bmscan not yet available on Canonical Csound Examples Here is an example of the vrandi opcode. It uses the file vrandi.csd. Example of the vrandi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also vrandh, randi Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/semitone.xml0000644000000000000000000000541212262561504016750 0ustar rootroot semitone Pitch Converters:Functions semitone semitone Calculates a factor to raise/lower a frequency by a given amount of semitones. Description Calculates a factor to raise/lower a frequency by a given amount of semitones. Syntax semitone(x) This function works at a-rate, i-rate, and k-rate. Initialization x -- a value expressed in semitones. Performance The value returned by the semitone function is a factor. You can multiply a frequency by this factor to raise/lower it by the given amount of semitones. Examples Here is an example of the semitone opcode. It uses the file semitone.csd. Example of the semitone opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i1 880.00000 i1 2.00000 i1 830.65625 i1 1.88786 i1 783.94665 i1 1.78170 i1 739.98885 i1 1.68179 i1 698.49586 i1 1.58749 i1 659.21793 i1 1.49822 i1 622.25397 i1 1.41421 i1 587.36267 i1 1.33492 i1 554.33399 i1 1.25985 i1 523.25113 i1 1.18921 i1 493.91116 i1 1.12253 i1 466.13747 i1 1.05940 i1 440.00000 i1 1.00000 ....... See Also cent, db, octave Credits New in version 4.16 manual_src~dfsg/opcodes/rigoto.xml0000644000000000000000000000234012262561504016425 0ustar rootroot rigoto Instrument Control:Initialization and Reinitialization rigoto rigoto Transfers control during a reinit pass. Description Similar to igoto, but effective only during a reinit pass (i.e., no-op at standard i-time). This statement is useful for bypassing units that are not to be reinitialized. Syntax rigoto label See Also cigoto, igoto, reinit, rireturn manual_src~dfsg/opcodes/mclock.xml0000644000000000000000000000400712262561504016374 0ustar rootroot mclock Real-time MIDI:System Realtime mclock mclock Sends a MIDI CLOCK message. Description Sends a MIDI CLOCK message. Syntax mclock ifreq Initialization ifreq -- clock message frequency rate in Hz Performance Sends a MIDI CLOCK message (0xF8) every 1/ifreq seconds. So ifreq is the frequency rate of CLOCK message in Hz. See Also mrtmsg Examples Here is an example of the mclock opcode. It uses the file mclock.csd. Example of the mclock opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; Italy New in Csound version 3.47 manual_src~dfsg/opcodes/pvsynth.xml0000644000000000000000000000643312262561504016644 0ustar rootroot pvsynth Spectral Processing:Streaming pvsynth pvsynth Resynthesise using a FFT overlap-add. Description Resynthesise phase vocoder data (f-signal) using a FFT overlap-add. Syntax ares pvsynth fsrc, [iinit] Performance ares -- output audio signal fsrc -- input signal iinit -- not yet implemented. Examples Example (using score-supplied f-table, assuming fsig <emphasis>fftsize = 1024</emphasis>) ; score f-table using cubic spline to define shaped peaks f1 0 513 8 0 2 1 3 0 4 1 6 0 10 1 12 0 16 1 32 0 1 0 436 0 asig buzz 20000, 199, 50, 1 ; pulsewave source fsig pvsanal asig, 1024, 256, 1024, 0 ; create fsig kmod linseg 0, p3/2, 1, p3/2, 0 ; simple control sig fsigout pvsmaska fsig, 2, kmod ; apply weird eq to fsig aout pvsynth fsigout ; resynthesize, dispfft aout, 0.1, 1024 ; and view the effect Here is an example of the pvsynth opcode. It uses the file pvsynth.csd. Example of the pvsynth opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsadsyn Credits Author: &namerichard; August 2001 New in version 4.13 February 2004. Thanks to a note from Francisco Vila, updated the example. manual_src~dfsg/opcodes/modmatrix.xml0000644000000000000000000001742112262561504017134 0ustar rootroot modmatrix Miscellaneous modmatrix modmatrix Modulation matrix opcode with optimizations for sparse matrices. Description The opcode can be used to let a large number of k-rate modulator variables modulate a large number of k-rate parameter variables, with arbitrary scaling of each modulator-to-parameter connection. Csound ftables are used to hold both the input (parameter) variables, the modulator variables, and the scaling coefficients. Output variables are written to another Csound ftable. Syntax modmatrix iresfn, isrcmodfn, isrcparmfn, imodscale, inum_mod, \\ inum_parm, kupdate Initialization iresfn -- ftable number for the parameter output variables isrcmodfn -- ftable number for the modulation source variables isrcparmfn -- ftable number for the parameter input variables imodscale -- scaling/routing coefficient matrix. This is also a csound ftable, used as a matrix of inum_mod rows and inum_parm columns. inum_mod -- number of modulation variables inum_parm -- number of parmeter (input and output) variables. The arguments inum_mod and inum_parm do not have to be set to power-of-two values. Performance kupdate -- flag to update the scaling coefficients. When the flag is set to a nonzero value, the scaling coefficients are read directly from the imodscale ftable. When the flag is set to zero, the scaling coefficients are scanned, and an optimized scaling matrix stored internally in the opcode. For each modulator in isrcmodfn, scale it with the coefficient (in imodscale) determining to what degree it should influence each parameter. Then sum all modulators for each parameter and add the resulting modulator value to the input parameter value read from iscparmfn. Finally, write the output parameter values to table iresfn. The following tables give insight into the processing performed by the modmatrix opcode, for a simplified example using 3 parameter and 2 modulators. Let’s call the parameters "cps1", "cps2", and "cutoff", and the modulators "lfo1" and "lfo2". The input variables may at a given point in time have these values: cps1 cps2 cutoff isrcparmfn 400 800 3
... while the modulator variables have these values: lfo1 lfo2 isrcmodfn 0.5 -0.2
The scaling/routing coefficients used: imodscale cps1 cps2 cutoff lfo1 40 0 -2 lfo2 -50 100 3
... and the resulting output values: cps1 cps2 cutoff iresfn 430 780 1.4 lfo2 -50 100 3
The output value for "cps1" is calculated as 400+(0.5*40)+(-0.2*-50), similarly for "cps2" 800+(0.5*0)+(-0.2*100), and for cutoff: 3+(0.5*-2)+(-0.2*3) The imodscale ftable may be specified in the score like this: f1 0 8 -2 200 0 2 50 300 -1.5 Or more conveniently using ftgen in the orchestra: gimodscale ftgen 0, 0, 8, -2, 200, 0, 2, 50, 300, -1.5 Obviously, the parameter and modulator variables need not be static values, and similarly, the scaling routing coefficient table may be continuously rewritten using opcodes like tablew. Examples Here is an example of the modmatrix opcode. It uses the file modmatrix.csd. Example of the modmatrix opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also Linear Algebra Opcodes, Vectorial Opcodes, tablew. Credits By: Oeyvind Brandtsegg and Thom Johansen New in version 5.12 manual_src~dfsg/opcodes/sprintf.xml0000644000000000000000000000536712262561504016623 0ustar rootroot sprintf Strings:Manipulation sprintf sprintf printf-style formatted output to a string variable. Description sprintf write printf-style formatted output to a string variable, similarly to the C function sprintf(). sprintf runs at i-time only. Syntax Sdst sprintf Sfmt, xarg1[, xarg2[, ... ]] Initialization Sfmt -- format string, has the same format as in printf() and other similar C functions, except length modifiers (l, ll, h, etc.) are not supported. The following conversion specifiers are allowed: d, i, o, u, x, X, e, E, f, F, g, G, c, s xarg1, xarg2, ... -- input arguments (max. 30) for format, should be i-rate for all conversion specifiers except %s, which requires a string argument. Integer formats like %d round the input values to the nearest integer. Performance Sdst -- output string variable Examples Here is an example of the sprintf opcode. It uses the file sprintf.csd. Example of the sprintf opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: The file name is: 'impuls20.aiff' soundin: opened 'impuls20.aiff' See Also sprintfk Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/flsettexttype.xml0000644000000000000000000001262112262561504020051 0ustar rootroot FLsetTextType FLTK:Appearance FLsetTextType FLsetTextType Sets some font attributes of the text label of a FLTK widget. Description FLsetTextType sets some attributes related to the fonts of the text label of the target widget. Syntax FLsetTextType itype, ihandle Initialization itype -- an integer number that modify the appearance of the target widget. The legal values of itype are: 0 - normal label 1 - no label (hides the text) 2 - symbol label (see below) 3 - shadow label 4 - engraved label 5- embossed label 6- bitmap label (not implemented yet) 7- pixmap label (not implemented yet) 8- image label (not implemented yet) 9- multi label (not implemented yet) 10- free-type label (not implemented yet) When using itype=3 (symbol label), it is possible to assign a graphical symbol instead of the text label of the target widget. In this case, the string of the target label must always start with @. If it starts with something else (or the symbol is not found), the label is drawn normally. The following symbols are supported: FLTK label supported symbols. FLTK label supported symbols. The @ sign may be followed by the following optional formatting characters, in this order: # forces square scaling rather than distortion to the widget's shape. +[1-9] or -[1-9] tweaks the scaling a little bigger or smaller. [1-9] rotates by a multiple of 45 degrees. 6 does nothing, the others point in the direction of that key on a numeric keypad. Notice that with FLbox and FLbutton, it is not necessary to call FLsetTextType opcode at all in order to use a symbol. In this case, it is sufficient to set a label starting with @ followed by the proper formatting string. ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/ictrl14.xml0000644000000000000000000000076712262561504016417 0ustar rootroot ictrl14 ictrl14 ictrl14 Deprecated. Description Deprecated as of version 3.52. Use the ctrl14 opcode instead. manual_src~dfsg/opcodes/butterhp.xml0000644000000000000000000000566112262561504016770 0ustar rootroot butterhp Signal Modifiers:Standard Filters butterhp butterhp A high-pass Butterworth filter. Description Implementation of second-order high-pass Butterworth filter. This opcode can also be written as buthp. Syntax ares butterhp asig, kfreq [, iskip] ares butterhp asig, afreq [, iskip] Initialization iskip (optional, default=0) -- Skip initialization if present and non-zero. Performance These filters are Butterworth second-order IIR filters. They are slightly slower than the original filters in Csound, but they offer an almost flat passband and very good precision and stopband attenuation. asig -- Input signal to be filtered. kfreq/afreq -- Cutoff or center frequency for each of the filters. Examples Here is an example of the butterhp opcode. It uses the file butterhp.csd. Example of the butterhp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also butterbp, butterbr, butterlp Credits Author: &nameparis; MIT, Cambridge 1995 Existed in 3.30 Audio rate parameters introduced in version 6.02 October 2013. manual_src~dfsg/opcodes/event.xml0000644000000000000000000001073712262561504016254 0ustar rootroot event Instrument Control:Invocation event event Generates a score event from an instrument. Description Generates a score event from an instrument. Syntax event "scorechar", kinsnum, kdelay, kdur, [, kp4] [, kp5] [, ...] event "scorechar", "insname", kdelay, kdur, [, kp4] [, kp5] [, ...] Initialization scorechar -- A string (in double-quotes) representing the first p-field in a score statement. This is usually e, f, or i. insname -- A string (in double-quotes) representing a named instrument. Performance kinsnum -- The instrument to use for the event. This corresponds to the first p-field, p1, in a score statement. kdelay -- When (in seconds) the event will occur from the current performance time. This corresponds to the second p-field, p2, in a score statement. kdur -- How long (in seconds) the event will happen. This corresponds to the third p-field, p3, in a score statement. kp4, kp5, ... (optional) -- Parameters representing additional p-field in a score statement. It starts with the fourth p-field, p4. Note that the event opcode cannot accept string p-fields. If you need to pass strings when instantiating an instrument, use the scoreline or scoreline_i opcode. Examples Here is an example of the event opcode. It uses the file event.csd. Example of the event opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an example of the event opcode using a named instrument. It uses the file event_named.csd. Example of the event opcode using a named instrument. See also event_i, schedule, schedwhen, schedkwhen, schedkwhennamed, scoreline, scoreline_i Credits Examples written by &namekevin;. New in version 4.17 Thanks goes to Matt Ingalls for helping to fix the example. Thanks goes to Matt Ingalls for helping clarify the kwhen/kdelay parameter. manual_src~dfsg/opcodes/ftlen.xml0000644000000000000000000000720212262561504016234 0ustar rootroot ftlen Table Control:Table Queries ftlen ftlen Returns the size of a stored function table. Description Returns the size of a stored function table. Syntax ftlen(x) (init-rate args only) Performance Returns the size (number of points, excluding guard point) of stored function table, number x. While most units referencing a stored table will automatically take its size into account (so tables can be of arbitrary length), this function reports the actual size if that is needed. Note that ftlen will always return a power-of-2 value, i.e. the function table guard point (see f Statement) is not included.As of Csound version 3.53, ftlen works with deferred function tables (see GEN01). ftlen differs from nsamp in that nsamp gives the number of sample frames loaded, while ftlen gives the total number of samples without the guard point. For example, with a stereo sound file of 10000 samples, ftlen() would return 19999 (i.e. a total of 20000 mono samples, not including a guard point), but nsamp() returns 10000. Examples Here is an example of the ftlen opcode. It uses the file ftlen.csd, fox.wav and beats.wav. Example of the ftlen opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The audio file fox.wav is 121569 samples long, beats.wav has a length of 88200 samples. The ftlen opcode reports them as 121568 and 88199 samples, because it reserves 1 point for the guard point. Its output should include lines like these: instr 1: ift = 121568.000 instr 1: ift = 88199.000 See Also ftchnls, ftlptim, ftsr, nsamp Credits Author: &namebarry; MIT Cambridge, Massachussetts 1997 manual_src~dfsg/opcodes/fmwurlie.xml0000644000000000000000000001045312262561504016760 0ustar rootroot fmwurlie Signal Generators:FM Synthesis fmwurlie fmwurlie Uses FM synthesis to create a Wurlitzer electric piano sound. Description Uses FM synthesis to create a Wurlitzer electric piano sound. It comes from a family of FM sounds, all using 4 basic oscillators and various architectures, as used in the TX81Z synthesizer. Syntax ares fmwurlie kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, \ ifn4, ivfn Initialization All these opcodes take 5 tables for initialization. The first 4 are the basic inputs and the last is the low frequency oscillator (LFO) used for vibrato. The last table should usually be a sine wave. The initial waves should be: ifn1 -- sine wave ifn2 -- sine wave ifn3 -- sine wave ifn4 -- fwavblnk.aiff Note The file fwavblnk.aiff is also available at ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. kc1, kc2 -- Controls for the synthesizer: kc1 -- Mod index 1 kc2 -- Crossfade of two outputs Algorithm -- 5 kvdepth -- Vibrator depth kvrate -- Vibrator rate Examples Here is an example of the fmwurlie opcode. It uses the file fmwurlie.csd, and fwavblnk.aiff. Example of the fmwurlie opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also fmb3, fmbell, fmmetal, fmpercfl, fmrhode Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/ATScross.xml0000644000000000000000000001712212262561504016627 0ustar rootroot ATScross Spectral Processing:ATS ATScross ATScross perform cross synthesis from ATS analysis files. Description ATScross uses data from an ATS analysis file and data from an ATSbufread to perform cross synthesis. Syntax ar ATScross ktimepnt, kfmod, iatsfile, ifn, kmylev, kbuflev, ipartials \ [, ipartialoffset, ipartialincr] Initialization iatsfile – integer or character-string denoting a control-file derived from ATS analysis of an audio signal. An integer denotes the suffix of a file ATS.m; a character-string (in double quotes) gives a filename, optionally a full pathname. If not full-path, the file is sought first in the current directory, then in the one given by the environment variable SADIR (if defined). ifn – table number of a stored function containing a sine wave. ipartials – number of partials that will be used in the resynthesis ipartialoffset (optional) – is the first partial used (defaults to 0). ipartialincr (optional) – sets an increment by which these synthesis opcodes counts up from ipartialoffset for ibins components in the re-synthesis (defaults to 1). Performance ktimepnt – The time pointer in seconds used to index the ATS file. Used for ATScross exactly the same as for pvoc. kfmod – an input for performing pitch transposition or frequency modulation on all of the synthesized partials, if no fm or pitch change is desired then use a 1 for this value. kmylev - scales the ATScross component of the frequency spectrum applied to the partials from the ATS file indicated by the ATScross opcode. The frequency spectrum information comes from the ATScross ATS file. A value of 1 (and 0 for kbuflev) gives the same results as ATSadd. kbuflev - scales the ATSbufread component of the frequency spectrum applied to the partials from the ATS file indicated by the ATScross opcode. The frequency spectrum information comes from the ATSbufread ATS file. A value of 1 (and 0 for kmylev) results in partials that have frequency information from the ATS file given by the ATScross, but amplitudes imposed by data from the ATS file given by ATSbufread. ATScross uses data from an ATS analysis file (indicated by iatsfile) and data from an ATSbufread to perform cross synthesis. ATScross uses ktimepnt, kfmod, ipartials, ipartialoffset and ipartialincr just like ATSadd. ATScross synthesizes a sine-wave for each partial selected by the user and uses the frequency of that partial (after scaling in frequency by kfmod) to index the table created by ATSbufread. Interpolation is used to get in-between values. ATScross uses the sum of the amplitude data from its ATS file (scaled by kmylev) and the amplitude data gained from an ATSbufread (scaled by kbuflev) to scale the amplitude of each partial it synthesizes. Setting kmylev to one and kbuflev to zero will make ATScross act exactly like ATSadd. Setting kmylev to zero and kbuflev to one will produce a sound that has all the partials selected by the ATScross ugen, but with amplitudes taken from an ATSbufread. The time pointers of the ATSbufread and ATScross do not need to be the same. Examples Here is an example of the ATScross opcode. It uses the file ATScross.csd. Example of the ATScross opcode. This example performs cross synthesis using two ATS-files, "fox.ats" and "beats.ats". The result of this will be a sound that starts out with the shape (in frequency) of fox.ats, and ends with the shape of beats.ats. All the sine-wave frequencies come from beats.ats. The kbuflev value is scaled because the energy produced by applying fox.ats's frequency spectrum to beats.ats's partials is very large. Notice also that the time pointers of the ATSbufread (fox.ats) and ATScross (beats.ats) need not have the same value, this way you can read through the two ATS files at different rates. See also ATSread, ATSreadnz, ATSinfo, ATSsinnoi, ATSbufread, ATSinterpread, ATSpartialtap, ATSadd, ATSaddnz Credits Author: Alex Norman Seattle,Washington 2004 manual_src~dfsg/opcodes/midinoteonpch.xml0000644000000000000000000001113112262561504017760 0ustar rootroot midinoteonpch Real-time MIDI:MIDI/Score Interoperability midinoteonpch midinoteonpch Gets a MIDI note number as a pitch-class value. Description midinoteonpch is designed to simplify writing instruments that can be used interchangeably for either score or MIDI input, and to make it easier to adapt instruments originally written for score input to work with MIDI input. In general, it should be possible to write instrument definitions that work identically with both scores and MIDI, including both MIDI files and real-time MIDI input, without using any conditional statements, and that take full advantage of MIDI voice messages. Note that correlating Csound instruments with MIDI channel numbers is done using the massign opcode for real-time performance,. For file-driven performance, instrument numbers default to MIDI channel number + 1, but the defaults are overridden by any MIDI program change messages in the file. Syntax midinoteonpch xpch, xvelocity Performance xpch -- returns MIDI key translated to octave.pch during MIDI activation, remains unchanged otherwise. xvelocity -- returns MIDI velocity during MIDI activation, remains unchanged otherwise. If the instrument was activated by MIDI input, the opcode overwrites the values of xpch and xvelocity with the corresponding value from MIDI input. If the instrument was NOT activated by MIDI input, the values of xpch and xvelocity remain unchanged. This enables score p-fields to receive MIDI input data during MIDI activation, and score values otherwise. Adapting a score-activated Csound instrument. See the MIDI interop opcodes section for details on adapting score driven instruments for MIDI or vice-versa. Examples Here is an example of the midinoteonpch opcode. It uses the file midinoteonpch.csd. Example of the midinoteonpch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i1 8.09000 i1 9.05000 See Also midichannelaftertouch, midicontrolchange, mididefault, midinoteoff, midinoteoncps, midinoteonkey, midinoteonoct, midipitchbend, midipolyaftertouch, midiprogramchange Credits Author: Michael Gogins New in version 4.20 manual_src~dfsg/opcodes/pop.xml0000644000000000000000000001030212262561504015715 0ustar rootroot pop Instrument Control:Stacks pop pop Pops values from the global stack. Description Pops values from the global stack. Syntax xval1, [xval2, ... , xval31] pop ival1, [ival2, ... , ival31] pop Initialization ival1 ... ival31 - values to be popped from the stack. Performance xval1 ... xval31 - values to be popped from the stack. The given values are poped from the stack. The global stack works in LIFO order: after multiple push calls, pop should be used in reverse order. Each push or pop operation can work on a "bundle" of multiple variables. When using pop, the number, type, and order of items must match those used by the corresponding push. That is, after a 'push Sfoo, ibar', you must call something like 'Sbar, ifoo pop', and not e.g. two separate 'pop' statements. push and pop opcodes can take variables of any type (i-, k-, a- and strings). Use of any combination of i, k, a, and S types is allowed. Variables of type 'a' and 'k' are passed at performance time only, while 'i' and 'S' are passed at init time only. push/pop for a, k, i, and S types copy data by value. By contrast, push_f only pushes a "reference" to the f-signal, and then the corresponding pop_f will copy directly from the original variable to its output signal. For this reason, changing the source f-signal of push_f before pop_f is called is not recommended, and if the instrument instance owning the variable that was passed by push_f is deactivated before pop_f is called, undefined behavior may occur. Any stack errors (trying to push when there is no more space, or pop from an empty stack, inconsistent number or type of arguments, etc.) are fatal and terminate performance. Examples Here is an example of the pop opcode. It uses the file pop.csd. Example of the pop opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also stack, push, pop_f and push_f. Using this opcode is somewhat hackish, as you can read here: Credits By: &nameistvan;. 2006 manual_src~dfsg/opcodes/flclosebutton.xml0000644000000000000000000001002412262561504020003 0ustar rootroot FLcloseButton FLTK:Other FLcloseButton FLcloseButton A FLTK widget opcode that creates a button that will close the panel window it is a part of. Description A FLTK widget opcode that creates a button that will close the panel window it is a part of. Syntax ihandle FLcloseButton "label", iwidth, iheight, ix, iy Initialization ihandle -- a handle value (an integer number) that unequivocally references a corresponding widget. This is used by other opcodes that modify a widget's properties (see Modifying FLTK Widget Appearance). It is automatically output by FLcloseButton and must not be set by the user label. (The user label is a double-quoted string containing some user-provided text placed near the widget.) label -- a double-quoted string containing some user-provided text, placed near the corresponding widget. Notice that with FLcloseButton, it is not necessary to call the FLsetTextType opcode at all in order to use a symbol. In this case, it is sufficient to set a label starting with @ followed by the proper formatting string. The following symbols are supported: FLTK label supported symbols. FLTK label supported symbols. The @ sign may be followed by the following optional formatting characters, in this order: # forces square scaling rather than distortion to the widget's shape. +[1-9] or -[1-9] tweaks the scaling a little bigger or smaller. [1-9] rotates by a multiple of 45 degrees. 6 does nothing, the others point in the direction of that key on a numeric keypad. iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). See Also FLbutton, FLbox, FLbutBank, FLprintk, FLprintk2, FLvalue Credits Author: &namesteven; New in version 5.05 manual_src~dfsg/opcodes/sfpassign.xml0000644000000000000000000001046312262561504017124 0ustar rootroot sfpassign Signal Generators:Sample Playback sfpassign sfpassign Assigns all presets of a SoundFont2 (SF2) sample file to a sequence of progressive index numbers. Description Assigns all presets of a previously loaded SoundFont2 (SF2) sample file to a sequence of progressive index numbers. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. sfpassign should be placed in the header section of a Csound orchestra. Syntax sfpassign istartindex, ifilhandle[, imsgs] Initialization istartindex -- starting index preset by the user in bulk preset assignments. ifilhandle -- unique number generated by sfload opcode to be used as an identifier for a SF2 file. Several SF2 files can be loaded and activated at the same time. imsgs -- if non-zero messages are suppressed. Performance sfpassign assigns all presets of a previously loaded SF2 file to a sequence of progressive index numbers, to be used later with the opcodes sfplay and sfplaym. istartindex specifies the starting index number. Any number of sfpassign instances can be placed in the header section of an orchestra, each one assigning presets belonging to different SF2 files. The user must take care that preset index numbers of different SF2 files do not overlap. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. Examples Here is an example of the sfpassign opcode. It uses the file sfpassign.csd. Example of the sfpassign opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also sfilist, sfinstr, sfinstrm, sfload, sfplay, sfplaym, sfplist, sfpreset Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/imagesetpixel.xml0000644000000000000000000000647012262561504017772 0ustar rootroot imagegetpixel Image Processing Opcodes imagesetpixel imagesetpixel Set the RGB value of a pixel inside a previously opened or created image. Description Set the RGB value of a pixel inside a previously opened or created image. An image can be loaded with imageload. An empty image can be created with imagecreate and saved with imagesave. Syntax imagesetpixel iimagenum, ax, ay, ared, agreen, ablue imagesetpixel iimagenum, kx, ky, kred, kgreen, kblue Initialization iimagenum -- the reference of the image.. It should be a value returned by imageload or imagecreate. Performance ax (kx) -- horizontal pixel position (must be a float from 0 to 1). ay (ky) -- vertical pixel position (must be a float from 0 to 1). ared (kred) -- red value of the pixel (mapped to a float from 0 to 1). agreen (kgreen) -- green value of the pixel (mapped to a float from 0 to 1). ablue (kblue) -- blue value of the pixel (mapped to a float from 0 to 1). Examples Here is an example of the imagesetpixel opcode. It uses the file imageopcodes.csd. Example of the imagesetpixel opcode. See Also imageload, imagecreate, imagesize, imagesave, imagegetpixel, imagefree Credits Author: Cesare Marilungo New in version 5.08 manual_src~dfsg/opcodes/sinh.xml0000644000000000000000000000431312262561504016065 0ustar rootroot sinh Mathematical Operations:Trigonometric Functions sinh sinh Performs a hyperbolic sine function. Description Returns the hyperbolic sine of x (x in radians). Syntax sinh(x) (no rate restriction) Examples Here is an example of the sinh opcode. It uses the file sinh.csd. Example of the sinh opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should a line like this: instr 1: i1 = 1.175 See Also cos, cosh, cosinv, sin, sininv, tan, tanh, taninv Credits Example written by &namekevin;. New in version 3.47 manual_src~dfsg/opcodes/flhvsbox.xml0000644000000000000000000000712712262561504016765 0ustar rootroot FLhvsBox FLTK:Other FLhvsBox FLhvsBox Displays a box with a grid useful for visualizing two-dimensional Hyper Vectorial Synthesis. Description FLhvsBox displays a box with a grid useful for visualizing two-dimensional Hyper Vectorial Synthesis. Syntax ihandle FLhvsBox inumlinesX, inumlinesY, iwidth, iheight, ix, iy [, image] Initialization ihandle – an integer number used a univocally-defined handle for identifying a specific HVS box (see below). inumlinesX, inumlinesY - number of vertical and horizontal lines delimiting the HVS squared areas iwidth, iheight - width and height of the HVS box ix, iy - the position of the HVS box image – (optional, default 0) an integer number denoting an RGB image opened with the bmopen opcode. A zero indicates no image. Performance FLhvsBox is a widget able to visualize current position of the HVS cursor in an HVS box (i.e. a squared area containing a grid). The number of horizontal and vertical lines of the grid can be defined with the inumlinesX, inumlinesY arguments. This opcode has to be declared inside an FLpanel - FLpanelEnd block. See the entry for hvs2 for an example of usage of FLhvsBox. FLhvsBoxSetValue is used to set the cursor position of an FLhvsBox widget. The opcode bmscan has not been implemented, so currently the parameter image has no effect. See Also hvs2, FLhvsBoxSetValue Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/foutir.xml0000644000000000000000000001256112262561504016440 0ustar rootroot foutir Signal I/O:File I/O foutir foutir Outputs i-rate signals from an arbitrary number of channels to a specified file. Description foutir output N i-rate signals to a specified file of N channels. Syntax foutir ihandle, iformat, iflag, iout1 [, iout2, iout3,....,ioutN] Initialization ihandle -- a number which specifies this file. iformat -- a flag to choose output file format: 0 - floating point in text format 1 - 32-bit floating point in binary format iflag -- choose the mode of writing to the ASCII file (valid only in ASCII mode; in binary mode iflag has no meaning, but it must be present anyway). iflag can be a value chosen among the following: 0 - line of text without instrument prefix 1 - line of text with instrument prefix (see below) 2 - reset the time of instrument prefixes to zero (to be used only in some particular cases. See below) iout,..., ioutN -- values to be written to the file Performance fouti and foutir write i-rate values to a file. The main use of these opcodes is to generate a score file during a realtime session. For this purpose, the user should set iformat to 0 (text file output) and iflag to 1, which enable the output of a prefix consisting of the strings inum, actiontime, and duration, before the values of iout1...ioutN arguments. The arguments in the prefix refer to instrument number, action time and duration of current note. The difference between fouti and foutir is that, in the case of fouti, when iflag is set to 1, the duration of the first opcode is undefined (so it is replaced by a dot). Whereas, foutir is defined at the end of note, so the corresponding text line is written only at the end of the current note (in order to recognize its duration). The corresponding file is linked by the ihandle value generated by the fiopen opcode. So fouti and foutir can be used to generate a Csound score while playing a realtime session. Notice that fout and foutk can use either a string containing a file pathname, or a handle-number generated by fiopen. Whereas, with fouti and foutir, the target file can be only specified by means of a handle-number. Examples Here is an example of the foutir opcode. It uses the file foutir.csdand creates a list "foutir.sco". It can be used as a score file. Example of the foutir opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also fiopen, fout, fouti, foutk Credits Author: &namegabriel; Italy 1999 New in Csound version 3.56 manual_src~dfsg/opcodes/temposcal.xml0000644000000000000000000000721412262561504017116 0ustar rootroot temposcal Spectral Processing:Other temposcal temposcal Phase-locked vocoder processing with onset detection/processing, 'tempo-scaling'. Description temposcal implements phase-locked vocoder processing using function tables containing sampled-sound sources, with GEN01, and temposcal will accept deferred allocation tables. This opcode allows for time and frequency-independent scaling. Time is advanced internally, but controlled by a tempo scaling parameter; when an onset is detected, timescaling is momentarily stopped to avoid smearing of attacks. The quality of the effect is generally improved with phase locking switched on. temposcal will also scale pitch, independently of frequency, using a transposition factor (k-rate). Syntax asig temposcal ktimescal, kamp, kpitch, ktab, klock [,ifftsize, idecim, ithresh] Initialization ifftsize -- FFT size (power-of-two), defaults to 2048. idecim -- decimation, defaults to 4 (meaning hopsize = fftsize/4) idbthresh -- threshold based on dB power spectrum ratio between two successive windows. A detected ratio above it will cancel timescaling momentarily, to avoid smearing (defaults to 1) Performance ktimescal -- timescaling ratio, < 1 stretch, > 1 contract. kamp -- amplitude scaling kpitch -- grain pitch scaling (1=normal pitch, < 1 lower, > 1 higher; negative, backwards) klock -- 0 or 1, to switch phase-locking on/off ktab -- source signal function table. Deferred-allocation tables (see GEN01) are accepted, but the opcode expects a mono source. Tables can be switched at k-rate. Examples Here is an example of the temposcal opcode. It uses the file temposcal.csd. Example of the temposcal opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; February 2010 New plugin in version 5.13 February 2005. manual_src~dfsg/opcodes/vbapgmove.xml0000644000000000000000000001404212262561504017112 0ustar rootroot vbapgmove Signal Modifiers:Panning and Spatialization vbapgmove vbapgmove Calculates the gains for a sound location between multiple channels with moving virtual sources. Description Calculates the gains for a sound location between multiple channels with moving virtual sources. Syntax kr1[, kr2...] vbapgmove idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] karray[] vbapgmove idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] Initialization idur -- the duration over which the movement takes place. ispread -- spreading of the virtual source (range 0 - 100). If value is zero, conventional amplitude panning is used. When ispread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. ifldnum -- number of fields (absolute value must be 2 or larger). If ifldnum is positive, the virtual source movement is a polyline specified by given directions. Each transition is performed in an equal time interval. If ifldnum is negative, specified angular velocities are applied to the virtual source during specified relative time intervals (see below). ifld1, ifld2, ... -- azimuth angles or angular velocities, and relative durations of movement phases (see below). Performance vbapgmove allows the use of moving virtual sources. If ifldnum is positive, the fields represent directions of virtual sources and equal times, iazi1, [iele1,] iazi2, [iele2,], etc. The position of the virtual source is interpolated between directions starting from the first direction and ending at the last. Each interval is interpolated in time that is fraction total_time / number_of_intervals of the duration of the sound event. If ifldnum is negative, the fields represent angular velocities and equal times. The first field is, however, the starting direction, iazi1, [iele1,] iazi_vel1, [iele_vel1,] iazi_vel2, [iele_vel2,] .... Each velocity is applied to the note that is fraction total_time / number_of_velocities of the duration of the sound event. If the elevation of the virtual source becomes greater than 90 degrees or less than 0 degrees, the polarity of angular velocity is changed. Thus the elevational angular velocity produces a virtual source that moves up and down between 0 and 90 degrees. Please note that all vbap panning opcodes require the vbap system to be initialized using vbaplsinit. Examples Here is an example of the vbapgmove opcode. It uses the file vbapgmove.csd. Example of the vbapg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. See Also vbap, vbap16, vbap16move, vbap4, vbap4move, vbap8, vbap8move, vbaplsinit, vbapz, vbapzmove Credits Author: Ville Pulkki Sibelius Academy Computer Music Studio Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Helsinki, Finland May 2000 Author: &namejohn; July 2012, Sept 2013 New in Csound Version 5.17.13 Array outputs new in Csound Version 6.01 manual_src~dfsg/opcodes/fiopen.xml0000644000000000000000000000753112262561504016411 0ustar rootroot fiopen Signal I/O:File I/O fiopen fiopen Opens a file in a specific mode. Description fiopen can be used to open a file in one of the specified modes. Syntax ihandle fiopen ifilename, imode Initialization ihandle -- a number which specifies this file. ifilename -- the output file's name (in double-quotes). imode -- choose the mode of opening the file. imode can be a value chosen among the following: 0 - open a text file for writing 1 - open a text file for reading 2 - open a binary file for writing 3 - open a binary file for reading Performance fiopen opens a file to be used by the fout family of opcodes. It is safer to use it in the header section, external to any instruments. It returns a number, ihandle, which unequivocally refers to the opened file. If fiopen is called on an already open file, it just returns the same handle again, and does not close the file. Notice that fout and foutk can use either a string containing a file pathname, or a handle-number generated by fiopen. Whereas, with fouti and foutir, the target file can be only specified by means of a handle-number. Examples Here is an example of the fiopen opcode. It uses the file fiopen.csd. Example of the fiopen opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also ficlose fout, fouti, foutir, foutk Credits Author: &namegabriel; Italy 1999 New in Csound version 3.56 manual_src~dfsg/opcodes/harmon.xml0000644000000000000000000001111312262561504016404 0ustar rootroot harmon Signal Modifiers:Special Effects harmon harmon Analyze an audio input and generate harmonizing voices in synchrony. Description Analyze an audio input and generate harmonizing voices in synchrony. Syntax ares harmon asig, kestfrq, kmaxvar, kgenfreq1, kgenfreq2, imode, \ iminfrq, iprd Initialization imode -- interpreting mode for the generating frequency inputs kgenfreq1, kgenfreq2. 0: input values are ratios with respect to the audio signal analyzed frequency. 1: input values are the actual requested frequencies in Hz. iminfrq -- the lowest expected frequency (in Hz) of the audio input. This parameter determines how much of the input is saved for the running analysis, and sets a lower bound on the internal pitch tracker. iprd -- period of analysis (in seconds). Since the internal pitch analysis can be time-consuming, the input is typically analyzed only each 20 to 50 milliseconds. Performance kestfrq -- estimated frequency of the input. kmaxvar -- the maximum variance (expects a value betwee 0 and 1). kgenfreq1 -- the first generated frequency. kgenfreq2 -- the second generated frequency. This unit is a harmonizer, able to provide up to two additional voices with the same amplitude and spectrum as the input. The input analysis is assisted by two things: an input estimated frequency kestfrq (in Hz), and a fractional maximum variance kmaxvar about that estimate which serves to limit the size of the search. Once the real input frequency is determined, the most recent pulse shape is used to generate the other voices at their requested frequencies. The three frequency inputs can be derived in various ways from a score file or MIDI source. The first is the expected pitch, with a variance parameter allowing for inflections or inaccuracies; if the expected pitch is zero the harmonizer will be silent. The second and third pitches control the output frequencies; if either is zero the harmonizer will output only the non-zero request; if both are zero the harmonizer will be silent. When the requested frequency is higher than the input, the process requires additional computation due to overlapped output pulses. This is currently limited for efficiency reasons, with the result that only one voice can be higher than the input at any one time. This unit is useful for supplying a background chorus effect on demand, or for correcting the pitch of a faulty input vocal. There is essentially no delay between input and output. Output includes only the generated parts, and does not include the input. Examples Here is an example of the harmon opcode. It uses the file harmon.csd. Example of the harmon opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namebarry; M.I.T., Cambridge, Mass 1997 New in version 3.47 manual_src~dfsg/opcodes/trigger.xml0000644000000000000000000000577712262561504016606 0ustar rootroot trigger Instrument Control:Sensing and Control trigger trigger Informs when a krate signal crosses a threshold. Description Informs when a krate signal crosses a threshold. Syntax kout trigger ksig, kthreshold, kmode Performance ksig -- input signal kthreshold -- trigger threshold kmode -- can be 0 , 1 or 2 Normally trigger outputs zeroes: only each time ksig crosses kthreshold trigger outputs a 1. There are three modes of using ktrig: kmode = 0 - (down-up) ktrig outputs a 1 when current value of ksig is higher than kthreshold, while old value of ksig was equal to or lower than kthreshold. kmode = 1 - (up-down) ktrig outputs a 1 when current value of ksig is lower than kthreshold while old value of ksig was equal or higher than kthreshold. kmode = 2 - (both) ktrig outputs a 1 in both the two previous cases. Examples Here is an example of the trigger opcode. It uses the file trigger.csd. Example of the trigger opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; Italy New in Csound version 3.49 manual_src~dfsg/opcodes/dconv.xml0000644000000000000000000000567712262561504016253 0ustar rootroot dconv Signal Modifiers:Convolution and Morphing dconv dconv A direct convolution opcode. Description A direct convolution opcode. Syntax ares dconv asig, isize, ifn Initialization isize -- the size of the convolution buffer to use. If the buffer size is smaller than the size of ifn, then only the first isize values will be used from the table. ifn -- table number of a stored function containing the impulse response for convolution. Performance Rather than the analysis/resynthesis method of the convolve opcode, dconv uses direct convolution to create the result. For small tables it can do this quite efficiently, however larger table require much more time to run. dconv does (isize * ksmps) multiplies on every k-cycle. Therefore, reverb and delay effects are best done with other opcodes (unless the times are short). dconv was designed to be used with time varying tables to facilitate new realtime filtering capabilities. Examples Here is an example of the dconv opcode. It uses the file dconv.csd. Example of the dconv opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also pconvolve, convolve, ftconv Credits Author: William Pete Moss 2001 New in version 4.12 manual_src~dfsg/opcodes/jitter2.xml0000644000000000000000000000665612262561504016523 0ustar rootroot jitter2 Signal Generators:Random (Noise) Generators jitter2 jitter2 Generates a segmented line with user-controllable random segments. Description Generates a segmented line with user-controllable random segments. Syntax kout jitter2 ktotamp, kamp1, kcps1, kamp2, kcps2, kamp3, kcps3 Performance ktotamp -- Resulting amplitude of jitter2 kamp1 -- Amplitude of the first jitter component kcps1 -- Speed of random variation of the first jitter component (expressed in cps) kamp2 -- Amplitude of the second jitter component kcps2 -- Speed of random variation of the second jitter component (expressed in cps) kamp3 -- Amplitude of the third jitter component kcps3 -- Speed of random variation of the third jitter component (expressed in cps) jitter2 also generates a segmented line such as jitter, but in this case the result is similar to the sum of three randi opcodes, each one with a different amplitude and frequency value (see randi for more details), that can be varied at k-rate. Different effects can be obtained by varying the input arguments. jitter2 can be used to make more natural and analog-sounding some static, dull sound. For best results, it is suggested to keep its amplitude moderate. Examples Here is an example of the jitter2 opcode. It uses the file jitter2.csd. Example of the jitter2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also jitter, vibr, vibrato Credits Author: &namegabriel; New in Version 4.15 manual_src~dfsg/opcodes/rireturn.xml0000644000000000000000000000347012262561504017001 0ustar rootroot rireturn Instrument Control:Initialization and Reinitialization rireturn rireturn Terminates a reinit pass. Description Terminates a reinit pass (i.e., no-op at standard i-time). This statement, or an endin, will cause normal performance to be resumed. Syntax rireturn Examples The following statements will generate an exponential control signal whose value moves from 440 to 880 exactly ten times over the duration p3. They use the file reinit.csd. Example of the rireturn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also reinit, rigoto manual_src~dfsg/opcodes/flgroupend.xml0000644000000000000000000000364712262561504017302 0ustar rootroot FLgroupEnd FLTK:Containers FLgroupEnd FLgroupEnd Marks the end of a group of FLTK child widgets. Description Marks the end of a group of FLTK child widgets. Syntax FLgroupEnd Performance Containers are useful to format the graphic appearance of the widgets. The most important container is FLpanel, that actually creates a window. It can be filled with other containers and/or valuators or other kinds of widgets. There are no k-rate arguments in containers. See Also FLgroup, FLpack, FLpackEnd, FLpanel, FLpanelEnd, FLscroll, FLscrollEnd, FLtabs, FLtabsEnd Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/mixersetlevel.xml0000644000000000000000000000755312262561504020025 0ustar rootroot MixerSetLevel Mixer Opcodes MixerSetLevel MixerSetLevel Sets the level of a send to a buss. Syntax MixerSetLevel isend, ibuss, kgain Description Sets the level at which signals from the send are added to the buss. The actual sending of the signal to the buss is performed by the MixerSend opcode. Initialization isend -- The number of the send, for example the number of the instrument sending the signal (but any integer can be used). ibuss -- The number of the buss, for example the number of the instrument receiving the signal (but any integer can be used). Setting the gain for a buss also creates the buss. Performance kgain -- The level (any real number) at which the signal from the send will be mixed onto the buss. The default is 0. Use of the mixer requires that instruments setting gains have smaller numbers than instruments sending signals, and that instruments sending signals have smaller numbers than instruments receiving those signals. However, an instrument may have any number of sends or receives. After the final signal is received, MixerClear must be invoked to reset the busses before the next kperiod. Examples In the orchestra, define an instrument to control mixer levels: instr 1 MixerSetLevel p4, p5, p6 endin In the score, use that instrument to set mixer levels: ; SoundFonts ; to Chorus i 1 0 0 100 200 0.9 ; to Reverb i 1 0 0 100 210 0.7 ; to Output i 1 0 0 100 220 0.3 ; Kelley Harpsichord ; to Chorus i 1 0 0 3 200 0.30 ; to Reverb i 1 0 0 3 210 0.9 ; to Output i 1 0 0 3 220 0.1 ; Chorus to Reverb i 1 0 0 200 210 0.5 ; Chorus to Output i 1 0 0 200 220 0.5 ; Reverb to Output i 1 0 0 210 220 0.2 Here is a complete example of the MixerSetLevel opcode. It uses the file Mixer.csd Complete example of the MixerSetLevel opcode. Credits Michael Gogins (gogins at pipeline dot com). manual_src~dfsg/opcodes/xscanmap.xml0000644000000000000000000000543612262561504016745 0ustar rootroot xscanmap Signal Generators:Scanned Synthesis xscanmap xscanmap Allows the position and velocity of a node in a scanned process to be read. Description Allows the position and velocity of a node in a scanned process to be read. Syntax kpos, kvel xscanmap iscan, kamp, kvamp [, iwhich] Initialization iscan -- which scan process to read iwhich (optional) -- which node to sense. The default is 0. Performance kamp -- amount to amplify the kpos value. kvamp -- amount to amplify the kvel value. The internal state of a node is read. This includes its position and velocity. They are amplified by the kamp and kvamp values. Examples Here is an example of the xscanmap opcode. It uses the file xscanmap.csd. Example of the xscanmap opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also More information on Scanned Synthesis (as well as several other matrices) is available on the Scanned Synthesis page at cSounds.com. Also an article on these opcodes: http://www.csounds.com/stevenyi/scanned/yi_scannedSynthesis.html , written by Steven Yi Credits Author: &namejohn; New in version 4.20 manual_src~dfsg/opcodes/STKSaxofony.xml0000644000000000000000000001047712262561504017324 0ustar rootroot STKSaxofony Signal Generators:STK Opcodes STKSaxofony STKSaxofony STKSaxofony is a faux conical bore reed instrument. Description STKSaxofony is a faux conical bore reed instrument. This opcode uses a "hybrid" digital waveguide instrument that can generate a variety of wind-like sounds. It has also been referred to as the "blowed string" model. The waveguide section is essentially that of a string, with one rigid and one lossy termination. The non-linear function is a reed table. The string can be "blown" at any point between the terminations, though just as with strings, it is impossible to excite the system at either end. If the excitation is placed at the string mid-point, the sound is that of a clarinet. At points closer to the "bridge", the sound is closer to that of a saxophone. Syntax asignal STKSaxofony ifrequency, iamplitude, [kstiff, kv1[, kapert, kv2[, kblow, kv3[, knoise, kv4[, klfo, kv5[, klfodepth, kv6[, kbreath, kv7]]]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kstiff -- controller 2, stiffness of reed. Value range of kv1 is 0-127. kapert -- controller 26, reed aperture. Value range of kv2 is 0-127. kblow -- controller 11, blow position. Value range of kv3 is 0-127. knoise -- controller 4, noise gain. Value range of kv4 is 0-127. klfo -- controller 29, speed of low-frequency oscillator. Value range of kv5 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv6 is 0-127. kbreath -- controller 128, breath pressure. Value range of kv7 is 0-127. Note The code for this opcode is taken directly from the Saxofony class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKSaxofony opcode. It uses the file STKSaxofony.csd. Example of the STKSaxofony opcode. Credits Author: &namemichael; (after Georg Essl) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/strrindexk.xml0000644000000000000000000000225712262561504017326 0ustar rootroot strrindexk Strings:Manipulation strrindexk strrindexk Return the position of the last occurence of a string in another string Description Return the position of the last occurence of S2 in S1, or -1 if not found. If S2 is empty, the length of S1 is returned. strrindexk runs both at init and performance time. Syntax kpos strrindexk S1, S2 See also strrindex Credits Author: Istvan Varga 2006 New in version 5.02 manual_src~dfsg/opcodes/sfpreset.xml0000644000000000000000000001063412262561504016762 0ustar rootroot sfpreset Signal Generators:Sample Playback sfpreset sfpreset Assigns an existing preset of a SoundFont2 (SF2) sample file to an index number. Description Assigns an existing preset of a previously loaded SoundFont2 (SF2) sample file to an index number. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. sfpreset should be placed in the header section of a Csound orchestra. Syntax ir sfpreset iprog, ibank, ifilhandle, ipreindex Initialization ir -- output to be used by other SF2 opcodes. For sfpreset, ir is ipreindex. iprog -- program number of a bank of presets in a SF2 file ibank -- number of a specific bank of a SF2 file ifilhandle -- unique number generated by sfload opcode to be used as an identifier for a SF2 file. Several SF2 files can be loaded and activated at the same time. ipreindex -- preset index Performance sfpreset assigns an existing preset of a previously loaded SF2 file to an index number, to be used later with the opcodes sfplay and sfplaym. The user must previously know the program and the bank numbers of the preset in order to fill the corresponding arguments. Any number of sfpreset instances can be placed in the header section of an orchestra, each one assigning a different preset belonging to the same (or different) SF2 file to different index numbers. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. Examples Here is an example of the sfpreset opcode. It uses the file sfpreset.csd. Example of the sfpreset opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also sfilist, sfinstr, sfinstrm, sfload, sfpassign, sfplay, sfplaym, sfplist Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/apcauchy.xml0000644000000000000000000000077612262561504016732 0ustar rootroot apcauchy apcauchy apcauchy Deprecated. Description Deprecated as of version 3.49. Use the pcauchy opcode instead. manual_src~dfsg/opcodes/ipoisson.xml0000644000000000000000000000077612262561504017000 0ustar rootroot ipoisson ipoisson ipoisson Deprecated. Description Deprecated as of version 3.49. Use the poisson opcode instead. manual_src~dfsg/opcodes/adsr.xml0000644000000000000000000000744712262561504016070 0ustar rootroot adsr Signal Generators:Envelope Generators adsr adsr Calculates the classical ADSR envelope using linear segments. Description Calculates the classical ADSR envelope using linear segments. Syntax ares adsr iatt, idec, islev, irel [, idel] kres adsr iatt, idec, islev, irel [, idel] Initialization iatt -- duration of attack phase idec -- duration of decay islev -- level for sustain phase irel -- duration of release phase idel -- period of zero before the envelope starts Performance The envelope generated is the range 0 to 1 and may need to be scaled further, depending on the amplitude required. If using 0dbfs = 1, scaling down will probably be required since playing more than one note might result in clipping. If not using 0dbfs, scaling to a large amplitude (e.g. 32000) might be required. The envelope may be described as: Picture of an ADSR envelope. Picture of an ADSR envelope. The length of the sustain is calculated from the length of the note. This means adsr is not suitable for use with MIDI events. The opcode madsr uses the linsegr mechanism, and so can be used in MIDI applications. adsr is new in Csound version 3.49. Examples Here is an example of the adsr opcode. It uses the file adsr.csd. Example of the adsr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also madsr, mxadsr, xadsr Credits Author: &namejohn; New in version 3.49 manual_src~dfsg/opcodes/flvslidbnk.xml0000644000000000000000000002363012262561504017265 0ustar rootroot FLvslidBnk FLTK:Other FLvslidBnk FLvslidBnk A FLTK widget containing a bank of vertical sliders. Description FLvslidBnk is a widget containing a bank of vertical sliders. Syntax FLvslidBnk "names", inumsliders [, ioutable] [, iwidth] [, iheight] [, ix] \ [, iy] [, itypetable] [, iexptable] [, istart_index] [, iminmaxtable] Initialization names -- a double-quoted string containing the names of each slider. Each slider can have a different name. Separate each name with @ character, for example: frequency@amplitude@cutoff. It is possible to not provide any name by giving a single space . In this case, the opcode will automatically assign a progressive number as a label for each slider. inumsliders -- the number of sliders. ioutable (optional, default=0) -- number of a previously-allocated table in which to store output values of each slider. The user must be sure that table size is large enough to contain all output cells, otherwise a segfault will crash Csound. By assigning zero to this argument, the output will be directed to the zak space in the k-rate zone. In this case, the zak space must be previously allocated with the zakinit opcode and the user must be sure that the allocation size is big enough to cover all sliders. The default value is zero (i.e. store output in zak space). istart_index (optional, default=0) -- an integer number referring to a starting offset of output cell locations. It can be positive to allow multiple banks of sliders to output in the same table or in the zak space. The default value is zero (no offset). iminmaxtable (optional, default=0) -- number of a previously-defined table containing a list of min-max pairs, referred to each slider. A zero value defaults to the 0 to 1 range for all sliders without necessity to provide a table. The default value is zero. iexptable (optional, default=0) -- number of a previously-defined table containing a list of identifiers (i.e. integer numbers) provided to modify the behaviour of each slider independently. Identifiers can assume the following values: -1 -- exponential curve response 0 -- linear response number > than 0 -- follow the curve of a previously-defined table to shape the response of the corresponding slider. In this case, the number corresponds to table number. You can assume that all sliders of the bank have the same response curve (exponential or linear). In this case, you can assign -1 or 0 to iexptable without worrying about previously defining any table. The default value is zero (all sliders have a linear response, without having to provide a table). itypetable (optional, default=0) -- number of a previously-defined table containing a list of identifiers (i.e. integer numbers) provided to modify the aspect of each individual slider independently. Identifiers can assume the following values: 0 = Nice slider 1 = Fill slider 3 = Normal slider 5 = Nice slider 7 = Nice slider with down-box You can assume that all sliders of the bank have the same aspect. In this case, you can assign a negative number to itypetable without worrying about previously defining any table. Negative numbers have the same meaning of the corresponding positive identifiers with the difference that the same aspect is assigned to all sliders. You can also assign a random aspect to each slider by setting itypetable to a negative number lower than -7. The default value is zero (all sliders have the aspect of nice sliders, without having to provide a table). You can add 20 to a value inside the table to make the slider "plastic", or subtract 20 if you want to set the value for all widgets without defining a table (e.g. -21 to set all sliders types to Plastic Fill slider). iwidth (optional) -- width of the rectangular area containing all sliders of the bank, excluding text labels, that are placed below that area. iheight (optional) -- height of the rectangular area containing all sliders of the bank, excluding text labels, that are placed below that area. ix (optional) -- horizontal position of the upper left corner of the rectangular area containing all sliders belonging to the bank. You have to leave enough space, below that rectangle, in order to make sure labels of sliders to be visible. This is because the labels themselves are external to the rectangular area. iy (optional) -- vertical position of the upper left corner of the rectangular area containing all sliders belonging to the bank. You have to leave enough space, below that rectangle, in order to make sure labels of sliders to be visible. This is because the labels themselves are external to the rectangular area. Performance There are no k-rate arguments, even if cells of the output table (or the zak space) are updated at k-rate. FLvslidBnk is a widget containing a bank of vertical sliders. Any number of sliders can be placed into the bank (inumsliders argument). The output of all sliders is stored into a previously allocated table or into the zak space (ioutable argument). It is possible to determine the first location of the table (or of the zak space) in which to store the output of the first slider by means of istart_index argument. Each slider can have an individual label that is placed below it. Labels are defined by the names argument. The output range of each slider can be individually set by means of an external table (iminmaxtable argument). The curve response of each slider can be set individually, by means of a list of identifiers placed in a table (iexptable argument). It is possible to define the aspect of each slider independently or to make all sliders have the same aspect (itypetable argument). The iwidth, iheight, ix, and iy arguments determine width, height, horizontal and vertical position of the rectangular area containing sliders. Notice that the label of each slider is placed below them and is not included in the rectangular area containing sliders. So the user should leave enough space below the bank by assigning a proper iy value in order to leave labels visible. FLvslidBnk is identical to FLslidBnk except it contains vertical sliders instead of horizontal. Since the width of each single slider is often small, it is recommended to leave only a single space in the names string (“ “), in this case each slider will be automatically numbered. IMPORTANT! Notice that the tables used by FLvslidBnk must be created with the ftgen opcode and placed in the orchestra before the corresponding valuator. They can not placed in the score. This is because tables placed in the score are created later than the initialization of the opcodes placed in the header section of the orchestra. Examples Here is an example of the FLvslidBnk opcode. It uses the file FLvslidBnk.csd. Example of the FLvslidBnk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLslider, FLslidBnk Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/lpreson.xml0000644000000000000000000001226712262561504016615 0ustar rootroot lpreson Spectral Processing:LPC lpreson lpreson Resynthesises a signal from the data passed internally by a previous lpread. Description Resynthesises a signal from the data passed internally by a previous lpread. Syntax ares lpreson asig Performance asig -- an audio driving function for resynthesis. lpreson receives values internally produced by a leading lpread.lpread gets its values from the control file according to the input value ktimpnt (in seconds). If ktimpnt proceeds at the analysis rate, time-normal synthesis will result; proceeding at a faster, slower, or variable rate will result in time-warped synthesis. At each k-period, lpread interpolates between adjacent frames to more accurately determine the parameter values (presented as output) and the filter coefficient settings (passed internally to a subsequent lpreson). The error signal kerr (between 0 and 1) derived during predictive analysis reflects the deterministic/random nature of the analyzed source. This will emerge low for pitched (periodic) material and higher for noisy material. The transition from voiced to unvoiced speech, for example, produces an error signal value of about .3. During synthesis, the error signal value can be used to determine the nature of the lpreson driving function: for example, by arbitrating between pitched and non-pitched input, or even by determining a mix of the two. In normal speech resynthesis, the pitched input to lpreson is a wideband periodic signal or pulse train derived from a unit such as buzz, and the nonpitched source is usually derived from rand. However, any audio signal can be used as the driving function, the only assumption of the analysis being that it has a flat response. lpfreson is a formant shifted lpreson, in which kfrqratio is the (cps) ratio of shifted to original formant positions. This permits synthesis in which the source object changes its apparent acoustic size. lpfreson with kfrqratio = 1 is equivalent to lpreson. Generally, lpreson provides a means whereby the time-varying content and spectral shaping of a composite audio signal can be controlled by the dynamic spectral content of another. There can be any number of lpread/lpreson (or lpfreson) pairs in an instrument or in an orchestra; they can read from the same or different control files independently. Examples Here is an example of the lpreson opcode. It uses the file lpreson.csd. Example of the lpreson opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The audio file fox.wav is 2.8 seconds long. So filelen's output should be a line like this: fox.wav = 2.756667 seconds Here is another example of the lpreson opcode. It uses the file lpreson-2.csd. Another example of the lpreson opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also lpfreson, lpread manual_src~dfsg/opcodes/max.xml0000644000000000000000000000560312262561504015714 0ustar rootroot max Signal Modifiers:Comparators and Accumulators max max Produces a signal that is the maximum of any number of input signals. Description The max opcode takes any number of a-rate or k-rate signals as input (all of the same rate), and outputs a signal at the same rate that is the maximum of all of the inputs. For a-rate signals, the inputs are compared one sample at a time (i.e. max does not scan an entire ksmps period of a signal for its local maximum as the max_k opcode does). Syntax amax max ain1, ain2 [, ain3] [, ain4] [...] kmax max kin1, kin2 [, kin3] [, kin4] [...] Performance ain1, ain2, ... -- a-rate signals to be compared. kin1, kin2, ... -- k-rate signals to be compared. Examples Here is an example of the max opcode. It uses the file max.csd. Example of the max opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also min, maxabs, minabs, maxaccum, minaccum, maxabsaccum, minabsaccum, max_k Credits Author: &nameanthony; March 2006 New in Csound version 5.01 manual_src~dfsg/opcodes/zkmod.xml0000644000000000000000000000535012262561504016252 0ustar rootroot zkmod Zak Patch System zkmod zkmod Facilitates the modulation of one signal by another. Description Facilitates the modulation of one signal by another. Syntax kres zkmod ksig, kzkmod Performance ksig -- the input signal kzkmod -- controls which zk variable is used for modulation. A positive value means additive modulation, a negative value means multiplicative modulation. A value of 0 means no change to ksig. kzkmod can be i-rate or k-rate zkmod facilitates the modulation of one signal by another, where the modulating signal comes from a zk variable. Either additive or mulitiplicative modulation can be specified. Examples Here is an example of the zkmod opcode. It uses the file zkmod.csd. Example of the zkmod opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zamod, zkcl, zkr, zkwm, zkw Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/areson.xml0000644000000000000000000001317712262561504016423 0ustar rootroot areson Signal Modifiers:Standard Filters:Resonant areson areson A notch filter whose transfer functions are the complements of the reson opcode. Description A notch filter whose transfer functions are the complements of the reson opcode. Syntax ares areson asig, kcf, kbw [, iscl] [, iskip] ares areson asig, acf, kbw [, iscl] [, iskip] ares areson asig, kcf, abw [, iscl] [, iskip] ares areson asig, acf, abw [, iscl] [, iskip] Initialization iscl (optional, default=0) -- coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf/acf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. (This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise.) A zero value signifies no scaling of the signal, leaving that to some later adjustment (see balance). The default value is 0. iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance ares -- the output signal at audio rate. asig -- the input signal at audio rate. kcf/acf -- the center frequency of the filter, or frequency position of the peak response. kbw/abw -- bandwidth of the filter (the Hz difference between the upper and lower half-power points). areson is a filter whose transfer functions is the complement of reson. Thus areson is a notch filter whose transfer functions represents the filtered out aspects of their complements. However, power scaling is not normalized in areson but remains the true complement of the corresponding unit. Thus an audio signal, filtered by parallel matching reson and areson units, would under addition simply reconstruct the original spectrum. This property is particularly useful for controlled mixing of different sources (see lpreson). Complex response curves such as those with multiple peaks can be obtained by using a bank of suitable filters in series. (The resultant response is the product of the component responses.) In such cases, the combined attenuation may result in a serious loss of signal power, but this can be regained by the use of balance. Examples Here is an example of the areson opcode. It uses the file areson.csd. Example of the areson opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also aresonk, atone, atonek, port, portk, reson, resonk, tone, tonek Credits Audio rate parameters introduced in version 6.02 October 2013. manual_src~dfsg/opcodes/vbapz.xml0000644000000000000000000001104312262561504016244 0ustar rootroot vbapz Signal Modifiers:Panning and Spatialization vbapz vbapz Writes a multi-channel audio signal to a ZAK array. Description Writes a multi-channel audio signal to a ZAK array. Syntax vbapz inumchnls, istartndx, asig, kazim [, kelev] [, kspread] Initialization inumchnls -- number of channels to write to the ZA array. Must be in the range 2 - 256. istartndx -- first index or position in the ZA array to use Performance asig -- audio signal to be panned kazim -- azimuth angle of the virtual source kelev (optional) -- elevation angle of the virtual source kspread (optional) -- spreading of the virtual source (range 0 - 100). If value is zero, conventional amplitude panning is used. When kspread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. The opcode vbapz is the multiple channel analog of the opcodes like vbap4, working on inumchnls and using a ZAK array for output. Please note that all vbap panning opcodes require the vbap system to be initialized using vbaplsinit. Examples See the entry for vbap8 for an example of usage of the vbap opcodes. Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. See Also vbap16, vbap16move, vbap4, vbap4move, vbap8, vbap8move, vbaplsinit, vbapzmove Credits &namejohn; University of Bath/Codemist Ltd. Bath, UK May 2000 New in Csound Version 4.07. Input parameters accept k-rate since Csund 5.09. manual_src~dfsg/opcodes/serialPrint.xml0000644000000000000000000000354312262561504017424 0ustar rootroot serialPrint Serial I/O serialPrint serialPrint Print data from a serial port. Description Print to the screen any bytes (up to 32k) in the input buffer. Note that these bytes will be cleared from the buffer. use this opcode mainly for debugging messages. If you want to mix debugging and other communication messages over the same port, you will need to manually parse the data with the serialRead opcode. Syntax serialPrint iPort Performance iPort -- port number optained from a serialBegin opcode. See Also serialBegin, serialEnd, serialWrite_i, serialWrite, serialRead, serialFlush. Credits Author: &namematt; 2011 New in version 5.14 manual_src~dfsg/opcodes/system.xml0000644000000000000000000000643012262561504016452 0ustar rootroot system_i system Miscellaneous system system Call an external program via the system call Description system and system_i call any external command understood by the operating system, similarly to the C function system(). system_i runs at i-time only, while system runs both at initialization and performance time. Syntax ires system_i itrig, Scmd, [inowait] kres system ktrig, Scmd, [knowait] Initialization Scmd -- command string itrig -- if greater than zero the opcode performs the printing; otherwise it is an null operation. Performance ktrig -- if greater than zero and different from the value on the previous control cycle the opcode performs the requested printing. Initially this previous value is taken as zero. inowait,knowait -- if given an non zero the command is run in the background and the command does not wait for the result. (default = 0) ires, kres -- the return code of the command in wait mode and if the command is run.In other cases returns zero. More than one system command (a script) can be executed with a single system opcode by using double braces strings {{ }}. This opcode is very system dependant, so should be used with extreme care (or not used) if platform neutrality is desired. Example Here is an example of the system_i opcode. It uses the file system.csd. Example of the system opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; 2007 New in version 5.06 manual_src~dfsg/opcodes/lua_exec.xml0000644000000000000000000000640012262561504016710 0ustar rootroot lua_exec Lua Opcodes lua_exec lua_exec Executes an arbitrary block of Lua code from the Csound orchestra. Description Executes an arbitrary block of Lua code from the Csound orchestra. The code is executed at initialization time, typically from the orchestra header. Syntax lua_exec Sluacode Initialization Sluacode -- A block of Lua code, of any length. Multi-line blocks may be enclosed in double braces (i.e. {{ }}). This code is evaluated once at initialization time, typically from the orchestra header. Global and local variables, functions, tables, and classes may be declared and defined. Objects defined at global Lua scope remain in scope throughout the performance, and are visible to any other Lua code in the same Csound thread. The running instance of Csound is stored as a Lua lightuserdata in a global variable csound. This can be passed to any Csound API function. One use of this would be to generate a score using Lua in the orchestra header, and schedule the events for performance using csoundInputMessage. By default, all objects defined in Lua are defined at global scope. In order to ensure that objects are confined to their own block of code, that is to ensure that the object is visible only in lexical scope, the object must be declared as local. This is the feature of Lua that beginners tend to find the most troublesome. Another thing to look out for is that Lua arrays use 1-based indexing, not the 0-based indexing used in C and many other programming languages. See Also lua_opdef, lua_opcall. Credits By: &namemichael; 2011 New in Csound version 5.13.2 manual_src~dfsg/opcodes/noteondur2.xml0000644000000000000000000000750512262561504017231 0ustar rootroot noteondur2 Real-time MIDI:Note Output noteondur2 noteondur2 Sends a noteon and a noteoff MIDI message both with the same channel, number and velocity. Description Sends a noteon and a noteoff MIDI message both with the same channel, number and velocity. Syntax noteondur2 ichn, inum, ivel, idur Initialization ichn -- MIDI channel number (1-16) inum -- note number (0-127) ivel -- velocity (0-127) idur -- how long, in seconds, this note should last. Performance noteondur2 (i-rate note on with duration) sends a noteon and a noteoff MIDI message both with the same channel, number and velocity. Noteoff message is sent after idur seconds are elapsed by the time noteondur2 was active. noteondur differs from noteondur2 in that noteondur truncates note duration when current instrument is deactivated by score or by real-time playing, while noteondur2 will extend performance time of current instrument until idur seconds have elapsed. In real-time playing, it is suggested to use noteondur also for undefined durations, giving a large idur value. Any number of noteondur2 opcodes can appear in the same Csound instrument, allowing chords to be played by a single instrument. Examples Here is an example of the noteondur2 opcode. It uses the file noteondur2.csd. Example of the noteondur2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This example generates notes for every note received on the MIDI input. It generates MIDI notes on csound's MIDI output, so be sure to connect something. See Also noteoff, noteon, noteondur, midion, midion2 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/vtabi.xml0000644000000000000000000000645112262561504016236 0ustar rootroot vtabi Vectorial:Tables vtabi vtabi Read vectors (from tables -or arrays of vectors). Description This opcode reads vectors from tables. Syntax vtabi indx, ifn, iout1 [, iout2, iout3, .... , ioutN ] Initialization indx - Index into f-table, either a positive number range matching the table length ifn - table number iout1...ioutN - output vector elements Performance This opcode is useful in all cases in which one needs to access sets of values associated to unique indexes (for example, multi-channel samples, STFT bin frames, spectral formants, p-field based scores etc.). The number of elements of each vector (length of the vector) is determined by the number of optional arguments on the right (iout1, iout2, iout3, .... ioutN). Notice that no wrap nor limit mode is implemented. So, if an index attempt to access to a zone not allocated by the table, Csound will probably crash. However this drawback can be easily avoided by using wrap or limit opcodes applied to indexes before using vtabi, in order to correct eventual out-of-range values. The vtab family is similar to vtable, but is much faster because interpolation is not available, table number cannot be changed after initialization, and only raw indexing is supported. Notice that vtabi's output arguments are placed at the right of the opcode name, differently from usual (this style is already used in other opcodes using undefined lists of output arguments such as fin or trigseq). Examples For an example of the vtabi opcode usage, see vtablei. See also vtabk, vtaba, vtablei, vtablewi, vtabwi, Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/OSCsend.xml0000644000000000000000000000701012262561504016417 0ustar rootroot OSCsend OSC OSCsend OSCsend Sends data to other processes using the OSC protocol Description Uses the OSC protocol to send message to other OSC listening processes. Syntax OSCsend kwhen, ihost, iport, idestination, itype [, kdata1, kdata2, ...] Initialization ihost -- a string that is the intended host computer domain name. An empty string is interpreted as the current computer. iport -- the number of the port that is used for the communication. idest -- a string that is the destination address. This takes the form of a file name with directories. Csound just passes this string to the raw sending code and makes no interpretation. itype -- a string that indicates the types of the optional arguments that are read at k-rate. The string can contain the characters "bcdfilmst" which stand for Boolean, character, double, float, 32-bit integer, 64-bit integer, MIDI, string and timestamp. Performance kwhen -- a message is sent whenever this value changes. A message will always be sent on the first call. The data is taken from the k-values that follow the format string. In a similar way to a printf format, the characters in order determine how the argument is interpreted. Note that a time stamp takes two arguments. Example The example shows a simple instrument, which when called, sends a group of 3 messages to a computer called "xenakis", on port 7770, to be read by a process that recognises /foo/bar as its address. instr 1 OSCsend 1, "xenakis.cs.bath.ac.uk",7770, "/foo/bar", "sis", "FOO", 42, "bar" endin See the entry for OSClisten, for an example of send/recieve usage using OSC. See Also OSClisten, OSCinit More information on this opcode: http://www.youtube.com/watch?v=JX1C3TqP_9Y , made by Andrés Cabrera Credits Author: &namejohn; 2005 manual_src~dfsg/opcodes/loscilx.xml0000644000000000000000000000232512262561504016602 0ustar rootroot loscilx Signal Generators:Sample Playback loscilx loscilx Loop oscillator. Description This file is currently a stub, but the syntax should be correct. Syntax ar1 [, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, ar13, ar14, \ ar15, ar16] loscilx xamp, kcps, ifn \ [, iwsize, ibas, istrt, imod1, ibeg1, iend1] See Also sndload loscil Credits Written by &nameistvan;. 2006 New in Csound 5.03 manual_src~dfsg/opcodes/JackoMidiInConnect.xml0000644000000000000000000000625412262561504020565 0ustar rootroot JackoMidiInConnect Jacko Opcodes JackoMidiInConnect JackoMidiInConnect Creates a MIDI connection from a Jack port to Csound. Description In the orchestra header, creates a MIDI connection from an external Jack MIDI output port to this instance of Csound. Syntax JackoMidiInConnect SexternalPortName, ScsoundPortName Initialization SexternalPortName -- The full name ("clientname:portname") of an external Jack MIDI output port. ScsoundPortName -- The short name ("portname") of the internal Jack MIDI input port. Must be used in conjunction with the -M0 -+rtmidi=null Csound command-line options. Can be used in with the MIDI inter-operability command-line options and/or opcodes to enable the use of ordinary Csound instrument definitions to render external scores or MIDI sequences. Note that Csound can connect to ALSA ports through Jack, but in that case you will have to identify the port by its alias in the JackInfo printout. Performance The actual MIDI events will be received in the regular Csound way, i.e. through a MIDI driver and the sensevents mechanism, rather than through a Jack input port opcode. The granularity of timing is Csound's kperiod. See Also JackoInfo, JackoInfo, JackoFreewheel, JackoAudioOutConnect, JackoAudioOutConnect, JackoMidiOutConnect, JackoOn, JackoAudioIn, JackoAudioOut, JackoMidiOut, JackoNoteOut, JackoTransport. Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/subtracts.xml0000644000000000000000000001022112262561504017131 0ustar rootroot - Mathematical Operations:Arithmetic and Logic Operations - - Subtraction operator. Description Arithmetic operators perform operations of change-sign (negate), don't-change-sign, logical AND logical OR, add, subtract, multiply and divide. Note that a value or an expression may fall between two of these operators, either of which could take it as its left or right argument, as in a + b * c. In such cases three rules apply: 1. * and / bind to their neighbors more strongly than + and −. Thus the above expression is taken as a + (b * c) with * taking b and c and then + taking a and b * c. 2. + and − bind more strongly than &&, which in turn is stronger than ||: a && b - c || d is taken as (a && (b - c)) || d 3. When both operators bind equally strongly, the operations are done left to right: a - b - c is taken as (a - b) - c Parentheses may be used as above to force particular groupings. Syntax −a (no rate restriction) a − b (no rate restriction) where the arguments a and b may be further expressions. Arguments The arguments of − can be scalar values or k-rate one dimensional arrays (vectors), or any combination. If one of the arguments is an array, so is the value. Examples Here is an example of the - operator. It uses the file subtracts.csd. Example of the - operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: sutain time= note duration - attack - decay --> 3.0-0.1-0.2 = 2.7 sutain time= note duration - attack - decay --> 3.0-0.5-1.5 = 1.0 sutain time= note duration - attack - decay --> 5.0-4.0-0.5 = 0.5 See Also +, &&, ||, *, /, ˆ, % Credits Arithmetic on vectors is new in version 6.00 manual_src~dfsg/opcodes/waveset.xml0000644000000000000000000000454412262561504016610 0ustar rootroot waveset Signal Generators:Sample Playback waveset waveset A simple time stretch by repeating cycles. Description A simple time stretch by repeating cycles. Syntax ares waveset ain, krep [, ilen] Initialization ilen (optional, default=0) -- the length (in samples) of the audio signal. If ilen is set to 0, it defaults to half the given note length (p3). Performance ain -- the input audio signal. krep -- the number of times the cycle is repeated. The input is read and each complete cycle (two zero-crossings) is repeated krep times. There is an internal buffer as the output is clearly slower that the input. Some care is taken if the buffer is too short, but there may be strange effects. Examples Here is an example of the waveset opcode. It uses the file waveset.csd. Example of the waveset opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; February 2001 New in version 4.11 manual_src~dfsg/opcodes/slider64.xml0000644000000000000000000001130612262561504016560 0ustar rootroot slider64 Real-time MIDI:Slider Banks slider64 slider64 Creates a bank of 64 different MIDI control message numbers. Description Creates a bank of 64 different MIDI control message numbers. Syntax i1,...,i64 slider64 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum64, imin64, imax64, init64, ifn64 k1,...,k64 slider64 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum64, imin64, imax64, init64, ifn64 Initialization i1 ... i64 -- output values ichan -- MIDI channel (1-16) ictlnum1 ... ictlnum64 -- MIDI control number (0-127) imin1 ... imin64 -- minimum values for each controller imax1 ... imax64 -- maximum values for each controller init1 ... init64 -- initial value for each controller ifn1 ... ifn64 -- function table for conversion for each controller Performance k1 ... k64 -- output values slider64 is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider64 allows a bank of 64 different MIDI control message numbers. As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. In the i-rate version of slider64, there is not an initial value input argument, because the output is gotten directly from current status of internal controller array of Csound. See Also s16b14, s32b14, slider16, slider16f, slider32, slider32f, slider64f slider8, slider8f Credits Author: &namegabriel; Italy December 1998 New in Csound version 3.50 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/strlen.xml0000644000000000000000000000200512262561504016427 0ustar rootroot strlen Strings:Manipulation strlen strlen Return the length of a string Description Return the length of a string, or zero if it is empty. strlen runs at init time only. Syntax ilen strlen Sstr See also strlenk Credits Author: Istvan Varga 2006 New in version 5.02 manual_src~dfsg/opcodes/pvsbufread.xml0000644000000000000000000001345112262561504017270 0ustar rootroot pvsbufread Spectral Processing:Streaming pvsbufread pvsbufread This opcode reads a circular buffer of f-signals (streaming PV signals). Description This opcode reads from a circular buffer of length ilen (secs), taking a handle for the buffer and a time pointer, which holds the current read position (also in seconds). It is used in conjunction with a pvsbuffer opocde. Reading is circular, wrapping around at the end of the buffer. Syntax fsig pvsbufread ktime, khandle[, ilo, ihi, iclear] Initialisation ilo, ihi -- set the lowest and highest freqs to be read from the buffer (defaults to 0, Nyquist). iclear -- set to 1 to clear output fsig before every write (default 1), 0 tells the opcode not to clear the output fsig. This is relevant when writing to subsets of an fsig frame using ilo, ihi. Performance fsig -- output pv stream ktime -- time position of reading pointer (in secs). khandle -- handle identifying the buffer to be read. When using k-rate handles, it is important to initialise the k-rate variable to a given existing handle. When changing buffers, fsig buffers need to be compatible (same fsig format). With this opcode and pvsbuffer, it is possible to, among other things: time-stretch/compress a fsig stream, by reading it at different rates delay a fsig or portions of it. 'brassage' two or more fsigs by switching buffers, since the reading handles are k-rate. Note that, when using k-rate handles, it is important to initialise the k-rate variable to a given handle (so that the fsig initialisation can take place) and it is only possible to switch handles between compatible fsig buffers (with the same fftsize and overlap). It is important that the handle value passed to pvsbufread is valid and was created by pvsbuffer. Csound will crash with invalid handles. Examples Here is an example of the pvsbufread opcode. It does 'brassage' by switching between two buffers. Example of the pvsbufread opcode See the sections Real-time Audio and Command Line Flags for more information on using command line flags. fsig1 pvsanal asig1, 1024, 256, 1024, 1 fsig2 pvsanal asig2, 1024, 256, 1024, 1 ibuf1, kt1 pvsbuffer fsig1, 10 ; 10-sec buf with fsig1 ibuf2, kt2 pvsbuffer fsig2, 7 ; 7-sec buf with fsig2 khan init ibuf1 ; initialise handle to buf1 if ktrig > 0 then ; switch buffers according to trigger khan = ibuf2 else khan = ibuf1 endif fsb pvsbufread kt1, khan ; read buffer Here is an example of the pvsbufread opcode. It uses the file pvsbufread.csd. Example of the pvsbufread opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsbuffer, pvsadsyn Credits Author: &namevictor; July 2007 manual_src~dfsg/opcodes/voice.xml0000644000000000000000000001123712262561504016234 0ustar rootroot voice Signal Generators:Models and Emulations voice voice An emulation of a human voice. Description An emulation of a human voice. Syntax ares voice kamp, kfreq, kphoneme, kform, kvibf, kvamp, ifn, ivfn Initialization ifn, ivfn -- two table numbers containing the carrier waveform and the vibrato waveform. The files impuls20.aiff, ahh.aiff, eee.aiff, or ooo.aiff are suitable for the first of these, and a sine wave for the second. These files are available from ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. It can be varied in performance. kphoneme -- an integer in the range 0 to 16, which select the formants for the sounds: eee, ihh, ehh, aaa, ahh, aww, ohh, uhh, uuu, ooo, rrr, lll, mmm, nnn, nng, ngg. At present the phonemes fff, sss, thh, shh, xxx, hee, hoo, hah, bbb, ddd, jjj, ggg, vvv, zzz, thz, zhh are not available (!) kform -- Gain on the phoneme. values 0.0 to 1.2 recommended. kvibf -- frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp -- amplitude of the vibrato Examples Here is an example of the voice opcode. It uses the file voice.csd, and impuls20.aiff. Example of the voice opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/imagesize.xml0000644000000000000000000000534112262561504017103 0ustar rootroot imagesize Image Processing Opcodes imagesize imagesize Return the width and height of a previously opened or created image. Description Return the width and height of a previously opened or created image. An image can be loaded with imageload. An empty image can be created with imagecreate. Syntax iwidth, iheight imagesize iimagenum Initialization iimagenum -- the reference of the image.. It should be a value returned by imageload or imagecreate. iwidth -- image width. iheight -- image height. Examples Here is an example of the imagesize opcode. It uses the file imageopcodes.csd. Example of the imagesize opcode. Its output should include a line like this: imagewidth = 180.000000 pixels, imageheigth = 135.000000 pixels See Also imageload, imagecreate, imagesave, imagegetpixel, imagesetpixel, imagefree Credits Author: Cesare Marilungo New in version 5.08 manual_src~dfsg/opcodes/flcolor2.xml0000644000000000000000000000674612262561504016662 0ustar rootroot FLcolor2 FLTK:Appearance FLcolor2 FLcolor2 A FLTK opcode that sets the secondary (selection) color. Description FLcolor2 is the same of FLcolor except it affects the secondary (selection) color. Syntax FLcolor2 ired, igreen, iblue Initialization ired -- The red color of the target widget. The range for each RGB component is 0-255 igreen -- The green color of the target widget. The range for each RGB component is 0-255 iblue -- The blue color of the target widget. The range for each RGB component is 0-255 Performance These opcodes modify the appearance of other widgets. There are two types of such opcodes: those that don't contain the ihandle argument which affect all subsequently declared widgets, and those with ihandle which affect only a target widget previously defined. FLcolor2 is the same of FLcolor except it affects the secondary (selection) color. Setting it to -1 turns off the influence of FLcolor2 on subsequent widgets. A value of -2 (or -3) makes all next widget secondary colors randomly selected. The difference is that -2 selects a light random color, while -3 selects a dark random color. See Also FLcolor, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/cpspch.xml0000644000000000000000000001417512262561504016413 0ustar rootroot cpspch Pitch Converters:Functions cpspch cpspch Converts a pitch-class value to cycles-per-second. Description Converts a pitch-class value to cycles-per-second. Syntax cpspch (pch) (init- or control-rate args only) where the argument within the parentheses may be a further expression. Performance cpspch and its related opcodes are really value converters with a special function of manipulating pitch data. Data concerning pitch and frequency can exist in any of the following forms: Pitch and Frequency Values Name Abbreviation octave point pitch-class (8ve.pc) pch octave point decimal oct cycles per second cps Midi note number (0-127) midinn
The first two forms consist of a whole number, representing octave registration, followed by a specially interpreted fractional part. For pch, the fraction is read as two decimal digits representing the 12 equal-tempered pitch classes from .00 for C to .11 for B. For oct, the fraction is interpreted as a true decimal fractional part of an octave. The two fractional forms are thus related by the factor 100/12. In both forms, the fraction is preceded by a whole number octave index such that 8.00 represents Middle C, 9.00 the C above, etc. Midi note number values range between 0 and 127 (inclusively) with 60 representing Middle C, and are usually whole numbers. Thus A440 can be represented alternatively by 440 (cps), 69 (midinn), 8.09 (pch), or 8.75 (oct). Microtonal divisions of the pch semitone can be encoded by using more than two decimal places. The mnemonics of the pitch conversion units are derived from morphemes of the forms involved, the second morpheme describing the source and the first morpheme the object (result). Thus cpspch(8.09) will convert the pitch argument 8.09 to its cps (or Hertz) equivalent, giving the value of 440. Since the argument is constant over the duration of the note, this conversion will take place at i-time, before any samples for the current note are produced. By contrast, the conversion cpsoct(8.75 + k1) which gives the value of A440 transposed by the octave interval k1. The calculation will be repeated every k-period since that is the rate at which k1 varies. Note The conversion from pch, oct, or midinn into cps is not a linear operation but involves an exponential process that could be time-consuming when executed repeatedly. Csound now uses a built-in table lookup to do this efficiently, even at audio rates. Because the table index is truncated without interpolation, pitch resolution when using one of these opcodes is limited to 8192 discrete and equal divisions of the octave, and some pitches of the standard 12-tone equally-tempered scale are very slightly mistuned (by at most 0.15 cents). If you need more precision in the calculation, use cps2pch or cpsxpch instead. Examples Here is an example of the cpspch opcode. It uses the file cpspch.csd. Example of the cpspch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: icps = 277.167 instr 1: icps = 293.656 instr 1: icps = 311.101 instr 1: icps = 54.995 See Also cps2pch, cpsoct, cpsxpch, octcps, octpch, pchoct, cpsmidinn, octmidinn, pchmidinn manual_src~dfsg/opcodes/pvstanal.xml0000644000000000000000000001036012262561504016753 0ustar rootroot pvstanal Spectral Processing:Streaming pvstanal pvstanal Phase vocoder analysis processing with onset detection/processing. Description pvstanal implements phase vocoder analysis by reading function tables containing sampled-sound sources, with GEN01, and pvstanal will accept deferred allocation tables. This opcode allows for time and frequency-independent scaling. Time is advanced internally, but controlled by a tempo scaling parameter; when an onset is detected, timescaling is momentarily stopped to avoid smearing of attacks. The quality of the effect is generally improved with phase locking switched on. pvstanal will also scale pitch, independently of frequency, using a transposition factor (k-rate). Syntax fsig pvstanal ktimescal, kamp, kpitch, ktab, [kdetect, kwrap, ioffset,ifftsize, ihop, idbthresh] Initialization ifftsize -- FFT size (power-of-two), defaults to 2048. ihop -- hopsize, defaults to 512 ioffset -- startup read offset into table, in secs. idbthresh -- threshold for onset detection, based on dB power spectrum ratio between two successive windows. A detected ratio above it will cancel timescaling momentarily, to avoid smearing (defaults to 1). By default anything more than a 1 dB inter-frame power difference will be detected as an onset. Performance ktimescal -- timescaling ratio, < 1 stretch, > 1 contract. kamp -- amplitude scaling kpitch -- grain pitch scaling (1=normal pitch, < 1 lower, > 1 higher; negative, backwards) kdetect -- 0 or 1, to switch onset detection/processing. The onset detector checks for power difference between analysis windows. If more than what has been specified in the dbthresh parameter, an onset is declared. It suspends timescaling momentarily so the onsets are not modified. ktab -- source signal function table. Deferred-allocation tables (see GEN01) are accepted, but the opcode expects a mono source. Tables can be switched at k-rate. kwrap -- 0 or 1, to switch on/off table wrap-around read (default to 1) Examples Here is an example of the pvstanal opcode. It uses the file pvstanal.csd. Example of the pvstanal opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; February 2010 New plugin in version 5.13 February 2005. manual_src~dfsg/opcodes/vmult_i.xml0000644000000000000000000001043512262561504016605 0ustar rootroot vmult_i Vectorial:Scalar operations vmult_i vmult_i Multiplies a vector in a table by a scalar value. Description Multiplies a vector in a table by a scalar value. Syntax vmult_i ifn, ival, ielements [, idstoffset] Initialization ifn - number of the table hosting the vectorial signal to be processed ival - scalar value to be multiplied ielements - number of elements of the vector idstoffset - index offset for the destination table Performance vmult_i multiplies each element of the vector contained in the table ifn by ival, starting from table index idstoffset. This enables you to process a specific section of a table by specifying the offset and the number of elements to be processed. Offset is counted starting from 0, so if no offset is specified (or set to 0), the table will be modified from the beginning. This opcode runs only on initialization, there is a k-rate version of this opcode called vmult. Negative values for idstoffset are valid. Elements from the vector that are outside the table, will be discarded, and they will not wrap around the table. In all these opcodes, the resulting vectors are stored in ifn, overriding the intial vectors. If you want to keep initial vector, use vcopy or vcopy_i to copy it in another table. All these operators are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2, etc. They can also be useful in conjunction with the spectral opcodes pvsftw and pvsftr. Examples Here is an example of the vmult_i opcode. It uses the file vmult_i.csd. Example of the vmult_i opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also vadd, vadd, vmult, vpow_i, vpow, vexp_i and vexp. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. Example by Andres Cabrera. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/vadd.xml0000644000000000000000000001335312262561504016046 0ustar rootroot vadd Vectorial:Scalar operations vadd vadd Adds a scalar value to a vector in a table. Description Adds a scalar value to a vector in a table. Syntax vadd ifn, kval, kelements [, kdstoffset] [, kverbose] Initialization ifn - number of the table hosting the vectorial signal to be processed Performance kval - scalar value to be added kelements - number of elements of the vector kdstoffset - index offset for the destination table (Optional, default=0) kverbose - Selects whether or not warnings are printed (Optional, default=0) vadd adds the value of kval to each element of the vector contained in the table ifn, starting from table index kdstoffset. This enables you to process a specific section of a table by specifying the offset and the number of elements to be processed. Offset is counted starting from 0, so if no offset is specified (or set to 0), the table will be modified from the beginning. Note that this opcode runs at k-rate so the value of kval is added every control period. Use with care or you will end up with very large numbers (or use vadd_i). These opcodes (vadd, vmult, vpow and vexp) perform numeric operations between a vectorial control signal (hosted by the table ifn), and a scalar signal (kval). Result is a new vector that overrides old values of ifn. All these opcodes work at k-rate. Negative values for kdstoffset are valid. Elements from the vector that are outside the table, will be discarded, and they will not wrap around the table. If the optional kverbose argument is different to 0, the opcode will print warning messages every k-pass if table lengths are exceeded. In all these opcodes, the resulting vectors are stored in ifn, overriding the intial vectors. If you want to keep initial vector, use vcopy or vcopy_i to copy it in another table. All these operators are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2, etc. They can also be useful in conjunction with the spectral opcodes pvsftw and pvsftr. Please note that the elements argument has changed in version 5.03 from i-rate to k-rate. This will change the opcode's behavior in the unusual cases where the i-rate variable ielements is changed inside the instrument, for example in: instr 1 ielements = 10 vadd 1, 1, ielements ielements = 20 vadd 2, 1, ielements turnoff endin Examples Here is an example of the vadd opcode. It uses the file vadd.csd. Example of the vadd opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also vadd_i, vmult, vpow and vexp. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/flprintk2.xml0000644000000000000000000000445012262561504017041 0ustar rootroot FLprintk2 FLTK:Other FLprintk2 FLprintk2 A FLTK opcode that prints a new value every time a control-rate variable changes. Description FLprintk2 is similar to FLprintk but shows a k-rate variable's value only when it changes. Syntax FLprintk2 kval, idisp Initialization idisp -- a handle value that was output from a previous instance of the FLvalue opcode to display the current value of the current valuator in the FLvalue widget itself. If the user doesn't want to use this feature that displays current values, it must be set to a negative number by the user. Performance kval -- k-rate signal to be displayed. FLprintk2 is similar to FLprintk, but shows the k-rate variable's value only each time it changes. Useful for monitoring MIDI control changes when using sliders. It should be used for debugging purposes only, since it slows-down performance. See Also FLbox, FLbutBank, FLbutton, FLprintk, FLvalue Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/inq.xml0000644000000000000000000000476112262561504015722 0ustar rootroot inq Signal I/O:Signal Input inq inq Reads quad audio data from an external device or stream. Description Reads quad audio data from an external device or stream. Warning This opcode is designed to be used only with orchestras that have nchnls_i=4. Doing so with orchestras with nchnls_i > 4 will cause incorrect audio input. Syntax ar1, ar2, ar3, a4 inq Performance Reads quad audio data from an external device or stream. If the command-line -i flag is set, sound is read continuously from the audio input stream (e.g. stdin or a soundfile) into an internal buffer. Any number of these opcodes can read freely from this buffer. Examples Here is an example of the inq opcode. It uses the file inq.csd. Example of the inq opcode. See Also diskin, in, inh, inh, ino, ins, soundin Credits Authors: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 manual_src~dfsg/opcodes/strrindex.xml0000644000000000000000000000346712262561504017157 0ustar rootroot strrindex Strings:Manipulation strrindex strrindex Return the position of the last occurence of a string in another string Description Return the position of the last occurence of S2 in S1, or -1 if not found. If S2 is empty, the length of S1 is returned. strrindex runs at init time only. Syntax ipos strrindex S1, S2 Examples Here is an example of the strrindex opcode. It uses the file strrindex.csd. Example of the strrindex opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also strrindexk Credits Author: Istvan Varga 2006 New in version 5.02 manual_src~dfsg/opcodes/vtaba.xml0000644000000000000000000000650012262561504016221 0ustar rootroot vtaba Vectorial:Tables vtaba vtaba Read vectors (from tables -or arrays of vectors). Description This opcode reads vectors from tables at a-rate. Syntax vtaba andx, ifn, aout1 [, aout2, aout3, .... , aoutN ] Initialization ifn - table number Performance andx - Index into f-table, either a positive number range matching the table length aout1...aoutN - output vector elements This opcode is useful in all cases in which one needs to access sets of values associated to unique indexes (for example, multi-channel samples, STFT bin frames, spectral formants, p-field based scores etc.). The number of elements of each vector (length of the vector) is determined by the number of optional arguments on the right (aout1, aout2, aout3, .... aoutN). Notice that no wrap nor limit mode is implemented. So, if an index attempt to access to a zone not allocated by the table, Csound will probably crash. However this drawback can be easily avoided by using wrap or limit opcodes applied to indexes before using vtaba, in order to correct eventual out-of-range values. The vtab family is similar to the vtable family, but is much faster because interpolation is not available, table number cannot be changed after initialization, and only raw indexing is supported. Notice that vtaba's output arguments are placed at the right of the opcode name, differently from usual (this style is already used in other opcodes using undefined lists of output arguments such as fin or trigseq). Examples The usage of vtaba is similar to vtablek. See also vtabk, vtabi, vtablea, vtablewa, vtabwa, Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/filevalid.xml0000644000000000000000000000464212262561504017070 0ustar rootroot filevalid Signal I/O:Soundfile Queries filevalid filevalid Checks that a file can be used. Description Returns 1 if the sound file is valid, or 0 if not. Syntax ir filevalid ifilcod Initialization ifilcod -- sound file to be queried Performance filevalid returns 1 if the sound file ifilcod can be used. Examples Here is an example of the filevalid opcode. It uses the file filevalid.csd. Example of the filevalid opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output should include a line like this: Audiofile 'frox.wav' does not exist! See Also filebit, filelen, filenchnls, filepeak, filesr Credits Author: &namematt; July 2010 New in Csound version 5.13 manual_src~dfsg/opcodes/vdelayxwq.xml0000644000000000000000000001035212262561504017150 0ustar rootroot vdelayxwq Signal Modifiers:Delay vdelayxwq vdelayxwq Variable delay opcodes with high quality interpolation. Description Variable delay opcodes with high quality interpolation. Syntax aout1, aout2, aout3, aout4 vdelayxwq ain1, ain2, ain3, ain4, adl, \ imd, iws [, ist] Initialization imd -- max. delay time (seconds) iws -- interpolation window size (see below) ist (optional) -- skip initialization if not zero Performance ain1, ain2, ain3, ain4 -- input audio signals aout1, aout2, aout3, aout4 -- output audio signals adl -- delay time in seconds These opcodes use high quality (and slow) interpolation, that is much more accurate than the currently available linear and cubic interpolation. The iws parameter sets the number of input samples used for calculating one output sample (allowed values are any integer multiply of 4 in the range 4 - 1024); higher values mean better quality and slower speed. The vdelayxw opcodes change the position of the write tap in the delay line (unlike all other delay ugens that move the read tap), and are most useful for implementing Doppler effects where the position of the listener is fixed, and the sound source is moving. The multichannel opcodes (eg. vdelayxq) allow delaying 2 or 4 variables at once (stereo or quad signals); this is much more efficient than using separate opcodes for each channel. Notes Delay time is measured in seconds (unlike in vdelay and vdelay3), and must be a-rate. The minimum allowed delay is iws/2 samples. Using the same variables as input and output is allowed in these opcodes. In vdelayxw*, changing the delay time has some effects on output volume: a = 1 / (1 + dt) where a is the output gain, and dt is the change of delay time per seconds. These opcodes are best used in the double-precision version of Csound. Examples Here is an example of the use of the vdelayxwq opcode. It uses the file vdelayxwq.csd. Example of the <emphasis>vdelayxwq</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vdelayx, vdelayxq, vdelayxs, vdelayxw, vdelayxws manual_src~dfsg/opcodes/midremot.xml0000644000000000000000000001005112262561504016740 0ustar rootroot midremot Remote Opcodes midremot midremot An opcode which can be used to implement a remote midi orchestra. This opcode will send midi events from a source machine to one destination. Description With the midremot and midglobal opcodes you are able to perform instruments on remote machines and control them from a master machine. The remote opcodes are implemented using the master/client model. All the machines involved contain the same orchestra but only the master machine contains the information of the midi score. During the performance the master machine sends the midi events to the clients. The midremot opcode will send events from a source machine to one destination if you want to send events to many destinations (broadcast) use the midglobal opcode instead. These two opcodes can be used in combination. Syntax midremot idestination, isource, instrnum [,instrnum...] Initialization idestination -- a string that is the intended host computer (e.g. 192.168.0.100). This is the destination host which receives the events from the given instrument. isource -- a string that is the intended host computer (e.g. 192.168.0.100). This is the source host which generates the events of the given instrument and sends it to the address given by idestination. instrnum -- a list of instrument numbers which will be played on the destination machine Examples Here is an example of the midremot opcode. It uses the files insremot.csd. Example of the insremot opcode. The example shows a Bach fugue played on 4 remote computers. The master machine is named "192.168.1.100", client1 "192.168.1.101" and so on. Start the clients on each machine (they will be waiting to receive the events from the master machine) and then start the master. Here is the command on linux to start a client (csound -dm0 -odac -+rtaudio=alsa midremot.csd -+rtmidi=Null), and the command on the master machine will look like this (csound -dm0 -odac -+rtaudio=alsa midremot.csd -F midremot.mid). See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also insglobal, insremot, midglobal, remoteport Credits Author: Simon Schampijer 2006 New in version 5.03 manual_src~dfsg/opcodes/dssilist.xml0000644000000000000000000000317012262561504016762 0ustar rootroot dssilist Plugin Hosting:DSSI and LADSPA dssilist dssilist Lists all available DSSI and LADSPA plugins. Syntax dssilist Description dssilist checks the variables DSSI_PATH and LADSPA_PATH and lists all plugins available in all plugin libraries there. LADSPA and DSSI libraries may contain more than one plugin which must be referenced by the index provided by dssilist. This opcode produces a long printout which may interrupt realtime audio output, so it should be run at the start of a performance. Examples Here is an example of the dssilist opcode. It uses the file dssilist.csd. Example of the dssilist opcode. (Remember to change the Library name) Credits 2005 By: &nameandres; Uses code from Richard Furse's LADSPA sdk. manual_src~dfsg/opcodes/outq1.xml0000644000000000000000000000615112262561504016177 0ustar rootroot outq1 Signal I/O:Signal Output outq1 outq1 Writes samples to quad channel 1 of an external device or stream. Description Writes samples to quad channel 1 of an external device or stream. Syntax outq1 asig Performance Sends audio samples to an accumulating output buffer (created at the beginning of performance) which serves to collect the output of all active instruments before the sound is written to disk. There can be any number of these output units in an instrument. The type (mono, stereo, quad, hex, or oct) should agree with nchnls. But as of version 3.50, Csound will attempt to change an incorrect opcode to agree with the nchnls statement. Opcodes can be chosen to direct sound to any particular channel: outs1 sends to stereo channel 1, outq3 to quad channel 3, etc. Examples Here is an example of the outq1 opcode. It uses the file outq1.csd. Example of the outq1 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also out, outh, outo, outq, outq2, outq3, outq4, outs, outs1, outs2, soundout Credits Author: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 manual_src~dfsg/opcodes/JackoAudioInConnect.xml0000644000000000000000000000476012262561504020744 0ustar rootroot JackoAudioInConnect Jacko Opcodes JackoAudioInConnect JackoAudioInConnect Creates an audio connection from a Jack port to Csound. Description In the orchestra header, creates an audio connection from an external Jack audio output port to a Jack audio input port inside this instance of Csound. Syntax JackoAudioInConnect SexternalPortName, ScsoundPortName Initialization SexternalPortName -- The full name ("clientname:portname") of an external Jack audio output port. ScsoundPortName -- The short name ("portname") of the internal Jack audio input port. Performance The actual audio must be read with the JackoAudioIn opcode. See Also JackoInfo, JackoInfo, JackoFreewheel, JackoAudioOutConnect, JackoMidiInConnect, JackoMidiOutConnect, JackoOn, JackoAudioIn, JackoAudioOut, JackoMidiOut, JackoNoteOut, JackoTransport. Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/ampmidi.xml0000644000000000000000000000613112262561504016544 0ustar rootroot ampmidi Real-time MIDI:Converters ampmidi ampmidi Get the velocity of the current MIDI event. Description Get the velocity of the current MIDI event. Syntax iamp ampmidi iscal [, ifn] Initialization iscal -- i-time scaling factor ifn (optional, default=0) -- function table number of a normalized translation table, by which the incoming value is first interpreted. The default value is 0, denoting no translation. Performance Get the velocity of the current MIDI event, optionally pass it through a normalized translation table, and return an amplitude value in the range 0 - iscal. Examples Here is an example of the ampmidi opcode. It uses the file ampmidi.csd. Example of the ampmidi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also aftouch, cpsmidi, cpsmidib, midictrl, notnum, octmidi, octmidib, pchbend, pchmidi, pchmidib, veloc Credits Author: &namebarry; - &namemike; MIT - Mills May 1997 manual_src~dfsg/opcodes/ftcps.xml0000644000000000000000000000453212262561504016246 0ustar rootroot ftcps Table Control:Table Queries ftcps ftcps Returns the base frequency of a stored function table in Hz. Description Returns the base frequency of a stored function table in Hz.. Syntax ftcps(x) (init-rate args only) Performance Returns the base frequency of stored function table, number x. ftcps is designed for tables storing audio waveforms read from files (see GEN01). ftcps returns -1 in case of an error (no base frequency set in table or non-existent table). Examples Here is an example of the ftcps opcode. Example of the ftcps opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also ftchnls, ftlptim, ftsr, nsamp Credits Author: Victor Lazzarini 2010 Example written by Victor Lazzarini manual_src~dfsg/opcodes/nchnls.xml0000644000000000000000000000504312262561504016412 0ustar rootroot nchnls Orchestra Syntax:Header nchnls nchnls Sets the number of channels of audio output. Description These statements are global value assignments, made at the beginning of an orchestra, before any instrument block is defined. Their function is to set certain reserved symbol variables that are required for performance. Once set, these reserved symbols can be used in expressions anywhere in the orchestra. Syntax nchnls = iarg Initialization nchnls = (optional) -- set number of channels of audio output to iarg. (1 = mono, 2 = stereo, 4 = quadraphonic.) The default value is 1 (mono). In addition, any global variable can be initialized by an init-time assignment anywhere before the first instr statement. All of the above assignments are run as instrument 0 (i-pass only) at the start of real performance. Examples Here is an example of the nchnls opcode. It uses the file nchnls.csd. Example of the nchnls opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also kr, ksmps, sr manual_src~dfsg/opcodes/crunch.xml0000644000000000000000000000732012262561504016407 0ustar rootroot crunch Signal Generators:Models and Emulations crunch crunch Semi-physical model of a crunch sound. Description crunch is a semi-physical model of a crunch sound. It is one of the PhISEM percussion opcodes. PhISEM (Physically Informed Stochastic Event Modeling) is an algorithmic approach for simulating collisions of multiple independent sound producing objects. Syntax ares crunch iamp, idettack [, inum] [, idamp] [, imaxshake] Initialization iamp -- Amplitude of output. Note: As these instruments are stochastic, this is only a approximation. idettack -- period of time over which all sound is stopped inum (optional) -- The number of beads, teeth, bells, timbrels, etc. If zero, the default value is 7. idamp (optional) -- the damping factor, as part of this equation: damping_amount = 0.998 + (idamp * 0.002) The default damping_amount is 0.99806 which means that the default value of idamp is 0.03. The maximum damping_amount is 1.0 (no damping). This means the maximum value for idamp is 1.0. The recommended range for idamp is usually below 75% of the maximum value. imaxshake (optional) -- amount of energy to add back into the system. The value should be in range 0 to 1. Examples Here is an example of the crunch opcode. It uses the file crunch.csd. Example of the crunch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cabasa, sandpaper, sekere, stix Credits Author: &nameperry;, part of the PhOLIES (Physically-Oriented Library of Imitated Environmental Sounds) Adapted by &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.07 Added notes by &namerasmus; on May 2002. manual_src~dfsg/opcodes/islider8.xml0000644000000000000000000000077612262561504016660 0ustar rootroot islider8 islider8 islider8 Deprecated. Description Deprecated as of version 3.52. Use the slider8 opcode instead. manual_src~dfsg/opcodes/qinf.xml0000644000000000000000000000275712262561504016073 0ustar rootroot qinf Mathematical Operations:Mathematical Functions qinf qinf Questions whether the argument is a infinite number Description Returns the number of times the argument is not a number, with the sign of the first infinity. Syntax qinf(x) (no rate restriction) Examples Here is an example of the qinf opcode. It uses the file qinf.csd. Example of the qinf opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by &namejohn;. New in Csound 5.14 manual_src~dfsg/opcodes/STKShakers.xml0000644000000000000000000001106012262561504017103 0ustar rootroot STKShakers Signal Generators:STK Opcodes STKShakers STKShakers STKShakers is an instrument that simulates environmental sounds or collisions of multiple independent sound producing objects. Description STKShakers are a set of PhISEM and PhOLIES instruments: PhISEM (Physically Informed Stochastic Event Modeling) is an algorithmic approach for simulating collisions of multiple independent sound producing objects. It can simulate a Maraca, Sekere, Cabasa, Bamboo Wind Chimes, Water Drops, Tambourine, Sleighbells, and a Guiro. On PhOLIES (Physically-Oriented Library of Imitated Environmental Sounds) there is a similar approach for the synthesis of environmental sounds. It simulates of breaking sticks, crunchy snow (or not), a wrench, sandpaper, and more.. Syntax asignal STKShakers ifrequency, iamplitude, [kenerg, kv1[, kdecay, kv2[, kshake, kv3[, knum, kv4[, kres, kv5[, kinstr, kv6]]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kenerg -- controller 2, shake energy. Value range of kv1 is 0-127. kdecay -- controller 4, system decay. Value range of kv2 is 0-127. kshake -- controller 128, shake energy. Value range of kv3 is 0-127. knum -- controller 11, number of objects. Value range of kv4 is 0-127. kres -- controller 1, resonance frequency. Value range of kv5 is 0-127. kinstr -- controller 1071, instrument selection (Maraca = 0, Cabasa = 1, Sekere = 2, Guiro = 3, Water Drops = 4, Bamboo Chimes = 5, Tambourine = 6, Sleigh Bells = 7, Sticks = 8, Crunch = 9, Wrench = 10, Sand Paper = 11, Coke Can = 12, Next Mug = 13, Penny + Mug = 14, Nickle + Mug = 15, Dime + Mug = 16, Quarter + Mug = 17, Franc + Mug = 18, Peso + Mug = 19, Big Rocks = 20, Little Rocks = 21, Tuned Bamboo Chimes = 22). Value range of kv6 is 0-22. Note The code for this opcode is taken directly from the Shakers class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKShakers opcode. It uses the file STKShakerscsd. Example of the STKShakers opcode. Credits Author: &namemichael; (after Georg Essl) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/vmirror.xml0000644000000000000000000000470112262561504016625 0ustar rootroot vmirror Vectorial:Limiting and Wrapping vmirror vmirror Limiting and Wrapping Vectorial Signals Description 'Reflects' elements of vectorial control signals on thresholds. Syntax vmirror ifn, kmin, kmax, ielements Initialization ifn - number of the table hosting the vector to be processed ielements - number of elements of the vector Performance kmin - minimum threshold value kmax - maximum threshold value vmirror 'reflects' each element of corresponding vector if it exceeds low or high thresholds. These opcodes are similar to limit, wrap and mirror, but operate with a vectorial signal instead of with a scalar signal. Result overrides old values of ifn1, if these are out of min/max interval. If you want to keep input vector, use vcopy opcode to copy it in another table. All these opcodes are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Note: bmscan not yet available on Canonical Csound Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/fmpercfl.xml0000644000000000000000000000751612262561504016732 0ustar rootroot fmpercfl Signal Generators:FM Synthesis fmpercfl fmpercfl Uses FM synthesis to create a percussive flute sound. Description Uses FM synthesis to create a percussive flute sound. It comes from a family of FM sounds, all using 4 basic oscillators and various architectures, as used in the TX81Z synthesizer. Syntax ares fmpercfl kamp, kfreq, kc1, kc2, kvdepth, kvrate[, ifn1, ifn2, \ ifn3, ifn4, ivfn] Initialization All these opcodes take 5 tables for initialization. The first 4 are the basic inputs and the last is the low frequency oscillator (LFO) used for vibrato. These all default to a sine wave table. The initial waves should be: ifn1 -- sine wave ifn2 -- sine wave ifn3 -- sine wave ifn4 -- sine wave Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. kc1, kc2 -- Controls for the synthesizer: kc1 -- Total mod index kc2 -- Crossfade of two modulators Algorithm -- 4 kvdepth -- Vibrator depth kvrate -- Vibrator rate Examples Here is an example of the fmpercfl opcode. It uses the file fmpercfl.csd. Example of the fmpercfl opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also fmb3, fmbell, fmmetal, fmrhode, fmwurlie Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/specsum.xml0000644000000000000000000000541312262561504016605 0ustar rootroot specsum Spectral Processing:Non-Standard specsum specsum Sums the magnitudes across all channels of the spectrum. Description Sums the magnitudes across all channels of the spectrum. Syntax ksum specsum wsig [, interp] Initialization interp (optional, default-0) -- if non-zero, interpolate the output signal (koct or ksum). The default value is 0 (repeat the signal value between changes). Performance ksum -- the output signal. wsig -- the input spectrum. Sums the magnitudes across all channels of the spectrum. At each new frame of wsig, the magnitudes are summed and released as a scalar ksum signal. Between frames, the output is either repeated or interpolated at the k-rate. This unit produces a k-signal summation of the magnitudes present in the spectral data, and is thereby a running measure of its moment-to-moment overall strength. Examples ksum specsum wsig, 1 ; sum the spec bins, and ksmooth if ksum < 2000 kgoto zero ; if sufficient amplitude koct specptrk wsig ; pitch-track the signal kgoto contin zero: koct = 0 ; else output zero contin: See Also specdisp manual_src~dfsg/opcodes/chn.xml0000644000000000000000000001373512262561504015704 0ustar rootroot chn Signal I/O:Software Bus chn chn Declare a channel of the named software bus. Description Declare a channel of the named software bus, with setting optional parameters in the case of a control channel. If the channel does not exist yet, it is created, with an inital value of zero or empty string. Otherwise, the type (control, audio, or string) of the existing channel must match the declaration, or an init error occurs. The input/output mode of an existing channel is updated so that it becomes the bitwise OR of the previous and the newly specified value. Syntax chn_k Sname, imode[, itype, idflt, imin, ima, ix, iy, iwidth, iheight, Sattributes] chn_a Sname, imode chn_S Sname, imode Initialization imode -- sum of at least one of 1 for input and 2 for output. itype (optional, defaults to 0) -- channel subtype for control channels only. Possible values are: 0: default/unspecified (idflt, imin, and imax are ignored) 1: integer values only 2: linear scale 3: exponential scale idflt (optional, defaults to 0) -- default value, for control channels with non-zero itype only. Must be greater than or equal to imin, and less than or equal to imax. imin (optional, defaults to 0) -- minimum value, for control channels with non-zero itype only. Must be non-zero for exponential scale (itype = 3). imax (optional, defaults to 0) -- maximum value, for control channels with non-zero itype only. Must be greater than imin. In the case of exponential scale, it should also match the sign of imin. ix -- suggested x position for controller. iy -- suggested y position for controller. iwidth -- suggested width position for controller. iheight -- suggested height position for controller. Sattributes -- attributes for controller. Notes The channel parameters (imode, itype, idflt, imin, and imax) are only hints for the host application or external software accessing the bus through the API, and do not actually restrict reading from or writing to the channel in any way. Also, the initial value of a newly created control channel is zero, regardless of the setting of idflt. For communication with external software, using chnexport may be preferred, as it allows direct access to orchestra variables exported as channels of the bus, eliminating the need for using chnset and chnget to send or receive data. Performance chn_k, chn_a, and chn_S declare a control, audio, or string channel, respectively. Example The example shows the software bus being used as an asynchronous control signal to select a filter cutoff. It assumes that an external program that has access to the API is feeding the values. sr = 44100 kr = 100 ksmps = 1 chn_k "cutoff", 1, 3, 1000, 500, 2000 instr 1 kc chnget "cutoff" a1 oscil p4, p5, 100 a2 lowpass2 a1, kc, 200 out a2 endin Credits Author: &nameistvan; 2005 manual_src~dfsg/opcodes/vtablek.xml0000644000000000000000000001054412262561504016557 0ustar rootroot vtablek Vectorial:Tables vtablek vtablek Read vectors (from tables -or arrays of vectors). Description This opcode reads vectors from tables at k-rate. Syntax vtablek kndx, kfn, kinterp, ixmode, kout1 [, kout2, kout3, .... , koutN ] Initialization ixmode - index data mode. The default value is 0. == 0 index is treated as a raw table location, == 1 index is normalized (0 to 1). kinterp - switch beetween interpolated or non-interpolated output. 0 -> non-interpolation , non-zero -> interpolation activated Performance kndx - Index into f-table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0). kfn - table number kout1...koutN - output vector elements This opcode is useful in all cases in which one needs to access sets of values associated to unique indexes (for example, multi-channel samples, STFT bin frames, spectral formants, p-field based scores etc.). The number of elements of each vector (length of the vector) is determined by the number of optional arguments on the right (kout1 , kout2, kout3, .... koutN). vtablek allows the user to switch beetween interpolated or non-interpolated output at k-rate by means of kinterp argument. vtablek allows also to switch the table number at k-rate (but this is possible only when vector frames of each used table have the same number of elements, otherwise unpredictable results could occurr), as well as to choose indexing style (raw or normalized, see also ixmode argument of table opcode ). Notice that no wrap nor limit mode is implemented. So, if an index attempt to access to a zone not allocated by the table, Csound will probably crash. However this drawback can be easily avoided by using wrap or limit opcodes applied to indexes before using vtable, in order to correct eventual out-of-range values. Notice that vtablek's output arguments are placed at the left of the opcode name, differently from usual (this style is already used in other opcodes using undefined lists of output arguments such as fin or trigseq). Examples Here is an example of the vtablek opcode. It uses the files vtablek.csd. Example of the vtablek opcode. See also vtablea, vtablei, vtabk, vtablewk, vtabwk, Credits Written by Gabriel Maldonado. Example written by Andres Cabrera. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/tanh.xml0000644000000000000000000000423112262561504016055 0ustar rootroot tanh Mathematical Operations:Trigonometric Functions tanh tanh Performs a hyperbolic tangent function. Description Returns the hyperbolic tangent of x. Syntax tanh(x) (no rate restriction) Examples Here is an example of the tanh opcode. It uses the file tanh.csd. Example of the tanh opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cos, cosh, cosinv, sin, sinh, sininv, tan, taninv Credits Author: &namejohn; New in version 3.47 manual_src~dfsg/opcodes/vbap4.xml0000644000000000000000000001213012262561504016134 0ustar rootroot vbap4 Signal Modifiers:Panning and Spatialization vbap4 vbap4 Distributes an audio signal among 4 channels. Description Distributes an audio signal among 4 channels. Syntax ar1, ar2, ar3, ar4 vbap4 asig, kazim [, kelev] [, kspread] Performance asig -- audio signal to be panned kazim -- azimuth angle of the virtual source kelev (optional) -- elevation angle of the virtual source kspread (optional) -- spreading of the virtual source (range 0 - 100). If value is zero, conventional amplitude panning is used. When kspread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. vbap4 takes an input signal, asig and distributes it among 4 outputs, according to the controls kazim and kelev, and the configured loudspeaker placement. If idim = 2, kelev is set to zero. The distribution is performed using Vector Base Amplitude Panning (VBAP - See reference). VBAP distributes the signal using loudspeaker data configured with vbaplsinit. The signal is applied to, at most, two loudspeakers in 2-D loudspeaker configurations, and three loudspeakers in 3-D loudspeaker configurations. If the virtual source is panned outside the region spanned by loudspeakers, the nearest loudspeakers are used in panning. Please note that all vbap panning opcodes require the vbap system to be initialized using vbaplsinit. Examples See the entry for vbap8 for an example of usage of the vbap opcodes. Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. Examples Here is an example of the vbap4 opcode. It uses the file vbap4.csd. Example of the vbap4 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: spread of source = 0 spread of source = 8 spread of source = 17 spread of source = 25 spread of source = 33 spread of source = 42 spread of source = 50 spread of source = 58 spread of source = 67 spread of source = 75 spread of source = 83 spread of source = 92 spread of source = 100 See Also vbap16, vbap16move, vbap4move, vbap8, vbap8move, vbaplsinit, vbapz, vbapzmove Credits Author: Ville Pulkki Sibelius Academy Computer Music Studio Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Helsinki, Finland May 2000 New in Csound Version 4.06. Input parameters accept k-rate since Csund 5.09. manual_src~dfsg/opcodes/flslidbnksetk.xml0000644000000000000000000001325312262561504017766 0ustar rootroot FLslidBnkSetk FLTK:Other FLslidBnkSetk FLslidBnkSetk modify the values of a slider bank. Description FLslidBnkSetk modifies the values of a slider bank according to an array of values stored in a table. Syntax FLslidBnkSetk ktrig, ihandle, ifn [, istartIndex, istartSlid, inumSlid] Initialization ihandle - handle of the sliderBnk (to be used to set its values). ifn - number of a table containing an array of values to set each slider to. istartIndex - (optional) starting index of the table element of to be evaluated firstly. Default value is zero istartSlid - (optional) starting slider to be evaluated. Default 0, denoting the first slider. inumSlid - (optional) number of sliders to be updated. Default 0, denoting all sliders. Performance ktrig – the output of FLslidBnkSetk consists of a trigger that informs if sliders have to be updated or not. A non-zero value forces the slider to be updated. FLslidBnkSetk is similar to FLslidBnkSet but allows k-rate to modify the values of FLslidBnk (FLslidBnkSetk can also be used with FLvslidBnk, obtaining identical result). It also allows the slider bank to be joined with MIDI. If you are using MIDI (for instance, when using the slider8table opcode), FLslidBnkSetk changes the values of FLslidBnk bank of sliders to a set of values located in a table. This opcode is actually able to serve as a MIDI bridge to the FLslidBnk widget when used together with the sliderXXtable set of opcodes (see slider8table entry for more information). Notice, that, when you want to use table indexing as a curve response, it is not possible to do it directly in the iconfigtable configuration of FLslidBnk2, when you intend to use the FLslidBnkSetk opcode. In fact, corresponding inputTable element of FLslidBnkSetk must be set in linear mode and respect the 0 to 1 range. Even the corresponding elements of sliderXXtable must be set in linear mode and in the normalized range. You can do table indexing later, by using the tab and tb opcodes, and rescaling output according to max and min values. By the other hand, it is possible to use linear and exponential curve response directly, by setting the actual min-max range and flag both in the iconfigtable of corresponding FLslidBnk2 and in sliderXXtable. FLslidBnkSetk the k-rate version of FLslidBnk2Set. See Also FLslider, FLslidBnkGetHandle, FLslidBnk, FLslidBnk2, FLvslidBnk, FLvslidBnk2 FLslidBnkSet, FLslidBnk2Set, slider8table Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/readscratch.xml0000644000000000000000000000410112262561504017402 0ustar rootroot readscratch Instrument Control:Sensing and Control readscratch readscratch returns a value stored in the instance of an instrument. Description The readscratch opcode returns one of four scalar values stored in the instance of an instrument. Syntax ival readscratch[index] Initialisation ival -- variable for result. index -- which value to read, defaulting to zero. Examples Here is an example of the readscratch opcode. It uses the file readscratch.csd. Example of the readscratch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also writescratch, Credits Author: &namejohn; March 2013 New in Csound version 6.00 manual_src~dfsg/opcodes/rnd.xml0000644000000000000000000000556712262561504015723 0ustar rootroot rnd Mathematical Operations:Random Functions rnd rnd Returns a random number in a unipolar range at the rate given by the input argument. Description Returns a random number in a unipolar range at the rate given by the input argument. Syntax rnd(x) (init- or control-rate only) Where the argument within the parentheses may be an expression. These value converters sample a global random sequence, but do not reference seed. The result can be a term in a further expression. Performance Returns a random number in the unipolar range 0 to x. Examples Here is an example of the rnd opcode. It uses the file rnd.csd. Example of the rnd opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should be: SECTION 1: new alloc for instr 1: instr 1: irand = 9.735 new alloc for instr 1: instr 1: irand = 1.394 new alloc for instr 1: instr 1: irand = 7.695 midi channel 1 now using instr 1 B 0.000 .. 2.000 T 2.000 TT 2.000 M: 0.00000 new alloc for instr 2: i 2 time 2.10000: 5.25005 i 2 time 2.20000: 6.22665 i 2 time 2.30000: 9.69511 i 2 time 2.40000: 7.16822 i 2 time 2.50000: 9.45134 i 2 time 2.60000: 1.34123 i 2 time 2.70000: 2.09879 i 2 time 2.80000: 2.36001 i 2 time 2.90000: 0.03553 See Also birnd Credits Author: &namebarry; MIT Cambridge, Massachussetts 1997 manual_src~dfsg/opcodes/fini.xml0000644000000000000000000000577712262561504016070 0ustar rootroot fini Signal I/O:File I/O fini fini Read signals from a file at i-rate. Description Read signals from a file at i-rate. Syntax fini ifilename, iskipframes, iformat, in1 [, in2] [, in3] [, ...] Initialization ifilename -- input file name (can be a string or a handle number generated by fiopen) iskipframes -- number of frames to skip at the start (every frame contains a sample of each channel) iformat -- a number specifying the input file format. If a header is found, this argument is ignored. 0 - floating points in text format (loop; see below) 1 - floating points in text format (no loop; see below) 2 - 32 bit floating points in binary format (no loop) Performance fini is the complement of fouti and foutir. It reads the values each time the corresponding instrument note is activated. When iformat is set to 0 and the end of file is reached, the file pointer is zeroed. This restarts the scan from the beginning. When iformat is set to 1 or 2, no looping is enabled and at the end of file the corresponding variables will be filled with zeroes. Please note that since this opcode generates its output using input parameters (on the right side of the opcode), these variables must be initialized before use, otherwise a 'used before defined' error will occur. You can use the init opcode for this. See Also fin, fink Credits Author: &namegabriel; Italy 1999 New in Csound version 3.56 manual_src~dfsg/opcodes/inleta.xml0000644000000000000000000000603612262561504016404 0ustar rootroot inleta Signal Flow Graph Opcodes inleta inleta Receives an arate signal into an instrument through a named port. Description Receives an arate signal into an instrument through a named port. Syntax asignal inleta Sname Initialization Sname -- String name of the inlet port. The name of the inlet is implicitly qualified by the instrument name or number, so it is valid to use the same inlet name in more than one instrument (but not to use the same inlet name twice in one instrument). Performance asignal -- audio input signal During performance, the arate inlet signal is received from each instance of an instrument containing an outlet port to which this inlet has been connected using the connect opcode. The signals of all the outlets connected to an inlet are summed in the inlet. Examples Here is an example of the inleta opcode. It uses the file inleta.csd. Example of the inleta opcode. See Also outleta outletk outletkid outletf inletk inletkid inletf connect alwayson ftgenonce More information on this opcode: http://www.csounds.com/journal/issue13/signalFlowGraphOpcodes.html , written by Michael Gogins Credits By: &namemichael; 2009 manual_src~dfsg/opcodes/tabplay.xml0000644000000000000000000000452312262561504016563 0ustar rootroot tabplay Table Control:Read/Write Opreations tabplay tabplay Playing-back control signals. Description Plays-back control-rate signals on trigger-temporization basis. Syntax tabplay ktrig, knumtics, kfn, kout1 [,kout2,..., koutN] Performance ktrig -- starts playing when non-zero. knumtics -- stop recording or reset playing pointer to zero when the number of tics defined by this argument is reached. kfn -- table where k-rate signals are recorded. kout1,...,koutN -- playback output signals. The tabplay and tabrec opcodes allow to record/playback control signals on trigger-temporization basis. tabplay plays back a group of k-rate signals, previously recorded by tabrec into a table. Each time ktrig argument is triggered, an internal counter is increased of one unit. After knumtics trigger impluses are received by ktrig argument, the internal counter is zeroed and playback is restarted from the beginning, in looping style. These opcodes can be used like a sort of ``middle-term'' memory that ``remembers'' generated signals. Such memory can be used to supply generative music with a coherent iterative compositional structure. See Also tabrec Credits Written by &namegabriel;. manual_src~dfsg/opcodes/turnoff2.xml0000644000000000000000000000472512262561504016700 0ustar rootroot turnoff2 Instrument Control:Duration Control turnoff2 turnoff2 Turn off instance(s) of other instruments at performance time. Description Turn off instance(s) of other instruments at performance time. Syntax turnoff2 kinsno, kmode, krelease Performance kinsno -- instrument to be turned off (can be fractional) if zero or negative, no instrument is turned off kmode -- sum of the following values: 0, 1, or 2: turn off all instances (0), oldest only (1), or newest only (2) 4: only turn off notes with exactly matching (fractional) instrument number, rather than ignoring fractional part 8: only turn off notes with indefinite duration (p3 < 0 or MIDI) krelease -- if non-zero, the turned off instances are allowed to release, otherwise are deactivated immediately (possibly resulting in clicks) As a rule of thumb, you should turn off instruments with a higher instrument number than the one where turnoff is called, as doing otherwise might cause initialization issues. See Also turnoff Credits Author: Istvan Varga 2005 New in Csound 5.00 manual_src~dfsg/opcodes/loop_gt.xml0000644000000000000000000000571612262561504016577 0ustar rootroot loop_gt Instrument Control:Program Flow Control loop_gt loop_gt Looping constructions. Description Construction of looping operations. Syntax loop_gt indx, idecr, imin, label loop_gt kndx, kdecr, kmin, label Initialization indx -- i-rate variable to count loop. idecr -- value to decrement the loop. imin -- minimum value of loop index. Performance kndx -- k-rate variable to count loop. kdecr -- value to decrement the loop. kmin -- minimum value of loop index. The actions of loop_gt are equivalent to the code indx = indx - idecr if (indx > imin) igoto label or kndx = kndx - kdecr if (kndx > kmin) kgoto label See Also loop_ge, loop_le and loop_lt. More information on this opcode: http://www.csounds.com/journal/2006summer/controlFlow_part2.html , written by Steven Yi. And in the Floss Manuals: http://en.flossmanuals.net/csound/ch018_c-control-structures Credits &nameistvan;. New in Csound version 5.01 manual_src~dfsg/opcodes/pvsinit.xml0000644000000000000000000000361112262561504016620 0ustar rootroot pvsinit Spectral Processing:Streaming pvsinit pvsinit Initialise a spectral (f) variable to zero. Description Performs the equivalent to an init operation on an f-variable. Syntax fsig pvsinit isize[, iolap, iwinsize, iwintype, iformat] Performance fsig -- output pv stream set to zero. isize -- size of the DFT frame. iolap -- size of the analysis overlap, defaults to isize/4. iwinsize -- size of the analysis window, defaults to isize. iwintype -- type of analysis window, defaults to 1, Hanning. iformat -- pvsdata format, defaults to 0:PVS_AMP_FREQ. Examples Example fsig pvsinit 1024 Credits Author: &namevictor; November 2004 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/readk.xml0000644000000000000000000001450312262561504016214 0ustar rootroot readk Signal I/O:File I/O readk readk Periodically reads an orchestra control-signal value from an external file. Description Periodically reads an orchestra control-signal value from a named external file in a specific format. Syntax kres readk ifilname, iformat, iprd Initialization ifilname -- an integer N denoting a file named "readk.N" or a character string (in double quotes, spaces permitted) denoting the external file name. For a string, it may either be a full path name with directory specified or a simple filename. In the later case, the file is sought first in the current directory, then in SSDIR, and finally in SFDIR. iformat -- specifies the input data format: 1 = 8-bit signed integers (char) 4 = 16-bit short integers 5 = 32-bit long integers 6 = 32-bit floats 7 = ASCII long integers (plain text) 8 = ASCII floats (plain text) Note that A-law and U-law formats are not available, and that all formats except the last two are binary. The input file should be a "raw", headerless data file. iprd -- the rate (period) in seconds, rounded to the nearest orchestra control period, at which the signal is read from the input file. A value of 0 implies one control period (the enforced minimum), which will read new values at the orchestra control rate. Longer periods will cause the same values to repeat for more than one control period. Performance kres -- output of the signal read from ifilname. This opcode allows a generated control signal value to be read from a named external file. The file should contain no header information but it should contain a regularly sampled time series of control values. For ASCII text formats, the values are assumed to be separated by at least one whitespace character. There may be any number of readk opcodes in an instrument or orchestra and they may read from the same or different files. Examples Here is an example of the readk opcode. It uses the file readk.csd. Example of the readk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the readk opcode. It uses the file readk-2.csd. Example 2 of the readk opcode. Its output should include lines like this: WARNING: Seeding from current time 683384022 i 1 time 1.00033: 463.64510 i 1 time 2.00000: 463.64510 i 1 time 3.00000: 483.14200 i 1 time 4.00000: 567.55973 i 1 time 5.00000: 576.37060 i 1 time 6.00000: 460.66550 i 2 time 6.00033: 463.64510 i 2 time 7.00000: 463.64510 i 2 time 8.00000: 483.14200 i 2 time 9.00000: 567.55970 i 2 time 10.00000: 576.37060 i 2 time 11.00000: 460.66550 See Also dumpk, dumpk2, dumpk3, dumpk4, readk2, readk3, readk4 Credits By: John ffitch and Barry Vercoe 1999 or earlier manual_src~dfsg/opcodes/cpsmidinn.xml0000644000000000000000000001752712262561504017123 0ustar rootroot cpsmidinn Pitch Converters:Functions cpsmidinn cpsmidinn Converts a Midi note number value to cycles-per-second. Description Converts a Midi note number value to cycles-per-second. Syntax cpsmidinn (MidiNoteNumber) (init- or control-rate args only) where the argument within the parentheses may be a further expression. Performance cpsmidinn is a function that takes an i-rate or k-rate value representing a Midi note number and returns the equivalent frequency value in cycles-per-second (Hertz). This conversion assumes that Middle C is Midi note number 60 and that Middle A is tuned to 440 Hz. Midi note number values are typically integers in the range from 0 to 127 but fractional values or values outside of this range will be interpreted consistently. cpsmidinn vs. cpsmidi The cpsmidinn opcode may be used in any Csound instrument instance whether it is activated from a Midi event, score event, line event, or from another instrument. The input value for cpsmidinn might for example come from a p-field in a textual score or it may have been retrieved from the real-time Midi event that activated the current note using the notnum opcode. You must specify an i-rate or k-rate expression for the Midi note number that is to be converted. On the other hand, the cpsmidi opcode only produces meaningful results in a Midi-activated note (either real-time or from a Midi score with the -F flag). With cpsmidi, the Midi note number value is taken from the Midi event associated with the instrument instance, and no location or expression for this value may be specified. cpsmidinn and its related opcodes are really value converters with a special function of manipulating pitch data. Data concerning pitch and frequency can exist in any of the following forms: Pitch and Frequency Values Name Abbreviation octave point pitch-class (8ve.pc) pch octave point decimal oct cycles per second cps Midi note number (0-127) midinn
The first two forms consist of a whole number, representing octave registration, followed by a specially interpreted fractional part. For pch, the fraction is read as two decimal digits representing the 12 equal-tempered pitch classes from .00 for C to .11 for B. For oct, the fraction is interpreted as a true decimal fractional part of an octave. The two fractional forms are thus related by the factor 100/12. In both forms, the fraction is preceded by a whole number octave index such that 8.00 represents Middle C, 9.00 the C above, etc. Midi note number values range between 0 and 127 (inclusively) with 60 representing Middle C, and are usually whole numbers. Thus A440 can be represented alternatively by 440 (cps), 69 (midinn), 8.09 (pch), or 8.75 (oct). Microtonal divisions of the pch semitone can be encoded by using more than two decimal places. The mnemonics of the pitch conversion units are derived from morphemes of the forms involved, the second morpheme describing the source and the first morpheme the object (result). Thus cpspch(8.09) will convert the pitch argument 8.09 to its cps (or Hertz) equivalent, giving the value of 440. Since the argument is constant over the duration of the note, this conversion will take place at i-time, before any samples for the current note are produced. By contrast, the conversion cpsoct(8.75 + k1) which gives the value of A440 transposed by the octave interval k1. The calculation will be repeated every k-period since that is the rate at which k1 varies. Note The conversion from pch, oct, or midinn into cps is not a linear operation but involves an exponential process that could be time-consuming when executed repeatedly. Csound now uses a built-in table lookup to do this efficiently, even at audio rates. Because the table index is truncated without interpolation, pitch resolution when using one of these opcodes is limited to 8192 discrete and equal divisions of the octave, and some pitches of the standard 12-tone equally-tempered scale are very slightly mistuned (by at most 0.15 cents). Examples Here is an example of the cpsmidinn opcode. It uses the file cpsmidinn.csd. Example of the cpsmidinn opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the cpsmidinn opcode. It uses the file cpsmidinn2.csd. Second example of the cpsmidinn opcode. See Also octmidinn, pchmidinn, cpsmidi, notnum, cpspch, cpsoct, octcps, octpch, pchoct Credits Derived from original value converters by Barry Vercoe. New in version 5.07 manual_src~dfsg/opcodes/faustcompile.xml0000644000000000000000000000317612262561504017625 0ustar rootroot faustcompile Faust Opcodes faustcompile faustcompile Invokes the just-in-time compiler to produce a instantiable DSP process from a Faust program. Description Faustcompile will compile a Faust program from a string, controlled by various arguments. Multi-line strings are accepted, using {{ }} to enclose the string. Syntax ihandle faustcompile Scode, Sargs Initialization Scode -- a string (in double-quotes or enclosed by {{ }}) containing a Faust program. Sargs -- a string (in double-quotes or enclosed by {{ }}) containing Faust compiler args. Examples Here is an example of the faustcompile opcode, with a simple program that mixes two inputs: ihandle faustcompile "process=+;", "-vec -lv 1" Credits Author: &namevictor;, 2013 manual_src~dfsg/opcodes/vtablewi.xml0000644000000000000000000000464612262561504016752 0ustar rootroot vtablewi Vectorial:Tables vtablewi vtablewi Write vectors (to tables -or arrays of vectors). Description This opcode writes vectors to tables at init time. Syntax vtablewi indx, ifn, ixmode, inarg1 [, inarg2, inarg3 , .... , inargN ] Initialization indx - Index into f-table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0). ifn - table number ixmode - index data mode. The default value is 0. == 0 index is treated as a raw table location, == 1 index is normalized (0 to 1). inarg1...inargN - input vector elements Performance This opcode is useful in all cases in which one needs to write sets of values associated to unique indexes (for example, multi-channel samples, STFT bin frames, spectral formants, p-field based scores etc.). The number of elements of each vector (length of the vector) is determined by the number of optional arguments on the right (inarg1, inarg2, inarg3, .... inargN). Notice that no wrap nor limit mode is implemented. So, if an index attempt to access to a zone not allocated by the table, Csound will probably crash. However this drawback can be easily avoided by using wrap or limit opcodes applied to indexes before using vtablewi, in order to correct eventual out-of-range values. Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/urandom.xml0000644000000000000000000000653712262561504016603 0ustar rootroot urandom Signal Generators:Random (Noise) Generators urandom urandom truly random opcodes with controllable range. Description truly random opcodes with controllable range. These units are for Linux only and use /dev/urandom to construct Csound random values Syntax ax urandom [imin, imax] ix urandom [imin, imax] kx urandom [imin, imax] Initialization ix -- i-rate output value. imin -- minimum value of range; defaults to -1. imax -- maximum value of range; defaults to +1. Notes The algorithm produces 2^64 different possible values which are scaled to fit the range requested. The randomness comes form the usual Linux /dev/urandom method, and there is no guarantee that it will be truly random, but there is a good chance. It does not show any cycles. Performance ax -- a-rate output value. kx -- k-rate output value. Examples Here is an example of the urandom opcode at a-rate. It uses the file urandom.csd. An example of the urandom opcode at a-rate. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an example of the urandom opcode at k-rate. It uses the file urandom_krate.csd. An example of the urandom opcode at k-rate. Its output should include lines like this: k1=0.229850 k1=-0.077047 k1=-0.199339 k1=-0.620577 k1=-0.119447 k1=-0.596258 k1=0.525800 k1=-0.171583 k1=-0.017196 k1=-0.974613 k1=-0.036276 Credits Author: &namejohn; New in version 5.13 manual_src~dfsg/opcodes/koutc.xml0000644000000000000000000000075312262561504016255 0ustar rootroot koutc koutc koutc Deprecated. Description Deprecated as of version 3.52. Use the outkc opcode instead. manual_src~dfsg/opcodes/strtolk.xml0000644000000000000000000000465612262561504016640 0ustar rootroot strtolk Strings:Conversion strtolk strtolk Converts a string to a signed integer (k-rate). Description Convert a string to a signed integer value at i- or k-rate. It is also possible to pass an strset index or a string p-field from the score instead of a string argument. If the string cannot be parsed as an integer number, an init or perf error occurs and the instrument is deactivated. Note If a k-rate index variable is used, it should be valid at i-time as well. Syntax kr strtolk Sstr kr strtolk kndx strtolk can parse numbers in decimal, octal (prefixed by 0), and hexadecimal (with a prefix of 0x) format. Performance kr -- Value of string as signed integer. Sstr -- String to convert. indx -- index of string set by strset Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/puts.xml0000644000000000000000000000462512262561504016125 0ustar rootroot puts Strings:Manipulation puts puts Print a string constant or variable Description puts prints a string with an optional newline at the end whenever the trigger signal is positive and changes. Syntax puts Sstr, ktrig[, inonl] Initialization Sstr -- string to be printed inonl (optional, defaults to 0) -- if non-zero, disables the default printing of a newline character at the end of the string Performance ktrig -- trigger signal, should be valid at i-time. The string is printed at initialization time if ktrig is positive, and at performance time whenever ktrig is both positive and different from the previous value. Use a constant value of 1 to print once at note initialization. Examples Here is an example of the puts opcode. It uses the file puts.csd. Example of the puts opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: frequency in Hertz : 440 frequency in Hertz : 441 frequency in Hertz : 442 .... frequency in Hertz : 459 frequency in Hertz : 460 Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/zir.xml0000644000000000000000000000434112262561504015731 0ustar rootroot zir Zak Patch System zir zir Reads from a location in zk space at i-rate. Description Reads from a location in zk space at i-rate. Syntax ir zir indx Initialization indx -- points to the zk location to be read. Performance zir reads the signal at indx location in zk space. Examples Here is an example of the zir opcode. It uses the file zir.csd. Example of the zir opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zar, zarg, zkr Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/vdelayxq.xml0000644000000000000000000001016612262561504016764 0ustar rootroot vdelayxq Signal Modifiers:Delay vdelayxq vdelayxq A 4-channel variable delay opcode with high quality interpolation. Description A 4-channel variable delay opcode with high quality interpolation. Syntax aout1, aout2, aout3, aout4 vdelayxq ain1, ain2, ain3, ain4, adl, imd, iws [, ist] Initialization imd -- max. delay time (seconds) iws -- interpolation window size (see below) ist (optional) -- skip initialization if not zero Performance aout1, aout2, aout3, aout4 -- output audio signals. ain1, ain2, ain3, ain4 -- input audio signals. adl -- delay time in seconds This opcode uses high quality (and slow) interpolation, that is much more accurate than the currently available linear and cubic interpolation. The iws parameter sets the number of input samples used for calculating one output sample (allowed values are any integer multiply of 4 in the range 4 - 1024); higher values mean better quality and slower speed. The multichannel opcodes (eg. vdelayxq) allow delaying 2 or 4 variables at once (stereo or quad signals); this is much more efficient than using separate opcodes for each channel. Notes Delay time is measured in seconds (unlike in vdelay and vdelay3), and must be a-rate. The minimum allowed delay is iws/2 samples. Using the same variables as input and output is allowed in these opcodes. In vdelayxw*, changing the delay time has some effects on output volume: a = 1 / (1 + dt) where a is the output gain, and dt is the change of delay time per seconds. These opcodes are best used in the double-precision version of Csound. Examples Here is an example of the vdelayxq opcode. It uses the file vdelayxq.csd. Example of the vdelayxq opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Two important points here. First, the delay time must be always positive. And second, even though the delay time can be controlled in k-rate, it is not advised to do so, since sudden time changes will create clicks. See Also vdelayx, vdelayxs, vdelayxw, vdelayxwq, vdelayxws manual_src~dfsg/opcodes/lorisplay.xml0000644000000000000000000001014112262561504017136 0ustar rootroot lorisplay Spectral Processing:Loris lorisplay lorisplay renders a stored set of bandwidth-enhanced partials using the method of Bandwidth-Enhanced Additive Synthesis implemented in the Loris software, applying control-rate frequency, amplitude, and bandwidth scaling envelopes. Syntax ar lorisplay ireadidx, kfreqenv, kampenv, kbwenv Description lorisplay renders a stored set of bandwidth-enhanced partials using the method of Bandwidth-Enhanced Additive Synthesis implemented in the Loris software, applying control-rate frequency, amplitude, and bandwidth scaling envelopes. Initialization istoreidx, ireadidx, isrcidx, itgtidx are labels that identify a stored set of bandwidth-enhanced partials. lorisread imports partials from a SDIF file and stores them with the integer label istoreidx. lorismorph morphs sets of partials labeled isrcidx and itgtidx, and stores the resulting partials with the integer label istoreidx. lorisplay renders the partials stored with the label ireadidx. The labels are used only at initialization time, and may be reused without any cost or benefit in efficiency, and without introducing any interaction between instruments or instances. Performance lorisplay implements signal reconstruction using Bandwidth-Enhanced Additive Synthesis. The control data is obtained from a stored set of bandwidth-enhanced partials imported from an SDIF file using lorisread or constructed by another unit generator such as lorismorph. kfreqenv is a control-rate transposition factor: a value of 1 incurs no transposition, 1.5 transposes up a perfect fifth, and .5 down an octave. kampenv is a control-rate scale factor that is applied to all partial amplitude envelopes. kbwenv is a control-rate scale factor that is applied to all partial bandwidth or noisiness envelopes. The bandwidth-enhanced partial data is stored in memory with a specified label for future access by another generator. Examples Here is an example of the lorisplay opcode. It uses the file lorisplay.csd. Example of the lorisplay opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits This implementation of the Loris unit generators was written by Kelly Fitz (loris@cerlsoundgroup.org). It is patterned after a prototype implementation of the lorisplay unit generator written by Corbin Champion, and based on the method of Bandwidth-Enhanced Additive Synthesis and on the sound morphing algorithms implemented in the Loris library for sound modeling and manipulation. The opcodes were further adapted as a plugin for Csound 5 by Michael Gogins. manual_src~dfsg/opcodes/ioutat.xml0000644000000000000000000000076212262561504016435 0ustar rootroot ioutat ioutat ioutat Deprecated. Description Deprecated as of version 3.52. Use the outiat opcode instead. manual_src~dfsg/opcodes/soundin.xml0000644000000000000000000001515712262561504016613 0ustar rootroot soundin Signal I/O:Signal Input soundin soundin Reads audio data from an external device or stream. Description Reads audio data from an external device or stream. Up to 24 channels may be read before v5.14, extended to 40 in later versions. Syntax ar1[, ar2[, ar3[, ... a24]]] soundin ifilcod [, iskptim] [, iformat] \ [, iskipinit] [, ibufsize] Initialization ifilcod -- integer or character-string denoting the source soundfile name. An integer denotes the file soundin.filcod; a character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the named file is sought first in the current directory, then in that given by the environment variable SSDIR (if defined) then by SFDIR. See also GEN01. iskptim (optional, default=0) -- time in seconds of input sound to be skipped. The default value is 0. In csound 5.00 and later, this may be negative to add a delay instead of skipping time. iformat (optional, default=0) -- specifies the audio data file format: 1 = 8-bit signed char (high-order 8 bits of a 16-bit integer) 2 = 8-bit A-law bytes 3 = 8-bit U-law bytes 4 = 16-bit short integers 5 = 32-bit long integers 6 = 32-bit floats 7 = 8-bit unsigned int (not available in Csound versions older than 5.00) 8 = 24-bit int (not available in Csound versions older than 5.00) 9 = 64-bit doubles (not available in Csound versions older than 5.00) iskipinit -- switches off all initialisation if non zero (default=0). This was introduced in 4_23f13 and csound5. ibufsize -- buffer size in mono samples (not sample frames). Not available in Csound versions older than 5.00. The default buffer size is 2048. If iformat = 0 it is taken from the soundfile header, and if no header from the Csound -o command-line flag. The default value is 0. Performance soundin is functionally an audio generator that derives its signal from a pre-existing file. The number of channels read in is controlled by the number of result cells, a1, a2, etc., which must match that of the input file. A soundin opcode opens this file whenever the host instrument is initialized, then closes it again each time the instrument is turned off. There can be any number of soundin opcodes within a single instrument or orchestra. Two or more of them can read simultaneously from the same external file. Note to Windows users Windows users typically use back-slashes, \, when specifying the paths of their files. As an example, a Windows user might use the path c:\music\samples\loop001.wav. This is problematic because back-slashes are normally used to specify special characters. To correctly specify this path in Csound, one may alternately: Use forward slashes: c:/music/samples/loop001.wav Use back-slash special characters, \\: c:\\music\\samples\\loop001.wav Examples Here is an example of the soundin opcode. It uses the file soundin.csd, fox.wav and kickroll.wav. Example of the soundin opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also diskin, in, inh, ino, inq, ins Credits Authors: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 Warning to Windows users added by &namekevin;, April 2002 manual_src~dfsg/opcodes/pvshift.xml0000644000000000000000000001047712262561504016617 0ustar rootroot pvshift Spectral Processing:Streaming pvshift pvshift Shift the frequency components of a pv stream, stretching/compressing its spectrum. Description Shift the frequency components of a pv stream, stretching/compressing its spectrum. Syntax fsig pvshift fsigin, kshift, klowest[, kkeepform, igain, kcoefs] Performance fsig -- output pv stream fsigin -- input pv stream kshift -- shift amount (in Hz, positive or negative). klowest -- lowest frequency to be shifted. kkeepform -- attempt to keep input signal formants; 0: do not keep formants; 1: keep formants using a liftered cepstrum method; 2: keep formants by using a true envelope method (defaults to 0). kgain -- amplitude scaling (defaults to 1). kcoefs -- number of cepstrum coefs used in formant preservation (defaults to 80). This opcode will shift the components of a pv stream, from a certain frequency upwards, up or down a fixed amount (in Hz). It can be used to transform a harmonic spectrum into an inharmonic one. The kkeepform flag can be used to try and preserve formants for possibly interesting and unusual spectral modifications. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Example asig in ; get the signal in fsig pvsanal asig, 1024, 256, 1024, 1 ; analyse it ftps pvshift fsig, 100, 0 ; add 100 Hz to each component atps pvsynth ftps ; synthesise it Depending on the input, this will transform a pitched sound into an inharmonic, bell-like sound. Here is an example of the use of the pvshift opcode. It uses the file pvshift.csd. Example of the <emphasis>pvshift</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namevictor; November 2004 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/kweibull.xml0000644000000000000000000000077612262561504016753 0ustar rootroot kweibull kweibull kweibull Deprecated. Description Deprecated as of version 3.49. Use the weibull opcode instead. manual_src~dfsg/opcodes/vco2init.xml0000644000000000000000000002037312262561504016665 0ustar rootroot vco2init Signal Generators:Dynamic Spectrum Oscillators vco2init vco2init Calculates tables for use by vco2 opcode. Description vco2init calculates tables for use by vco2 opcode. Optionally, it is also possible to access these tables as standard Csound function tables. In this case, vco2ft can be used to find the correct table number for a given oscillator frequency. In most cases, this opcode is called from the orchestra header. Using vco2init in instruments is possible but not recommended. This is because replacing tables during performance can result in a Csound crash if other opcodes are accessing the tables at the same time. Note that vco2init is not required for vco2 to work (tables are automatically allocated by the first vco2 call, if not done yet), however it can be useful in some cases: Pre-calculate tables at orchestra load time. This is useful to avoid generating the tables during performance, which could interrupt real-time processing. Share the tables as Csound ftables. By default, the tables can be accessed only by vco2. Change the default parameters of tables (e.g. size) or use an user-defined waveform specified in a function table. Syntax ifn vco2init iwave [, ibasfn] [, ipmul] [, iminsiz] [, imaxsiz] [, isrcft] Initialization ifn -- the first free ftable number after the allocated tables. If ibasfn was not specified, -1 is returned. iwave -- sum of the following values selecting which waveforms are to be calculated: 16: triangle 8: square wave 4: pulse (not normalized) 2: 4 * x * (1 - x) (integrated sawtooth) 1: sawtooth Alternatively, iwave can be set to a negative integer that selects an user-defined waveform. This also requires the isrcft parameter to be specified. vco2 can access waveform number -1. However, other user-defined waveforms are usable only with vco2ft or vco2ift. ibasfn (optional, default=-1) -- ftable number from which the table set(s) can be accessed by opcodes other than vco2. This is required by user defined waveforms, with the exception of -1. If this value is less than 1, it is not possible to access the tables calculated by vco2init as Csound function tables. ipmul (optional, default=1.05) -- multiplier value for number of harmonic partials. If one table has n partials, the next one will have n * ipmul (at least n + 1). The allowed range for ipmul is 1.01 to 2. Zero or negative values select the default (1.05). iminsiz (optional, default=-1) -- minimum table size. imaxsiz (optional, default=-1) -- maximum table size. The actual table size is calculated by multiplying the square root of the number of harmonic partials by iminsiz, rounding up the result to the next power of two, and limiting this not to be greater than imaxsiz. Both parameters, iminsiz and imaxsiz, must be power of two, and in the allowed range. The allowed range is 16 to 262144 for iminsiz to up to 16777216 for imaxsiz. Zero or negative values select the default settings: The minimum size is 128 for all waveforms except pulse (iwave=4). Its minimum size is 256. The default maximum size is usually the minimum size multiplied by 64, but not more than 16384 if possible. It is always at least the minimum size. isrcft (optional, default=-1) -- source ftable number for user-defined waveforms (if iwave < 0). isrcft should point to a function table containing the waveform to be used for generating the table array. The table size is recommended to be at least imaxsiz points. If iwave is not negative (built-in waveforms are used), isrcft is ignored. Warning The number and size of tables is not fixed. Orchestras should not depend on these parameters, as they are subject to changes between releases. If the selected table set already exists, it is replaced. If any opcode is accessing the tables at the same time, it is very likely that a crash will occur. This is why it is recommended to use vco2init only in the orchestra header. These tables should not be replaced/overwritten by GEN routines or the ftgen opcode. Otherwise, unpredictable behavior or a Csound crash may occur if vco2 is used. The first free ftable after the table array(s) is returned in ifn. Examples Here is an example of the vco2init opcode. It uses the file vco2init.csd. Example of the vco2init opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like these: i 1 time 0.00002: 103.00000 i 1 time 0.10000: 103.00000 i 1 time 0.20000: 103.00000 i 1 time 0.30002: 103.00000 i 1 time 0.40000: 104.00000 i 1 time 0.50000: 104.00000 ....... ...... i 1 time 4.80002: 135.00000 i 1 time 4.90000: 136.00000 i 1 time 5.00000: 138.00000 See the example for the vco2 opcode too. See Also vco2ft, vco2ift, and vco2. Credits Author: &nameistvan; New in version 4.22 manual_src~dfsg/opcodes/ptable3.xml0000644000000000000000000000602012262561504016453 0ustar rootroot ptable3 Signal Generators:Table Access ptable3 ptable3 Accesses table values by direct indexing with cubic interpolation. Description Accesses table values by direct indexing with cubic interpolation. Syntax ares ptable3 andx, ifn [, ixmode] [, ixoff] [, iwrap] ires ptable3 indx, ifn [, ixmode] [, ixoff] [, iwrap] kres ptable3 kndx, ifn [, ixmode] [, ixoff] [, iwrap] Initialization ifn -- function table number. ixmode (optional) -- index data mode. The default value is 0. 0 = raw index 1 = normalized (0 to 1) ixoff (optional) -- amount by which index is to be offset. For a table with origin at center, use tablesize/2 (raw) or .5 (normalized). The default value is 0. iwrap (optional) -- wraparound index flag. The default value is 0. 0 = nowrap (index < 0 treated as index=0; index > tablesize sticks at index=size) 1 = wraparound. Performance ptable3 is identical to table3, except that it uses does not require the table to have a power of two size. See Also table, tablei, table3, ptable, ptablei, oscil1, oscil1i, osciln Credits Author: &namejohn; Jan 2012 New in Csound version 5.16 manual_src~dfsg/opcodes/kpoisson.xml0000644000000000000000000000077612262561504017002 0ustar rootroot kpoisson kpoisson kpoisson Deprecated. Description Deprecated as of version 3.49. Use the poisson opcode instead. manual_src~dfsg/opcodes/scanhammer.xml0000644000000000000000000000351012262561504017240 0ustar rootroot scanhammer Signal Generators:Scanned Synthesis scanhammer scanhammer Copies from one table to another with a gain control. Description This is is a variant of tablecopy, copying from one table to another, starting at ipos, and with a gain control. The number of points copied is determined by the length of the source. Other points are not changed. This opcode can be used to hit a string in the scanned synthesis code. Syntax scanhammer isrc, idst, ipos, imult Initialization isrc -- source function table. idst -- destination function table. ipos -- starting position (in points). imult -- gain multiplier. A value of 0 will leave values unchanged. See Also scantable Credits Author: Matt Gilliard April 2002 New in version 4.20 manual_src~dfsg/opcodes/vtabk.xml0000644000000000000000000000646412262561504016244 0ustar rootroot vtabk Vectorial:Tables vtabk vtabk Read vectors (from tables -or arrays of vectors). Description This opcode reads vectors from tables at k-rate. Syntax vtabk kndx, ifn, kout1 [, kout2, kout3, .... , koutN ] Initialization ifn - table number Performance kndx - Index into f-table, either a positive number range matching the table length kout1...koutN - output vector elements This opcode is useful in all cases in which one needs to access sets of values associated to unique indexes (for example, multi-channel samples, STFT bin frames, spectral formants, p-field based scores etc.) . The number of elements of each vector (length of the vector) is determined by the number of optional arguments on the right (kout1, kout2, kout3, .... koutN). Notice that no wrap nor limit mode is implemented. So, if an index attempt to access to a zone not allocated by the table, Csound will probably crash. However this drawback can be easily avoided by using wrap or limit opcodes applied to indexes before using vtabk, in order to correct eventual out-of-range values. The vtab family is similar to vtable, but is much faster because interpolation is not available, table number cannot be changed after initialization, and only raw indexing is supported. Notice that vtabk's output arguments are placed at the right of the opcode name, differently from usual (this style is already used in other opcodes using undefined lists of output arguments such as fin or trigseq). Examples For an example of the vtabk opcode usage, see vtablek. See also vtabi, vtaba, vtablek, vtablewk, vtabwk, Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/ioff.xml0000644000000000000000000000075212262561504016052 0ustar rootroot ioff ioff ioff Deprecated. Description Deprecated as of version 3.52. Use the noteoff opcode instead. manual_src~dfsg/opcodes/cossegr.xml0000644000000000000000000001241512262561504016573 0ustar rootroot cossegr Signal Generators:Linear and Exponential Generators cossegr cossegr Trace a series of line segments between specified points with cosine interpolation, including a release segment. Description Trace a series of line segments between specified points with cosine interpolation, including a release segment. Syntax ares cossegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz kres cossegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz Initialization ia -- starting value. ib, ic, etc. -- value after dur1 seconds, etc. idur1 -- duration in seconds of first segment. A zero or negative value will cause all initialization to be skipped. idur2, idur3, etc. -- duration in seconds of subsequent segments. A zero or negative value will terminate the initialization process with the preceding point, permitting the last-defined line or curve to be continued indefinitely in performance. The default is zero. irel, iz -- duration in seconds and final value of a note releasing segment. Performance These units generate control or audio signals whose values can pass through 2 or more specified points. The sum of dur values may or may not equal the instrument's performance time: a shorter performance will truncate the specified pattern, while a longer one will cause the last-defined segment to continue on in the same direction. cossegr is amongst the Csound r units that contain a note-off sensor and release time extender. When each senses an event termination or MIDI noteoff, it immediately extends the performance time of the current instrument by irel seconds, and sets out to reach the value iz by the end of that period (no matter which segment the unit is in). r units can also be modified by MIDI noteoff velocities. For two or more extenders in an instrument, extension is by the greatest period. You can use other pre-made envelopes which start a release segment upon recieving a note off message, like linenr and expsegr, or you can construct more complex envelopes using xtratim and release. Note that you don't need to use xtratim if you are using cossegr, since the time is extended automatically. Examples Here is an example of the cossegr opcode. It uses the file cossegr.csd. Example of the cossegr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also linenr, linsegr, expsegr, envlpxr, mxadsr, madsr expon, expseg, expsega line, linseg, xtratim, transegr Credits Author: &namejohn; August 2012. New in Csound 5.18 manual_src~dfsg/opcodes/looptseg.xml0000644000000000000000000001155312262561504016764 0ustar rootroot looptseg Signal Generators:Linear and Exponential Generators looptseg looptseg Generate control signal consisting of exponential or linear segments delimited by two or more specified points. Description Generate control signal consisting of controllable exponential segments or linear segments delimited by two or more specified points. The entire envelope is looped at kfreq rate. Each parameter can be varied at k-rate. Syntax ksig looptseg kfreq, ktrig, iphase, kvalue0, ktype0, ktime0, [, kvalue1] [,ktype1] [, ktime1] \ [, kvalue2] [,ktype2] [, ktime2] [...] [, kvalueN] [,ktypeN] [, ktimeN] Initialization iphase -- A value between 0 and 1 to say where to start the loop. Zero, the commonest value, indicates the beginning. Performance ksig -- Output signal. kfreq -- Repeat rate in Hz or fraction of Hz. ktrig -- If non-zero, retriggers the envelope from start (see trigger opcode), before the envelope cycle is completed. kvalue0...kvalueN -- Values of points ktime0...ktimeN -- Times between points; expressed in fractions of a cycle (see below). The final time designates a ramp between the final value and the first value. ktype0...ktypeN -- shape of the envelope. If the value is 0 then the shap eis linear; otherwise it is an concave exponential (positive type) or a convex exponential (negative type). looptseg opcode is similar to transeg, but the entire envelope is looped at kfreq rate. Notice that times are not expressed in seconds but in fraction of a cycle. Actually each duration represent is proportional to the other, and the entire cycle duration is proportional to the sum of all duration values. The sum of all duration is then rescaled according to kfreq argument. For example, considering an envelope made up of 3 segments, each segment having 100 as duration value, their sum will be 300. This value represents the total duration of the envelope, and is actually divided into 3 equal parts, a part for each segment. Actually, the real envelope duration in seconds is determined by kfreq. Again, if the envelope is made up of 3 segments, but this time the first and last segments have a duration of 50, whereas the central segment has a duration of 100 again, their sum will be 200. This time 200 represent the total duration of the 3 segments, so the central segment will be twice as long as the other segments. All parameters can be varied at k-rate. Negative frequency values are allowed, reading the envelope backward. ktime0 should always be set to 0, except if the user wants some special effect. Examples Here is an example of the looptseg opcode. It uses the file looptseg.csd. Example of the looptseg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also lpshold loopseg Credits Author: &namejohn; New in Version 5.12 manual_src~dfsg/opcodes/mirror.xml0000644000000000000000000000537312262561504016445 0ustar rootroot mirror Signal Modifiers:Signal Limiters mirror mirror Reflects the signal that exceeds the low and high thresholds. Description Reflects the signal that exceeds the low and high thresholds. Syntax ares mirror asig, klow, khigh ires mirror isig, ilow, ihigh kres mirror ksig, klow, khigh Initialization isig -- input signal ilow -- low threshold ihigh -- high threshold Performance xsig -- input signal klow -- low threshold khigh -- high threshold mirror reflects the signal that exceeds the low and high thresholds. This opcode is useful in several situations, such as table indexing or for clipping and modeling a-rate, i-rate or k-rate signals. Examples Here is an example of the mirror opcode. It uses the file mirror.csd. Example of the mirror opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also limit, wrap Credits Author: &namegabriel; Italy New in Csound version 3.49 manual_src~dfsg/opcodes/lpf18.xml0000644000000000000000000000712412262561504016061 0ustar rootroot lpf18 Signal Modifiers:Standard Filters:Resonant lpf18 lpf18 A 3-pole sweepable resonant lowpass filter. Description Implementation of a 3 pole sweepable resonant lowpass filter. Syntax ares lpf18 asig, xfco, xres, xdist [, iskip] Initialization iskip (optional, default=0) -- Skip initialization if present and non-zero. Performance xfco -- the filter cutoff frequency in Hz. Should be in the range 0 to sr/2. xres -- the amount of resonance. Self-oscillation occurs when xres is approximately 1. Should usually be in the range 0 to 1, however, values slightly greater than 1 are possible for more sustained oscillation and an overdrive effect. xdist -- amount of distortion. kdist = 0 gives a clean output. xdist > 0 adds tanh() distortion controlled by the filter parameters, in such a way that both low cutoff and high resonance increase the distortion amount. Some experimentation is encouraged. lpf18 is a digital emulation of a 3 pole (18 dB/oct.) lowpass filter capable of self-oscillation with a built-in distortion unit. It is really a 3-pole version of moogvcf, retuned, recalibrated and with some performance improvements. The tuning and feedback tables use no more than 6 adds and 6 multiplies per control rate. The distortion unit, itself, is based on a modified tanh function driven by the filter controls. Note This filter requires that the input signal be normalized to one. Examples Here is an example of the lpf18 opcode. It uses the file lpf18.csd. Example of the lpf18 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Josep M Comajuncosas Spain December 2000 Example written by &namekevin; with help from Iain Duncan. Thanks goes to Iain for helping with the example. New in Csound version 4.10 Audio rate parameters introduced in version 6.02 October 2013. manual_src~dfsg/opcodes/gbuzz.xml0000644000000000000000000001227112262561504016267 0ustar rootroot gbuzz Signal Generators:Dynamic Spectrum Oscillators gbuzz gbuzz Output is a set of harmonically related cosine partials. Description Output is a set of harmonically related cosine partials. Syntax ares gbuzz xamp, xcps, knh, klh, kmul, ifn [, iphs] Initialization ifn -- table number of a stored function containing a cosine wave. A large table of at least 8192 points is recommended. iphs (optional, default=0) -- initial phase of the fundamental frequency, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is zero Performance The buzz units generate an additive set of harmonically related cosine partials of fundamental frequency xcps, and whose amplitudes are scaled so their summation peak equals xamp. The selection and strength of partials is determined by the following control parameters: knh -- total number of harmonics requested. If knh is negative, the absolute value is used. If knh is zero, a value of 1 is used. klh -- lowest harmonic present. Can be positive, zero or negative. In gbuzz the set of partials can begin at any partial number and proceeds upwards; if klh is negative, all partials below zero will reflect as positive partials without phase change (since cosine is an even function), and will add constructively to any positive partials in the set. kmul -- specifies the multiplier in the series of amplitude coefficients. This is a power series: if the klhth partial has a strength coefficient of A, the (klh + n)th partial will have a coefficient of A * (kmul ** n), i.e. strength values trace an exponential curve. kmul may be positive, zero or negative, and is not restricted to integers. buzz and gbuzz are useful as complex sound sources in subtractive synthesis. buzz is a special case of the more general gbuzz in which klh = kmul = 1; it thus produces a set of knh equal-strength harmonic partials, beginning with the fundamental. (This is a band-limited pulse train; if the partials extend to the Nyquist, i.e. knh = int (sr / 2 / fundamental freq.), the result is a real pulse train of amplitude xamp.) Although both knh and klh may be varied during performance, their internal values are necessarily integer and may cause pops due to discontinuities in the output. kmul, however, can be varied during performance to good effect. gbuzz can be amplitude- and/or frequency-modulated by either control or audio signals. N.B. This unit has its analog in GEN11, in which the same set of cosines can be stored in a function table for sampling by an oscillator. Although computationally more efficient, the stored pulse train has a fixed spectral content, not a time-varying one as above. Examples Here is an example of the gbuzz opcode. It uses the file gbuzz.csd. Example of the gbuzz opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also buzz Credits September 2003. Thanks to &namekanata; for correcting the mentions of the kmul parameter. manual_src~dfsg/opcodes/wiiconnect.xml0000644000000000000000000000574412262561504017277 0ustar rootroot wiiconnect Instrument Control:Sensing and Control wiiconnect wiiconnect Reads data from a number of external Nintendo Wiimote controllers. Description Opens and at control-rate polls up to four external Nintendo Wiimote controllers. Syntax ires wiiconnect [itimeout, imaxnum] Initialization itimeout -- integer number of seconds the system should wait for all Wiimotes to be connected. If not given it defaults to 10 seconds. imaxnum -- maximum number of Wiimotes to locate. If not given it defaults to 4. Initially each Wiimote has its numeric allocation indicated by lighting one of the four LEDs. ires -- return value is 1 if sucess or zero on failure. Performance Please note that these opcodes are currently only supported on Linux. Every control cycle each Wiimote is polled for its status and position. These values are read by the wiidata opcode. The result returned is 1 in most cases, but will be zero if a Wiimote disconnects, Example Here is an example of the wii opcodes. It uses the file wii.csd. Example of the wii opcodes. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also wiidata, wiirange, wiisend Credits Author: &namejohn; Codemist Ltd 2009 New in version 5.11 manual_src~dfsg/opcodes/zawm.xml0000644000000000000000000000717512262561504016113 0ustar rootroot zawm Zak Patch System zawm zawm Writes to a za variable at a-rate with mixing. Description Writes to a za variable at a-rate with mixing. Syntax zawm asig, kndx [, imix] Initialization imix (optional, default=1) -- indicates if mixing should occur. Performance asig -- value to be written to the za location. kndx -- points to the zk or za location to which to write. These opcodes are fast, and always check that the index is within the range of zk or za space. If not, an error is reported, 0 is returned, and no writing takes place. zawm is a mixing opcode, it adds the signal to the current value of the variable. If no imix is specified, mixing always occurs. imix = 0 will cause overwriting like ziw, zkw, and zaw. Any other value will cause mixing. Caution: When using the mixing opcodes ziwm, zkwm, and zawm, care must be taken that the variables mixed to, are zeroed at the end (or start) of each k- or a-cycle. Continuing to add signals to them, can cause their values can drift to astronomical figures. One approach would be to establish certain ranges of zk or za variables to be used for mixing, then use zkcl or zacl to clear those ranges. Examples Here is an example of the zawm opcode. It uses the file zawm.csd. Example of the zawm opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zaw, ziw, ziwm, zkw, zkwm Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/cpsoct.xml0000644000000000000000000001341712262561504016424 0ustar rootroot cpsoct Pitch Converters:Functions cpsoct cpsoct Converts an octave-point-decimal value to cycles-per-second. Description Converts an octave-point-decimal value to cycles-per-second. Syntax cpsoct (oct) (no rate restriction) where the argument within the parentheses may be a further expression. Performance cpsoct and its related opcodes are really value converters with a special function of manipulating pitch data. Data concerning pitch and frequency can exist in any of the following forms: Pitch and Frequency Values Name Abbreviation octave point pitch-class (8ve.pc) pch octave point decimal oct cycles per second cps Midi note number (0-127) midinn
The first two forms consist of a whole number, representing octave registration, followed by a specially interpreted fractional part. For pch, the fraction is read as two decimal digits representing the 12 equal-tempered pitch classes from .00 for C to .11 for B. For oct, the fraction is interpreted as a true decimal fractional part of an octave. The two fractional forms are thus related by the factor 100/12. In both forms, the fraction is preceded by a whole number octave index such that 8.00 represents Middle C, 9.00 the C above, etc. Midi note number values range between 0 and 127 (inclusively) with 60 representing Middle C, and are usually whole numbers. Thus A440 can be represented alternatively by 440 (cps), 69 (midinn), 8.09 (pch), or 8.75 (oct). Microtonal divisions of the pch semitone can be encoded by using more than two decimal places. The mnemonics of the pitch conversion units are derived from morphemes of the forms involved, the second morpheme describing the source and the first morpheme the object (result). Thus cpspch(8.09) will convert the pitch argument 8.09 to its cps (or Hertz) equivalent, giving the value of 440. Since the argument is constant over the duration of the note, this conversion will take place at i-time, before any samples for the current note are produced. By contrast, the conversion cpsoct(8.75 + k1) which gives the value of A440 transposed by the octave interval k1. The calculation will be repeated every k-period since that is the rate at which k1 varies. Note The conversion from pch, oct, or midinn into cps is not a linear operation but involves an exponential process that could be time-consuming when executed repeatedly. Csound now uses a built-in table lookup to do this efficiently, even at audio rates. Because the table index is truncated without interpolation, pitch resolution when using one of these opcodes is limited to 8192 discrete and equal divisions of the octave, and some pitches of the standard 12-tone equally-tempered scale are very slightly mistuned (by at most 0.15 cents). Examples Here is an example of the cpsoct opcode. It uses the file cpsoct.csd. Example of the cpsoct opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: icps = 440.000 instr 1: icps = 446.110 instr 1: icps = 452.344 instr 1: icps = 80.521 See Also cpspch, octcps, octpch, pchoct, cpsmidinn, octmidinn, pchmidinn manual_src~dfsg/opcodes/sndwarp.xml0000644000000000000000000002522712262561504016611 0ustar rootroot sndwarp Signal Generators:Granular Synthesis sndwarp sndwarp Reads a mono sound sample from a table and applies time-stretching and/or pitch modification. Description sndwarp reads sound samples from a table and applies time-stretching and/or pitch modification. Time and frequency modification are independent from one another. For example, a sound can be stretched in time while raising the pitch! The window size and overlap arguments are important to the result and should be experimented with. In general they should be as small as possible. For example, start with iwsize=sr/10 and ioverlap=15. Try irandw=iwsize*0.2. If you can get away with less overlaps, the program will be faster. But too few may cause an audible flutter in the amplitude. The algorithm reacts differently depending upon the input sound and there are no fixed rules for the best use in all circumstances. But with proper tuning, excellent results can be achieved. Syntax ares [, ac] sndwarp xamp, xtimewarp, xresample, ifn1, ibeg, iwsize, \ irandw, ioverlap, ifn2, itimemode Initialization ifn1 -- the number of the table holding the sound samples which will be subjected to the sndwarp processing. GEN01 is the appropriate function generator to use to store the sound samples from a pre-existing soundfile. ibeg -- the time in seconds to begin reading in the table (or soundfile). When itimemode is non- zero, the value of xtimewarp is offset by ibeg. iwsize -- the window size in samples used in the time scaling algorithm. irandw -- the bandwidth of a random number generator. The random numbers will be added to iwsize. ioverlap -- determines the density of overlapping windows. ifn2 -- a function used to shape the window. It is usually used to create a ramp of some kind from zero at the beginning and back down to zero at the end of each window. Try using a half sine (i.e.: f1 0 16384 9 .5 1 0) which works quite well. Other shapes can also be used. Performance ares -- the single channel of output from the sndwarp unit generator. sndwarp assumes that the function table holding the sampled signal is a mono one. This simply means that sndwarp will index the table by single-sample frame increments. The user must be aware then that if a stereo signal is used with sndwarp, time and pitch will be altered accordingly. ac (optional) -- a single-layer (no overlaps), unwindowed versions of the time and/or pitch altered signal. They are supplied in order to be able to balance the amplitude of the signal output, which typically contains many overlapping and windowed versions of the signal, with a clean version of the time-scaled and pitch-shifted signal. The sndwarp process can cause noticeable changes in amplitude, (up and down), due to a time differential between the overlaps when time-shifting is being done. When used with a balance unit, ac can greatly enhance the quality of sound. xamp -- the value by which to scale the amplitude (see note on the use of this when using ac). xtimewarp -- determines how the input signal will be stretched or shrunk in time. There are two ways to use this argument depending upon the value given for itimemode. When the value of itimemode is 0, xtimewarp will scale the time of the sound. For example, a value of 2 will stretch the sound by 2 times. When itimemode is any non-zero value then xtimewarp is used as a time pointer in a similar way in which the time pointer works in lpread and pvoc. An example below illustrates this. In both cases, the pitch will not be altered by this process. Pitch shifting is done independently using xresample. xresample -- the factor by which to change the pitch of the sound. For example, a value of 2 will produce a sound one octave higher than the original. The timing of the sound, however, will not be altered. Examples Here is an example of the sndwarp opcode. It uses the file sndwarp.csd. Example of the sndwarp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Other examples The below example shows a slowing down or stretching of the sound stored in the stored table (ifn1). Over the duration of the note, the stretching will grow from no change from the original to a sound which is ten times slower than the original. At the same time the overall pitch will move upward over the duration by an octave. iwindfun = 1 isampfun = 2 ibeg = 0 iwindsize = 2000 iwindrand = 400 ioverlap = 10 awarp line 1, p3, 1 aresamp line 1, p3, 2 kenv line 1, p3, .1 asig sndwarp kenv, awarp, aresamp, isampfun, ibeg, iwindsize, iwindrand, ioverlap, iwindfun, 0 Now, here's an example using xtimewarp as a time pointer and using stereo: itimemode = 1 atime line 0, p3, 10 ar1, ar2 sndwarpst kenv, atime, aresamp, sampfun, ibeg, iwindsize, iwindrand, ioverlap, \ iwindfun, itimemode In the above, atime advances the time pointer used in the sndwarpst from 0 to 10 over the duration of the note. If p3 is 20 then the sound will be two times slower than the original. Of course you can use a more complex function than just a single straight line to control the time factor. Now the same as above but using the balance function with the optional outputs: asig,acmp sndwarp 1, awarp, aresamp, isampfun, ibeg, iwindsize, iwindrand, ioverlap, iwindfun, itimemode abal balance asig, acmp asig1,asig2,acmp1,acmp2 sndwarpst 1, atime, aresamp, sampfun, ibeg, iwindsize, iwindrand, ioverlap, \ iwindfun, itimemode abal1 balance asig1, acmp1 abal2 balance asig2, acmp2 In the above two examples notice the use of the balance unit. The output of balance can then be scaled, enveloped, sent to an out or outs, and so on. Notice that the amplitude arguments to sndwarp and sndwarpst are 1 in these examples. By scaling the signal after the sndwarp process, abal, abal1, and abal2 should contain signals that have nearly the same amplitude as the original input signal to the sndwarp process. This makes it much easier to predict the levels and avoid samples out of range or sample values that are too small. More Advice Only use the stereo version when you really need to be processing a stereo file. It is somewhat slower than the mono version and if you use the balance function it is slower again. There is nothing wrong with using a mono sndwarp in a stereo orchestra and sending the result to one or both channels of the stereo output! See Also sndwarpst Credits Author: Richard Karpen Seattle, WA USA 1997 manual_src~dfsg/opcodes/pluck.xml0000644000000000000000000001254012262561504016243 0ustar rootroot pluck Signal Generators:Waveguide Physical Modeling pluck pluck Produces a naturally decaying plucked string or drum sound. Description Audio output is a naturally decaying plucked string or drum sound based on the Karplus-Strong algorithms. Syntax ares pluck kamp, kcps, icps, ifn, imeth [, iparm1] [, iparm2] Initialization icps -- intended pitch value in Hz, used to set up a buffer of 1 cycle of audio samples which will be smoothed over time by a chosen decay method. icps normally anticipates the value of kcps, but may be set artificially high or low to influence the size of the sample buffer. ifn -- table number of a stored function used to initialize the cyclic decay buffer. If ifn = 0, a random sequence will be used instead. imeth -- method of natural decay. There are six, some of which use parameters values that follow. simple averaging. A simple smoothing process, uninfluenced by parameter values. stretched averaging. As above, with smoothing time stretched by a factor of iparm1 (>=1). simple drum. The range from pitch to noise is controlled by a 'roughness factor' in iparm1 (0 to 1). Zero gives the plucked string effect, while 1 reverses the polarity of every sample (octave down, odd harmonics). The setting .5 gives an optimum snare drum. stretched drum. Combines both roughness and stretch factors. iparm1 is roughness (0 to 1), and iparm2 the stretch factor (>=1). weighted averaging. As method 1, with iparm1 weighting the current sample (the status quo) and iparm2 weighting the previous adjacent one. iparm1 + iparm2 must be <= 1. 1st order recursive filter, with coefs .5. Unaffected by parameter values. iparm1, iparm2 (optional) -- parameter values for use by the smoothing algorithms (above). The default values are both 0. Performance kamp -- the output amplitude. kcps -- the resampling frequency in cycles-per-second. An internal audio buffer, filled at i-time according to ifn, is continually resampled with periodicity kcps and the resulting output is multiplied by kamp. Parallel with the sampling, the buffer is smoothed to simulate the effect of natural decay. Plucked strings (1, 2, 5, 6) are best realized by starting with a random noise source, which is rich in initial harmonics. Drum sounds (methods 3, 4) work best with a flat source (wide pulse), which produces a deep noise attack and sharp decay. The original Karplus-Strong algorithm used a fixed number of samples per cycle, which caused serious quantization of the pitches available and their intonation. This implementation resamples a buffer at the exact pitch given by kcps, which can be varied for vibrato and glissando effects. For low values of the orch sampling rate (e.g. sr = 10000), high frequencies will store only very few samples (sr / icps). Since this may cause noticeable noise in the resampling process, the internal buffer has a minimum size of 64 samples. This can be further enlarged by setting icps to some artificially lower pitch. Examples Here is an example of the pluck opcode. It uses the file pluck.csd. Example of the pluck opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. manual_src~dfsg/opcodes/mode.xml0000644000000000000000000000601612262561504016052 0ustar rootroot mode Signal Modifiers:Standard Filters mode mode A filter that simulates a mass-spring-damper system Description Filters the incoming signal with the specified resonance frequency and quality factor. It can also be seen as a signal generator for high quality factor, with an impulse for the excitation. You can combine several modes to built complex instruments such as bells or guitar tables. Syntax aout mode ain, xfreq, xQ [, iskip] Initialization iskip (optional, default=0) -- if non zero skip the initialisation of the filter. Performance aout -- filtered signal ain -- signal to filter xfreq -- resonant frequency of the filter Warning This filter becomes unstable if sr/xfreq < pi (e.g xfreq > 14037 Hz @ 44 kHz) xQ -- quality factor of the filter The resonance time is roughly proportional to xQ/xfreq. See Modal Frequency Ratios for frequency ratios of real instruments which can be used to determine the values of xfreq. Examples Here is an example of the mode opcode. It uses the file mode.csd. Example of the mode opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Original UDO and documentation/example by François Blanc Opcode translation to C-code by Steven Yi New in version 5.04 Audio rate parameters introduced in version 6.02 November 2013. manual_src~dfsg/opcodes/flprintk.xml0000644000000000000000000000555012262561504016761 0ustar rootroot FLprintk FLTK:Other FLprintk FLprintk A FLTK opcode that prints a k-rate value at specified intervals. Description FLprintk is similar to printk but shows values of a k-rate signal in a text field instead of on the console. Syntax FLprintk itime, kval, idisp Initialization itime -- how much time in seconds is to elapse between updated displays. idisp -- a handle value that was output from a previous instance of the FLvalue opcode to display the current value of the current valuator in the FLvalue widget itself. If the user doesn't want to use this feature that displays current values, it must be set to a negative number by the user. Performance kval -- k-rate signal to be displayed. FLprintk is similar to printk, but shows values of a k-rate signal in a text field instead of showing it in the console. The idisp argument must be filled with the ihandle return value of a previous FLvalue opcode. While FLvalue should be placed in the header section of an orchestra inside an FLpanel/FLpanelEnd block, FLprintk must be placed inside an instrument to operate correctly. For this reason, it slows down performance and should be used for debugging purposes only. See Also FLbox, FLbutBank, FLbutton, FLprintk2, FLvalue Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/schedkwhennamed.xml0000644000000000000000000001174312262561504020261 0ustar rootroot schedkwhennamed Instrument Control:Invocation schedkwhennamed schedkwhennamed Similar to schedkwhen but uses a named instrument at init-time. Description Similar to schedkwhen but uses a named instrument at init-time. Syntax schedkwhennamed ktrigger, kmintim, kmaxnum, "name", kwhen, kdur \ [, ip4] [, ip5] [...] Initialization ip4, ip5, ... -- Equivalent to p4, p5, etc., in a score i statement Performance ktrigger -- triggers a new score event. If ktrigger is 0, no new event is triggered. kmintim -- minimum time between generated events, in seconds. If kmintim is less than or equal to 0, no time limit exists. kmaxnum -- maximum number of simultaneous instances of named instrument allowed. If the number of existant instances of the named instrument is greater than or equal to kmaxnum, no new event is generated. If kmaxnum is less than or equal to 0, it is not used to limit event generation. "name" -- the named instrument's name. kwhen -- start time of the new event. Equivalent to p2 in a score i statement. Measured from the time of the triggering event. kwhen must be greater than or equal to 0. If kwhen greater than 0, the instrument will not be initialized until the actual time when it should start performing. kdur -- duration of event. Equivalent to p3 in a score i statement. If kdur is 0, the instrument will only do an initialization pass, with no performance. If kdur is negative, a held note is initiated. (See ihold and i statement.) While waiting for events to be triggered by schedkwhennamed, the performance must be kept going, or Csound may quit if no score events are expected. To guarantee continued performance, an f0 statement may be used in the score. Note that the schedkwhennamed opcode can't accept string p-fields. If you need to pass strings when instantiating an instrument, use the scoreline or scoreline_i opcode. Examples Here is an example of the schedkwhennamed opcode. It uses the file schedkwhennamed.csd. Example of the schedkwhennamed opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also event, event_i, schedule, schedwhen, schedkwhen, scoreline, scoreline_i Credits Author: &namerasmus; EMS, Stockholm, Sweden New in Csound version 4.23 manual_src~dfsg/opcodes/ifndef.xml0000644000000000000000000000376212262561504016366 0ustar rootroot #ifndeforchestra #ifndef Orchestra Syntax:Macros #ifndef #ifndef Conditional reading of code. Description If the specified macro is not defined then #ifndef can incorporate text into an orchestra upto the next #end. This is similar to, but independent of, the macro system in the score language. Syntax #ifndef NAME .... #else  .... #end  Performance Note that the #ifndef can be nested, like in the C preprocessor language. See Also $NAME, #define, #ifdef. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April 2005 New in Csound5 (and 4.23f13) manual_src~dfsg/opcodes/filepeak.xml0000644000000000000000000000602312262561504016704 0ustar rootroot filepeak Signal I/O:Soundfile Queries filepeak filepeak Returns the peak absolute value of a sound file. Description Returns the peak absolute value of a sound file. Syntax ir filepeak ifilcod [, ichnl] Initialization ifilcod -- sound file to be queried ichnl (optional, default=0) -- channel to be used in calculating the peak value. Default is 0. ichnl = 0 returns peak value of all channels ichnl > 0 returns peak value of ichnl Performance filepeak returns the peak absolute value of the sound file ifilcod. Examples Here is an example of the filepeak opcode. It uses the file filepeak.csd, and Church.wav. Example of the filepeak opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The filepeak's output should include lines like this: Peak value in file 'Church.wav' is 0.909363 (-0.825 dB). Peak value in file 'Church_norm.wav' is 0.501190 (-6.000 dB). See Also filelen, filenchnls, filesr Credits Author: &namematt; July 1999 New in Csound version 3.57 manual_src~dfsg/opcodes/flbox.xml0000644000000000000000000002036512262561504016243 0ustar rootroot FLbox FLTK:Other FLbox FLbox A FLTK widget that displays text inside of a box. Description A FLTK widget that displays text inside of a box. Syntax ihandle FLbox "label", itype, ifont, isize, iwidth, iheight, ix, iy [, image] Initialization ihandle -- a handle value (an integer number) that unequivocally references a corresponding widget. This is used by other opcodes that modify a widget's properties (see Modifying FLTK Widget Appearance). It is automatically output by FLbox and must not be set by the user label. (The user label is a double-quoted string containing some user-provided text placed near the widget.) label -- a double-quoted string containing some user-provided text, placed near corresponding widget. Notice that with FLbox, it is not necessary to call the FLsetTextType opcode at all in order to use a symbol. In this case, it is sufficient to set a label starting with @ followed by the proper formatting string. The following symbols are supported: FLTK label supported symbols. FLTK label supported symbols. The @ sign may be followed by the following optional formatting characters, in this order: # forces square scaling rather than distortion to the widget's shape. +[1-9] or -[1-9] tweaks the scaling a little bigger or smaller. [1-9] rotates by a multiple of 45 degrees. 6 does nothing, the others point in the direction of that key on a numeric keypad. itype -- an integer number denoting the appearance of the widget. The following values are legal for itype: 1 - flat box 2 - up box 3 - down box 4 - thin up box 5 - thin down box 6 - engraved box 7 - embossed box 8 - border box 9 - shadow box 10 - rounded box 11 - rounded box with shadow 12 - rounded flat box 13 - rounded up box 14 - rounded down box 15 - diamond up box 16 - diamond down box 17 - oval box 18 - oval shadow box 19 - oval flat box ifont -- an integer number denoting the font of FLbox. ifont argument to set the font type. The following values are legal for ifont: 1 - helvetica (same as "Arial" under Windows) 2 - helvetica bold 3 - helvetica italic 4 - helvetica bold italic 5 - courier 6 - courier bold 7 - courier italic 8 - courier bold italic 9 - times 10 - times bold 11 - times italic 12 - times bold italic 13 - symbol 14 - screen 15 - screen bold 16 - dingbats isize -- size of the font. iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of the upper left corner of the valuator, relative to the upper left corner of corresponding window. (Expressed in pixels.) iy -- vertical position of the upper left corner of the valuator, relative to the upper left corner of corresponding window. (Expressed in pixels.) image -- a handle referring to an eventual image opened with bmopen opcode. If it is set, it allows a skin for that widget. Note about the bmopen opcode Although the documentation mentions the bmopen opcode, it has not been implemented in Csound 4.22. Performance FLbox is useful to show some text in a window. The text is bounded by a box, whose aspect depends on itype argument. Note that FLbox is not a valuator and its value is fixed. Its value cannot be modified. Examples Here is an example of the FLbox opcode. It uses the file FLbox.csd. Example of the FLbox opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLbutBank, FLbutton, FLprintk, FLprintk2, FLvalue Credits Author: &namegabriel; New in version 4.22 Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/trirand.xml0000644000000000000000000001033712262561504016572 0ustar rootroot x-class noise generators trirand Signal Generators:Random (Noise) Generators trirand trirand Triangular distribution random number generator Description Triangular distribution random number generator. This is an x-class noise generator. Syntax ares trirand krange ires trirand krange kres trirand krange Performance krange -- the range of the random numbers (-krange to +krange). For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the trirand opcode. It uses the file trirand.csd. Example of the trirand opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: i 1 time 0.00067: -4.97993 i 1 time 0.20067: 1.20909 i 1 time 0.40067: 17.45873 i 1 time 0.60067: 52.55409 i 1 time 0.80067: -1.92888 i 1 time 1.00000: -11.01149 i 1 time 1.20067: 9.79521 i 1 time 1.40067: 26.98504 i 1 time 1.60067: 24.67405 i 1 time 1.80000: -67.59846 i 1 time 2.00000: 64.24861 WARNING: Seeding from current time 521999639 i 2 time 3.00067: 3.28969 i 2 time 3.20067: 54.98986 i 2 time 3.40067: -33.84788 i 2 time 3.60000: -41.93523 i 2 time 3.80067: -6.61742 i 2 time 4.00000: 39.67097 i 2 time 4.20000: 2.95123 i 2 time 4.40067: 45.59255 i 2 time 4.60067: 16.57259 i 2 time 4.80067: -18.80273 i 2 time 5.00000: -2.01697 See Also betarand, bexprnd, cauchy, exprand, gauss, linrand, pcauchy, poisson, unirand, weibull Credits Author: &nameparis; MIT, Cambridge 1995 manual_src~dfsg/opcodes/vpow_i.xml0000644000000000000000000001071212262561504016427 0ustar rootroot vpow_i Vectorial:Scalar operations vpow_i vpow_i Raises each element of a vector to a scalar power Description Raises each element of a vector to a scalar power Syntax vpow_i ifn, ival, ielements [, idstoffset] Initialization ifn - number of the table hosting the vectorial signal to be processed ielements - number of elements of the vector ival - scalar value to which the elements of ifn will be raised idstoffset - index offset for the destination table Performance vpow_i elevates each element of the vector contained in the table ifn to the power of ival, starting from table index idstoffset. This enables you to process a specific section of a table by specifying the offset and the number of elements to be processed. Offset is counted starting from 0, so if no offset is specified (or set to 0), the table will be modified from the beginning. This opcode runs only on initialization, there is a k-rate version of this opcode called vpow. Negative values for idstoffset are valid. Elements from the vector that are outside the table, will be discarded, and they will not wrap around the table. In all these opcodes, the resulting vectors are stored in ifn, overriding the intial vectors. If you want to keep initial vector, use vcopy or vcopy_i to copy it in another table. All these operators are designed to be used together with other opcodes that operate with vectorial signals such as vcella, adsynt, adsynt2, etc. They can also be useful in conjunction with the spectral opcodes pvsftw and pvsftr. Examples Here is an example of the vpow_i opcode. It uses the file vpow_i.csd. Examples of the vpow_i opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the vpow_i opcode. It uses the file vpow_i-2.csd. See also vadd_i, vmult_i, vpow and vexp_i. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/vpow.xml0000644000000000000000000001410012262561504016112 0ustar rootroot vpow Vectorial:Scalar operations vpow vpow Raises each element of a vector to a scalar power. Description Raises each element of a vector to a scalar power. Syntax vpow ifn, kval, kelements [, kdstoffset] [, kverbose] Initialization ifn - number of the table hosting the vectorial signal to be processed Performance kval - scalar value to which the elements of ifn will be raised kelements - number of elements of the vector kdstoffset - index offset for the destination table (Optional, default = 0) kverbose - Selects whether or not warnings are printed (Default=0) vpow raises each element of the vector contained in the table ifn to the power of kval, starting from table index kdstoffset. This enables you to process a specific section of a table by specifying the offset and the number of elements to be processed. Offset is counted starting from 0, so if no offset is specified (or set to 0), the table will be modified from the beginning. Note that this opcode runs at k-rate so the value of kval is processed every control period. Use with care or you will end up with very large (or small) numbers (or use vpow_i). These opcodes (vadd, vmult, vpow and vexp) perform numeric operations between a vectorial control signal (hosted by the table ifn), and a scalar signal (kval). Result is a new vector that overrides old values of ifn. All these opcodes work at k-rate. Negative values for kdstoffset are valid. Elements from the vector that are outside the table, will be discarded, and they will not wrap around the table. If the optional kverbose argument is different to 0, the opcode will print warning messages every k-pass if table lengths are exceeded. In all these opcodes, the resulting vectors are stored in ifn, overriding the intial vectors. If you want to keep initial vector, use vcopy or vcopy_i to copy it in another table. All these operators are designed to be used together with other opcodes that operate with vectorial signals such as vcella, adsynt, adsynt2, etc. They can also be useful in conjunction with the spectral opcodes pvsftw and pvsftr. Please note that the elements argument has changed in version 5.03 from i-rate to k-rate. This will change the opcode's behavior in the unusual cases where the i-rate variable ielements is changed inside the instrument, for example in: instr 1 ielements = 10 vadd 1, 1, ielements ielements = 20 vadd 2, 1, ielements turnoff endin Examples Here is an example of the vpow opcode. It uses the file vpow.csd. Examples of the vpow opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the vpow opcode. It uses the file vpow-2.csd. See also vadd_i, vmult, vpow and vexp. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/ptable.xml0000644000000000000000000000775412262561504016407 0ustar rootroot ptable Signal Generators:Table Access ptable ptable Accesses table values by direct indexing. Description Accesses table values by direct indexing. Syntax ares ptable andx, ifn [, ixmode] [, ixoff] [, iwrap] ires ptable indx, ifn [, ixmode] [, ixoff] [, iwrap] kres ptable kndx, ifn [, ixmode] [, ixoff] [, iwrap] Initialization ifn -- function table number. ixmode (optional) -- index data mode. The default value is 0. 0 = raw index 1 = normalized (0 to 1) ixoff (optional) -- amount by which index is to be offset. For a table with origin at center, use tablesize/2 (raw) or .5 (normalized). The default value is 0. iwrap (optional) -- wraparound index flag. The default value is 0. 0 = nowrap (index < 0 treated as index=0; index > tablesize sticks at index=size) 1 = wraparound. Performance ptable invokes table lookup on behalf of init, control or audio indices. These indices can be raw entry numbers (0, 1, 2,... size - 1) or scaled values (0 to 1). Indices are first modified by the offset value then checked for range before table lookup (see iwrap). If index is likely to be full scale, or if interpolation is being used, the table should have an extended guard point. table indexed by a periodic phasor ( see phasor) will simulate an oscillator. Examples Here is an example of the ptable opcode. It uses the file ptable.csd. Example of the ptable opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also table, tablei, table3, ptable3, ptablei, oscil1, oscil1i, osciln Credits Author: &namejohn; Jan 2012 New in Csound version 5.16 manual_src~dfsg/opcodes/comb.xml0000644000000000000000000000750412262561504016051 0ustar rootroot comb Signal Modifiers:Reverberation comb comb Reverberates an input signal with a colored frequency response. Description Reverberates an input signal with a colored frequency response. Syntax ares comb asig, krvt, ilpt [, iskip] [, insmps] Initialization ilpt -- loop time in seconds, which determines the echo density of the reverberation. This in turn characterizes the color of the comb filter whose frequency response curve will contain ilpt * sr/2 peaks spaced evenly between 0 and sr/2 (the Nyquist frequency). Loop time can be as large as available memory will permit. The space required for an n second loop is n*sr floating or double numbers (usually 4 or 8 bytes). Delay space is allocated and returned as in delay. iskip (optional, default=0) -- initial disposition of delay-loop data space (cf. reson). The default value is 0. insmps (optional, default=0) -- delay amount, as a number of samples. Performance krvt -- the reverberation time (defined as the time in seconds for a signal to decay to 1/1000, or 60dB down from its original amplitude). This filter reiterates input with an echo density determined by loop time ilpt. The attenuation rate is independent and is determined by krvt, the reverberation time (defined as the time in seconds for a signal to decay to 1/1000, or 60dB down from its original amplitude). Output from a comb filter will appear only after ilpt seconds. Examples Here is an example of the comb opcode. It uses the file comb.csd. Example of the comb opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also alpass, reverb, valpass, vcomb Credits &namebarry; manual_src~dfsg/opcodes/plusbecomes.xml0000644000000000000000000000442512262561504017451 0ustar rootroot = Instrument Control:Initialization and Reinitialization = = Performs add and assignment. Syntax ares += xarg ires += iarg kres += karg table [ kval] += karg Description Performs an add and assign. Initialization += - Adds the value of the expression iarg (karg, xarg) into the named result. This provides a means of saving an evaluated result for later use. Examples Here is an example of the plusbecomes opcode. It uses the file reverb.csd. Example of the plusbecomes opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also assign, Credits Author: &namejohn; University of Bath, and Codemist Ltd. Bath, UK 2013 New in version 6.00 manual_src~dfsg/opcodes/prints.xml0000644000000000000000000000535112262561504016446 0ustar rootroot prints Signal I/O:Printing and Display prints prints Prints at init-time using a printf() style syntax. Description Prints at init-time using a printf() style syntax. Syntax prints "string" [, kval1] [, kval2] [...] Initialization "string" -- the text string to be printed. Can be up to 8192 characters and must be in double quotes. Performance kval1, kval2, ... (optional) -- The k-rate values to be printed. These are specified in string with the standard C value specifier (%f, %d, etc.) in the order given. Use 0 for those which are not used. prints is similar to the printks opcode except it operates at init-time instead of k-rate. For more information about output formatting, please look at printks's documentation. Examples Here is an example of the prints opcode. It uses the file prints.csd. Example of the prints opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: 1234.568 ;;;;;;semicolons! See Also printks Credits Author: &namematt; January 2003 manual_src~dfsg/opcodes/dumpk.xml0000644000000000000000000001306712262561504016252 0ustar rootroot dumpk Signal I/O:File I/O dumpk dumpk Periodically writes an orchestra control-signal value to an external file. Description Periodically writes an orchestra control-signal value to a named external file in a specific format. Syntax dumpk ksig, ifilname, iformat, iprd Initialization ifilname -- character string (in double quotes, spaces permitted) denoting the external file name. May either be a full path name with target directory specified or a simple filename to be created within the current directory iformat -- specifies the output data format: 1 = 8-bit signed char(high order 8 bits of a 16-bit integer 4 = 16-bit short integers 5 = 32-bit long integers 6 = 32-bit floats 7 = ASCII long integers 8 = ASCII floats (2 decimal places) Note that A-law and U-law output are not available, and that all formats except the last two are binary. The output file contains no header information. iprd -- the period of ksig output in seconds, rounded to the nearest orchestra control period. A value of 0 implies one control period (the enforced minimum), which will create an output file sampled at the orchestra control rate. Performance ksig -- a control-rate signal This opcode allows a generated control signal value to be saved in a named external file. The file contains no self-defining header information. But it contains a regularly sampled time series, suitable for later input or analysis. There may be any number of dumpk opcodes in an instrument or orchestra but each must write to a different file. Examples Here is an example of the dumpk opcode. It uses the file dumpk.csd. Example of the dumpk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the dumpk opcode. It uses the file dumpk-2.csd. Example 2 of the dumpk opcode. Its output should include lines like this: WARNING: Seeding from current time 683384022 i 1 time 1.00033: 463.64510 i 1 time 2.00000: 463.64510 i 1 time 3.00000: 483.14200 i 1 time 4.00000: 567.55973 i 1 time 5.00000: 576.37060 i 1 time 6.00000: 460.66550 i 2 time 6.00033: 463.64510 i 2 time 7.00000: 463.64510 i 2 time 8.00000: 483.14200 i 2 time 9.00000: 567.55970 i 2 time 10.00000: 576.37060 i 2 time 11.00000: 460.66550 See Also dumpk2, dumpk3, dumpk4, readk, readk2, readk3, readk4 Credits By: John ffitch and Barry Vercoe 1999 or earlier manual_src~dfsg/opcodes/outch.xml0000644000000000000000000000706212262561504016252 0ustar rootroot outch Signal I/O:Signal Output outch outch Writes multi-channel audio data, with user-controllable channels, to an external device or stream. Description Writes multi-channel audio data, with user-controllable channels, to an external device or stream. Syntax outch kchan1, asig1 [, kchan2] [, asig2] [...] Performance outch outputs asig1 on the channel determined by kchan1, asig2 on the channel determined by kchan2, etc. The highest number for kchanX available for use with outch depends on nchnls. If kchanX is greater than nchnls, asigX will be silent. Note that outch will give a warning but not an error in this case. Examples Here is an example of the outch opcode. It uses the file outch.csd. Example of the outch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. signal is sent to channel 5 signal is sent to channel 4 signal is sent to channel 3 signal is sent to channel 2 signal is sent to channel 1 signal is sent to channel 5 ........ Here is another example of the outch opcode. It uses the file outch-2.csd. Another example of the outch opcode. See Also out32, outc, outx, outz More information on this opcode: http://www.csounds.com/journal/issue16/audiorouting.html , written by Andreas Russo Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/vaddv.xml0000644000000000000000000001352012262561504016230 0ustar rootroot vaddv Vectorial:Vectorial operations vaddv vaddv Performs addition between two vectorial control signals. Description Performs addition between two vectorial control signals. Syntax vaddv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] Initialization ifn1 - number of the table hosting the first vector to be processed ifn2 - number of the table hosting the second vector to be processed Performance kelements - number of elements of the two vectors kdstoffset - index offset for the destination (ifn1) table (Default=0) ksrcoffset - index offset for the source (ifn2) table (Default=0) kverbose - Selects whether or not warnings are printed (Default=0) vaddv adds two vectorial control signals, that is, each element of the first vector is processed (only) with the corresponding element of the other vector. Each vectorial signal is hosted by a table (ifn1 and ifn2). The number of elements contained in both vectors must be the same. The result is a new vectorial control signal that overrides old values of ifn1. If you want to keep the old ifn1 vector, use vcopy_iopcode to copy it in another table. You can use kdstoffset and ksrcoffset to specify vectors in any location of the tables. Negative values for kdstoffset and ksrcoffset are acceptable. If kdstoffset is negative, the out of range section of the vector will be discarded. If ksrcoffset is negative, the out of range elements will be assumed to be 0 (i.e. the destination elements will not be changed). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 0 (i.e. the destination vector will not be changed for these elements). Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at k-rate (this means that every k-pass the vectors are added). There's an i-rate version of this opcode called vaddv_i. Please note that the elements argument has changed in version 5.03 from i-rate to k-rate. This will change the opcode's behavior in the unusual cases where the i-rate variable ielements is changed inside the instrument, for example in: instr 1 ielements = 10 vadd 1, 1, ielements ielements = 20 vadd 2, 1, ielements turnoff endin All these operators (vaddv,vsubv,vmultv,vdivv,vpowv,vexpv, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Examples Here is an example of the vaddv opcode. It uses the file vaddv.csd. Example of the vaddv opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/pvsarp.xml0000644000000000000000000000667112262561504016450 0ustar rootroot pvsarp Spectral Processing:Streaming pvsarp pvsarp Arpeggiate the spectral components of a streaming pv signal. Description This opcode arpeggiates spectral components, by amplifying one bin and attenuating all the others around it. Used with an LFO it will provide a spectral arpeggiator similar to Trevor Wishart's CDP program specarp. Syntax fsig pvsarp fsigin, kbin, kdepth, kgain Performance fsig -- output pv stream fsigin -- input pv stream kbin -- target bin, normalised 0 - 1 (0Hz - Nyquist). kdepth -- depth of attenuation of surrounding bins kgain -- gain boost applied to target bin It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the pvsarp opcode. It uses the file pvsarp.csd Example of the pvsarp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the pvsarp opcode. It uses the file pvsarp2.csd Example of the pvsarp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namevictor; April 2005 New plugin in version 5 April 2005. manual_src~dfsg/opcodes/pvcross.xml0000644000000000000000000001404712262561504016630 0ustar rootroot pvcross Spectral Processing:STFT pvcross pvcross Applies the amplitudes from one phase vocoder analysis file to the data from a second file. Description pvcross applies the amplitudes from one phase vocoder analysis file to the data from a second file and then performs the resynthesis. The data is passed, as described above, from a previously called pvbufread unit. The two k-rate amplitude arguments are used to scale the amplitudes of each files separately before they are added together and used in the resynthesis (see below for further explanation). The frequencies of the first file are not used at all in this process. This unit simply allows for cross-synthesis through the application of the amplitudes of the spectra of one signal to the frequencies of a second signal. Unlike pvinterp, pvcross does allow for the use of the ispecwp as in pvoc and vpvoc. Syntax ares pvcross ktimpnt, kfmod, ifile, kampscale1, kampscale2 [, ispecwp] Initialization ifile -- the pvoc number (n in pvoc.n) or the name in quotes of the analysis file made using pvanal. (See pvoc.) ispecwp (optional, default=0) -- if non-zero, attempts to preserve the spectral envelope while its frequency content is varied by kfmod. The default value is zero. Performance ktimpnt -- the passage of time, in seconds, through this file. ktimpnt must always be positive, but can move forwards or backwards in time, be stationary or discontinuous, as a pointer into the analysis file. kfmod -- a control-rate transposition factor: a value of 1 incurs no transposition, 1.5 transposes up a perfect fifth, and 0.5 down an octave. kampscale1, kampscale2 -- used to scale the amplitudes stored in each frame of the phase vocoder analysis file. kampscale1 scale the amplitudes of the data from the file read by the previously called pvbufread. kampscale2 scale the amplitudes of the file named by ifile. By using these arguments, it is possible to adjust these values before applying the interpolation. For example, if file1 is much louder than file2, it might be desirable to scale down the amplitudes of file1 or scale up those of file2 before interpolating. Likewise one can adjust the frequencies of each to bring them more in accord with one another (or just the opposite, of course!) before the interpolation is performed. Examples Below is an example using pvbufread with pvcross. In this example the amplitudes used in the resynthesis gradually change from those of the oboe to those of the clarinet. The frequencies, of course, remain those of the clarinet throughout the process since pvcross does not use the frequency data from the file read by pvbufread. ktime1 line 0, p3, 3.5 ; used as index in the "oboe.pvoc" file ktime2 line 0, p3, 4.5 ; used as index in the "clar.pvoc" file kcross expon 0.001, p3, 1 pvbufread ktime1, "oboe.pvoc" apv pvcross ktime2, 1, "clar.pvoc", 1-kcross, kcross Here is a complete example of the pvcross opcode. It uses the file pvcross.csd Example of the pvcross opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvbufread, pvinterp, pvread, tableseg, tablexseg Credits Author: Richard Karpen Seattle, Wash 1997 New in version 3.44 manual_src~dfsg/opcodes/crossfm.xml0000644000000000000000000001364112262561504016604 0ustar rootroot crossfm Signal Generators:FM Synthesis crossfm crossfm Two mutually frequency and/or phase modulated oscillators. Description Two oscillators, mutually frequency and/or phase modulated by each other. Syntax a1, a2 crossfm xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] a1, a2 crossfmi xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] a1, a2 crosspm xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] a1, a2 crosspmi xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] a1, a2 crossfmpm xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] a1, a2 crossfmpmi xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] Initialization ifn1 -- function table number for oscillator #1. Requires a wrap-around guard point. ifn2 -- function table number for oscillator #2. Requires a wrap-around guard point. iphs1 (optional, default=0) -- initial phase of waveform in table ifn1, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. iphs2 (optional, default=0) -- initial phase of waveform in table ifn2, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. Performance xfrq1 -- a factor that, when multipled by the kcps parameter, gives the frequency of oscillator #1. xfrq2 -- a factor that, when multipled by the kcps parameter, gives the frequency of oscillator #2. xndx1 -- the index of the modulation of oscillator #2 by oscillator #1. xndx2 -- the index of the modulation of oscillator #1 by oscillator #2. kcps -- a common denominator, in cycles per second, for both oscillators frequencies. crossfm implements a crossed frequency modulation algorithm. The audio-rate output of oscillator #1 is used to modulate the frequency input of oscillator #2 while the audio-rate output of oscillator #2 is used to modulate the frequency input of oscillator #1. This double feedback structure produces a rich set of sounds with some chaotic behaviour. crossfmi behaves like crossfm except that linear interpolation is used for table lookup. crosspm and crosspmi implement cross phase modulation between two oscillators. crossfmpm and crossfmpmi implement cross frequency/phase modulation between two oscillators. Oscillator #1 is frequency-modulated by oscillator #2 while oscillator #2 is phase-modulated by oscillator #1. You can read my paper in the Csound Journal for more information. Warning Those opcodes may produce very rich spectra, especially with high modulation indexes, and in some cases foldover aliases may occur if the sampling rate is not high enough. Moreover the audio output may vary in function of the sampling rate, due to the non-linearity of the algorithm. In Csound, two other opcodes have this characteristic: planet and chuap. Examples Here is an example of the crossfm opcode. It uses the file crossfm.csd. Example of the crossfm opcode. In this example, an FLTK GUI is used to control in real-time the oscillators frequency with two Flcount widgets and the cross modulation indexes with one FLjoy widget. It uses a sampling rate of 96000Hz. See Also More information on this opcode: http://www.csounds.com/journal/issue12/crossfm.html , written by François Pinot Credits Author: &namepinot; 2005-2009 New in version 5.12 manual_src~dfsg/opcodes/cpumeter.xml0000644000000000000000000000510412262561504016747 0ustar rootroot cpumeter Instrument Control:Realtime Performance Control cpumeter cpumeter Reports the usage of cpu either total or per core. Description Reports the usage of cpu either total or per core to monitor how close to max-out the processing is. Syntax ktot[,kcpu1, kcpu2,...]cpumeter ifreq Initialization ifreq is the time in seconds that the meter is refreshed. If this is too low then mainly figures of zero or one hundred occur. A value of 0.1 seems acceptable. Performance cpumeter reads the total idle time in the last ifreq seconds and reports it as a percentage usage. If more than just ktot results are requested these report the same value for individual cores. Examples Here is an example of the cpumeter opcode. It uses the file cpumeter.csd. Example of the cpumeter opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also maxalloc, prealloc, cpuprc Credits Author: &namejohn; May 2011 New in Csound version 5.14, for Linux/Unix and OSX only manual_src~dfsg/opcodes/aunirand.xml0000644000000000000000000000077612262561504016736 0ustar rootroot aunirand aunirand aunirand Deprecated. Description Deprecated as of version 3.49. Use the unirand opcode instead. manual_src~dfsg/opcodes/fmrhode.xml0000644000000000000000000001045512262561504016554 0ustar rootroot fmrhode Signal Generators:FM Synthesis fmrhode fmrhode Uses FM synthesis to create a Fender Rhodes electric piano sound. Description Uses FM synthesis to create a Fender Rhodes electric piano sound. It comes from a family of FM sounds, all using 4 basic oscillators and various architectures, as used in the TX81Z synthesizer. Syntax ares fmrhode kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, \ ifn3, ifn4, ivfn Initialization All these opcodes take 5 tables for initialization. The first 4 are the basic inputs and the last is the low frequency oscillator (LFO) used for vibrato. The last table should usually be a sine wave. The initial waves should be: ifn1 -- sine wave ifn2 -- sine wave ifn3 -- sine wave ifn4 -- fwavblnk.aiff Note The file fwavblnk.aiff is also available at ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. kc1, kc2 -- Controls for the synthesizer: kc1 -- Mod index 1 kc2 -- Crossfade of two outputs Algorithm -- 5 kvdepth -- Vibrator depth kvrate -- Vibrator rate Examples Here is an example of the fmrhode opcode. It uses the file fmrhode.csd, and fwavblnk.aiff. Example of the fmrhode opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also fmb3, fmbell, fmmetal, fmpercfl, fmwurlie Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/pgmassign.xml0000644000000000000000000001070012262561504017111 0ustar rootroot pgmassign Real-time MIDI:Input pgmassign pgmassign Assigns an instrument number to a specified MIDI program. Description Assigns an instrument number to a specified (or all) MIDI program(s). By default, the instrument is the same as the program number. If the selected instrument is zero or negative or does not exist, the program change is ignored. This opcode is normally used in the orchestra header. Although, like massign, it also works in instruments. Syntax pgmassign ipgm, inst[, ichn] pgmassign ipgm, "insname"[, ichn] Initialization ipgm -- MIDI program number (1 to 128). A value of zero selects all programs. inst -- instrument number. If set to zero, or negative, MIDI program changes to ipgm are ignored. Currently, assignment to an instrument that does not exist has the same effect. This may be changed in a later release to print an error message. insname -- A string (in double-quotes) representing a named instrument. ichn (optional, defaults to zero) -- channel number. If zero, program changes are assigned on all channels. You can disable the turning on of any instruments by using the following in the header: massign 0, 0 pgmassign 0, 0 Examples Here is an example of the pgmassign opcode. It uses the file pgmassign.csd. Example of the pgmassign opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an example of the pgmassign opcode that will ignore program change events. It uses the file pgmassign_ignore.csd. Example of the pgmassign opcode that will ignore program change events. Here is an advanced example of the pgmassign opcode. It uses the file pgmassign_advanced.csd. Don't forget that you must include the -F flag when using an external MIDI file like pgmassign_advanced.mid. An advanced example of the pgmassign opcode. See Also midichn and massign Credits Author: &nameistvan; May 2002 New in version 4.20 manual_src~dfsg/opcodes/taninv.xml0000644000000000000000000000450612262561504016427 0ustar rootroot taninv Mathematical Operations:Trigonometric Functions taninv taninv Performs an arctangent function. Description Returns the arctangent of x (x in radians). Syntax taninv(x) (no rate restriction) Examples Here is an example of the taninv opcode. It uses the file taninv.csd. Example of the taninv opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: i1 = 0.464 See Also cos, cosh, cosinv, sin, sinh, sininv, tan, tanh, taninv2 Credits Author: &namejohn; New in version 3.48 Example written by &namekevin;. manual_src~dfsg/opcodes/serialEnd.xml0000644000000000000000000000304512262561504017033 0ustar rootroot serialEnd Serial I/O serialEnd serialEnd Close a serial port. Description Close a serial port for arduino. Syntax serialEnd iPort Initialization iPort -- port number optained from a serialBeginopcode. See Also serialBegin, serialWrite_i, serialWrite, serialRead, serialPrint, serialFlush. Credits Author: &namematt; 2011 New in version 5.14 manual_src~dfsg/opcodes/linsegr.xml0000644000000000000000000001272512262561504016575 0ustar rootroot linsegr Signal Generators:Linear and Exponential Generators linsegr linsegr Trace a series of line segments between specified points including a release segment. Description Trace a series of line segments between specified points including a release segment. Syntax ares linsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz kres linsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz Initialization ia -- starting value. ib, ic, etc. -- value after dur1 seconds, etc. idur1 -- duration in seconds of first segment. A zero or negative value will cause all initialization to be skipped. idur2, idur3, etc. -- duration in seconds of subsequent segments. A zero or negative value will terminate the initialization process with the preceding point, permitting the last-defined line or curve to be continued indefinitely in performance. The default is zero. irel, iz -- duration in seconds and final value of a note releasing segment. For Csound versions prior to 5.00, the release time cannot be longer than 32767/kr seconds. This limit has been extended to (231-1)/kr. Performance These units generate control or audio signals whose values can pass through 2 or more specified points. The sum of dur values may or may not equal the instrument's performance time: a shorter performance will truncate the specified pattern, while a longer one will cause the last-defined segment to continue on in the same direction. linsegr is amongst the Csound r units that contain a note-off sensor and release time extender. When each senses an event termination or MIDI noteoff, it immediately extends the performance time of the current instrument by irel seconds, and sets out to reach the value iz by the end of that period (no matter which segment the unit is in). r units can also be modified by MIDI noteoff velocities. For two or more extenders in an instrument, extension is by the greatest period. You can use other pre-made envelopes which start a release segment upon recieving a note off message, like linenr and expsegr, or you can construct more complex envelopes using xtratim and release. Note that you don't need to use xtratim if you are using linsegr, since the time is extended automatically. Examples Here is an example of the linsegr opcode. It uses the file linsegr.csd. Example of the linsegr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also linenr, expsegr, envlpxr, mxadsr, madsr expon, expseg, expsega line, linseg, xtratim, transegr Credits Author: &namebarry; December 2002, December 2006. Thanks to &nameistvan;, added documentation about the maximum release time. New in Csound 3.47 manual_src~dfsg/opcodes/ftsavek.xml0000644000000000000000000000423312262561504016570 0ustar rootroot ftsavek Table Control:Read/Write Opreations ftsavek ftsavek Save a set of previously-allocated tables to a file. Description Save a set of previously-allocated tables to a file. Syntax ftsavek "filename", ktrig, iflag, ifn1 [, ifn2] [...] Initialization "filename" -- A quoted string containing the name of the file to save. iflag -- Type of the file to save. (0 = binary file, Non-zero = text file) ifn1, ifn2, ... -- Numbers of tables to save. Performance ktrig -- The trigger signal. Save the file each time it is non-zero. ftsavek saves a list of tables to a file. The file's format can be binary or text. Unlike ftsave, the saving operation can be repeated numerous times within the same note by using a trigger signal. Warning The file's format is not compatible with a WAV-file and is not endian-safe. See Also ftloadk, ftload, ftsave Credits Author: &namegabriel; New in version 4.21 manual_src~dfsg/opcodes/cpstuni.xml0000644000000000000000000001164212262561504016614 0ustar rootroot cpstuni Pitch Converters:Tuning Opcodes cpstuni cpstuni Returns micro-tuning values at init-rate. Description Returns micro-tuning values at init-rate. Syntax icps cpstuni index, ifn Initialization icps -- Return value in cycles per second. index -- An integer number denoting an index of scale. ifn -- Function table containing the parameters (numgrades, interval, basefreq, basekeymidi) and the tuning ratios. Performance These opcodes are similar to cpstmid, but work without necessity of MIDI. cpstuni works at init-rate. It allows fully customized micro-tuning scales. It requires a function table number containing the tuning ratios, and some other parameters stored in the function table itself. The index argument should be filled with integer numbers expressing the grade of given scale to be converted in cps. The function table ifn should be generated by GEN02 and the first four values stored in this function are parameters that express: numgrades -- The number of grades of the micro-tuning scale. interval -- The frequency range covered before repeating the grade ratios, for example 2 for one octave, 1.5 for a fifth etcetera. basefreq -- The base frequency of the scale in cycles per second. basekey -- The integer index of the scale to which to assign basefreq unmodified. After these four values, the user can begin to insert the tuning ratios. For example, for a standard 12-grade scale with the base-frequency of 261 cps assigned to the key-number 60, the corresponding f-statement in the score to generate the table should be: ; numgrades basefreq tuning-ratios (eq.temp) ....... ; interval basekey f1 0 64 -2 12 2 261 60 1 1.059463 1.12246 1.18920 ..etc... Another example with a 24-grade scale with a base frequency of 440 assigned to the key-number 48, and a repetition interval of 1.5: ; numgrades basefreq tuning-ratios ....... ; interval basekey f1 0 64 -2 24 1.5 440 48 1 1.01 1.02 1.03 ..etc... Examples Here is an example of the cpstuni opcode. It uses the file cpstuni.csd. Example of the cpstuni opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: i1 = 440.110 See Also cpstmid, cpstun, GEN02 Credits Written by &namegabriel;. Example written by &namekevin;. manual_src~dfsg/opcodes/pvsfwrite.xml0000644000000000000000000000435012262561504017156 0ustar rootroot pvsfwrite Spectral Processing:Streaming pvsfwrite pvsfwrite Write a fsig to a PVOCEX file. Description This opcode writes a fsig to a PVOCEX file (which in turn can be read by pvsfread or other programs that support PVOCEX file input). Syntax pvsfwrite fsig, ifile Initialisation fsig -- fsig input data. ifile -- filename (a string in double-quotes) . Examples Here is an example of the pvsfwrite opcode. It uses the file pvsfwrite.csd. This example uses realtime audio input. Example of the pvsfwrite opcode See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namevictor; November 2004 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/powoftwo.xml0000644000000000000000000000606612262561504017017 0ustar rootroot powoftwo Mathematical Operations:Mathematical Functions powoftwo powoftwo Performs a power-of-two calculation. Description Performs a power-of-two calculation. Syntax powoftwo(x) (init-rate or control-rate args only) Performance powoftwo() function returns 2x and allows positive and negatives numbers as argument. The range of values admitted in powoftwo() is -5 to +5 allowing a precision more fine than one cent in a range of ten octaves. If a greater range of values is required, use the slower opcode pow. These functions are fast, because they read values stored in tables. Also they are very useful when working with tuning ratios. They work at i- and k-rate. Examples Here is an example of the powoftwo opcode. It uses the file powoftwo.csd. Example of the powoftwo opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: iQuantize = 32768.000 instr 1: iQuantize = 2048.000 instr 1: iQuantize = 128.000 instr 1: iQuantize = 8.000 instr 1: iQuantize = 2.000 instr 1: iQuantize = 1.000 See Also logbtwo, pow Credits Author: &namegabriel; Italy June 1998 Author: &namejohn; University of Bath, Codemist, Ltd. Bath, UK July 1999 New in Csound version 3.57 manual_src~dfsg/opcodes/opbitand.xml0000644000000000000000000000424212262561504016725 0ustar rootroot & Mathematical Operations:Arithmetic and Logic Operations & & Bitwise AND operator. Description The bitwise operators perform operations of bitwise AND, bitwise OR, bitwise NOT and bitwise non-equivalence. Syntax a & b (bitwise AND) where the arguments a and b may be further expressions. They are converted to the nearest integer to machine precision and then the operation is performed. Performance The priority of these operators is less binding that the arithmetic ones, but more binding that the comparisons. Parentheses may be used as above to force particular groupings. Examples Here is an example of the bitwise AND and OR operators. It uses the file bitwise.csd. Example of the bitwise operators. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also |, #, ¬ manual_src~dfsg/opcodes/cosinv.xml0000644000000000000000000000437512262561504016435 0ustar rootroot cosinv Mathematical Operations:Trigonometric Functions cosinv cosinv Performs a arccosine function. Description Returns the arccosine of x (x in radians). Syntax cosinv(x) (no rate restriction) Examples Here is an example of the cosinv opcode. It uses the file cosinv.csd. Example of the cosinv opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: i1 = 1.047 See Also cos, cosh, sin, sinh, sininv, tan, tanh, taninv Credits Author: &namejohn; New in version 3.48 Example written by &namekevin;. manual_src~dfsg/opcodes/klinrand.xml0000644000000000000000000000077612262561504016737 0ustar rootroot klinrand klinrand klinrand Deprecated. Description Deprecated as of version 3.49. Use the linrand opcode instead. manual_src~dfsg/opcodes/spsend.xml0000644000000000000000000000702012262561504016416 0ustar rootroot spsend Signal Modifiers:Panning and Spatialization spsend spsend Generates output signals based on a previously defined space opcode. Description spsend depends upon the existence of a previously defined space. The output signals from spsend are derived from the values given for xy and reverb in the space and are ready to be sent to local or global reverb units (see example below). Syntax a1, a2, a3, a4 spsend Performance The configuration of the xy coordinates in space places the signal in the following way: a1 is -1, 1 a2 is 1, 1 a3 is -1, -1 a4 is 1, -1 This assumes a loudspeaker set up as a1 is left front, a2 is right front, a3 is left back, a4 is right back. Values greater than 1 will result in sounds being attenuated, as if in the distance. space considers the speakers to be at a distance of 1; smaller values of xy can be used, but space will not amplify the signal in this case. It will, however balance the signal so that it can sound as if it were within the 4 speaker space. x=0, y=1, will place the signal equally balanced between left and right front channels, x=y=0 will place the signal equally in all 4 channels, and so on. Although there must be 4 output signals from space, it can be used in a 2 channel orchestra. If the xy's are kept so that Y>=1, it should work well to do panning and fixed localization in a stereo field. Examples Here is a stereo example of the spsend opcode. It uses the file spsend.csd. Example of the spsend opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also space, spdist Credits Author: Richard Karpen Seattle, WA USA 1998 New in Csound version 3.48 manual_src~dfsg/opcodes/hrtfmove.xml0000644000000000000000000001305412262561504016760 0ustar rootroot hrtfmove Signal Modifiers:Panning and Spatialization hrtfmove hrtfmove Generates dynamic 3d binaural audio for headphones using magnitude interpolation and phase truncation. Description This opcode takes a source signal and spatialises it in the 3 dimensional space around a listener by convolving the source with stored head related transfer function (HRTF) based filters. Syntax aleft, aright hrtfmove asrc, kAz, kElev, ifilel, ifiler [, imode, ifade, isr] Initialization ifilel -- left HRTF spectral data file ifiler -- right HRTF spectral data file Spectral datafiles (based on the MIT HRTF database) are available in 3 different sampling rates: 44.1, 48 and 96 khz and are labelled accordingly. Input and processing sr should match datafile sr. Files should be in the current directory or the SADIR (see Environment Variables). HRTF Data files for use with hrtfmove, hrtfmove2, hrtfstat, hrtfearly, hrtfreverb were updated for Csound 5.15 and later (the code was updated and is more efficient). Old datafiles are now deprecated. imode -- optional, default 0 for phase truncation, 1 for minimum phase ifade -- optional, number of processing buffers for phase change crossfade (default 8). Legal range is 1-24. A low value is recommended for complex sources (4 or less: a higher value may make the crossfade audible), a higher value (8 or more: a lower value may make the inconsistency when the filter changes phase values audible) for narrowband sources. Does not effect minimum phase processing. Ocassionally fades may overlap (when unnaturally fast/complex trajectories are requested). In this case, a warning will be printed. Use a smaller crossfade or slightly change trajectory to avoid any possible inconsistencies that may arise. isr - optional, default 44.1kHz, legal values: 44100, 48000 and 96000. Performance kAz -- azimuth value in degrees. Positive values represent position on the right, negative values are positions on the left. kElev -- elevation value in degrees. Positive values represent position above horizontal, negative values are positions below horizontal (min -40). Artifact-free user-defined trajectories are made possible using an interpolation algorithm based on spectral magnitude interpolation and phase truncation. Crossfades are implemented to minimise/eliminate any inconsistencies caused by updating phase values. These crossfades are performed over a user definable number of convolution processing buffers. Complex sources may only need to crossfade over 1 buffer; narrow band sources may need several. The opcode also offers minimum phase based processing, a more traditional and complex method. In this mode, the hrtf filters used are reduced to minimum phase representations and the interpolation process then uses the relationship between minimum phase magnitude and phase spectra. Interaural time difference, which is inherent to the phase truncation process, is reintroduced in the minimum phase process using variable delay lines. Examples Here is an example of the hrtfmove opcode. It uses the file hrtfmove.csd. Example of the htrfmove opcode. See Also hrtfmove2, hrtfstat, hrtfer. More information on this opcode: http://www.csounds.com/journal/issue9/newHRTFOpcodes.html , written by Brian Carty Credits Author: Brian Carty Maynooth 2008 manual_src~dfsg/opcodes/STKDrummer.xml0000644000000000000000000000462412262561504017126 0ustar rootroot STKDrummer Signal Generators:STK Opcodes STKDrummer STKDrummer STKDrummer is a drum sampling synthesizer. Description STKDrummer is a drum sampling synthesizer using raw waves and one-pole filters, The drum rawwave files are sampled at 22050 Hz, but will be appropriately interpolated for other sample rates. Syntax asignal STKDrummer ifrequency, iamplitude Initialization ifrequency -- Samples being played. iamplitude -- Amplitude of note played (range 0-1). Note The code for this opcode is taken directly from the Drummer class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html Examples Here is an example of the STKDrummer opcode. It uses the file STKDrummer.csd. Example of the STKDrummer opcode. See Also STKClarinet. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/ptablei.xml0000644000000000000000000001026112262561504016543 0ustar rootroot ptablei Signal Generators:Table Access ptablei ptablei Accesses table values by direct indexing with linear interpolation. Description Accesses table values by direct indexing with linear interpolation. Syntax ares ptablei andx, ifn [, ixmode] [, ixoff] [, iwrap] ires ptablei indx, ifn [, ixmode] [, ixoff] [, iwrap] kres ptablei kndx, ifn [, ixmode] [, ixoff] [, iwrap] Initialization ifn -- function table number. The table need not be a power of 2 long. ixmode (optional) -- index data mode. The default value is 0. 0 = raw index 1 = normalized (0 to 1) ixoff (optional) -- amount by which index is to be offset. For a table with origin at center, use tablesize/2 (raw) or .5 (normalized). The default value is 0. iwrap (optional) -- wraparound index flag. The default value is 0. 0 = nowrap (index < 0 treated as index=0; index > tablesize sticks at index=size) 1 = wraparound. Performance ptablei is a interpolating unit in which the fractional part of index is used to interpolate between adjacent table entries. The smoothness gained by interpolation is at some small cost in execution time (see also oscili, etc.), but the interpolating and non-interpolating units are otherwise interchangeable. Examples Here is an example of the ptablei opcode. It uses the file ptablei.csd. Example of the ptablei opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also table, tablei, table3, ptable, ptable3, oscil1, oscil1i, osciln More information on this opcode: http://www.csounds.com/journal/issue12/genInstruments.html , written by Jacob Joaquin Credits Author: &namejohn; Jan 2012 New in Csound version 5.16 manual_src~dfsg/opcodes/setksmps.xml0000644000000000000000000001224512262561504017000 0ustar rootroot setksmps Signal I/O:Software Bus setksmps setksmps Sets the local ksmps value in an instrument or user-defined opcode block Description Sets the local ksmps value in an instrument or user-defined opcode block. Syntax setksmps iksmps Initialization iksmps -- sets the local ksmps value. If iksmps is set to zero, the ksmps of the caller instrument or opcode is used (this is the default behavior). Note The local ksmps is implemented by splitting up a control period into smaller sub-kperiods and setting up the instrument local ksmps to the new value. This also requires converting the rate of k-rate input and output arguments (input variables receive the same value in all sub-kperiods, while outputs are written only in the last one). It also means that you cannot use a local ksmps that is higher than the global ksmps. Warning about local ksmps When the local ksmps is not the same as the orchestra level ksmps value (as specified in the orchestra header), global a-rate operations must be carefully coded to access the data in a vector according to the local ksmps. The audio rate bus channel opcodes (chnget/chnset) can be used freely, however, as they will do the necessary conversion between ksmp sizes. Other opcodes that require some care include: any access to ga variables a-rate zak opcodes (zar, zaw, etc.) tablera and tablewa (these two opcodes may in fact work, but caution is needed) The in and out opcode family cannot be used in local-ksmps UDOs (these read from, and write to global a-rate buffers), but are safe in local-ksmps instruments In general, the local ksmps should be used with care as it is an experimental feature. Though it works correctly in most cases. The setksmps statement can be used to set the local ksmps value of the instrument or user-defined opcode block. It has one i-time parameter specifying the new ksmps value (which is left unchanged if zero is used). setksmps should be used before any other opcodes (but allowed after xin in UDOs), otherwise unpredictable results may occur. Performance The syntax of a user-defined opcode block is as follows: opcode name, outtypes, intypes xinarg1 [, xinarg2] [, xinarg3] ... [xinargN] xin [setksmps iksmps] ... the rest of the instrument's code. xout xoutarg1 [, xoutarg2] [, xoutarg3] ... [xoutargN] endop The new opcode can then be used with the usual syntax: [xinarg1] [, xinarg2] ... [xinargN] name [xoutarg1] [, xoutarg2] ... [xoutargN] [, iksmps] Examples See the example for the opcode opcode. See Also endop, opcode, xin, xout Credits Author: Istvan Varga, 2002; based on code by Matt J. Ingalls; Modified by V Lazzarini for Csound 6 New in version 4.22 manual_src~dfsg/opcodes/kunirand.xml0000644000000000000000000000077612262561504016750 0ustar rootroot kunirand kunirand kunirand Deprecated. Description Deprecated as of version 3.49. Use the unirand opcode instead. manual_src~dfsg/opcodes/mandol.xml0000644000000000000000000000625312262561504016403 0ustar rootroot mandol Signal Generators:Models and Emulations mandol mandol An emulation of a mandolin. Description An emulation of a mandolin. Syntax ares mandol kamp, kfreq, kpluck, kdetune, kgain, ksize, ifn [, iminfreq] Initialization ifn -- table number containing the pluck wave form. The file mandpluk.aiff is suitable for this. It is also available at ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. iminfreq (optional, default=0) -- Lowest frequency to be played on the note. If it is omitted it is taken to be the same as the initial kfreq. Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. kpluck -- The pluck position, in range 0 to 1. Suggest 0.4. kdetune -- The proportional detuning between the two strings. Suggested range 0.9 to 1. kgain -- the loop gain of the model, in the range 0.97 to 1. ksize -- The size of the body of the mandolin. Range 0 to 2. Examples Here is an example of the mandol opcode. It uses the file mandol.csd, and mandpluk.aiff. Example of the mandol opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/cent.xml0000644000000000000000000000470212262561504016057 0ustar rootroot cent Pitch Converters:Functions cent cent Calculates a factor to raise/lower a frequency by a given amount of cents. Description Calculates a factor to raise/lower a frequency by a given amount of cents. Syntax cent(x) This function works at a-rate, i-rate, and k-rate. Initialization x -- a value expressed in cents. Performance The value returned by the cent function is a factor. You can multiply a frequency by this factor to raise/lower it by the given amount of cents. Examples Here is an example of the cent opcode. It uses the file cent.csd. Example of the cent opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: instr 1: iroot = 440.000 instr 2: iroot = 440.000 instr 2: ifactor = 1.189 instr 2: inew = 523.251 instr 2: iroot = 440.000 instr 2: ifactor = 2.000 instr 2: inew = 880.000 See Also db, octave, semitone New in version 4.16 manual_src~dfsg/opcodes/vexp_i.xml0000644000000000000000000001022112262561504016411 0ustar rootroot vexp_i Vectorial:Scalar operations vexp_i vexp_i Performs power-of operations between a vector and a scalar Description Performs power-of operations between a vector and a scalar Syntax vexp_i ifn, ival, ielements[, idstoffset] Initialization ifn - number of the table hosting the vectorial signal to be processed ival - scalar operand to be processed ielements - number of elements of the vector ival - scalar value to be added idstoffset - index offset for the destination table Performance vexp_i rises ival to each element contained in a vector from table ifn, starting from table index idstoffset. This enables you to process a specific section of a table by specifying the offset and the number of elements to be processed. Offset is counted starting from 0, so if no offset is specified (or set to 0), the table will be modified from the beginning. Negative values for idstoffset are valid. Elements from the vector that are outside the table, will be discarded, and they will not wrap around the table. This opcode runs only on initialization, there is a k-rate version of this opcode called vexp. In all these opcodes, the resulting vectors are stored in ifn, overriding the initial vectors. If you want to keep initial vector, use vcopy or vcopy_i to copy it in another table. All these operators are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2, etc. They can also be useful in conjunction with the spectral opcodes pvsftw and pvsftr. Examples Here is an example of the vexp_i opcode. It uses the file vexp_i.csd. Example of the vexp_i opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also vadd, vmult_i, vpow_i and vexp_i. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/printk.xml0000644000000000000000000000674012262561504016441 0ustar rootroot printk Signal I/O:Printing and Display printk printk Prints one k-rate value at specified intervals. Description Prints one k-rate value at specified intervals. Syntax printk itime, kval [, ispace] Initialization itime -- time in seconds between printings. ispace (optional, default=0) -- number of spaces to insert before printing. (default: 0, max: 130) Performance kval -- The k-rate values to be printed. printk prints one k-rate value on every k-cycle, every second or at intervals specified. First the instrument number is printed, then the absolute time in seconds, then a specified number of spaces, then the kval value. The variable number of spaces enables different values to be spaced out across the screen - so they are easier to view. This opcode can be run on every k-cycle it is run in the instrument. To every accomplish this, set itime to 0. When itime is not 0, the opcode print on the first k-cycle it is called, and subsequently when every itime period has elapsed. The time cycles start from the time the opcode is initialized - typically the initialization of the instrument. Examples Here is an example of the printk opcode. It uses the file printk.csd. Example of the printk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: i 1 time 0.00002: 0.00000 i 1 time 1.00002: 20.01084 i 1 time 2.00002: 40.02999 i 1 time 3.00002: 60.04914 i 1 time 4.00002: 79.93327 See Also printk2 and printks Credits Author: Robin Whittle Australia May 1997 Example written by &namekevin;. Thanks goes to Luis Jure for pointing out a mistake with the itime parameter. manual_src~dfsg/opcodes/peakk.xml0000644000000000000000000000075112262561504016221 0ustar rootroot peakk peakk peakk Deprecated. Description Deprecated as of version 3.63. Use the peak opcode instead. manual_src~dfsg/opcodes/downsamp.xml0000644000000000000000000000464112262561504016760 0ustar rootroot downsamp Signal Modifiers:Sample Level Operators downsamp downsamp Modify a signal by down-sampling. Description Modify a signal by down-sampling. Syntax kres downsamp asig [, iwlen] Initialization iwlen (optional) -- window length in samples over which the audio signal is averaged to determine a downsampled value. Maximum length is ksmps; 0 and 1 imply no window averaging. The default value is 0. Performance downsamp converts an audio signal to a control signal by downsampling. It produces one kval for each audio control period. The optional window invokes a simple averaging process to suppress foldover. Examples Here is an example of the downsamp opcode. It uses the file downsamp.csd. Example of the downsamp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also diff, integ, interp, samphold, upsamp manual_src~dfsg/opcodes/tempoval.xml0000644000000000000000000000504612262561504016757 0ustar rootroot tempoval Instrument Control:Sensing and Control tempoval tempoval Reads the current value of the tempo. Description Reads the current value of the tempo. Syntax kres tempoval Performance kres -- the value of the tempo. If you use a positive value with the -t command-line flag, tempoval returns the percentage increase/decrease from the original tempo of 60 beats per minute. If you don't, its value will be 60 (for 60 beats per minute). Examples Here is an example of the tempoval opcode. Remember, it only works if you use the -t flag with Csound. It uses the file tempoval.csd. Example of the tempoval opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Since 120 beats per minute is a 50% increase over the original 60 beats per minute, its output should include lines like: kval = 0.500000 See Also tempo and miditempo Credits Example written by &namekevin;. New in version 4.15 December 2002. Thanks to Drake Wilson for pointing out unclear documentation. manual_src~dfsg/opcodes/mixersetlevel_i.xml0000644000000000000000000000472512262561504020333 0ustar rootroot MixerSetLevel_i Mixer Opcodes MixerSetLevel_i MixerSetLevel_i Sets the level of a send to a buss. Syntax MixerSetLevel_i isend, ibuss, igain Description Sets the level at which signals from the send are added to the buss. This opcode, because all parameters are irate, may be used in the orchestra header. The actual sending of the signal to the buss is performed by the MixerSend opcode. Initialization isend -- The number of the send, for example the number of the instrument sending the signal (but any integer can be used). ibuss -- The number of the buss, for example the number of the instrument receiving the signal (but any integer can be used). igain -- The level (any real number) at which the signal from the send will be mixed onto the buss. The default is 0. Setting the gain for a buss also creates the buss. Performance Use of the mixer requires that instruments setting gains have smaller numbers than instruments sending signals, and that instruments sending signals have smaller numbers than instruments receiving those signals. However, an instrument may have any number of sends or receives. After the final signal is received, MixerClear must be invoked to reset the busses before the next kperiod. Examples In the orchestra header, set the gain for the send from buss 3 to buss 4: MixerSetLevel_i 3, 4, 0.76 Credits Michael Gogins (gogins at pipeline dot com). manual_src~dfsg/opcodes/midinoteonoct.xml0000644000000000000000000001117312262561504020001 0ustar rootroot midinoteonoct Real-time MIDI:MIDI/Score Interoperability midinoteonoct midinoteonoct Gets a MIDI note number value as octave-point-decimal value. Description midinoteonoct is designed to simplify writing instruments that can be used interchangeably for either score or MIDI input, and to make it easier to adapt instruments originally written for score input to work with MIDI input. In general, it should be possible to write instrument definitions that work identically with both scores and MIDI, including both MIDI files and real-time MIDI input, without using any conditional statements, and that take full advantage of MIDI voice messages. Note that correlating Csound instruments with MIDI channel numbers is done using the massign opcode for real-time performance,. For file-driven performance, instrument numbers default to MIDI channel number + 1, but the defaults are overridden by any MIDI program change messages in the file. Syntax midinoteonoct xoct, xvelocity Performance xoct -- returns MIDI key translated to linear octaves during MIDI activation, remains unchanged otherwise. xvelocity -- returns MIDI velocity during MIDI activation, remains unchanged otherwise. If the instrument was activated by MIDI input, the opcode overwrites the values of xoct and xvelocity with the corresponding value from MIDI input. If the instrument was NOT activated by MIDI input, the values of xoct and xvelocity remain unchanged. This enables score p-fields to receive MIDI input data during MIDI activation, and score values otherwise. Adapting a score-activated Csound instrument. See the MIDI interop opcodes section for details on adapting score driven instruments for MIDI or vice-versa. Examples Here is an example of the midinoteonoct opcode. It uses the file midinoteonoct.csd. Example of the midinoteonoct opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i1 8.00000 i1 8.91700 i1 9.33333 See Also midichannelaftertouch, midicontrolchange, mididefault, midinoteoff, midinoteoncps, midinoteonkey, midinoteonpch, midipitchbend, midipolyaftertouch, midiprogramchange Credits Author: Michael Gogins New in version 4.20 manual_src~dfsg/opcodes/adsynt.xml0000644000000000000000000000774012262561504016435 0ustar rootroot adsynt Signal Generators:Additive Synthesis/Resynthesis adsynt adsynt Performs additive synthesis with an arbitrary number of partials, not necessarily harmonic. Description Performs additive synthesis with an arbitrary number of partials, not necessarily harmonic. Syntax ares adsynt kamp, kcps, iwfn, ifreqfn, iampfn, icnt [, iphs] Initialization iwfn -- table containing a waveform, usually a sine. Table values are not interpolated for performance reasons, so larger tables provide better quality. ifreqfn -- table containing frequency values for each partial. ifreqfn may contain beginning frequency values for each partial, but is usually used for generating parameters at runtime with tablew. Frequencies must be relative to kcps. Size must be at least icnt. iampfn -- table containing amplitude values for each partial. iampfn may contain beginning amplitude values for each partial, but is usually used for generating parameters at runtime with tablew. Amplitudes must be relative to kamp. Size must be at least icnt. icnt -- number of partials to be generated iphs -- initial phase of each oscillator, if iphs = -1, initialization is skipped. If iphs > 1, all phases will be initialized with a random value. Performance kamp -- amplitude of note kcps -- base frequency of note. Partial frequencies will be relative to kcps. Frequency and amplitude of each partial is given in the two tables provided. The purpose of this opcode is to have an instrument generate synthesis parameters at k-rate and write them to global parameter tables with the tablew opcode. Examples Here is an example of the adsynt opcode. It uses the file adsynt.csd. These two instruments perform additive synthesis. The output of each sounds like a Tibetan bowl. The first one is static, as parameters are only generated at init-time. In the second one, parameters are continuously changed. Example of the adsynt opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Peter Neubäcker Munich, Germany August, 1999 New in Csound version 3.58 manual_src~dfsg/opcodes/copyf2array.xml0000644000000000000000000000402412262561504017364 0ustar rootroot copyf2array Array Opcodes copyf2array copyf2array Copy data from an f-table to a vector. Description The copyf2array opcode takes an f-table and copies the contents to a t-var. Syntax copyf2array tab, kftbl Performance tab -- tables for destination. kftbl -- f-tables for source. Example Here is an example of the copyf2array opcode. It uses the file copyf2array.csd. Example of the copyf2array opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also copya2ftab Credits Author: &namejohn; October 2011 New in Csound version 5.15 Renamed in Csound version 6.00 manual_src~dfsg/opcodes/specscal.xml0000644000000000000000000000567612262561504016736 0ustar rootroot specscal Spectral Processing:Non-Standard specscal specscal Scales an input spectral datablock with spectral envelopes. Description Scales an input spectral datablock with spectral envelopes. Syntax wsig specscal wsigin, ifscale, ifthresh Initialization ifscale -- scale function table. A function table containing values by which a value's magnitude is rescaled. ifthresh -- threshold function table. If ifthresh is non-zero, each magnitude is reduced by its corresponding table-value (to not less than zero) Performance wsig -- the output spectrum wsigin -- the input spectra Scales an input spectral datablock with spectral envelopes. Function tables ifthresh and ifscale are initially sampled across the (logarithmic) frequency space of the input spectrum; then each time a new input spectrum is sensed the sampled values are used to scale each of its magnitude channels as follows: if ifthresh is non-zero, each magnitude is reduced by its corresponding table-value (to not less than zero); then each magnitude is rescaled by the corresponding ifscale value, and the resulting spectrum written to wsig. Examples wsig2 specdiff wsig1 ; sense onsets wsig3 specfilt wsig2, 2 ; absorb slowly specdisp wsig2, 0.1 ; & display both spectra specdisp wsig3, 0.1 See Also specaddm, specdiff, specfilt, spechist manual_src~dfsg/opcodes/apow.xml0000644000000000000000000000074212262561504016074 0ustar rootroot apow apow apow Deprecated. Description Deprecated as of version 3.48. Use the pow opcode instead. manual_src~dfsg/opcodes/islider32.xml0000644000000000000000000000100512262561504016717 0ustar rootroot islider32 islider32 islider32 Deprecated. Description Deprecated as of version 3.52. Use the slider32 opcode instead. manual_src~dfsg/opcodes/vdelayxws.xml0000644000000000000000000001026212262561504017152 0ustar rootroot vdelayxws Signal Modifiers:Delay vdelayxws vdelayxws Variable delay opcodes with high quality interpolation. Description Variable delay opcodes with high quality interpolation. Syntax aout1, aout2 vdelayxws ain1, ain2, adl, imd, iws [, ist] Initialization imd -- max. delay time (seconds) iws -- interpolation window size (see below) ist (optional) -- skip initialization if not zero Performance ain1, ain2 -- input audio signals aout1, aout2 -- output audio signals adl -- delay time in seconds These opcodes use high quality (and slow) interpolation, that is much more accurate than the currently available linear and cubic interpolation. The iws parameter sets the number of input samples used for calculating one output sample (allowed values are any integer multiply of 4 in the range 4 - 1024); higher values mean better quality and slower speed. The vdelayxw opcodes change the position of the write tap in the delay line (unlike all other delay ugens that move the read tap), and are most useful for implementing Doppler effects where the position of the listener is fixed, and the sound source is moving. The multichannel opcodes (eg. vdelayxq) allow delaying 2 or 4 variables at once (stereo or quad signals); this is much more efficient than using separate opcodes for each channel. Notes Delay time is measured in seconds (unlike in vdelay and vdelay3), and must be a-rate. The minimum allowed delay is iws/2 samples. Using the same variables as input and output is allowed in these opcodes. In vdelayxw*, changing the delay time has some effects on output volume: a = 1 / (1 + dt) where a is the output gain, and dt is the change of delay time per seconds. These opcodes are best used in the double-precision version of Csound. Examples Here is an example of the use of the vdelayxws opcode. It uses the file vdelayxws.csd. Example of the <emphasis>vdelayxws</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vdelayx, vdelayxq, vdelayxs, vdelayxw, vdelayxwq manual_src~dfsg/opcodes/pvsfread.xml0000644000000000000000000000566312262561504016747 0ustar rootroot pvsfread Spectral Processing:Streaming pvsfread pvsfread Read a selected channel from a PVOC-EX analysis file. Description Create an fsig stream by reading a selected channel from a PVOC-EX analysis file loaded into memory, with frame interpolation. Only format 0 files (amplitude+frequency) are currently supported. The operation of this opcode mirrors that of pvoc, but outputs an fsig instead of a resynthesized signal. Syntax fsig pvsfread ktimpt, ifn [, ichan] Initialization ifn -- Name of the analysis file. This must have the .pvx file extension. A multi-channel PVOC-EX file can be generated using the extended pvanal utility. ichan -- (optional) The channel to read (counting from 0). Default is 0. Performance ktimpt -- Time pointer into analysis file, in seconds. See the description of the same parameter of pvoc for usage. Note that analysis files can be very large, especially if multi-channel. Reading such files into memory will very likely incur breaks in the audio during real-time performance. As the file is read only once, and is then available to all other interested opcodes, it can be expedient to arrange for a dedicated instrument to preload all such analysis files at startup. Examples Here is an example of the pvsfread opcode. It uses the file pvsfread.csd. Example of the pvsfread opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namerichard; August 2001 New in version 4.13 manual_src~dfsg/opcodes/expseg.xml0000644000000000000000000000745612262561504016432 0ustar rootroot expseg Signal Generators:Linear and Exponential Generators expseg expseg Trace a series of exponential segments between specified points. Description Trace a series of exponential segments between specified points. Syntax ares expseg ia, idur1, ib [, idur2] [, ic] [...] kres expseg ia, idur1, ib [, idur2] [, ic] [...] Initialization ia -- starting value. Zero is illegal for exponentials. ib, ic, etc. -- value after dur1 seconds, etc. For exponentials, must be non-zero and must agree in sign with ia. idur1 -- duration in seconds of first segment. A zero or negative value will cause all initialization to be skipped. idur2, idur3, etc. -- duration in seconds of subsequent segments. A zero or negative value will terminate the initialization process with the preceding point, permitting the last-defined line or curve to be continued indefinitely in performance. The default is zero. Performance These units generate control or audio signals whose values can pass through 2 or more specified points. The sum of dur values may or may not equal the instrument's performance time: a shorter performance will truncate the specified pattern, while a longer one will cause the last-defined segment to continue on in the same direction. Note that the expseg opcode does not operate correctly at audio rate when segments are shorter than a k-period. Try the expsega opcode instead. Examples Here is an example of the expseg opcode. It uses the file expseg.csd. Example of the expseg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also expon, expsega, expsegr, line, linseg, linsegr transeg Credits Author: &namegabriel; Example written by &namekevin;. New in Csound 3.57 manual_src~dfsg/opcodes/tabrec.xml0000644000000000000000000000510012262561504016357 0ustar rootroot tabrec Table Control:Read/Write Opreations tabrec tabrec Recording of control signals. Description Records control-rate signals on trigger-temporization basis. Syntax tabrec ktrig_start, ktrig_stop, knumtics, kfn, kin1 [,kin2,...,kinN] Performance ktrig_start -- start recording when non-zero. ktrig_stop -- stop recording when knumtics trigger impulses are received by this input argument. knumtics -- stop recording or reset playing pointer to zero when the number of tics defined by this argument is reached. kfn -- table where k-rate signals are recorded. kin1,...,kinN -- input signals to record. The tabrec and tabplay opcodes allow to record/playback control signals on trigger-temporization basis. tabrec opcode records a group of k-rate signals by storing them into kfn table. Each time ktrig_start is triggered, tabrec resets the table pointer to zero and begins to record. Recording phase stops after knumtics trigger impulses have been received by ktrig_stop argument. These opcodes can be used like a sort of ``middle-term'' memory that ``remembers'' generated signals. Such memory can be used to supply generative music with a coherent iterative compositional structure. See Also tabplay Credits Written by &namegabriel;. manual_src~dfsg/opcodes/insremot.xml0000644000000000000000000001150012262561504016760 0ustar rootroot insremot Remote Opcodes insremot insremot An opcode which can be used to implement a remote orchestra. This opcode will send note events from a source machine to one destination. Description With the insremot and insglobal opcodes you are able to perform instruments on remote machines and control them from a master machine. The remote opcodes are implemented using the master/client model. All the machines involved contain the same orchestra but only the master machine contains the information of the score. During the performance the master machine sends the note events to the clients. The insremot opcode will send events from a source machine to one destination if you want to send events to many destinations (broadcast) use the insglobal opcode instead. These two opcodes can be used in combination. Syntax insremot idestination, isource, instrnum [,instrnum...] Initialization idestination -- a string that is the intended client computer (e.g. 192.168.0.100). This is the destination host which receives the events from the given instrument. isource -- a string that is the intended server computer (e.g. 192.168.0.100). This is the source host which generates the events of the given instrument and sends it to the address given by idestination. instrnum -- a list of instrument numbers which will be played on the destination machine Performance If is essential the computers using this opcode have the same byte-order, data size (double or float) and pointer size. One cannot use it with mixed 32 and 64 computers for instance. Internally this opcode makes use of the gethostname and gethostbyname functions to determine the client's and server's IP address for checking which messages are for which machine. If a computer has more than one IP address there is no way to control which IP address is found. The remote operation does not allow the sending of strings at all. Examples Here is an example of the insremot opcode. It uses the files insremot.csd and insremotM.csd. Example of the insremot opcode. The simple example below shows the bilbar example played on a remote machine. The master machine is named "192.168.1.100" and the client "192.168.1.101". Start the client on the machine (it will wait to receive the events from the master machine) and then start the master. Here is the command on linux to start a client (csound -+rtaudio=alsa -odac -dm0 insremot.csd), and the command on the master machine will look like this (csound -+rtaudio=alsa -odac -dm0 insremotM.csd). See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also insglobal, midglobal, midremot, remoteport Credits Author: Simon Schampijer 2006 New in version 5.03 manual_src~dfsg/opcodes/specptrk.xml0000644000000000000000000001716512262561504016770 0ustar rootroot specptrk Spectral Processing:Non-Standard specptrk spectrk Estimates the pitch of the most prominent complex tone in the spectrum. Description Estimate the pitch of the most prominent complex tone in the spectrum. Syntax koct, kamp specptrk wsig, kvar, ilo, ihi, istr, idbthresh, inptls, \ irolloff [, iodd] [, iconfs] [, interp] [, ifprd] [, iwtflg] Initialization ilo, ihi, istr -- pitch range conditioners (low, high, and starting) expressed in decimal octave form. idbthresh -- energy threshold (in decibels) for pitch tracking to occur. Once begun, tracking will be continuous until the energy falls below one half the threshold (6 dB down), whence the koct and kamp outputs will be zero until the full threshold is again surpassed. idbthresh is a guiding value. At initialization it is first converted to the idbout mode of the source spectrum (and the 6 dB down point becomes .5, .25, or 1/root 2 for modes 0, 2 and 3). The values are also further scaled to allow for the weighted partial summation used during correlation.The actual thresholding is done using the internal weighted and summed kamp value that is visible as the second output parameter. inptls, irolloff -- number of harmonic partials used as a matching template in the spectrally-based pitch detection, and an amplitude rolloff for the set expressed as some fraction per octave (linear, so don't roll off to negative). Since the partials and rolloff fraction can affect the pitch following, some experimentation will be useful: try 4 or 5 partials with .6 rolloff as an initial setting; raise to 10 or 12 partials with rolloff .75 for complex timbres like the bassoon (weak fundamental). Computation time is dependent on the number of partials sought. The maximum number is 16. iodd (optional) -- if non-zero, employ only odd partials in the above set (e.g. inptls of 4 would employ partials 1,3,5,7). This improves the tracking of some instruments like the clarinet The default value is 0 (employ all partials). iconfs (optional) -- number of confirmations required for the pitch tracker to jump an octave, pro-rated for fractions of an octave (i.e. the value 12 implies a semitone change needs 1 confirmation (two hits) at the spectrum generating iprd). This parameter limits spurious pitch analyses such as octave errors. A value of 0 means no confirmations required; the default value is 10. interp (optional) -- if non-zero, interpolate each output signal (koct, kamp) between incoming wsig frames. The default value is 0 (repeat the signal values between frames). ifprd (optional) -- if non-zero, display the internally computed spectrum of candidate fundamentals. The default value is 0 (no display). iwtftg (optional) -- wait flag. If non-zero, hold each display until released by the user. The default value is 0 (no wait). Performance At note initialization this unit creates a template of inptls harmonically related partials (odd partials, if iodd non-zero) with amplitude rolloff to the fraction irolloff per octave. At each new frame of wsig, the spectrum is cross-correlated with this template to provide an internal spectrum of candidate fundamentals (optionally displayed). A likely pitch/amp pair (koct, kamp, in decimal octave and summed idbout form) is then estimated. koct varies from the previous koct by no more than plus or minus kvar decimal octave units. It is also guaranteed to lie within the hard limit range ilo -- ihi (decimal octave low and high pitch). kvar can be dynamic, e.g. onset amp dependent. Pitch resolution uses the originating spectrum ifrqs bins/octave, with further parabolic interpolation between adjacent bins. Settings of root magnitude, ifrqs = 24, iq = 15 should capture all the inflections of interest. Between frames, the output is either repeated or interpolated at the k-rate. (See spectrum.) Examples a1, a2 ins ; read a stereo clarinet input krms rms a1, 20 ; find a monaural rms value kvar = 0.6 + krms/8000 ; & use to gate the pitch variance wsig spectrum a1, 0.01, 7, 24, 15, 0, 3 ; get a 7-oct spectrum, 24 bibs/oct specdisp wsig, 0.2 ; display this and now estimate koct, ka spectrk wsig, kvar, 7.0, 10, 9, 20, 4, 0.7, 1, 5, 1, 0.2 ; the pch and amp aosc oscil ka * ka * 10, cpsoct(koct), 2 ; & generate \ new tone with these koct = (koct < 7.0 ? 7.0 : koct) ; replace non pitch with low C display koct - 7.0, 0.25, 20 ; & display the pitch track display ka, 0.25, 20 ; plus the summed root mag outs a1, aosc ; output 1 original and 1 new track manual_src~dfsg/opcodes/randh.xml0000644000000000000000000000776212262561504016233 0ustar rootroot randh Signal Generators:Random (Noise) Generators randh randh Generates random numbers and holds them for a period of time. Description Generates random numbers and holds them for a period of time. Syntax ares randh xamp, xcps [, iseed] [, isize] [, ioffset] kres randh kamp, kcps [, iseed] [, isize] [, ioffset] Initialization iseed (optional, default=0.5) -- seed value for the recursive pseudo-random formula. A value between 0 and +1 will produce an initial output of kamp * iseed. A negative value will cause seed re-initialization to be skipped. A value greater than 1 will seed from system time, this is the best option to generate a different random sequence for each run. isize (optional, default=0) -- if zero, a 16 bit number is generated. If non-zero, a 31-bit random number is generated. Default is 0. ioffset (optional, default=0) -- a base value added to the random result. New in Csound version 4.03. Performance kamp, xamp -- range over which random numbers are distributed. kcps, xcps -- the frequency which new random numbers are generated. The internal pseudo-random formula produces values which are uniformly distributed over the range -kamp to +kamp. rand will thus generate uniform white noise with an R.M.S value of kamp / root 2. The remaining units produce band-limited noise: the kcps and xcps parameters permit the user to specify that new random numbers are to be generated at a rate less than the sampling or control frequencies. randh will hold each new number for the period of the specified cycle. Examples Here is an example of the randh opcode. It uses the file randh.csd. Example of the randh opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example will produce the following output: i1 50.00000 i1 50.00305 i1 97.68677 i1 -44.25354 i1 -61.56006 i1 -75.91248 i1 67.57202 i1 12.83875 i1 5.39551 i1 -95.18738 WARNING: Seeding from current time 684387922 i2 -13.81226 i2 -16.49475 i2 69.51904 i2 35.04944 i2 47.47925 i2 63.25378 i2 -59.61914 i2 50.93079 i2 -6.46362 i2 5.89294 See Also rand, randi manual_src~dfsg/opcodes/STKFlute.xml0000644000000000000000000000716012262561504016570 0ustar rootroot STKFlute Signal Generators:STK Opcodes STKFlute STKFlute STKFlute uses a simple flute physical model. Description STKFlute uses a simple flute physical model. The jet model uses a polynomial, a la Cook. Syntax asignal STKFlute ifrequency, iamplitude, [kjet, kv1[, knoise, kv2[, klfo, kv3[, klfodepth, kv4[, kbreath, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kjet -- controller 2, jet delay. Value range of kv1 is 0-127. knoise -- controller 4, gain of noise. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kbreath -- controller 128, breath pressure. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the Flute class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKFlute opcode. It uses the file STKFlute.csd. Example of the STKFlute opcode. See Also STKClarinet. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/fluidLoad.xml0000644000000000000000000000712312262561504017031 0ustar rootroot fluidLoad Signal Generators:Sample Playback fluidLoad fluidLoad Loads a SoundFont into a fluidEngine, optionally listing SoundFont contents. Syntax isfnum fluidLoad soundfont, ienginenum[, ilistpresets] Description Loads a SoundFont into an instance of a fluidEngine, optionally listing banks and presets for SoundFont. Initialization isfnum -- number assigned to just-loaded soundfont. soundfont -- string specifying a SoundFont filename. Note that any number of SoundFonts may be loaded (obviously, by different invocations of fluidLoad). ienginenum -- engine number assigned from fluidEngine ilistpresets -- optional, if specified, lists all Fluidsynth programs for the just-loaded SoundFont. A Fluidsynth program is a combination of SoundFont ID, bank number, and preset number that is assigned to a MIDI channel. Performance Invoke fluidLoad in the orchestra header, any number of times. The same SoundFont may be invoked to assign programs to MIDI channels any number of times; the SoundFont is only loaded the first time. Examples Here is an example of the fluidLoad opcode. It uses the file fluidLoad.csd and 07AcousticGuitar.sf2. Example of the fluidLoad opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output should include a line like this: SoundFont: 1 Bank: 0 Preset: 0 Seagul Acoustic Git See Also fluidEngine, fluidNote More information on soundfonts in the Floss Manuals: http://en.flossmanuals.net/csound/ch048_d-reading-midi-files Other information on soundfonts on Wikipedia: http://en.wikipedia.org/wiki/Soundfont Credits Michael Gogins (gogins at pipeline dot com), Steven Yi. Thanks to Peter Hanappe for Fluidsynth. New in Csound5.00 manual_src~dfsg/opcodes/log10.xml0000644000000000000000000000527012262561504016051 0ustar rootroot log10 Mathematical Operations:Mathematical Functions log10 log10 Returns a base 10 log. Description Returns the base 10 log of x (x positive only). The argument value is restricted for log, log10, and sqrt. Syntax log10(x) (no rate restriction) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Examples Here is an example of the log10 opcode. It uses the file log10.csd. Example of the log10 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: i1 = 0.903 See Also abs, exp, frac, int, log, log, i, sqrt Credits Written by &namejohn;. New in version 3.47 Example written by &namekevin;. manual_src~dfsg/opcodes/locsend.xml0000644000000000000000000001410412262561504016552 0ustar rootroot locsend Signal Modifiers:Panning and Spatialization locsend locsend Distributes the audio signals of a previous locsig opcode. Description locsend depends upon the existence of a previously defined locsig. The number of output signals must match the number in the previous locsig. The output signals from locsend are derived from the values given for distance and reverb in the locsig and are ready to be sent to local or global reverb units (see example below). The reverb amount and the balance between the 2 or 4 channels are calculated in the same way as described in the Dodge book (an essential text!). Syntax a1, a2 locsend a1, a2, a3, a4 locsend Examples asig some audio signal kdegree line 0, p3, 360 kdistance line 1, p3, 10 a1, a2, a3, a4 locsig asig, kdegree, kdistance, .1 ar1, ar2, ar3, ar4 locsend ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 reverb2 ga1, 2.5, .5 a2 reverb2 ga2, 2.5, .5 a3 reverb2 ga3, 2.5, .5 a4 reverb2 ga4, 2.5, .5 outq a1, a2, a3, a4 ga1=0 ga2=0 ga3=0 ga4=0 In the above example, the signal, asig, is sent around a complete circle once during the duration of a note while at the same time it becomes more and more distant from the listeners' location. locsig sends the appropriate amount of the signal internally to locsend. The outputs of the locsend are added to global accumulators in a common Csound style and the global signals are used as inputs to the reverb units in a separate instrument. For an example, see locsig. locsig is useful for quad and stereo panning as well as fixed placed of sounds anywhere between two loudspeakers. Below is an example of the fixed placement of sounds in a stereo field. It uses the file locsend_stereo.csd. Example of the locsend opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. A few notes: ;place the sound in the left speaker and near: i1 0 1 0 1 ;place the sound in the right speaker and far: i1 1 1 90 25 ;place the sound equally between left and right and in the middle ground distance: i1 2 1 45 12 e The next example shows a simple intuitive use of the distance value to simulate Doppler shift. The same value is used to scale the frequency as is used as the distance input to locsig. kdistance line 1, p3, 10 kfreq = (ifreq * 340) / (340 + kdistance) asig oscili iamp, kfreq, 1 kdegree line 0, p3, 360 a1, a2, a3, a4 locsig asig, kdegree, kdistance, .1 ar1, ar2, ar3, ar4 locsend See Also locsig Credits Author: Richard Karpen Seattle, WA USA 1998 New in Csound version 3.48 manual_src~dfsg/opcodes/min.xml0000644000000000000000000000560212262561504015711 0ustar rootroot min Signal Modifiers:Comparators and Accumulators min min Produces a signal that is the minimum of any number of input signals. Description The min opcode takes any number of a-rate or k-rate signals as input (all of the same rate), and outputs a signal at the same rate that is the minimum of all of the inputs. For a-rate signals, the inputs are compared one sample at a time (i.e. min does not scan an entire ksmps period of a signal for its local minimum as the max_k opcode does). Syntax amin min ain1, ain2 [, ain3] [, ain4] [...] kmin min kin1, kin2 [, kin3] [, kin4] [...] Performance ain1, ain2, ... -- a-rate signals to be compared. kin1, kin2, ... -- k-rate signals to be compared. Examples Here is an example of the min opcode. It uses the file min.csd. Example of the min opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also max, maxabs, minabs, maxaccum, minaccum, maxabsaccum, minabsaccum, max_k Credits Author: &nameanthony; March 2006 New in Csound version 5.01 manual_src~dfsg/opcodes/valpass.xml0000644000000000000000000000732212262561504016600 0ustar rootroot valpass Signal Modifiers:Reverberation valpass valpass Variably reverberates an input signal with a flat frequency response. Description Variably reverberates an input signal with a flat frequency response. Syntax ares valpass asig, krvt, xlpt, imaxlpt [, iskip] [, insmps] Initialization imaxlpt -- maximum loop time for klpt iskip (optional, default=0) -- initial disposition of delay-loop data space (cf. reson). The default value is 0. insmps (optional, default=0) -- delay amount, as a number of samples. Performance krvt -- the reverberation time (defined as the time in seconds for a signal to decay to 1/1000, or 60dB down from its original amplitude). xlpt -- variable loop time in seconds, same as ilpt in comb. Loop time can be as large as imaxlpt. This filter reiterates input with an echo density determined by loop time xlpt. The attenuation rate is independent and is determined by krvt, the reverberation time (defined as the time in seconds for a signal to decay to 1/1000, or 60dB down from its original amplitude). Its output will begin to appear immediately. Examples Here is an example of the valpass opcode. It uses the file valpass.csd. Example of the valpass opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the valpass opcode. It uses the file valpass-2.csd. Second example of the valpass opcode. See Also alpass, comb, reverb, vcomb Credits Author: William Pete Moss University of Texas at Austin Austin, Texas USA January 2002 manual_src~dfsg/opcodes/sndload.xml0000644000000000000000000001341312262561504016551 0ustar rootroot sndload Table Control sndload sndload Loads a sound file into memory for use by loscilx Description sndload loads a sound file into memory for use by loscilx. Syntax sndload Sfname[, ifmt[, ichns[, isr[, ibas[, iamp[, istrt \ [, ilpmod[, ilps[, ilpe]]]]]]]]] Initialization Sfname - file name as a string constant or variable, string p-field, or a number that is used either as an index to strings set with strset, or, if that is not available, a fine name in the format soundin.n is used. If the file name does not include a full path, the file is searched in the current directory first, then those specified by SSDIR (if defined), and finally SFDIR. If the same file was already loaded previously, it will not be read again, but the parameters ibas, iamp, istrt, ilpmod, ilps, and ilpe are still updated. ifmt (optional, defaults to zero) - default sample format for raw (headerless) sound files; if the file has a header, this is ignored. Can be one of the following: -1: do not allow headerless files (fail with an init error) 0: use the same format as the one specified on the command line 1: 8 bit signed integers 2: a-law 3: u-law 4: 16 bit signed integers 5: 32 bit signed integers 6: 32 bit floats 7: 8 bit unsigned integers 8: 24 bit signed integers 9: 64 bit floats ichns (optional, defaults to zero) - default number of channels for raw (headerless) sound files; if the file has a header, this is ignored. Zero or negative values are interpreted as 1 channel. isr (optional, defaults to zero) - default sample rate for raw (headerless) sound files; if the file has a header, this is ignored. Zero or negative values are interpreted as the orchestra sample rate (sr). ibas (optional, defaults to zero) - base frequency in Hz. If positive, overrides the value specified in the sound file header; otherwise, the value from the header is used if present, and 1.0 if the file does not include such information. iamp (optional, defaults to zero) - amplitude scale. If non-zero, overrides the value specified in the sound file header (note: negative values are allowed, and will invert the sound output); otherwise, the value from the header is used if present, and 1.0 if the file does not include such information. istrt (optional, defaults to -1) - starting position in sample frames, can be fractional. If non-negative, overrides the value specified in the sound file header; otherwise, the value from the header is used if present, and 0 if the file does not include such information. Note: even if this parameter is specified, the whole file is still read into memory. ilpmod (optional, defaults to -1) - loop mode, can be one of the following: any negative value: use the loop information specified in the sound file header, ignoring ilps and ilpe 0: no looping (ilps and ilpe are ignored) 1: forward looping (wrap around loop end if it is crossed in forward direction, and wrap around loop start if it is crossed in backward direction) 2: backward looping (change direction at loop end if it is crossed in forward direction, and wrap around loop start if it is crossed in backward direction) 3: forward-backward looping (change direction at both loop points if they are crossed as described above) ilps (optional, defaults to 0) - loop start in sample frames (fractional values are allowed), or loop end if ilps is greater than ilpe. Ignored unless ilpmod is set to 1, 2, or 3. If the loop points are equal, the whole sample is looped. ilpe (optional, defaults to 0) - loop end in sample frames (fractional values are allowed), or loop start if ilps is greater than ilpe. Ignored unless ilpmod is set to 1, 2, or 3. If the loop points are equal, the whole sample is looped. Credits Written by &nameistvan;. 2006 New in Csound 5.03 manual_src~dfsg/opcodes/grain2.xml0000644000000000000000000001322612262561504016311 0ustar rootroot grain2 Signal Generators:Granular Synthesis grain2 grain2 Easy-to-use granular synthesis texture generator. Description Generate granular synthesis textures. grain2 is simpler to use, but grain3 offers more control. Syntax ares grain2 kcps, kfmd, kgdur, iovrlp, kfn, iwfn [, irpow] \ [, iseed] [, imode] Initialization iovrlp -- (fixed) number of overlapping grains. iwfn -- function table containing window waveform (Use GEN20 to calculate iwfn). irpow (optional, default=0) -- this value controls the distribution of grain frequency variation. If irpow is positive, the random distribution (x is in the range -1 to 1) is abs(x) ˆ ((1 / irpow) - 1); for negative irpow values, it is (1 - abs(x)) ˆ ((-1 / irpow) - 1)Setting irpow to -1, 0, or 1 will result in uniform distribution (this is also faster to calculate). The image below shows some examples for irpow. The default value of irpow is 0. [A graph of distributions for different values of irpow.] A graph of distributions for different values of irpow. iseed (optional, default=0) -- seed value for random number generator (positive integer in the range 1 to 2147483646 (231 - 2)). Zero or negative value seeds from current time (this is also the default). imode (optional default=0) -- sum of the following values: 8: interpolate window waveform (slower). 4: do not interpolate grain waveform (fast, but lower quality). 2: grain frequency is continuously modified by kcps and kfmd (by default, each grain keeps the frequency it was launched with). This may be slower at high control rates. 1: skip initialization. Performance ares -- output signal. kcps -- grain frequency in Hz. kfmd -- random variation (bipolar) in grain frequency in Hz. kgdur -- grain duration in seconds. kgdur also controls the duration of already active grains (actually the speed at which the window function is read). This behavior does not depend on the imode flags. kfn -- function table containing grain waveform. Table number can be changed at k-rate (this is useful to select from a set of band-limited tables generated by GEN30, to avoid aliasing). Note grain2 internally uses the same random number generator as rnd31. So reading its documentation is also recommended. Examples Here is an example of the grain2 opcode. It uses the file grain2.csd. Example of the grain2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also grain3 Credits Author: &nameistvan; New in version 4.15 Updated April 2002 by &nameistvan; manual_src~dfsg/opcodes/imageload.xml0000644000000000000000000000451212262561504017047 0ustar rootroot imageload Image Processing Opcodes imageload imageload Load an image. Description Load an image and return a reference to it. Individual pixel values can then be accessed with imagegetpixel. The image processing opcodes can only load png images Syntax iimagenum imageload filename Initialization iimagenum -- number assigned to the loaded image. filename -- The filename of the image to load (should be a valid PNG image file). Examples Here is an example of the imageload opcode. It uses the file imageopcodes.csd. Example of the imageload opcode. See Also imagecreate, imagesize, imagesave, imagegetpixel, imagesetpixel, imagefree Credits Author: Cesare Marilungo New in version 5.08 manual_src~dfsg/opcodes/zfilter2.xml0000644000000000000000000001073012262561504016665 0ustar rootroot zfilter2 Signal Modifiers:Specialized Filters zfilter2 zfilter2 Performs filtering using a transposed form-II digital filter lattice with radial pole-shearing and angular pole-warping. Description General purpose custom filter with time-varying pole control. The filter coefficients implement the following difference equation: (1)*y(n) = b0*x[n] + b1*x[n-1] +...+ bM*x[n-M] - a1*y[n-1] -...- aN*y[n-N] the system function for which is represented by: B(Z) b0 + b1*Z-1 + ... + bM*Z-M H(Z) = ---- = -------------------------- A(Z) 1 + a1*Z-1 + ... + aN*Z-N Syntax ares zfilter2 asig, kdamp, kfreq, iM, iN, ib0, ib1, ..., ibM, \ ia1,ia2, ..., iaN Initialization At initialization the number of zeros and poles of the filter are specified along with the corresponding zero and pole coefficients. The coefficients must be obtained by an external filter-design application such as Matlab and specified directly or loaded into a table via GEN01. With zfilter2, the roots of the characteristic polynomials are solved at initialization so that the pole-control operations can be implemented efficiently. Performance The filter2 opcodes perform filtering using a transposed form-II digital filter lattice with no time-varying control. zfilter2 uses the additional operations of radial pole-shearing and angular pole-warping in the Z plane. Pole shearing increases the magnitude of poles along radial lines in the Z-plane. This has the affect of altering filter ring times. The k-rate variable kdamp is the damping parameter. Positive values (0.01 to 0.99) increase the ring-time of the filter (hi-Q), negative values (-0.01 to -0.99) decrease the ring-time of the filter, (lo-Q). Pole warping changes the frequency of poles by moving them along angular paths in the Z plane. This operation leaves the shape of the magnitude response unchanged but alters the frequencies by a constant factor (preserving 0 and p). The k-rate variable kfreq determines the frequency warp factor. Positive values (0.01 to 0.99) increase frequencies toward p and negative values (-0.01 to -0.99) decrease frequencies toward 0. Since filter2 implements generalized recursive filters, it can be used to specify a large range of general DSP algorithms. For example, a digital waveguide can be implemented for musical instrument modeling using a pair of delayr and delayw opcodes in conjunction with the filter2 opcode. Examples A controllable second-order IIR filter operating on an a-rate signal: a1 zfilter2 asig, kdamp, kfreq, 1, 2, 1, ia1, ia2 ;; controllable a-rate IIR filter See Also filter2 Credits Author: Michael A. Casey M.I.T. Cambridge, Mass. 1997 New in Version 3.47 manual_src~dfsg/opcodes/syncgrain.xml0000644000000000000000000001302012262561504017114 0ustar rootroot syncgrain Signal Generators:Granular Synthesis syncgrain syncgrain Synchronous granular synthesis. Description syncgrain implements synchronous granular synthesis. The source sound for the grains is obtained by reading a function table containing the samples of the source waveform. For sampled-sound sources, GEN01 is used. syncgrain will accept deferred allocation tables. The grain generator has full control of frequency (grains/sec), overall amplitude, grain pitch (a sampling increment) and grain size (in secs), both as fixed or time-varying (signal) parameters. An extra parameter is the grain pointer speed (or rate), which controls which position the generator will start reading samples in the table for each successive grain. It is measured in fractions of grain size, so a value of 1 (the default) will make each successive grain read from where the previous grain should finish. A value of 0.5 will make the next grain start at the midway position from the previous grain start and finish, etc.. A value of 0 will make the generator read always from a fixed position of the table (wherever the pointer was last at). A negative value will decrement pointer positions. This control gives extra flexibility for creating timescale modifications in the resynthesis. syncgrain will generate any number of parallel grain streams (which will depend on grain density/frequency), up to the iolaps value (default 100). The number of streams (overlapped grains) is determined by grainsize*grain_freq. More grain overlaps will demand more calculations and the synthesis might not run in realtime (depending on processor power). syncgrain can simulate FOF-like formant synthesis, provided that a suitable shape is used as grain envelope and a sinewave as the grain wave. For this use, grain sizes of around 0.04 secs can be used. The formant centre frequency is determined by the grain pitch. Since this is a sampling increment, in order to use a frequency in Hz, that value has to be scaled by tablesize/sr. Grain frequency will determine the fundamental. syncgrain uses floating-point indexing, so its precision is not affected by large-size tables. This opcode is based on the SndObj library SyncGrain class. Syntax asig syncgrain kamp, kfreq, kpitch, kgrsize, kprate, ifun1, \ ifun2, iolaps Initialization ifun1 -- source signal function table. Deferred-allocation tables (see GEN01) are accepted, but the opcode expects a mono source. ifun2 -- grain envelope function table. iolaps -- maximum number of overlaps, max(kfreq)*max(kgrsize). Estimating a large value should not affect performance, but exceeding this value will probably have disastrous consequences. Performance kamp -- amplitude scaling kfreq -- frequency of grain generation, or density, in grains/sec. kpitch -- grain pitch scaling (1=normal pitch, < 1 lower, > 1 higher; negative, backwards) kgrsize -- grain size in secs. kprate -- readout pointer rate, in grains. The value of 1 will advance the reading pointer 1 grain ahead in the source table. Larger values will time-compress and smaller values will time-expand the source signal. Negative values will cause the pointer to run backwards and zero will freeze it. Examples Here is an example of the syncgrain opcode. It uses the file syncgrain.csd. Example of the syncgrain opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; January 2005 New plugin in version 5 January 2005. manual_src~dfsg/opcodes/timeinstk.xml0000644000000000000000000000605412262561504017137 0ustar rootroot timeinstk Instrument Control:Time Reading timeinstk timeinstk Read absolute time in k-rate cycles. Description Read absolute time, in k-rate cycles, since the start of an instance of an instrument. Called at both i-time as well as k-time. Syntax kres timeinstk Performance timeinstk is for time in k-rate cycles. So with: sr = 44100 kr = 6300 ksmps = 7 then after half a second, the timeinstk opcode would report 3150. It will always report an integer. timeinstk produces a k-rate variable for output. There are no input parameters. timeinstk is similar to timek except it returns the time since the start of this instance of the instrument. Examples Here is an example of the timeinstk opcode. It uses the file timeinstk.csd. Example of the timeinstk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: k1 = 1.000000 samples k1 = 2205.000000 samples k1 = 4410.000000 samples k1 = 6615.000000 samples k1 = 8820.000000 samples See Also timeinsts, timek, times Credits Author: Robin Whittle Australia May 1997 Example written by &namekevin;. manual_src~dfsg/opcodes/kcauchy.xml0000644000000000000000000000076712262561504016564 0ustar rootroot kcauchy kcauchy kcauchy Deprecated. Description Deprecated as of version 3.49. Use the cauchy opcode instead. manual_src~dfsg/opcodes/pvsftw.xml0000644000000000000000000001013712262561504016456 0ustar rootroot pvsftw Spectral Processing:Streaming pvsftw pvsftw Writes amplitude and/or frequency data to function tables. Description Writes amplitude and/or frequency data to function tables. Syntax kflag pvsftw fsrc, ifna [, ifnf] Initialization ifna -- A table, at least inbins in size, that stores amplitude data. Ignored if ifna = 0 ifnf -- A table, at least inbins in size, that stores frequency data. Ignored if ifnf = 0 Performance kflag -- A flag that has the value of 1 when new data is available, 0 otherwise. fsrc -- a PVOC-EX formatted source. Enables the contents of fsrc to be exchanged with function tables, for custom processing. Except when the frame overlap equals ksmps (which will generally not be the case), the frame data is not updated each control period. The data in ifna, ifnf should only be processed when kflag is set to 1. To process only frequency data, set ifna to zero. As the functions tables are required only to store data from fsrc, there is no advantage in defining then in the score. They should generally be created in the instrument using ftgen. By exporting amplitude data, say, from one fsig and importing it into another, basic cross-synthesis (as in pvscross) can be performed, with the option to modify the data beforehand using the table manipulation opodes. Note that the format data in the source fsig is not written to the tables. This therefore offers a means of transferring amplitude and frequency data between non-identical fsigs. Used this way, these opcodes become potentially pathological, and can be relied upon to produce unexpected results. In such cases, resynthesis using pvsadsyn would almost certainly be required. To perform a straight copy from one fsig to another one of identical format, the conventional assignment syntax can be used: fsig1 = fsig2 It is not necessary to use function tables in this case. Examples Here is an example of the pvsftw opcode. It uses the file pvsftw.csd. Example of the pvsftw opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsftr Credits Author: &namerichard; August 2001 New in version 4.13 manual_src~dfsg/opcodes/ftsave.xml0000644000000000000000000000522312262561504016415 0ustar rootroot ftsave Table Control:Read/Write Opreations ftsave ftsave Save a set of previously-allocated tables to a file. Description Save a set of previously-allocated tables to a file. Syntax ftsave "filename", iflag, ifn1 [, ifn2] [...] Initialization "filename" -- A quoted string containing the name of the file to save. iflag -- Type of the file to save. (0 = binary file, Non-zero = text file) ifn1, ifn2, ... -- Numbers of tables to save. Performance ftsave saves a list of tables to a file. The file's format can be binary or text. Warning The file's format is not compatible with a WAV-file and is not endian-safe. Examples Here is an example of the ftsave opcode. It uses the file ftsave.csd. Example of the ftsave opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also ftloadk, ftload, ftsavek Credits Author: &namegabriel; Example written by &namekevin;. New in version 4.21 manual_src~dfsg/opcodes/ibexprnd.xml0000644000000000000000000000077612262561504016750 0ustar rootroot ibexprnd ibexprnd ibexprnd Deprecated. Description Deprecated as of version 3.49. Use the bexprnd opcode instead. manual_src~dfsg/opcodes/hilbert.xml0000644000000000000000000001655612262561504016571 0ustar rootroot hilbert Signal Modifiers:Specialized Filters hilbert hilbert A Hilbert transformer. Description An IIR implementation of a Hilbert transformer. Syntax ar1, ar2 hilbert asig Performance asig -- input signal ar1 -- sine output of asig ar2 -- cosine output of asig hilbert is an IIR filter based implementation of a broad-band 90 degree phase difference network. The input to hilbert is an audio signal, with a frequency range from 15 Hz to 15 kHz. The outputs of hilbert have an identical frequency response to the input (i.e. they sound the same), but the two outputs have a constant phase difference of 90 degrees, plus or minus some small amount of error, throughout the entire frequency range. The outputs are in quadrature. hilbert is useful in the implementation of many digital signal processing techniques that require a signal in phase quadrature. ar1 corresponds to the cosine output of hilbert, while ar2 corresponds to the sine output. The two outputs have a constant phase difference throughout the audio range that corresponds to the phase relationship between cosine and sine waves. Internally, hilbert is based on two parallel 6th-order allpass filters. Each allpass filter implements a phase lag that increases with frequency; the difference between the phase lags of the parallel allpass filters at any given point is approximately 90 degrees. Unlike an FIR-based Hilbert transformer, the output of hilbert does not have a linear phase response. However, the IIR structure used in hilbert is far more efficient to compute, and the nonlinear phase response can be used in the creation of interesting audio effects, as in the second example below. Examples The first example implements frequency shifting, or single sideband amplitude modulation. Frequency shifting is similar to ring modulation, except the upper and lower sidebands are separated into individual outputs. By using only one of the outputs, the input signal can be "detuned," where the harmonic components of the signal are shifted out of harmonic alignment with each other, e.g. a signal with harmonics at 100, 200, 300, 400 and 500 Hz, shifted up by 50 Hz, will have harmonics at 150, 250, 350, 450, and 550 Hz. Here is the first example of the hilbert opcode. It uses the file hilbert.csd, and beats.wav. Example of the hilbert opcode implementing frequency shifting. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The second example is a variation of the first, but with the output being fed back into the input. With very small shift amounts (i.e. between 0 and +-6 Hz), the result is a sound that has been described as a barberpole phaser or Shepard tone phase shifter. Several notches appear in the spectrum, and are constantly swept in the direction opposite that of the shift, producing a filtering effect that is reminiscent of Risset's endless glissando. Here is the second example of the hilbert opcode. It uses the file hilbert_barberpole.csd. Example of the hilbert opcode sounding like a <quote>barberpole phaser</quote>. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Technical History The use of phase-difference networks in frequency shifters was pioneered by Harald Bode1. Bode and Bob Moog provide an excellent description of the implementation and use of a frequency shifter in the analog realm in 2; this would be an excellent first source for those that wish to explore the possibilities of single sideband modulation. Bernie Hutchins provides more applications of the frequency shifter, as well as a detailed technical analysis3. A recent paper by Scott Wardle4 describes a digital implementation of a frequency shifter, as well as some unique applications. References H. Bode, "Solid State Audio Frequency Spectrum Shifter." AES Preprint No. 395 (1965). H. Bode and R.A. Moog, "A High-Accuracy Frequency Shfiter for Professional Audio Applications." Journal of the Audio Engineering Society, July/August 1972, vol. 20, no. 6, p. 453. B. Hutchins. Musical Engineer's Handbook (Ithaca, NY: Electronotes, 1975), ch. 6a. S. Wardle, "A Hilbert-Transformer Frequency Shifter for Audio." Available online at http://www.iua.upf.es/dafx98/papers/. Credits Author: &namesean; Seattle, Washington 1999 New in Csound version 3.55 The examples were updated April 2002. Thanks go to &namesean; for fixing the barberpole example. manual_src~dfsg/opcodes/shaker.xml0000644000000000000000000000571212262561504016405 0ustar rootroot shaker Signal Generators:Models and Emulations shaker shaker Sounds like the shaking of a maraca or similar gourd instrument. Description Audio output is a tone related to the shaking of a maraca or similar gourd instrument. The method is a physically inspired model developed from &nameperry;, but re-coded for Csound. Syntax ares shaker kamp, kfreq, kbeans, kdamp, ktimes [, idecay] Initialization idecay -- If present indicates for how long at the end of the note the shaker is to be damped. The default value is zero. Performance A note is played on a maraca-like instrument, with the arguments as below. kamp -- Amplitude of note. kfreq -- Frequency of note played. kbeans -- The number of beans in the gourd. A value of 8 seems suitable. kdamp -- The damping value of the shaker. Values of 0.98 to 1 seems suitable, with 0.99 a reasonable default. ktimes -- Number of times shaken. Note The argument knum was redundant, so it was removed in version 3.49. Examples Here is an example of the shaker opcode. It uses the file shaker.csd. Example of the shaker opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 Fixed the example thanks to a message from &nameistvan;. manual_src~dfsg/opcodes/pvadd.xml0000644000000000000000000002563012262561504016227 0ustar rootroot pvadd Spectral Processing:STFT pvadd pvadd Reads from a pvoc file and uses the data to perform additive synthesis. Description pvadd reads from a pvoc file and uses the data to perform additive synthesis using an internal array of interpolating oscillators. The user supplies the wave table (usually one period of a sine wave), and can choose which analysis bins will be used in the re-synthesis. Syntax ares pvadd ktimpnt, kfmod, ifilcod, ifn, ibins [, ibinoffset] \ [, ibinincr] [, iextractmode] [, ifreqlim] [, igatefn] Initialization ifilcod -- integer or character-string denoting a control-file derived from pvanal analysis of an audio signal. An integer denotes the suffix of a file pvoc.m; a character-string (in double quotes) gives a filename, optionally a full pathname. If not fullpath, the file is sought first in the current directory, then in the one given by the environment variable SADIR (if defined). pvoc control files contain data organized for fft resynthesis. Memory usage depends on the size of the files involved, which are read and held entirely in memory during computation but are shared by multiple calls (see also lpread). ifn -- table number of a stored function containing a sine wave. ibins -- number of bins that will be used in the resynthesis (each bin counts as one oscillator in the re-synthesis) ibinoffset (optional) -- is the first bin used (it is optional and defaults to 0). ibinincr (optional) -- sets an increment by which pvadd counts up from ibinoffset for ibins components in the re-synthesis (see below for a further explanation). iextractmode (optional) -- determines if spectral extraction will be carried out and if so whether components that have changes in frequency below ifreqlim or above ifreqlim will be discarded. A value for iextractmode of 1 will cause pvadd to synthesize only those components where the frequency difference between analysis frames is greater than ifreqlim. A value of 2 for iextractmode will cause pvadd to synthesize only those components where the frequency difference between frames is less than ifreqlim. The default values for iextractmode and ifreqlim are 0, in which case a simple resynthesis will be done. See examples below. igatefn (optional) -- is the number of a stored function which will be applied to the amplitudes of the analysis bins before resynthesis takes place. If igatefn is greater than 0 the amplitudes of each bin will be scaled by igatefn through a simple mapping process. First, the amplitudes of all of the bins in all of the frames in the entire analysis file are compared to determine the maximum amplitude value. This value is then used create normalized amplitudes as indeces into the stored function igatefn. The maximum amplitude will map to the last point in the function. An amplitude of 0 will map to the first point in the function. Values between 0 and 1 will map accordingly to points along the function table.This will be made clearer in the examples below. Performance ktimpnt and kfmod are used in the same way as in pvoc. Examples ktime line 0, p3, p3 asig pvadd ktime, 1, oboe.pvoc, 1, 100, 2 In the above, ibins is 100 and ibinoffset is 2. Using these settings the resynthesis will contain 100 components beginning with bin #2 (bins are counted starting with 0). That is, resynthesis will be done using bins 2-101 inclusive. It is usually a good idea to begin with bin 1 or 2 since the 0th and often 1st bin have data that is neither necessary nor even helpful for creating good clean resynthesis. ktime line 0, p3, p3 asig pvadd ktime, 1, oboe.pvoc, 1, 100, 2, 2 The above is the same as the previous example with the addition of the value 2 used for the optional ibinincr argument. This result will still result in 100 components in the resynthesis, but pvadd will count through the bins by 2 instead of by 1. It will use bins 2, 4, 6, 8, 10, and so on. For ibins=10, ibinoffset=10, and ibinincr=10, pvadd would use bins 10, 20, 30, 40, up to and including 100. Below is an example using spectral extraction. In this example iextractmode is 1 and ifreqlim is 9. This will cause pvadd to synthesize only those bins where the frequency deviation, averaged over 6 frames, is greater than 9. ktime line 0, p3, p3 asig pvadd ktime, 1, oboe.pvoc, 1, 100, 2, 2, 1, 9 If iextractmode were 2 in the above, then only those bins with an average frequency deviation of less than 9 would be synthesized. If tuned correctly, this technique can be used to separate the pitched parts of the spectrum from the noisy parts. In practice this depends greatly on the type of sound, the quality of the recording and digitization, and also on the analysis window size and frame increment. Next is an example using amplitude gating. The last 2 in the argument list stands for f2 in the score. asig pvadd ktime, 1, oboe.pvoc, 1, 100, 2, 2, 0, 0, 2 Suppose the score for the above were to contain: f2 0 512 7 0 256 1 256 1 Then those bins with amplitudes of 50% of the maximum or greater would be left unchanged, while those with amplitudes less than 50% of the maximum would be scaled down. In this case the lower the amplitude the more severe the scaling down would be. But suppose the score contains: f2 0 512 5 1 512 .001 In this case lower amplitudes will be left unchanged and greater ones will be scaled down, turning the sound upside-down in terms of the amplitude spectrum! Functions can be arbitrarily complex. Just remember that the normalized amplitude values of the analysis are themselves the indeces into the function. Finally, both spectral extraction and amplitude gating can be used together. The example below will synthesize only those components that with a frequency deviation of less than 5Hz per frame and it will scale the amplitudes according to f2. asig pvadd ktime, 1, oboe.pvoc, 1, 100, 1, 1, 2, 5, 2 USEFUL HINTS By using several pvadd units together, one can gradually fade in different parts of the resynthesis, creating various filtering effects. The author uses pvadd to synthesis one bin at a time to have control over each separate component of the re-synthesis. If any combination of ibins, ibinoffset, and ibinincr, creates a situation where pvadd is asked to used a bin number greater than the number of bins in the analysis, it will just use all of the available bins, and give no complaint. So to use every bin just make ibins a big number (ie. 2000). Expect to have to scale up the amplitudes by factors of 10-100, by the way. Here is a complete example of the pvadd opcode. It uses the file pvadd.csd Example of the pvadd opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Richard Karpen Seattle, WA USA 1998 New in Csound version 3.48, additional arguments version 3.56 manual_src~dfsg/opcodes/midiin.xml0000644000000000000000000000644612262561504016406 0ustar rootroot midiin Real-time MIDI:Generic I/O midiin midiin Returns a generic MIDI message received by the MIDI IN port. Description Returns a generic MIDI message received by the MIDI IN port Syntax kstatus, kchan, kdata1, kdata2 midiin Performance kstatus -- the type of MIDI message. Can be: 128 (note off) 144 (note on) 160 (polyphonic aftertouch) 176 (control change) 192 (program change) 208 (channel aftertouch) 224 (pitch bend 0 if no MIDI message are pending in the MIDI IN buffer kchan -- MIDI channel (1-16) kdata1, kdata2 -- message-dependent data values midiin has no input arguments, because it reads at the MIDI in port implicitly. It works at k-rate. Normally (i.e., when no messages are pending) kstatus is zero, only when MIDI data are present in the MIDI IN buffer, is kstatus set to the type of the relevant messages. Be careful when using midiin in low numbered instruments, since a MIDI note will launch additional instances of the instrument, resulting in duplicate events and weird behaviour. Use massign to direct MIDI note on messages to a different instrument or to disable triggering of instruments from MIDI. Examples Here is an example of the midiin opcode. It uses the file midiin.csd. Example of the midiin opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; Italy 1998 New in Csound version 3.492 manual_src~dfsg/opcodes/tempest.xml0000644000000000000000000001210012262561504016576 0ustar rootroot tempest Instrument Control:Sensing and Control tempest tempest Estimate the tempo of beat patterns in a control signal. Description Estimate the tempo of beat patterns in a control signal. Syntax ktemp tempest kin, iprd, imindur, imemdur, ihp, ithresh, ihtim, ixfdbak, \ istartempo, ifn [, idisprd] [, itweek] Initialization iprd -- period between analyses (in seconds). Typically about .02 seconds. imindur -- minimum duration (in seconds) to serve as a unit of tempo. Typically about .2 seconds. imemdur -- duration (in seconds) of the kin short-term memory buffer which will be scanned for periodic patterns. Typically about 3 seconds. ihp -- half-power point (in Hz) of a low-pass filter used to smooth input kin prior to other processing. This will tend to suppress activity that moves much faster. Typically 2 Hz. ithresh -- loudness threshold by which the low-passed kin is center-clipped before being placed in the short-term buffer as tempo-relevant data. Typically at the noise floor of the incoming data. ihtim -- half-time (in seconds) of an internal forward-masking filter that masks new kin data in the presence of recent, louder data. Typically about .005 seconds. ixfdbak -- proportion of this unit's anticipated value to be mixed with the incoming kin prior to all processing. Typically about .3. istartempo -- initial tempo (in beats per minute). Typically 60. ifn -- table number of a stored function (drawn left-to-right) by which the short-term memory data is attenuated over time. idisprd (optional) -- if non-zero, display the short-term past and future buffers every idisprd seconds (normally a multiple of iprd). The default value is 0 (no display). itweek (optional) -- fine-tune adjust this unit so that it is stable when analyzing events controlled by its own output. The default value is 1 (no change). Performance tempest examines kin for amplitude periodicity, and estimates a current tempo. The input is first low-pass filtered, then center-clipped, and the residue placed in a short-term memory buffer (attenuated over time) where it is analyzed for periodicity using a form of autocorrelation. The period, expressed as a tempo in beats per minute, is output as ktemp. The period is also used internally to make predictions about future amplitude patterns, and these are placed in a buffer adjacent to that of the input. The two adjacent buffers can be periodically displayed, and the predicted values optionally mixed with the incoming signal to simulate expectation. This unit is useful for sensing the metric implications of any k-signal (e.g.- the RMS of an audio signal, or the second derivative of a conducting gesture), before sending to a tempo statement. Examples Here is an example of the tempest opcode. It uses the file tempest.csd, and beats.wav. Example of the tempest opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The tempo of the audio file beats.wav is 120 beats per minute. In this examples, tempest will print out its best guess as the audio file plays. Its output should include lines like this: . i1 118.24654 . i1 121.72949 manual_src~dfsg/opcodes/lpshold.xml0000644000000000000000000000633012262561504016572 0ustar rootroot lpshold Signal Generators:Linear and Exponential Generators lpshold lpshold Generate control signal consisting of held segments. Description Generate control signal consisting of held segments delimited by two or more specified points. The entire envelope is looped at kfreq rate. Each parameter can be varied at k-rate. Syntax ksig lpshold kfreq, ktrig, iphase, ktime0, kvalue0 [, kvalue1] [, ktime1] [, kvalue2] [, ktime2] [...] Performance ksig -- Output signal kfreq -- Repeat rate in Hz or fraction of Hz ktrig -- If non-zero, retriggers the envelope from start (see trigger opcode), before the envelope cycle is completed. iphase -- A vaue between 0 and 1 to say where to start the loop. Zero, the commonest value, indicates the beginning. kvalue0...kvalueN -- Values of points ktime0...ktimeN -- Times between points; expressed in fractions of a cycle (see below). The final time designates a ramp between the final value and the first value. lpshold is similar to loopseg, but can generate only horizontal segments, i.e. holds values for each time interval placed between ktimeN and ktimeN+1. It can be useful, among other things, for melodic control, like old analog sequencers. Examples Here is an example of the lpshold opcode. It uses the file lpshold.csd. Example of the lpshold opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also loopseg Credits Author: &namegabriel; New in Version 4.13 manual_src~dfsg/opcodes/vexpv_i.xml0000644000000000000000000001026612262561504016610 0ustar rootroot vexpv_i Vectorial:Vectorial operations vexpv_i vexpv_i Performs exponential operations between two vectorial control signals at init time. Description Performs exponential operations between two vectorial control signals at init time. Syntax vexpv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Initialization ifn1 - number of the table hosting the first vector to be processed ifn2 - number of the table hosting the second vector to be processed ielements - number of elements of the two vectors idstoffset - index offset for the destination (ifn1) table (Default=0) isrcoffset - index offset for the source (ifn2) table (Default=0) Performance vexpv_i elevates each element of ifn2 to the corresponding element of ifn1. Each vectorial signal is hosted by a table (ifn1 and ifn2). The number of elements contained in both vectors must be the same. The result is a new vectorial control signal that overrides old values of ifn1. If you want to keep the old ifn1 vector, use vcopy_i opcode to copy it in another table. You can use idstoffset and isrcoffset to specify vectors in any location of the tables. Negative values for idstoffset and isrcoffset are acceptable. If idstoffset is negative, the out of range section of the vector will be discarded. If isrcoffset is negative, the out of range elements will be assumed to be 1 (i.e. the destination elements will be set to 1). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 1 (i.e. the destination vector elements will be set to 1). Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at init time. There's an k-rate version of this opcode called vexpv. All these operators (vaddv_i,vsubv_i,vmultv_i,vdivv_i,vpowv_i,vexpv_i, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/zacl.xml0000644000000000000000000000520512262561504016056 0ustar rootroot zacl Zak Patch System zacl zacl Clears one or more variables in the za space. Description Clears one or more variables in the za space. Syntax zacl kfirst, klast Performance kfirst -- first zk or za location in the range to clear. klast -- last zk or za location in the range to clear. zacl clears one or more variables in the za space. This is useful for those variables which are used as accumulators for mixing a-rate signals at each cycle, but which must be cleared before the next set of calculations. Examples Here is an example of the zacl opcode. It uses the file zacl.csd. Example of the zacl opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zamod, zar, zaw, zawm, ziw, ziwm Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/octcps.xml0000644000000000000000000001345612262561504016427 0ustar rootroot octcps Pitch Converters:Functions octcps octcps Converts a cycles-per-second value to octave-point-decimal. Description Converts a cycles-per-second value to octave-point-decimal. Syntax octcps (cps) (init- or control-rate args only) where the argument within the parentheses may be a further expression. Performance octcps and its related opcodes are really value converters with a special function of manipulating pitch data. Data concerning pitch and frequency can exist in any of the following forms: Pitch and Frequency Values Name Abbreviation octave point pitch-class (8ve.pc) pch octave point decimal oct cycles per second cps Midi note number (0-127) midinn
The first two forms consist of a whole number, representing octave registration, followed by a specially interpreted fractional part. For pch, the fraction is read as two decimal digits representing the 12 equal-tempered pitch classes from .00 for C to .11 for B. For oct, the fraction is interpreted as a true decimal fractional part of an octave. The two fractional forms are thus related by the factor 100/12. In both forms, the fraction is preceded by a whole number octave index such that 8.00 represents Middle C, 9.00 the C above, etc. Midi note number values range between 0 and 127 (inclusively) with 60 representing Middle C, and are usually whole numbers. Thus A440 can be represented alternatively by 440 (cps), 69 (midinn), 8.09 (pch), or 8.75 (oct). Microtonal divisions of the pch semitone can be encoded by using more than two decimal places. The mnemonics of the pitch conversion units are derived from morphemes of the forms involved, the second morpheme describing the source and the first morpheme the object (result). Thus cpspch(8.09) will convert the pitch argument 8.09 to its cps (or Hertz) equivalent, giving the value of 440. Since the argument is constant over the duration of the note, this conversion will take place at i-time, before any samples for the current note are produced. By contrast, the conversion cpsoct(8.75 + k1) which gives the value of A440 transposed by the octave interval k1. The calculation will be repeated every k-period since that is the rate at which k1 varies. Note The conversion from pch, oct, or midinn into cps is not a linear operation but involves an exponential process that could be time-consuming when executed repeatedly. Csound now uses a built-in table lookup to do this efficiently, even at audio rates. Because the table index is truncated without interpolation, pitch resolution when using one of these opcodes is limited to 8192 discrete and equal divisions of the octave, and some pitches of the standard 12-tone equally-tempered scale are very slightly mistuned (by at most 0.15 cents). Examples Here is an example of the octcps opcode. It uses the file octcps.csd. Example of the octcps opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: ioct = 8.750 See Also cpsoct, cpspch, octpch, pchoct, cpsmidinn, octmidinn, pchmidinn Credits Example written by &namekevin;. manual_src~dfsg/opcodes/resonz.xml0000644000000000000000000001701312262561504016445 0ustar rootroot resonz Signal Modifiers:Standard Filters:Resonant resonz resonz A bandpass filter with variable frequency response. Description Implementations of a second-order, two-pole two-zero bandpass filter with variable frequency response. Syntax ares resonz asig, xcf, xbw [, iscl] [, iskip] Initialization The optional initialization variables for resonr and resonz are identical to the i-time variables for reson. iskip -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. iscl -- coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise. A zero value signifies no scaling of the signal, leaving that to some later adjustment (see balance). The default value is 0. Performance resonr and resonz are variations of the classic two-pole bandpass resonator (reson). Both filters have two zeroes in their transfer functions, in addition to the two poles. resonz has its zeroes located at z = 1 and z = -1. resonr has its zeroes located at +sqrt(R) and -sqrt(R), where R is the radius of the poles in the complex z-plane. The addition of zeroes to resonr and resonz results in the improved selectivity of the magnitude response of these filters at cutoff frequencies close to 0, at the expense of less selectivity of frequencies above the cutoff peak. resonr and resonz are very close to constant-gain as the center frequency is swept, resulting in a more efficient control of the magnitude response than with traditional two-pole resonators such as reson. resonr and resonz produce a sound that is considerably different from reson, especially for lower center frequencies; trial and error is the best way of determining which resonator is best suited for a particular application. asig -- input signal to be filtered xcf -- cutoff or resonant frequency of the filter, measured in Hz xbw -- bandwidth of the filter (the Hz difference between the upper and lower half-power points) Technical History resonr and resonz were originally described in an article by Julius O. Smith and James B. Angell.1 Smith and Angell recommended the resonz form (zeros at +1 and -1) when computational efficiency was the main concern, as it has one less multiply per sample, while resonr (zeroes at + and - the square root of the pole radius R) was recommended for situations when a perfectly constant-gain center peak was required. Ken Steiglitz, in a later article 2, demonstrated that resonz had constant gain at the true peak of the filter, as opposed to resonr, which displayed constant gain at the pole angle. Steiglitz also recommended resonz for its sharper notches in the gain curve at zero and Nyquist frequency. Steiglitz's recent book 3 features a thorough technical discussion of reson and resonz, while Dodge and Jerse's textbook 4 illustrates the differences in the response curves of reson and resonz. Examples Here is an example of the resonr and resonz opcodes. It uses the file resonr.csd. Example of the resonr and resonz opcodes. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. References Smith, Julius O. and Angell, James B., "A Constant-Gain Resonator Tuned by a Single Coefficient," Computer Music Journal, vol. 6, no. 4, pp. 36-39, Winter 1982. Steiglitz, Ken, "A Note on Constant-Gain Digital Resonators," Computer Music Journal, vol. 18, no. 4, pp. 8-10, Winter 1994. Ken Steiglitz, A Digital Signal Processing Primer, with Applications to Digital Audio and Computer Music. Addison-Wesley Publishing Company, Menlo Park, CA, 1996. Dodge, Charles and Jerse, Thomas A., Computer Music: Synthesis, Composition, and Performance. New York: Schirmer Books, 1997, 2nd edition, pp. 211-214. See Also resonr Credits Author: &namesean; Seattle, Washington 1999 New in Csound version 3.55 Audio rate parameters introduced in version 6.02 November 2013. manual_src~dfsg/opcodes/dates.xml0000644000000000000000000000404512262561504016226 0ustar rootroot dates Instrument Control:Time Reading dates dates Returns as a string the date and time specified. Description Returns as a string the date and time specified. Syntax Sir dates [ itime] Initialization itime -- the time is seconds since the start of the epoch. If omited or negative the current time is taken. Sir -- the date and time as a string. Examples Here is an example of the dates opcode. It uses the file dates.csd. Example of the dates opcode. Its output should include lines like this: File '2011_Jan_05_19_14_46.wav' written to the same directory as this CSD file is! Closing file '/home/user/csound/Output/2011_Jan_05_19_14_46.wav'... See Also date Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK December 2006 New in Csound version 5.05 manual_src~dfsg/opcodes/trlowest.xml0000644000000000000000000000505512262561504017013 0ustar rootroot trlowest Spectral Processing:Streaming trlowest trlowest Extracts the lowest-frequency track from a streaming track input signal. Description The trlowest opcode takes an input containg TRACKS pv streaming signals (as generated, for instance by partials) and outputs only the lowest track. In addition it outputs two k-rate signals, corresponding to the frequency and amplitude of the lowest track signal. Syntax fsig, kfr, kamp trlowest fin1, kscal Performance fsig -- output pv stream in TRACKS format kfr -- frequency (in Hz) of the lowest-frequency track kamp -- amplitude of the lowest-frequency track fin -- input pv stream in TRACKS format. kscal -- amplitude scaling of output. Examples Here is an example of the trlowest opcode. It uses the file trlowest.csd. Example of the trlowest opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of an ifd-analysis signal, extraction of the lowest frequency and resynthesis. Credits Author: &namevictor; February 2006 New in Csound 5.01 manual_src~dfsg/opcodes/sfplist.xml0000644000000000000000000000743112262561504016614 0ustar rootroot sfplist Signal Generators:Sample Playback sfplist sfplist Prints a list of all presets of a SoundFont2 (SF2) sample file. Description Prints a list of all presets of a previously loaded SoundFont2 (SF2) sample file. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. Syntax sfplist ifilhandle Initialization ifilhandle -- unique number generated by sfload opcode to be used as an identifier for a SF2 file. Several SF2 files can be loaded and activated at the same time. Performance sfplist prints a list of all presets of a previously loaded SF2 file to the console. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. Examples Here is an example of the sfplist opcode. It uses the file sfplist.csd. Example of the sfplist opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: Preset list of "sf_GMbank.sf2" 0) Piano 1 prog:0 bank:0 1) Piano 2 prog:1 bank:0 2) Piano 3 prog:2 bank:0 3) Honky Tonk prog:3 bank:0 4) E.Piano 1 prog:4 bank:0 5) E.Piano 2 prog:5 bank:0 6) Harpsichord prog:6 bank:0 ........ 146) Car-Pass prog:125 bank:3 ........ See Also sfilist, sfinstr, sfinstrm, sfload, sfpassign, sfplay, sfplaym, sfpreset Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/randomi.xml0000644000000000000000000001127212262561504016557 0ustar rootroot randomi Signal Generators:Random (Noise) Generators randomi randomi Generates a user-controlled random number series with interpolation between each new number. Description Generates a user-controlled random number series with interpolation between each new number. Syntax ares randomi kmin, kmax, xcps [,imode] [,ifirstval] kres randomi kmin, kmax, kcps [,imode] [,ifirstval] Initialization imode (optional, default=0) -- first interpolation cycle mode (see below) ifirstval (optional, default=0) -- first output value Performance kmin -- minimum range limit kmax -- maximum range limit kcps, xcps -- rate of random break-point generation The randomi opcode is similar to randi but allows the user to set arbitrary minimum and maximum values. When imode = 0 (the default), the kmin argument value is outputted during 1/kcps (resp. 1/xcps) seconds at the beginning of the note, before the first random number is generated. Then the normal interpolation process takes place, first between kmin and the first random number generated, and then between successive generated random numbers, each interpolation cycle having a duration of 1/kcps (resp. 1/xcps) seconds. When imode = 1, a random number is generated at initialization and interpolation begins immediately between the kmin argument value and that random number. When imode = 2, a random number is generated at initialization and interpolation begins immediately between the ifirstval argument value and that random number. When imode = 3, two random numbers are generated at initialization as breakpoints for the first interpolation cycle. Examples Here is an example of the randomi opcode. It uses the file randomi.csd. Example of the randomi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: Mode: 0 k1 = 220.000000 k1 = 220.000000 k1 = 220.146093 k1 = 246.827703 k1 = 395.595775 ... Mode: 1 k1 = 220.000000 k1 = 224.325329 k1 = 274.370074 k1 = 343.216049 k1 = 414.324347 ... Mode: 2 k1 = 330.000000 k1 = 292.628171 k1 = 334.519777 k1 = 290.610602 k1 = 394.905366 ... Mode: 3 k1 = 360.727674 k1 = 431.680412 k1 = 380.625254 k1 = 289.267139 k1 = 303.038109 ... See Also randi, random, randomh Credits Author: &namegabriel; Arguments imode and ifirstval added by &namepinot;, Jan. 2011, after a discussion with Peiman Khosravi on the csnd list. Example written by &namekevin;, adapted for new args by &namepinot;. manual_src~dfsg/opcodes/ion.xml0000644000000000000000000000074312262561504015714 0ustar rootroot ion ion ion Deprecated. Description Deprecated as of version 3.52. Use the noteon opcode instead. manual_src~dfsg/opcodes/distort1.xml0000644000000000000000000000647012262561504016703 0ustar rootroot distort1 Signal Modifiers:Special Effects distort1 distort1 Modified hyperbolic tangent distortion. Description Implementation of modified hyperbolic tangent distortion. distort1 can be used to generate wave shaping distortion based on a modification of the tanh function. exp(asig * (shape1 + pregain)) - exp(asig * (shape2 - pregain)) aout = --------------------------------------------------------------- exp(asig * pregain) + exp(-asig * pregain) Syntax ares distort1 asig, kpregain, kpostgain, kshape1, kshape2[, imode] Initialization imode (Csound version 5.00 and later only; optional, defaults to 0) -- scales kpregain, kpostgain, kshape1, and kshape2 for use with audio signals in the range -32768 to 32768 (imode=0), -0dbfs to 0dbfs (imode=1), or disables scaling of kpregain and kpostgain and scales kshape1 by kpregain and kshape2 by -kpregain (imode=2). Performance asig -- is the input signal. kpregain -- determines the amount of gain applied to the signal before waveshaping. A value of 1 gives slight distortion. kpostgain -- determines the amount of gain applied to the signal after waveshaping. kshape1 -- determines the shape of the positive part of the curve. A value of 0 gives a flat clip, small positive values give sloped shaping. kshape2 -- determines the shape of the negative part of the curve. Examples Here is an example of the distort1 opcode. It uses the file distort1.csd. Example of the distort1 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namehans; December 1998 New in Csound version 3.50 manual_src~dfsg/opcodes/expon.xml0000644000000000000000000000543712262561504016265 0ustar rootroot expon Signal Generators:Linear and Exponential Generators expon expon Trace an exponential curve between specified points. Description Trace an exponential curve between specified points. Syntax ares expon ia, idur, ib kres expon ia, idur, ib Initialization ia -- starting value. Zero is illegal for exponentials. ib -- value after idur seconds. For exponentials, must be non-zero and must agree in sign with ia. idur -- duration in seconds of the segment. A zero or negative value will cause all initialization to be skipped. Performance These units generate control or audio signals whose values can pass through 2 specified points. The idur value may or may not equal the instrument's performance time: a shorter performance will truncate the specified pattern, while a longer one will cause the defined segment to continue on in the same direction. Examples Here is an example of the expon opcode. It uses the file expon.csd. Example of the expon opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also expseg, expsegr, line, linseg, linsegr manual_src~dfsg/opcodes/outrg.xml0000644000000000000000000000544112262561504016267 0ustar rootroot outrg Signal I/O:Signal Output outrg outrg Allow output to a range of adjacent audio channels on the audio output device Description outrg outputs audio to a range of adjacent audio channels on the audio output device. Syntax outrg kstart, aout1 [,aout2, aout3, ..., aoutN] Performance kstart - the number of the first channel of the output device to be accessed (channel numbers starts with 1, which is the first channel) aout1, aout2, ... aoutN - the arguments containing the audio to be output to the corresponding output channels. outrg allows to output a range of adjacent channels to the output device. kstart indicates the first channel to be accessed (channel 1 is the first channel). The user must be sure that the number obtained by summing kstart plus the number of accessed channels -1 is <= nchnls. Examples Here is an example of the outrg opcode. It uses the file outrg.csd. Example of the outrg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: speaker 1 position 0.000200 speaker 1 position 0.000400 speaker 1 position 0.000600 ..... speaker 1 position 1.000000 speaker 2 position 0.000000 speaker 2 position 0.000200 .... speaker 2 position 0.999800 speaker 3 position 0.000000 speaker 3 position 0.000200 .... speaker 3 position 0.999600 speaker 4 position 0.000000 ..... Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/mixersend.xml0000644000000000000000000000744712262561504017135 0ustar rootroot MixerSend Mixer Opcodes MixerSend MixerSend Mixes an arate signal into a channel of a buss. Syntax MixerSend asignal, isend, ibuss, ichannel Description Mixes an arate signal into a channel of a buss. Initialization isend -- The number of the send, for example the number of the instrument sending the signal. The gain of the send is controlled by the MixerSetLevel opcode. The reason that the sends are numbered is to enable different levels for different sends to be set independently of the actual level of the signals. ibuss -- The number of the buss, for example the number of the instrument receiving the signal. ichannel -- The number of the channel. Each buss has nchnls channels. Performance asignal -- The signal that will be mixed into the indicated channel of the buss. Use of the mixer requires that instruments setting gains have smaller numbers than instruments sending signals, and that instruments sending signals have smaller numbers than instruments receiving those signals. However, an instrument may have any number of sends or receives. After the final signal is received, MixerClear must be invoked to reset the busses to 0 before the next kperiod. Examples instr 100 ; Fluidsynth output ; INITIALIZATION ; Normalize so iamplitude for p5 of 80 == ampdb(80). iamplitude = ampdb(p5) * 2.0 ; AUDIO aleft, aright fluidAllOut giFluidsynth asig1 = aleft * iamplitude asig2 = aright * iamplitude ; To the chorus. MixerSend asig1, 100, 200, 0 MixerSend asig2, 100, 200, 1 ; To the reverb. MixerSend asig1, 100, 210, 0 MixerSend asig2, 100, 210, 1 ; To the output. MixerSend asig1, 100, 220, 0 MixerSend asig2, 100, 220, 1 endin Here is a complete example of the MixerSend opcode. It uses the file Mixer.csd Complete example of the MixerSend opcode. Credits Michael Gogins (gogins at pipeline dot com). manual_src~dfsg/opcodes/outkc.xml0000644000000000000000000000755512262561504016264 0ustar rootroot outkc Real-time MIDI:Output outkc outkc Sends MIDI controller messages at k-rate. Description Sends MIDI controller messages at k-rate. Syntax outkc kchn, knum, kvalue, kmin, kmax Performance kchn -- MIDI channel number (1-16) knum -- controller number (0-127 for example 1 = ModWheel; 2 = BreathControl etc.) kvalue -- floating point value kmin -- minimum floating point value (converted in MIDI integer value 0) kmax -- maximum floating point value (converted in MIDI integer value 127 (7 bit)) outkc (k-rate MIDI controller output) sends controller messages to MIDI OUT device. It works only with MIDI instruments which recognize them. It can drive a different value of a parameter for each note currently active. It can scale the k-value floating-point argument according to the kmin and kmax values. For example: set kmin = 1.0 and kmax = 2.0. When the kvalue argument receives a 2.0 value, the opcode will send a 127 value to the MIDI OUT device. When the kvalue argument receives a 1.0 value, it will send a 0 value. k-rate opcodes send a message each time the MIDI converted value of argument kvalue changes. Examples Here is an example of the outkc opcode. It uses the file outkc.csd. Example of the outkc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also outiat, outic14, outic, outipat, outipb, outipc, outkat, outkc14, outkpat, outkpb, outkpc Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/sfplay3m.xml0000644000000000000000000001614612262561504016671 0ustar rootroot sfplay3m Signal Generators:Sample Playback sfplay3m sfplay3m Plays a SoundFont2 (SF2) sample preset, generating a mono sound with cubic interpolation. Description Plays a SoundFont2 (SF2) sample preset, generating a mono sound with cubic interpolation. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. Syntax ares sfplay3m ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset] [, ienv] Initialization ivel -- velocity value inotenum -- MIDI note number value ipreindex -- preset index iflag (optional) -- flag regarding the behavior of xfreq and inotenum ioffset (optional) -- start playing at offset, in samples. ienv (optional) -- enables and determines amplitude envelope. 0 = no envelope, 1 = linear attack and decay, 2 = linear attack, exponential decay (see below). Default = 0. Performance xamp -- amplitude correction factor xfreq -- frequency value or frequency multiplier, depending by iflag. When iflag = 0, xfreq is a multiplier of a the default frequency, assigned by SF2 preset to the inotenum value. When iflag = 1, xfreq is the absolute frequency of the output sound, in Hz. Default is 0. When iflag = 0, inotenum sets the frequency of the output according to the MIDI note number used, and xfreq is used as a multiplier. When iflag = 1, the frequency of the output, is set directly by xfreq. This allows the user to use any kind of micro-tuning based scales. However, this method is designed to work correctly only with presets tuned to the default equal temperament. Attempts to use this method with a preset already having non-standard tunings, or with drum-kit-based presets, could give unexpected results. Adjustment of the amplitude can be done by varying the xamp argument, which acts as a multiplier. Notice that both xamp and xfreq can use k-rate as well as a-rate signals. Both arguments must use variables of the same rate, or sfplay3m will not work correctly. ipreindex must contain the number of a previously assigned preset, or Csound will crash. The ioffset parameter allows the sound to start from a sample different than the first one. The user should make sure that its value is within the length of the specific sound. Otherwise, Csound will probably crash. The ienv parameter enables and determines the type of amplitude envelope used. The default value is 0, or no envelope. If ienv is set to 1, the attack and decay portions are linear. If set to 2, the attack is linear and the decay is exponential. The release portion of the envelope has not yet been implemented. sfplay3m is a mono version of sfplay3. It should be used with mono preset, or with the stereo presets in which stereo output is not required. It is faster than sfplay3. sfplay3m is also a cubic-interpolation version of sfplaym. Difference of sound-quality is noticeable specially in bass-frequency-transposed samples. In high-freq-transposed samples the difference is less noticeable, and I suggest to use linear-interpolation versions, because they are faster. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. Examples Here is an example of the sfplay3m opcode. It uses the file sfplay3m.csd. Example of the sfplay3m opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also sfilist, sfinstr3, sfinstr3m, sfinstr, sfinstrm, sfload, sfpassign, sfplay3, sfplaym, sfplay, sfplist, sfpreset Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 New optional parameter ienv in version 5.09 manual_src~dfsg/opcodes/timek.xml0000644000000000000000000000606212262561504016240 0ustar rootroot timek Instrument Control:Time Reading timek timek Read absolute time in k-rate cycles. Description Read absolute time, in k-rate cycles, since the start of the performance. Syntax ires timek kres timek Performance timek is for time in k-rate cycles. So with: sr = 44100 kr = 6300 ksmps = 7 then after half a second, the timek opcode would report 3150. It will always report an integer. timek can produce a k-rate variable for output. There are no input parameters. timek can also operate only at the start of the instance of the instrument. It produces an i-rate variable (starting with i or gi) as its output. Examples Here is an example of the timek opcode. It uses the file timek.csd. Example of the timek opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: k1 = 1.000000 samples k1 = 2205.000000 samples k1 = 4410.000000 samples k1 = 6615.000000 samples k1 = 8820.000000 samples See Also timeinstk, timensts, times Credits Author: Robin Whittle Australia May 1997 New in version 3.47 Example written by &namekevin;. manual_src~dfsg/opcodes/linen.xml0000644000000000000000000001007412262561504016232 0ustar rootroot linen Signal Generators:Envelope Generators linen linen Applies a straight line rise and decay pattern to an input amp signal. Description linen -- apply a straight line rise and decay pattern to an input amp signal. Syntax ares linen xamp, irise, idur, idec kres linen kamp, irise, idur, idec Initialization irise -- rise time in seconds. A zero or negative value signifies no rise modification. idur -- overall duration in seconds. A zero or negative value will cause initialization to be skipped. idec -- decay time in seconds. Zero means no decay. An idec > idur will cause a truncated decay. Performance kamp, xamp -- input amplitude signal. Rise modifications are applied for the first irise seconds, and decay from time idur - idec. If these periods are separated in time there will be a steady state during which amp will be unmodified. If linen rise and decay periods overlap then both modifications will be in effect for that time. If the overall duration idur is exceeded in performance, the final decay will continue on in the same direction, going negative. [Envelope generated by the linen opcode] Envelope generated by the linen opcode A common error with this opcode is to assume that the value of 0 is the held after the envelope has finished at idur. linen does not automatically end or stop at the end of the duration given. If your note length is longer than idur seconds, kres (or ares) will not come to rest at 0, but will instead continue to fall with the same rate. If a decay and then hold is required then the linseg opcode should be considered instead. Examples Here is an example of the linen opcode. It uses the file linen.csd. Example of the linen opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also envlpx, envlpxr, linenr manual_src~dfsg/opcodes/vdivv_i.xml0000644000000000000000000001033112262561504016567 0ustar rootroot vdivv_i Vectorial:Vectorial operations vdivv_i vdivv_i Performs division between two vectorial control signals at init time. Description Performs division between two vectorial control signals at init time. Syntax vdivv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Initialization ifn1 - number of the table hosting the first vector to be processed ifn2 - number of the table hosting the second vector to be processed ielements - number of elements of the two vectors idstoffset - index offset for the destination (ifn1) table (Default=0) isrcoffset - index offset for the source (ifn2) table (Default=0) Performance vdivv_i divides two vectorial control signals, that is, each element of ifn1 is divided by the corresponding element of ifn2. Each vectorial signal is hosted by a table (ifn1 and ifn2). The number of elements contained in both vectors must be the same. The result is a new vectorial control signal that overrides old values of ifn1. If you want to keep the old ifn1 vector, use vcopy_i opcode to copy it in another table. You can use idstoffset and isrcoffset to specify vectors in any location of the tables. Negative values for idstoffset and isrcoffset are acceptable. If idstoffset is negative, the out of range section of the vector will be discarded. If isrcoffset is negative, the out of range elements will be assumed to be 1 (i.e. the destination elements will not be changed). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 1 (i.e. the destination vector will not be changed for these elements). Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at init time. There's an k-rate version of this opcode called vdivv. All these operators (vaddv_i,vsubv_i,vmultv_i,vdivv_i,vpowv_i,vexpv_i, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/serialWrite_i.xml0000644000000000000000000000335312262561504017731 0ustar rootroot serialrite_i Serial I/O serialWrite_i serialWrite_i Write data to a serial port. Description Write data to a serial port for arduino. Syntax serialWrite_i iPort, iByte serialWrite_i iPort, SBytes Initialization iPort -- port number optained from a serialBegin opcode. iByte -- a byte of data to write. See Also serialBegin, serialEnd, serialWrite, serialRead, serialPrint, serialFlush. Credits Author: &namematt; 2011 New in version 5.14 manual_src~dfsg/opcodes/noteon.xml0000644000000000000000000000503512262561504016430 0ustar rootroot noteon Real-time MIDI:Note Output noteon noteon Send a noteon message to the MIDI OUT port. Description Send a noteon message to the MIDI OUT port. Syntax noteon ichn, inum, ivel Initialization ichn -- MIDI channel number (1-16) inum -- note number (0-127) ivel -- velocity (0-127) Performance noteon (i-rate note on) and noteoff (i-rate note off) are the simplest MIDI OUT opcodes. noteon sends a MIDI noteon message to MIDI OUT port, and noteoff sends a noteoff message. A noteon opcode must always be followed by an noteoff with the same channel and number inside the same instrument, otherwise the note will play endlessly. These noteon and noteoff opcodes are useful only when introducing a timout statement to play a non-zero duration MIDI note. For most purposes, it is better to use noteondur and noteondur2. See Also noteoff, noteondur, noteondur2 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/STKModalBar.xml0000644000000000000000000000750612262561504017176 0ustar rootroot STKModalBar Signal Generators:STK Opcodes STKModalBar STKModalBar STKModalBar is a resonant bar instrument. Description This opcode is a resonant bar instrument.It has a number of different struck bar instruments. Syntax asignal STKModalBar ifrequency, iamplitude, [khard, kv1[, kpos, kv2[, klfo, kv3[, klfodepth, kv4[, kmix, kv5[, kvol, kv6[, kinstr, kv7]]]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance khard -- controller 2, hardness of the stick. Value range of kv1 is 0-127. kpos -- controller 4, stick position. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kmix -- controller 8, direct stick mix. Value range of kv5 is 0-127. kvol -- controller 128, volume. Value range of kv6 is 0-127. kinstr -- controller 16, instrument presets (0 = marimba, 1 = vibraphone, 2 = agogo, 3 = wood1, 4 = reso, 5 = wood2, 6 = beats, 7 = two fixed, 8 = clump). Value range of kv7 is 0-16. Note The code for this opcode is taken directly from the ModalBar class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKModalBar opcode. It uses the file STKModalBar.csd. Example of the STKModalBar opcode. Credits Author: &namemichael; (after Georg Essl) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/vco.xml0000644000000000000000000001321012262561504015707 0ustar rootroot vco Signal Generators:Dynamic Spectrum Oscillators vco vco Implementation of a band limited, analog modeled oscillator. Description Implementation of a band limited, analog modeled oscillator, based on integration of band limited impulses. vco can be used to simulate a variety of analog wave forms. Syntax ares vco xamp, xcps, iwave, kpw [, ifn] [, imaxd] [, ileak] [, inyx] \ [, iphs] [, iskip] Initialization iwave -- determines the waveform: iwave = 1 - sawtooth iwave = 2 - square/PWM iwave = 3 - triangle/saw/ramp ifn (optional, default = 1) -- should be the table number of a of a stored sine wave. Must point to a valid table which contains a sine wave. Csound will report an error if this parameter is not set and table 1 doesn't exist. imaxd (optional, default = 1) -- is the maximum delay time. A time of 1/ifqc may be required for the PWM and triangle waveform. To bend the pitch down this value must be as large as 1/(minimum frequency). ileak (optional, default = 0) -- if ileak is between zero and one (0 < ileak < 1) then ileak is used as the leaky integrator value. Otherwise a leaky integrator value of .999 is used for the saw and square waves and .995 is used for the triangle wave. This can be used to flatten the square wave or straighten the saw wave at low frequencies by setting ileak to .99999 or a similar value. This should give a hollower sounding square wave. inyx (optional, default = .5) -- this is used to determine the number of harmonics in the band limited pulse. All overtones up to sr * inyx will be used. The default gives sr * .5 (sr/2). For sr/4 use inyx = .25. This can generate a fatter sound in some cases. iphs (optional, default = 0) -- this is a phase value. There is an artifact (bug-like feature) in vco which occurs during the first half cycle of the square wave which causes the waveform to be greater in magnitude than all others. The value of iphs has an effect on this artifact. In particular setting iphs to .5 will cause the first half cycle of the square wave to resemble a small triangle wave. This may be more desirable than the large wave artifact which is the current default. iskip (optional, default = 0) -- if non zero skip the initialisation of the filter. (New in Csound version 4.23f13 and 5.0) Performance kpw -- determines either the pulse width (if iwave is 2) or the saw/ramp character (if iwave is 3) The value of kpw should be greater than 0 and less than 1. A value of 0.5 will generate either a square wave (if iwave is 2) or a triangle wave (if iwave is 3). xamp -- determines the amplitude xcps -- is the frequency of the wave in cycles per second. Examples Here is an example of the vco opcode. It uses the file vco.csd. Example of the vco opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vco2 Credits Author: &namehans; December 1998 New in Csound version 3.50 November 2002. Corrected the documentation for the kpw parameter thanks to &nameluis; and &namehans;. manual_src~dfsg/opcodes/imidic14.xml0000644000000000000000000000077612262561504016540 0ustar rootroot imidic14 imidic14 imidic14 Deprecated. Description Deprecated as of version 3.52. Use the midic14 opcode instead. manual_src~dfsg/opcodes/else.xml0000644000000000000000000000540612262561504016060 0ustar rootroot else Instrument Control:Program Flow Control else else Executes a block of code when an "if...then" condition is false. Description Executes a block of code when an "if...then" condition is false. Syntax else Performance else is used inside of a block of code between the "if...then" and endif opcodes. It defines which statements are executed when a "if...then" condition is false. Only one else statement may occur and it must be the last conditional statement before the endif opcode. Examples Here is an example of the else opcode. It uses the file else.csd. Example of the else opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also elseif, endif, goto, if, igoto, kgoto, tigoto, timout More information on this opcode: http://www.csounds.com/journal/2006spring/controlFlow.html , written by Steven Yi Credits New in version 4.21 manual_src~dfsg/opcodes/grain.xml0000644000000000000000000001075612262561504016234 0ustar rootroot grain Signal Generators:Granular Synthesis grain grain Generates granular synthesis textures. Description Generates granular synthesis textures. Syntax ares grain xamp, xpitch, xdens, kampoff, kpitchoff, kgdur, igfn, \ iwfn, imgdur [, igrnd] Initialization igfn -- The ftable number of the grain waveform. This can be just a sine wave or a sampled sound. iwfn -- Ftable number of the amplitude envelope used for the grains (see also GEN20). imgdur -- Maximum grain duration in seconds. This is the biggest value to be assigned to kgdur. igrnd (optional) -- if non-zero, turns off grain offset randomness. This means that all grains will begin reading from the beginning of the igfn table. If zero (the default), grains will start reading from random igfn table positions. Performance xamp -- Amplitude of each grain. xpitch -- Grain pitch. To use the original frequency of the input sound, use the formula: sndsr / ftlen(igfn) where sndsr is the original sample rate of the igfn sound. xdens -- Density of grains measured in grains per second. If this is constant then the output is synchronous granular synthesis, very similar to fof. If xdens has a random element (like added noise), then the result is more like asynchronous granular synthesis. kampoff -- Maximum amplitude deviation from xamp. This means that the maximum amplitude a grain can have is xamp + kampoff and the minimum is xamp. If kampoff is set to zero then there is no random amplitude for each grain. kpitchoff -- Maximum pitch deviation from xpitch in Hz. Similar to kampoff. kgdur -- Grain duration in seconds. The maximum value for this should be declared in imgdur. If kgdur at any point becomes greater than imgdur, it will be truncated to imgdur. The grain generator is based primarily on work and writings of Barry Truax and Curtis Roads. Examples This example generates a texture with gradually shorter grains and wider amp and pitch spread. It uses the file grain.csd, and beats.wav. Example of the grain opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &nameparis; MIT May 1997 manual_src~dfsg/opcodes/strindex.xml0000644000000000000000000000217612262561504016771 0ustar rootroot strindex Strings:Manipulation strindex strindex Return the position of the first occurence of a string in another string Description Return the position of the first occurence of S2 in S1, or -1 if not found. If S2 is empty, 0 is returned. strindex runs at init time only. Syntax ipos strindex S1, S2 See also strindexk Credits Author: Istvan Varga 2006 New in version 5.02 manual_src~dfsg/opcodes/hrtfmove2.xml0000644000000000000000000001221712262561504017042 0ustar rootroot hrtfmove2 Signal Modifiers:Panning and Spatialization hrtfmove2 hrtfmove2 Generates dynamic 3d binaural audio for headphones using a Woodworth based spherical head model with improved low frequency phase accuracy. Description This opcode takes a source signal and spatialises it in the 3 dimensional space around a listener using head related transfer function (HRTF) based filters. Syntax aleft, aright hrtfmove2 asrc, kAz, kElev, ifilel, ifiler [,ioverlap, iradius, isr] Initialization ifilel -- left HRTF spectral data file ifiler -- right HRTF spectral data file Spectral datafiles (based on the MIT HRTF database) are available in 3 different sampling rates: 44.1, 48 and 96 khz and are labelled accordingly. Input and processing sr should match datafile sr. Files should be in the current directory or the SADIR (see Environment Variables). HRTF Data files for use with hrtfmove, hrtfmove2, hrtfstat, hrtfearly, hrtfreverb were updated for Csound 5.15 and later (the code was updated and is more efficient). Old datafiles are now deprecated. ioverlap -- optional, number of overlaps for STFT processing (default 4). See STFT section of manual. iradius -- optional, head radius used for phase spectra calculation in centimeters (default 9.0) isr - optional, default 44.1kHz, legal values: 44100, 48000 and 96000. Performance asrc -- Input/source signal. kAz -- azimuth value in degrees. Positive values represent position on the right, negative values are positions on the left. kElev -- elevation value in degrees. Positive values represent position above horizontal, negative values are positions below horizontal (min -40). Artifact-free user-defined trajectories are made possible using an interpolation algorithm based on spectral magnitude interpolation and a derived phase spectrum based on the Woodworth spherical head model. Accuracy is increased for the data set provided by extracting and applying a frequency dependent scaling factor to the phase spectra, leading to a more precise low frequency interaural time difference. Users can control head radius for the phase derivation, allowing a crude level of individualisation. The dynamic source version of the opcode uses a Short Time Fourier Transform algorithm to avoid artefacts caused by derived phase spectra changes. STFT processing means this opcode is more computationally intensive than hrtfmove using phase truncation, but phase is constantly updated by hrtfmove2. Examples Here is an example of the hrtfmove2 opcode. It uses the file hrtfmove2.csd. Example of the htrfmove2 opcode. See Also hrtfmove, hrtfstat, hrtfer. More information on this opcode: http://www.csounds.com/journal/issue9/newHRTFOpcodes.html , written by Brian Carty Credits Author: Brian Carty Maynooth 2008 manual_src~dfsg/opcodes/ioutpc.xml0000644000000000000000000000076212262561504016433 0ustar rootroot ioutpc ioutpc ioutpc Deprecated. Description Deprecated as of version 3.52. Use the outipc opcode instead. manual_src~dfsg/opcodes/tableicopy.xml0000644000000000000000000000512512262561504017261 0ustar rootroot tableicopy Table Control:Read/Write Opreations tableicopy tableicopy Simple, fast table copy opcode. Description Simple, fast table copy opcode. Syntax tableicopy idft, isft Initialization idft -- Destination function table. isft -- Number of source function table. Performance tableicopy -- Simple, fast table copy opcodes. Takes the table length from the destination table, and reads from the start of the source table. For speed reasons, does not check the source length - just copies regardless - in "wrap" mode. This may read through the source table several times. A source table with length 1 will cause all values in the destination table to be written to its value. tableicopy cannot read or write the guardpoint. To read it use table, with ndx = the table length. Likewise use table write to write it. To write the guardpoint to the value in location 0, use tablegpw. This is primarily to change function tables quickly in a real-time situation. See Also tablecopy, tablegpw, tablemix, tableigpw, tableimix Credits Author: Robin Whittle Australia May 1997 New in version 3.47 manual_src~dfsg/opcodes/opk.xml0000644000000000000000000000356312262561504015723 0ustar rootroot a Signal Modifiers:Sample Level Operators k k Converts a i-rate parameter to an k-rate value. Or converts an a-rate value to a k-rate value by down-sampling. Description Converts an i-rate value to control rate, for example to be used with rnd() and birnd() to generate random numbers at k-rate. Syntax k(x) (i-rate args only) k(x) (a-rate args only) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. See Also i a More information on this opcode: http://www.csounds.com/journal/issue10/CsoundRates.html , written by Andrés Cabrera Credits Author: &nameistvan; New in version Csound 5.00 Author: &namejohn; arate to krate new in version Csound 6.00 manual_src~dfsg/opcodes/tempo.xml0000644000000000000000000000547612262561504016263 0ustar rootroot tempo Instrument Control:Sensing and Control tempo tempo Apply tempo control to an uninterpreted score. Description Apply tempo control to an uninterpreted score. Syntax tempo ktempo, istartempo Initialization istartempo -- initial tempo (in beats per minute). Typically 60. Performance ktempo -- The tempo to which the score will be adjusted. tempo allows the performance speed of Csound scored events to be controlled from within an orchestra. It operates only in the presence of the Csound -t flag. When that flag is set, scored events will be performed from their uninterpreted p2 and p3 (beat) parameters, initially at the given command-line tempo. When a tempo statement is activated in any instrument (ktempo 0.), the operating tempo will be adjusted to ktempo beats per minute. There may be any number of tempo statements in an orchestra, but coincident activation is best avoided. Examples Here is an example of the tempo opcode. Remember, it only works if you use the -t flag with Csound. The example uses the file tempo.csd. Example of the tempo opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also tempoval Credits Example written by &namekevin;. manual_src~dfsg/opcodes/readk3.xml0000644000000000000000000001357712262561504016311 0ustar rootroot readk3 Signal I/O:File I/O readk3 readk3 Periodically reads three orchestra control-signal values from an external file. Description Periodically reads three orchestra control-signal values from an external file. Syntax kr1, kr2, kr3 readk3 ifilname, iformat, iprd Initialization ifilname -- an integer N denoting a file named "readk.N" or a character string (in double quotes, spaces permitted) denoting the external file name. For a string, it may either be a full path name with directory specified or a simple filename. In the later case, the file is sought first in the current directory, then in SSDIR, and finally in SFDIR. iformat -- specifies the input data format: 1 = 8-bit signed integers (char) 4 = 16-bit short integers 5 = 32-bit long integers 6 = 32-bit floats 7 = ASCII long integers (plain text) 8 = ASCII floats (plain text) Note that A-law and U-law formats are not available, and that all formats except the last two are binary. The input file should be a "raw", headerless data file. iprd -- the rate (period) in seconds, rounded to the nearest orchestra control period, at which the signals are read from the input file. A value of 0 implies one control period (the enforced minimum), which will read new values at the orchestra control rate. Longer periods will cause the same values to repeat for more than one control period. Performance kr1, kr2, kr3 -- output of the signals read from ifilname. This opcode allows three generated control signal values to be read from a named external file. The file should contain no header information but it should contain a regularly sampled time series of control values. For binary formats, the individual samples of each signal are interleaved. For ASCII text formats, the values are assumed to be separated by at least one whitespace character. The three "channels" in a sample frame may be on the same line or separated by newline characters, it does not matter. There may be any number of readk3 opcodes in an instrument or orchestra and they may read from the same or different files. Examples Here is an example of the readk3 opcode. It uses the file readk3.csd. Example of the readk3 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output should include lines like these: WRITING: kfreq = 473.352855, kdb = -15.197657, kpan = 0.366764 kfreq = 473.352855, kdb = -15.197657, kpan = 0.366764 kfreq = 441.426368, kdb = -19.026206, kpan = 0.207327 kfreq = 452.965140, kdb = -21.447486, kpan = 0.553270 kfreq = 585.106328, kdb = -11.903852, kpan = 0.815665 kfreq = 482.056760, kdb = -4.046744, kpan = 0.876537 READING: kfreq = 473.352900, kdb = -15.197700, kpan = 0.366800 kfreq = 473.352900, kdb = -15.197700, kpan = 0.366800 kfreq = 441.426400, kdb = -19.026200, kpan = 0.207300 kfreq = 452.965100, kdb = -21.447500, kpan = 0.553300 kfreq = 585.106300, kdb = -11.903900, kpan = 0.815700 kfreq = 482.056800, kdb = -4.046700, kpan = 0.876500 See Also dumpk, dumpk2, dumpk3, dumpk4, readk, readk2, readk4 Credits By: John ffitch and Barry Vercoe 1999 or earlier manual_src~dfsg/opcodes/atrirand.xml0000644000000000000000000000077612262561504016741 0ustar rootroot atrirand atrirand atrirand Deprecated. Description Deprecated as of version 3.49. Use the trirand opcode instead. manual_src~dfsg/opcodes/cuserrnd.xml0000644000000000000000000001177712262561504016765 0ustar rootroot cuserrnd Signal Generators:Random (Noise) Generators cuserrnd cuserrnd Continuous USER-defined-distribution RaNDom generator. Description Continuous USER-defined-distribution RaNDom generator. Syntax aout cuserrnd kmin, kmax, ktableNum iout cuserrnd imin, imax, itableNum kout cuserrnd kmin, kmax, ktableNum Initialization imin -- minimum range limit imax -- maximum range limit itableNum -- number of table containing the random-distribution function. Such table is generated by the user. See GEN40, GEN41, and GEN42. The table length does not need to be a power of 2 Performance ktableNum -- number of table containing the random-distribution function. Such table is generated by the user. See GEN40, GEN41, and GEN42. The table length does not need to be a power of 2 kmin -- minimum range limit kmax -- maximum range limit cuserrnd (continuous user-defined-distribution random generator) generates random values according to a continuous random distribution created by the user. In this case the shape of the distribution histogram can be drawn or generated by any GEN routine. The table containing the shape of such histogram must then be translated to a distribution function by means of GEN40 (see GEN40 for more details). Then such function must be assigned to the XtableNum argument of cuserrnd. The output range can then be rescaled according to the Xmin and Xmax arguments. cuserrnd linearly interpolates between table elements, so it is not recommended for discrete distributions (GEN41 and GEN42). For a tutorial about random distribution histograms and functions see: D. Lorrain. "A panoply of stochastic cannons". In C. Roads, ed. 1989. Music machine. Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the cuserrnd opcode. It uses the file cuserrnd.csd. Example of the cuserrnd opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i 1 time 0.00067: 53.14918 i 1 time 0.20067: 0.00000 i 1 time 0.40067: 0.00000 i 1 time 0.60067: 96.80406 i 1 time 0.80067: 94.20729 i 1 time 1.00000: 0.00000 i 1 time 1.20067: 86.13032 i 1 time 1.40067: 31.37096 i 1 time 1.60067: 70.35889 i 1 time 1.80000: 0.00000 i 1 time 2.00000: 49.18914 WARNING: Seeding from current time 2006647442 i 2 time 3.00067: 21.45002 i 2 time 3.20067: 44.32333 i 2 time 3.40067: 46.05420 i 2 time 3.60000: 0.00000 i 2 time 3.80067: 41.32175 i 2 time 4.00000: 0.00000 i 2 time 4.20000: 63.72019 i 2 time 4.40067: 0.00000 i 2 time 4.60067: 0.00000 i 2 time 4.80067: 0.00000 i 2 time 5.00000: 74.49330 See Also duserrnd, urd Credits Author: &namegabriel; New in Version 4.16 manual_src~dfsg/opcodes/ftgenonce.xml0000644000000000000000000000671512262561504017104 0ustar rootroot ftgenonce Signal Flow Graph Opcodes ftgenonce ftgenonce Generate a function table from within an instrument definition, without duplication of data. Description Enables the creation of function tables entirely inside instrument definitions, without any duplication of data. The ftgenonce opcode is designed to simplify writing instrument definitions that can be re-used in different orchestras simply by #including them and plugging them into some output instrument. There is no need to define function tables either in the score, or in the orchestra header. The ftgenonce opcode is similar to ftgentmp, and has identical arguments. However, function tables are neither duplicated nor deleted. Instead, all of the arguments to the opcode are concatenated to form the key to a lookup table that points to the function table number. Thus, every request to ftgenonce with the same arguments receives the same instance of the function table data. Every change in the value of any ftgenonce argument causes the creation of a new function table. Syntax ifno ftgenonce ip1, ip2dummy, isize, igen, iarga, iargb, ... Initialization ifno -- an automatically assigned table number. ip1 -- the number of the table to be generated or 0 if the number is to be assigned. ip2dummy -- ignored. isize -- table size. Corresponds to p3 of the score f statement. igen -- function table GEN routine. Corresponds to p4 of the score f statement. iarga, iargb, ... -- function table arguments. Correspond to p5 through pn of the score f statement. Csound was originally designed to support tables with power of two sizes only. Though this has changed in recent versions (you can use any size by using a negative number), many opcodes will not accept them. Examples Here is an example of the ftgenonce opcode. It uses the file ftgenonce.csd. Example of the ftgenonce opcode. Credits Authors: &namemichael; 2009 manual_src~dfsg/opcodes/invalue.xml0000644000000000000000000000421712262561504016572 0ustar rootroot invalue Signal I/O:Signal Input invalue invalue Reads a k-rate signal from a user-defined channel. Description Reads a k-rate signal or string from a user-defined channel. Syntax kvalue invalue "channel name" Sname invalue "channel name" Performance kvalue -- The k-rate value that is read from the channel. Sname -- The string variable that is read from the channel. "channel name" -- An integer, string (in double-quotes), or string variable identifying the channel. Examples Here is an example of the invalue opcode. It uses the file invalue.csd. Example of the invalue opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also outvalue Credits Author: Matt Ingalls manual_src~dfsg/opcodes/opbitor.xml0000644000000000000000000000322212262561504016600 0ustar rootroot | Mathematical Operations:Arithmetic and Logic Operations | | Bitwise OR operator. Description The bitwise operators perform operations of bitwise AND, bitwise OR, bitwise NOT and bitwise non-equivalence. Syntax a | b (bitwise OR) where the arguments a and b may be further expressions. They are converted to the nearest integer to machine precision and then the operation is performed. Performance The priority of these operators is less binding that the arithmetic ones, but more binding that the comparisons. Parentheses may be used as above to force particular groupings. For an example of usage, see the entry for & See Also &, #, ¬ manual_src~dfsg/opcodes/chnmix.xml0000644000000000000000000000342712262561504016417 0ustar rootroot chnmix Signal I/O:Software Bus chnmix chnmix Writes audio data to the named software bus, mixing to the previous output. Description Adds an audio signal to a channel of the named software bus. Implies declaring the channel with imode=2 (see also chn_a). Syntax chnmix aval, Sname Initialization Sname -- a string that indicates which named channel of the software bus to write. Performance aval -- the audio signal to write at performance time. Examples Here is an example of the chnmix opcode. It uses the file chnmix.csd. Example of the chnmix opcode. Credits Author: &nameistvan; 2006 manual_src~dfsg/opcodes/transeg.xml0000644000000000000000000000755312262561504016600 0ustar rootroot transeg Signal Generators:Linear and Exponential Generators transeg transeg Constructs a user-definable envelope. Description Constructs a user-definable envelope. Syntax ares transeg ia, idur, itype, ib [, idur2] [, itype] [, ic] ... kres transeg ia, idur, itype, ib [, idur2] [, itype] [, ic] ... Initialization ia -- starting value. ib, ic, etc. -- value after idur seconds. idur -- duration in seconds of first segment. A zero or negative value will cause all initialization to be skipped. idur2,... idurx etc. -- duration in seconds of segment itype, itype2, etc. -- if 0, a straight line is produced. If non-zero, then transeg creates the following curve, for n steps: ibeg + (ivalue - ibeg) * (1 - exp( i*itype/(n-1) )) / (1 - exp(itype)) Performance If itype > 0, there is a slowly rising (concave) or slowly decaying (convex) curve, while if itype < 0, the curve is fast rising (convex) or fast decaying (concave). See also GEN16. Examples Here is an example of the transeg opcode. It uses the file transeg.csd. The example produces the following output: Output of the transeg example. Output of the transeg example. Example of the transeg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also expsega, expsegr, linseg, linsegr, transegr Credits Author: &namejohn; University of Bath, Codemist. Ltd. Bath, UK October 2000 New in Csound version 4.09 Thanks goes to Matt Gerassimoff for pointing out the correct command syntax. manual_src~dfsg/opcodes/loop_ge.xml0000644000000000000000000000573112262561504016555 0ustar rootroot loop_ge Instrument Control:Program Flow Control loop_ge loop_ge Looping constructions. Description Construction of looping operations. Syntax loop_ge indx, idecr, imin, label loop_ge kndx, kdecr, kmin, label Initialization indx -- i-rate variable to count loop. idecr -- value to decrement the loop. imin -- minimum value of loop index. Performance kndx -- k-rate variable to count loop. kdecr -- value to decrement the loop. kmin -- minimum value of loop index. The actions of loop_ge are equivalent to the code indx = indx - idecr if (indx >= imin) igoto label or kndx = kndx - kdecr if (kndx >= kmin) kgoto label See Also loop_gt, loop_le and loop_lt. More information on this opcode: http://www.csounds.com/journal/2006summer/controlFlow_part2.html , written by Steven Yi. And in the Floss Manuals: http://en.flossmanuals.net/csound/ch018_c-control-structures Credits &nameistvan;. 2006 New in Csound version 5.01 manual_src~dfsg/opcodes/mixergetlevel.xml0000644000000000000000000000471512262561504020006 0ustar rootroot MixerGetLevel Mixer Opcodes MixerGetLevel MixerGetLevel Gets the level of a send to a buss. Syntax kgain MixerGetLevel isend, ibuss Description Gets the level at which signals from the send are being added to the buss. The actual sending of the signal to the buss is performed by the MixerSend opcode. Initialization isend -- The number of the send, for example the number of the instrument sending the signal. ibuss -- The number of the buss, for example the number of the instrument receiving the signal. Performance kgain -- The level (any real number) at which the signal from the send will be mixed onto the buss. This opcode reports the level set by MixerSetLevel for a send and buss pair. Use of the mixer requires that instruments setting gains have smaller numbers than instruments sending signals, and that instruments sending signals have smaller numbers than instruments receiving those signals. However, an instrument may have any number of sends or receives. After the final signal is received, MixerClear must be invoked to reset the busses to 0 before the next kperiod. Examples Here is an example of the MixerGetLevel opcode. It uses the file Mixer.csd Example of the MixerGetLevel opcode. Credits Michael Gogins (gogins at pipeline dot com). manual_src~dfsg/opcodes/xscans.xml0000644000000000000000000001117512262561504016427 0ustar rootroot xscans Signal Generators:Scanned Synthesis xscans xscans Fast scanned synthesis waveform and the wavetable generator. Description Experimental version of scans. Allows much larger matrices and is faster and smaller but removes some (unused?) flexibility. If liked, it will replace the older opcode as it is syntax compatible but extended. Syntax ares xscans kamp, kfreq, ifntraj, id [, iorder] Initialization ifntraj -- table containing the scanning trajectory. This is a series of numbers that contains addresses of masses. The order of these addresses is used as the scan path. It should not contain values greater than the number of masses, or negative numbers. See the introduction to the scanned synthesis section. id -- If positive, the ID of the opcode. This will be used to point the scanning opcode to the proper waveform maker. If this value is negative, the absolute of this value is the wavetable on which to write the waveshape. That wavetable can be used later from an other opcode to generate sound. The initial contents of this table will be destroyed. iorder (optional, default=0) -- order of interpolation used internally. It can take any value in the range 1 to 4, and defaults to 4, which is quartic interpolation. The setting of 2 is quadratic and 1 is linear. The higher numbers are slower, but not necessarily better. Performance kamp -- output amplitude. Note that the resulting amplitude is also dependent on instantaneous value in the wavetable. This number is effectively the scaling factor of the wavetable. kfreq -- frequency of the scan rate Matrix Format The new matrix format is a list of connections, one per line linking point x to point y. There is no weight given to the link; it is assumed to be unity. The list is preceded by the line <MATRIX> and ends with a </MATRIX> line For example, a circular string of 8 would be coded as <MATRIX> 0 1 1 0 1 2 2 1 2 3 3 2 3 4 4 3 4 5 5 4 5 6 6 5 6 7 7 6 0 7 </MATRIX> Examples Here is an example of the xscans opcode. It uses the file xscans.csd. Example of the xscans opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. For similar examples, see the documentation on scans. See Also More information on Scanned Synthesis (as well as several other matrices) is available on the Scanned Synthesis page at cSounds.com. Also an article on these opcodes: http://www.csounds.com/stevenyi/scanned/yi_scannedSynthesis.html , written by Steven Yi scans, xscanu Credits Written by &namejohn;. New in version 4.20 manual_src~dfsg/opcodes/atonex.xml0000644000000000000000000000601712262561504016425 0ustar rootroot atonex Signal Modifiers:Standard Filters atonex atonex Emulates a stack of filters using the atone opcode. Description atonex is equivalent to a filter consisting of more layers of atone with the same arguments, serially connected. Using a stack of a larger number of filters allows a sharper cutoff. They are faster than using a larger number instances in a Csound orchestra of the old opcodes, because only one initialization and k- cycle are needed at time and the audio loop falls entirely inside the cache memory of processor. Syntax ares atonex asig, khp [, inumlayer] [, iskip] ares atonex asig, ahp [, inumlayer] [, iskip] Initialization inumlayer (optional) -- number of elements in the filter stack. Default value is 4. iskip (optional, default=0) -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance asig -- input signal khp/ahp -- the response curve's half-power point. Half power is defined as peak power / root 2. Examples Here is an example of the atonex opcode. It uses the file atonex.csd. Example of the atonex opcode. See Also resonx, tonex Credits Author: &namegabriel; (adapted by &namejohn;) Italy New in Csound version 3.49 Audio rate parameters introduced in version 6.02 October 2013. manual_src~dfsg/opcodes/ftlptim.xml0000644000000000000000000000542212262561504016605 0ustar rootroot ftlptim Table Control:Table Queries ftlptim ftlptim Returns the loop segment start-time of a stored function table number. Description Returns the loop segment start-time of a stored function table number. Syntax ftlptim(x) (init-rate args only) Performance Returns the loop segment start-time (in seconds) of stored function table number x. This reports the duration of the direct recorded attack and decay parts of a sound sample, prior to its looped segment. Returns zero (and a warning message) if the sample does not contain loop points. Examples Here is an example of the ftlptim opcode. It uses the files ftlptim.csd, and Church.wav. Example of the ftlptim opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The audio file Church.wav is a looped sample, its output should include lines like these: Base Note : 60 Detune : 0 Low Note : 0 High Note : 127 Low Vel. : 0 High Vel. : 127 Gain : 1 Count : 1 mode : 801 start : 4529 end : 4912 count :0 mode : 0 start : 0 end : 0 count :0 See Also ftchnls, ftlen, ftsr, nsamp Credits Author: &namebarry; MIT Cambridge, Massachussetts 1997 manual_src~dfsg/opcodes/kr.xml0000644000000000000000000000577012262561504015550 0ustar rootroot kr Orchestra Syntax:Header kr kr Sets the control rate. Description These statements are global value assignments, made at the beginning of an orchestra, before any instrument block is defined. Their function is to set certain reserved symbol variables that are required for performance. Once set, these reserved symbols can be used in expressions anywhere in the orchestra. Syntax kr = iarg Initialization kr = (optional) -- set control rate to iarg samples per second. The default value is 4410. In addition, any global variable can be initialized by an init-time assignment anywhere before the first instr statement. All of the above assignments are run as instrument 0 (i-pass only) at the start of real performance. Beginning with Csound version 3.46, kr can be omitted. Csound will use the default values, or calculate kr from defined ksmps and sr. It is usually better to just specify ksmps and sr and let csound calculate kr. Examples sr = 10000 kr = 500 ksmps = 20 gi1 = sr/2. ga init 0 itranspose = octpch(.0l) See Also ksmps, nchnls, nchnls_i, sr manual_src~dfsg/opcodes/midic14.xml0000644000000000000000000000720112262561504016355 0ustar rootroot midic14 Real-time MIDI:Input midic14 midic14 Allows a floating-point 14-bit MIDI signal scaled with a minimum and a maximum range. Description Allows a floating-point 14-bit MIDI signal scaled with a minimum and a maximum range. Syntax idest midic14 ictlno1, ictlno2, imin, imax [, ifn] kdest midic14 ictlno1, ictlno2, kmin, kmax [, ifn] Initialization idest -- output signal ictln1o -- most-significant byte controller number (0-127) ictlno2 -- least-significant byte controller number (0-127) imin -- user-defined minimum floating-point value of output imax -- user-defined maximum floating-point value of output ifn (optional) -- table to be read when indexing is required. Table must be normalized. Output is scaled according to imin and imax values. Performance kdest -- output signal kmin -- user-defined minimum floating-point value of output kmax -- user-defined maximum floating-point value of output midic14 (i- and k-rate 14 bit MIDI control) allows a floating-point 14-bit MIDI signal scaled with a minimum and a maximum range. The minimum and maximum values can be varied at k-rate. It can use optional interpolated table indexing. It requires two MIDI controllers as input. Please note that the midic family of opcodes are designed for MIDI triggered events, and don't require a channel number since they will respond to the same channel as the one that triggered the instrument (see massign). However they will crash if called from a score driven event. See Also ctrl7, ctrl14, ctrl21, initc7, initc14, initc21, midic7, midic21 Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct controller number range. manual_src~dfsg/opcodes/outic.xml0000644000000000000000000000760012262561504016251 0ustar rootroot outic Real-time MIDI:Output outic outic Sends MIDI controller output at i-rate. Description Sends MIDI controller output at i-rate. Syntax outic ichn, inum, ivalue, imin, imax Initialization ichn -- MIDI channel number (1-16) inum -- controller number (0-127 for example 1 = ModWheel; 2 = BreathControl etc.) ivalue -- floating point value imin -- minimum floating point value (converted in MIDI integer value 0) imax -- maximum floating point value (converted in MIDI integer value 127 (7 bit)) Performance outic (i-rate MIDI controller output) sends controller messages to the MIDI OUT device. It works only with MIDI instruments which recognize them. It can drive a different value of a parameter for each note currently active. It can scale an i-value floating-point argument according to the imin and imax values. For example, set imin = 1.0 and imax = 2.0. When the ivalue argument receives a 2.0 value, the opcode will send a 127 value to the MIDI OUT device. When the ivalue argument receives a 1.0 value, it will send a 0 value. i-rate opcodes send their message once during instrument initialization. Examples Here is an example of the outic opcode. It uses the file outic.csd. Example of the outic opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also outiat, outic14, outipat, outipb, outipc, outkat, outkc14, outkc, outkpat, outkpb, outkpc Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/splitrig.xml0000644000000000000000000001001212262561504016752 0ustar rootroot splitrig Instrument Control:Sensing and Control splitrig splitrig Split a trigger signal Description splitrig splits a trigger signal (i.e. a timed sequence of control-rate impulses) into several channels following a structure designed by the user. Syntax splitrig ktrig, kndx, imaxtics, ifn, kout1 [,kout2,...,koutN] Initialization imaxtics - number of tics belonging to largest pattern ifn - number of table containing channel-data structuring Performance asig - incoming (input) signal ktrig - trigger signal The splitrig opcode splits a trigger signal into several output channels according to one or more patterns provided by the user. Normally the regular timed trigger signal generated by metro opcode is used to be transformed into rhythmic pattern that can trig several independent melodies or percussion riffs. But you can also start from non-isocronous trigger signals. This allows to use some "interpretative" and less "mechanic" groove variations. Patterns are looped and each numtics_of_pattern_N the cycle is repeated. The scheme of patterns is defined by the user and is stored into ifn table according to the following format: This scheme can contain more than one pattern, each one with a different number of rows. Each pattern is preceded by a a special row containing a single numtics_of_pattern_N field; this field expresses the number of tics that makes up the corresponding pattern. Each pattern's row makes up a tic. Each pattern's column corresponds to a channel, and each field of a row is a number that makes up the value outputted by the corresponding koutXX channel (if number is a zero, corresponding output channel will not trigger anything in that particular arguments). Obviously, all rows must contain the same number of fields that must be equal to the number of koutXX channel. All patterns must contain the same number of rows, this number must be equal to the largest pattern and is defined by imaxtics variable. Even if a pattern has less tics than the largest pattern, it must be made up of the same number of rows, in this case, some of these rows, at the end of the pattern itself, will not be used (and can be set to any value, because it doesn't matter). The kndx variable chooses the number of the pattern to be played, zero indicating the first pattern. Each time the integer part of kndx changes, tic counter is reset to zero. Patterns are looped and each numtics_of_pattern_N the cycle is repeated. examples 4 - calculate average value of asig in the time interval This opcode can be useful in several situations, for example to implement a vu-meter Credits Written by &namegabriel;. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/nlfilt2.xml0000644000000000000000000000755612262561504016512 0ustar rootroot nlfilt2 Signal Modifiers:Specialized Filters nlfilt2 nlfilt2 A filter with a non-linear effect and blowup protection. Description Implements the filter: Y{n} =tanh(a Y{n-1} + b Y{n-2} + d Y^2{n-L} + X{n} - C) described in Dobson and Fitch (ICMC'96) as modified by Risto Holopainen. Syntax ares nlfilt2 ain, ka, kb, kd, kC, kL Performance Non-linear effect. The range of parameters are: a = b = 0 d = 0.8, 0.9, 0.7 C = 0.4, 0.5, 0.6 L = 20 This affects the lower register most but there are audible effects over the whole range. We suggest that it may be useful for coloring drums, and for adding arbitrary highlights to notes. Low Pass with non-linear. The range of parameters are: a = 0.4 b = 0.2 d = 0.7 C = 0.11 L = 20, ... 200 There are instability problems with this variant but the effect is more pronounced of the lower register, but is otherwise much like the pure comb. Short values of L can add attack to a sound. High Pass with non-linear. The range of parameters are: a = 0.35 b = -0.3 d = 0.95 C = 0,2, ... 0.4 L = 200 High Pass with non-linear. The range of parameters are: a = 0.7 b = -0.2, ... 0.5 d = 0.9 C = 0.12, ... 0.24 L = 500, 10 The high pass version is less likely to oscillate. It adds scintillation to medium-high registers. With a large delay L it is a little like a reverberation, while with small values there appear to be formant-like regions. There are arbitrary color changes and resonances as the pitch changes. Works well with individual notes. Warning The "useful" ranges of parameters are not yet mapped. Examples Here is an example of the nlfilt2 opcode. It uses the file nlfilt2.csd. Example of the nlfilt2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK 2012 New in version 5.19 manual_src~dfsg/opcodes/maxabs.xml0000644000000000000000000000616212262561504016403 0ustar rootroot maxabs Signal Modifiers:Comparators and Accumulators maxabs maxabs Produces a signal that is the maximum of the absolute values of any number of input signals. Description The maxabs opcode takes any number of a-rate or k-rate signals as input (all of the same rate), and outputs a signal at the same rate that is the maximum of all of the inputs. It is identical to the max opcode except that it takes the absolute value of each input before comparing them. Therefore, the output is always non-negative. For a-rate signals, the inputs are compared one sample at a time (i.e. maxabs does not scan an entire ksmps period of a signal for its local maximum as the max_k opcode does). Syntax amax maxabs ain1, ain2 [, ain3] [, ain4] [...] kmax maxabs kin1, kin2 [, kin3] [, kin4] [...] Performance ain1, ain2, ... -- a-rate signals to be compared. kin1, kin2, ... -- k-rate signals to be compared. Examples Here is an example of the maxabs opcode. It uses the file maxabs.csd. Example of the maxabs opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also minabs, max, min, maxaccum, minaccum, maxabsaccum, minabsaccum, max_k Credits Author: &nameanthony; March 2006 New in Csound version 5.01 manual_src~dfsg/opcodes/lpsholdp.xml0000644000000000000000000000425012262561504016751 0ustar rootroot lpsholdp Signal Generators:Linear and Exponential Generators lpsholdp lpsholdp Control signals based on held segments. Description Generate control signal consisiting of held segments delimited by two or more specified points. The entire envelope can be looped at time-variant rate. Each segment coordinate can also be varied at k-rate. Syntax ksig lpsholdp kphase, kvalue0, ktime0 [, kvalue1] [, ktime1] \ [, kvalue2] [, ktime2] [...] Performance ksig - output signal kphase -- point of the sequence read, expressed as a fraction of a cycle (0 to 1) kvalue0...kvalueN -- Values of points ktime0...ktimeN -- Times between points; expressed in fractions of a cycle (see below). The final time designates a ramp between the final value and the first value. lpsholdp opcode is similar to lpshold; the only difference is that, instead of frequency, a time-variant phase is required. If you use a phasor to get the phase value, you will have a behaviour identical to lpshold, but interesting results can be achieved when using phases having non-linear motions, making lpsholdp more powerful and general than lpshold. Credits Written by &namegabriel;. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/strcat.xml0000644000000000000000000000336612262561504016433 0ustar rootroot strcat Strings:Manipulation strcat strcat Concatenate strings Description Concatenate two strings and store the result in a variable. strcat runs at i-time only. It is allowed for any of the input arguments to be the same as the output variable. Syntax Sdst strcat Ssrc1, Ssrc2 Examples Here is an example of the strcat opcode. It uses the file strcat.csd. Example of the strcat opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also strcatk Credits Author: Istvan Varga 2005 New in version 5.02 manual_src~dfsg/opcodes/outkpb.xml0000644000000000000000000000732212262561504016433 0ustar rootroot outkpb Real-time MIDI:Output outkpb outkpb Sends MIDI pitch-bend messages at k-rate. Description Sends MIDI pitch-bend messages at k-rate. Syntax outkpb kchn, kvalue, kmin, kmax Performance kchn -- MIDI channel number (1-16) kvalue -- floating point value kmin -- minimum floating point value (converted in MIDI integer value 0) kmax -- maximum floating point value (converted in MIDI integer value 127 (7 bit)) outkpb (k-rate pitch-bend output) sends pitch-bend messages. It works only with MIDI instruments which recognize them. It can drive a different value of a parameter for each note currently active. It can scale the k-value floating-point argument according to the kmin and kmax values. For example: set kmin = 1.0 and kmax = 2.0. When the kvalue argument receives a 2.0 value, the opcode will send a 127 value to the MIDI OUT device. When the kvalue argument receives a 1.0 value, it will send a 0 value. k-rate opcodes send a message each time the MIDI converted value of argument kvalue changes. Examples Here is an example of the outkpb opcode. It uses the file outkpb.csd. Example of the outkpb opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also outiat, outic14, outic, outipat, outipb, outipc, outkat, outkc14, outkc, outkpat, outkpc Credits Author: &namegabriel; Italy New in Csound version 3.47 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/lpslot.xml0000644000000000000000000000552112262561504016443 0ustar rootroot lpslot Spectral Processing:LPC lpslot lpslot Selects the slot to be use by further lp opcodes. Description Selects the slot to be use by further lp opcodes. Syntax lpslot islot Initialization islot -- number of slot to be selected. Performance lpslot selects the slot to be use by further lp opcodes. This is the way to load and reference several analyses at the same time. Examples Here is a typical orc using the opcodes: ipower init 50000 ; Define sound generator ifreq init 440 asrc buzz ipower, ifreq, 10, 1 ktime line 0, p3, p3 ; Define time lin lpslot 0 ; Read square data poles krmsr, krmso, kerr, kcps lpread ktime,"square.pol" lpslot 1 ; Read triangle data poles krmsr, krmso, kerr, kcps lpread ktime,"triangle.pol" kmix line 0, p3, 1 ; Compute result of mixing lpinterp 0, 1, kmix ; and balance power ares lpreson asrc aout balance ares, asrc out aout See Also lpinterp Credits Author: Mark Resibois Brussels 1996 New in version 3.44 manual_src~dfsg/opcodes/cpsxpch.xml0000644000000000000000000001172112262561504016575 0ustar rootroot cpsxpch Pitch Converters:Tuning Opcodes cpsxpch cpsxpch Converts a pitch-class value into cycles-per-second (Hz) for equal divisions of any interval. Description Converts a pitch-class value into cycles-per-second (Hz) for equal divisions of any interval. There is a restriction of no more than 100 equal divisions. Syntax icps cpsxpch ipch, iequal, irepeat, ibase Initialization ipch -- Input number of the form 8ve.pc, indicating an 'octave' and which note in the octave. iequal -- if positive, the number of equal intervals into which the 'octave' is divided. Must be less than or equal to 100. If negative, is the number of a table of frequency multipliers. irepeat -- Number indicating the interval which is the 'octave.' The integer 2 corresponds to octave divisions, 3 to a twelfth, 4 is two octaves, and so on. This need not be an integer, but must be positive. ibase -- The frequency which corresponds to pitch 0.0 Note The following are essentially the same ia = cpspch(8.02) ib cps2pch 8.02, 12 ic cpsxpch 8.02, 12, 2, 1.02197503906 These are opcodes not functions Negative values of ipch are allowed, but not negative irepeat, iequal or ibase. Examples Here is an example of the cpsxpch opcode. It uses the file cpsxpch.csd. Example of the cpsxpch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: icps = 293.666 Here is an example of the cpsxpch opcode using a 10.5 ET scale. It uses the file cpsxpch_105et.csd. Example of the cpsxpch opcode using a 10.5 ET scale. Its output should include lines like this: instr 1: icps = 4776.824 Here is an example of the cpsxpch opcode using a Pierce scale centered on middle A. It uses the file cpsxpch_pierce.csd. Example of the cpsxpch opcode using a Pierce scale centered on middle A. Its output should include lines like this: instr 1: icps = 2827.762 See Also cpspch, cps2pch Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK 1997 New in Csound version 3.492 manual_src~dfsg/opcodes/dumpk4.xml0000644000000000000000000001206012262561504016326 0ustar rootroot dumpk4 Signal I/O:File I/O dumpk4 dumpk4 Periodically writes four orchestra control-signal values to an external file. Description Periodically writes four orchestra control-signal values to a named external file in a specific format. Syntax dumpk4 ksig1, ksig2, ksig3, ksig4, ifilname, iformat, iprd Initialization ifilname -- character string (in double quotes, spaces permitted) denoting the external file name. May either be a full path name with target directory specified or a simple filename to be created within the current directory iformat -- specifies the output data format: 1 = 8-bit signed char(high order 8 bits of a 16-bit integer 4 = 16-bit short integers 5 = 32-bit long integers 6 = 32-bit floats 7 = ASCII long integers 8 = ASCII floats (2 decimal places) Note that A-law and U-law output are not available, and that all formats except the last two are binary. The output file contains no header information. iprd -- the period of ksig output in seconds, rounded to the nearest orchestra control period. A value of 0 implies one control period (the enforced minimum), which will create an output file sampled at the orchestra control rate. Performance ksig1, ksig2, ksig3, ksig4 -- control-rate signals This opcode allows four generated control signal values to be saved in a named external file. The file contains no self-defining header information. But it contains a regularly sampled time series, suitable for later input or analysis. There may be any number of dumpk4 opcodes in an instrument or orchestra but each must write to a different file. Examples Here is an example of the dumpk4 opcode. It uses the file dumpk4.csd. Example of the dumpk4 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output should include lines like these: WRITING: kcf = 1122.469723, kq = 11.762839, kdb = -14.313445, kpan = 0.538142 kcf = 1122.469723, kq = 11.762839, kdb = -14.313445, kpan = 0.538142 kcf = 1148.638412, kq = 12.040490, kdb = -14.061868, kpan = 0.552205 kcf = 165.796855, kq = 18.523179, kdb = -15.816977, kpan = 0.901528 kcf = 147.729960, kq = 13.071911, kdb = -11.924531, kpan = 0.982518 kcf = 497.430113, kq = 13.605512, kdb = -21.586611, kpan = 0.179229 READING: WARNING: Seeding from current time 3308160476 kcf = 1122.469700, kq = 11.762800, kdb = -14.313400, kpan = 0.538100 kcf = 1122.469700, kq = 11.762800, kdb = -14.313400, kpan = 0.538100 kcf = 1148.638400, kq = 12.040500, kdb = -14.061900, kpan = 0.552200 kcf = 165.796900, kq = 18.523200, kdb = -15.817000, kpan = 0.901500 kcf = 147.730000, kq = 13.071900, kdb = -11.924500, kpan = 0.982500 kcf = 497.430100, kq = 13.605500, kdb = -21.586600, kpan = 0.179200 See Also dumpk, dumpk2, dumpk3, readk, readk2, readk3, readk4 Credits By: John ffitch and Barry Vercoe 1999 or earlier manual_src~dfsg/opcodes/flsetsize.xml0000644000000000000000000000557512262561504017147 0ustar rootroot FLsetSize FLTK:Appearance FLsetSize FLsetSize Resizes a FLTK widget. Description FLsetSize resizes the target widget (not the size of its text) according to the iwidth and iheight arguments. Syntax FLsetSize iwidth, iheight, ihandle Initialization iwidth -- width of widget. iheight -- height of widget. ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/unirand.xml0000644000000000000000000001063612262561504016571 0ustar rootroot x-class noise generators unirand Signal Generators:Random (Noise) Generators unirand unirand Uniform distribution random number generator (positive values only). Description Uniform distribution random number generator (positive values only). This is an x-class noise generator. Syntax ares unirand krange ires unirand krange kres unirand krange Performance krange -- the range of the random numbers (0 - krange). For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the unirand opcode. It uses the file unirand.csd. Example of the unirand opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i 1 time 0.00067: 81.47237 i 1 time 0.20067: 41.72671 i 1 time 0.40067: 5.96189 i 1 time 0.60067: 91.59912 i 1 time 0.80067: 85.07127 i 1 time 1.00000: 92.50948 i 1 time 1.20067: 98.79347 i 1 time 1.40067: 98.91449 i 1 time 1.60067: 50.37808 i 1 time 1.80000: 72.02497 i 1 time 2.00000: 52.94362 WARNING: Seeding from current time 4007444022 i 2 time 3.00067: 91.86294 i 2 time 3.20067: 94.68759 i 2 time 3.40067: 1.05825 i 2 time 3.60000: 78.57628 i 2 time 3.80067: 27.67408 i 2 time 4.00000: 76.46347 i 2 time 4.20000: 77.10071 i 2 time 4.40067: 34.28921 i 2 time 4.60067: 37.72286 i 2 time 4.80067: 54.96646 i 2 time 5.00000: 11.67566 B 3.000 .. 5.000 T 5.000 TT 5.000 M: 0.80000 0.80000 Score finished in csoundPerform(). See Also seed, betarand, bexprnd, cauchy, exprand, gauss, linrand, pcauchy, poisson, trirand, weibull Credits Author: &nameparis; MIT, Cambridge 1995 manual_src~dfsg/opcodes/pvsftr.xml0000644000000000000000000000774512262561504016464 0ustar rootroot pvsftr Spectral Processing:Streaming pvsftr pvsftr Reads amplitude and/or frequency data from function tables. Description Reads amplitude and/or frequency data from function tables. Syntax pvsftr fsrc, ifna [, ifnf] Initialization ifna -- A table, at least inbins in size, that stores amplitude data. Ignored if ifna = 0 ifnf (optional) -- A table, at least inbins in size, that stores frequency data. Ignored if ifnf = 0 Performance fsrc -- a PVOC-EX formatted source. Enables the contents of fsrc to be exchanged with function tables for custom processing. Except when the frame overlap equals ksmps (which will generally not be the case), the frame data is not updated each control period. The data in ifna, ifnf should only be processed when kflag is set to 1. To process only frequency data, set ifna to zero. As the function tables are required only to store data from fsrc, there is no advantage in defining then in the score, and they should generally be created in the instrument, using ftgen. By exporting amplitude data, say, from one fsig and importing it into another, basic cross-synthesis (as in pvscross) can be performed, with the option to modify the data beforehand using the table manipulation opodes. Note that the format data in the source fsig is not written to the tables. This therefore offers a means of transferring amplitude and frequency data between non-identical fsigs. Used this way, these opcodes become potentially pathological, and can be relied upon to produce unexpected results. In such cases, resynthesis using pvsadsyn would almost certainly be required. To perform a straight copy from one fsig to another one of identical format, the conventional assignment syntax can be used: fsig1 = fsig2 It is not necessary to use function tables in this case. Examples Here is an example of the pvsftr opcode. It uses the file pvsftr.csd. Example of the pvsftr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsftw Credits Author: &namerichard; August 2001 New in version 4.13 manual_src~dfsg/opcodes/random.xml0000644000000000000000000000620512262561504016406 0ustar rootroot random Signal Generators:Random (Noise) Generators random random Generates a controlled pseudo-random number series between min and max values. Description Generates is a controlled pseudo-random number series between min and max values. Syntax ares random kmin, kmax ires random imin, imax kres random kmin, kmax Initialization imin -- minimum range limit imax -- maximum range limit Performance kmin -- minimum range limit kmax -- maximum range limit The random opcode is similar to linrand and trirand but sometimes I [&namegabriel;] find it more convenient because allows the user to set arbitrary minimum and maximum values. Examples Here is an example of the random opcode. It uses the file random.csd. Example of the random opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i 1 time 0.00067: 894.58566 i 1 time 0.50000: 748.44281 i 1 time 1.00000: 328.29916 WARNING: Seeding from current time 1656666052 i 2 time 2.00067: 690.71466 i 2 time 2.50000: 459.42445 i 2 time 3.00000: 100.85594 See Also linrand, randomh, randomi, trirand Credits Author: &namegabriel; manual_src~dfsg/opcodes/vmultv.xml0000644000000000000000000001413012262561504016457 0ustar rootroot vmultv Vectorial:Vectorial operations vmultv vmultv Performs mutiplication between two vectorial control signals Description Performs mutiplication between two vectorial control signals Syntax vmultv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] Initialization ifn1 - number of the table hosting the first vector to be processed ifn2 - number of the table hosting the second vector to be processed Performance kelements - number of elements of the two vectors kdstoffset - index offset for the destination (ifn1) table (Default=0) ksrcoffset - index offset for the source (ifn2) table (Default=0) kverbose - Selects whether or not warnings are printed (Default=0) vmultv multiplies two vectorial control signals, that is, each element of the first vector is processed (only) with the corresponding element of the other vector. Each vectorial signal is hosted by a table (ifn1 and ifn2). The number of elements contained in both vectors must be the same. The Result is a new vectorial control signal that overrides old values of ifn1. If you want to keep the old ifn1 vector, use vcopy_i opcode to copy it in another table. You can use kdstoffset and ksrcoffset to specify vectors in any location of the tables. Negative values for kdstoffset and ksrcoffset are acceptable. If kdstoffset is negative, the out of range section of the vector will be discarded. If ksrcoffset is negative, the out of range elements will be assumed to be 1 (i.e. the destination elements will not be changed). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 1 (i.e. the destination vector will not be changed for these elements). If the optional kverbose argument is different to 0, the opcode will print warning messages every k-pass if table lengths are exceeded. Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at k-rate (this means that every k-pass the vectors are multiplied). There's an i-rate version of this opcode called vmultv_i. Please note that the elements argument has changed in version 5.03 from i-rate to k-rate. This will change the opcode's behavior in the unusual cases where the i-rate variable ielements is changed inside the instrument, for example in: instr 1 ielements = 10 vadd 1, 1, ielements ielements = 20 vadd 2, 1, ielements turnoff endin All these operators (vaddv, vsubv, vmultv, vdivv, vpowv, vexpv, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Examples Here is an example of the vmultv opcode. It uses the file vmultv.csd. Example of the vmultv opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/ATSreadnz.xml0000644000000000000000000001345112262561504016762 0ustar rootroot ATSreadnz Spectral Processing:ATS ATSreadnz ATSreadnz reads data from an ATS file. Description ATSreadnz returns the energy (kenergy) of a user specified noise band (1-25 bands) at the time indicated by the time pointer ktimepnt. Syntax kenergy ATSreadnz ktimepnt, iatsfile, iband Initialization iatsfile – the ATS number (n in ats.n) or the name in quotes of the analysis file made using ATS. iband – the number of the noise band to return the energy data. Performance kenergy outputs the linearly interpolated energy of the noise band indicated in iband. The output is dependent on the data in the analysis file and the ktimepnt. ktimepnt – The time pointer in seconds used to index the ATS file. Used for ATSreadnz exactly the same as for pvoc and ATSadd. ATSaddnz reads from an ATS file and resynthesizes the noise from noise energy data contained in the ATS file. It uses a modified randi function to create band limited noise and modulates that with a user supplied wave table (one period of a cosine wave), to synthesize a user specified selection of frequency bands. Modulating the noise is required to put the band limited noise in the correct place in the frequency spectrum. An ATS analysis differs from a pvanal in that ATS tracks the partials and computes the noise energy of the sound being analyzed. For more info about ATS analysis read Juan Pampin's description on the the ATS web-page. Examples ktime line 2.5, p3, 0 kenergy ATSreadnz ktime, "clarinet.ats", 5 Here we are extracting the noise energy from band 5 in the 'clarinet.ats' ATS analysis file. We're actually reading backwards from 2.5 seconds to the beginning of the analysis file. We could use this to synthesize noise like this: anoise randi sqrt(kenergy), 55 aout oscili 4000000000000000000000000, 455, 2 aout = aout * anoise Function table 2 used in the oscillator is a cosine, which is needed to shift the band limited noise into the correct place in the frequency spectrum. The randi function creates a band of noise centered about 0 Hz that has a bandwidth of about 110 Hz; multiplying it by a cosine will shift it to be centered at 455 Hz, which is the center frequency of the 5th critical noise band. This is only an example, for synthesizing the noise you'd be better off just using ATSaddnz unless you want to use your own noise synthesis algorithm. Maybe you could use the noise energy for something else like applying a small amount of jitter to specific partials or for controlling something totally unrelated to the source sound? Examples Here is another example of the ATSreadnz opcode. It uses the file ATSreadnz.csd. Another example of the ATSreadnz opcode. See also ATSread, ATSinfo, ATSbufread, ATScross, ATSinterpread, ATSpartialtap, ATSadd, ATSaddnz, ATSsinnoi Credits Author: Alex Norman Seattle,Washington 2004 manual_src~dfsg/opcodes/midion2.xml0000644000000000000000000000573012262561504016471 0ustar rootroot midion2 Real-time MIDI:Note Output midion2 midion2 Sends noteon and noteoff messages to the MIDI OUT port. Description Sends noteon and noteoff messages to the MIDI OUT port when triggered by a value different than zero. Syntax midion2 kchn, knum, kvel, ktrig Performance kchn -- MIDI channel (1-16) knum -- MIDI note number (0-127) kvel -- note velocity (0-127) ktrig -- trigger input signal (normally 0) Similar to midion, this opcode sends noteon and noteoff messages to the MIDI out port, but only when ktrig is non-zero. This opcode is can work together with the output of the trigger opcode. Examples Here is an example of the midion2 opcode. It uses the file midion2.csd. Example of the midion2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also moscil, midion, noteon, noteoff, noteondur, noteondur2 Credits Author: &namegabriel; Italy 1998 New in Csound version 3.492 Thanks goes to &namerasmus; for pointing out the correct MIDI channel and controller number ranges. manual_src~dfsg/opcodes/display.xml0000644000000000000000000000627412262561504016601 0ustar rootroot display Signal I/O:Printing and Display display display Displays the audio or control signals as an amplitude vs. time graph. Description These units will print orchestra init-values, or produce graphic display of orchestra control signals and audio signals. Uses X11 windows if enabled, else (or if -g flag is set) displays are approximated in ASCII characters. Syntax display xsig, iprd [, inprds] [, iwtflg] Initialization iprd -- the period of display in seconds. inprds (optional, default=1) -- Number of display periods retained in each display graph. A value of 2 or more will provide a larger perspective of the signal motion. The default value is 1 (each graph completely new). inprds is a scaling factor for the displayed waveform, controlling how many iprd-sized frames of samples are drawn in the window (the default and minimum value is 1.0). Higher inprds values are slower to draw (more points to draw) but will show the waveform scrolling through the window, which is useful with low iprd values. iwtflg (optional, default=0) -- wait flag. If non-zero, each display is held until released by the user. The default value is 0 (no wait). Performance display -- displays the audio or control signal xsig every iprd seconds, as an amplitude vs. time graph. Examples Here is an example of the display opcode. It uses the file display.csd. Example of the display opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also dispfft, print Credits Comments about the inprds parameter contributed by &namerasmus;. manual_src~dfsg/opcodes/init.xml0000644000000000000000000001073212262561504016071 0ustar rootroot init Instrument Control:Initialization and Reinitialization init init Puts the value of the i-time expression into a k-, a-rate or t- variable. Syntax ares init iarg ires init iarg kres init iarg ares, ... init iarg, ... ires, ... init iarg, ... kres, ... init iarg, ... tab init isize[, ival] Description Put the value of the i-time expression into a k- or a-rate variable. Initialization Puts the value of the i-time expression iarg into a k-, a-rate or t- variable, i.e., initialize the result. Note that init provides the only case of an init-time statement being permitted to write into a perf-time (k- or a-rate) result cell; the statement has no effect at perf-time. Since version 5.13 it is possible to initialise upto 24 variables of the same class in one statement. If there are more output variables than input expressions then the last one is repeated. It is an error to have more inputs than outputs. The t-variable form was introduced in 5.14 and allocated space for a vector or the given size, initialised to the given value (default value is zero). Examples Here is an example of the init opcode. It uses the file init.csd. Example of the init opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i 1 time 0.00073: 1.00000 i 1 time 0.10014: 138.00000 i 1 time 0.20027: 276.00000 i 1 time 0.30041: 414.00000 i 1 time 0.40054: 552.00000 i 1 time 0.50068: 690.00000 i 1 time 0.60009: 827.00000 i 1 time 0.70023: 965.00000 i 1 time 0.80036: 1103.00000 i 1 time 0.90050: 1241.00000 i 2 time 2.00054: 1.00000 i 2 time 2.09995: 1.00000 i 2 time 2.20009: 1.00000 i 2 time 2.30023: 1.00000 i 2 time 2.40036: 1.00000 i 2 time 2.50050: 1.00000 i 2 time 2.59991: 1.00000 i 2 time 2.70005: 1.00000 i 2 time 2.80018: 1.00000 i 2 time 2.90032: 1.00000 See Also =, divz, tival Other information on this opcode in the Floss Manuals: http://www.flossmanuals.net/csound/ch016_a-initialization-and-performance-pass Credits Init first appeared in the original Csound, but the extension to multiple values is by Author: &namejohn; University of Bath, and Codemist Ltd. Bath, UK February 2010 Multiple form new in version 5.13; t-variable form new in 5.14. manual_src~dfsg/opcodes/loopxseg.xml0000644000000000000000000001036312262561504016766 0ustar rootroot loopxseg Signal Generators:Linear and Exponential Generators loopxseg loopxseg Generate control signal consisting of exponential segments delimited by two or more specified points. Description Generate control signal consisting of exponential segments delimited by two or more specified points. The entire envelope is looped at kfreq rate. Each parameter can be varied at k-rate. Syntax ksig loopxseg kfreq, ktrig, iphase, ktime0, kvalue0 [, ktime1] [, kvalue1] \ [, ktime2] [, kvalue2] [...] Performance ksig -- Output signal. kfreq -- Repeat rate in Hz or fraction of Hz. ktrig -- If non-zero, retriggers the envelope from start (see trigger opcode), before the envelope cycle is completed. iphase -- A value between 0 and 1 to say where to start the loop. Zero, the commonest value, indicates the beginning. ktime0...ktimeN -- Times of points; expressed in fraction of a cycle. kvalue0...kvalueN -- Values of points loopxseg opcode is similar to expseg, but the entire envelope is looped at kfreq rate. Notice that times are not expressed in seconds but in fraction of a cycle. Actually each duration represent is proportional to the other, and the entire cycle duration is proportional to the sum of all duration values. The sum of all duration is then rescaled according to kfreq argument. For example, considering an envelope made up of 3 segments, each segment having 100 as duration value, their sum will be 300. This value represents the total duration of the envelope, and is actually divided into 3 equal parts, a part for each segment. Actually, the real envelope duration in seconds is determined by kfreq. Again, if the envelope is made up of 3 segments, but this time the first and last segments have a duration of 50, whereas the central segment has a duration of 100 again, their sum will be 200. This time 200 represent the total duration of the 3 segments, so the central segment will be twice as long as the other segments. All parameters can be varied at k-rate. Negative frequency values are allowed, reading the envelope backward. ktime0 should always be set to 0, except if the user wants some special effect. Examples Here is an example of the loopxseg opcode. It uses the file loopxseg.csd. Example of the loopxseg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also lpshold loopseg Credits Author: &namejohn; New in Version 5.12 manual_src~dfsg/opcodes/maxaccum.xml0000644000000000000000000000757712262561504016741 0ustar rootroot maxaccum Signal Modifiers:Comparators and Accumulators maxaccum maxaccum Accumulates the maximum value of audio signals. Description maxaccum compares two audio-rate variables and stores the maximum value between them into the first. Syntax maxaccum aAccumulator, aInput Performance aAccumulator -- audio variable to store the maximum value aInput -- signal that aAccumulator is compared to The maxaccum opcode is designed to accumulate the maximum value from among many audio signals that may be in different note instances, different channels, or otherwise cannot all be compared at once using the max opcode. Its semantics are similar to vincr since aAccumulator is used as both an input and an output variable, except that maxaccum keeps the maximum value instead of adding the signals together. maxaccum performs the following operation on each pair of samples: if (aInput > aAccumulator) aAccumulator = aInput aAccumulator will usually be a global audio variable. At the end of any given computation cycle (k-period), after its value is read and used in some way, the accumulator variable should usually be reset to zero (perhaps by using the clear opcode). Care must be taken however if aInput is negative at any point, in which case the accumulator should be initialized and reset to some large enough negative value that will always be less than the input signals to which it is compared. Examples Here is an example of the maxaccum opcode. It uses the file maxaccum.csd. Example of the maxabs opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also minaccum, maxabsaccum, minabsaccum, max, min, maxabs, minabs, vincr, clear Credits Author: &nameanthony; March 2006 New in Csound version 5.01 manual_src~dfsg/opcodes/pvread.xml0000644000000000000000000000772712262561504016421 0ustar rootroot pvread Spectral Processing:STFT pvread pvread Reads from a phase vocoder analysis file and returns the frequency and amplitude from a single analysis channel or bin. Description pvread reads from a pvoc file and returns the frequency and amplitude from a single analysis channel or bin. The returned values can be used anywhere else in the Csound instrument. For example, one can use them as arguments to an oscillator to synthesize a single component from an analyzed signal or a bank of pvreads can be used to resynthesize the analyzed sound using additive synthesis by passing the frequency and magnitude values to a bank of oscillators. Syntax kfreq, kamp pvread ktimpnt, ifile, ibin Initialization ifile -- the pvoc number (n in pvoc.n) or the name in quotes of the analysis file made using pvanal. (See pvoc.) ibin -- the number of the analysis channel from which to return frequency in Hz and magnitude. Performance kfreq, kamp -- outputs of the pvread unit. These values, retrieved from a phase vocoder analysis file, represent the values of frequency and amplitude from a single analysis channel specified in the ibin argument. Interpolation between analysis frames is performed at k-rate resolution and dependent of course upon the rate and direction of ktimpnt. ktimpnt -- the passage of time, in seconds, through this file. ktimpnt must always be positive, but can move forwards or backwards in time, be stationary or discontinuous, as a pointer into the analysis file. Examples The example below shows the use pvread to synthesize a single component at a time from a phase vocoder analysis file. It should be noted that the kfreq and kamp outputs can be used for any kind of synthesis, filtering, processing, and so on. Example of the pvread opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvbufread, pvcross, pvinterp, tableseg, tablexseg Credits Author: Richard Karpen Seattle, Wash 1997 New in version 3.44 manual_src~dfsg/opcodes/space.xml0000644000000000000000000002312512262561504016221 0ustar rootroot space Signal Modifiers:Panning and Spatialization space space Distributes an input signal among 4 channels using cartesian coordinates. Description space takes an input signal and distributes it among 4 channels using Cartesian xy coordinates to calculate the balance of the outputs. The xy coordinates can be defined in a separate text file and accessed through a Function statement in the score using Gen28, or they can be specified using the optional kx, ky arguments. The advantages to the former are: A graphic user interface can be used to draw and edit the trajectory through the Cartesian plane The file format is in the form time1 X1 Y1 time2 X2 Y2 time3 X3 Y3 allowing the user to define a time-tagged trajectory space then allows the user to specify a time pointer (much as is used for pvoc, lpread and some other units) to have detailed control over the final speed of movement. Syntax a1, a2, a3, a4 space asig, ifn, ktime, kreverbsend, kx, ky Initialization ifn -- number of the stored function created using Gen28. This function generator reads a text file which contains sets of three values representing the xy coordinates and a time-tag for when the signal should be placed at that location. The file should look like: 0 -1 1 1 1 1 2 4 4 2.1 -4 -4 3 10 -10 5 -40 0 If that file were named move then the Gen28 call in the score would like: f1 0 0 28 "move" Gen28 takes 0 as the size and automatically allocates memory. It creates values to 10 milliseconds of resolution. So in this case there will be 500 values created by interpolating X1 to X2 to X3 and so on, and Y1 to Y2 to Y3 and so on, over the appropriate number of values that are stored in the function table. In the above example, the sound will begin in the left front, over 1 second it will move to the right front, over another second it move further into the distance but still in the right front, then in just 1/10th of a second it moves to the left rear, a bit distant. Finally over the last .9 seconds the sound will move to the right rear, moderately distant, and it comes to rest between the two left channels (due west!), quite distant. Since the values in the table are accessed through the use of a time-pointer in the space unit, the actual timing can be made to follow the file's timing exactly or it can be made to go faster or slower through the same trajectory. If you have access to the GUI that allows one to draw and edit the files, there is no need to create the text files manually. But as long as the file is ASCII and in the format shown above, it doesn't matter how it is made! Important If ifn is 0, then space will take its values for the xy coordinates from kx and ky. Performance The configuration of the xy coordinates in space places the signal in the following way: a1 is -1, 1 a2 is 1, 1 a3 is -1, -1 a4 is 1, -1 This assumes a loudspeaker set up as a1 is left front, a2 is right front, a3 is left back, a4 is right back. Values greater than 1 will result in sounds being attenuated, as if in the distance. space considers the speakers to be at a distance of 1; smaller values of xy can be used, but space will not amplify the signal in this case. It will, however balance the signal so that it can sound as if it were within the 4 speaker space. x=0, y=1, will place the signal equally balanced between left and right front channels, x=y=0 will place the signal equally in all 4 channels, and so on. Although there must be 4 output signals from space, it can be used in a 2 channel orchestra. If the xy's are kept so that y>=1, it should work well to do panning and fixed localization in a stereo field. asig -- input audio signal. ktime -- index into the table containing the xy coordinates. If used like: ktime line 0, 5, 5 a1, a2, a3, a4 space asig, 1, ktime, ... with the file move described above, the speed of the signal's movement will be exactly as described in that file. However: ktime line 0, 10, 5 the signal will move at half the speed specified. Or in the case of: ktime line 5, 15, 0 the signal will move in the reverse direction as specified and 3 times slower! Finally: ktime line 2, 10, 3 will cause the signal to move only from the place specified in line 3 of the text file to the place specified in line 5 of the text file, and it will take 10 seconds to do it. kreverbsend -- the percentage of the direct signal that will be factored along with the distance as derived from the xy coordinates to calculate signal amounts that can be sent to reverb units such as reverb, or reverb2. kx, ky -- when ifn is 0, space and spdist will use these values as the xy coordinates to localize the signal. Examples Here is an example of the space opcode. It uses the file space_quad.csd. Example of the space opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. In the above example, the signal, asig, is moved according to the data in Function #1 indexed by ktime. space sends the appropriate amount of the signal internally to spsend. The outputs of the spsend are added to global accumulators in a common Csound style and the global signals are used as inputs to the reverb units in a separate instrument. space can be useful for quad and stereo panning as well as fixed placed of sounds anywhere between two loudspeakers. Below is an example of the fixed placement of sounds in a stereo field using xy values from the score instead of a function table. It uses the file space_stereo.csd. Second example of the space opcode. spdist demonstrates an example of a simple intuitive use of the distance values to simulate Doppler shift. See Also spdist, spsend Credits Author: Richard Karpen Seattle, WA USA 1998 New in Csound version 3.48 manual_src~dfsg/opcodes/urd.xml0000644000000000000000000000665612262561504015732 0ustar rootroot urd Signal Generators:Random (Noise) Generators urd urd A discrete user-defined-distribution random generator that can be used as a function. Description A discrete user-defined-distribution random generator that can be used as a function. Syntax aout = urd(ktableNum) iout = urd(itableNum) kout = urd(ktableNum) Initialization itableNum -- number of table containing the random-distribution function. Such table is generated by the user. See GEN40, GEN41, and GEN42. The table length does not need to be a power of 2 Performance ktableNum -- number of table containing the random-distribution function. Such table is generated by the user. See GEN40, GEN41, and GEN42. The table length does not need to be a power of 2 urd is the same opcode as duserrnd, but can be used in function fashion. For a tutorial about random distribution histograms and functions see: D. Lorrain. "A panoply of stochastic cannons". In C. Roads, ed. 1989. Music machine. Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the urd opcode. It uses the file urd.csd. Example of the urd opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i1 184.61538 i1 130.76923 i1 169.23077 i1 12.00000 ....... WARNING: Seeding from current time 3751086165 i2 138.46154 i2 12.00000 i2 123.07692 i2 161.53846 i2 123.07692 i2 153.84615 ...... See Also cuserrnd, duserrnd Credits Author: &namegabriel; New in Version 4.16 manual_src~dfsg/opcodes/strset.xml0000644000000000000000000000444012262561504016451 0ustar rootroot strset Strings:Definition strset strset Allows a string to be linked with a numeric value. Description Allows a string to be linked with a numeric value. Syntax strset iarg, istring Initialization iarg -- the numeric value. istring -- the alphanumeric string (in double-quotes). strset (optional) allows a string, such as a filename, to be linked with a numeric value. Its use is optional. Examples The following statement, used in the orchestra header, will allow the numeric value 10 to be substituted anywhere the soundfile asound.wav is called for. strset 10, "asound.wav" Examples Here is an example of the strset opcode. It uses the file strset.csd. Example of the strset opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pset and strget manual_src~dfsg/opcodes/koutpat.xml0000644000000000000000000000077112262561504016617 0ustar rootroot koutpat koutpat koutpat Deprecated. Description Deprecated as of version 3.52. Use the outkpat opcode instead. manual_src~dfsg/opcodes/clip.xml0000644000000000000000000001205612262561504016056 0ustar rootroot clip Signal Modifiers:Amplitude Modifiers clip clip Clips a signal to a predefined limit. Description Clips an a-rate signal to a predefined limit, in a soft manner, using one of three methods. Syntax ares clip asig, imeth, ilimit [, iarg] Initialization imeth -- selects the clipping method. The default is 0. The methods are: 0 = Bram de Jong method (default) 1 = sine clipping 2 = tanh clipping ilimit -- limiting value iarg (optional, default=0.5) -- when imeth = 0, indicates the point at which clipping starts, in the range 0 - 1. Not used when imeth = 1 or imeth = 2. Default is 0.5. Performance asig -- a-rate input signal The Bram de Jong method (imeth = 0) applies the algorithm (denoting ilimit as limit and iarg as a): |x| >= 0 and |x| <= (limit*a): f(x) = f(x) |x| > (limit*a) and |x| <= limit: f(x) = sign(x) * (limit*a+(x-limit*a)/(1+((x-limit*a)/(limit*(1-a)))2)) |x| > limit: f(x) = sign(x) * (limit*(1+a))/2 The second method (imeth = 1) is the sine clip: |x| < limit: f(x) = limit * sin(π*x/(2*limit)), |x| >= limit: f(x) = limit * sign(x) The third method (imeth = 2) is the tanh clip: |x| < limit: f(x) = limit * tanh(x/limit)/tanh(1), |x| >= limit: f(x) = limit * sign(x) Note Method 1 appears to be non-functional at release of Csound version 4.07. Examples Here is an example of the clip opcode. It uses the file clip.csd. Example of the clip opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; University of Bath, Codemist Ltd. Bath, UK August, 2000 New in Csound version 4.07 September 2009: Thanks to a note from Paolo Dell'Osso, corrected the formula. manual_src~dfsg/opcodes/dssiaudio.xml0000644000000000000000000000515612262561504017116 0ustar rootroot dssiaudio Plugin Hosting:DSSI and LADSPA dssiaudio dssiaudio Processes audio using a LADSPA or DSSI plugin. Syntax [aout1, aout2, ..., aout9] dssiaudio ihandle, [ain1, ain2, ..., ain9] Description dssiaudio generates audio by processing an input signal through a LADSPA plugin. Initialization ihandle - handle for the plugin returned by dssiinit Performance aout1, aout2, etc - Audio ouput generated by the plugin ain1, ain2, etc - Audio provided to the plugin for processing dssiaudio runs a plugin on the provided audio and produces audio output. Currently upto four inputs and outputs are provided. You should provide signal for all the plugins audio inputs, otherwise unpredictable results may occur. If the plugin doesn't have any input (e.g Noise generator) you must still provide at least one input variable, which will be ignored with a message. Only one dssiaudio should be executed once per plugin, or strange results may occur. Examples Here is an example of the dssiaudio opcode. It uses the file dssiaudio.csd. Example of the dssiaudio opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits 2005 By: &nameandres; Uses code from Richard Furse's LADSPA sdk. manual_src~dfsg/opcodes/subinstr.xml0000644000000000000000000000737212262561504017005 0ustar rootroot subinstr Instrument Control:Subinstrument Control subinstr subinstr Creates and runs a numbered instrument instance. Description Creates an instance of another instrument and is used as if it were an opcode. Syntax a1, [...] [, a8] subinstr instrnum [, p4] [, p5] [...] a1, [...] [, a8] subinstr "insname" [, p4] [, p5] [...] Initialization instrnum -- Number of the instrument to be called. insname -- A string (in double-quotes) representing a named instrument. For more information about specifying input and output interfaces, see Calling an Instrument within an Instrument. Performance a1, ..., a8 -- The audio output from the called instrument. This is generated using the signal output opcodes. p4, p5, ... -- Additional input values the are mapped to the called instrument p-fields, starting with p4. The called instrument's p2 and p3 values will be identical to the host instrument's values. While the host instrument can control its own duration, any such attempts inside the called instrument will most likely have no effect. See Also Calling an Instrument within an Instrument, event, schedule, subinstrinit Examples Here is an example of the subinstr opcode. It uses the file subinstr.csd. Example of the subinstr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an example of the subinstr opcode using a named instrument. It uses the file subinstr_named.csd. Example of the subinstr opcode using a named instrument. Credits New in version 4.21 manual_src~dfsg/opcodes/balance.xml0000644000000000000000000000557212262561504016521 0ustar rootroot balance Signal Modifiers:Amplitude Modifiers balance balance Adjust one audio signal according to the values of another. Description The rms power of asig can be interrogated, set, or adjusted to match that of a comparator signal. Syntax ares balance asig, acomp [, ihp] [, iskip] Initialization ihp (optional) -- half-power point (in Hz) of a special internal low-pass filter. The default value is 10. iskip (optional, default=0) -- initial disposition of internal data space (see reson). The default value is 0. Performance asig -- input audio signal acomp -- the comparator signal balance outputs a version of asig, amplitude-modified so that its rms power is equal to that of a comparator signal acomp. Thus a signal that has suffered loss of power (eg., in passing through a filter bank) can be restored by matching it with, for instance, its own source. It should be noted that gain and balance provide amplitude modification only - output signals are not altered in any other respect. Examples Here is an example of the balance opcode. It uses the file balance.csd. Example of the balance opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also gain, rms manual_src~dfsg/opcodes/imagecreate.xml0000644000000000000000000000447212262561504017400 0ustar rootroot imagecreate Image Processing Opcodes imagecreate imagecreate Create an empty image of a given size. Description Create an empty image of a given size. Individual pixel values can then be set with. imagegetpixel. Syntax iimagenum imagecreate iwidth, iheight Initialization iimagenum -- number assigned to the created image. iwidth -- desired image width. iheight -- desired image height. Examples Here is an example of the imagecreate opcode. It uses the file imageopcodes.csd. Example of the imageload opcode. See Also imageload, imagesize, imagesave, imagegetpixel, imagesetpixel, imagefree Credits Author: Cesare Marilungo New in version 5.08 manual_src~dfsg/opcodes/ibetarand.xml0000644000000000000000000000100512262561504017050 0ustar rootroot ibetarand ibetarand ibetarand Deprecated. Description Deprecated as of version 3.49. Use the betarand opcode instead. manual_src~dfsg/opcodes/ATSread.xml0000644000000000000000000001067612262561504016420 0ustar rootroot ATSread Spectral Processing:ATS ATSread ATSread reads data from an ATS file. Description ATSread returns the amplitude (kamp) and frequency (kfreq) information of a user specified partial contained in the ATS analysis file at the time indicated by the time pointer ktimepnt. Syntax kfreq, kamp ATSread ktimepnt, iatsfile, ipartial Initialization iatsfile – the ATS number (n in ats.n) or the name in quotes of the analysis file made using ATS. ipartial – the number of the analysis partial to return the frequency in Hz and amplitude. Performance kfreq, kamp - outputs of the ATSread unit. These values represent the frequency and amplitude of a specific partial selected by the user using ipartial. The partials' informations are derived from an ATS analysis. ATSread linearly interpolates the frequency and amplitude between frames in the ATS analysis file at k-rate. The output is dependent on the data in the analysis file and the pointer ktimepnt. ktimepnt – The time pointer in seconds used to index the ATS file. Used for ATSread exactly the same as for pvoc and ATSadd. Examples Here is an example of the ATSread opcode. It uses the file ATSread.csd. Example of the ATSread opcode. Here we're using ATSread to get the 100th partial's frequency and amplitude data out of the 'beats.ats' ATS analysis file. We're using that data to drive an oscillator, but we could use it for anything else that can take a k-rate input, like the bandwidth and resonance of a filter etc. Here is another example of the ATSread opcode. It uses the file ATSread-2.csd. Example 2 of the ATSread opcode. See also ATSreadnz, ATSinfo, ATSbufread, ATScross, ATSinterpread, ATSpartialtap, ATSadd, ATSaddnz, ATSsinnoi Credits Author: Alex Norman Seattle,Washington 2004 manual_src~dfsg/opcodes/zkcl.xml0000644000000000000000000000522512262561504016072 0ustar rootroot zkcl Zak Patch System zkcl zkcl Clears one or more variables in the zk space. Description Clears one or more variables in the zk space. Syntax zkcl kfirst, klast Performance ksig -- the input signal kfirst -- first zk or za location in the range to clear. klast -- last zk or za location in the range to clear. zkcl clears one or more variables in the zk space. This is useful for those variables which are used as accumulators for mixing k-rate signals at each cycle, but which must be cleared before the next set of calculations. Examples Here is an example of the zkcl opcode. It uses the file zkcl.csd. Example of the zkcl opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also zacl, zkwm, zkw, zkmod, zkr Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/tambourine.xml0000644000000000000000000001040212262561504017265 0ustar rootroot tambourine Signal Generators:Models and Emulations tambourine tambourine Semi-physical model of a tambourine sound. Description tambourine is a semi-physical model of a tambourine sound. It is one of the PhISEM percussion opcodes. PhISEM (Physically Informed Stochastic Event Modeling) is an algorithmic approach for simulating collisions of multiple independent sound producing objects. Syntax ares tambourine kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ [, ifreq1] [, ifreq2] Initialization idettack -- period of time over which all sound is stopped inum (optional) -- The number of beads, teeth, bells, timbrels, etc. If zero, the default value is 32. idamp (optional) -- the damping factor, as part of this equation: damping_amount = 0.9985 + (idamp * 0.002) The default damping_amount is 0.9985 which means that the default value of idamp is 0. The maximum damping_amount is 1.0 (no damping). This means the maximum value for idamp is 0.75. The recommended range for idamp is usually below 75% of the maximum value. imaxshake (optional, default=0) -- amount of energy to add back into the system. The value should be in range 0 to 1. ifreq (optional) -- the main resonant frequency. The default value is 2300. ifreq1 (optional) -- the first resonant frequency. The default value is 5600. ifreq2 (optional) -- the second resonant frequency. The default value is 8100. Performance kamp -- Amplitude of output. Note: As these instruments are stochastic, this is only an approximation. Examples Here is an example of the tambourine opcode. It uses the file tambourine.csd. Example of the tambourine opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also bamboo, dripwater, guiro, sleighbells Credits Author: &nameperry;, part of the PhISEM (Physically Informed Stochastic Event Modeling) Adapted by &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.07 Added notes by &namerasmus; on May 2002. manual_src~dfsg/opcodes/slider16tablef.xml0000644000000000000000000001525612262561504017743 0ustar rootroot slider16tablef Real-time MIDI:Slider Banks slider16tablef slider16tablef Stores a bank of 16 different MIDI control messages to a table, filtered before output. Description Stores a bank of 16 different MIDI control messages to a table, filtered before output. Syntax kflag slider16tablef ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, icutoff1, .... , ictlnum16, imin16, imax16, init16, ifn16, icutoff16 Initialization ichan -- MIDI channel (1-16) ioutTable -- number of the table that will contain the output ioffset -- output table offset. A zero means that the output of the first slider will affect the first table element. A 10 means that the output of the first slider will affect the 11th table element. ictlnum1 ... ictlnum16 -- MIDI control number (0-127) imin1 ... imin16 -- minimum values for each controller imax1 ... imax16 -- maximum values for each controller init1 ... init16 -- initial value for each controller ifn1 ... ifn16 -- function table for conversion for each controller icutoff1 ... icutoff16 -- low-pass filter cutoff frequency for each controller Performance kflag -- a flag that informs if any control-change message in the bank has been received. In this case kflag is set to 1. Otherwise is set to zero. slider16tablef is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider16tablef allows a bank of 16 different MIDI control message numbers. It filters the signal before output. This eliminates discontinuities due to the low resolution of the MIDI (7 bit). The cutoff frequency can be set separately for each controller (suggested range: .1 to 5 Hz). As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. slider16tablef is very similar to slider16f and sliderNf family of opcodes (see their manual for more information). The actual difference is that the output is not stored to k-rate variables, but to a table, denoted by the ioutTable argument. It is possible to define a starting index in order to use the same table for more than one slider bank (or other purposes). It is possible to use this opcode together with FLslidBnk2Setk and FLslidBnk2, so you can synchronize the position of the MIDI values to the position of the FLTK valuator widgets of FLslidBnk2. Notice that you have to specify the same min/max values as well the linear/exponential responses in both sliderNtable(f) and FLslidBnk2. The exception is when using table-indexed response instead of a lin/exp response. In this case, in order to achieve a useful result, the table-indexed response and actual min/max values must be set only in FLslidBnk2, whereas, in sliderNtable(f), you have to set a linear response and a minimum of zero and a maximum of one in all sliders. slider16tablef does not output the required initial value immediately, but only after some k-cycles because the filter slightly delays the output. See Also slider16table, slider32table, slider32tablef, slider64table, slider64tablef, slider8table, slider8tablef Credits Author: &namegabriel; New in Csound version 5.06 manual_src~dfsg/opcodes/strfromurl.xml0000644000000000000000000000366212262561504017351 0ustar rootroot strfromurl Strings:Definition strfromurl strfromurl Set string variable to value read from an URL Description strfromurl sets a string variable at initialization time to the value found from reading an URL. Syntax Sdst strfromurl StringURL Initialization StringURL -- string naming an URL. Sdst -- destination string variable Examples Here is an example of the strfromurl opcode. It uses the file strfromurl.csd. Example of the strfromurl opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also strset Credits Author: &namejohn; 2013 URLs available from Csound 6.02 if built with support manual_src~dfsg/opcodes/print.xml0000644000000000000000000000554512262561504016270 0ustar rootroot print Signal I/O:Printing and Display print print Displays the values init (i-rate) variables. Description These units will print orchestra init-values. Syntax print iarg [, iarg1] [, iarg2] [...] Initialization iarg, iarg2, ... -- i-rate arguments. Performance print -- print the current value of the i-time arguments (or expressions) iarg at every i-pass through the instrument. The print opcode will truncate decimal places and may not show the complete value. Csound's precision depends on whether it is the floats (32-bit) or double (64-bit) version, since most internal calculations use one of these formats. If you need more resolution in the console output, you can try printf. Examples Here is an example of the print opcode. It uses the file print.csd. Example of the print opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: instr 1: irand = 2.930 instr 1: irand = 0.420 instr 1: irand = 2.316 See Also dispfft, display, printk, printk2, printks , printf and prints manual_src~dfsg/opcodes/xadsr.xml0000644000000000000000000000716012262561504016250 0ustar rootroot xadsr Signal Generators:Envelope Generators xadsr xadsr Calculates the classical ADSR envelope. Description Calculates the classical ADSR envelope Syntax ares xadsr iatt, idec, islev, irel [, idel] kres xadsr iatt, idec, islev, irel [, idel] Initialization iatt -- duration of attack phase idec -- duration of decay islev -- level for sustain phase irel -- duration of release phase idel -- period of zero before the envelope starts Performance The envelope generated is the range 0 to 1 and may need to be scaled further, depending on the amplitude required. If using 0dbfs = 1, scaling down will probably be required since playing more than one note might result in clipping. If not using 0dbfs, scaling to a large amplitude (e.g. 32000) might be required. The envelope may be described as: Picture of an ADSR envelope. Picture of an ADSR envelope. The length of the sustain is calculated from the length of the note. This means xadsr is not suitable for use with MIDI events, use mxadsr instead. The opcode xadsr is identical to adsr except it uses exponential, rather than linear, line segments. Examples Here is an example of the xadsr opcode. It uses the file xadsr.csd. Example of the xadsr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also adsr, madsr, mxadsr Credits Author: &namejohn; xadsr is new in Csound version 3.51. manual_src~dfsg/opcodes/deltapx.xml0000644000000000000000000000762712262561504016600 0ustar rootroot deltapx Signal Modifiers:Delay deltapx deltapx Read from or write to a delay line with interpolation. Description deltapx is similar to deltapi or deltap3. However, it allows higher quality interpolation. This opcode can read from and write to a delayr/delayw delay line with interpolation. Syntax aout deltapx adel, iwsize Initialization iwsize -- interpolation window size in samples. Allowed values are integer multiplies of 4 in the range 4 to 1024. iwsize = 4 uses cubic interpolation. Increasing iwsize improves sound quality at the expense of CPU usage, and minimum delay time. Performance aout -- Output signal. adel -- Delay time in seconds. a1 delayr idlr deltapxw a2, adl1, iws1 a3 deltapx adl2, iws2 deltapxw a4, adl3, iws3 delayw a5 Minimum and maximum delay times: idlr >= 1/kr Delay line length adl1 >= (iws1/2)/sr Write before read adl1 <= idlr - (1 + iws1/2)/sr (allows shorter delays) adl2 >= 1/kr + (iws2/2)/sr Read time adl2 <= idlr - (1 + iws2/2)/sr adl2 >= adl1 + (iws1 + iws2) / (2*sr) adl2 >= 1/kr + adl3 + (iws2 + iws3) / (2*sr) adl3 >= (iws3/2)/sr Write after read adl3 <= idlr - (1 + iws3/2)/sr (allows feedback) Note Window sizes for opcodes other than deltapx are: deltap, deltapn: 1, deltapi: 2 (linear), deltap3: 4 (cubic) Examples Here is an example of the deltapx opcode. It uses the file deltapx.csd. Example of the deltapx opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also deltapxw Credits Author: &nameistvan; August 2001 New in version 4.13 manual_src~dfsg/opcodes/svfilter.xml0000644000000000000000000001056312262561504016766 0ustar rootroot svfilter Signal Modifiers:Standard Filters:Resonant svfilter svfilter A resonant second order filter, with simultaneous lowpass, highpass and bandpass outputs. Description Implementation of a resonant second order filter, with simultaneous lowpass, highpass and bandpass outputs. Syntax alow, ahigh, aband svfilter asig, kcf, kq [, iscl] Initialization iscl -- coded scaling factor, similar to that in reson. A non-zero value signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A zero value signifies no scaling of the signal, leaving that to some later adjustment (see balance). The default value is 0. Performance svfilter is a second order state-variable filter, with k-rate controls for cutoff frequency and Q. As Q is increased, a resonant peak forms around the cutoff frequency. svfilter has simultaneous lowpass, highpass, and bandpass filter outputs; by mixing the outputs together, a variety of frequency responses can be generated. The state-variable filter, or "multimode" filter was a common feature in early analog synthesizers, due to the wide variety of sounds available from the interaction between cutoff, resonance, and output mix ratios. svfilter is well suited to the emulation of "analog" sounds, as well as other applications where resonant filters are called for. asig -- Input signal to be filtered. kcf -- Cutoff or resonant frequency of the filter, measured in Hz. kq -- Q of the filter, which is defined (for bandpass filters) as bandwidth/cutoff. kq should be in a range between 1 and 500. As kq is increased, the resonance of the filter increases, which corresponds to an increase in the magnitude and "sharpness" of the resonant peak. When using svfilter without any scaling of the signal (where iscl is either absent or 0), the volume of the resonant peak increases as Q increases. For high values of Q, it is recommended that iscl be set to a non-zero value, or that an external scaling function such as balance is used. svfilter is based upon an algorithm in Hal Chamberlin's Musical Applications of Microprocessors (Hayden Books, 1985). Examples Here is an example of the svfilter opcode. It uses the file svfilter.csd. Example of the svfilter opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namesean; Seattle, Washington 1999 New in Csound version 3.55 manual_src~dfsg/opcodes/JackoNoteOut.xml0000644000000000000000000000602112262561504017467 0ustar rootroot JackoNoteOut Jacko Opcodes JackoNoteOut JackoNoteOut Sends a MIDI channel message to a Jack port. Description Sends a MIDI channel message to a Jack MIDI output port inside this instance of Csound, and in turn to its connected external Jack MIDI input port. Syntax JackoNoteOut ScsoundPortName, kstatus, kchannel, kdata1[, kdata2] Initialization ScsoundPortName -- The short name ("portname") of the internal Jack MIDI output port. Performance kstatus -- MIDI status byte; must indicate a MIDI channel message. kchannel -- MIDI channel (from 0 through 15). kdata1 -- First data byte of a MIDI channel message. kdata2 -- Optional second data byte of a MIDI channel message. This opcode can be called any number of times in the same kperiod. Messages from multiple instances of the opcode sending to the same port are collected before sending. Running status, system exclusive messages, and real-time messages are not supported. The granularity of timing is Csound's kperiod. See Also JackoInfo, JackoInfo, JackoFreewheel, JackoAudioInConnect, JackoAudioOutConnect, JackoMidiInConnect, JackoMidiOutConnect, JackoOn, JackoAudioIn, JackoMidiOut, The Jacko Opcodes. Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/flsetposition.xml0000644000000000000000000000615612262561504020035 0ustar rootroot FLsetPosition FLTK:Appearance FLsetPosition FLsetPosition Sets the position of a FLTK widget. Description FLsetPosition sets the position of the target widget according to the ix and iy arguments. Syntax FLsetPosition ix, iy, ihandle Initialization ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/pvscross.xml0000644000000000000000000000635412262561504017015 0ustar rootroot pvscross Spectral Processing:Streaming pvscross pvscross Performs cross-synthesis between two source fsigs. Description Performs cross-synthesis between two source fsigs. Syntax fsig pvscross fsrc, fdest, kamp1, kamp2 Performance The operation of this opcode is identical to that of pvcross (q.v.), except in using fsigs rather than analysis files, and the absence of spectral envelope preservation. The amplitudes from fsrc and fdest (using scale factors kamp1 for fsrc and kamp2 for fdest) are applied to the frequencies of fsrc. kamp1 and kamp2 must not exceed the range 0 to 1. With this opcode, cross-synthesis can be performed on real-time audio input, by using pvsanal to generate fsrc and fdest. These must have the same format. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the use of the pvscross opcode. It uses the file pvscross.csd. Example of the <emphasis>pvscross</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namerichard; August 2001 November 2003. Thanks to &namekanata;, fixed the link to the pvcross opcode. New in version 4.13 manual_src~dfsg/opcodes/clockoff.xml0000644000000000000000000000445712262561504016723 0ustar rootroot clockoff Instrument Control:Clock Control clockoff clockoff Stops one of a number of internal clocks. Description Stops one of a number of internal clocks. Syntax clockoff inum Initialization inum -- the number of a clock. There are 32 clocks numbered 0 through 31. All other values are mapped to clock number 32. Performance Between a clockon and a clockoff opcode, the CPU time used is accumulated in the clock. The precision is machine dependent but is the millisecond range on UNIX and Windows systems. The readclock opcode reads the current value of a clock at initialization time. Examples Here is an example of the clockoff opcode. It uses the file clockoff.csd. Example of the clockoff opcode. See Also clockon, readclock Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK July, 1999 New in Csound version 3.56 manual_src~dfsg/opcodes/flgetsnap.xml0000644000000000000000000000631012262561504017106 0ustar rootroot FLgetsnap FLTK:Other FLgetsnap FLgetsnap Retrieves a previously stored FLTK snapshot. Description Retrieves a previously stored snapshot (in memory), i.e. sets all valuator to the corresponding values stored in that snaphot. Syntax inumsnap FLgetsnap index [, igroup] Initialization inumsnap -- current number of snapshots. index -- a number referring unequivocally to a snapshot. Several snapshots can be stored in the same bank. igroup -- (optional) an integer number referring to a snapshot-related group of widget. It allows to get/set, or to load/save the state of a subset of valuators. Default value is zero that refers to the first group. The group number is determined by the opcode FLsetSnapGroup. The igroup parameter has not been yet fully implemented in the current version of csound. Please do not rely on it yet. Performance FLgetsnap retrieves a previously stored snapshot (in memory), i.e. sets all valuator to the corresponding values stored in that snapshot. The index argument unequivocally must refer to an already existing snapshot. If the index argument refers to an empty snapshot or to a snapshot that doesn't exist, no action is done. FLsetsnap outputs the current number of snapshots (inumsnap argument). For purposes of snapshot saving, widgets can be grouped, so that snapshots affect only a defined group of widgets. The opcode FLsetSnapGroup is used to specify the group for all widgets declared after it, until the next FLsetSnapGroup statement. See Also FLloadsnap, FLrun, FLsavesnap, FLsetsnap, FLsetSnapGroup, FLupdate Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/vecdelay.xml0000644000000000000000000000450312262561504016721 0ustar rootroot vecdelay Vectorial:Delay Paths vecdelay vecdelay Vectorial Control-rate Delay Paths Description Generate a sort of 'vectorial' delay Syntax vecdelay ifn, ifnIn, ifnDel, ielements, imaxdel [, iskip] Initialization ifn - number of the table containing the output vector ifnIn - number of the table containing the input vector ifnDel - number of the table containing a vector whose elements contain delay values in seconds ielements - number of elements of the two vectors imaxdel - Maximum value of delay in seconds. iskip (optional) - initial disposition of delay-loop data space (see reson). The default value is 0. Performance vecdelay is similar to vdelay, but it works at k-rate and, instead of delaying a single signal, it delays a vector. ifnIn is the input vector of signals, ifn is the output vector of signals, and ifnDel is a vector containing delay times for each element, expressed in seconds. Elements of ifnDel can be updated at k-rate. Each single delay can be different from that of the other elements, and can vary at k-rate. imaxdel sets the maximum delay allowed for all elements of ifnDel. Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/foutk.xml0000644000000000000000000000750612262561504016263 0ustar rootroot foutk Signal I/O:File I/O foutk foutk Outputs k-rate signals of an arbitrary number of channels to a specified file, in raw (headerless) format. Description foutk outputs N k-rate signals to a specified file of N channels. Syntax foutk ifilename, iformat, kout1 [, kout2, kout3,....,koutN] Initialization ifilename -- the output file's name (in double-quotes). iformat -- a flag to choose output file format (note: Csound versions older than 5.0 may only support formats 0 and 1): 0 - 32-bit floating point samples without header (binary PCM multichannel file) 1 - 16-bit integers without header (binary PCM multichannel file) 2 - 16-bit integers without header (binary PCM multichannel file) 3 - u-law samples without header 4 - 16-bit integers without header 5 - 32-bit integers without header 6 - 32-bit floats without header 7 - 8-bit unsigned integers without header 8 - 24-bit integers without header 9 - 64-bit floats without header Performance kout1,...koutN -- control-rate signals to be written to the file. The expected range of the signals is determined by the selected sample format. foutk operates in the same way as fout, but with k-rate signals. iformat can be set only in the range 0 to 9, or 0 to 1 with an old version of Csound. Notice that fout and foutk can use either a string containing a file pathname, or a handle-number generated by fiopen. Whereas, with fouti and foutir, the target file can be only specified by means of a handle-number. See Also fiopen, fout, fouti, foutir Credits Author: &namegabriel; Italy 1999 New in Csound version 3.56 manual_src~dfsg/opcodes/oscilx.xml0000644000000000000000000000104612262561504016425 0ustar rootroot oscilx Signal Generators:Basic Oscillators oscilx oscilx Same as the osciln opcode. Description Same as the osciln opcode. manual_src~dfsg/opcodes/minaccum.xml0000644000000000000000000000717012262561504016724 0ustar rootroot minaccum Signal Modifiers:Comparators and Accumulators minaccum minaccum Accumulates the minimum value of audio signals. Description minaccum compares two audio-rate variables and stores the minimum value between them into the first. Syntax minaccum aAccumulator, aInput Performance aAccumulator -- audio variable to store the minimum value aInput -- signal that aAccumulator is compared to The minaccum opcode is designed to accumulate the minimum value from among many audio signals that may be in different note instances, different channels, or otherwise cannot all be compared at once using the min opcode. Its semantics are similar to vincr since aAccumulator is used as both an input and an output variable, except that minaccum keeps the minimum value instead of adding the signals together. minaccum performs the following operation on each pair of samples: if (aInput < aAccumulator) aAccumulator = aInput aAccumulator will usually be a global audio variable. At the end of any given computation cycle (k-period), after its value is read and used in some way, the accumulator variable should usually be reset to some large enough positive value that will always be greater than the input signals to which it is compared. Examples Here is an example of the minaccum opcode. It uses the file minaccum.csd. Example of the minaccum opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also maxaccum, maxabsaccum, minabsaccum, max, min, maxabs, minabs, vincr Credits Author: &nameanthony; March 2006 New in Csound version 5.01 manual_src~dfsg/opcodes/outletk.xml0000644000000000000000000000640212262561504016614 0ustar rootroot outletk Signal Flow Graph Opcodes outletk outletk Sends a krate signal out from an instrument to a named port. Description Sends a krate signal out from an instrument to a named port. Syntax outletk Sname, ksignal Initialization Sname -- String name of the outlet port. The name of the outlet is implicitly qualified by the instrument name or number, so it is valid to use the same outlet name in more than one instrument (but not to use the same outlet name twice in one instrument). Performance ksignal -- krate output signal During performance, the krate output signal is sent to each instance of an instrument containing an inlet port to which this outlet has been connected using the connect opcode. The signals of all the outlets connected to an inlet are summed in the inlet. Examples Here is an example of the outletk opcode. It uses the file outletk.csd. Example of the outletk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also outleta outletf outletkid inleta inletk inletkid inletf connect alwayson ftgenonce More information on this opcode: http://www.csounds.com/journal/issue13/signalFlowGraphOpcodes.html , written by Michael Gogins Credits By: &namemichael; 2009 manual_src~dfsg/opcodes/spechist.xml0000644000000000000000000000406612262561504016753 0ustar rootroot spechist Spectral Processing:Non-Standard spechist spechist Accumulates the values of successive spectral frames. Description Accumulates the values of successive spectral frames. Syntax wsig spechist wsigin Performance wsigin -- the input spectra. Accumulates the values of successive spectral frames. At each new frame of wsigin, the accumulations-to-date in each magnitude track are written to the output spectrum. This unit thus provides a running histogram of spectral distribution. Examples wsig2 specdiff wsig1 ; sense onsets wsig3 specfilt wsig2, 2 ; absorb slowly specdisp wsig2, 0.1 ; & display both spectra specdisp wsig3, 0.1 See Also specaddm, specdiff, specfilt, specscal manual_src~dfsg/opcodes/partikkel.xml0000644000000000000000000004655612262561504017131 0ustar rootroot partikkel Signal Generators:Granular Synthesis partikkel partikkel Granular synthesizer with "per grain" control over many of its parameters. Has a sync input to sychronize its internal grain scheduler clock to an external clock source. Description partikkel was conceived after reading Curtis Roads' book "Microsound", and the goal was to create an opcode that was capable of all time-domain varieties of granular synthesis described in this book. The idea being that most of the techniques only differ in parameter values, and by having a single opcode that can do all varieties of granular synthesis makes it possible to interpolate between techniques. Granular synthesis is sometimes dubbed particle synthesis, and it was thought apt to name the opcode partikkel to distinguish it from other granular opcodes. Some of the input parameters to partikkel is table numbers, pointing to tables where values for the "per grain" parameter changes are stored. partikkel can use single-cycle or complex (e.g. sampled sound) waveforms as source waveforms for grains. Each grain consists of a mix of 4 source waveforms. Individual tuning of the base frequency can be done for each of the 4 source waveforms. Frequency modulation inside each grain is enabled via an auxillary audio input (awavfm). Trainlet synthesis is available, and trainlets can be mixed with wavetable based grains. Up to 8 separate audio outputs can be used. Syntax a1 [, a2, a3, a4, a5, a6, a7, a8] partikkel agrainfreq, \ kdistribution, idisttab, async, kenv2amt, ienv2tab, ienv_attack, \ ienv_decay, ksustain_amount, ka_d_ratio, kduration, kamp, igainmasks, \ kwavfreq, ksweepshape, iwavfreqstarttab, iwavfreqendtab, awavfm, \ ifmamptab, kfmenv, icosine, ktraincps, knumpartials, kchroma, \ ichannelmasks, krandommask, kwaveform1, kwaveform2, kwaveform3, \ kwaveform4, iwaveamptab, asamplepos1, asamplepos2, asamplepos3, \ asamplepos4, kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains \ [, iopcode_id] Initialization idisttab -- function table number, distribution for random grain displacements over time. The table values are interpreted as "displacement amount" scaled by 1/grainrate. This means that a value of 0.5 in the table will displace a grain by half the grainrate period. The table values are read randomly, and scaled by kdistribution. For realistic stochastic results, it is advisable not to use a too small table size, as this limits the amount of possible displacement values. This can also be utilized for other purposes, e.g. using quantized displacement values to work with controlled time displacement from the periodic grain rate. If kdistribution is negative, the table values will be read sequentially. A default table might be selected by using -1 as the ftable number, for idisttab the default uses a zero distribution (no displacement). ienv_attack -- function table number, attack shape of grain. Needs extended guard point. A default table might be selected by using -1 as the ftable number, for ienv_attack the default uses a square window (no enveloping). ienv_decay -- function table number, decay shape of grain. Needs extended guard point. A default table might be selected by using -1 as the ftable number, for ienv_decay the default uses a square window (no enveloping). ienv2tab -- function table number, additional envelope applied to grain, done after attack and decay envelopes. Can be used e.g. for fof formant synthesis. Needs extended guard point. A default table might be selected by using -1 as the ftable number, for ienv2tab the default uses a square window (no enveloping). icosine -- function table number, must contain a cosine, used for trainlets. Table size should be at least 2048 for good quality trainlets. igainmasks -- function table number, gain per grain. The sequence of values in the table is as follows: index 0 is used as a loop start point in reading the values, index 1 is used as a loop end point. Remaining indices contain gain values (normally in range 0 - 1, but other values are allowed, negative values will invert phase of waveform inside grain) for a sequence of grains, these are read at grain rate enabling exact patterns of "gain per grain". The loop start and end points are zero based with an origin at index 2, e.g. a loop start value of 0 and loop end value of 3 will read indices 2,3,4,5 in a loop at grain rate. A default table might be selected by using -1 as the ftable number, for igainmasks the default disables gain masking (all grains are given a gain masking value of 1). ichannelmasks -- function table number, see igainmasks for a description of how the values in the table are read. Range is 0 to N, where N is the number of output channels minus 1. A value of zero will send the grain to audio output 1 from the opcode. Fractional values are allowed, e.g. a value of 3.5 will mix the grain equally to outputs 4 and 5. The user is responsible for keeping the values in range, no range checking is done. The opcode will crash with out of range values. A default table might be selected by using -1 as the ftable number, for ichannelmasks the default disables channel masking (all grains are given a channel masking value of 0 and are sent to partikkel audio out 1). iwavfreqstarttab -- function table number, see igainmasks for a description of how the values in the table are read. Start frequency multiplicator for each grain. Pitch will glide from start frequency to end frequency following a line or curve as set by ksweepshape. A default table might be selected by using -1 as the ftable number, for iwavfreqstarttab the default uses a multiplicator of 1, disabling any start frequency modification. iwavfreqendtab -- function table number, see iwavfreqstarttab. End frequency multiplicator for each grain. A default table might be selected by using -1 as the ftable number, for iwavfreqendtab the default uses a multiplicator of 1, disabling any end frequency modification. ifmamptab -- function table number, see igainmasks for a description of how the values in the table are read. Frequency modulation index per grain. The signal awavfm will be multiplied by values read from this table. A default table might be selected by using -1 as the ftable number, for ifmamptab the default uses 1 as the index multiplicator, enabling fm for all grains. iwaveamptab -- function table number, the indices are read in a similar way to what is used for igainmasks. Index 0 is used as a loop start point, and index 1 is used as a loop end point. The rest of the indices are read in groups of 5, where each value represent a gain value for each of the 4 source waveforms, and the 5th value represent trainlet amplitude. A default table might be selected by using -1 as the ftable number, for iwaveamptab the default uses an equal mix of all 4 source waveforms (each with an amplitude of 0.5) and setting trainlet amp to zero. Computation of trainlets can be CPU intensive, and setting ktrainamp to zero will skip most of the trainlet computations. Trainlets will be normalized to peak (ktrainamp), compensating for amplitude variations caused by variations in kpartials and kchroma. imax_grains -- maximum number of grains per k-period. Estimating a large value should not affect performance, exceeding this value will lead to "oldest grains" being deleted. iopcode_id -- the opcode id, linking an instance of partikkel to an instance of partikkelsync, the linked partikkelsync will output trigger pulses synchronized to partikkel's grain maker scheduler. The default value is zero, which means no connection to any partikkelsync instances. Performance xgrainfreq -- number of grains per second. A value of zero is allowed, and this will defer all grain scheduling to the sync input. async -- sync input. Input values are added to the phase value of the internal grain maker clock, enabling tempo synchronization with an external clock source. As this is an a-rate signal, inputs are usually pulses of length 1/sr. Using such pulses, the internal phase value can be "nudged" up or down, enabling soft or hard synchronization. Negative input values decrements the internal phase, while positive values in the range 0 to 1 increments the internal phase. An input value of 1 will always make partikkel generate a grain. If the value remains at 1, the internal grain scheduler clock will pause but any currently playing grains will still play to end. kdistribution -- periodic or stochastic distribution of grains, 0 = periodic. Stochastic grain displacement is in the range of kdistribution/grainrate seconds. The stochastic distribution profile (random distribution) can be set in the idisttab table. If kdistribution is negative, the result is deterministic time displacement as described by idisttab (sequential read of displacement values). Maximum grain displacement in all cases is limited to 10 seconds, and a grain will keep the values (duration, pitch etc) it was given when it was first generated (before time displacement). Since grain displacement is relative to the grain rate, displacement amount is undefined at 0Hz grain rate and kdistribution is completely disabled in this case. kenv2amt -- amount of enveloping for the secondary envelope for each grain. Range 0 to 1, where 0 is no secondary enveloping (square window), a value of 0.5 will use an interpolation between a square window and the shape set by ienv2tab. ksustain_amount -- sustain time as fraction of grain duration. I.e. balance between enveloped time(attack+decay) and sustain level time. The sustain level is taken from the last value of the ienv_attack ftable. ka_d_ratio -- balance between attack time and decay time. For example, with ksustain_amount set to 0.5 and ka_d_ratio set to 0.5, the attack envelope of each grain will take 25% of the grain duration, full amplitude (sustain) will be held for 50% of the grain duration, and the decay envelope will take the remaining 25% of the grain duration. kduration -- grain duration in milliseconds. kamp -- amplitude scaling of the opcode's output. Multiplied by per grain amplitude read from igainmasks. Source waveform playback inside grains can consume a significant amount of CPU cycles, especially if grain duration is long so that we have a lot of overlapping grains. Setting kamp to zero will skip waveform playback inside grains (and not generate any sound, obviously). This can be used as a "soft" bypass method if we want to keep the opcode active but silent for some periods of time. kwavfreq -- transposition scaling. Multiplied with start and end transposition values read from iwavfreqstarttab and iwavfreqendtab. ksweepshape -- transposition sweep shape, controls the curvature of the transposition sweep. Range 0 to 1. Low values will hold the transposition at the start value longer and then drop to the end value quickly, high values will drop to the end value quickly. A value of 0.5 will give a linear sweep. A value of exactly 0 will bypass sweep and only use the start frequency, while a value of exactly 1 will bypass sweep and only use the end frequency. The sweep generator might be slightly inaccurate in hitting the end frequency when using a steep curve and very long grains. awavfm -- audio input for frequency modulation inside grain. kfmenv -- function table number, envelope for FM modulator signal enabling the modulation index to change over the duration of a grain. ktraincps -- trainlet fundamental frequency. knumpartials -- number of partials in trainlets. kchroma -- chroma of trainlets. A value of 1 give equal amplitude to each partial, higher values will reduce the amplitude of lower partials while strengthening the amplitude of the higher partials. krandommask -- random masking (muting) of individual grains. Range 0 to 1, where a value of 0 will give no masking (all grains are played), and a value of 1 will mute all grains. kwaveform1 -- table number for source waveform 1. kwaveform2 -- table number for source waveform 2. kwaveform3 -- table number for source waveform 3. kwaveform4 -- table number for source waveform 4. asamplepos1 -- start position for reading source waveform 1. asamplepos2 -- start position for reading source waveform 2. asamplepos3 -- start position for reading source waveform 3. asamplepos4 -- start position for reading source waveform 4. kwavekey1 -- original key of source waveform 1. Can be used to transpose each source waveform independently. kwavekey2 -- as kwavekey1, but for source waveform 2. kwavekey3 -- as kwavekey1, but for source waveform 3. kwavekey4 -- as kwavekey1, but for source waveform 4. Examples Here is an example of the partikkel opcode. It uses the file partikkel.csd. Example of the partikkel opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the partikkel opcode. It uses the file partikkel-2.csd. Example 2 of the partikkel opcode. See Also fof, fof2, fog, grain, grain2, grain3, granule, sndwarp, sndwarpst, syncgrain, syncloop, partikkelsync Credits Author: Thom Johansen Author: Torgeir Strand Henriksen Author: &nameoeyvind; April 2007 Examples written by Joachim Heintz and &nameoeyvind;. New in version 5.06 manual_src~dfsg/opcodes/table3.xml0000644000000000000000000000507412262561504016303 0ustar rootroot table3 Signal Generators:Table Access table3 table3 Accesses table values by direct indexing with cubic interpolation. Description Accesses table values by direct indexing with cubic interpolation. Syntax ares table3 andx, ifn [, ixmode] [, ixoff] [, iwrap] ires table3 indx, ifn [, ixmode] [, ixoff] [, iwrap] kres table3 kndx, ifn [, ixmode] [, ixoff] [, iwrap] Initialization ifn -- function table number. ixmode (optional) -- index data mode. The default value is 0. 0 = raw index 1 = normalized (0 to 1) ixoff (optional) -- amount by which index is to be offset. For a table with origin at center, use tablesize/2 (raw) or .5 (normalized). The default value is 0. iwrap (optional) -- wraparound index flag. The default value is 0. 0 = nowrap (index < 0 treated as index=0; index > tablesize sticks at index=size) 1 = wraparound. Performance table3 is identical to tablei, except that it uses cubic interpolation. (New in Csound version 3.50). See Also table, tablei, oscil1, oscil1i, osciln manual_src~dfsg/opcodes/STKTubeBell.xml0000644000000000000000000000716012262561504017207 0ustar rootroot STKTubeBell Signal Generators:STK Opcodes STKTubeBell STKTubeBell STKTubeBell is a tubular bell (orchestral chime) FM synthesis instrument. Description STKTubeBell is a tubular bell (orchestral chime) FM synthesis instrument. It uses two simple FM Pairs summed together, also referred to as algorithm 5 of the TX81Z. Syntax asignal STKTubeBell ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kmod -- controller 2, modulator index 1. Value range of kv1 is 0-127. kcross -- controller 4, crossfade of outputs. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kadsr -- controller 128, ADSR 2 and 4 target. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the TubeBell class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKTubeBell opcode. It uses the file STKTubeBell.csd. Example of the STKTubeBell opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/qnan.xml0000644000000000000000000000277112262561504016067 0ustar rootroot qnan Mathematical Operations:Mathematical Functions qnan qnan Questions whether the argument is a infinite number Description Returns the number of times the argument is not a number. Syntax qnan(x) (no rate restriction) Examples Here is an example of the qnan opcode. It uses the file qnan.csd. Example of the qnan opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by &namejohn;. New in Csound 5.14 manual_src~dfsg/opcodes/pinkish.xml0000644000000000000000000001400712262561504016572 0ustar rootroot pinkish Signal Generators:Random (Noise) Generators pinkish pinkish Generates approximate pink noise. Description Generates approximate pink noise (-3dB/oct response) by one of two different methods: a multirate noise generator after Moore, coded by Martin Gardner a filter bank designed by Paul Kellet Syntax ares pinkish xin [, imethod] [, inumbands] [, iseed] [, iskip] Initialization imethod (optional, default=0) -- selects filter method: 0 = Gardner method (default). 1 = Kellet filter bank. 2 = A somewhat faster filter bank by Kellet, with less accurate response. inumbands (optional) -- only effective with Gardner method. The number of noise bands to generate. Maximum is 32, minimum is 4. Higher levels give smoother spectrum, but above 20 bands there will be almost DC-like slow fluctuations. Default value is 20. iseed (optional, default=0) -- only effective with Gardner method. If non-zero, seeds the random generator. If zero, the generator will be seeded from current time. Default is 0. iskip (optional, default=0) -- if non-zero, skip (re)initialization of internal state (useful for tied notes). Default is 0. Performance xin -- for Gardner method: k- or a-rate amplitude. For Kellet filters: normally a-rate uniform random noise from rand (31-bit) or unirand, but can be any a-rate signal. The output peak value varies widely (±15%) even over long runs, and will usually be well below the input amplitude. Peak values may also occasionally overshoot input amplitude or noise. pinkish attempts to generate pink noise (i.e., noise with equal energy in each octave), by one of two different methods. The first method, by Moore & Gardner, adds several (up to 32) signals of white noise, generated at octave rates (sr, sr/2, sr/4 etc). It obtains pseudo-random values from an internal 32-bit generator. This random generator is local to each opcode instance and seedable (similar to rand). The second method is a lowpass filter with a response approximating -3dB/oct. If the input is uniform white noise, it outputs pink noise. Any signal may be used as input for this method. The high quality filter is slower, but has less ripple and a slightly wider operating frequency range than less computationally intense versions. With the Kellet filters, seeding is not used. The Gardner method output has some frequency response anomalies in the low-mid and high-mid frequency ranges. More low-frequency energy can be generated by increasing the number of bands. It is also a bit faster. The refined Kellet filter has very smooth spectrum, but a more limited effective range. The level increases slightly at the high end of the spectrum. Examples Here is an example of the pinkish opcode. It uses the file pinkish.csd. Example of the pinkish opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Kellet-filtered noise for a tied note (iskip is non-zero). Credits Authors: Phil Burk and &namejohn; University of Bath/Codemist Ltd. Bath, UK May 2000 New in Csound Version 4.07 Adapted for Csound by &namerasmus; The noise bands method is due to F. R. Moore (or R. F. Voss), and was presented by Martin Gardner in an oft-cited article in Scientific American. The present version was coded by Phil Burk as the result of discussion on the music-dsp mailing list, with significant optimizations suggested by James McCartney. The filter bank was designed by Paul Kellet, posted to the music-dsp mailing list. The whole pink noise discussion was collected on a HTML page by Robin Whittle, which is currently available at http://www.firstpr.com.au/dsp/pink-noise/. Added notes by &namerasmus; on September 2002. manual_src~dfsg/opcodes/dust.xml0000644000000000000000000000430012262561504016077 0ustar rootroot dust Signal Generators:Random (Noise) Generators dust dust Random impulses. Description Generates random impulses from 0 to 1. Syntax ares dust kamp, kdensity kres dust kamp, kdensity Performance kamp -- amplitude. kdensity -- average number of impulses per second. Examples Here is an example of the dust opcode. It uses the file dust.csd. Example of the dust opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also dust2 gausstrig mpulse Credits Based on James McCartney's Dust ugen (SuperCollider) Author: &nametito; January 2012 New in Csound version 5.16 manual_src~dfsg/opcodes/butbr.xml0000644000000000000000000000127012262561504016241 0ustar rootroot butbr Signal Modifiers:Standard Filters butbr butbr Same as the butterbr opcode. Description Same as the butterbr opcode. Syntax ares butbr asig, kfreq, kband [, iskip] manual_src~dfsg/opcodes/pvsin.xml0000644000000000000000000000501112262561504016257 0ustar rootroot pvsin Spectral Processing:Streaming pvsin pvsin Retrieve an fsig from the input software bus; a pvs equivalent to chani. Description This opcode retrieves an f-sig from the pvs in software bus, which can be used to get data from an external source, using the Csound 5 API. A channel is created if not already existing. The fsig channel is in that case initialised with the given parameters. It is important to note that the pvs input and output (pvsout opcode) busses are independent and data is not shared between them. Syntax fsig pvsin kchan[, isize, iolap, iwinsize, iwintype, iformat] Initialization isize -- initial DFT size,defaults to 1024. iolap -- size of overlap, defaults to isize/4. iwinsize -- size of analysis window, defaults to isize. iwintype -- type of window, defaults to Hanning (1) (see pvsanal) iformat -- data format, defaults 0 (PVS_AMP_FREQ). Other possible values are 1 (PVS_AMP_PHASE), 2 (PVS_COMPLEX) or 3 (PVS_TRACKS). Performance fsig -- output fsig. kchan -- channel number. If non-existent, a channel will be created. Examples Example fsig pvsin 0 ; get data from pvs in bus channel 0 Credits Author: &namevictor; August 2006 manual_src~dfsg/opcodes/strcmp.xml0000644000000000000000000000270212262561504016434 0ustar rootroot strcmp Strings:Manipulation strcmp strcmp Compare strings Description Compare strings and set the result to -1, 0, or 1 if the first string is less than, equal to, or greater than the second, respectively. strcmp compares at i-time only. Syntax ires strcmp S1, S2 Examples Here is an example of the strcmp opcode. It uses the file strcmp.csd. Example of the strcmp opcode. See also strcmpk Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/spectrum.xml0000644000000000000000000001052612262561504016771 0ustar rootroot spectrum Spectral Processing:Non-Standard spectrum spectrum Generate a constant-Q, exponentially-spaced DFT. Description Generate a constant-Q, exponentially-spaced DFT across all octaves of a multiply-downsampled control or audio input signal. Syntax wsig spectrum xsig, iprd, iocts, ifrqa [, iq] [, ihann] [, idbout] \ [, idsprd] [, idsinrs] Initialization ihann (optional) -- apply a Hamming or Hanning window to the input. The default is 0 (Hamming window) idbout (optional) -- coded conversion of the DFT output: 0 = magnitude 1 = dB 2 = mag squared 3 = root magnitude The default value is 0 (magnitude). idisprd (optional) -- if non-zero, display the composite downsampling buffer every idisprd seconds. The default value is 0 (no display). idsines (optional) -- if non-zero, display the Hamming or Hanning windowed sinusoids used in DFT filtering. The default value is 0 (no sinusoid display). Performance This unit first puts signal asig or ksig through iocts of successive octave decimation and downsampling, and preserves a buffer of down-sampled values in each octave (optionally displayed as a composite buffer every idisprd seconds). Then at every iprd seconds, the preserved samples are passed through a filter bank (ifrqs parallel filters per octave, exponentially spaced, with frequency/bandwidth Q of iq), and the output magnitudes optionally converted (idbout ) to produce a band-limited spectrum that can be read by other units. The stages in this process are computationally intensive, and computation time varies directly with iocts, ifrqs, iq, and inversely with iprd. Settings of ifrqs = 12, iq = 10, idbout = 3, and iprd = .02 will normally be adequate, but experimentation is encouraged. ifrqs currently has a maximum of 120 divisions per octave. For audio input, the frequency bins are tuned to coincide with A440. This unit produces a self-defining spectral datablock wsig, whose characteristics used (iprd, iocts, ifrqs, idbout) are passed via the data block itself to all derivative wsigs. There can be any number of spectrum units in an instrument or orchestra, but all wsig names must be unique. Examples asig in ; get external audio wsig spectrum asig, 0.02, 6, 12, 33, 0, 1, 1 ; downsample in 6 octs & calc a 72 pt dft ; (Q 33, dB out) every 20 msecs manual_src~dfsg/opcodes/sqrt.xml0000644000000000000000000000545112262561504016121 0ustar rootroot sqrt Mathematical Operations:Mathematical Functions sqrt sqrt Returns a square root value. Description Returns the square root of x (x non-negative). The argument value is restricted for log, log10, and sqrt. Syntax sqrt(x) (no rate restriction) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Examples Here is an example of the sqrt opcode. It uses the file sqrt.csd. Example of the sqrt opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: square root of left channel = 1.000000 square root of left channel = 0.948688 square root of left channel = 0.894437 square root of left channel = 0.836676 square root of left channel = 0.774620 square root of left channel = 0.707139 square root of left channel = 0.632499 square root of left channel = 0.547781 square root of left channel = 0.447295 square root of left channel = 0.316242 See Also abs, exp, frac, int, log, log10, i manual_src~dfsg/opcodes/faustgen.xml0000644000000000000000000000330712262561504016742 0ustar rootroot faustgen Faust Opcodes faustgen faustgen Compiles, Instantiates and runs a compiled Faust program. Description Faustgen will invoke the just-in-time compiler, instantiate and run a Faust program. Syntax ihandle,a1[,a2,...] faustgen SCode[,ain1,...] Initialization Scode -- a string containing a Faust program. ihandle -- a handle to the Faust DSP instance, which can be used to access its controls with faustctl. Performance ain1,... -- input signals a1,... -- output signals Examples Here is an example of the faustgen opcode, with a simple program that mixes two inputs: idsp,a1 faustgen "process=+;",ain1,ain2 Credits Author: &namevictor;, 2013 manual_src~dfsg/opcodes/multiplies.xml0000644000000000000000000001020612262561504017311 0ustar rootroot * Mathematical Operations:Arithmetic and Logic Operations * * Multiplication operator. Description Arithmetic operators perform operations of change-sign (negate), don't-change-sign, logical AND logical OR, add, subtract, multiply and divide. Note that a value or an expression may fall between two of these operators, either of which could take it as its left or right argument, as in a + b * c. In such cases three rules apply: 1. * and / bind to their neighbors more strongly than + and −. Thus the above expression is taken as a + (b * c) with * taking b and c and then + taking a and b * c. 2. + and − bind more strongly than &&, which in turn is stronger than ||: a && b - c || d is taken as (a && (b - c)) || d 3. When both operators bind equally strongly, the operations are done left to right: a - b - c is taken as (a - b) - c Parentheses may be used as above to force particular groupings. Syntax a * b (no rate restriction) where the arguments a and b may be further expressions. Arguments The arguments of * can be scalar values or k-rate one dimensional arrays (vectors), or any combination. If one of the arguments is an array, so is the value. Examples Here is an example of the * operator. It uses the file multiplies.csd. Example of the * operator. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: base amplitude * rising expon output = 0.8 * 0.010000 base amplitude * rising expon output = 0.8 * 0.012589 .......... base amplitude * rising expon output = 0.8 * 0.794328 base amplitude * rising expon output = 0.8 * 0.998466 See Also -, +, &&, ||, /, ˆ, % Credits Arithmetic on vectors is new in version 6.00 manual_src~dfsg/opcodes/imidic7.xml0000644000000000000000000000076712262561504016462 0ustar rootroot imidic7 imidic7 imidic7 Deprecated. Description Deprecated as of version 3.52. Use the midic7 opcode instead. manual_src~dfsg/opcodes/dssiactivate.xml0000644000000000000000000000552412262561504017614 0ustar rootroot dssiactivate Plugin Hosting:DSSI and LADSPA dssiactivate dssiactivate Activates or deactivates a DSSI or LADSPA plugin. Syntax dssiactivate ihandle, ktoggle Description dssiactivate is used to activate or deactivate a DSSI or LADSPA plugin. It calles the plugin's activate() and deactivate() functions if they are provided. Initialization ihandle - the number which identifies the plugin, generated by dssiinit. Performance ktoggle - Selects between activation (ktoggle=1) and deactivation (ktoggle=0). dssiactivate is used to turn on and off plugins if they provide this facility. This may help conserve CPU processing in some cases. For consistency, all plugins must be activated to produce sound. An inactive plugin produces silence. Depending on the plugin's implementation, this may cause interruptions in the realtime audio process, so use with caution. dssiactivate may cause audio stream breakups when used in realtime, so it is recommended to load all plugins to be used before playing. Please note that even if activate() and deactivate() functions are not present in a plugin, dssiactivate must be called for the plugin to produce sound. Examples Here is an example of the dssiactivate opcode. It uses the file dssiactivate.csd. Example of the dssiactivate opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits 2005 By: &nameandres; Uses code from Richard Furse's LADSPA sdk. manual_src~dfsg/opcodes/pvsmix.xml0000644000000000000000000000441112262561504016451 0ustar rootroot pvsmix Spectral Processing:Streaming pvsmix pvsmix Mix 'seamlessly' two pv signals. Description Mix 'seamlessly' two pv signals. This opcode combines the most prominent components of two pvoc streams into a single mixed stream. Syntax fsig pvsmix fsigin1, fsigin2 Performance fsig -- output pv stream fsigin1 -- input pv stream. fsigin2 -- input pv stream, which must have same format as fsigin1. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the pvsmix opcode. It uses the file pvsmix.csd. Example of the pvsmix opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; November 2004 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/pow.xml0000644000000000000000000000552012262561504015732 0ustar rootroot pow Mathematical Operations:Opcode Equivalents of Functions pow pow Computes one argument to the power of another argument. Description Computes xarg to the power of kpow (or ipow) and scales the result by inorm. Syntax ares pow aarg, kpow [, inorm] ires pow iarg, ipow [, inorm] kres pow karg, kpow [, inorm] Initialization inorm (optional, default=1) -- The number to divide the result (default to 1). This is especially useful if you are doing powers of a- or k- signals where samples out of range are extremely common! Performance aarg, iarg, karg -- the base. ipow, kpow -- the exponent. Use ˆ with caution in arithmetical statements, as the precedence may not be correct. New in Csound version 3.493. Examples Here is an example of the pow opcode. It uses the file pow.csd. Example of the pow opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i1 0.50000 i1 0.50007 i1 0.50014 ....... i1 14.45986 i1 14.46130 Credits Author: &nameparis; MIT, Cambridge 1995 manual_src~dfsg/opcodes/flslider.xml0000644000000000000000000002121712262561504016732 0ustar rootroot FLslider FLTK:Valuators FLslider FLslider Puts a slider into the corresponding FLTK container. Description FLslider puts a slider into the corresponding container. Syntax kout, ihandle FLslider "label", imin, imax, iexp, itype, idisp, iwidth, \ iheight, ix, iy Initialization ihandle -- a handle value (an integer number) that unequivocally references a corresponding widget. This is used by other opcodes that modify a widget's properties (see Modifying FLTK Widget Appearance). It is automatically output by FLslider and must not be set by the user label. (The user label is a double-quoted string containing some user-provided text placed near the widget.) label -- a double-quoted string containing some user-provided text, placed near the corresponding widget. imin -- minimum value of output range (corresponds to the left value for horizontal sliders, and the top value for vertical sliders). imax -- maximum value of output range (corresponds to the right value for horizontal sliders, and the bottom value for vertical sliders). The imin argument may be greater than imax argument. This has the effect of reversing the object so the larger values are in the opposite direction. This also switches which end of the filled sliders is filled. iexp -- an integer number denoting the behaviour of valuator: 0 = valuator output is linear -1 = valuator output is exponential All other positive numbers for iexp indicate the number of an existing table that is used for indexing. Linear interpolation is provided in table indexing. A negative table number suppresses interpolation. IMPORTANT! Notice that the tables used by valuators must be created with the ftgen opcode and placed in the orchestra before the corresponding valuator. They can not placed in the score. This is because tables placed in the score are created later than the initialization of the opcodes placed in the header section of the orchestra. itype -- an integer number denoting the appearance of the valuator. The itype argument can be set to the following values: 1 - shows a horizontal fill slider 2 - a vertical fill slider 3 - a horizontal engraved slider 4 - a vertical engraved slider 5 - a horizontal nice slider 6 - a vertical nice slider 7 - a horizontal up-box nice slider 8 - a vertical up-box nice slider FLslider - a horizontal fill slider (itype=1). FLslider - a horizontal fill slider (itype=1). FLslider - a horizontal engraved slider (itype=3). FLslider - a horizontal engraved slider (itype=3). FLslider - a horizontal nice slider (itype=5). FLslider - a horizontal nice slider (itype=5). You can also create "plastic" looking sliders by adding 20 to itype. idisp -- a handle value that was output from a previous instance of the FLvalue opcode to display the current value of the current valuator in the FLvalue widget itself. If the user doesn't want to use this feature that displays current values, it must be set to a negative number by the user. iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). Performance kout -- output value FLsliders are created with the minimum value by default in the left/at the top. If you want to reverse the slider, reverse the values. See the example below. Examples Here is an example of the FLslider opcode. It uses the file FLslider.csd. Example of the FLslider opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the FLslider opcode, showing the slider types and other options. It also shows the usage of FLvalue to display a widget's contents. It uses the file FLslider-2.csd. More complex example of the FLslider opcode. See Also FLcount, FLjoy, FLknob, FLroller, FLslidBnk, FLtext Credits Author: &namegabriel; New in version 4.22 February 2004. Thanks to a note from Dave Phillips, deleted the extraneous istep parameter. Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/itrirand.xml0000644000000000000000000000077612262561504016751 0ustar rootroot itrirand itrirand itrirand Deprecated. Description Deprecated as of version 3.49. Use the trirand opcode instead. manual_src~dfsg/opcodes/cpstun.xml0000644000000000000000000001201612262561504016437 0ustar rootroot cpstun Pitch Converters:Tuning Opcodes cpstun cpstun Returns micro-tuning values at k-rate. Description Returns micro-tuning values at k-rate. Syntax kcps cpstun ktrig, kindex, kfn Performance kcps -- Return value in cycles per second. ktrig -- A trigger signal used to trigger the evaluation. kindex -- An integer number denoting an index of scale. kfn -- Function table containing the parameters (numgrades, interval, basefreq, basekeymidi) and the tuning ratios. These opcodes are similar to cpstmid, but work without necessity of MIDI. cpstun works at k-rate. It allows fully customized micro-tuning scales. It requires a function table number containing the tuning ratios, and some other parameters stored in the function table itself. kindex arguments should be filled with integer numbers expressing the grade of given scale to be converted in cps. In cpstun, a new value is evaluated only when ktrig contains a non-zero value. The function table kfn should be generated by GEN02 and the first four values stored in this function are parameters that express: numgrades -- The number of grades of the micro-tuning scale. interval -- The frequency range covered before repeating the grade ratios, for example 2 for one octave, 1.5 for a fifth etcetera. basefreq -- The base frequency of the scale in cycles per second. basekey -- The integer index of the scale to which to assign basefreq unmodified. After these four values, the user can begin to insert the tuning ratios. For example, for a standard 12-grade scale with the base-frequency of 261 cps assigned to the key-number 60, the corresponding f-statement in the score to generate the table should be: ; numgrades basefreq tuning-ratios (eq.temp) ....... ; interval basekey f1 0 64 -2 12 2 261 60 1 1.059463 1.12246 1.18920 ..etc... Another example with a 24-grade scale with a base frequency of 440 assigned to the key-number 48, and a repetition interval of 1.5: ; numgrades basefreq tuning-ratios ....... ; interval basekey f1 0 64 -2 24 1.5 440 48 1 1.01 1.02 1.03 ..etc... Examples Here is an example of the cpstun opcode. It uses the file cpstun.csd. Example of the cpstun opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: i1 440.11044 See Also cpstmid, cpstuni, GEN02 Credits Example written by &namekevin;. manual_src~dfsg/opcodes/itablegpw.xml0000644000000000000000000000100712262561504017077 0ustar rootroot itablegpw itablegpw itablegpw Deprecated. Description Deprecated as of version 3.52. Use the tableigpw opcode instead. manual_src~dfsg/opcodes/trigseq.xml0000644000000000000000000001135012262561504016601 0ustar rootroot trigseq Instrument Control:Sensing and Control trigseq trigseq Accepts a trigger signal as input and outputs a group of values. Description Accepts a trigger signal as input and outputs a group of values. Syntax trigseq ktrig_in, kstart, kloop, kinitndx, kfn_values, kout1 [, kout2] [...] Performance ktrig_in -- input trigger signal kstart -- start index of looped section kloop -- end index of looped section kinitndx -- initial index Note Although kinitndx is listed as k-rate, it is in fact accessed only at init-time. So if you are using a k-rate argument, it must be assigned with init. kfn_values -- number of a table containing a sequence of groups of values kout1 -- output values kout2, ... (optional) -- more output values This opcode handles timed-sequences of groups of values stored into a table. trigseq accepts a trigger signal (ktrig_in) as input and outputs group of values (contained in the kfn_values table) each time ktrig_in assumes a non-zero value. Each time a group of values is triggered, table pointer is advanced of a number of positions corresponding to the number of group-elements, in order to point to the next group of values. The number of elements of groups is determined by the number of koutX arguments. It is possible to start the sequence from a value different than the first, by assigning to kinitndx an index different than zero (which corresponds to the first value of the table). Normally the sequence is looped, and the start and end of loop can be adjusted by modifying kstart and kloop arguments. User must be sure that values of these arguments (as well as kinitndx) correspond to valid table numbers, otherwise Csound will crash because no range-checking is implemented. It is possible to disable loop (one-shot mode) by assigning the same value both to kstart and kloop arguments. In this case, the last read element will be the one corresponding to the value of such arguments. Table can be read backward by assigning a negative kloop value. trigseq is designed to be used together with seqtime or trigger opcodes. Examples Here is an example of the trigseq opcode. It uses the file trigseq.csd. Example of the trigseq opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also seqtime, trigger Credits Author: &namegabriel; November 2002. Added a note about the kinitndx parameter, thanks to &namerasmus;. January 2003. Thanks to a note from &nameoeyvind;, I corrected the credits. New in version 4.06 manual_src~dfsg/opcodes/vcopy_i.xml0000644000000000000000000000775712262561504016613 0ustar rootroot vcopy Vectorial:Vectorial operations vcopy_i vcopy_i Copies a vector from one table to another. Description Copies a vector from one table to another. Syntax vcopy_i ifn1, ifn2, ielements [,idstoffset, isrcoffset] Initialization ifn1 - number of the table where the vectorial signal will be copied ifn2 - number of the table hosting the vectorial signal to be copied ielements - number of elements of the vector idstoffset - index offset for destination table isrcoffset - index offset for source table Performance vcopy_i copies ielements elements from ifn2 (starting from position isrcoffset) to ifn1 (starting from position idstoffset). Useful to keep old vector values, by storing them in another table. This opcode is exactly the same as vcopy but performs all the copying on the intialization pass only. Negative values for idstoffset and isrcoffset are acceptable. If idstoffset is negative, the out of range section of the vector will be discarded. If isrcoffset is negative, the out of range elements will be assumed to be 0 (i.e. the destination elements will be set to 0). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 0 (i.e. the destination vector elements will be 0). Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. All these operators (vaddv,vsubv,vmultv,vdivv,vpowv,vexp, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Note: bmscan not yet available on Canonical Csound Examples See vcopy for an example. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/exprand.xml0000644000000000000000000001044212262561504016565 0ustar rootroot x-class noise generators exprand Signal Generators:Random (Noise) Generators exprand exprand Exponential distribution random number generator (positive values only). Description Exponential distribution random number generator (positive values only). This is an x-class noise generator. Syntax ares exprand klambda ires exprand klambda kres exprand klambda Performance klambda -- lambda parameter for the exponential distribution. The probablity density function of an exponential distribution is an exponential curve, whose mean is 0.69515/lambda. For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the exprand opcode. It uses the file exprand.csd. Example of the exprand opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: i 1 time 0.00033: 4.09813 i 1 time 0.20033: 56.39567 i 1 time 0.40033: 3.23362 i 1 time 0.60033: 0.24277 i 1 time 0.80033: 13.71228 i 1 time 1.00000: 12.71885 i 1 time 1.20033: 32.36737 i 1 time 1.40033: 0.29747 i 1 time 1.60033: 4.04450 i 1 time 1.80000: 35.75676 i 1 time 2.00000: 3.69845 Seeding from current time 3034472128 i 2 time 3.00033: 6.67934 i 2 time 3.20033: 2.72431 i 2 time 3.40033: 14.51822 i 2 time 3.60000: 12.10120 i 2 time 3.80033: 1.12266 i 2 time 4.00000: 26.90772 i 2 time 4.20000: 0.43554 i 2 time 4.40033: 28.59836 i 2 time 4.60033: 27.01831 i 2 time 4.80033: 18.19911 i 2 time 5.00000: 4.45125 See Also seed, betarand, bexprnd, cauchy, gauss, linrand, pcauchy, poisson, trirand, unirand, weibull Credits Author: &nameparis; MIT, Cambridge 1995 manual_src~dfsg/opcodes/moog.xml0000644000000000000000000000654512262561504016076 0ustar rootroot moog Signal Generators:Models and Emulations moog moog An emulation of a mini-Moog synthesizer. Description An emulation of a mini-Moog synthesizer. Syntax ares moog kamp, kfreq, kfiltq, kfiltrate, kvibf, kvamp, iafn, iwfn, ivfn Initialization iafn, iwfn, ivfn -- three table numbers containing the attack waveform (unlooped), the main looping wave form, and the vibrato waveform. The files mandpluk.aiff and impuls20.aiff are suitable for the first two, and a sine wave for the last. Note The files mandpluk.aiff and impuls20.aiff are also available at ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. kfiltq -- Q of the filter, in the range 0.8 to 0.9 kfiltrate -- rate control for the filter in the range 0 to 0.0002 kvibf -- frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp -- amplitude of the vibrato Examples Here is an example of the moog opcode. It uses the file moog.csd, mandpluk.aiff, and impuls20.aiff. Example of the moog opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/STKBlowBotl.xml0000644000000000000000000000666212262561504017243 0ustar rootroot STKBlowBotl Signal Generators:STK Opcodes STKBlowBotl STKBlowBotl STKBlowBotl uses a helmholtz resonator (biquad filter) with a polynomial jet excitation. Description This opcode implements a helmholtz resonator (biquad filter) with a polynomial jet excitation (a la Cook). Syntax asignal STKBlowBotl ifrequency, iamplitude, [knoise, kv1[, klfo, kv2[, klfodepth, kv3[, kvol, kv4]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance knoise -- controller 4, gain of noise. Value range of kv1 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv2 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv3 is 0-127. kvol -- controller 128, volume. Value range of kv4 is 0-127. Notes The code for this opcode is taken directly from the BlowBotl class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKBlowBotl opcode. It uses the file STKBlowBotl.csd. Example of the STKBlowBotl opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/STKRhodey.xml0000644000000000000000000000756312262561504016752 0ustar rootroot STKRhodey Signal Generators:STK Opcodes STKRhodey STKRhodey STK Fender Rhodes-like electric piano FM synthesis instrument. Description STK Fender Rhodes-like electric piano FM synthesis instrument. This opcode implements an instrument based on two simple FM Pairs summed together, also referred to as algorithm 5 of the Yamaha TX81Z. It simulates the sound of a Rhodes electric piano, and some related sounds. Syntax asignal STKRhodey ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kmod -- controller 2, modulator index 1. Value range of kv1 is 0-127. kcross -- controller 4, crossfade of outputs. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kadsr -- controller 128, ADSR 2 and 4 target. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the Rhodey class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKRhodey opcode. It uses the file STKRhodey.csd. Example of the STKRhodey opcode. See Also STKWurley. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/rtclock.xml0000644000000000000000000000345712262561504016575 0ustar rootroot rtclock Instrument Control:Time Reading rtclock rtclock Read the real time clock from the operating system. Description Read the real-time clock from the operating system. Syntax ires rtclock kres rtclock Performance Read the real-time clock from operating system. Under Windows, this changes only once per second. Under GNU/Linux, it ticks every microsecond. Performance under other systems varies. Examples Here is an example of the rtclock opcode. It uses the file rtclock.csd. Example of the rtclock opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; New in version 4.10 manual_src~dfsg/opcodes/prepiano.xml0000644000000000000000000001145112262561504016742 0ustar rootroot prepiano Signal Generators:Models and Emulations prepiano prepiano Creates a tone similar to a piano string prepared in a Cageian fashion. Description Audio output is a tone similar to a piano string, prepared with a number of rubbers and rattles. The method uses a physical model developed from solving the partial differential equation. Syntax ares prepiano ifreq, iNS, iD, iK, \ iT30,iB, kbcl, kbcr, imass, ifreq, iinit, ipos, ivel, isfreq, \ isspread[, irattles, irubbers] al,ar prepiano ifreq, iNS, iD, iK, \ iT30,iB, kbcl, kbcr, imass, ifreq, iinit, ipos, ivel, isfreq, \ isspread[, irattles, irubbers] Initialization ifreq -- The base frequency of the string. iNS -- the number of strings involved. In a real piano 1, 2 or 3 strings are found in different frequency regions. iD -- the amount each string other that the first is detuned from the main frequency, measured in cents. iK -- dimensionless stiffness parameter. iT30 -- 30 db decay time in seconds. ib -- high-frequency loss parameter (keep this small). imass -- the mass of the piano hammer. ifreq -- the frequency of the natural vibrations of the hammer. iinit -- the initial position of the hammer. ipos -- position along the string that the strike occurs. ivel -- normalized strike velocity. isfreq -- scanning frequency of the reading place. isspread -- scanning frequency spread. irattles -- table number giving locations of any rattle(s). irubbers -- table number giving locations of any rubbers(s). The rattles and rubbers tables are collections of four values, preceeded by a count. In the case of a rattle the four are position, mass density ratio of rattle/string, frequency of rattle and vertical length of the rattle. For the rubber the fours are position, mass density ratio of rubber/string, frequency of rubber and the loss parameter. Performance A note is played on a piano string, with the arguments as below. kbcL -- Boundary condition at left end of string (1 is clamped, 2 pivoting and 3 free). kbcR -- Boundary condition at right end of string (1 is clamped, 2 pivoting and 3 free). Note that changing the boundary conditions during playing may lead to glitches and is made available as an experiment. Examples Here is an example of the prepiano opcode. It uses the file prepiano.csd. Example of the prepiano opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Stefan Bilbao University of Edinburgh, UK Author: &namejohn; University of Bath, Codemist Ltd. Bath, UK New in Csound version 5.05 manual_src~dfsg/opcodes/tablewa.xml0000644000000000000000000002664212262561504016554 0ustar rootroot tablewa Table Control:Read/Write Opreations tablewa tablewa Writes tables in sequential locations. Description This opcode writes to a table in sequential locations to and from an a-rate variable. Some thought is required before using it. It has at least two major, and quite different, applications which are discussed below. Syntax kstart tablewa kfn, asig, koff Performance kstart -- Where in table to read or write. kfn -- i- or k-rate number of the table to read or write. asig -- a-rate signal to read from when writing to the table. koff -- i- or k-rate offset into table. Range unlimited - see explanation at end of this section. In one application, it is intended to be used with one or with several tablera opcodes before a tablewa -- all sharing the same kstart variable. These read from and write to sequential locations in a table at audio rates, with ksmps floats being written and read each cycle. tablera starts reading from location kstart. tablewa starts writing to location kstart, and then writes to kstart with the number of the location one more than the one it last wrote. (Note that for tablewa, kstart is both an input and output variable.) If the writing index reaches the end of the table, then no further writing occurs and zero is written to kstart. For instance, if the table's length was 16 (locations 0 to 15), and ksmps was 5. Then the following steps would occur with repetitive runs of the tablewa opcode, assuming that kstart started at 0. Run Number Initial kstart Final kstart Locations Written 1 0 5 0 1 2 3 4 2 5 10 5 6 7 8 9 3 10 15 10 11 12 13 14 4 15 0 15 This is to facilitate processing table data using standard a-rate orchestra code between the tablera and tablewa opcodes. They allow all Csound k-rate operators to be used (with caution) on a-rate variables - something that would only be possible otherwise by ksmps = 1, downsamp and upsamp. Several cautions The k-rate code in the processing loop is really running at a-rate, so time dependent functions like port and oscil work faster than normal - their code is expecting to be running at k-rate. This system will produce undesirable results unless the ksmps fits within the table length. For instance a table of length 16 will accommodate 1 to 16 samples, so this example will work with ksmps = 1 to 16. Both these opcodes generate an error and deactivate the instrument if a table with length < ksmps is selected. Likewise an error occurs if kstart is below 0 or greater than the highest entry in the table - if kstart = table length. kstart is intended to contain integer values between 0 and (table length - 1). Fractional values above this should not affect operation but do not achieve anything useful. These opcodes are not interpolating, and the kstart and koff parameters always have a range of 0 to (table length - 1) - not 0 to 1 as is available in other table read/write opcodes. koff can be outside this range but it is wrapped around by the final AND operation. These opcodes are permanently in wrap mode. When koff is 0, no wrapping needs to occur, since the kstart++ index will always be within the table's normal range. koff not equal to 0 can lead to wrapping. The offset does not affect the number of read/write cycles performed, or the value written to kstart by tablewa. These opcodes cannot read or write the guardpoint. Use tablegpw to write the guardpoint after manipulations have been done with tablewa. Examples kstart = 0 lab1: atemp tablera ktabsource, kstart, 0 ; Read 5 values from table into an ; a-rate variable. atemp = log(atemp) ; Process the values using a-rate ; code. kstart tablewa ktabdest, atemp, 0 ; Write it back to the table if ktemp 0 goto lab1 ; Loop until all table locations ; have been processed. The above example shows a processing loop, which runs every k-cycle, reading each location in the table ktabsource, and writing the log of those values into the same locations of table ktabdest. This enables whole tables, parts of tables (with offsets and different control loops) and data from several tables at once to be manipulated with a-rate code and written back to another (or to the same) table. This is a bit of a fudge, but it is faster than doing it with k-rate table read and write code. Another application is: kzero = 0 kloop = 0 kzero tablewa 23, asignal, 0 ; ksmps a-rate samples written ; into locations 0 to (ksmps -1) of table 23. lab1: ktemp table kloop, 23 ; Start a loop which runs ksmps times, ; in which each cycle processes one of [ Some code to manipulate ] ; table 23's values with k-rate orchestra [ the value of ktemp. ] ; code. tablew ktemp, kloop, 23 ; Write the processed value to the table. kloop = kloop + 1 ; Increment the kloop, which is both the ; pointer into the table and the loop if kloop < ksmps goto lab1 ; counter. Keep looping until all values ; in the table have been processed. asignal tablera 23, 0, 0 ; Copy the table contents back ; to an a-rate variable. koff -- This is an offset which is added to the sum of kstart and the internal index variable which steps through the table. The result is then ANDed with the lengthmask (000 0111 for a table of length 8 - or 9 with guardpoint) and that final index is used to read or write to the table. koff can be any value. It is converted into a long using the ANSI floor() function so that -4.3 becomes -5. This is what we would want when using offsets which range above and below zero. Ideally this would be an optional variable, defaulting to 0, however with the existing Csound orchestra read code, such default parameters must be init time only. We want k-rate here, so we cannot have a default. Credits Author: Robin Whittle Australia manual_src~dfsg/opcodes/gendyx.xml0000644000000000000000000001646112262561504016431 0ustar rootroot gendyx Signal Generators:Models and Emulations gendyx gendyx Variation of the dynamic stochastic approach to waveform synthesis conceived by Iannis Xenakis. Description gendyx (gendy eXtended) is an implementation of the Génération Dynamique Stochastique (GENDYN), a dynamic stochastic approach to waveform synthesis conceived by Iannis Xenakis, using curves instead of segments. Syntax ares gendyx kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl, kcurveup, kcurvedown [, initcps] [, knum] kres gendyx kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl, kcurveup, kcurvedown [, initcps] [, knum] Initialization initcps (optional, default=12) -- max number of control points. Performance kamp -- amplitude. kampdist -- choice of probability distribution for the next perturbation of the amplitude of a control point. The valid distributions are: 0 - LINEAR 1 - CAUCHY 2 - LOGIST 3 - HYPERBCOS 4 - ARCSINE 5 - EXPON 6 - SINUS (external k-rate signal) If kampdist=6, the user can use an external k-rate signal through kadpar. kdurdist -- choice of distribution for the perturbation of the current inter control point duration. See kampdist for the valid distributions. If kdurdist=6, the user can use an external k-rate signal through kddpar. kadpar -- parameter for the kampdist distribution. Should be in the range of 0.0001 to 1. kddpar -- parameter for the kdurdist distribution. Should be in the range of 0.0001 to 1. kminfreq -- minimum allowed frequency of oscillation. kmaxfreq -- maximum allowed frequency of oscillation. kampscl -- multiplier for the distribution's delta value for amplitude (1.0 is full range). kdurscl -- multiplier for the distribution's delta value for duration. kcurveup -- controls the curve for the increasing amplitudes between two points; it has to be non negative. 0: step function (like a sample & hold) <1: concave 1: linear (like gendy) >1: convex kcurvedown -- controls the curve for the decreasing amplitudes between two points; it has to be non negative. 0: step function <1: convex 1: linear >1: concave knum (optional, default=initcps) -- current number of utilized control points. The waveform is generated by knum - 1 curves and is repeated in the time. The vertexes (control points) are moved according to a stochastic action and they are limited within the boundaries of a mirror formed by an amplitude barrier and a time barrier. [Extract of a waveform generated with gendyx.] Extract of a waveform generated with gendyx. Examples Here is an example of the gendyx opcode. It uses the file gendyx.csd. Example of the gendyx opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an example of the gendyx opcode with some modulations. It uses the file gendyx-2.csd. Example of the gendyx opcode with some modulations. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. References Formalized Music (1992, Stuyvesant, NY: Pendragon Press), pp. 246 - 254, 289 - 322. See Also gendy gendyc Credits Variation of the Nick Collins's Gendy1 ugen (SuperCollider) Author: &nametito; January 2012 New in Csound version 5.16 manual_src~dfsg/opcodes/vdelay.xml0000644000000000000000000000570112262561504016412 0ustar rootroot vdelay Signal Modifiers:Delay vdelay vdelay An interpolating variable time delay. Description This is an interpolating variable time delay, it is not very different from the existing implementation (deltapi), it is only easier to use. Syntax ares vdelay asig, adel, imaxdel [, iskip] Initialization imaxdel -- Maximum value of delay in milliseconds. If adel gains a value greater than imaxdel it is folded around imaxdel. This should not happen. iskip -- Skip initialization if present and non­zero Performance With this unit generator it is possible to do Doppler effects or chorusing and flanging. asig -- Input signal. adel -- Current value of delay in milliseconds. Note that linear functions have no pitch change effects. Fast changing values of adel will cause discontinuities in the waveform resulting noise. Examples Here is an example of the vdelay opcode. It uses the file vdelay.csd. Example of the vdelay opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Two important points here. First, the delay time must be always positive. And second, even though the delay time can be controlled in k-rate, it is not advised to do so, since sudden time changes will create clicks. See Also vdelay3 Credits Author: &nameparis; MIT, Cambridge 1995 manual_src~dfsg/opcodes/outs2.xml0000644000000000000000000000615512262561504016206 0ustar rootroot outs2 Signal I/O:Signal Output outs2 outs2 Writes samples to stereo channel 2 of an external device or stream. Description Writes samples to stereo channel 2 of an external device or stream. Syntax outs2 asig Performance Sends audio samples to an accumulating output buffer (created at the beginning of performance) which serves to collect the output of all active instruments before the sound is written to disk. There can be any number of these output units in an instrument. The type (mono, stereo, quad, hex, or oct) should agree with nchnls. But as of version 3.50, Csound will attempt to change an incorrect opcode to agree with the nchnls statement. Opcodes can be chosen to direct sound to any particular channel: outs1 sends to stereo channel 1, outq3 to quad channel 3, etc. Examples Here is an example of the outs2 opcode. It uses the file outs2.csd. Example of the outs2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also out, outh, outo, outq, outq1, outq2, outq3, outq4, outs, outs1, soundout Credits Author: &namebarry;, &namematt;/&namemike; MIT, Mills College 1993-1997 manual_src~dfsg/opcodes/compilestr.xml0000644000000000000000000000516212262561504017310 0ustar rootroot compilestr Instrument Control:Compilation compilestr compilestr compiles a new orchestra passed in as an ASCII string Description Compilestr will compile one or more instruments at init time, which will be added to the running engine. In case of existing instrument numbers or names, these will be replaced, but any instance still running of the old instrument definition will still perform until it terminates. Only new instances will use the new definition. Multi-line strings are accepted, using {{ }} to enclose the string. Syntax ires compilestr Sorch Initialization Sorch -- a string (in double-quotes or enclosed by {{ }}) containing one or more instruments. ires -- returns 0 if compilation was successful, or -1 if not. Examples Here is an example of the compilestr opcode. It uses the file compilestr.csd. Example of the compilestr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also event, event_i, schedule, schedwhen, schedkwhen, schedkwhennamed Credits Author: &namevictor;, 2013 manual_src~dfsg/opcodes/hrtfearly.xml0000644000000000000000000001675612262561504017142 0ustar rootroot hrtfearly Signal Modifiers:Panning and Spatialization hrtfearly hrtfearly Generates 3D binaural audio with high-fidelity early reflections in a parametric room using a Phase Truncation algorithm. Description This opcode essentially nests the hrtfmove opcode in an image model for a user-definable shoebox-shaped room. A default room can be selected, or advanced room parameters can be used. Room surfaces can be controlled with high and low-frequency absorption coefficients and gain factors of a three-band equaliser. Although valid as a stand alone opcode, hrtfearly is designed to work with hrtfreverb to provide spatially accurate, dynamic binaural reverberation. A number of sources can be processed dynamically using a number of hrtfearly instances. All can then be processed with one instance of hrtfreverb. Syntax aleft, aright, irt60low, irt60high, imfp hrtfearly asrc, ksrcx, ksrcy, ksrcz, klstnrx, klstnry, klstnrz, \ ifilel, ifiler, idefroom [,ifade, isr, iorder, ithreed, kheadrot, iroomx, iroomy, iroomz, iwallhigh, \ iwalllow, iwallgain1, iwallgain2, iwallgain3, ifloorhigh, ifloorlow, ifloorgain1, ifloorgain2, \ ifloorgain3, iceilinghigh, iceilinglow, iceilinggain1, iceilinggain2, iceilinggain3] Initialization ifilel - left HRTF spectral data file. ifiler - right HRTF spectral data file. Spectral datafiles (based on the MIT HRTF database) are available in 3 different sampling rates: 44.1, 48 and 96 kHz and are labelled accordingly. Input and processing sr should match datafile sr. Files should be in the current directory or the SADIR (see Environment Variables). HRTF Data files for use with hrtfmove, hrtfmove2, hrtfstat, hrtfearly, hrtfreverb were updated for Csound 5.15 and later (the code was updated and is more efficient). Old datafiles are now deprecated. idefroom - default room, medium (1: 10*10*3), small (2: 3*4*3) or large (3: 20*25*7). Wall details (high coef, low coef, gain1, gain2, gain3): .3, .1, .75, .95, .9. Floor: .6, .1, .95, .6, .35. Ceiling: .2, .1, 1, 1, 1. If 0 is entered, optional room parameters will be read. ifade - optional, number of processing buffers for phase change crossfade (default 8). Legal range is 1-24. See hrtfmove. isr - optional, default 44.1kHz, legal values: 44100, 48000 and 96000. iorder - optional, order of images processed: higher order: more reflections. Defaults to 1, legal range: 0-4. ithreed - optional, process image sources in three dimensions (1) or two (0: default). iroomx - optional, x room size in metres, will be read if no valid default room is entered (all below parameters behave similarly). Minimum room size is 2*2*2. iroomy - optional, y room size. iroomz - optional, z room size. iwallhigh - optional, high frequency wall absorption coefficient (all 4 walls are assumed identical). Absorption coefficients will affect reverb time output. iwalllow - optional, low frequency wall absorption coefficient. iwallgain1 - optional, gain on filter centred at 250 Hz (all filters have a Q implying 4 octaves). iwallgain2 - optional, as above, centred on 1000 Hz. iwallgain3 - optional, as above, centred on 4000 Hz. ifloorhigh, ifloorlow, ifloorgain1, ifloorgain2, ifloorgain3 - as above for floor. iceilinghigh, iceilinglow, iceilinggain1, iceilinggain2, iceilinggain3 - as above for ceiling. Performance ksrcx source x location, must be 10 cm inside room. Also, near-field HRTFs are not processed, so source will not change spatially within a 45 cm radius of the listener. These restrictions also apply to location parameters below. ksrcy source y location. ksrcz source z location. klstnrx, klstnry, klstnrz listener location, as above. kheadrot - optional, angular value for head rotation. asrc - Input/source signal. Output irt60low - suggested low frequency reverb time for later binaural reverb. irt60high - as above, for high frequency. imfp - mean free path of room, to be used with later reverb. Examples Here is an example of the hrtfearly and hrtfreverb opcodes. It uses the file hrtfearly.csd. Example of the htrfearly opcode. See Also hrtfreverb hrtfmove, hrtfmove2, hrtfstat, hrtfer. Credits Author: Brian Carty Maynooth 2011 manual_src~dfsg/opcodes/fprints.xml0000644000000000000000000000563612262561504016622 0ustar rootroot fprints Signal I/O:File I/O fprints fprints Similar to prints but prints to a file. Description Similar to prints but prints to a file. Syntax fprints "filename", "string" [, ival1] [, ival2] [...] Initialization "filename" -- name of the output file. "string" -- the text string to be printed. Can be up to 8192 characters and must be in double quotes. ival1, ival2, ... (optional) -- The i-rate values to be printed. These are specified in string with the standard C value specifier (%f, %d, etc.) in the order given. Performance fprints is similar to the prints opcode except it outputs to a file. For more information about output formatting, please look at printks's documentation. Examples Here is an example of the fprints opcode. It uses the file fprints.csd. Example of the fprints opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This example will generate a file called my.sco. It should contain a line like this: ;Generated score by ma++ See Also prints Credits Author: &namematt; January 2003 manual_src~dfsg/opcodes/vstaudio.xml0000644000000000000000000000630212262561504016762 0ustar rootroot vstaudio Plugin Hosting:VST vstaudio, vstaudiog vstaudio VST audio output. Syntax aout1,aout2 vstaudio instance, [ain1, ain2] aout1,aout2 vstaudiog instance, [ain1, ain2] Description vstaudio and vstaudiog are used for sending and receiving audio from a VST plugin. vstaudio is used within an instrument definition that contains a vstmidiout or vstnote opcode. It outputs audio for only that one instrument. Any audio remaining in the plugin after the end of the note, for example a reverb tail, will be cut off and should be dealt with using a damping envelope. vstaudiog (vstaudio global) is used in a separate instrument to process audio from any number of VST notes or MIDI events that share the same VST plugin instance (instance). The vstaudiog instrument must be numbered higher than all the instruments receiving notes or MIDI data, and the note controlling the vstplug instrument must have an indefinite duration, or at least a duration as long as the VST plugin is active. Initialization instance - the number which identifies the plugin, to be passed to other vst4cs opcodes. Performance aout1, aout2 - the audio output received from the plugin. ain1, ain2 - the audio input sent to the plugin. Examples Here is an example of the use of the vstaudio opcode. It uses the file vst4cs.csd. Example of the <emphasis>vstaudio</emphasis> opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits By: &nameandres; and &namemichael; Uses code from Hermann Seib's VSTHost and Thomas Grill's vst~ object. VST is a trademark of Steinberg Media Technologies GmbH. VST Plug-In Technology by Steinberg. manual_src~dfsg/opcodes/vbap16move.xml0000644000000000000000000001241412262561504017113 0ustar rootroot vbap16move Signal Modifiers:Panning and Spatialization vbap16move vbap16move Distribute an audio signal among 16 channels with moving virtual sources. Description Distribute an audio signal among 16 channels with moving virtual sources. Syntax ar1, ..., ar16 vbap16move asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] Initialization idur -- the duration over which the movement takes place. ispread -- spreading of the virtual source (range 0 - 100). If value is zero, conventional amplitude panning is used. When ispread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. ifldnum -- number of fields (absolute value must be 2 or larger). If ifldnum is positive, the virtual source movement is a polyline specified by given directions. Each transition is performed in an equal time interval. If ifldnum is negative, specified angular velocities are applied to the virtual source during specified relative time intervals (see below). ifld1, ifld2, ... -- azimuth angles or angular velocities, and relative durations of movement phases. Performance asig -- audio signal to be panned vbap16move allows the use of moving virtual sources. If ifldnum is positive, the fields represent directions of virtual sources and equal times, iazi1, [iele1,] iazi2, [iele2,], etc. The position of the virtual source is interpolated between directions starting from the first direction and ending at the last. Each interval is interpolated in time that is fraction total_time / number_of_intervals of the duration of the sound event. If ifldnum is negative, the fields represent angular velocities and equal times. The first field is, however, the starting direction, iazi1, [iele1,] iazi_vel1, [iele_vel1,] iazi_vel2, [iele_vel2,] .... Each velocity is applied to the note that is fraction total_time / number_of_velocities of the duration of the sound event. If the elevation of the virtual source becomes greater than 90 degrees or less than 0 degrees, the polarity of angular velocity is changed. Thus the elevational angular velocity produces a virtual source that moves up and down between 0 and 90 degrees. Please note that all vbap panning opcodes require the vbap system to be initialized using vbaplsinit. Examples See the entry for vbap8move for an example of usage of the vbapXmove opcodes. Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. See Also vbap16, vbap4, vbap4move, vbap8, vbap8move, vbaplsinit, vbapz, vbapzmove, vbapzmove Credits Author: Ville Pulkki Sibelius Academy Computer Music Studio Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Helsinki, Finland May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/pwd.xml0000644000000000000000000000345312262561504015722 0ustar rootroot pwd Miscellaneous pwd pwd Asks the underlying operating system for the current directory name as a string. Description pwd call the operating system to determine the current directory (folder). pwd runs at i-time only. Syntax Sres pwd Performance Sres -- the returned string. Example Here is an example of the pwd opcode. It uses the file pwd.csd. Example of the pwd opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; July 2012 New in version 5.18 manual_src~dfsg/opcodes/midinoteoncps.xml0000644000000000000000000001115312262561504017777 0ustar rootroot midinoteoncps Real-time MIDI:MIDI/Score Interoperability midinoteoncps midinoteoncps Gets a MIDI note number as a cycles-per-second frequency. Description midinoteoncps is designed to simplify writing instruments that can be used interchangeably for either score or MIDI input, and to make it easier to adapt instruments originally written for score input to work with MIDI input. In general, it should be possible to write instrument definitions that work identically with both scores and MIDI, including both MIDI files and real-time MIDI input, without using any conditional statements, and that take full advantage of MIDI voice messages. Note that correlating Csound instruments with MIDI channel numbers is done using the massign opcode for real-time performance,. For file-driven performance, instrument numbers default to MIDI channel number + 1, but the defaults are overridden by any MIDI program change messages in the file. Syntax midinoteoncps xcps, xvelocity Performance xcps -- returns MIDI key translated to cycles per second during MIDI activation, remains unchanged otherwise. xvelocity -- returns MIDI velocity during MIDI activation, remains unchanged otherwise. If the instrument was activated by MIDI input, the opcode overwrites the values of xcps and xvelocity with the corresponding values from MIDI input. If the instrument was NOT activated by MIDI input, the values of xcps and xvelocity remain unchanged. This enables score p-fields to receive MIDI input data during MIDI activation, and score values otherwise. Adapting a score-activated Csound instrument. See the MIDI interop opcodes section for details on adapting score driven instruments for MIDI or vice-versa. Examples Here is an example of the midinoteoncps opcode. It uses the file midinoteoncps.csd. Example of the midinoteoncps opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i1 261.62561 i1 440.00006 See Also midichannelaftertouch, midicontrolchange, mididefault, midinoteoff, midinoteonkey, midinoteonoct, midinoteonpch, midipitchbend, midipolyaftertouch, midiprogramchange Credits Author: Michael Gogins New in version 4.20 manual_src~dfsg/opcodes/fof2.xml0000644000000000000000000001745512262561504015773 0ustar rootroot fof2 Signal Generators:Granular Synthesis fof2 fof2 Produces sinusoid bursts including k-rate incremental indexing with each successive burst. Description Audio output is a succession of sinusoid bursts initiated at frequency xfund with a spectral peak at xform. For xfund above 25 Hz these bursts produce a speech-like formant with spectral characteristics determined by the k-input parameters. For lower fundamentals this generator provides a special form of granular synthesis. fof2 implements k-rate incremental indexing into ifna function with each successive burst. Syntax ares fof2 xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, \ ifna, ifnb, itotdur, kphs, kgliss [, iskip] Initialization iolaps -- number of preallocated spaces needed to hold overlapping burst data. Overlaps are frequency dependent, and the space required depends on the maximum value of xfund * kdur. Can be over-estimated at no computation cost. Uses less than 50 bytes of memory per iolap. ifna, ifnb -- table numbers of two stored functions. The first is a sine table for sineburst synthesis (size of at least 4096 recommended). The second is a rise shape, used forwards and backwards to shape the sineburst rise and decay; this may be linear (GEN07) or perhaps a sigmoid (GEN19). itotdur -- total time during which this fof will be active. Normally set to p3. No new sineburst is created if it cannot complete its kdur within the remaining itotdur. iskip (optional, default=0) -- If non-zero, skip initialization (allows legato use). Performance xamp -- peak amplitude of each sineburst, observed at the true end of its rise pattern. The rise may exceed this value given a large bandwidth (say, Q < 10) and/or when the bursts are overlapping. xfund -- the fundamental frequency (in Hertz) of the impulses that create new sinebursts. xform -- the formant frequency, i.e. freq of the sinusoid burst induced by each xfund impulse. This frequency can be fixed for each burst or can vary continuously (see ifmode). koct -- octaviation index, normally zero. If greater than zero, lowers the effective xfund frequency by attenuating odd-numbered sinebursts. Whole numbers are full octaves, fractions transitional. kband -- the formant bandwidth (at -6dB), expressed in Hz. The bandwidth determines the rate of exponential decay throughout the sineburst, before the enveloping described below is applied. kris, kdur, kdec -- rise, overall duration, and decay times (in seconds) of the sinusoid burst. These values apply an enveloped duration to each burst, in similar fashion to a Csound linen generator but with rise and decay shapes derived from the ifnb input. kris inversely determines the skirtwidth (at -40 dB) of the induced formant region. kdur affects the density of sineburst overlaps, and thus the speed of computation. Typical values for vocal imitation are .003,.02,.007. kphs -- allows k-rate indexing of function table ifna with each successive burst, making it suitable for time-warping applications. Values of kphs are normalized from 0 to 1, 1 being the end of the function table ifna. kgliss -- sets the end pitch of each grain relative to the initial pitch, in octaves. Thus kgliss = 2 means that the grain ends two octaves above its initial pitch, while kgliss = -3/4 has the grain ending a major sixth below. Each 1/12 added to kgliss raises the ending frequency one half-step. If you want no glissando, set kgliss to 0. Csound's fof generator is loosely based on Michael Clarke's C-coding of IRCAM's CHANT program (Xavier Rodet et al.). Each fof produces a single formant, and the output of four or more of these can be summed to produce a rich vocal imitation. fof synthesis is a special form of granular synthesis, and this implementation aids transformation between vocal imitation and granular textures. Computation speed depends on kdur, xfund, and the density of any overlaps. The ending frequency of any grain is equal to kform * (2 ^ kgliss), so setting kgliss too high may result in aliasing. For example, kform = 3000 and kgliss = 3 places the ending frequency over the Nyquist if sr = 44100. It is a good idea to scale kgliss accordingly. Examples Here is an example of the fof2 opcode. It uses the file fof2.csd. Example of the fof2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the fof2 opcode, which generates vowel tones using formants generated by fof2 coinciding with values from the Formant Values appendix. It uses the file fof2-2.csd. Example of the fof2 opcode to produce vowel sounds. See Also fof Credits Author: &namerasmus; fof2 is a modification of fof by &namerasmus; New in Csound 3.45 manual_src~dfsg/opcodes/if.xml0000644000000000000000000001460112262561504015523 0ustar rootroot if Instrument Control:Program Flow Control if if Branches conditionally at initialization or during performance time. Description if...igoto -- conditional branch at initialization time, depending on the truth value of the logical expression ia R ib. The branch is taken only if the result is true. if...kgoto -- conditional branch during performance time, depending on the truth value of the logical expression ka R kb. The branch is taken only if the result is true. if...goto -- combination of the above. Condition tested on every pass. if...then -- allows the ability to specify conditional if/else/endif blocks. All if...then blocks must end with an endif statement. elseif and else statements are optional. Any number of elseif statements are allowed. Only one else statement may occur and it must be the last conditional statement before the endif statement. Nested if...then blocks are allowed. Note Note that if the condition uses a k-rate variable (for instance, if kval > 0), the if...goto or if...then statement will be ignored during the i-time pass. This allows for opcode initialization, even if the k-rate variable has already been assigned an appropriate value by an earlier init statement. Syntax if ia R ib igoto label if ka R kb kgoto label if xa R xb goto label if xa R xb then where label is in the same instrument block and is not an expression, and where R is one of the Relational operators (<, =, <=, ==, !=) (and = for convenience, see also under Conditional Values). If goto or then is used instead of kgoto or igoto, the behavior is determined by the type being compared. If the comparison used k-type variables, kgoto is used and viceversa. If/then/goto statements cannot do audio-type comparisons. You can't put a-type variables in the comparison expressions for these opcodes. The reason for this is that audio variables are actually vectors, which can't be compared in the same way as scalars. If you need to compare individua audio samples, use kr = 1 or Comparators Examples Here is an example of the if...igoto combination. It uses the file igoto.csd. Example of the if...igoto combination. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: iparam = 0.000 instr 1: ifreq = 440.000 instr 1: iparam = 1.000 instr 1: ifreq = 880.000 Here is an example of the if...kgoto combination. It uses the file kgoto.csd. Example of the if...kgoto combination. Its output should include lines like this: kval = 0.000000, kfreq = 440.000000 kval = 0.999732, kfreq = 440.000000 kval = 1.999639, kfreq = 880.000000 Examples Here is an example of the if...then combo. It uses the file ifthen.csd. Example of the if...then combo. See Also elseif, else, endif, goto, igoto, kgoto, tigoto, timout Credits Examples written by &namekevin;. Added a note by Jim Aikin. February 2004. Added a note by &namematt;. manual_src~dfsg/opcodes/lowres.xml0000644000000000000000000000543612262561504016446 0ustar rootroot lowres Signal Modifiers:Standard Filters:Resonant lowres lowres Another resonant lowpass filter. Description lowres is a resonant lowpass filter. Syntax ares lowres asig, kcutoff, kresonance [, iskip] Initialization iskip -- initial disposition of internal data space. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance asig -- input signal kcutoff -- filter cutoff frequency point kresonance -- resonance amount lowres is a resonant lowpass filter derived from a &namehans; orchestra. This implementation is much faster than implementing it in Csound language, and it allows kr lower than sr. kcutoff is not in Hz and kresonance is not in dB, so experiment for the finding best results. Examples Here is an example of the lowres opcode. It uses the file lowres.csd. Example of the lowres opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also lowresx Credits Author: &namegabriel; (adapted by &namejohn;) Italy New in Csound version 3.49 manual_src~dfsg/opcodes/oscil1i.xml0000644000000000000000000000565112262561504016475 0ustar rootroot oscil1i Signal Generators:Table Access oscil1i oscil1i Accesses table values by incremental sampling with linear interpolation. Description Accesses table values by incremental sampling with linear interpolation. Syntax kres oscil1i idel, kamp, idur [, ifn] Initialization idel -- delay in seconds before oscil1i incremental sampling begins. idur -- duration in seconds to sample through the oscil1i table just once. A zero or negative value will cause all initialization to be skipped. ifn -- (optional) function table number. oscil1i requires the extended guard point. The default value is -1 indicating a sine wave. Performance kamp -- amplitude factor oscil1i is an interpolating unit in which the fractional part of index is used to interpolate between adjacent table entries. The smoothness gained by interpolation is at some small cost in execution time (see also oscili, etc.), but the interpolating and non-interpolating units are otherwise interchangeable. Examples Here is an example of the oscil1i opcode. It uses the file oscil1i.csd. Example of the oscil1i opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also table, tablei, table3, oscil1, osciln manual_src~dfsg/opcodes/tableseg.xml0000644000000000000000000000626012262561504016715 0ustar rootroot tableseg Spectral Processing:STFT tableseg tableseg Creates a new function table by making linear segments between values in stored function tables. Description tableseg is like linseg but interpolate between values in a stored function tables. The result is a new function table passed internally to any following vpvoc which occurs before a subsequent tableseg (much like lpread/lpreson pairs work). The uses of these are described below under vpvoc. Syntax tableseg ifn1, idur1, ifn2 [, idur2] [, ifn3] [...] Initialization ifn1, ifn2, ifn3, etc. -- function table numbers. ifn1, ifn2, and so on, must be the same size. idur1, idur2, etc. -- durations during which interpolation from one table to the next will take place. Examples Here is an example of the tableseg opcode. It uses the file tableseg.csd. Example of the tableseg opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvbufread, pvcross, pvinterp, pvread, tablexseg Credits Author: Richard Karpen Seattle, Wash 1997 New in version 3.44 manual_src~dfsg/opcodes/kill.xml0000644000000000000000000000356512262561504016067 0ustar rootroot kill Instrument Control:Duration Control kill kill Stops a given instrument instance. Description Given a valid instrument instance, this opcode stops it. Syntax kill kinstance Performance kinstance -- a valid instrument instance. Whether of finite duration or held, the instance currently being performed is immediately removed from the active instance list. No other instances are affected. Examples The following example uses the kill opcode. It uses the file kill.csd. Example of the kill opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also ihold, turnon manual_src~dfsg/opcodes/cngoto.xml0000644000000000000000000000531412262561504016417 0ustar rootroot cngoto Instrument Control:Program Flow Control cngoto cngoto Transfers control on every pass when a condition is not true. Description Transfers control on every pass when the condition is not true. Syntax cngoto condition, label where label is in the same instrument block and is not an expression, and where condition uses one of the Relational operators (<, =, <=, ==, !=) (and = for convenience, see also under Conditional Values). Examples Here is an example of the cngoto opcode. It uses the file cngoto.csd. Example of the cngoto opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: kval = 0.000000, kfreq = 880.000000 kval = 0.999732, kfreq = 880.000000 kval = 1.999639, kfreq = 440.000000 See Also cggoto, cigoto, ckgoto, goto, if, igoto, tigoto, timout Credits Example written by &namekevin;. New in version 4.21 manual_src~dfsg/opcodes/logcurve.xml0000644000000000000000000000547512262561504016764 0ustar rootroot logcurve Signal Generators:Linear and Exponential Generators logcurve logcurve This opcode implements a formula for generating a normalised logarithmic curve in range 0 - 1. It is based on the Max / MSP work of Eric Singer (c) 1994. Description Generates a logarithmic curve in range 0 to 1 of arbitrary steepness. Steepness index equal to or lower than 1.0 will result in Not-a-Number errors and cause unstable behavior. The formula used to calculate the curve is: log(x * (y-1)+1) / (log(y) where x is equal to kindex and y is equal to ksteepness. Syntax kout logcurve kindex, ksteepness Performance kindex -- Index value. Expected range 0 to 1. ksteepness -- Steepness of the generated curve. Values closer to 1.0 result in a straighter line while larger values steepen the curve. kout -- Scaled output. Examples Here is an example of the logcurve opcode. It uses the file logcurve.csd. Example of the logcurve opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also scale, gainslider, expcurve Credits Author: David Akbari October 2006 manual_src~dfsg/opcodes/zakinit.xml0000644000000000000000000000547612262561504016610 0ustar rootroot zakinit Zak Patch System zakinit zakinit Establishes zak space. Description Establishes zak space. Must be called only once. Syntax zakinit isizea, isizek Initialization isizea -- the number of audio rate locations for a-rate patching. Each location is actually an array which is ksmps long. isizek -- the number of locations to reserve for floats in the zk space. These can be written and read at i- and k-rates. Performance At least one location each is always allocated for both za and zk spaces. There can be thousands or tens of thousands za and zk ranges, but most pieces probably only need a few dozen for patching signals. These patching locations are referred to by number in the other zak opcodes. To run zakinit only once, put it outside any instrument definition, in the orchestra file header, after sr, kr, ksmps, and nchnls. Zak channels count from 0, so if you define 1 channel, the only valid channel is channel 0. Examples Here is an example of the zakinit opcode. It uses the file zakinit.csd. Example of the zakinit opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Robin Whittle Australia May 1997 New in version 3.45 Example written by &namekevin;. manual_src~dfsg/opcodes/igoto.xml0000644000000000000000000000466112262561504016253 0ustar rootroot igoto Instrument Control:Program Flow Control igoto igoto Transfer control during the i-time pass. Description During the i-time pass only, unconditionally transfer control to the statement labeled by label. Syntax igoto label where label is in the same instrument block and is not an expression. Examples Here is an example of the igoto opcode. It uses the file igoto.csd. Example of the igoto opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: iparam = 0.000 instr 1: ifreq = 440.000 instr 1: iparam = 1.000 instr 1: ifreq = 880.000 See Also cggoto, cigoto, ckgoto, goto, if, kgoto, rigoto, tigoto, timout Credits Example written by &namekevin;. manual_src~dfsg/opcodes/octpch.xml0000644000000000000000000001344212262561504016407 0ustar rootroot octpch Pitch Converters:Functions octpch octpch Converts a pitch-class value to octave-point-decimal. Description Converts a pitch-class value to octave-point-decimal. Syntax octpch (pch) (init- or control-rate args only) where the argument within the parentheses may be a further expression. Performance octpch and its related opcodes are really value converters with a special function of manipulating pitch data. Data concerning pitch and frequency can exist in any of the following forms: Pitch and Frequency Values Name Abbreviation octave point pitch-class (8ve.pc) pch octave point decimal oct cycles per second cps Midi note number (0-127) midinn
The first two forms consist of a whole number, representing octave registration, followed by a specially interpreted fractional part. For pch, the fraction is read as two decimal digits representing the 12 equal-tempered pitch classes from .00 for C to .11 for B. For oct, the fraction is interpreted as a true decimal fractional part of an octave. The two fractional forms are thus related by the factor 100/12. In both forms, the fraction is preceded by a whole number octave index such that 8.00 represents Middle C, 9.00 the C above, etc. Midi note number values range between 0 and 127 (inclusively) with 60 representing Middle C, and are usually whole numbers. Thus A440 can be represented alternatively by 440 (cps), 69 (midinn), 8.09 (pch), or 8.75 (oct). Microtonal divisions of the pch semitone can be encoded by using more than two decimal places. The mnemonics of the pitch conversion units are derived from morphemes of the forms involved, the second morpheme describing the source and the first morpheme the object (result). Thus cpspch(8.09) will convert the pitch argument 8.09 to its cps (or Hertz) equivalent, giving the value of 440. Since the argument is constant over the duration of the note, this conversion will take place at i-time, before any samples for the current note are produced. By contrast, the conversion cpsoct(8.75 + k1) which gives the value of A440 transposed by the octave interval k1. The calculation will be repeated every k-period since that is the rate at which k1 varies. Note The conversion from pch, oct, or midinn into cps is not a linear operation but involves an exponential process that could be time-consuming when executed repeatedly. Csound now uses a built-in table lookup to do this efficiently, even at audio rates. Because the table index is truncated without interpolation, pitch resolution when using one of these opcodes is limited to 8192 discrete and equal divisions of the octave, and some pitches of the standard 12-tone equally-tempered scale are very slightly mistuned (by at most 0.15 cents). Examples Here is an example of the octpch opcode. It uses the file octpch.csd. Example of the octpch opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: ioct = 8.750 See Also cpsoct, cpspch, octcps, pchoct, cpsmidinn, octmidinn, pchmidinn Credits Example written by &namekevin;. manual_src~dfsg/opcodes/ilinrand.xml0000644000000000000000000000077612262561504016735 0ustar rootroot ilinrand ilinrand ilinrand Deprecated. Description Deprecated as of version 3.49. Use the linrand opcode instead. manual_src~dfsg/opcodes/midglobal.xml0000644000000000000000000000556312262561504017066 0ustar rootroot midglobal Remote Opcodes midglobal midglobal An opcode which can be used to implement a remote midi orchestra. This opcode will broadcast the midi events to all the machines involved in the remote concert. Description With the midremot and midglobal opcodes you are able to perform instruments on remote machines and control them from a master machine. The remote opcodes are implemented using the master/client model. All the machines involved contain the same orchestra but only the master machine contains the information of the midi score. During the performance the master machine sends the midi events to the clients. The midglobal opcode sends the events to all the machines involved in the remote concert. These machines are determined by the midremot definitions made above the midglobal command. To send events to only one machine use midremot. Syntax midglobal isource, instrnum [,instrnum...] Initialization isource -- a string that is the intended host computer (e.g. 192.168.0.100). This is the source host which generates the events of the given instrument(s) and sends it to all the machines involved in the remote concert. instrnum -- a list of instrument numbers which will be played on the destination machines Examples See the entry for midremot for an example of usage. See also insglobal, insremot, midremot, remoteport Credits Author: Simon Schampijer 2006 New in version 5.03 manual_src~dfsg/opcodes/ampmidid.xml0000644000000000000000000000736412262561504016721 0ustar rootroot ampmidid Real-time MIDI:Converters ampmidid ampmidid Musically map MIDI velocity to peak amplitude within a specified dynamic range in decibels. Description Musically map MIDI velocity to peak amplitude within a specified dynamic range in decibels. Syntax iamplitude ampmidid ivelocity, idecibels kamplitude ampmidid kvelocity, idecibels Initialization iamplitude -- Amplitude. ivelocity -- MIDI velocity number, ranging from 0 through 127. idecibels -- Desired dynamic range in decibels. Performance kamplitude -- Amplitude. kvelocity -- MIDI velocity number, ranging from 0 through 127. Musically map MIDI velocity to peak amplitude within a specified dynamic range in decibels: a = (m * v + b) ^ 2, where a = amplitude, v = MIDI velocity, r = 10 ^ (R / 20), b = 127 / (126 * sqrt( r )) - 1 / 126, m = (1 - b) / 127, and R = specified dynamic range in decibels. See Roger Dannenberg, "The Interpretation of MIDI Velocity," in Georg Essl and Ichiro Fujinaga (Eds.), Proceedings of the 2006 International Computer Music Conference, November 6-11, 2006 (San Francisco: The International Computer Music Association), pp. 193-196. Examples Here is an example of the ampmidid opcode. It uses the file ampmidid.csd. Example of the ampmidid opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also aftouch, cpsmidi, cpsmidib, midictrl, notnum, octmidi, octmidib, pchbend, pchmidi, pchmidib, veloc Credits Author: &namemichael; 2006 manual_src~dfsg/opcodes/sr.xml0000644000000000000000000000771512262561504015561 0ustar rootroot sr Orchestra Syntax:Header sr sr Sets the audio sampling rate. Description These statements are global value assignments, made at the beginning of an orchestra, before any instrument block is defined. Their function is to set certain reserved symbol variables that are required for performance. Once set, these reserved symbols can be used in expressions anywhere in the orchestra. Syntax sr = iarg Initialization sr = (optional) -- set sampling rate to iarg samples per second per channel. The default value is 44100. In addition, any global variable can be initialized by an init-time assignment anywhere before the first instr statement. All of the above assignments are run as instrument 0 (i-pass only) at the start of real performance. Beginning with Csound version 3.46, sr may be omitted. The sample rate will be calculated from kr and ksmps, but this must evaluate to an integer. If none of these global values is defined, the sample rate will default to 44100. You will usually want to use a value that your soundcard supports, like 44100 or 48000, otherwise, the audio generated by csound may be unplayable, or you will get an error if you attempt to run in real-time. You may naturally use a sample rate like 96000, for off-line rendering even if your soundcard doesn't support it. Csound will generate a valid file that can be played on capable systems. Examples sr = 10000 kr = 500 ksmps = 20 gi1 = sr/2. ga init 0 itranspose = octpch(.0l) Here is another example of the sr opcode. It uses the file sr.csd. Example of the sr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: maximum number of harmonics = 240 maximum number of harmonics = 24 maximum number of harmonics = 2 See Also kr, ksmps, nchnls manual_src~dfsg/opcodes/oscil.xml0000644000000000000000000000654612262561504016247 0ustar rootroot oscil Signal Generators:Basic Oscillators oscil oscil A simple oscillator. Description oscil reads table ifn sequentially and repeatedly at a frequency xcps. The amplitude is scaled by xamp. Syntax ares oscil xamp, xcps [, ifn, iphs] kres oscil kamp, kcps [, ifn, iphs] Initialization ifn -- (optional) function table number. Requires a wrap-around guard point. The table number defaults to -1 which indicates a sinewave. iphs (optional, default=0) -- initial phase of sampling, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is 0. Performance kamp, xamp -- amplitude kcps, xcps -- frequency in cycles per second. The oscil opcode generates periodic control (or audio) signals consisting of the value of kamp (xamp) times the value returned from control rate (audio rate) sampling of a stored function table. The internal phase is simultaneously advanced in accordance with the kcps or xcps input value. Table ifn is incrementally sampled modulo the table length and the value obtained is multiplied by amp. If you need to change the oscillator table with a k-rate signal, you can use oscilikt. Examples Here is an example of the oscil opcode. It uses the file oscil.csd. Example of the oscil opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also oscili, oscilikt, oscil3, poscil, poscil3 manual_src~dfsg/opcodes/tableiw.xml0000644000000000000000000001673212262561504016563 0ustar rootroot tableiw Table Control:Read/Write Opreations tableiw tableiw Change the contents of existing function tables. Description This opcode operates on existing function tables, changing their contents. tableiw is used when all inputs are init time variables or constants and you only want to run it at the initialization of the instrument. The valid combinations of variable types are shown by the first letter of the variable names. Syntax tableiw isig, indx, ifn [, ixmode] [, ixoff] [, iwgmode] Initialization isig -- Input value to write to the table. indx -- Index into table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode not equal to 0) ifn -- Table number. Must be >= 1. Floats are rounded down to an integer. If a table number does not point to a valid table, or the table has not yet been loaded (GEN01) then an error will result and the instrument will be de-activated. ixmode (optional, default=0) -- index mode. 0 = indx and ixoff ranges match the length of the table. not equal to 0 = indx and ixoff have a 0 to 1 range. ixoff (optional, default=0) -- index offset. 0 = Total index is controlled directly by indx, i.e. the indexing starts from the start of the table. Not equal to 0 = Start indexing from somewhere else in the table. Value must be positive and less than the table length (ixmode = 0) or less than 1 (ixmode not equal to 0). iwgmode (optional, default=0) -- Wrap and guard point mode. 0 = Limit mode. 1 = Wrap mode. 2 = Guardpoint mode. Performance Limit mode (0) Limit the total index (indx + ixoff) to between 0 and the guard point. For a table of length 5, this means that locations 0 to 3 and location 4 (the guard point) can be written. A negative total index writes to location 0. Wrap mode (1) Wrap total index value into locations 0 to E, where E is either one less than the table length or the factor of 2 number which is one less than the table length. For example, wrap into a 0 to 3 range - so that total index 6 writes to location 2. Guardpoint mode (2) The guardpoint is written at the same time as location 0 is written - with the same value. This facilitates writing to tables which are intended to be read with interpolation for producing smooth cyclic waveforms. In addition, before it is used, the total index is incremented by half the range between one location and the next, before being rounded down to the integer address of a table location. Normally (iwgmode = 0 or 1) for a table of length 5 - which has locations 0 to 3 as the main table and location 4 as the guard point, a total index in the range of 0 to 0.999 will write to location 0. ("0.999" means just less than 1.0.) 1.0 to 1.999 will write to location 1 etc. A similar pattern holds for all total indexes 0 to 4.999 (igwmode = 0) or to 3.999 (igwmode = 1). igwmode = 0 enables locations 0 to 4 to be written - with the guardpoint (4) being written with a potentially different value from location 0. With a table of length 5 and the iwgmode = 2, then when the total index is in the range 0 to 0.499, it will write to locations 0 and 4. Range 0.5 to 1.499 will write to location 1 etc. 3.5 to 4.0 will also write to locations 0 and 4. This way, the writing operation most closely approximates the results of interpolated reading. Guard point mode should only be used with tables that have a guardpoint. Guardpoint mode is accomplished by adding 0.5 to the total index, rounding to the next lowest integer, wrapping it modulo the factor of two which is one less than the table length, writing the table (locations 0 to 3 in our example) and then writing to the guard point if index = 0. Examples Here is an example of the tableiw opcode. It uses the file tableiw.csd. Example of the tableiw opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also tablew, tablewkt More information on this opcode: http://www.csounds.com/journal/issue12/genInstruments.html , written by Jacob Joaquin Credits Author: Robin Whittle Australia May 1997 New in version 3.47 Updated August 2002, thanks go to Abram Hindle for pointing out the correct syntax. manual_src~dfsg/opcodes/flsetbox.xml0000644000000000000000000001002712262561504016751 0ustar rootroot FLsetBox FLTK:Appearance FLsetBox FLsetBox Sets the appearance of a box surrounding a FLTK widget. Description FLsetBox sets the appearance of a box surrounding the target widget. Syntax FLsetBox itype, ihandle Initialization itype -- an integer number that modify the appearance of the target widget. Legal values for the itype argument are: 1 - flat box 2 - up box 3 - down box 4 - thin up box 5 - thin down box 6 - engraved box 7 - embossed box 8 - border box 9 - shadow box 10 - rounded box 11 - rounded box with shadow 12 - rounded flat box 13 - rounded up box 14 - rounded down box 15 - diamond up box 16 - diamond down box 17 - oval box 18 - oval shadow box 19 - oval flat box ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/ftsr.xml0000644000000000000000000000506012262561504016102 0ustar rootroot ftsr Table Control:Table Queries ftsr ftsr Returns the sampling-rate of a stored function table. Description Returns the sampling-rate of a stored function table. Syntax ftsr(x) (init-rate args only) Performance Returns the sampling-rate of a GEN01 generated table. The sampling-rate is determined from the header of the original file. If the original file has no header or the table was not created by these GEN01, ftsr returns 0. New in Csound version 3.49. Examples Here is an example of the ftsr opcode. It uses the file ftsr.csd. Example of the ftsr opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. its output should a line like these: sampling-rate of table number 1 = 44100 sampling-rate of table number 2 = 22050 sampling-rate of table number 3 = 44100 sampling-rate of table number 4 = 44100 See Also ftchnls, ftlen, ftlptim, nsamp Credits Author: &namegabriel; Italy October 1998 manual_src~dfsg/opcodes/scalearray.xml0000644000000000000000000000455012262561504017255 0ustar rootroot scalearray Array Opcodes scalearray scalearray Scales the values in a range of a vector (one dimensional array). Description The scalearray opcode scales a subregion of a vector to a given minimum/maximum. Syntax scalearray tarray, kmin, kmax[, kleft, kright] Performance karray -- array for operation. kmin, kmax -- target minimum and maximum values. kleft, kright -- range of table to use, defaulting to 0 and size of the vector. Examples Here is an example of the scalearray opcode. It uses the file scalearray.csd. Example of the scalearray opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also maxarray, minarray, sumarray, Credits Author: &namejohn; October 2011 New in Csound version 5.14 Renamed in Csound version 6.00 manual_src~dfsg/opcodes/filenchnls.xml0000644000000000000000000000607412262561504017257 0ustar rootroot filenchnls Signal I/O:Soundfile Queries filenchnls filenchnls Returns the number of channels in a sound file. Description Returns the number of channels in a sound file. Syntax ir filenchnls ifilcod [, iallowraw] Initialization ifilcod -- sound file to be queried iallowraw -- (Optional) Allow raw sound files (default=1) Performance filenchnls returns the number of channels in the sound file ifilcod. filechnls can return the number of channels of convolve and PVOC files if the iallowraw flag is not zero (it is non-zero by default). Examples Here is an example of the filenchnls opcode. It uses the file filenchnls.csd, fox.wav, and kickroll.wav. Example of the filenchnls opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The audio file fox.wav is monoaural (1 channel), while kickroll.wav is stereo (2 channels) So filenchnls's output should include lines like this: instr 1: ichn = 1.000 instr 1: ichn = 2.000 See Also filebit, filelen, filepeak, filesr Credits Author: &namematt; July 1999 New in Csound version 3.57 manual_src~dfsg/opcodes/tablegpw.xml0000644000000000000000000000354712262561504016741 0ustar rootroot tablegpw Table Control:Read/Write Opreations tablegpw tablegpw Writes a table's guard point. Description Writes a table's guard point. Syntax tablegpw kfn Performance kfn -- Table number to be interrogated tablegpw -- For writing the table's guard point, with the value which is in location 0. Does nothing if table does not exist. Likely to be useful after manipulating a table with tablemix or tablecopy. See Also tablecopy, tablemix, tableicopy, tableigpw, tableimix Credits Author: Robin Whittle Australia May 1997 New in version 3.47 manual_src~dfsg/opcodes/phasorbnk.xml0000644000000000000000000000625012262561504017115 0ustar rootroot phasorbnk Signal Generators:Phasors phasorbnk phasorbnk Produce an arbitrary number of normalized moving phase values. Description Produce an arbitrary number of normalized moving phase values, accessable by an index. Syntax ares phasorbnk xcps, kndx, icnt [, iphs] kres phasorbnk kcps, kndx, icnt [, iphs] Initialization icnt -- maximum number of phasors to be used. iphs -- initial phase, expressed as a fraction of a cycle (0 to 1). If -1 initialization is skipped. If iphas>1 each phasor will be initialized with a random value. Performance kndx -- index value to access individual phasors For each independent phasor, an internal phase is successively accumulated in accordance with the kcps or xcps frequency to produce a moving phase value, normalized to lie in the range 0 <= phs < 1. Each individual phasor is accessed by index kndx. This phasor bank can be used inside a k-rate loop to generate multiple independent voices, or together with the adsynt opcode to change parameters in the tables used by adsynt. Examples Here is an example of the phasorbnk opcode. It uses the file phasorbnk.csd. Example of the phasorbnk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Generate multiple voices with independent partials. This example is better with adsynt. See also the example under adsynt, for k-rate use of phasorbnk. Credits Author: Peter Neubäcker Munich, Germany August 1999 New in Csound version 3.58 manual_src~dfsg/opcodes/flpanel.xml0000644000000000000000000001463712262561504016557 0ustar rootroot FLpanel FLTK:Containers FLpanel FLpanel Creates a window that contains FLTK widgets. Description Creates a window that contains FLTK widgets. Syntax FLpanel "label", iwidth, iheight [, ix] [, iy] [, iborder] [, ikbdcapture] [, iclose] Initialization label -- a double-quoted string containing some user-provided text, placed near the corresponding widget. iwidth -- width of widget. iheight -- height of widget. ix (optional) -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy (optional) -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iborder (optional) -- border type of the container. It is expressed by means of an integer number chosen from the following: 0 - no border 1 - down box border 2 - up box border 3 - engraved border 4 - embossed border 5 - black line border 6 - thin down border 7 - thin up border ikbdcapture (default = 0) -- If this flag is set to 1, keyboard events are captured by the window (for use with sensekey and FLkeyIn) iclose (default = 0) -- If this flag is set to anything other than 0, the close button of the window is disabled, and the window cannot be closed by the user directly. It will close when csound exits. Performance Containers are useful to format the graphic appearance of the widgets. The most important container is FLpanel, that actually creates a window. It can be filled with other containers and/or valuators or other kinds of widgets. There are no k-rate arguments in containers. FLpanel creates a window. It must be followed by the opcode FLpanelEnd when all widgets internal to it are declared. For example: FLpanel "PanelPluto",450,550,100,100 ;***** start of container gk1, ih1 FLslider "FLslider 1", 500, 1000, 2 ,1, -1, 300,15, 20,50 gk2, ih2 FLslider "FLslider 2", 300, 5000, 2 ,3, -1, 300,15, 20,100 gk3, ih3 FLslider "FLslider 3", 350, 1000, 2 ,5, -1, 300,15, 20,150 gk4, ih4 FLslider "FLslider 4", 250, 5000, 1 ,11,-1, 300,30, 20,200 FLpanelEnd ;***** end of container will output the following result: FLpanel. FLpanel. If the ikbdcapture flag is set, the window captures keyboard events, and sends them to all sensekey. This flag modifies the behavior of sensekey, and makes it receive events from the FLTK window instead of stdin. Examples Here is an example of the FLpanel opcode. It uses the file FLpanel.csd. Example of the FLpanel opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLgroup, FLgroupEnd, FLpack, FLpackEnd, FLpanelEnd, FLscroll, FLscrollEnd, FLtabs, FLtabsEnd, sensekey Credits Author: &namegabriel; New in version 4.22 Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/STKMandolin.xml0000644000000000000000000000720112262561504017246 0ustar rootroot STKMandolin Signal Generators:STK Opcodes STKMandolin STKMandolin STKMandolin produces mamdolin-like sounds. Description STKMandolin produces mamdolin-like sounds, using "commuted synthesis" techniques to model a mandolin instrument. Syntax asignal STKMandolin ifrequency, iamplitude, [kbody, kv1[, kpos, kv2[, ksus, kv3[, kdetune, kv4[, kmic, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kbody -- controller 2, size of body. Value range of kv1 is 0-127. kpos -- controller 4, pluck position. Value range of kv2 is 0-127. ksus -- controller 11, string sustain. Value range of kv3 is 0-127. kdetune -- controller 1, string detuning. Value range of kv4 is 0-127. kmic -- controller 128, position of microphone. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the Mandolin class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKMandolin opcode. It uses the file STKMandolin.csd. Example of the STKMandolin opcode. See Also STKPlucked. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/fluidOut.xml0000644000000000000000000000665212262561504016727 0ustar rootroot fluidOut Signal Generators:Sample Playback fluidOut fluidOut Outputs sound from a given fluidEngine Syntax aleft, aright fluidOut ienginenum Description Outputs the sound from a fluidEngine. Initialization ienginenum -- engine number assigned from fluidEngine Performance aleft -- Left channel audio output. aright -- Right channel audio output. Invoke fluidOut in an instrument definition numbered higher than any fluidcontrol instrument definitions. All SoundFonts used in the fluidEngine numbered ienginenum send their audio output to this one opcode. Send a note with an indefinite duration to this instrument to turn the SoundFonts on for as long as required. Examples Here is an example of the fluidOut opcode with two fluidOuts. It uses the file fluidOut.csd, 01hpschd.sf2 and 22Bassoon.sf2. Example of the fluidOut opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output should include lines like these: chnl 1 using instr 1 chnl 2 using instr 2 SoundFont: 1 Bank: 0 Preset: 0 Harpsichord I-8 SoundFont: 1 Bank: 0 Preset: 70 Ethan Bassoon mono See Also fluidEngine, fluidNote, fluidLoad More information on soundfonts in the Floss Manuals: http://en.flossmanuals.net/csound/ch048_d-reading-midi-files Other information on soundfonts on Wikipedia: http://en.wikipedia.org/wiki/Soundfont Credits Michael Gogins (gogins at pipeline dot com), Steven Yi. Thanks to Peter Hanappe for Fluidsynth. New in Csound5.00 manual_src~dfsg/opcodes/ksmps.xml0000644000000000000000000000624112262561504016263 0ustar rootroot ksmps Orchestra Syntax:Header ksmps ksmps Sets the number of samples in a control period. Description These statements are global value assignments, made at the beginning of an orchestra, before any instrument block is defined. Their function is to set certain reserved symbol variables that are required for performance. Once set, these reserved symbols can be used in expressions anywhere in the orchestra. Syntax ksmps = iarg Initialization ksmps = (optional) -- set the number of samples in a control period. This value must equal sr/kr. The default value is 10. In addition, any global variable can be initialized by an init-time assignment anywhere before the first instr statement. All of the above assignments are run as instrument 0 (i-pass only) at the start of real performance. Beginning with Csound version 3.46, either ksmps may be omitted. Csound will attempt to calculate the omitted value from the specified sr and krvalues, but it should evaluate to an integer. Warning ksmps must be an integer value. Examples sr = 10000 kr = 500 ksmps = 20 gi1 = sr/2. ga init 0 itranspose = octpch(.0l) See Also kr, nchnls, nchnls_i, sr Credits Thanks to a note from &namegabriel;, added a warning about integer values. manual_src~dfsg/opcodes/table.xml0000644000000000000000000000721012262561504016212 0ustar rootroot table Signal Generators:Table Access table table Accesses table values by direct indexing. Description Accesses table values by direct indexing. Syntax ares table andx, ifn [, ixmode] [, ixoff] [, iwrap] ires table indx, ifn [, ixmode] [, ixoff] [, iwrap] kres table kndx, ifn [, ixmode] [, ixoff] [, iwrap] Initialization ifn -- function table number. ixmode (optional) -- index data mode. The default value is 0. 0 = raw index 1 = normalized (0 to 1) ixoff (optional) -- amount by which index is to be offset. For a table with origin at center, use tablesize/2 (raw) or .5 (normalized). The default value is 0. iwrap (optional) -- wraparound index flag. The default value is 0. 0 = nowrap (index < 0 treated as index=0; index > tablesize sticks at index=size) 1 = wraparound. Performance table invokes table lookup on behalf of init, control or audio indices. These indices can be raw entry numbers (0,l,2...size - 1) or scaled values (0 to 1-e). Indices are first modified by the offset value then checked for range before table lookup (see iwrap). If index is likely to be full scale, or if interpolation is being used, the table should have an extended guard point. table indexed by a periodic phasor ( see phasor) will simulate an oscillator. Examples Here is an example of the table opcode. It uses the file table.csd. Example of the table opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also tablei, table3, oscil1, oscil1i, osciln Credits Example written by &namekevin;. manual_src~dfsg/opcodes/log2.xml0000644000000000000000000000502712262561504015772 0ustar rootroot log2 Mathematical Operations:Mathematical Functions log2 log2 Returns a base 2 log. Description Returns the base 2 log of x (x positive only). The argument value is restricted for log, log2, and sqrt. Syntax log2(x) (no rate restriction) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Examples Here is an example of the log2 opcode. It uses the file log2.csd. Example of the log2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: i1 = 3.000 See Also abs, exp, frac, int, log, log, i, sqrt Credits Written by &namejohn;. New in version 5.17.12 manual_src~dfsg/opcodes/tableng.xml0000644000000000000000000000521212262561504016537 0ustar rootroot tableng Table Control:Table Queries tableng tableng Interrogates a function table for length. Description Interrogates a function table for length. Syntax ires tableng ifn kres tableng kfn Initialization ifn -- Table number to be interrogated Performance kfn -- Table number to be interrogated tableng returns the length of the specified table. This will be a power of two number in most circumstances. It will not show whether a table has a guardpoint or not. It seems this information is not available in the table's data structure. If the specified table is not found, then 0 will be returned. Likely to be useful for setting up code for table manipulation operations, such as tablemix and tablecopy. Examples Here is an example of the tableng opcode. It uses the file tableng.csd. Example of the tableng opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: instr 1: isize = 115506.000 Credits Author: Robin Whittle Australia May 1997 manual_src~dfsg/opcodes/remoteport.xml0000644000000000000000000000247412262561504017332 0ustar rootroot remoteport Network remoteport remoteport Defines the port for use with the remote system. Description Defines the port for use with the insremot, midremot, insglobal and midglobal opcodes. Syntax remoteport iportnum Initialization iportnum -- number of the port to be used. If zero or negative the default port 40002 is selected. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK November, 2006 New in Csound version 5.05 manual_src~dfsg/opcodes/pvinterp.xml0000644000000000000000000002041512262561504016774 0ustar rootroot pvinterp Spectral Processing:STFT pvinterp pvinterp Interpolates between the amplitudes and frequencies of two phase vocoder analysis files. Description pvinterp interpolates between the amplitudes and frequencies, on a bin by bin basis, of two phase vocoder analysis files (one from a previously called pvbufread unit and the other from within its own argument list), allowing for user defined transitions between analyzed sounds. It also allows for general scaling of the amplitudes and frequencies of each file separately before the interpolated values are calculated and sent to the resynthesis routines. The kfmod argument in pvinterp performs its frequency scaling on the frequency values after their derivation from the separate scaling and subsequent interpolation is performed so that this acts as an overall scaling value of the new frequency components. Syntax ares pvinterp ktimpnt, kfmod, ifile, kfreqscale1, kfreqscale2, \ kampscale1, kampscale2, kfreqinterp, kampinterp Initialization ifile -- the pvoc number (n in pvoc.n) or the name in quotes of the analysis file made using pvanal. (See pvoc.) Performance ktimpnt -- the passage of time, in seconds, through this file. ktimpnt must always be positive, but can move forwards or backwards in time, be stationary or discontinuous, as a pointer into the analysis file. kfmod -- a control-rate transposition factor: a value of 1 incurs no transposition, 1.5 transposes up a perfect fifth, and .5 down an octave. kfreqscale1, kfreqscale2, kampscale1, kampscale2 -- used in pvinterp to scale the frequencies and amplitudes stored in each frame of the phase vocoder analysis file. kfreqscale1 and kampscale1 scale the frequencies and amplitudes of the data from the file read by the previously called pvbufread (this data is passed internally to the pvinterp unit). kfreqscale2 and kampscale2 scale the frequencies and amplitudes of the file named by ifile in the pvinterp argument list and read within the pvinterp unit. By using these arguments, it is possible to adjust these values before applying the interpolation. For example, if file1 is much louder than file2, it might be desirable to scale down the amplitudes of file1 or scale up those of file2 before interpolating. Likewise one can adjust the frequencies of each to bring them more in accord with one another (or just the opposite, of course!) before the interpolation is performed. kfreqinterp, kampinterp -- used in pvinterp, determine the interpolation distance between the values of one phase vocoder file and the values of a second file. When the value of kfreqinterp is 1, the frequency values will be entirely those from the first file (read by the pvbufread), post scaling by the kfreqscale1 argument. When the value of kfreqinterp is 0 the frequency values will be those of the second file (read by the pvinterp unit itself), post scaling by kfreqscale2. When kfreqinterp is between 0 and 1 the frequency values will be calculated, on a bin, by bin basis, as the percentage between each pair of frequencies (in other words, kfreqinterp=0.5 will cause the frequencies values to be half way between the values in the set of data from the first file and the set of data from the second file). kampinterp works in the same way upon the amplitudes of the two files. Since these are k-rate arguments, the percentages can change over time making it possible to create many kinds of transitions between sounds. Examples The example below shows an example using pvbufread with pvinterp to interpolate between the sound of an oboe and the sound of a clarinet. The value of kinterp returned by a linseg is used to determine the timing of the transitions between the two sounds. The interpolation of frequencies and amplitudes are controlled by the same factor in this example, but for other effects it might be interesting to not have them synchronized in this way. In this example the sound will begin as a clarinet, transform into the oboe and then return again to the clarinet sound. The value of kfreqscale2 is 1.065 because the oboe in this case is a semitone higher in pitch than the clarinet and this brings them approximately to the same pitch. The value of kampscale2 is 0.75 because the analyzed clarinet was somewhat louder than the analyzed oboe. The setting of these two parameters make the transition quite smooth in this case, but such adjustments are by no means necessary or even advocated. ktime1 line 0, p3, 3.5 ; used as index in the "oboe.pvoc" file ktime2 line 0, p3, 4.5 ; used as index in the "clar.pvoc" file kinterp linseg 1, p3*0.15, 1, p3*0.35, 0, p3*0.25, 0, p3*0.15, 1, p3*0.1, 1 pvbufread ktime1, "oboe.pvoc" apv pvinterp ktime2,1,"clar.pvoc", 1, 1.065, 1, 0.75, 1-kinterp, 1-kinterp Here is a complete example of the pvinterp opcode. It uses the file pvinterp.csd Example of the pvinterp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvbufread, pvcross, pvread, tableseg, tablexseg Credits Author: Richard Karpen Seattle, Wash 1997 manual_src~dfsg/opcodes/pvslock.xml0000644000000000000000000000452412262561504016611 0ustar rootroot pvslock Spectral Processing:Streaming pvslock pvslock Frequency lock an input fsig Description This opcode searches for spectral peaks and then locks the frequencies around those peaks. This is similar to phase-locking in non-streaming PV processing. It can be used to improve timestretching and pitch-shifting quality in PV processing. Syntax fsig pvslock fsigin, klock Performance fsig -- output pv stream fsigin -- input pv stream. klock -- frequency lock, 1 -> lock, 0 -> unlock (bypass). It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the pvslock opcode. It uses the file pvslock.csd. Example of the pvslock opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; November 2004 New plugin in version 5 November 2004. manual_src~dfsg/opcodes/kdump3.xml0000644000000000000000000000076212262561504016333 0ustar rootroot kdump3 kdump3 kdump3 Deprecated. Description Deprecated as of version 3.49. Use the dumpk3 opcode instead. manual_src~dfsg/opcodes/xin.xml0000644000000000000000000000700012262561504015716 0ustar rootroot xin Signal I/O:Software Bus xin xin Passes variables to a user-defined opcode block, Description The xin and xout opcodes copy variables to and from the opcode definition, allowing communication with the calling instrument. The types of input and output variables are defined by the parameters intypes and outtypes. Notes xin and xout should be called only once, and xin should precede xout, otherwise an init error and deactivation of the current instrument may occur. These opcodes actually run only at i-time. Performance time copying is done by the user opcode call. This means that skipping xin or xout with kgoto has no effect, while skipping with igoto affects both init and performance time operation. Syntax xinarg1 [, xinarg2] ... [xinargN] xin Performance xinarg1, xinarg2, ... - input arguments. The number and type of variables must agree with the user-defined opcode's intypes declaration. However, xin does not check for incorrect use of init-time and control-rate variables. The syntax of a user-defined opcode block is as follows: opcode name, outtypes, intypes xinarg1 [, xinarg2] [, xinarg3] ... [xinargN] xin [setksmps iksmps] ... the rest of the instrument's code. xout xoutarg1 [, xoutarg2] [, xoutarg3] ... [xoutargN] endop The new opcode can then be used with the usual syntax: [xinarg1] [, xinarg2] ... [xinargN] name [xoutarg1] [, xoutarg2] ... [xoutargN] [, iksmps] Examples See the example for the opcode opcode. See Also endop, opcode, setksmps, xout Credits Author: Istvan Varga, 2002; based on code by Matt J. Ingalls New in version 4.22 manual_src~dfsg/opcodes/abexprnd.xml0000644000000000000000000000077512262561504016737 0ustar rootroot abexprnd abexprnd abexprnd Deprecated. Description Deprecated as of version 3.49. Use the bexprnd opcode instead. manual_src~dfsg/opcodes/vexpv.xml0000644000000000000000000001402412262561504016274 0ustar rootroot vexpv Vectorial:Vectorial operations vexpv vexpv Performs exponential operations between two vectorial control signals Description Performs exponential operations between two vectorial control signals Syntax vexpv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] Initialization ifn1 - number of the table hosting the first vector to be processed ifn2 - number of the table hosting the second vector to be processed Performance kelements - number of elements of the two vectors kdstoffset - index offset for the destination (ifn1) table (Default=0) ksrcoffset - index offset for the source (ifn2) table (Default=0) kverbose - Selects whether or not warnings are printed (Default=0) vexpv elevates each element of ifn2 to the corresponding element of ifn1. Each vectorial signal is hosted by a table (ifn1 and ifn2). The number of elements contained in both vectors must be the same. The result is a new vectorial control signal that overrides old values of ifn1. If you want to keep the old ifn1 vector, use vcopy_i opcode to copy it in another table. You can use kdstoffset and ksrcoffset to specify vectors in any location of the tables. Negative values for kdstoffset and ksrcoffset are acceptable. If kdstoffset is negative, the out of range section of the vector will be discarded. If ksrcoffset is negative, the out of range elements will be assumed to be 0 (i.e. the destination elements will be set to 1). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 0 (i.e. the destination elements will be set to 1). If the optional kverbose argument is different to 0, the opcode will print warning messages every k-pass if table lengths are exceeded. Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at k-rate (this means that every k-pass the vectors are processed). There's an i-rate version of this opcode called vexpv_i. Please note that the elements argument has changed in version 5.03 from i-rate to k-rate. This will change the opcode's behavior in the unusual cases where the i-rate variable ielements is changed inside the instrument, for example in: instr 1 ielements = 10 vadd 1, 1, ielements ielements = 20 vadd 2, 1, ielements turnoff endin All these operators (vaddv, vsubv, vmultv, vdivv, vpowv, vexpv, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Examples Here is an example of the vexpv opcode. It uses the file vexpv.csd. Example of the vexpv opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/flxyin.xml0000644000000000000000000001210712262561504016435 0ustar rootroot FLxyin FLTK:Other FLxyin FLxyin Senses the mouse cursor position in a user-defined area inside an FLpanel. Description Similar to xyin, sense the mouse cursor position in a user-defined area inside an FLpanel. Syntax koutx, kouty, kinside FLxyin ioutx_min, ioutx_max, iouty_min, iouty_max, \ iwindx_min, iwindx_max, iwindy_min, iwindy_max [, iexpx, iexpy, ioutx, iouty] Initialization ioutx_min, ioutx_max - the minimum and maximum limits of the interval to be output (X or horizontal axis). iouty_min, iouty_max - the minimum and maximum limits of the interval to be output (Y or vertical axis). iwindx_min, iwindx_max - the X coordinate of the horizontal edges of the sensible area, relative to the FLpanel ,in pixels. iwindy_min, iwindy_max - the Y coordinates of the vertical edges of the sensible area, relative to the FLpanel, in pixels. iexpx, iexpy - (optional) integer numbers denoting the behavior of the x or y output: 0 -> output is linear; -1 -> output is exponential; any other number indicates the number of an existing table that is used for indexing. With a positive value for table number, linear interpolation is provided in table indexing. A negative table number suppresses interpolation. Notice that in normal operations, the table should be normalized and unipolar (i.e. all table elements should be in the zero to one range). In this case all table elements will be rescaled according to imax and imin. Anyway, it is possible to use non-normalized tables (created with a negative table number, that can contain elements of any value), in order to access the actual values of table elements, without rescaling, by assigning 0 to iout_min and 1 to iout_max. ioutx, iouty – (optional) initial output values. Performance koutx, kouty - output values, scaled according to user choices. kinside - a flag that informs if the mouse cursor falls out of the rectangle of the user-defined area. If it is out of the area, kinside is set to zero. FLxyin senses the mouse cursor position in a user-defined area inside an FLpanel. When FLxyin is called, the position of the mouse within the chosen area is returned at k-rate. It is possible to define the sensible area, as well the minimum and maximum values corresponding to the minimum and maximum mouse positions. Mouse buttons don’t need to be pressed to make FLxyin to operate. It is able to function correctly even if other widgets (present in the FLpanel) overlap the sensible area. FLxyin unlike most other FLTK opcodes can't be used inside the header, since it is not a widget. It is just a definition of an area for mouse sensing within an FLTK panel. Examples Here is an example of the FLxyin opcode. It uses the file FLxyin.csd. Example of the FLxyin opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the FLxyin opcode. It uses the file FLxyin-2.csd. Example of the FLxyin opcode. See Also FLpanel Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/lfo.xml0000644000000000000000000000600212262561504015701 0ustar rootroot lfo Signal Generators:Basic Oscillators lfo lfo A low frequency oscillator of various shapes. Description A low frequency oscillator of various shapes. Syntax kres lfo kamp, kcps [, itype] ares lfo kamp, kcps [, itype] Initialization itype (optional, default=0) -- determine the waveform of the oscillator. Default is 0. itype = 0 - sine itype = 1 - triangles itype = 2 - square (bipolar) itype = 3 - square (unipolar) itype = 4 - saw-tooth itype = 5 - saw-tooth(down) The sine wave is implemented as a 4096 table and linear interpolation. The others are calculated. Performance kamp -- amplitude of output kcps -- frequency of oscillator Examples Here is an example of the lfo opcode. It uses the file lfo.csd. Example of the lfo opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK November 1998 New in Csound version 3.491 manual_src~dfsg/opcodes/diff.xml0000644000000000000000000000550612262561504016041 0ustar rootroot diff Signal Modifiers:Sample Level Operators diff diff Modify a signal by differentiation. Description Modify a signal by differentiation. Syntax ares diff asig [, iskip] kres diff ksig [, iskip] Initialization iskip (optional) -- initial disposition of internal save space (see reson). The default value is 0. Performance integ and diff perform integration and differentiation on an input control signal or audio signal. Each is the converse of the other, and applying both will reconstruct the original signal. Since these units are special cases of low-pass and high-pass filters, they produce a scaled (and phase shifted) output that is frequency-dependent. Thus diff of a sine produces a cosine, with amplitude 2 * pi * Hz / sr that of the original (for each component partial); integ will inversely affect the magnitudes of its component inputs. With this understanding, these units can provide useful signal modification. Examples Here is an example of the diff opcode. It uses the file diff.csd. Example of the diff opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also downsamp, integ, interp, samphold, upsamp manual_src~dfsg/opcodes/slider8tablef.xml0000644000000000000000000001524012262561504017655 0ustar rootroot slider8tablef Real-time MIDI:Slider Banks slider8tablef slider8tablef Stores a bank of 8 different MIDI control messages to a table, filtered before output. Description Stores a bank of 8 different MIDI control messages to a table, filtered before output. Syntax kflag slider8tablef ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, icutoff1, .... , ictlnum8, imin8, imax8, init8, ifn8, icutoff8 Initialization ichan -- MIDI channel (1-16) ioutTable -- number of the table that will contain the output ioffset -- output table offset. A zero means that the output of the first slider will affect the first table element. A 10 means that the output of the first slider will affect the 11th table element. ictlnum1 ... ictlnum8 -- MIDI control number (0-127) imin1 ... imin8 -- minimum values for each controller imax1 ... imax8 -- maximum values for each controller init1 ... init8 -- initial value for each controller ifn1 ... ifn8 -- function table for conversion for each controller icutoff1 ... icutoff8 -- low-pass filter cutoff frequency for each controller Performance kflag -- a flag that informs if any control-change message in the bank has been received. In this case kflag is set to 1. Otherwise is set to zero. slider8tablef is a bank of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider8tablef allows a bank of 8 different MIDI control message numbers. It filters the signal before output. This eliminates discontinuities due to the low resolution of the MIDI (7 bit). The cutoff frequency can be set separately for each controller (suggested range: .1 to 5 Hz). As the input and output arguments are many, you can split the line using '\' (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. slider8tablef is very similar to slider8f and sliderNf family of opcodes (see their manual for more information). The actual difference is that the output is not stored to k-rate variables, but to a table, denoted by the ioutTable argument. It is possible to define a starting index in order to use the same table for more than one slider bank (or other purposes). It is possible to use this opcode together with FLslidBnk2Setk and FLslidBnk2, so you can synchronize the position of the MIDI values to the position of the FLTK valuator widgets of FLslidBnk2. Notice that you have to specify the same min/max values as well the linear/exponential responses in both sliderNtable(f) and FLslidBnk2. The exception is when using table-indexed response instead of a lin/exp response. In this case, in order to achieve a useful result, the table-indexed response and actual min/max values must be set only in FLslidBnk2, whereas, in sliderNtable(f), you have to set a linear response and a minimum of zero and a maximum of one in all sliders. slider8tablef opcodes do not output the required initial value immediately, but only after some k-cycles because the filter slightly delays the output. See Also slider16table, slider16tablef, slider32table, slider32tablef, slider64table, slider64tablef, slider8table Credits Author: &namegabriel; New in Csound version 5.06 manual_src~dfsg/opcodes/vtabwa.xml0000644000000000000000000000411012262561504016403 0ustar rootroot vtabwa Vectorial:Tables vtabwa vtabwa Write vectors (to tables -or arrays of vectors). Description This opcode writes vectors to tables at a-rate. Syntax vtabwa andx, ifn, ainarg1 [, ainarg2, ainarg3 , .... , ainargN ] Initialization ifn - table number Performance andx - Index into f-table, a positive number range matching the table length ainarg1...ainargN - input vector elements This opcode is useful in all cases in which one needs to write sets of values associated to unique indexes (for example, multi-channel samples, STFT bin frames, spectral formants, p-field based scores etc.). The number of elements of each vector (length of the vector) is determined by the number of optional arguments on the right (ainarg1, ainarg2, ainarg3, .... ainargN). Notice that no wrap nor limit mode is implemented. So, if an index attempt to access to a zone not allocated by the table, Csound will probably crash. However this drawback can be easily avoided by using wrap or limit opcodes applied to indexes before using vtabwa, in order to correct eventual out-of-range values. Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/butterbp.xml0000644000000000000000000000566212262561504016763 0ustar rootroot butterbp Signal Modifiers:Standard Filters butterbp butterbp A band-pass Butterworth filter. Description Implementation of a second-order band-pass Butterworth filter. This opcode can also be written as butbp. Syntax ares butterbp asig, xfreq, xband [, iskip] Initialization iskip (optional, default=0) -- Skip initialization if present and non-zero. Performance These filters are Butterworth second-order IIR filters. They are slightly slower than the original filters in Csound, but they offer an almost flat passband and very good precision and stopband attenuation. asig -- Input signal to be filtered. xfreq -- Cutoff or center frequency for each of the filters. xband -- Bandwidth of the bandpass and bandreject filters. Examples Here is an example of the butterbp opcode. It uses the file butterbp.csd. Example of the butterbp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also butterbr, butterhp, butterlp Credits Author: &nameparis; MIT, Cambridge 1995 Existed in 3.30 Audio rate parameters introduced in version 6.02 October 2013. manual_src~dfsg/opcodes/midinoteoff.xml0000644000000000000000000001103412262561504017425 0ustar rootroot midinoteoff Real-time MIDI:MIDI/Score Interoperability midinoteoff midinoteoff Gets a MIDI noteoff value. Description midinoteoff is designed to simplify writing instruments that can be used interchangeably for either score or MIDI input, and to make it easier to adapt instruments originally written for score input to work with MIDI input. In general, it should be possible to write instrument definitions that work identically with both scores and MIDI, including both MIDI files and real-time MIDI input, without using any conditional statements, and that take full advantage of MIDI voice messages. Note that correlating Csound instruments with MIDI channel numbers is done using the massign opcode for real-time performance,. For file-driven performance, instrument numbers default to MIDI channel number + 1, but the defaults are overridden by any MIDI program change messages in the file. Syntax midinoteoff xkey, xvelocity Performance xkey -- returns MIDI key during MIDI activation, remains unchanged otherwise. xvelocity -- returns MIDI velocity during MIDI activation, remains unchanged otherwise. If the instrument was activated by MIDI input, the opcode overwrites the values of the xkey and xvelocity with the corresponding values from MIDI input. If the instrument was NOT activated by MIDI input, the values of xkey and xvelocity remain unchanged. This enables score p-fields to receive MIDI input data during MIDI activation, and score values otherwise. Adapting a score-activated Csound instrument. See the MIDI interop opcodes section for details on adapting score driven instruments for MIDI or vice-versa. Examples Here is an example of the midinoteoff opcode. It uses the file midinoteoff.csd. Example of the midinoteoff opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i1 60.00000 i1 76.00000 See Also midichannelaftertouch, midicontrolchange, mididefault, midinoteoncps, midinoteonkey, midinoteonoct, midinoteonpch, midipitchbend, midipolyaftertouch, midiprogramchange Credits Author: Michael Gogins New in version 4.20 manual_src~dfsg/opcodes/strlenk.xml0000644000000000000000000000203212262561504016602 0ustar rootroot strlenk Strings:Manipulation strlenk strlenk Return the length of a string Description Return the length of a string, or zero if it is empty. strlenk runs both at init and performance time. Syntax klen strlenk Sstr See also strlen Credits Author: Istvan Varga 2006 New in version 5.02 manual_src~dfsg/opcodes/readk4.xml0000644000000000000000000001414512262561504016302 0ustar rootroot readk4 Signal I/O:File I/O readk4 readk4 Periodically reads four orchestra control-signal values from an external file. Description Periodically reads four orchestra control-signal values from an external file. Syntax kr1, kr2, kr3, kr4 readk4 ifilname, iformat, iprd Initialization ifilname -- an integer N denoting a file named "readk.N" or a character string (in double quotes, spaces permitted) denoting the external file name. For a string, it may either be a full path name with directory specified or a simple filename. In the later case, the file is sought first in the current directory, then in SSDIR, and finally in SFDIR. iformat -- specifies the input data format: 1 = 8-bit signed integers (char) 4 = 16-bit short integers 5 = 32-bit long integers 6 = 32-bit floats 7 = ASCII long integers (plain text) 8 = ASCII floats (plain text) Note that A-law and U-law formats are not available, and that all formats except the last two are binary. The input file should be a "raw", headerless data file. iprd -- the rate (period) in seconds, rounded to the nearest orchestra control period, at which the signals are read from the input file. A value of 0 implies one control period (the enforced minimum), which will read new values at the orchestra control rate. Longer periods will cause the same values to repeat for more than one control period. Performance kr1, kr2, kr3, kr4 -- output of the signals read from ifilname. This opcode allows four generated control signal values to be read from a named external file. The file should contain no header information but it should contain a regularly sampled time series of control values. For binary formats, the individual samples of each signal are interleaved. For ASCII text formats, the values are assumed to be separated by at least one whitespace character. The four "channels" in a sample frame may be on the same line or separated by newline characters, it does not matter. There may be any number of readk4 opcodes in an instrument or orchestra and they may read from the same or different files. Examples Here is an example of the readk4 opcode. It uses the file readk4.csd. Example of the readk4 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output should include lines like these: WRITING: kcf = 1122.469723, kq = 11.762839, kdb = -14.313445, kpan = 0.538142 kcf = 1122.469723, kq = 11.762839, kdb = -14.313445, kpan = 0.538142 kcf = 1148.638412, kq = 12.040490, kdb = -14.061868, kpan = 0.552205 kcf = 165.796855, kq = 18.523179, kdb = -15.816977, kpan = 0.901528 kcf = 147.729960, kq = 13.071911, kdb = -11.924531, kpan = 0.982518 kcf = 497.430113, kq = 13.605512, kdb = -21.586611, kpan = 0.179229 READING: WARNING: Seeding from current time 3308160476 kcf = 1122.469700, kq = 11.762800, kdb = -14.313400, kpan = 0.538100 kcf = 1122.469700, kq = 11.762800, kdb = -14.313400, kpan = 0.538100 kcf = 1148.638400, kq = 12.040500, kdb = -14.061900, kpan = 0.552200 kcf = 165.796900, kq = 18.523200, kdb = -15.817000, kpan = 0.901500 kcf = 147.730000, kq = 13.071900, kdb = -11.924500, kpan = 0.982500 kcf = 497.430100, kq = 13.605500, kdb = -21.586600, kpan = 0.179200 See Also dumpk, dumpk2, dumpk3, dumpk4, readk, readk2, readk3 Credits By: John ffitch and Barry Vercoe 1999 or earlier manual_src~dfsg/opcodes/vaset.xml0000644000000000000000000000511112262561504016243 0ustar rootroot vaset Signal Modifiers:Sample Level Operators vaset vaset Write value of into the current buffer of an a-rate variable by index. Description Write values into the current buffer of an a-rate variable at the given index. Useful for doing sample-by-sample manipulation at k-rate without using setksmps 1. Note Because this opcode does not do any bounds checking, the user must be careful not to try to write values past ksmps (the size of a buffer for an a-rate variable) by using index values greater than ksmps. Note In Csound6 this opcode does do bounds checking. Syntax vaset kval, kndx, avar Performance kval - value to write into avar kndx - index of the sample to write to the current buffer of the given avar variable avar - a-rate variable to write to Examples Here is an example of the vaset opcode. It uses the file vaset.csd. Example of the vaset opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vaget Credits Author: Steven Yi New in version 5.04 September 2006. manual_src~dfsg/opcodes/tableigpw.xml0000644000000000000000000000364512262561504017111 0ustar rootroot tableigpw Table Control:Read/Write Opreations tableigpw tableigpw Writes a table's guard point. Description Writes a table's guard point. Syntax tableigpw ifn Initialization ifn -- Table number to be interrogated Performance tableigpw -- For writing the table's guard point, with the value which is in location 0. Does nothing if table does not exist. Likely to be useful after manipulating a table with tablemix or tablecopy. See Also tablecopy, tablegpw, tablemix, tableicopy, tableimix Credits Author: Robin Whittle Australia May 1997 New in version 3.47 manual_src~dfsg/opcodes/lowresx.xml0000644000000000000000000000665312262561504016640 0ustar rootroot lowresx Signal Modifiers:Standard Filters:Resonant lowresx lowresx Simulates layers of serially connected resonant lowpass filters. Description lowresx is equivalent to more layers of lowres with the same arguments serially connected. Syntax ares lowresx asig, xcutoff, xresonance [, inumlayer] [, iskip] Initialization inumlayer -- number of elements in a lowresx stack. Default value is 4. There is no maximum. iskip -- initial disposition of internal data space. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. Performance asig -- input signal xcutoff -- filter cutoff frequency point xresonance -- resonance amount lowresx is equivalent to more layer of lowres with the same arguments serially connected. Using a stack of a larger number of filters allows a sharper cutoff. This is faster than using a larger number of instances of lowres in a Csound orchestra because only one initialization and k cycle are needed at time and the audio loop falls entirely inside the cache memory of processor. Based on an orchestra by &namehans; Examples Here is an example of the lowresx opcode. It uses the file lowresx.csd. Example of the lowresx opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also lowres Credits Author: &namegabriel; (adapted by &namejohn;) Italy New in Csound version 3.49 Audio rate parameters introduced in version 6.02 November 2013. manual_src~dfsg/opcodes/vexpseg.xml0000644000000000000000000000535612262561504016615 0ustar rootroot vexpseg Vectorial:Envelopes vexpseg vexpseg Vectorial envelope generator Description Generate exponential vectorial segments Syntax vexpseg ifnout, ielements, ifn1, idur1, ifn2 [, idur2, ifn3 [...]] Initialization ifnout - number of table hosting output vectorial signal ifn1 - starting vector ifn2, ifn3,etc. - vector after idurx seconds idur1 - duration in seconds of first segment. idur2, idur3, etc. - duration in seconds of subsequent segments. ielements - number of elements of vectors. Performance These opcodes are similar to linseg and expseg, but operate with vectorial signals instead of with scalar signals. Output is a vectorial control signal hosted by ifnout (that must be previously allocated), while each break-point of the envelope is actually a vector of values. All break-points must contain the same number of elements (ielements). All these operators are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2, etc. Example Here is an example of the vexpseg opcode. It uses the files vexpseg.csd. Example of the vexpseg opcode. Credits Written by Gabriel Maldonado. Example by Andres Cabrera. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/STKResonate.xml0000644000000000000000000000703412262561504017271 0ustar rootroot STKResonate Signal Generators:STK Opcodes STKResonate STKResonate STKResonate is a noise driven formant filter. Description STKResonate is a noise driven formant filter. This instrument contains a noise source, which excites a biquad resonance filter, with volume controlled by an ADSR. Syntax asignal STKResonate ifrequency, iamplitude, [kfreq, kv1[, kpole, kv2[, knotch, kv3[, kzero, kv4[, kenv, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kfreq -- controller 2, frequency of resonance. Value range of kv1 is 0-127. kpole -- controller 4, pole radii. Value range of kv2 is 0-127. knotch -- controller 11, notch frequency. Value range of kv3 is 0-127. kzero -- controller 1, zero radii. Value range of kv4 is 0-127. kenv -- controller 128, gain of envelope. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the Resonate class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKResonate opcode. It uses the file STKResonate.csd. Example of the STKResonate opcode. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/sprintfk.xml0000644000000000000000000000656212262561504016774 0ustar rootroot sprintfk Strings:Manipulation sprintfk sprintfk printf-style formatted output to a string variable at k-rate. Description sprintfk writes printf-style formatted output to a string variable, similarly to the C function sprintf(). sprintfk runs both at initialization and performance time. Syntax Sdst sprintfk Sfmt, xarg1[, xarg2[, ... ]] Initialization Sfmt -- format string, has the same format as in printf() and other similar C functions, except length modifiers (l, ll, h, etc.) are not supported. The following conversion specifiers are allowed: d, i, o, u, x, X, e, E, f, F, g, G, c, s xarg1, xarg2, ... -- input arguments (max. 30) for format, should be i-rate for all conversion specifiers except %s, which requires a string argument. sprintfk also allows k-rate number arguments, but these should still be valid at init time as well (unless sprintfk is skipped with igoto). Integer formats like %d round the input values to the nearest integer. Performance Sdst -- output string variable Examples Here is an example of the sprintfk opcode. It uses the file sprintfk.csd. Example of the sprintfk opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: 1 + 1 = 2 1 + 1 + 1 = 3 1 + 1 + 1 + 1 = 4 1 + 1 + 1 + 1 + 1 = 5 1 + 1 + 1 + 1 + 1 + 1 = 6 1 + 1 + 1 + 1 + 1 + 1 + 1 = 7 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 = 8 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 = 9 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 = 10 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 = 11 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 + 1 = 12 See also sprintf, puts, strcatk Credits Author: Istvan Varga 2005 Example by Jonathan Murphy manual_src~dfsg/opcodes/wiirange.xml0000644000000000000000000000415412262561504016734 0ustar rootroot wiirange Instrument Control:Sensing and Control wiirange wiirange Sets scaling and range limits for certain Wiimote fields. Description Sets scaling and range limits for certain Wiimote fields. Syntax wiirange icontrol, iminimum, imaximum[, inum] Initialization This opcode must be used in conjuction with a running wiiconnect opcode. icontrol -- which control is to be scaled. This must be one of 20 (WII_PITCH), 21 (WII_ROLL), 30 (WII_NUNCHUK_PITCH), 31 (WII_NUNCHUK_ROLL). iminimum -- minimun value for control. imaximum -- maximum value for control. Please note that these opcodes are currently only supported on Linux. Examples See the example for wiiconnect. See Also wiiconnect, wiidata, wiisend, Credits Author: &namejohn; Codemist Ltd 2009 New in version 5.11 manual_src~dfsg/opcodes/deltapxw.xml0000644000000000000000000001041312262561504016752 0ustar rootroot deltapxw Signal Modifiers:Delay deltapxw deltapxw Mixes the input signal to a delay line. Description deltapxw mixes the input signal to a delay line. This opcode can be mixed with reading units (deltap, deltapn, deltapi, deltap3, and deltapx) in any order; the actual delay time is the difference of the read and write time. This opcode can read from and write to a delayr/delayw delay line with interpolation. Syntax deltapxw ain, adel, iwsize Initialization iwsize -- interpolation window size in samples. Allowed values are integer multiplies of 4 in the range 4 to 1024. iwsize = 4 uses cubic interpolation. Increasing iwsize improves sound quality at the expense of CPU usage, and minimum delay time. Performance ain -- Input signal. adel -- Delay time in seconds. a1 delayr idlr deltapxw a2, adl1, iws1 a3 deltapx adl2, iws2 deltapxw a4, adl3, iws3 delayw a5 Minimum and maximum delay times: idlr >= 1/kr Delay line length adl1 >= (iws1/2)/sr Write before read adl1 <= idlr - (1 + iws1/2)/sr (allows shorter delays) adl2 >= 1/kr + (iws2/2)/sr Read time adl2 <= idlr - (1 + iws2/2)/sr adl2 >= adl1 + (iws1 + iws2) / (2*sr) adl2 >= 1/kr + adl3 + (iws2 + iws3) / (2*sr) adl3 >= (iws3/2)/sr Write after read adl3 <= idlr - (1 + iws3/2)/sr (allows feedback) Note Window sizes for opcodes other than deltapx are: deltap, deltapn: 1, deltapi: 2 (linear), deltap3: 4 (cubic) Examples Here is an example of the deltapxw opcode. It uses the file deltapxw.csd. Example of the deltapxw opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also deltapx Credits Author: &nameistvan; August 2001 New in version 4.13 manual_src~dfsg/opcodes/kpow.xml0000644000000000000000000000074212262561504016106 0ustar rootroot kpow kpow kpow Deprecated. Description Deprecated as of version 3.48. Use the pow opcode instead. manual_src~dfsg/opcodes/db.xml0000644000000000000000000000444312262561504015515 0ustar rootroot db Mathematical Operations:Amplitude Functions db db Returns the amplitude equivalent for a given decibel amount. Description Returns the amplitude equivalent for a given decibel amount. This opcode is the same as ampdb. Syntax db(x) This function works at a-rate, i-rate, and k-rate. Initialization x -- a value expressed in decibels. Performance Returns the amplitude for a given decibel amount. Examples Here is an example of the db opcode. It uses the file db.csd. Example of the db opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: instr 1: iamp = 316.252 instr 1: iamp = 1211.582 instr 1: iamp = 4641.643 instr 1: iamp = 17782.420 See Also ampdb, cent, octave, semitone New in version 4.16 manual_src~dfsg/opcodes/pvsout.xml0000644000000000000000000000336112262561504016466 0ustar rootroot pvsout Spectral Processing:Streaming pvsout pvsout Write a fsig to the pvs output bus. Description This opcode writes a fsig to a channel of the pvs output bus. Note that the pvs out bus and the pvs in bus are separate and independent. A new channel is created if non-existent. Syntax pvsout fsig, kchan Performance fsig -- fsig input data. kchan -- pvs out bus channel number. Examples Example asig in ; input fsig pvsanal asig, 1024, 256, 1024, 1 ; analysis pvsout fsig, 0 ; write signal to pvs out bus channel 0 Credits Author: &namevictor; August 2006 manual_src~dfsg/opcodes/vadd_i.xml0000644000000000000000000001017712262561504016357 0ustar rootroot vadd_i Vectorial:Scalar operations vadd_i vadd_i Adds a scalar value to a vector in a table. Description Adds a scalar value to a vector in a table. Syntax vadd_i ifn, ival, ielements [, idstoffset] Initialization ifn - number of the table hosting the vectorial signal to be processed ielements - number of elements of the vector ival - scalar value to be added idstoffset - index offset for the destination table Performance vadd_i adds the value of ival to each element of the vector contained in the table ifn, starting from table index idstoffset. This enables you to process a specific section of a table by specifying the offset and the number of elements to be processed. Offset is counted starting from 0, so if no offset is specified (or set to 0), the table will be modified from the beginning. This opcode runs only on initialization, there is a k-rate version of this opcode called vadd. Negative values for idstoffset are valid. Elements from the vector that are outside the table, will be discarded, and they will not wrap around the table. In all these opcodes, the resulting vectors are stored in ifn, overriding the intial vectors. If you want to keep initial vector, use vcopy or vcopy_i to copy it in another table. All these operators are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2, etc. They can also be useful in conjunction with the spectral opcodes pvsftw and pvsftr. Examples Here is an example of the vadd_i opcode. It uses the file vadd_i.csd. Example of the vadd_i opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also vadd, vmult_i, vpow_i and vexp_i. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/sfplaym.xml0000644000000000000000000001500512262561504016577 0ustar rootroot sfplaym Signal Generators:Sample Playback sfplaym sfplaym Plays a SoundFont2 (SF2) sample preset, generating a mono sound. Description Plays a SoundFont2 (SF2) sample preset, generating a mono sound. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. Syntax ares sfplaym ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset] [, ienv] Initialization ivel -- velocity value inotenum -- MIDI note number value ipreindex -- preset index iflag (optional) -- flag regarding the behavior of xfreq and inotenum ioffset (optional) -- start playing at offset, in samples. ienv (optional) -- enables and determines amplitude envelope. 0 = no envelope, 1 = linear attack and decay, 2 = linear attack, exponential decay (see below). Default = 0. Performance xamp -- amplitude correction factor xfreq -- frequency value or frequency multiplier, depending by iflag. When iflag = 0, xfreq is a multiplier of a the default frequency, assigned by SF2 preset to the inotenum value. When iflag = 1, xfreq is the absolute frequency of the output sound, in Hz. Default is 0. When iflag = 0, inotenum sets the frequency of the output according to the MIDI note number used, and xfreq is used as a multiplier. When iflag = 1, the frequency of the output, is set directly by xfreq. This allows the user to use any kind of micro-tuning based scales. However, this method is designed to work correctly only with presets tuned to the default equal temperament. Attempts to use this method with a preset already having non-standard tunings, or with drum-kit-based presets, could give unexpected results. Adjustment of the amplitude can be done by varying the xamp argument, which acts as a multiplier. Notice that both xamp and xfreq can use k-rate as well as a-rate signals. Both arguments must use variables of the same rate, or sfplay will not work correctly. ipreindex must contain the number of a previously assigned preset, or Csound will crash. The ioffset parameter allows the sound to start from a sample different than the first one. The user should make sure that its value is within the length of the specific sound. Otherwise, Csound will probably crash. The ienv parameter enables and determines the type of amplitude envelope used. The default value is 0, or no envelope. If ienv is set to 1, the attack and decay portions are linear. If set to 2, the attack is linear and the decay is exponential. The release portion of the envelope has not yet been implemented. sfplaym is a mono version of sfplay. It should be used with mono preset, or with the stereo presets in which stereo output is not required. It is faster than sfplay. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. Examples Here is an example of the sfplaym opcode. It uses the file sfplaym.csd. Example of the sfplaym opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also sfilist, sfinstr, sfinstrm, sfload, sfpassign, sfplay, sfplay3, sfplay3m, sfplist, sfpreset Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 New optional parameter ienv in version 5.09 manual_src~dfsg/opcodes/osciliktp.xml0000644000000000000000000000635612262561504017136 0ustar rootroot osciliktp Signal Generators:Basic Oscillators osciliktp osciliktp A linearly interpolated oscillator that allows allows phase modulation. Description osciliktp allows phase modulation (which is actually implemented as k-rate frequency modulation, by differentiating phase input). The disadvantage is that there is no amplitude control, and frequency can be varied only at the control-rate. This opcode can be faster or slower than oscilikt, depending on the control-rate. Syntax ares osciliktp kcps, kfn, kphs [, istor] Initialization istor (optional, defaults to 0) -- Skips initialization. Performance ares -- audio-rate ouptut signal. kcps -- frequency in Hz. Zero and negative values are allowed. However, the absolute value must be less than sr (and recommended to be less than sr/2). kfn -- function table number. Can be varied at control rate (useful to morph waveforms, or select from a set of band-limited tables generated by GEN30). kphs -- phase (k-rate), the expected range is 0 to 1. The absolute value of the difference of the current and previous value of kphs must be less than ksmps. Examples Here is an example of the osciliktp opcode. It uses the file osciliktp.csd. Example of the osciliktp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also oscilikt and oscilikts. Credits Author: &nameistvan; New in version 4.22 manual_src~dfsg/opcodes/ATSbufread.xml0000644000000000000000000001271112262561504017105 0ustar rootroot ATSbufread Spectral Processing:ATS ATSbufread ATSbufread reads data from and ATS data file and stores it in an internal data table of frequency, amplitude pairs. Description ATSbufread reads data from and ATS data file and stores it in an internal data table of frequency, amplitude pairs. Syntax ATSbufread ktimepnt, kfmod, iatsfile, ipartials[, ipartialoffset, \ ipartialincr] Initialization iatsfile – the ATS number (n in ats.n) or the name in quotes of the analysis file made using ATS. ipartials – number of partials that will be used in the resynthesis (the noise has a maximum of 25 bands) ipartialoffset (optional) – is the first partial used (defaults to 0). ipartialincr (optional) – sets an increment by which these synthesis opcodes counts up from ipartialoffset for ibins components in the re-synthesis (defaults to 1). Performance ktimepnt – The time pointer in seconds used to index the ATS file. Used for ATSbufread exactly the same as for pvoc. kfmod – an input for performing pitch transposition or frequency modulation on all of the synthesized partials, if no fm or pitch change is desired then use a 1 for this value. ATSbufread is based on pvbufread by Richard Karpen. ATScross, ATSinterpread and ATSpartialtap are all dependent on ATSbufread just as pvcross and pvinterp are on pvbufread. ATSbufread reads data from and ATS data file and stores it in an internal data table of frequency, amplitude pairs. The data stored by an ATSbufread can only be accessed by other unit generators, and therefore, due to the architecture of Csound, an ATSbufread must come before (but not necessarily directly) any dependent unit generator. Besides the fact that ATSbufread doesn't output any data directly, it works almost exactly as ATSadd. The ugen uses a time pointer (ktimepnt) to index the data in time, ipartials, ipartialoffset and ipartialincr to select which partials to store in the table and kfmod to scale partials in frequency. Examples Here is an example of the ATSbufread opcode. It uses the file ATSbufread.csd. Example of the ATSbufread opcode. See also the examples for ATScross, ATSinterpread and ATSpartialtap See also ATSread, ATSreadnz, ATSinfo, ATSsinnoi, ATScross, ATSinterpread, ATSpartialtap, ATSadd, ATSaddnz Credits Author: Alex Norman Seattle,Washington 2004 manual_src~dfsg/opcodes/mp3len.xml0000644000000000000000000000375212262561504016330 0ustar rootroot mp3len Signal I/O:Soundfile Queries mp3len mp3len Returns the length of an MP3 sound file. Description Returns the length of an MP3 sound file. Syntax ir mp3len ifilcod Initialization ifilcod -- sound file to be queried Performance mp3len returns the length of the sound file ifilcod in seconds. Examples Here is an example of the mp3len opcode. It uses the file mp3len.csd. Example of the mp3len opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also mp3in Credits Author: &namejohn; Feb 2011 New in Csound version 5.14 manual_src~dfsg/opcodes/inx.xml0000644000000000000000000000374612262561504015733 0ustar rootroot inx Signal I/O:Signal Input inx inx Reads a 16-channel audio signal from an external device or stream. Description Reads a 16-channel audio signal from an external device or stream. Warning This opcode is designed to be used only with orchestras that have nchnls=16. Doing so with orchestras with nchnls > 16 will cause incorrect audio input. Syntax ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, \ ar13, ar14, ar15, ar16 inx Performance inx reads a 16-channel audio signal from an external device or stream. If the command-line -i flag is set, sound is read continuously from the audio input stream (e.g. stdin or a soundfile) into an internal buffer. See Also in32, inch, inz Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/ftconv.xml0000644000000000000000000000773612262561504016437 0ustar rootroot ftconv Signal Modifiers:Convolution and Morphing ftconv ftconv Low latency multichannel convolution, using a function table as impulse response source. Description Low latency multichannel convolution, using a function table as impulse response source. The algorithm is to split the impulse response to partitions of length determined by the iplen parameter, and delay and mix partitions so that the original, full length impulse response is reconstructed without gaps. The output delay (latency) is iplen samples, and does not depend on the control rate, unlike in the case of other convolve opcodes. Syntax a1[, a2[, a3[, ... a8]]] ftconv ain, ift, iplen[, iskipsamples \ [, iirlen[, iskipinit]]] Initialization ift -- source ftable number. The table is expected to contain interleaved multichannel audio data, with the number of channels equal to the number of output variables (a1, a2, etc.). An interleaved table can be created from a set of mono tables with GEN52. iplen -- length of impulse response partitions, in sample frames; must be an integer power of two. Lower settings allow for shorter output delay, but will increase CPU usage. iskipsamples (optional, defaults to zero) -- number of sample frames to skip at the beginning of the table. Useful for reverb responses that have some amount of initial delay. If this delay is not less than iplen samples, then setting iskipsamples to the same value as iplen will eliminate any additional latency by ftconv. iirlen (optional) -- total length of impulse response, in sample frames. The default is to use all table data (not including the guard point). iskipinit (optional, defaults to zero) -- if set to any non-zero value, skip initialization whenever possible without causing an error. Performance ain -- input signal. a1 ... a8 -- output signal(s). Example Here is an example of the ftconv opcode. It uses the file ftconv.csd. Example of the ftconv opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also pconvolve, convolve, dconv. Credits Author: Istvan Varga 2005 manual_src~dfsg/opcodes/serialBegin.xml0000644000000000000000000000565712262561504017364 0ustar rootroot serialBegin Serial I/O serialBegin serialBegin Open a serial port. Description Open a serial port for arduino. Syntax iPort serialBegin SPortName [, ibaudRate] Initialization SPortName -- port name number ibaudrate -- serial speed, defaulting to 9600 bps. Examples Here is an example of the serialBegin opcode. It uses the file serialBegin.csd. Example of the serialBegin opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. This is the matching Arduino code : void setup() { // enable serial communication Serial.begin(9600); // declare pin 9 to be an output: pinMode(9, OUTPUT); } void loop() { // only do something if we received something (this should be at csound's k-rate) if (Serial.available()) { // set the brightness of LED (connected to pin 9) to our input value int brightness = Serial.read(); analogWrite(9, brightness); // while we are here, get our knob value and send it to csound int sensorValue = analogRead(A0); Serial.write(sensorValue/4); // scale to 1-byte range (0-255) } } ..... See Also serialEnd, serialWrite_i, serialWrite, serialRead, serialPrint, serialFlush. Credits Author: &namematt; 2011 New in version 5.14 manual_src~dfsg/opcodes/dssiinit.xml0000644000000000000000000000643712262561504016763 0ustar rootroot dssiinit Plugin Hosting:DSSI and LADSPA dssiinit dssiinit Loads a DSSI or LADSPA plugin. Syntax ihandle dssiinit ilibraryname, iplugindex [, iverbose] Description dssiinit is used to load a DSSI or LADSPA plugin into memory for use with the other dssi4cs opcodes. Both LADSPA effects and DSSI instruments can be used. Initialization ihandle - the number which identifies the plugin, to be passed to other dssi4cs opcodes. ilibraryname - the name of the .so (shared object) file to load. iplugindex - The index of the plugin to be used. iverbose (optional) - show plugin information and parameters when loading. (default = 1) dssiinit looks for ilibraryname on LADSPA_PATH and DSSI_PATH. One of these variables must be set, otherwise dssiinit will return an error. LADSPA and DSSI libraries may contain more than one plugin which must be referenced by its index. dssiinit then attempts to find plugin index iplugindex in the library and load the plugin into memory if it is found. To find out which plugins you have available and their index numbers you can use: dssilist. If iverbose is not 0 (the default), information about the plugin detailing its characteristics and its ports will be shown. This information is important for opcodes like dssictls. Plugins are set to inactive by default, so you *must* use dssiactivate to get the plugin to produce sound. This is required even if the plugin doesn't provide an activate() function. dssiinit may cause audio stream breakups when used in realtime, so it is recommended to load all plugins to be used before playing. Examples Here is an example of the dssinit opcode. It uses the file dssiinit.csd. Example of the dssiinit opcode. (Remember to change the Library name) Credits 2005 By: &nameandres; Uses code from Richard Furse's LADSPA sdk. manual_src~dfsg/opcodes/compress.xml0000644000000000000000000001203412262561504016756 0ustar rootroot compress Signal Modifiers:Amplitude Modifiers compress compress Compress, limit, expand, duck or gate an audio signal. Description This unit functions as an audio compressor, limiter, expander, or noise gate, using either soft-knee or hard-knee mapping, and with dynamically variable performance characteristics. It takes two audio input signals, aasig and acsig, the first of which is modified by a running analysis of the second. Both signals can be the same, or the first can be modified by a different controlling signal. compress first examines the controlling acsig by performing envelope detection. This is directed by two control values katt and krel, defining the attack and release time constants (in seconds) of the detector. The detector rides the peaks (not the RMS) of the control signal. Typical values are .01 and .1, the latter usually being similar to ilook. The running envelope is next converted to decibels, then passed through a mapping function to determine what compresser action (if any) should be taken. The mapping function is defined by four decibel control values. These are given as positive values, where 0 db corresponds to an amplitude of 1, and 90 db corresponds to an amplitude of 32768. Syntax ar compress aasig, acsig, kthresh, kloknee, khiknee, kratio, katt, krel, ilook Initialization ilook -- lookahead time in seconds, by which an internal envelope release can sense what is coming. This induces a delay between input and output, but a small amount of lookahead improves the performance of the envelope detector. Typical value is .05 seconds, sufficient to sense the peaks of the lowest frequency in acsig. Performance kthresh -- sets the lowest decibel level that will be allowed through. Normally 0 or less, but if higher the threshold will begin removing low-level signal energy such as background noise. kloknee, khiknee -- decibel break-points denoting where compression or expansion will begin. These set the boundaries of a soft-knee curve joining the low-amplitude 1:1 line and the higher-amplitude compression ratio line. Typical values are 48 and 60 db. If the two breakpoints are equal, a hard-knee (angled) map will result. kratio -- ratio of compression when the signal level is above the knee. The value 2 will advance the output just one decibel for every input gain of two; 3 will advance just one in three; 20 just one in twenty, etc. Inverse ratios will cause signal expansion: .5 gives two for one, .25 four for one, etc. The value 1 will result in no change. The actions of compress will depend on the parameter settings given. A hard-knee compressor-limiter, for instance, is obtained from a near-zero attack time, equal-value break-points, and a very high ratio (say 100). A noise-gate plus expander is obtained from some positive threshold, and a fractional ratio above the knee. A voice-activated music compressor (ducker) will result from feeding the music into aasig and the speech into acsig. A voice de-esser will result from feeding the voice into both, with the acsig version being preceded by a band-pass filter that emphasizes the sibilants. Each application will require some experimentation to find the best parameter settings; these have been made k-variable to make this practical. Examples Here is an example of the compress opcode. It uses the file compress.csd. Example of the compress opcode. See Also dam Credits Written by &namebarry; for Extended Csound and released in csound5.02. manual_src~dfsg/opcodes/max_k.xml0000644000000000000000000000542412262561504016227 0ustar rootroot max_k Signal Modifiers:Comparators and Accumulators max_k max_k Local maximum (or minimum) value of an incoming asig signal Description max_k outputs the local maximum (or minimum) value of the incoming asig signal, checked in the time interval between ktrig has become true twice. Syntax knumkout max_k asig, ktrig, itype Initialization itype - itype determinates the behaviour of max_k (see below) Performance asig - incoming (input) signal ktrig - trigger signal max_k outputs the local maximum (or minimum) value of the incoming asig signal, checked in the time interval between ktrig has become true twice. itype determinates the behaviour of max_k: 1 - absolute maximum (sign of negative values is changed to positive before evaluation). 2 - actual maximum. 3 - actual minimum. 4 - calculate average value of asig in the time interval since the last trigger. This opcode can be useful in several situations, for example to implement a vu-meter. Examples Here is an example of the max_k opcode. It uses the file max_k.csd. Example of the max_k opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by &namegabriel;. New in Csound 5 (Previously available only on CsoundAV) Fixed to agree with documentation in 5.15 manual_src~dfsg/opcodes/pvsmorph.xml0000644000000000000000000000740312262561504017005 0ustar rootroot pvsmorph Spectral Processing:Streaming pvsmorph pvsmorph Performs morphing (or interpolation) between two source fsigs. Description Performs morphing (or interpolation) between two source fsigs. Syntax fsig pvsmorph fsig1, fsig2, kampint, kfrqint Performance The operation of this opcode is similar to that of pvinterp (q.v.), except in using fsigs rather than analysis files, and the absence of spectral envelope preservation. The amplitudes and frequencies of fsig1 are interpolated with those of fsig2, depending on the values of kampint and kfrqint, respectively. These range between 0 and 1, where 0 means fsig1 and 1, fsig2. Anything in between will interpolate amps and/or freqs of the two fsigs. With this opcode, morphing can be performed on real-time audio input, by using pvsanal to generate fsig1 and fsig2. These must have the same format. It is unsafe to use the same f-variable for both input and output of pvs opcodes. Using the same one might lead to undefined behavior on some opcodes. Use a different one on the left and right sides of the opcode. Examples Here is an example of the pvsmorph opcode. It uses the file pvsmorph.csd. Example of the pvsmorph opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the pvsmorph opcode. It uses the file pvsmorph2.csd. Example of the pvsmorph opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also pvsanal, pvsynth, pvsadsyn Credits Author: &namevictor; April 2007 New in Csound 5.06 manual_src~dfsg/opcodes/adsynt2.xml0000644000000000000000000001117012262561504016507 0ustar rootroot adsynt2 Signal Generators:Additive Synthesis/Resynthesis adsynt2 adsynt2 Performs additive synthesis with an arbitrary number of partials -not necessarily harmonic- with interpolation. Description Performs additive synthesis with an arbitrary number of partials, not necessarily harmonic. (see adsynt for detailed manual) Syntax ar adsynt2 kamp, kcps, iwfn, ifreqfn, iampfn, icnt [, iphs] Initialization iwfn -- table containing a waveform, usually a sine. Table values are not interpolated for performance reasons, so larger tables provide better quality. ifreqfn -- table containing frequency values for each partial. ifreqfn may contain beginning frequency values for each partial, but is usually used for generating parameters at runtime with tablew. Frequencies must be relative to kcps. Size must be at least icnt. iampfn -- table containing amplitude values for each partial. iampfn may contain beginning amplitude values for each partial, but is usually used for generating parameters at runtime with tablew. Amplitudes must be relative to kamp. Size must be at least icnt. icnt -- number of partials to be generated iphs -- initial phase of each oscillator, if iphs = -1, initialization is skipped. If iphs > 1, all phases will be initialized with a random value. Performance kamp -- amplitude of note kcps -- base frequency of note. Partial frequencies will be relative to kcps. Frequency and amplitude of each partial is given in the two tables provided. The purpose of this opcode is to have an instrument generate synthesis parameters at k-rate and write them to global parameter tables with the tablew opcode. adsynt2 is identical to adsynt (by Peter Neubäcker), except it provides linear interpolation for amplitude envelopes of each partial. It is a bit slower than adsynt, but interpolation higly improves sound quality in fast amplitude envelope transients when kr < sr (i.e. when ksmps > 1). No interpolation is provided for pitch envelopes, since in this case sound quality degradation is not so evident even with high values of ksmps. It is not recommended when kr = sr, in this case adsynt is better (since it is faster). Examples Here is an example of the adsynt2 opcode. It uses the file adsynt2.csd. These two instruments perform additive synthesis. The output of each sounds like a Tibetan bowl. The first one is static, as parameters are only generated at init-time. In the second one, parameters are continuously changed. Example of the adsynt2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by &namegabriel;. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/tablei.xml0000644000000000000000000001005212262561504016361 0ustar rootroot tablei Signal Generators:Table Access tablei tablei Accesses table values by direct indexing with linear interpolation. Description Accesses table values by direct indexing with linear interpolation. Syntax ares tablei andx, ifn [, ixmode] [, ixoff] [, iwrap] ires tablei indx, ifn [, ixmode] [, ixoff] [, iwrap] kres tablei kndx, ifn [, ixmode] [, ixoff] [, iwrap] Initialization ifn -- function table number. tablei requires the extended guard point. ixmode (optional) -- index data mode. The default value is 0. 0 = raw index 1 = normalized (0 to 1) ixoff (optional) -- amount by which index is to be offset. For a table with origin at center, use tablesize/2 (raw) or .5 (normalized). The default value is 0. iwrap (optional) -- wraparound index flag. The default value is 0. 0 = nowrap (index < 0 treated as index=0; index > tablesize sticks at index=size) 1 = wraparound. Performance tablei is a interpolating unit in which the fractional part of index is used to interpolate between adjacent table entries. The smoothness gained by interpolation is at some small cost in execution time (see also oscili, etc.), but the interpolating and non-interpolating units are otherwise interchangeable. Note that when tablei uses a periodic index whose modulo n is less than the power of 2 table length, the interpolation process requires that there be an (n + 1)th table value that is a repeat of the 1st (see f Statement in score). Examples Here is an example of the tablei opcode. It uses the file tablei.csd. Example of the tablei opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also table, table3, oscil1, oscil1i, osciln More information on this opcode: http://www.csounds.com/journal/issue12/genInstruments.html , written by Jacob Joaquin manual_src~dfsg/opcodes/tbvcf.xml0000644000000000000000000000621612262561504016234 0ustar rootroot tbvcf Signal Modifiers:Standard Filters:Resonant tbvcf tbvcf Models some of the filter characteristics of a Roland TB303 voltage-controlled filter. Description This opcode attempts to model some of the filter characteristics of a Roland TB303 voltage-controlled filter. Euler's method is used to approximate the system, rather than traditional filter methods. Cutoff frequency, Q, and distortion are all coupled. Empirical methods were used to try to unentwine, but frequency is only approximate as a result. Future fixes for some problems with this opcode may break existing orchestras relying on this version of tbvcf. Syntax ares tbvcf asig, xfco, xres, kdist, kasym [, iskip] Initialization iskip (optional, default=0) -- if non zero skip the initialisation of the filter. (New in Csound version 4.23f13 and 5.0) Performance asig -- input signal. Should be normalized to ±1. xfco -- filter cutoff frequency. Optimum range is 10,000 to 1500. Values below 1000 may cause problems. xres -- resonance or Q. Typically in the range 0 to 2. kdist -- amount of distortion. Typical value is 2. Changing kdist significantly from 2 may cause odd interaction with xfco and xres. kasym -- asymmetry of resonance. Typically in the range 0 to 1. Examples Here is an example of the tbvcf opcode. It uses the file tbvcf.csd. Example of the tbvcf opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namehans; December, 2000 -- January, 2001 New in Csound 4.10 manual_src~dfsg/opcodes/button.xml0000644000000000000000000000316112262561504016437 0ustar rootroot button Instrument Control:Sensing and Control button button Sense on-screen controls. Description Sense on-screen controls. Requires Winsound or TCL/TK. Syntax kres button knum Performance Note that this opcode is not available on Windows due to the implimentation of pipes on that system kres -- value of the button control. If the button has been pushed since the last k-period, then return 1, otherwise return 0. knum -- the number of the button. If it does not exist, it is made on-screen at initialization. See Also checkbox Credits Author: &namejohn; University of Bath, Codemist. Ltd. Bath, UK September 2000 New in Csound version 4.08 manual_src~dfsg/opcodes/specdiff.xml0000644000000000000000000000426312262561504016713 0ustar rootroot specdiff Spectral Processing:Non-Standard specdiff specdiff Finds the positive difference values between consecutive spectral frames. Description Finds the positive difference values between consecutive spectral frames. Syntax wsig specdiff wsigin Performance wsig -- the output spectrum. wsigin -- the input spectra. Finds the positive difference values between consecutive spectral frames. At each new frame of wsigin, each magnitude value is compared with its predecessor, and the positive changes written to the output spectrum. This unit is useful as an energy onset detector. Examples wsig2 specdiff wsig1 ; sense onsets wsig3 specfilt wsig2, 2 ; absorb slowly specdisp wsig2, 0.1 ; & display both spectra specdisp wsig3, 0.1 See Also specaddm, specfilt, spechist, specscal manual_src~dfsg/opcodes/midictrl.xml0000644000000000000000000000622512262561504016737 0ustar rootroot midictrl Real-time MIDI:Input midictrl midictrl Get the current value (0-127) of a specified MIDI controller. Description Get the current value (0-127) of a specified MIDI controller. Syntax ival midictrl inum [, imin] [, imax] kval midictrl inum [, imin] [, imax] Initialization inum -- MIDI controller number (0-127) imin, imax -- set minimum and maximum limits on values obtained. Performance Get the current value (0-127) of a specified MIDI controller. Warning midictrl should only be used in notes that were triggered from MIDI, so that an associated channel number is available. For notes activated from the score, line events, or orchestra, the ctrl7 opcode that takes an explicit channel number should be used instead. Examples Here is an example of the midictrl opcode. It uses the file midictrl.csd Example of the midictrl opcode. See Also aftouch, ampmidi, cpsmidi, cpsmidib, notnum, octmidi, octmidib, pchbend, pchmidi, pchmidib, veloc Credits Author: &namebarry; - &namemike; MIT - Mills May 1997 Thanks goes to &namerasmus; for pointing out the correct controller number range. manual_src~dfsg/opcodes/flvslidbnk2.xml0000644000000000000000000002113212262561504017342 0ustar rootroot FLvslidBnk2 FLTK:Other FLvslidBnk2 FLvslidBnk2 A FLTK widget containing a bank of vertical sliders. Description FLvslidBnk2 is a widget containing a bank of vertical sliders. Syntax FLvslidBnk2 "names", inumsliders, ioutable, iconfigtable [,iwidth, iheight, ix, iy, istart_index] Initialization names -- a double-quoted string containing the names of each slider. Each slider can have a different name. Separate each name with @ character, for example: frequency@amplitude@cutoff. It is possible to not provide any name by giving a single space . In this case, the opcode will automatically assign a progressive number as a label for each slider. inumsliders -- the number of sliders. ioutable (optional, default=0) -- number of a previously-allocated table in which to store output values of each slider. The user must be sure that table size is large enough to contain all output cells, otherwise a segfault will crash Csound. By assigning zero to this argument, the output will be directed to the zak space in the k-rate zone. In this case, the zak space must be previously allocated with the zakinit opcode and the user must be sure that the allocation size is big enough to cover all sliders. The default value is zero (i.e. store output in zak space). iconfigtable -- in the FLslidBnk2 and FLvslidBnk2 opcodes, this table replaces iminmaxtable, iexptable and istyletable, all these parameters being placed into a single table. This table has to be filled with a group of 4 parameters for each slider in this way: min1, max1, exp1, style1, min2, max2, exp2, style2, min3, max3, exp3, style3 etc. for example using GEN02 you can type: inum ftgen 1,0,256, -2, 0,1,0,1, 100, 5000, -1, 3, 50, 200, -1, 5,….. [etcetera] In this example the first slider will be affected by the [0,1,0,1] parameters (the range will be 0 to 1, it will have linear response, and its aspect will be a fill slider), the second slider will be affected by the [100,5000,-1,3] parameters (the range is 100 to 5000, the response is exponential and the aspect is a normal slider), the third slider will be affected by the [50,200,-1,5] parameters (the range is 50 to 200, the behavior exponential, and the aspect is a nice slider), and so on. iwidth (optional) -- width of the rectangular area containing all sliders of the bank, excluding text labels, that are placed below that area. iheight (optional) -- height of the rectangular area containing all sliders of the bank, excluding text labels, that are placed below that area. ix (optional) -- horizontal position of the upper left corner of the rectangular area containing all sliders belonging to the bank. You have to leave enough space, below that rectangle, in order to make sure labels of sliders to be visible. This is because the labels themselves are external to the rectangular area. iy (optional) -- vertical position of the upper left corner of the rectangular area containing all sliders belonging to the bank. You have to leave enough space, below that rectangle, in order to make sure labels of sliders to be visible. This is because the labels themselves are external to the rectangular area. istart_index (optional, default=0) -- an integer number referring to a starting offset of output cell locations. It can be positive to allow multiple banks of sliders to output in the same table or in the zak space. The default value is zero (no offset). Performance There are no k-rate arguments, even if cells of the output table (or the zak space) are updated at k-rate. FLvslidBnk2 is a widget containing a bank of vertical sliders. Any number of sliders can be placed into the bank (inumsliders argument). The output of all sliders is stored into a previously allocated table or into the zak space (ioutable argument). It is possible to determine the first location of the table (or of the zak space) in which to store the output of the first slider by means of istart_index argument. Each slider can have an individual label that is placed below it. Labels are defined by the names argument. The output range of each slider can be individually set by means of the min and max values inside the iconfigtable table. The curve response of each slider can be set individually, by means of a list of identifiers placed in the iconfigtable table (exp argument). It is possible to define the aspect of each slider independently or to make all sliders have the same aspect (style argument in the iconfigtable table). The iwidth, iheight, ix, and iy arguments determine width, height, horizontal and vertical position of the rectangular area containing sliders. Notice that the label of each slider is placed below them and is not included in the rectangular area containing sliders. So the user should leave enough space below the bank by assigning a proper iy value in order to leave labels visible. FLvslidBnk2 is identical to FLslidBnk2 except it contains vertical sliders instead of horizontal. Since the width of each single slider is often small, it is recommended to leave only a single space in the names string (“ “), in this case each slider will be automatically numbered. IMPORTANT! Notice that the tables used by FLvslidBnk2 must be created with the ftgen opcode and placed in the orchestra before the corresponding valuator. They can not placed in the score. This is because tables placed in the score are created later than the initialization of the opcodes placed in the header section of the orchestra. See Also FLslider, FLslidBnk, FLslidBnk2, FLvslidBnk2 Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/lineto.xml0000644000000000000000000000561512262561504016424 0ustar rootroot lineto Signal Modifiers:Standard Filters:Control lineto lineto Generate glissandos starting from a control signal. Description Generate glissandos starting from a control signal. Syntax kres lineto ksig, ktime Performance kres -- Output signal. ksig -- Input signal. ktime -- Time length of glissando in seconds. lineto adds glissando (i.e. straight lines) to a stepped input signal (for example, produced by randh or lpshold). It generates a straight line starting from previous step value, reaching the new step value in ktime seconds. When the new step value is reached, such value is held until a new step occurs. Be sure that ktime argument value is smaller than the time elapsed between two consecutive steps of the original signal, otherwise discontinuities will occur in output signal. When used together with the output of lpshold it emulates the glissando effect of old analog sequencers. No new value for ksig or ktime will have effect until the previous ktime has elapsed. Examples Here is an example of the lineto opcode. It uses the file lineto.csd. Example of the lineto opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also tlineto Credits Author: &namegabriel; New in Version 4.13 manual_src~dfsg/opcodes/vtabwi.xml0000644000000000000000000000412312262561504016417 0ustar rootroot vtabwi Vectorial:Tables vtabwi vtabwi Write vectors (to tables -or arrays of vectors). Description This opcode writes vectors to tables at init time. Syntax vtabwi indx, ifn, inarg1 [, inarg2, inarg3 , .... , inargN ] Initialization indx - Index into f-table, a positive number range matching the table length ifn - table number inarg1...inargN - input vector elements Performance This opcode is useful in all cases in which one needs to write sets of values associated to unique indexes (for example, multi-channel samples, STFT bin frames, spectral formants, p-field based scores etc.). The number of elements of each vector (length of the vector) is determined by the number of optional arguments on the right (inarg1, inarg2, inarg3, .... inargN). Notice that no wrap nor limit mode is implemented. So, if an index attempt to access to a zone not allocated by the table, Csound will probably crash. However this drawback can be easily avoided by using wrap or limit opcodes applied to indexes before using vtabwi, in order to correct eventual out-of-range values. Credits Written by Gabriel Maldonado. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/wrap.xml0000644000000000000000000000607312262561504016102 0ustar rootroot wrap Signal Modifiers:Signal Limiters wrap wrap Wraps-around the signal that exceeds the low and high thresholds. Description Wraps-around the signal that exceeds the low and high thresholds. Syntax ares wrap asig, klow, khigh ires wrap isig, ilow, ihigh kres wrap ksig, klow, khigh Initialization isig -- input signal ilow -- low threshold ihigh -- high threshold Performance xsig -- input signal klow -- low threshold khigh -- high threshold wrap wraps-around the signal that exceeds the low and high thresholds. This opcode is useful in several situations, such as table indexing or for clipping and modeling a-rate, i-rate or k-rate signals. wrap is also useful for wrap-around of table data when the maximum index is not a power of two (see table and tablei). Another use of wrap is in cyclical event repeating, with arbitrary cycle length. Examples Here is an example of the wrap opcode. It uses the file wrap.csd. Example of the wrap opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also limit, mirror Credits Author: &namegabriel; Italy New in Csound version 3.49 manual_src~dfsg/opcodes/vsubv.xml0000644000000000000000000001412612262561504016274 0ustar rootroot vsubv Vectorial:Vectorial operations vsubv vsubv Performs subtraction between two vectorial control signals Description Performs subtraction between two vectorial control signals Syntax vsubv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] Initialization ifn1 - number of the table hosting the first vector to be processed ifn2 - number of the table hosting the second vector to be processed Performance kelements - number of elements of the two vectors kdstoffset - index offset for the destination (ifn1) table (Default=0) ksrcoffset - index offset for the source (ifn2) table (Default=0) kverbose - Selects whether or not warnings are printed (Default=0) vsubv subtracts two vectorial control signals, that is, each element of ifn2 is substracted from the corresponding element of ifn1. Each vectorial signal is hosted by a table (ifn1 and ifn2). The number of elements contained in both vectors must be the same. The result is a new vectorial control signal that overrides old values of ifn1. If you want to keep the old ifn1 vector, use vcopy_i opcode to copy it in another table. You can use kdstoffset and ksrcoffset to specify vectors in any location of the tables. Negative values for kdstoffset and ksrcoffset are acceptable. If kdstoffset is negative, the out of range section of the vector will be discarded. If ksrcoffset is negative, the out of range elements will be assumed to be 0 (i.e. the destination elements will not be changed). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 0 (i.e. the destination vector will not be changed for these elements). If the optional kverbose argument is different to 0, the opcode will print warning messages every k-pass if table lengths are exceeded. Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at k-rate (this means that every k-pass the vectors are subtracted). There's an i-rate version of this opcode called vsubv_i. Please note that the elements argument has changed in version 5.03 from i-rate to k-rate. This will change the opcode's behavior in the unusual cases where the i-rate variable ielements is changed inside the instrument, for example in: instr 1 ielements = 10 vadd 1, 1, ielements ielements = 20 vadd 2, 1, ielements turnoff endin All these operators (vaddv, vsubv, vmultv, vdivv, vpowv, vexpv, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2, etc. Examples Here is an example of the vsubv opcode. It uses the file vsubv.csd. Example of the vsubv opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/prealloc.xml0000644000000000000000000000514112262561504016725 0ustar rootroot prealloc Instrument Control:Realtime Performance Control prealloc prealloc Creates space for instruments but does not run them. Description Creates space for instruments but does not run them. Syntax prealloc insnum, icount prealloc "insname", icount Initialization insnum -- instrument number icount -- number of instrument allocations insname -- A string (in double-quotes) representing a named instrument. Performance All instances of prealloc must be defined in the header section, not in the instrument body. Examples Here is an example of the prealloc opcode. It uses the file prealloc.csd. Example of the prealloc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also cpuprc, maxalloc Credits Author: &namegabriel; Italy July 1999 Example written by &namekevin;. New in Csound version 3.57 manual_src~dfsg/opcodes/midiout.xml0000644000000000000000000000563612262561504016607 0ustar rootroot midiout Real-time MIDI:Generic I/O midiout midiout Sends a generic MIDI message to the MIDI OUT port. Description Sends a generic MIDI message to the MIDI OUT port. Syntax midiout kstatus, kchan, kdata1, kdata2 Performance kstatus -- the type of MIDI message. Can be: 128 (note off) 144 (note on) 160 (polyphonic aftertouch) 176 (control change) 192 (program change) 208 (channel aftertouch) 224 (pitch bend) 0 when no MIDI messages must be sent to the MIDI OUT port kchan -- MIDI channel (1-16) kdata1, kdata2 -- message-dependent data values midiout has no output arguments, because it sends a message to the MIDI OUT port implicitly. It works at k-rate. It sends a MIDI message only when kstatus is non-zero. Warning: Normally kstatus should be set to 0. Only when the user intends to send a MIDI message, can it be set to the corresponding message type number. Examples Here is an example of the midiout opcode. It uses the file midiout.csd. Example of the midiout opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; Italy 1998 New in Csound version 3.492 manual_src~dfsg/opcodes/ftgentmp.xml0000644000000000000000000000755212262561504016760 0ustar rootroot ftgentmp Table Control ftgentmp ftgentmp Generate a score function table from within the orchestra, which is deleted at the end of the note. Description Generate a score function table from within the orchestra, which is optionally deleted at the end of the note. Syntax ifno ftgentmp ip1, ip2dummy, isize, igen, iarga, iargb, ... Initialization ifno -- either a requested or automatically assigned table number above 100. ip1 -- the number of the table to be generated or 0 if the number is to be assigned, in which case the table is deleted at the end of the note activation. ip2dummy -- ignored. isize -- table size. Corresponds to p3 of the score f statement. igen -- function table GEN routine. Corresponds to p4 of the score f statement. iarga, iargb, ... -- function table arguments. Correspond to p5 through pn of the score f statement. Csound was originally designed to support tables with power of two sizes only. Though this has changed in recent versions (you can use any size by using a negative number), many opcodes will not accept them. Examples Here is an example of the ftgentmp opcode. It uses the file ftgentmp.csd. Example of the ftgentmp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The output of this csd shows that the tables are destroyed after the instrument instances that created them terminates, causing an init error if the tables are queried. SECTION 1: new alloc for instr 1: ftable 101: instr 1: ifno = 101.000 B 0.000 .. 2.000 T 2.000 TT 2.000 M: 0.00000 0.00000 new alloc for instr 2: instr 2: #i0 = 512.000 B 2.000 .. 5.000 T 5.001 TT 5.001 M: 0.00000 0.00000 new alloc for instr 1: ftable 102: instr 1: ifno = 102.000 B 5.000 .. 7.000 T 7.001 TT 7.001 M: 0.00000 0.00000 instr 2: #i0 = 512.000 B 7.000 .. 12.000 T 11.999 TT 11.999 M: 0.00000 0.00000 INIT ERROR in instr 2: Invalid ftable no. 101.000000 #i0 ftlen.i p4 instr 2: #i0 = -1.000 B 12.000 - note deleted. i2 had 1 init errors B 12.000 .. 17.000 T 17.000 TT 17.000 M: 0.00000 0.00000 INIT ERROR in instr 2: Invalid ftable no. 102.000000 #i0 ftlen.i p4 instr 2: #i0 = -1.000 B 17.000 - note deleted. i2 had 1 init errors Credits Authors: &nameistvan; 2005 manual_src~dfsg/opcodes/biquada.xml0000644000000000000000000000427712262561504016543 0ustar rootroot biquada Signal Modifiers:Standard Filters biquada biquada A sweepable general purpose biquadratic digital filter with a-rate parameters. Description A sweepable general purpose biquadratic digital filter. Syntax ares biquada asig, ab0, ab1, ab2, aa0, aa1, aa2 [, iskip] Initialization iskip (optional, default=0) -- if non-zero, intialization will be skipped. Default value 0. (New in Csound version 3.50) Performance asig -- input signal biquada is a general purpose biquadratic digital filter of the form: a0*y(n) + a1*y[n-1] + a2*y[n-2] = b0*x[n] + b1*x[n-1] + b2*x[n-2] This filter has the following frequency response: B(Z) b0 + b1*Z-1 + b2*Z-2 H(Z) = ---- = ------------------ A(Z) a0 + a1*Z-1 + a2*Z-2 This type of filter is often encountered in digital signal processing literature. It allows six user-defined a-rate coefficients. See Also biquad Credits Author: &namehans; October 1998 New in Csound version 3.49 manual_src~dfsg/opcodes/splitrig.txt0000644000000000000000000000137212262561504017002 0ustar rootrootgi1 ftgen 1,0,1024, -2 \ ; table is generated with GEN02 in this case \ ; numtics_of_pattern_1, \ ;pattern 1 tic1_out1, tic1_out2, ... , tic1_outN,\ tic2_out1, tic2_out2, ... , tic2_outN,\ tic3_out1, tic3_out2, ... , tic3_outN,\ ..... ticN_out1, ticN_out2, ... , ticN_outN,\ \ numtics_of_pattern_2, \ ;pattern 2 tic1_out1, tic1_out2, ... , tic1_outN,\ tic2_out1, tic2_out2, ... , tic2_outN,\ tic3_out1, tic3_out2, ... , tic3_outN,\ ..... ticN_out1, ticN_out2, ... , ticN_outN,\ ..... \ numtics_of_pattern_N,\ ;pattern N tic1_out1, tic1_out2, ... , tic1_outN,\ tic2_out1, tic2_out2, ... , tic2_outN,\ tic3_out1, tic3_out2, ... , tic3_outN,\ ..... ticN_out1, ticN_out2, ... , ticN_outN,\manual_src~dfsg/opcodes/strchar.xml0000644000000000000000000000336212262561504016575 0ustar rootroot strchar Strings:Conversion strchar strchar Return the ASCII code of a character in a string Description Return the ASCII code of the character in Sstr at ipos (defaults to zero which means the first character), or zero if ipos is out of range. strchar runs at init time only. Syntax ichr strchar Sstr[, ipos] Examples Here is an example of the strchar opcode. It uses the file strchar.csd. Example of the strchar opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also strchark Credits Author: Istvan Varga 2006 manual_src~dfsg/opcodes/agogobel.xml0000644000000000000000000000104312262561504016700 0ustar rootroot agogobel agogobel agogobel Deprecated. Description New in version 3.47 Deprecated as of version 3.52. Use the gogobel opcode instead. manual_src~dfsg/opcodes/samphold.xml0000644000000000000000000000724012262561504016735 0ustar rootroot samphold Signal Modifiers:Sample Level Operators samphold samphold Performs a sample-and-hold operation on its input. Description Performs a sample-and-hold operation on its input. Syntax ares samphold asig, agate [, ival] [, ivstor] kres samphold ksig, kgate [, ival] [, ivstor] Initialization ival, ivstor (optional) -- controls initial disposition of internal save space. If ivstor is zero the internal hold value is set to ival ; else it retains its previous value. Defaults are 0,0 (i.e. init to zero) Performance kgate, xgate -- controls whether to hold the signal. samphold performs a sample-and-hold operation on its input according to the value of gate. If gate != 0, the input samples are passed to the output; if gate = 0, the last output value is repeated. The controlling gate can be a constant, a control signal, or an audio signal. Examples asrc buzz 10000, 440, 20, 1 ; band-limited pulse train adif diff asrc ; emphasize the highs anew balance adif, asrc ; but retain the power agate reson asrc, 0, 440 ; use a lowpass of the original asamp samphold anew, agate ; to gate the new audiosig aout tone asamp, 100 ; smooth out the rough edges Here is another example of the samphold opcode. It uses the file samphold.csd. Example of the samphold opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also diff, downsamp, integ, interp, upsamp manual_src~dfsg/opcodes/vlinseg.xml0000644000000000000000000000532012262561504016572 0ustar rootroot vlinseg Vectorial:Envelopes vlinseg vlinseg Vectorial envelope generator Description Generate linear vectorial segments Syntax vlinseg ifnout, ielements, ifn1, idur1, ifn2 [, idur2, ifn3 [...]] Initialization ifnout - number of table hosting output vectorial signal ifn1 - starting vector ifn2,ifn3,etc. - vector after idurx seconds idur1 - duration in seconds of first segment. idur2, idur3, etc. - duration in seconds of subsequent segments. ielements - number of elements of vectors. Performance These opcodes are similar to linseg and expseg, but operate with vectorial signals instead of with scalar signals. Output is a vectorial control signal hosted by ifnout (that must be previously allocated), while each break-point of the envelope is actually a vector of values. All break-points must contain the same number of elements (ielements). All these operators are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2, etc. Example Here is an example of the vlinseg opcode. It uses the files vlinseg.csd. Example of the vlinseg opcode. Credits Written by Gabriel Maldonado. Example by Andres Cabrera. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/JackoInfo.xml0000644000000000000000000000546212262561504016775 0ustar rootroot JackoInfo Jacko Opcodes JackoInfo JackoInfo Prints information about the Jack system. Description Prints the Jack daemon and client names, the sampling rate and frames per period, and all active Jack port names, types, states, and connections. Syntax JackoInfo Initialization May be called any number of times in the orchestra header, for example both before and after creating Jack ports in the Csound orchestra header. Examples Here is an example of the JackoInfo opcode. It uses the file JackInfo.csd. Example of the JackoInfo opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also JackoInit, JackoFreewheel, JackoAudioInConnect, JackoAudioOutConnect, JackoMidiInConnect, JackoMidiOutConnect, JackoOn, JackoAudioIn, JackoAudioOut, JackoMidiOut, JackoNoteOut, JackoTransport. Credits By: &namemichael; 2010 manual_src~dfsg/opcodes/expsegba.xml0000644000000000000000000000604012262561504016721 0ustar rootroot expsegba Signal Generators:Linear and Exponential Generators expsegba expsegba An exponential segment generator operating at a-rate with absolute times. Description An exponential segment generator operating at a-rate. This unit is almost identical to expsegb, but more precise when defining segments with very short durations (i.e., in a percussive attack phase) at audio rate. Syntax ares expsegba ia, itim1, ib [, itim2] [, ic] [...] Initialization ia -- starting value. Zero is illegal. ib, ic, etc. -- value after itim1 seconds, etc. must be non-zero and must agree in sign with ia. itim1 -- time in seconds at end of first segment. itim2, itim3, etc. -- time in seconds at the end of subsequent segments. Performance This unit generate audio signals whose values can pass through two or more specified points. The final tim value may or may not equal the instrument's performance time. A shorter performance will truncate the specified pattern, while a longer one will cause the last defined segment to continue on in the same direction. Examples Here is an example of the expsegba opcode. It uses the file expsegba.csd. Example of the expsegba opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also expsegb, expsegr Credits Author: &namejohn; June 2011 New in Csound 5.14 manual_src~dfsg/opcodes/flsettextcolor.xml0000644000000000000000000000615412262561504020212 0ustar rootroot FLsetTextColor FLTK:Appearance FLsetTextColor FLsetTextColor Sets the color of the text label of a FLTK widget. Description FLsetTextColor sets the color of the text label of the target widget. Syntax FLsetTextColor ired, iblue, igreen, ihandle Initialization ired -- The red color of the target widget. The range for each RGB component is 0-255 igreen -- The green color of the target widget. The range for each RGB component is 0-255 iblue -- The blue color of the target widget. The range for each RGB component is 0-255 ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/dcblock2.xml0000644000000000000000000000535712262561504016620 0ustar rootroot dcblock2 Signal Modifiers:Specialized Filters dcblock2 dcblock2 A DC blocking filter. Description Implements a DC blocking filter with improved DC attenuation. Syntax ares dcblock2 ain [, iorder] [, iskip] Initialization iorder -- filter order, minimum 4th order, defaults to 128. iskip -- set to 1 to skip initialization (defaults to 0). Performance ares -- filered audio signal ain -- input audio signal Using a value for iorder less that ksmps will not reduce DC offset efficiently. Examples The result can be viewed in a graphical audio editor program like audacity. audacity is available for Linux, Windows, and the MacOS and may be downloaded from http://audacity.sourceforge.net. Here is an example of the dcblock2 opcode. It uses the file dcblock2.csd, and beats.wav. Example of the dcblock2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also dcblock Credits By: &namevictor; New in Csound version 5.09 manual_src~dfsg/opcodes/syncloop.xml0000644000000000000000000001021212262561504016765 0ustar rootroot syncloop Signal Generators:Granular Synthesis syncloop syncloop Synchronous granular synthesis. Description syncloop is a variation on syncgrain, which implements synchronous granular synthesis. syncloop adds loop start and end points and an optional start position. Loop start and end control grain start positions, so the actual grains can go beyond the loop points (if the loop points are not at the extremes of the table), enabling seamless crossfading. For more information on the granular synthesis process, check the syncgrain manual page. Syntax asig syncloop kamp, kfreq, kpitch, kgrsize, kprate, klstart, \ klend, ifun1, ifun2, iolaps[,istart, iskip] Initialization ifun1 -- source signal function table. Deferred-allocation tables (see GEN01) are accepted, but the opcode expects a mono source. ifun2 -- grain envelope function table. iolaps -- maximum number of overlaps, max(kfreq)*max(kgrsize). Estimating a large value should not affect performance, but execeeding this value will probably have disastrous consequences. istart -- starting point of synthesis in secs (defaults to 0). iskip -- if 1, the opcode initialisation is skipped, for tied notes, performance continues from the position in the loop where the previous note stopped. The default, 0, does not skip initialisation Performance kamp -- amplitude scaling kfreq -- frequency of grain generation, or density, in grains/sec. kpitch -- grain pitch scaling (1=normal pitch, < 1 lower, > 1 higher; negative, backwards) kgrsize -- grain size in secs. kprate -- readout pointer rate, in grains. The value of 1 will advance the reading pointer 1 grain ahead in the source table. Larger values will time-compress and smaller values will time-expand the source signal. Negative values will cause the pointer to run backwards and zero will freeze it. klstart -- loop start in secs. klend -- loop end in secs. Examples Here is an example of the syncloop opcode. It uses the file syncloop.csd. Example of the syncloop opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; January 2005 New plugin in version 5 January 2005. manual_src~dfsg/opcodes/igauss.xml0000644000000000000000000000076012262561504016421 0ustar rootroot igauss igauss igauss Deprecated. Description Deprecated as of version 3.49. Use the gauss opcode instead. manual_src~dfsg/opcodes/fmvoice.xml0000644000000000000000000000504012262561504016552 0ustar rootroot fmvoice Signal Generators:FM Synthesis fmvoice fmvoice FM Singing Voice Synthesis Description FM Singing Voice Synthesis Syntax ares fmvoice kamp, kfreq, kvowel, ktilt, kvibamt, kvibrate[, ifn1, \ ifn2, ifn3, ifn4, ivibfn] Initialization ifn1, ifn2, ifn3,ifn3, ivibfn -- Tables, usually of sinewaves. The last is for vibrato Performance kamp -- Amplitude of note. kfreq -- Frequency of note played. kvowel -- the vowel being sung, in the range 0-64 ktilt -- the spectral tilt of the sound in the range 0 to 99 kvibamt -- Depth of vibrato kvibrate -- Rate of vibrato Examples Here is an example of the fmvoice opcode. It uses the file fmvoice.csd. Example of the fmvoice opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; (after &nameperry;) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47 manual_src~dfsg/opcodes/dbamp.xml0000644000000000000000000000357412262561504016217 0ustar rootroot dbamp Mathematical Operations:Amplitude Functions dbamp dbamp Returns the decibel equivalent of the raw amplitude x. Description Returns the decibel equivalent of the raw amplitude x. Syntax dbamp(x) (init-rate or control-rate args only) Examples Here is an example of the dbamp opcode. It uses the file dbamp.csd. Example of the dbamp opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like this: instr 1: idb = 40.000 instr 1: idb = 60 instr 1: idb = 80 instr 1: idb = 86.021 See Also ampdb, ampdbfs, dbfsamp manual_src~dfsg/opcodes/opa.xml0000644000000000000000000000367312262561504015713 0ustar rootroot a Signal Modifiers:Sample Level Operators a a Converts a k-rate parameter to an a-rate value with interpolation. Description Converts a k-rate parameter to an a-rate value with interpolation. Syntax a(x) (control-rate args only) where the argument within the parentheses may be an expression. Value converters perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. Examples Here is an example of the a opcode. It uses the file opa.csd. Example of the a opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also i, k Credits Author: &namegabriel; New in version 4.21 manual_src~dfsg/opcodes/vsubv_i.xml0000644000000000000000000001034512262561504016603 0ustar rootroot vsubv_i Vectorial:Vectorial operations vsubv_i vsubv_i Performs subtraction between two vectorial control signals at init time. Description Performs subtraction between two vectorial control signals at init time. Syntax vsubv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Initialization ifn1 - number of the table hosting the first vector to be processed ifn2 - number of the table hosting the second vector to be processed ielements - number of elements of the two vectors idstoffset - index offset for the destination (ifn1) table (Default=0) isrcoffset - index offset for the source (ifn2) table (Default=0) Performance vsubv_i subtracts two vectorial control signals, that is, each element of ifn2 is subrtacted from the corresponding element of ifn1. Each vectorial signal is hosted by a table (ifn1 and ifn2). The number of elements contained in both vectors must be the same. The result is a new vectorial control signal that overrides old values of ifn1. If you want to keep the old ifn1 vector, use vcopy_i opcode to copy it in another table. You can use idstoffset and isrcoffset to specify vectors in any location of the tables. Negative values for idstoffset and isrcoffset are acceptable. If idstoffset is negative, the out of range section of the vector will be discarded. If isrcoffset is negative, the out of range elements will be assumed to be 0 (i.e. the destination elements will not be changed). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 0 (i.e. the destination vector will not be changed for these elements). Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at init time. There's an k-rate version of this opcode called vsubv. All these operators (vaddv_i,vsubv_i,vmultv_i,vdivv_i,vpowv_i,vexpv_i, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/nlfilt.xml0000644000000000000000000000744712262561504016427 0ustar rootroot nlfilt Signal Modifiers:Specialized Filters nlfilt nlfilt A filter with a non-linear effect. Description Implements the filter: Y{n} =a Y{n-1} + b Y{n-2} + d Y^2{n-L} + X{n} - C described in Dobson and Fitch (ICMC'96) Syntax ares nlfilt ain, ka, kb, kd, kC, kL Performance Non-linear effect. The range of parameters are: a = b = 0 d = 0.8, 0.9, 0.7 C = 0.4, 0.5, 0.6 L = 20 This affects the lower register most but there are audible effects over the whole range. We suggest that it may be useful for coloring drums, and for adding arbitrary highlights to notes. Low Pass with non-linear. The range of parameters are: a = 0.4 b = 0.2 d = 0.7 C = 0.11 L = 20, ... 200 There are instability problems with this variant but the effect is more pronounced of the lower register, but is otherwise much like the pure comb. Short values of L can add attack to a sound. High Pass with non-linear. The range of parameters are: a = 0.35 b = -0.3 d = 0.95 C = 0,2, ... 0.4 L = 200 High Pass with non-linear. The range of parameters are: a = 0.7 b = -0.2, ... 0.5 d = 0.9 C = 0.12, ... 0.24 L = 500, 10 The high pass version is less likely to oscillate. It adds scintillation to medium-high registers. With a large delay L it is a little like a reverberation, while with small values there appear to be formant-like regions. There are arbitrary color changes and resonances as the pitch changes. Works well with individual notes. Warning The "useful" ranges of parameters are not yet mapped. Examples Here is an example of the nlfilt opcode. It uses the file nlfilt.csd. Example of the nlfilt opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK 1997 New in version 3.44 manual_src~dfsg/opcodes/vbap4move.xml0000644000000000000000000001354712262561504017040 0ustar rootroot vbap4move Signal Modifiers:Panning and Spatialization vbap4move vbap4move Distributes an audio signal among 4 channels with moving virtual sources. Description Distributes an audio signal among 4 channels with moving virtual sources. Syntax ar1, ar2, ar3, ar4 vbap4move asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] Initialization idur -- the duration over which the movement takes place. ispread -- spreading of the virtual source (range 0 - 100). If value is zero, conventional amplitude panning is used. When ispread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. ifldnum -- number of fields (absolute value must be 2 or larger). If ifldnum is positive, the virtual source movement is a polyline specified by given directions. Each transition is performed in an equal time interval. If ifldnum is negative, specified angular velocities are applied to the virtual source during specified relative time intervals (see below). ifld1, ifld2, ... -- azimuth angles or angular velocities, and relative durations of movement phases (see below). Performance asig -- audio signal to be panned vbap4move allows the use of moving virtual sources. If ifldnum is positive, the fields represent directions of virtual sources and equal times, iazi1, [iele1,] iazi2, [iele2,], etc. The position of the virtual source is interpolated between directions starting from the first direction and ending at the last. Each interval is interpolated in time that is fraction total_time / number_of_intervals of the duration of the sound event. If ifldnum is negative, the fields represent angular velocities and equal times. The first field is, however, the starting direction, iazi1, [iele1,] iazi_vel1, [iele_vel1,] iazi_vel2, [iele_vel2,] .... Each velocity is applied to the note that is fraction total_time / number_of_velocities of the duration of the sound event. If the elevation of the virtual source becomes greater than 90 degrees or less than 0 degrees, the polarity of angular velocity is changed. Thus the elevational angular velocity produces a virtual source that moves up and down between 0 and 90 degrees. Please note that all vbap panning opcodes require the vbap system to be initialized using vbaplsinit. Examples See the entry for vbap8move for an example of usage of the vbapXmove opcodes. Reference Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456. Examples Here is an example of the vbap4move opcode. It uses the file vbap4move.csd. Example of the vbap4move opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also vbap16, vbap16move, vbap4, vbap8, vbap8move, vbaplsinit, vbapz, vbapzmove Credits Author: Ville Pulkki Sibelius Academy Computer Music Studio Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Helsinki, Finland May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/flooper.xml0000644000000000000000000000640512262561504016576 0ustar rootroot flooper Signal Generators:Sample Playback flooper flooper Function-table-based crossfading looper. Description This opcode reads audio from a function table and plays it back in a loop with user-defined start time, duration and crossfade time. It also allows the pitch of the loop to be controlled, including reversed playback. It accepts non-power-of-two tables, such as deferred-allocation GEN01 tables. Syntax asig flooper kamp, kpitch, istart, idur, ifad, ifn Initialization istart -- loop start pos in seconds idur -- loop duration in seconds ifad -- crossfade duration in seconds ifn -- function table number, generally created using GEN01 Performance asig -- output sig kamp -- amplitude control kpitch -- pitch control (transposition ratio); negative values play the loop back in reverse Examples aout flooper 16000, 1, 1, 4, 0.05, 1 ; loop starts at 1 sec, for 4 secs, 0.05 crossfade out aout The example above shows the basic operation of flooper. Pitch can be controlled at the k-rate, as well as amplitude. The example assumes table 1 to contain at least 5.05 seconds of audio (4 secs loop duration, starting 1 sec into the table, using 0.05 secs after the loop end for the crossfade). Here is another example of the flooper opcode. It uses the file flooper.csd and fox.wav. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namevictor; April 2005 New plugin in version 5 April 2005. manual_src~dfsg/opcodes/vcopy.xml0000644000000000000000000001274512262561504016274 0ustar rootroot vcopy Vectorial:Vectorial operations vcopy vcopy Copies between two vectorial control signals Description Copies between two vectorial control signals Syntax vcopy ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [, kverbose] Initialization ifn1 - number of the table where the vectorial signal will be copied (destination) ifn2 - number of the table hosting the vectorial signal to be copied (source) Performance kelements - number of elements of the vector kdstoffset - index offset for the destination (ifn1) table (Default=0) ksrcoffset - index offset for the source (ifn2) table (Default=0) kverbose - Selects whether or not warnings are printed (Default=0) vcopy copies kelements elements from ifn2 (starting from position ksrcoffset) to ifn1 (starting from position kdstoffset). Useful to keep old vector values, by storing them in another table. Negative values for kdstoffset and ksrcoffset are acceptable. If kdstoffset is negative, the out of range section of the vector will be discarded. If kdstoffset is negative, the out of range elements will be assumed to be 1 (i.e. the destination elements will not be changed). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 1 (i.e. the destination vector will not be changed for these elements). If the optional kverbose argument is different to 0, the opcode will print warning messages every k-pass if table lengths are exceeded. Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at k-rate (this means that every k-pass the vectors are copied). There's an i-rate version of this opcode called vcopy_i. Please note that the elements argument has changed in version 5.03 from i-rate to k-rate. This will change the opcode's behavior in the unusual cases where the i-rate variable ielements is changed inside the instrument, for example in: instr 1 ielements = 10 vadd 1, 1, ielements ielements = 20 vadd 2, 1, ielements turnoff endin All these operators (vaddv, vsubv, vmultv, vdivv, vpowv, vexp, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Note: bmscan not yet available on Canonical Csound Examples Here is an example of the vcopy opcode. It uses the file vcopy.csd. Example of the vcopy opcode. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/schedule.xml0000644000000000000000000001155612262561504016727 0ustar rootroot schedule Instrument Control:Invocation schedule schedule Adds a new score event. Description Adds a new score event. Syntax schedule insnum, iwhen, idur [, ip4] [, ip5] [...] schedule "insname", iwhen, idur [, ip4] [, ip5] [...] Initialization insnum -- instrument number. Equivalent to p1 in a score i statement. insnum must be a number greater than the number of the calling instrument. insname -- A string (in double-quotes) representing a named instrument. iwhen -- start time of the new event. Equivalent to p2 in a score i statement. iwhen must be nonnegative. If iwhen is zero, insum must be greater than or equal to the p1 of the current instrument. idur -- duration of event. Equivalent to p3 in a score i statement. ip4, ip5, ... -- Equivalent to p4, p5, etc., in a score i statement. Performance schedule adds a new score event. The arguments, including options, are the same as in a score. The iwhen time (p2) is measured from the time of this event. If the duration is zero or negative the new event is of MIDI type, and inherits the release sub-event from the scheduling instruction. Note that the schedule opcode can't accept string p-fields. If you need to pass strings when instantiating an instrument, use the scoreline or scoreline_i opcode. Examples Here is an example of the schedule opcode. It uses the file schedule.csd. Example of the schedule opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also schedwhen See also event, event_i, schedwhen, schedkwhen, schedkwhennamed, scoreline, scoreline_i More information on this opcode: http://www.csounds.com/journal/issue15/phrase_loops.html , written by Jim Aikin Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK November 1998 Example written by &namekevin;. New in Csound version 3.491 Based on work by &namegabriel; Thanks goes to David Gladstein, for clarifying the iwhen parameter. manual_src~dfsg/opcodes/fltabsend.xml0000644000000000000000000000362712262561504017075 0ustar rootroot FLtabsEnd FLTK:Containers FLtabsEnd FLtabsEnd Marks the end of a tabbed FLTK interface. Description Marks the end of a tabbed FLTK interface. Syntax FLtabsEnd Performance Containers are useful to format the graphic appearance of the widgets. The most important container is FLpanel, that actually creates a window. It can be filled with other containers and/or valuators or other kinds of widgets. There are no k-rate arguments in containers. See Also FLgroup, FLgroupEnd, FLpack, FLpackEnd, FLpanel, FLpanelEnd, FLscroll, FLscrollEnd, FLtabs Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/fold.xml0000644000000000000000000000457312262561504016060 0ustar rootroot fold Signal Modifiers:Sample Level Operators fold fold Adds artificial foldover to an audio signal. Description Adds artificial foldover to an audio signal. Syntax ares fold asig, kincr Performance asig -- input signal kincr -- amount of foldover expressed in multiple of sampling rate. Must be >= 1 fold is an opcode which creates artificial foldover. For example, when kincr is equal to 1 with sr=44100, no foldover is added. When kincr is set to 2, the foldover is equivalent to a downsampling to 22050, when it is set to 4, to 11025 etc. Fractional values of kincr are possible, allowing a continuous variation of foldover amount. This can be used for a wide range of special effects. Examples Here is an example of the fold opcode. It uses the file fold.csd. Example of the fold opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &namegabriel; Italy 1999 New in Csound version 3.56 manual_src~dfsg/opcodes/clfilt.xml0000644000000000000000000000741112262561504016403 0ustar rootroot clfilt Signal Modifiers:Standard Filters clfilt clfilt Implements low-pass and high-pass filters of different styles. Description Implements the classical standard analog filter types: low-pass and high-pass. They are implemented with the four classical kinds of filters: Butterworth, Chebyshev Type I, Chebyshev Type II, and Elliptical. The number of poles may be any even number from 2 to 80. Syntax ares clfilt asig, kfreq, itype, inpol [, ikind] [, ipbr] [, isba] [, iskip] Initialization itype -- 0 for low-pass, 1 for high-pass. inpol -- The number of poles in the filter. It must be an even number from 2 to 80. ikind (optional) -- 0 for Butterworth, 1 for Chebyshev Type I, 2 for Chebyshev Type II, 3 for Elliptical. Defaults to 0 (Butterworth) ipbr (optional) -- The pass-band ripple in dB. Must be greater than 0. It is ignored by Butterworth and Chebyshev Type II. The default is 1 dB. isba (optional) -- The stop-band attenuation in dB. Must be less than 0. It is ignored by Butterworth and Chebyshev Type I. The default is -60 dB. iskip (optional) -- 0 initializes all filter internal states to 0. 1 skips initialization. The default is 0. Performance asig -- The input audio signal. kfreq -- The corner frequency for low-pass or high-pass. Examples Here is an example of the clfilt opcode as a low-pass filter. It uses the file clfilt_lowpass.csd. Example of the clfilt opcode as a low-pass filter. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is an example of the clfilt opcode as a high-pass filter. It uses the file clfilt_highpass.csd. Example of the clfilt opcode as a high-pass filter. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: Erik Spjut New in version 4.20 manual_src~dfsg/opcodes/duserrnd.xml0000644000000000000000000001122512262561504016752 0ustar rootroot duserrnd Signal Generators:Random (Noise) Generators duserrnd duserrnd Discrete USER-defined-distribution RaNDom generator. Description Discrete USER-defined-distribution RaNDom generator. Syntax aout duserrnd ktableNum iout duserrnd itableNum kout duserrnd ktableNum Initialization itableNum -- number of table containing the random-distribution function. Such table is generated by the user. See GEN40, GEN41, and GEN42. The table length does not need to be a power of 2 Performance ktableNum -- number of table containing the random-distribution function. Such table is generated by the user. See GEN40, GEN41, and GEN42. The table length does not need to be a power of 2 duserrnd (discrete user-defined-distribution random generator) generates random values according to a discrete random distribution created by the user. The user can create the discrete distribution histogram by using GEN41. In order to create that table, the user has to define an arbitrary amount of number pairs, the first number of each pair representing a value and the second representing its probability (see GEN41 for more details). When used as a function, the rate of generation depends by the rate type of input variable XtableNum. In this case it can be embedded into any formula. Table number can be varied at k-rate, allowing to change the distribution histogram during the performance of a single note. duserrnd is designed be used in algorithmic music generation. duserrnd can also be used to generate values following a set of ranges of probabilities by using distribution functions generated by GEN42 (See GEN42 for more details). In this case, in order to simulate continuous ranges, the length of table XtableNum should be reasonably big, as duserrnd does not interpolate between table elements. For a tutorial about random distribution histograms and functions see: D. Lorrain. "A panoply of stochastic cannons". In C. Roads, ed. 1989. Music machine. Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the duserrnd opcode. It uses the file duserrnd.csd. Example of the duserrnd opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like these: i 1 time 0.00067: 8.00000 i 1 time 0.00133: 8.00000 i 1 time 0.00200: 8.00000 i 1 time 0.00267: 8.00000 i 1 time 0.00333: 2.00000 i 1 time 0.00400: 8.00000 i 1 time 0.00533: 8.00000 i 1 time 0.00600: 8.00000 .......... See Also cuserrnd, urd Credits Author: &namegabriel; New in Version 4.16 manual_src~dfsg/opcodes/cauchy.xml0000644000000000000000000001054212262561504016401 0ustar rootroot x-class noise generators cauchy Signal Generators:Random (Noise) Generators cauchy cauchy Cauchy distribution random number generator. Description Cauchy distribution random number generator. This is an x-class noise generator. Syntax ares cauchy kalpha ires cauchy kalpha kres cauchy kalpha Performance kalpha -- controls the spread from zero (big kalpha = big spread). Outputs both positive and negative numbers. For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the cauchy opcode. It uses the file cauchy.csd. Example of the cauchy opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include lines like: i 1 time 0.00267: -0.20676 i 1 time 0.20267: -0.28814 i 1 time 0.40267: 0.61651 i 1 time 0.60267: -18.18552 i 1 time 0.80267: 1.18140 i 1 time 1.00000: -0.75432 i 1 time 1.20267: -0.02002 i 1 time 1.40267: 0.01785 i 1 time 1.60267: -0.48834 i 1 time 1.80000: 9.69401 i 1 time 2.00000: -0.41257 Seeding from current time 3112109827 i 2 time 3.00267: -0.46887 i 2 time 3.20267: 0.06189 i 2 time 3.40267: -0.40303 i 2 time 3.60000: 0.89312 i 2 time 3.80267: -0.40374 i 2 time 4.00000: 0.86557 i 2 time 4.20000: 0.09192 i 2 time 4.40267: -0.16748 i 2 time 4.60267: 0.30133 i 2 time 4.80267: 0.31657 i 2 time 5.00000: 0.44681 See Also seed, betarand, bexprnd, exprand, gauss, linrand, pcauchy, poisson, trirand, unirand, weibull Credits Author: &nameparis; MIT, Cambridge 1995 Existed in 3.30 manual_src~dfsg/opcodes/fluidControl.xml0000644000000000000000000001006212262561504017566 0ustar rootroot fluidControl Signal Generators:Sample Playback fluidControl fluidControl Sends MIDI note on, note off, and other messages to a SoundFont preset. Syntax fluidControl ienginenum, kstatus, kchannel, kdata1, kdata2 Description The fluid opcodes provide a simple Csound opcode wrapper around Peter Hanappe's Fluidsynth SoundFont2 synthesizer. This implementation accepts any MIDI note on, note off, controller, pitch bend, or program change message at k-rate. Maximum polyphony is 4096 simultaneously sounding voices. Any number of SoundFonts may be loaded and played simultaneously. Initialization ienginenum -- engine number assigned from fluidEngine Performance kstatus -- MIDI channel message status byte: 128 for note off, 144 for note on, 176 for control change, 192 for program change, or 224 for pitch bend. kchannel -- MIDI channel number to which the Fluidsynth program is assigned: from 0 to 255. MIDI channels numbered 16 or higher are virtual channels. kdata1 -- For note on, MIDI key number: from 0 (lowest) to 127 (highest), where 60 is middle C. For continuous controller messages, controller number. kdata2 -- For note on, MIDI key velocity: from 0 (no sound) to 127 (loudest). For continous controller messages, controller value. Invoke fluidControl in instrument definitions that actually play notes and send control messages. Each instrument definition must consistently use one MIDI channel that was assigned to a Fluidsynth program using fluidLoad. In this implementation, SoundFont effects such as chorus or reverb are used if and only if they are defaults for the preset. There is no means of turning such effects on or off, or of changing their parameters, from Csound. Examples Here is a more complex example of the fluidsynth opcodes written by Istvan Varga. It uses the file fluidcomplex.csd. See Also fluidEngine, fluidNote, fluidLoad More information on soundfonts in the Floss Manuals: http://en.flossmanuals.net/csound/ch048_d-reading-midi-files Other information on soundfonts on Wikipedia: http://en.wikipedia.org/wiki/Soundfont Credits Opcodes by Michael Gogins (gogins at pipeline dot com). Thanks to Peter Hanappe for Fluidsynth, and to Steven Yi for seeing that it is necessary to break up the Fluidsynth into several different Csound opcodes. New in Csound5.00 manual_src~dfsg/opcodes/serialFlush.xml0000644000000000000000000000354312262561504017411 0ustar rootroot serialFlush Serial I/O serialFlush serialFlush Flush data from a serial port. Description Flush to the screen any bytes (up to 32k) in the input buffer. Note that these bytes will be cleared from the buffer. use this opcode mainly for debugging messages. If you want to mix debugging and other communication messages over the same port, you will need to manually parse the data with the serialRead opcode. Syntax serialFlush iPort Performance iPort -- port number optained from a serialBegin opcode. See Also serialBegin, serialEnd, serialWrite_i, serialWrite, serialRead, serialPrint. Credits Author: &namematt; 2011 New in version 5.14 manual_src~dfsg/opcodes/sfload.xml0000644000000000000000000001053312262561504016375 0ustar rootroot sfload Signal Generators:Sample Playback sfload sfload Loads an entire SoundFont2 (SF2) sample file into memory. Description Loads an entire SoundFont2 (SF2) sample file into memory. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the SoundFont2 File Format Appendix. sfload should be placed in the header section of a Csound orchestra. Syntax ir sfload "filename" Initialization ir -- output to be used by other SF2 opcodes. For sfload, ir is ifilhandle. filename -- name of the SF2 file, with its complete path. It must be a string typed within double-quotes with / to separate directories (this applies to DOS and Windows as well, where using a backslash will generate an error), or an integer that has been the subject of a strset operation Performance sfload loads an entire SF2 file into memory. It returns a file handle to be used by other opcodes. Several instances of sfload can placed in the header section of an orchestra, allowing use of more than one SF2 file in a single orchestra. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard. It should be noted that before version 5.12 a maximum of 10 sound fonts could be loaded, a restriction since relaxed. Examples Here is an example of the sfload opcode. It uses the file sfload.csd. Example of the sfload opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also sfilist, sfinstr, sfinstrm, sfpassign, sfplay, sfplaym, sfplist, sfpreset Credits Author: &namegabriel; Italy May 2000 New in Csound Version 4.07 manual_src~dfsg/opcodes/rms.xml0000644000000000000000000000643512262561504015734 0ustar rootroot rms Instrument Control:Sensing and Control rms rms Determines the root-mean-square amplitude of an audio signal. Description Determines the root-mean-square amplitude of an audio signal. It low-pass filters the actual value, to average in the manner of a VU meter. Syntax kres rms asig [, ihp] [, iskip] Initialization ihp (optional, default=10) -- half-power point (in Hz) of a special internal low-pass filter. The default value is 10. iskip (optional, default=0) -- initial disposition of internal data space (see reson). The default value is 0. Performance asig -- input audio signal kres -- low-pass filtered rms value of the input signal rms output values kres will trace the root-mean-square value of the audio input asig. This unit is not a signal modifier, but functions rather as a signal power-gauge. It uses an internal low-pass filter to make the response smoother. ihp can be used to control this smoothing. The higher the value, the "snappier" the measurement. This opcode can also be used as an evelope follower. The kres output from this opcode is given in amplitude and depends on 0dbfs. If you want the output in decibels, you can use dbamp Examples arms rms asig ; get rms value of signal asig Here is an example of the rms opcode. It uses the file rms.csd. Example of the rms opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also balance, gain manual_src~dfsg/opcodes/flsettext.xml0000644000000000000000000000712612262561504017153 0ustar rootroot FLsetText FLTK:Appearance FLsetText FLsetText Sets the label of a FLTK widget. Description FLsetText sets the label of the target widget to the double-quoted text string provided with the itext argument. Syntax FLsetText "itext", ihandle Initialization itext -- a double-quoted string denoting the text of the label of the widget. ihandle -- an integer number (used as unique identifier) taken from the output of a previously located widget opcode (which corresponds to the target widget). It is used to unequivocally identify the widget when modifying its appearance with this class of opcodes. The user must not set the ihandle value directly, otherwise a Csound crash will occur. Examples Here is an example of the FLsetText opcode. It uses the file FLsetText.csd. Example of the FLsetText opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/cross2.xml0000644000000000000000000000554212262561504016344 0ustar rootroot cross2 Signal Modifiers:Convolution and Morphing cross2 cross2 Cross synthesis using FFT's. Description This is an implementation of cross synthesis using FFT's. Syntax ares cross2 ain1, ain2, isize, ioverlap, iwin, kbias Initialization isize -- This is the size of the FFT to be performed. The larger the size the better the frequency response but a sloppy time response. ioverlap -- This is the overlap factor of the FFT's, must be a power of two. The best settings are 2 and 4. A big overlap takes a long time to compile. iwin -- This is the function table that contains the window to be used in the analysis. One can use the GEN20 routine to create this window. Performance ain1 -- The stimulus sound. Must have high frequencies for best results. ain2 -- The modulating sound. Must have a moving frequency response (like speech) for best results. kbias -- The amount of cross synthesis. 1 is the normal, 0 is no cross synthesis. Examples Here is an example of the cross2 opcode. It uses the file cross2.csd and fox.wav. Example of the cross2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Author: &nameparis; MIT, Cambridge 1997 manual_src~dfsg/opcodes/STKFMVoices.xml0000644000000000000000000000731112262561504017162 0ustar rootroot STKFMVoices Signal Generators:STK Opcodes STKFMVoices STKFMVoices STKFMVoices is a singing FM synthesis instrument. Description STKFMVoices is a singing FM synthesis instrument. It has 3 carriers and a common modulator, also referred to as algorithm 6 of the TX81Z. Syntax asignal STKFMVoices ifrequency, iamplitude, [kvowel, kv1[, kspec, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] Initialization ifrequency -- Frequency of note played, in Hertz. iamplitude -- Amplitude of note played (range 0-1). Performance kvowel -- controller 2, vowel. Value range of kv1 is 0-127. kspec -- controller 4, spectral tilt. Value range of kv2 is 0-127. klfo -- controller 11, speed of low-frequency oscillator. Value range of kv3 is 0-127. klfodepth -- controller 1, depth of low-frequency oscillator. Value range of kv4 is 0-127. kadsr -- controller 128, ADSR 2 and 4 Target. Value range of kv5 is 0-127. Note The code for this opcode is taken directly from the FMVoices class in the Synthesis Toolkit in C++ by Perry R. Cook and Gary P. Scavone. More on the STK classes can be found here: https://ccrma.stanford.edu/software/stk/classes.html kc1, kv1, kc2, kv2, kc3, kv3, kc4, kv4, kc5, kv5, kc6, kv6, kc7, kv7, kc8, kv8 -- Up to 8 optional k-rate controller pairs for the STK opcodes. Each controller pair consists of a controller number (kc) followed by a controller value (kv). Both the controller numbers and the controller values are krate variables. However, during a performance, normally the controller numbers stay fixed while the corresponding controller values may change at any time. The order of the controller pair is arbitrary, as long as they are after iamplitude. Also, it is not needed that all controller pairs are used. Examples Here is an example of the STKFMVoices opcode. It uses the file STKFMVoices.csd. Example of the STKFMVoices opcode. See Also STKBeeThree. Credits Author: &namemichael; (after &nameperry;) Irreducible Productions New York, NY New in Csound version 5.11 manual_src~dfsg/opcodes/flsetsnapgroup.xml0000644000000000000000000000516412262561504020205 0ustar rootroot FLsetSnapGroup FLTK:Other FLsetSnapGroup FLsetSnapGroup Determines the snapshot group for FL valuators. Description FLsetSnapGroup determines the snapshot group of valuators declared after it. Syntax FLsetSnapGroup igroup Initialization igroup -- (optional) an integer number referring to a snapshot-related group of widget. It allows to get/set, or to load/save the state of a subset of valuators. The igroup parameter has not been yet fully implemented in the current version of csound. Please do not rely on it yet. Performance For purposes of snapshot saving, widgets can be grouped, so that snapshots affect only a defined group of widgets. The opcode FLsetSnapGroup is used to specify the group for all widgets declared after it, until the next FLsetSnapGroup statement. FLsetSnapGroup determines the snapshot group of a declared valuator. To make a valuator belong to a stated group, you have to place FLsetSnapGroup just before the declaration of the widget itself. The group stated by FLsetSnapGroup lasts for all valuators declared after it, until a new FLsetSnapGroup statement with a different group is encountered. If no FLsetSnapGroup statement are present in an orchestra, the default group for all widgets will be group zero. See Also FLgetsnap, FLsetsnap, FLloadsnap, FLsavesnap Credits Author: &namegabriel; New in version 4.22 manual_src~dfsg/opcodes/p5gdata.xml0000644000000000000000000000667712262561504016470 0ustar rootroot p5gdata Instrument Control:Sensing and Control p5gdata p5gdata Reads data fields from an external P5 Glove. Description Reads data fields from a P5 Glove controller. Syntax kres p5gdata kcontrol Initialization This opcode must be used in conjuction with a running p5gconnect opcode. Performance kcontrol -- the code for which control to read On each access a particular data item of the P5 glove is read. The currently implemented controls are given below, together with the macro name defined in the file p5g_mac: 0 (P5G_BUTTONS): returns a bit pattern for all buttons that were pressed. 1 (P5G_BUTTON_A): returns 1 if the button has just been pressed, or 0 otherwise. 2 (P5G_BUTTON_B): as above. 4 (P5G_BUTTON_C): as above. 8 (P5G_JUSTPUSH): returns a bit pattern for all buttons that have just been pressed. 9 (P5G_JUSTPU_A): returns 1 if the A button has just been pressed. 10 (P5G_JUSTPU_B): as above. 12 (P5G_JUSTPU_C): as above. 16 (P5G_RELEASED): returns a bit pattern for all buttons that have just been released. 17 (P5G_RELSED_A): returns 1 if the A button has just been released. 18 (P5G_RELSED_B): as above. 20 (P5G_RELSED_C): as above. 32 (P5G_FINGER_INDEX): returns the clench value of the index finger. 33 (P5G_FINGER_MIDDLE): as above. 34 (P5G_FINGER_RING): as above. 35 (P5G_FINGER_PINKY): as above with little finger. 36 (P5G_FINGER_THUMB): as above. 37 (P5G_DELTA_X): The X position of the glove. 38 (P5G_DELTA_Y): The Y position of the glove. 39 (P5G_DELTA_Z): The Z position of the glove. 40 (P5G_DELTA_XR): The X axis change (angle). 41 (P5G_DELTA_YR): as above. 42 (P5G_DELTA_ZR): as above. 43 (P5G_ANGLES): The general angle Examples See the example for p5gconnect. See Also p5gconnect, Credits Author: &namejohn; Codemist Ltd 2009 New in version 5.12 manual_src~dfsg/opcodes/randi.xml0000644000000000000000000000770312262561504016227 0ustar rootroot randi Signal Generators:Random (Noise) Generators randi rand Generates a controlled random number series with interpolation between each new number. Description Generates a controlled random number series with interpolation between each new number. Syntax ares randi xamp, xcps [, iseed] [, isize] [, ioffset] kres randi kamp, kcps [, iseed] [, isize] [, ioffset] Initialization iseed (optional, default=0.5) -- seed value for the recursive pseudo-random formula. A value between 0 and +1 will produce an initial output of kamp * iseed. A negative value will cause seed re-initialization to be skipped. A value greater than 1 will seed from system time, this is the best option to generate a different random sequence for each run. isize (optional, default=0) -- if zero, a 16 bit number is generated. If non-zero, a 31-bit random number is generated. Default is 0. ioffset (optional, default=0) -- a base value added to the random result. New in Csound version 4.03. Performance kamp, xamp -- range over which random numbers are distributed. kcps, xcps -- the frequency which new random numbers are generated. The internal pseudo-random formula produces values which are uniformly distributed over the range kamp to -kamp. rand will thus generate uniform white noise with an R.M.S value of kamp / root 2. The remaining units produce band-limited noise: the kcps and xcps parameters permit the user to specify that new random numbers are to be generated at a rate less than the sampling or control frequencies. randi will produce straight-line interpolation between each new number and the next. Examples Here is an example of the randi opcode. It uses the file randi.csd. Example of the randi opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example will produce the following output: i 1 time 0.00067: 50.00000 i 1 time 0.50000: -75.81672 i 1 time 1.00000: 95.93833 WARNING: Seeding from current time 1482746120 i 2 time 2.00067: -17.94434 i 2 time 2.50000: -14.11875 i 2 time 3.00000: -72.41545 See Also rand, randh manual_src~dfsg/opcodes/maca.xml0000644000000000000000000000433312262561504016027 0ustar rootroot maca Mathematical Operations:Opcode Equivalents of Functions maca maca Multiply and accumulate a-rate signals only. Description Multiply and accumulate a-rate signals only. Syntax ares maca asig1 , asig2 [, asig3] [, asig4] [, asig5] [...] Performance asig1, asig2, ... -- a-rate input signals maca multiplies and accumulates a-rate signals only. It is equivalent to: ares = asig1*asig2 + asig3*asig4 + asig5*asig6 + ... Examples Here is an example of the maca opcode. It uses the file maca.csd. It is written for *NIX systems, and will generate errors on Windows. Example of the maca opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also mac Credits Author: &namejohn; University of Bath, Codemist, Ltd. Bath, UK May 1999 New in Csound version 3.54 manual_src~dfsg/opcodes/inch.xml0000644000000000000000000000556012262561504016052 0ustar rootroot inch Signal I/O:Signal Input inch inch Reads from numbered channels in an external audio signal or stream. Description Reads from numbered channels in an external audio signal or stream. Syntax ain1[, ...] inch kchan1[,...] Performance ain1, ... - input audio signals kchan1,... - channel numbers inch reads from numbered channels determined by the corresponding kchan into the associated ain. If the command-line -i flag is set, sound is read continuously from the audio input stream (e.g. stdin or a soundfile). inch can also be used to receive audio in realtime from the audio interface using -iadc. The highest number for kchan available for use with inch depends on nchnls_i. If kchan is greater than nchnls_i, ain will be silent. Note that inch will give a warning but not an error in this case. Examples Here is an example of the inch opcode. It uses the file inch.csd. Example of the inch opcode. See Also in32, inx, inz Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK May 2000 New in Csound Version 4.07 Multiple arguments from version 5.13 manual_src~dfsg/opcodes/trsplit.xml0000644000000000000000000000572312262561504016633 0ustar rootroot trsplit Spectral Processing:Streaming trsplit trsplit Streaming partial track frequency splitting. Description The trsplit opcode takes an input containg a TRACKS pv streaming signal (as generated, for instance by partials) and splits it into two signals according to a k-rate frequency 'split point'. The first output will contain all tracks up from 0Hz to the split frequency and the second will contain the tracks from the split frequency up to the Nyquist. It can also, optionally, scale the gain of the output signals by a k-rate amount (default 1). The result is two output signals containing only part of the original spectrum. Syntax fsiglow, fsighi trsplit fin, ksplit[, kgainlow, kgainhigh] Performance fsiglow -- output pv stream in TRACKS format containing the tracks below the split point. fsighi -- output pv stream in TRACKS format containing the tracks above and including the split point. fin -- input pv stream in TRACKS format ksplit -- frequency split point in Hz kgainlow, kgainhig -- amplitude scaling of each one of the outputs (default 1). Examples Here is an example of the trsplit opcode. It uses the file trsplit.csd. Example of the trsplit opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. The example above shows partial tracking of an ifd-analysis signal and linear additive resynthesis of the upper part of the spectrum (from 1500Hz). Credits Author: &namevictor; February 2006 New in Csound 5.01 manual_src~dfsg/opcodes/flknob.xml0000644000000000000000000001721412262561504016403 0ustar rootroot FLknob FLTK:Valuators FLknob FLknob A FLTK widget opcode that creates a knob. Description A FLTK widget opcode that creates a knob. Syntax kout, ihandle FLknob "label", imin, imax, iexp, itype, idisp, iwidth, \ ix, iy [, icursorsize] Initialization ihandle -- a handle value (an integer number) that unequivocally references a corresponding widget. This is used by other opcodes that modify a widget's properties (see Modifying FLTK Widget Appearance). It is automatically utput by FLknob and must not be set by the user label. (The user label is a double-quoted string containing some user-provided text placed near the widget.) label -- a double-quoted string containing some user-provided text, placed near the corresponding widget. imin -- minimum value of output range. imax -- maximum value of output range. iexp -- an integer number denoting the behaviour of valuator: 0 = valuator output is linear -1 = valuator output is exponential All other positive numbers for iexp indicate the number of an existing table that is used for indexing. Linear interpolation is provided in table indexing. A negative table number suppresses interpolation. IMPORTANT! Notice that the tables used by valuators must be created with the ftgen opcode and placed in the orchestra before the corresponding valuator. They can not placed in the score. In fact, tables placed in the score are created later than the initialization of the opcodes placed in the header section of the orchestra. itype -- an integer number denoting the appearance of the valuator. The itype argument can be set to the following values: 1 - a 3-D knob 2 - a pie-like knob 3 - a clock-like knob 4 - a flat knob A 3-D knob. A 3-D knob. A pie knob. A pie knob. A clock knob. A clock knob. A flat knob. A flat knob. idisp -- a handle value that was output from a previous instance of the FLvalue opcode to display the current value of the current valuator in the FLvalue widget itself. If the user doesn't want to use this feature that displays current values, it must be set to a negative number by the user. iwidth -- width of widget. iheight -- height of widget. ix -- horizontal position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). iy -- vertical position of upper left corner of the valuator, relative to the upper left corner of corresponding window (expressed in pixels). icursorsize (optional) -- If FLknob's itype is set to 1 (3D knob), this parameter controls the size of knob cursor. Performance kout -- output value FLknob puts a knob in the corresponding container. Examples Here is an example of the FLknob opcode. It uses the file FLknob.csd. Example of the FLknob opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Here is another example of the FLknob opcode, showing the different styles of knobs and the usage of FLvalue to display a knob's value. It uses the file FLknob-2.csd. More complex example of the FLknob opcode. See Also FLcount, FLjoy, FLroller, FLslider, FLtext Credits Author: &namegabriel; New in version 4.22 Example written by Iain McCurdy, edited by &namekevin;. manual_src~dfsg/opcodes/ipow.xml0000644000000000000000000000074212262561504016104 0ustar rootroot ipow ipow ipow Deprecated. Description Deprecated as of version 3.48. Use the pow opcode instead. manual_src~dfsg/opcodes/flslidbnk2.xml0000644000000000000000000002042512262561504017160 0ustar rootroot FLslidBnk2 FLTK:Other FLslidBnk2 FLslidBnk2 A FLTK widget containing a bank of horizontal sliders. Description FLslidBnk2 is a widget containing a bank of horizontal sliders. Syntax FLslidBnk2 "names", inumsliders, ioutable, iconfigtable [,iwidth, iheight, ix, iy, istart_index] Initialization names -- a double-quoted string containing the names of each slider. Each slider can have a different name. Separate each name with @ character, for example: frequency@amplitude@cutoff. It is possible to not provide any name by giving a single space . In this case, the opcode will automatically assign a progressive number as a label for each slider. inumsliders -- the number of sliders. ioutable (optional, default=0) -- number of a previously-allocated table in which to store output values of each slider. The user must be sure that table size is large enough to contain all output cells, otherwise a segfault will crash Csound. By assigning zero to this argument, the output will be directed to the zak space in the k-rate zone. In this case, the zak space must be previously allocated with the zakinit opcode and the user must be sure that the allocation size is big enough to cover all sliders. The default value is zero (i.e. store output in zak space). iconfigtable -- in the FLslidBnk2 and FLvslidBnk2 opcodes, this table replaces iminmaxtable, iexptable and istyletable, all these parameters being placed into a single table. This table has to be filled with a group of 4 parameters for each slider in this way: min1, max1, exp1, style1, min2, max2, exp2, style2, min3, max3, exp3, style3 etc. for example using GEN02 you can type: inum ftgen 1,0,256, -2, 0,1,0,1, 100, 5000, -1, 3, 50, 200, -1, 5,….. [etcetera] In this example the first slider will be affected by the [0,1,0,1] parameters (the range will be 0 to 1, it will have linear response, and its aspect will be a fill slider), the second slider will be affected by the [100,5000,-1,3] parameters (the range is 100 to 5000, the response is exponential and the aspect is a normal slider), the third slider will be affected by the [50,200,-1,5] parameters (the range is 50 to 200, the behavior exponential, and the aspect is a nice slider), and so on. iwidth (optional) -- width of the rectangular area containing all sliders of the bank, excluding text labels, that are placed to the left of that area. iheight (optional) -- height of the rectangular area containing all sliders of the bank, excluding text labels, that are placed to the left of that area. ix (optional) -- horizontal position of the upper left corner of the rectangular area containing all sliders belonging to the bank. You have to leave enough space, at the left of that rectangle, in order to make sure labels of sliders to be visible. This is because the labels themselves are external to the rectangular area. iy (optional) -- vertical position of the upper left corner of the rectangular area containing all sliders belonging to the bank. You have to leave enough space, at the left of that rectangle, in order to make sure labels of sliders to be visible. This is because the labels themselves are external to the rectangular area. istart_index (optional, default=0) -- an integer number referring to a starting offset of output cell locations. It can be positive to allow multiple banks of sliders to output in the same table or in the zak space. The default value is zero (no offset). Performance There are no k-rate arguments, even if cells of the output table (or the zak space) are updated at k-rate. FLslidBnk2 is a widget containing a bank of horizontal sliders. Any number of sliders can be placed into the bank (inumsliders argument). The output of all sliders is stored into a previously allocated table or into the zak space (ioutable argument). It is possible to determine the first location of the table (or of the zak space) in which to store the output of the first slider by means of istart_index argument. Each slider can have an individual label that is placed to the left of it. Labels are defined by the names argument. The output range of each slider can be individually set by means of the min and max values inside the iconfigtable table. The curve response of each slider can be set individually, by means of a list of identifiers placed in the iconfigtable table (exp argument). It is possible to define the aspect of each slider independently or to make all sliders have the same aspect (style argument in the iconfigtable table). The iwidth, iheight, ix, and iy arguments determine width, height, horizontal and vertical position of the rectangular area containing sliders. Notice that the label of each slider is placed to the left of them and is not included in the rectangular area containing sliders. So the user should leave enough space to the left of the bank by assigning a proper ix value in order to leave labels visible. IMPORTANT! Notice that the tables used by FLslidBnk2 must be created with the ftgen opcode and placed in the orchestra before the corresponding valuator. They can not placed in the score. This is because tables placed in the score are created later than the initialization of the opcodes placed in the header section of the orchestra. Examples Here is an example of the FLslidBnk2 opcode. It uses the file FLslidBnk2.csd. Example of the FLslidBnk2 opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See Also FLslider, FLslidBnk, FLvslidBnk, FLvslidBnk2 Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/ioutpat.xml0000644000000000000000000000077112262561504016615 0ustar rootroot ioutpat ioutpat ioutpat Deprecated. Description Deprecated as of version 3.52. Use the outipat opcode instead. manual_src~dfsg/opcodes/sockrecv.xml0000644000000000000000000000541612262561504016750 0ustar rootroot sockrecv Network sockrecv sockrecv Receives data from other processes using the low-level UDP or TCP protocols Description Receives directly using the UDP (sockrecv and sockrecvs) or TCP (strecv) protocol onto a network. The data is not subject to any encoding or special routing. The sockrecvs opcode receives a stereo signal interleaved. Syntax asig sockrecv iport, ilength asigl, asigr sockrecvs iport, ilength asig strecv Sipaddr, iport Initialization Sipaddr -- a string that is the IP address of the sender in standard 4-octet dotted form. iport -- the number of the port that is used for the communication. ilength -- the length of the individual packets in UDP transmission. This number must be sufficiently small to fit a single MTU, which is set to the save value of 1456. In UDP transmissions the sender and receiver needs agree on this value Performance asig, asigl, asigr -- audio data to be received. Example The example shows a mono signal being received on port 7777 using UDP. sr = 44100 ksmps = 100 nchnls = 1 instr 1 a1 sockrecv 7777, 200 out a1 endin Credits Author: &namejohn; 2006 manual_src~dfsg/opcodes/connect.xml0000644000000000000000000000623112262561504016556 0ustar rootroot connect Signal Flow Graph Opcodes connect connect Connects a source outlet to a sink inlet. Description The connect opcode, valid only in the orchestra header, sends the signals from the indicated outlet in all instances of the indicated source instrument to the indicated inlet in all instances of the indicated sink instrument. Each inlet instance receives the sum of the signals in all outlet instances. Thus multiple instances of an outlet may fan in to one instance of an inlet, or one instance of an outlet may fan out to multiple instances of an inlet. When Csound creates a new instance of an instrument template, new instances of its connections also are created. Syntax connect Tsource1, Soutlet1, Tsink1, Sinlet1 Initialization Tsource1 -- String name of the source instrument definition. Soutlet1 -- String name of the source outlet in the source instrument. Tsink1 -- String name of the sink instrument definition. Sinlet1 -- String name of the sink inlet in the sink instrument. Examples Here is an example of the connect opcode. It uses the file connect.csd. Example of the connect opcode. See Also outleta outletk outletf inleta inletk inletf alwayson ftgenonce More information on this opcode: http://www.csounds.com/journal/issue13/signalFlowGraphOpcodes.html , written by Michael Gogins Credits By: &namemichael; 2009 manual_src~dfsg/opcodes/evalstr.xml0000644000000000000000000000321112262561504016600 0ustar rootroot evalstr Instrument Control:Compilation evalstr evalstr Evalstrs evaluates a string containing Csound code, returning a value. Description Evalstr compiles and runs Csound code and returns a value from the global space (instr 0). This opcode can be also used to compile new instruments (as compilestr). Syntax ires evalstr Scode kres evalstr Scode, ktrig Initialization Scode -- a string to be compiled and evaluated. Performance ktrig -- triggers the compilation/evaluation if non-zero. Examples Here is an example of the evalstr opcode in conjunction with return: ival evalstr "return 2 + 2" print ival Credits Author: &namevictor;, 2013 manual_src~dfsg/opcodes/linrand.xml0000644000000000000000000001053112262561504016552 0ustar rootroot x-class noise generators linrand Signal Generators:Random (Noise) Generators linrand linrand Linear distribution random number generator (positive values only). Description Linear distribution random number generator (positive values only). This is an x-class noise generator. Syntax ares linrand krange ires linrand krange kres linrand krange Performance krange -- the range of the random numbers (0 - krange). Outputs only positive numbers. For more detailed explanation of these distributions, see: C. Dodge - T.A. Jerse 1985. Computer music. Schirmer books. pp.265 - 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 - 379. Examples Here is an example of the linrand opcode. It uses the file linrand.csd. Example of the linrand opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Its output should include a line like this: i 1 time 0.00033: 13.54770 i 1 time 0.20033: 32.38746 i 1 time 0.40033: 47.69304 i 1 time 0.60033: 19.82218 i 1 time 0.80033: 42.98293 i 1 time 1.00000: 81.13174 i 1 time 1.20033: 47.39585 i 1 time 1.40033: 12.53248 i 1 time 1.60033: 35.70722 i 1 time 1.80000: 65.25774 i 1 time 2.00000: 23.24811 Seeding from current time 392575384 i 2 time 3.00033: 23.05609 i 2 time 3.20033: 76.15114 i 2 time 3.40033: 22.78861 i 2 time 3.60000: 0.79064 i 2 time 3.80033: 43.49438 i 2 time 4.00000: 34.10963 i 2 time 4.20000: 31.88702 i 2 time 4.40033: 59.78054 i 2 time 4.60033: 4.96821 i 2 time 4.80033: 24.69674 i 2 time 5.00000: 21.88815 See Also seed, betarand, bexprnd, cauchy, exprand, gauss, pcauchy, poisson, trirand, unirand, weibull Credits Author: &nameparis; MIT, Cambridge 1995 manual_src~dfsg/opcodes/inrg.xml0000644000000000000000000000555312262561504016072 0ustar rootroot inrg Signal I/O:Signal Input inrg inrg Allow input from a range of adjacent audio channels from the audio input device Description inrg reads audio from a range of adjacent audio channels from the audio input device. Syntax inrg kstart, ain1 [,ain2, ain3, ..., ainN] Performance kstart - the number of the first channel of the input device to be accessed (channel numbers starts with 1, which is the first channel) ain1, ain2, ... ainN - the output arguments filled with the incoming audio coming from corresponding channels. inrg allows input from a range of adjacent channels from the input device. kstart indicates the first channel to be accessed (channel 1 is the first channel). The user must be sure that the number obtained by summing kstart plus the number of accessed channels -1 is <= nchnls_i. Note that this opcode is exceptional in that it produces its output on the parameters to the right. Credits Author: &namegabriel; New in version 5.06 manual_src~dfsg/opcodes/compileorc.xml0000644000000000000000000000475112262561504017266 0ustar rootroot compileorc Instrument Control:Compilation compileorc compileorc compiles a new orchestra from an ASCII file Description Compileorc will compile one or more instruments at init time, which will be added to the running engine. In case of existing instrument numbers or names, these will be replaced, but any instance still running of the old instrument definition will still perform until it terminates. Syntax ires compileorc Sfilename Initialization Sfilename -- a string containing the name of the file containing the orchestra. ires -- returns 0 if compilation was successful, or -1 if not. Examples Here is an example of the compileorc opcode. It uses the file compileorc.csd. Example of the compileorc opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. See also event, event_i, schedule, schedwhen, schedkwhen, schedkwhennamed Credits Author: &namevictor;, 2013 manual_src~dfsg/opcodes/vpowv.xml0000644000000000000000000001401512262561504016305 0ustar rootroot vpowv Vectorial:Vectorial operations vpowv vpowv Performs power-of operations between two vectorial control signals Description Performs power-of operations between two vectorial control signals Syntax vpowv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] Initialization ifn1 - number of the table hosting the first vector to be processed ifn2 - number of the table hosting the second vector to be processed Performance kelements - number of elements of the two vectors kdstoffset - index offset for the destination (ifn1) table (Default=0) ksrcoffset - index offset for the source (ifn2) table (Default=0) kverbose - Selects whether or not warnings are printed (Default=0) vpowv raises each element of ifn1 to the corresponding element of ifn2. Each vectorial signal is hosted by a table (ifn1 and ifn2). The number of elements contained in both vectors must be the same. The result is a new vectorial control signal that overrides old values of ifn1. If you want to keep the old ifn1 vector, use vcopy_i opcode to copy it in another table. You can use kdstoffset and ksrcoffset to specify vectors in any location of the tables. Negative values for kdstoffset and ksrcoffset are acceptable. If kdstoffset is negative, the out of range section of the vector will be discarded. If ksrcoffset is negative, the out of range elements will be assumed to be 1 (i.e. the destination elements will not be changed). If elements for the destination vector are beyond the size of the table (including guard point), these elements are discarded (i.e. elements do not wrap around the tables). If elements for the source vector are beyond the table length, these elements are taken as 1 (i.e. the destination vector will not be changed for these elements). If the optional kverbose argument is different to 0, the opcode will print warning messages every k-pass if table lengths are exceeded. Using the same table as source and destination table in versions earlier than 5.04, might produce unexpected behavior, so use with care. This opcode works at k-rate (this means that every k-pass the vectors are processed). There's an i-rate version of this opcode called vpowv_i. Please note that the elements argument has changed in version 5.03 from i-rate to k-rate. This will change the opcode's behavior in the unusual cases where the i-rate variable ielements is changed inside the instrument, for example in: instr 1 ielements = 10 vadd 1, 1, ielements ielements = 20 vadd 2, 1, ielements turnoff endin All these operators (vaddv, vsubv, vmultv, vdivv, vpowv, vexpv, vcopy and vmap) are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2, etc. Examples Here is an example of the vpowv opcode. It uses the file vpowv.csd. Example of the vpowv opcode. See the sections Real-time Audio and Command Line Flags for more information on using command line flags. Credits Written by Gabriel Maldonado. Optional arguments added by Andres Cabrera and Istvan Varga. New in Csound 5 (Previously available only on CsoundAV) manual_src~dfsg/opcodes/kdump2.xml0000644000000000000000000000076212262561504016332 0ustar rootroot kdump2 kdump2 kdump2 Deprecated. Description Deprecated as of version 3.49. Use the dumpk2 opcode instead. manual_src~dfsg/imageopcodes/0000755000000000000000000000000012262561504015404 5ustar rootrootmanual_src~dfsg/imageopcodes/top.xml0000644000000000000000000000154412262561504016734 0ustar rootroot Image processing opcodes Here is a list of opcodes that read/write image files: imagecreate imagesize imagegetpixel imagesetpixel imagesave imageload imagefree manual_src~dfsg/vectorial/0000755000000000000000000000000012262561504014735 5ustar rootrootmanual_src~dfsg/vectorial/limiting.xml0000644000000000000000000000277712262561504017310 0ustar rootroot
Limiting and wrapping of vectorial control signals The opcodes to perform limiting and wrapping of elements within a vector are: vlimit vwrap vmirror These opcodes are similar to limit, wrap and mirror, but operate on a vector instead of a scalar signal. The old values of the vector contained in an f-table are over-written if they are out of min/max interval. If you want to keep the original values of the input vector, use the vcopy opcode to copy it in another table. All these opcodes work at k-rate. All these operators are designed to be used together with other opcodes that operate with vectorial signals such as vcella, adsynt, adsynt2 etc.
manual_src~dfsg/vectorial/delay.xml0000644000000000000000000000104012262561504016550 0ustar rootroot
Vectorial Control-rate Delay Paths Vectorial Control-rate Delay Paths: vdelayk vport vecdelay
manual_src~dfsg/vectorial/scalar.xml0000644000000000000000000000273212262561504016730 0ustar rootroot
Operations Between a Vectorial and a Scalar Signal These opcodes perform numeric operations between a vectorial control signal (hosted inside a function table), and a scalar signal. Result is a new vector that overrides old values of the table. There are k-rate and i-rate versions of the opcodes. All these operators are designed to be used together with other opcodes that operate with vectorial signals such as bmscan, vcella, adsynt, adsynt2 etc. Operations Between a Vectorial and a Scalar Signal: vadd vmult vpow vexp vadd_i vmult_i vpow_i vexp_i
manual_src~dfsg/vectorial/vectorial.xml0000644000000000000000000000512212262561504017447 0ustar rootroot
Operations Between two Vectorial Signals These opcodes perform operations between two vectors, that is, each element of the first vector is processed with the corresponding element of the other vector. The result is a new vector that overrides the old values of the source vector. Operations Between two Vectorial Signals: vaddv vsubv vmultv vdivv vpowv vexpv vcopy vmap vaddv_i vsubv_i vmultv_i vdivv_i vpowv_i vexpv_i vcopy_i All these operators are designed to be used together with other opcodes that operate with vectorial signals such as vcella, adsynt, adsynt2, etc.
manual_src~dfsg/vectorial/tables.xml0000644000000000000000000000303012262561504016725 0ustar rootroot
Tables of vectors operators The following Vectorial opocodes support read/write access to arrays of vectors (or arrays of arrays): vtablei vtable1k vtablek vtablea vtablewi vtablewk vtablewa vtabi vtabk vtaba vtabwi vtabwk vtabwa
manual_src~dfsg/vectorial/top.xml0000644000000000000000000000076312262561504016267 0ustar rootroot Vectorial Opcodes The vectorial opcode family is designed to allow sections of f-tables to be treated as vectors for diverse operations on them. &vectorialtables; &vectorialscalar; &vectorialvectorial; &vectorialenvelopeGenerators; &vectoriallimiting; &vectorialdelay; &vectorialrandom; manual_src~dfsg/vectorial/vectorialrandom.xml0000644000000000000000000000114112262561504020645 0ustar rootroot
Vectorial Random Signal Generators These opcodes generate vectors of random numbers to be stored in tables. They generate a sort of 'vectorial band-limited noise'. All these opcodes work at k-rate. Vectorial random signal generators: vrandh and vrandi. Cellular automata vectors can be generated using: vcella.
manual_src~dfsg/vectorial/envelopeGenerators.xml0000644000000000000000000000222212262561504021324 0ustar rootroot
Vectorial Envelope Generators The opcodes to generate vectors containing envelopes are vlinseg and vexpseg. These opcodes are similar to linseg and expseg, but operate with vectorial signals instead of with scalar signals. Output is a vector hosted by an f-table (that must be previously allocated), while each break-point of the envelope is actually a vector of values. All break-points must contain the same number of elements (ielements). These operators are designed to be used together with other opcodes that operate with vectorial signals such as vcella, adsynt, adsynt2, etc.
manual_src~dfsg/readme.txt0000644000000000000000000001503212262561504014744 0ustar rootroot[ABOUT THE MANUAL]-------------------------------------------------------------- The Csound Reference Manual is written in Docbook-XML. More information about Docbook-XML can be found at the following links: http://docbook.org/ http://www.sagehill.net/book-description.html If you find problems or have suggestions, please report them on the csound developer mailing list. [BUILDING THE MANUAL]----------------------------------------------------------- To build the manual, a number of different tools are required, depending on what target output is desired. All targets require xsltproc and the Docbook XML DTDs to be installed and findable by xsltproc. The Docbook XSL stylesheets have to be installed too. The location of the stylesheets can be noted via the XSL_BASE_PATH variable: $ make XSL_BASE_PATH=path/to/xsl/installation The default is /usr/share/xml/docbook/stylesheet/nwalsh/ On Linux, most distributions include these programs, or are available on packaging repositories for the respective distribution. For Windows, xsltproc can be found as part of the Cygwin environment (http://www.cywgin.com). A natively compiled version for windows can be found at: http://zlatkovic.com/libxml.en.html Mac OS 10.5 (leopard) includes xsltproc. (NOTE: If someone works on OSX or OS9 and finds xsltproc for these platforms or another suitable XSLT processor, please update this documentation.) The manual is made using make with the included Makefile. For example to make the html (canonical) version: $ make html is all that is needed. To begin a new build, use $ make clean Other targets include pdf, pdfA4 and htmlXO. See the makefile for distribution and other targets. [HTML] Use: $ make html Requires: Python (to generate the frames version, and to update the Quick Reference) [PDF] Use: $ make pdf or $ make pdfA4 Requires: Apache FOP (http://xml.apache.org/fop/) Java Advanced Imaging Library (https://jai.dev.java.net/binary-builds.html) A Java Runtime Environment to run the above FOP can require a lot of memory, so it is recommended to edit the fop.sh file and add "-Xmx384m" to the last line, so it reads as such: $JAVACMD -Xmx384m -classpath "$LOCALCLASSPATH" $FOP_OPTS org.apache.fop.apps.Fop "$@" This increases the max ram the VM can use to 384megs, which you'll need for FOP to run as the manual is fairly large. [HTMLHELP] Use: $ make htmlhelp Requires: Microsoft HTML Help Workshop (http://msdn.microsoft.com/library/default.asp?url=/library/en-us/htmlhelp/html/hwMicrosoftHTMLHelpDownloads.asp) This target compiles a Windows .chm file, a format that replaced WinHelp. [EDITING THE MANUAL]------------------------------------------------------------ [MODIFYING AN ENTRY] In general, be careful to be mindful that these files are XML and must be valid. All tags that are openned must be closed, i.e. My Information. Also, Docbook-XML has a DTD that describes what is well-formed for the document, meaning that certain tags only allow other tags within it. For more information on valid Docbook tags, see "Docbook: The Definitive Guide" by Norm Walsh, available at http://docbook.org/. [ADDING AN OPCODE ENTRY] First, the best place to start is to take an existing entry and use that as a template. All opcode entries are organized in the opcodes folder. You can base your entry on an existing entry, or use Opcodes/templates.xml. To incorporate a new entry into the manual, a few things are required. 1)Add the entry as an entity in manual.xml. For example, if you put the myOpcodeEntry.xml in the opcodes directory, then this entry should be the one to add in manual.xml: 2)Add the entry to opcodes/top.xml by using the entity. This will actually add the entry to the reference manual. The entries are alphabetically ordered, so just find where your opcode should be in the list and add: &opcodesmyopcodeentry; The above uses the entity that was defined in the manual.xml file (think of it like an #include). 3)After that, you'll probably want to find the section headers where the opcode should be cross-referenced from. For example, if myOpcodeEntry should be categorized with other pvs opcodes in spectral/realtime.xml, it would be added in that file as: My Opcode Entry The linkend attribute above points to a docbook ID. The "myOpcodeEntry" ID should be defined in the myOpcodeEntry.xml, probably to be done at the top level like this: Repeat step 3 for any sections you think this opcode should be categorized in. 4) Add the appropriate info tag so the opcode will be properly categorized in the Quick Reference. If no info tag is provided the opcode will appear in 'Miscellaneous'. Consult available categories in quickref.py. 5) If possible reference the opcode in the appropriate section in Section II of the manual, and add the necessary cross references. 6) For transforming the myOpcodeEntry.csd example file to myOpcodeEntry.csd.xml (in examples-xml), use the csd2docbook.py or csd2docbook2.py script. The former uses the Csound API for an opcode list, while the latter uses a file opcode_list.txt (which is similar to the result of the command 'csound -z'). For instance: $ python csd2docbook2.py -f genarray_i.csd will create the file genarray_i.csd.xml from genarray_i.csd for use in the genarray_i opcode entry. [FOR MAINTAINERS]--------------------------------------------------------------- There are several targets that prepare files for release. It is important to remember to change the version number so that files and content are generated with that number. It must be changed both in manual.xml and the Makefile. It is also a good idea to update the What's new section for each release. [GENERAL XML EDITING TIPS]------------------------------------------------------ The "<" and ">" symbol, when used as text within XML tags, need to be escaped. The corresponding entities to use are "<" and ">". The refsect1 tag ends with the number 1, not the letter l. You can test your XML file by opening it in a web browser (i.e. Firefox, Mozilla, Internet Explorer, etc.). Most browsers will display the XML file if it is correct or will give you an error message if it the XML file is not valid, as well as give you information on where it is in error. Using a dedicated XML editor takes some of the pain out of editing the manual. Something like emacs in sgml mode or Kate in XML mode are very helpful. manual_src~dfsg/Makefile0000644000000000000000000007432212262561504014415 0ustar rootroot# ************************************************************************* # Module: Makefile # Purpose: Generates different versions of the Csound manual. # Note: Modified from the Makefile example in the DocBook-Install-HOWTO. # You may need to change program names and paths to get this to work # on your system. # ************************************************************************* BASENAME=Csound VERSION=6.02 # export PATH=$PATH:/home/andres/bin/fop-0.20.5/ # export JAVA_HOME=/usr/lib/jvm/java-6-sun/ # Output Directories HTML_DIR=html HTML_ONE_DIR=singleHTML HTMLHELP_DIR=htmlhelp MANPAGES_DIR=manpages # Stylesheets ifneq "$(wildcard /usr/share/xml/docbook/stylesheet/nwalsh/*/docbook.xsl)" "" XSL_BASE_PATH=/usr/share/xml/docbook/stylesheet/nwalsh/ else ifneq "$(wildcard /usr/share/xml/docbook/stylesheet/nwalsh/*/*/docbook.xsl)" "" # some distros version the docbook stylesheets text__=$(patsubst %manpages/docbook.xsl,%, $(wildcard /usr/share/xml/docbook/stylesheet/nwalsh/*/manpages/docbook.xsl)) # We want the newest version XSL_BASE_PATH=$(word $(words $(text__)), $(sort $(text__))) else endif XSL_HTML=xsl/customXHTMLchunk.xsl XSL_HTML_ONECHUNK=xsl/customXHTMLOneChunk.xsl XSL_PRINT=xsl/customPDF.xsl XSL_HTMLHELP=xsl/customHTMLHelp.xsl # Manpages don't need custom anything XSL_MANPAGES=$(XSL_BASE_PATH)/manpages/docbook.xsl COMMAND = command/desc.xml command/flags.xml command/preproc.xml \ command/top.xml command/unifile.xml CONTROL = control/clockctl.xml control/conditional.xml control/durctl.xml \ control/fltkappearance.xml control/fltkcontainers.xml \ control/fltkgeneral.xml control/fltkintro.xml control/fltkother.xml \ control/fltkvaluators.xml control/invocat.xml \ control/pgmctl.xml control/realtime.xml \ control/reinitn.xml control/sensing.xml control/subinstr.xml \ control/timeread.xml control/top.xml XMLS = cscore/advancd.xml cscore/compile.xml \ cscore/events.xml cscore/main.xml cscore/top.xml \ examples/opcode1.xml \ math/amp.xml math/artlogic.xml \ math/mathfunc.xml math/opeqfunc.xml math/rndfunc.xml \ math/top.xml math/trig.xml pitch/funcs.xml \ pitch/top.xml pitch/tuning.xml \ preface/copy.xml preface/preface.xml preface/top.xml \ spectral/nonstand.xml spectral/realtime.xml spectral/top.xml \ table/queries.xml table/readwrit.xml table/select.xml \ table/top.xml zak/top.xml MIDI = midi/convert.xml midi/extender.xml \ midi/generic.xml midi/onoff.xml midi/output.xml \ midi/realtime.xml midi/sliderbk.xml midi/top.xml \ MISC = misc/amp.xml misc/csound64.xml misc/formants.xml misc/glossary.xml \ misc/pitch.xml misc/quickref.xml misc/sf2.xml \ misc/top.xml misc/windows.xml OPCODES = opcodes/0dbfs.xml opcodes/abetarand.xml \ opcodes/abexprnd.xml opcodes/abs.xml \ opcodes/acauchy.xml opcodes/active.xml opcodes/adds.xml \ opcodes/adsr.xml opcodes/adsyn.xml opcodes/adsynt.xml \ opcodes/adsynt2.xml opcodes/aexprand.xml opcodes/aftouch.xml \ opcodes/agauss.xml opcodes/agogobel.xml opcodes/alinrand.xml \ opcodes/alpass.xml opcodes/ampdb.xml opcodes/ampdbfs.xml \ opcodes/ampmidid.xml \ opcodes/ampmidi.xml opcodes/apcauchy.xml opcodes/apoisson.xml \ opcodes/apow.xml opcodes/areson.xml opcodes/aresonk.xml \ opcodes/assign.xml opcodes/atone.xml opcodes/atonek.xml \ opcodes/atonex.xml opcodes/atrirand.xml opcodes/aunirand.xml \ opcodes/aweibull.xml opcodes/babo.xml opcodes/balance.xml \ opcodes/bamboo.xml opcodes/bbcutm.xml opcodes/bbcuts.xml \ opcodes/betarand.xml opcodes/bexprnd.xml opcodes/biquad.xml \ opcodes/biquada.xml opcodes/birnd.xml opcodes/bqrez.xml \ opcodes/butbp.xml opcodes/butbr.xml opcodes/buthp.xml \ opcodes/butlp.xml opcodes/butterbp.xml opcodes/butterbr.xml \ opcodes/butterhp.xml opcodes/butterlp.xml opcodes/button.xml \ opcodes/buzz.xml opcodes/cabasa.xml opcodes/cauchy.xml \ opcodes/ceil.xml opcodes/cent.xml opcodes/cggoto.xml \ opcodes/chanctrl.xml opcodes/changed.xml opcodes/checkbox.xml \ opcodes/cigoto.xml opcodes/ckgoto.xml opcodes/clear.xml \ opcodes/clfilt.xml opcodes/clip.xml opcodes/clock.xml \ opcodes/clockoff.xml opcodes/clockon.xml opcodes/cngoto.xml \ opcodes/comb.xml opcodes/control.xml opcodes/convle.xml \ opcodes/convolve.xml opcodes/copya2ftab.xml opcodes/copyf2array.xml \ opcodes/cos.xml opcodes/cosh.xml \ opcodes/cosinv.xml opcodes/cps2pch.xml opcodes/cpsmidi.xml \ opcodes/cpsmidib.xml opcodes/cpsoct.xml opcodes/cpspch.xml \ opcodes/cpstmid.xml opcodes/cpstun.xml opcodes/cpstuni.xml \ opcodes/cpsxpch.xml opcodes/cpuprc.xml opcodes/cross2.xml \ opcodes/crunch.xml opcodes/ctrl14.xml opcodes/ctrl21.xml \ opcodes/ctrl7.xml opcodes/ctrlinit.xml opcodes/cuserrnd.xml \ opcodes/dam.xml opcodes/db.xml opcodes/dbamp.xml \ opcodes/dbfsamp.xml opcodes/dcblock.xml opcodes/dconv.xml \ opcodes/define.xml opcodes/delay.xml opcodes/delay1.xml \ opcodes/delayr.xml opcodes/delayw.xml opcodes/deltap.xml \ opcodes/deltap3.xml opcodes/deltapi.xml opcodes/deltapn.xml \ opcodes/deltapx.xml opcodes/deltapxw.xml opcodes/diff.xml \ opcodes/diskin.xml opcodes/dispfft.xml opcodes/display.xml \ opcodes/distort1.xml opcodes/divides.xml opcodes/divz.xml \ opcodes/dollar.xml opcodes/downsamp.xml opcodes/dripwater.xml \ opcodes/dumpk.xml opcodes/dumpk2.xml opcodes/dumpk3.xml \ opcodes/dumpk4.xml opcodes/duserrnd.xml opcodes/dust.xml \ opcodes/dust2.xml opcodes/else.xml \ opcodes/elseif.xml opcodes/endif.xml opcodes/endin.xml \ opcodes/endop.xml opcodes/envlpx.xml opcodes/envlpxr.xml \ opcodes/equals.xml opcodes/event.xml opcodes/exitnow.xml \ opcodes/exp.xml opcodes/expon.xml opcodes/exprand.xml \ opcodes/expseg.xml opcodes/expsega.xml opcodes/expsegr.xml \ opcodes/filelen.xml opcodes/filenchnls.xml opcodes/filepeak.xml \ opcodes/filesr.xml opcodes/filter2.xml opcodes/fin.xml \ opcodes/fini.xml opcodes/fink.xml opcodes/fiopen.xml \ opcodes/flanger.xml opcodes/flashtxt.xml opcodes/flbox.xml \ opcodes/flbutbank.xml opcodes/flbutton.xml opcodes/flclosebutton.xml \ opcodes/flcolor.xml opcodes/flcolor2.xml opcodes/flcount.xml \ opcodes/flexecbutton.xml opcodes/flgetsnap.xml \ opcodes/flgroup.xml opcodes/flgroup_end.xml opcodes/flgroupend.xml \ opcodes/flhide.xml opcodes/fljoy.xml \ opcodes/flknob.xml opcodes/fllabel.xml opcodes/flloadsnap.xml \ opcodes/flooper.xml opcodes/floor.xml opcodes/flpack.xml \ opcodes/flpack_end.xml opcodes/flpackend.xml opcodes/flpanel.xml \ opcodes/flpanel_end.xml opcodes/flpanelend.xml opcodes/flprintk.xml \ opcodes/flprintk2.xml opcodes/flroller.xml opcodes/flrun.xml \ opcodes/flsavesnap.xml opcodes/flscroll.xml opcodes/flscroll_end.xml \ opcodes/flscrollend.xml opcodes/flsetalign.xml opcodes/flsetbox.xml \ opcodes/flsetcolor.xml opcodes/flsetcolor2.xml opcodes/flsetfont.xml \ opcodes/flsetposition.xml opcodes/flsetsize.xml \ opcodes/flsetsnap.xml opcodes/flsettext.xml \ opcodes/flsettextcolor.xml opcodes/flsettextsize.xml \ opcodes/flsettexttype.xml opcodes/flsetval.xml \ opcodes/flsetvali.xml opcodes/ifdef.xml \ opcodes/flshow.xml opcodes/flslidbnk.xml opcodes/flslider.xml \ opcodes/fltabs.xml opcodes/fltabs_end.xml opcodes/fltabsend.xml \ opcodes/fltext.xml opcodes/flupdate.xml opcodes/flvalue.xml \ opcodes/flvkeybd.xml \ opcodes/fmb3.xml opcodes/fmbell.xml opcodes/fmmetal.xml \ opcodes/fmpercfl.xml opcodes/fmrhode.xml opcodes/fmvoice.xml \ opcodes/fmwurlie.xml opcodes/fof.xml opcodes/fof2.xml \ opcodes/fofilter.xml opcodes/fog.xml opcodes/fold.xml \ opcodes/follow.xml opcodes/follow2.xml opcodes/foscil.xml \ opcodes/foscili.xml opcodes/fout.xml opcodes/fouti.xml \ opcodes/foutir.xml opcodes/foutk.xml opcodes/fprintks.xml \ opcodes/fprints.xml opcodes/frac.xml opcodes/fractalnoise.xml \ opcodes/ftchnls.xml \ opcodes/ftgen.xml opcodes/ftlen.xml opcodes/ftload.xml \ opcodes/ftloadk.xml opcodes/ftlptim.xml opcodes/ftmorf.xml \ opcodes/ftsave.xml opcodes/ftsavek.xml opcodes/ftsr.xml \ opcodes/gain.xml opcodes/gauss.xml opcodes/gausstrig.xml \ opcodes/gbuzz.xml opcodes/gendy.xml opcodes/gendyc.xml \ opcodes/gendyx.xml \ opcodes/gogobel.xml opcodes/goto.xml opcodes/grain.xml \ opcodes/grain2.xml opcodes/grain3.xml opcodes/granule.xml \ opcodes/greaterequal.xml opcodes/greaterthan.xml opcodes/guiro.xml \ opcodes/harmon.xml opcodes/hilbert.xml opcodes/hrtfer.xml \ opcodes/hsboscil.xml opcodes/ibetarand.xml opcodes/ibexprnd.xml \ opcodes/icauchy.xml opcodes/ictrl14.xml opcodes/ictrl21.xml \ opcodes/ictrl7.xml opcodes/iexprand.xml opcodes/if.xml \ opcodes/pvsifd.xml opcodes/igauss.xml opcodes/igoto.xml \ opcodes/ihold.xml opcodes/ilinrand.xml opcodes/imidic14.xml \ opcodes/imidic21.xml opcodes/imidic7.xml opcodes/in.xml \ opcodes/in32.xml opcodes/inch.xml opcodes/include.xml \ opcodes/inh.xml opcodes/init.xml opcodes/initc14.xml \ opcodes/initc21.xml opcodes/initc7.xml opcodes/ficlose.xml \ opcodes/ino.xml opcodes/inq.xml opcodes/ins.xml \ opcodes/instimek.xml opcodes/instimes.xml opcodes/instr.xml \ opcodes/int.xml opcodes/integ.xml opcodes/interp.xml \ opcodes/invalue.xml opcodes/inx.xml opcodes/inz.xml \ opcodes/ioff.xml opcodes/ion.xml opcodes/iondur.xml \ opcodes/iondur2.xml opcodes/ioutat.xml opcodes/ioutc.xml \ opcodes/ioutc14.xml opcodes/ioutpat.xml opcodes/ioutpb.xml \ opcodes/ioutpc.xml opcodes/ipcauchy.xml opcodes/ipoisson.xml \ opcodes/ipow.xml opcodes/is16b14.xml opcodes/is32b14.xml \ opcodes/islider16.xml opcodes/islider32.xml opcodes/islider64.xml \ opcodes/islider8.xml opcodes/itablecopy.xml opcodes/itablegpw.xml \ opcodes/itablemix.xml opcodes/itablew.xml opcodes/itrirand.xml \ opcodes/iunirand.xml opcodes/iweibull.xml opcodes/jitter.xml \ opcodes/jacktransport.xml \ opcodes/jitter2.xml opcodes/jspline.xml opcodes/kbetarand.xml \ opcodes/kbexprnd.xml opcodes/kcauchy.xml opcodes/kdump.xml \ opcodes/kdump2.xml opcodes/kdump3.xml opcodes/kdump4.xml \ opcodes/kexprand.xml opcodes/kfilter2.xml opcodes/kgauss.xml \ opcodes/kgoto.xml opcodes/klinrand.xml opcodes/kon.xml \ opcodes/koutat.xml opcodes/koutc.xml opcodes/koutc14.xml \ opcodes/koutpat.xml opcodes/koutpb.xml opcodes/koutpc.xml \ opcodes/kpcauchy.xml opcodes/kpoisson.xml opcodes/kpow.xml \ opcodes/kr.xml opcodes/kread.xml opcodes/kread2.xml \ opcodes/kread3.xml opcodes/kread4.xml opcodes/ksmps.xml \ opcodes/ktableseg.xml opcodes/ktrirand.xml opcodes/kunirand.xml \ opcodes/kweibull.xml opcodes/lessequal.xml opcodes/lessthan.xml \ opcodes/lfo.xml opcodes/limit.xml opcodes/line.xml \ opcodes/linen.xml opcodes/linenr.xml opcodes/lineto.xml \ opcodes/linrand.xml opcodes/linseg.xml opcodes/linsegr.xml \ opcodes/locsend.xml opcodes/locsig.xml opcodes/log.xml \ opcodes/log10.xml opcodes/logbtwo.xml opcodes/loopseg.xml \ opcodes/lorenz.xml opcodes/loscil.xml opcodes/loscil3.xml \ opcodes/lowpass2.xml opcodes/lowres.xml opcodes/lowresx.xml \ opcodes/lpf18.xml opcodes/lpfreson.xml opcodes/lphasor.xml \ opcodes/lpinterp.xml opcodes/lposcil.xml opcodes/lposcil3.xml \ opcodes/lpread.xml opcodes/lpreson.xml opcodes/lpshold.xml \ opcodes/lpslot.xml opcodes/mac.xml opcodes/maca.xml \ opcodes/madsr.xml opcodes/mandel.xml opcodes/mandol.xml \ opcodes/marimba.xml opcodes/massign.xml opcodes/max_k.xml \ opcodes/maxalloc.xml opcodes/mclock.xml opcodes/mdelay.xml \ opcodes/metro.xml opcodes/midic14.xml opcodes/midic21.xml \ opcodes/midic7.xml opcodes/midichannelaftertouch.xml \ opcodes/midichn.xml opcodes/midicontrolchange.xml \ opcodes/midictrl.xml opcodes/mididefault.xml \ opcodes/midiin.xml opcodes/midinoteoff.xml opcodes/midinoteoncps.xml \ opcodes/midinoteonkey.xml opcodes/midinoteonoct.xml \ opcodes/midinoteonpch.xml opcodes/midion.xml \ opcodes/midion2.xml opcodes/midiout.xml opcodes/midipitchbend.xml \ opcodes/midipolyaftertouch.xml opcodes/midiprogramchange.xml \ opcodes/miditempo.xml opcodes/mirror.xml opcodes/modulus.xml \ opcodes/moog.xml opcodes/moogladder.xml opcodes/moogvcf.xml \ opcodes/moscil.xml opcodes/mpulse.xml opcodes/mrtmsg.xml \ opcodes/multiplies.xml opcodes/multitap.xml opcodes/mute.xml \ opcodes/mxadsr.xml opcodes/nchnls.xml opcodes/nestedap.xml \ opcodes/nlfilt.xml opcodes/noise.xml opcodes/date.xml \ opcodes/noteoff.xml opcodes/noteon.xml opcodes/noteondur.xml \ opcodes/noteondur2.xml opcodes/notequal.xml opcodes/notnum.xml \ opcodes/nreverb.xml opcodes/nrpn.xml opcodes/nsamp.xml \ opcodes/nstrnum.xml opcodes/ntrpol.xml opcodes/octave.xml \ opcodes/octcps.xml opcodes/octmidi.xml opcodes/octmidib.xml \ opcodes/octpch.xml opcodes/opa.xml opcodes/opand.xml \ opcodes/opbitand.xml opcodes/opbitnot.xml opcodes/opbitor.xml \ opcodes/opcode.xml opcodes/opi.xml opcodes/opk.xml \ opcodes/opnonequiv.xml opcodes/opor.xml opcodes/oscbnk.xml \ opcodes/oscil.xml opcodes/oscil1.xml opcodes/oscil1i.xml \ opcodes/oscil3.xml opcodes/oscili.xml opcodes/oscilikt.xml \ opcodes/osciliktp.xml opcodes/oscilikts.xml opcodes/osciln.xml \ opcodes/oscils.xml opcodes/oscilx.xml opcodes/out.xml \ opcodes/out32.xml opcodes/outc.xml opcodes/outch.xml \ opcodes/outh.xml opcodes/outiat.xml opcodes/outic.xml \ opcodes/outic14.xml opcodes/outipat.xml opcodes/outipb.xml \ opcodes/outipc.xml opcodes/outkat.xml opcodes/dates.xml \ opcodes/outkc.xml opcodes/outkc14.xml opcodes/outkpat.xml \ opcodes/outkpb.xml opcodes/outkpc.xml opcodes/outo.xml \ opcodes/outq.xml opcodes/outq1.xml opcodes/outq2.xml \ opcodes/outq3.xml opcodes/outq4.xml opcodes/outs.xml \ opcodes/outs1.xml opcodes/outs2.xml opcodes/outvalue.xml \ opcodes/outx.xml opcodes/outz.xml opcodes/p.xml \ opcodes/pan.xml opcodes/pareq.xml opcodes/partials.xml \ opcodes/pcauchy.xml opcodes/pchbend.xml opcodes/pchmidi.xml \ opcodes/pchmidib.xml opcodes/pchoct.xml opcodes/peak.xml \ opcodes/peakk.xml opcodes/pgmassign.xml opcodes/phaser1.xml \ opcodes/phaser2.xml opcodes/phasor.xml opcodes/phasorbnk.xml \ opcodes/pinkish.xml opcodes/pitch.xml opcodes/pitchamdf.xml \ opcodes/planet.xml opcodes/pluck.xml opcodes/poisson.xml \ opcodes/polyaft.xml opcodes/port.xml opcodes/portk.xml \ opcodes/poscil.xml opcodes/poscil3.xml opcodes/pow.xml \ opcodes/powoftwo.xml opcodes/prealloc.xml opcodes/print.xml \ opcodes/printk.xml opcodes/printk2.xml opcodes/printks.xml \ opcodes/prints.xml opcodes/product.xml opcodes/pset.xml \ opcodes/ptable.xml opcodes/ptable3.xml opcodes/ptablei.xml \ opcodes/ptablew.xml \ opcodes/pvadd.xml opcodes/pvbufread.xml opcodes/pvcross.xml \ opcodes/pvinterp.xml opcodes/pvoc.xml opcodes/pvread.xml \ opcodes/pvsadsyn.xml opcodes/pvsanal.xml opcodes/pvsarp.xml \ opcodes/pvsblur.xml opcodes/pvscale.xml opcodes/pvscent.xml \ opcodes/pvscross.xml opcodes/pvsdemix.xml opcodes/pvsfilter.xml \ opcodes/pvsfread.xml opcodes/pvsftr.xml opcodes/pvsftw.xml \ opcodes/pvshift.xml opcodes/pvsinfo.xml opcodes/pvsinit.xml \ opcodes/pvsmaska.xml opcodes/pvsmix.xml opcodes/pvstencil.xml \ opcodes/pvsvoc.xml opcodes/pvsynth.xml opcodes/raises.xml \ opcodes/rand.xml opcodes/randh.xml opcodes/randi.xml \ opcodes/random.xml opcodes/randomh.xml opcodes/randomi.xml \ opcodes/readclock.xml opcodes/readk.xml opcodes/readk2.xml \ opcodes/readk3.xml opcodes/readk4.xml opcodes/reinit.xml \ opcodes/release.xml opcodes/repluck.xml opcodes/reson.xml \ opcodes/resonk.xml opcodes/resonr.xml opcodes/resonx.xml \ opcodes/resonxk.xml opcodes/resony.xml opcodes/resonz.xml \ opcodes/resyn.xml opcodes/reverb.xml opcodes/reverb2.xml \ opcodes/rezzy.xml opcodes/rigoto.xml opcodes/rireturn.xml \ opcodes/rms.xml opcodes/rnd.xml opcodes/rnd31.xml opcodes/tb.xml \ opcodes/round.xml opcodes/rspline.xml opcodes/rtclock.xml \ opcodes/s16b14.xml opcodes/s32b14.xml opcodes/samphold.xml \ opcodes/sandpaper.xml opcodes/scanhammer.xml opcodes/scans.xml \ opcodes/scantable.xml opcodes/scanu.xml opcodes/schedkwhen.xml \ opcodes/schedkwhennamed.xml opcodes/schedule.xml opcodes/tab.xml \ opcodes/schedwhen.xml opcodes/seqtime2.xml opcodes/tabrec.xml \ opcodes/seed.xml opcodes/sekere.xml opcodes/semitone.xml \ opcodes/sense.xml opcodes/sensekey.xml opcodes/seqtime.xml \ opcodes/setctrl.xml opcodes/setksmps.xml opcodes/sfilist.xml \ opcodes/sfinstr.xml opcodes/sfinstr3.xml opcodes/sfinstr3m.xml \ opcodes/sfinstrm.xml opcodes/sfload.xml opcodes/sfpassign.xml \ opcodes/sfplay.xml opcodes/sfplay3.xml opcodes/sfplay3m.xml \ opcodes/sfplaym.xml opcodes/sfplist.xml opcodes/sfpreset.xml \ opcodes/shaker.xml opcodes/sin.xml opcodes/sinh.xml \ opcodes/sininv.xml opcodes/sinsyn.xml opcodes/sleighbells.xml \ opcodes/slider16.xml opcodes/slider16f.xml opcodes/slider32.xml \ opcodes/slider32f.xml opcodes/slider64.xml opcodes/slider64f.xml \ opcodes/slider8.xml opcodes/slider8f.xml opcodes/sndloop.xml \ opcodes/sndwarp.xml opcodes/sndwarpst.xml opcodes/soundin.xml \ opcodes/soundout.xml opcodes/space.xml opcodes/spat3d.xml \ opcodes/spat3di.xml opcodes/spat3dt.xml opcodes/spdist.xml \ opcodes/specaddm.xml opcodes/specdiff.xml opcodes/specdisp.xml \ opcodes/specfilt.xml opcodes/spechist.xml opcodes/specptrk.xml \ opcodes/specscal.xml opcodes/specsum.xml opcodes/spectrum.xml \ opcodes/splitrig.xml opcodes/spsend.xml opcodes/sqrt.xml \ opcodes/sr.xml opcodes/statevar.xml opcodes/stix.xml \ opcodes/streson.xml opcodes/strset.xml opcodes/subinstr.xml \ opcodes/subinstrinit.xml opcodes/subtracts.xml opcodes/sum.xml \ opcodes/svfilter.xml opcodes/syncgrain.xml opcodes/table.xml \ opcodes/table3.xml opcodes/tablecopy.xml opcodes/tablegpw.xml \ opcodes/tablei.xml opcodes/tableicopy.xml opcodes/tableigpw.xml \ opcodes/tableikt.xml opcodes/tableimix.xml opcodes/tableiw.xml \ opcodes/tablekt.xml opcodes/tablemix.xml opcodes/tableng.xml \ opcodes/tablera.xml opcodes/tableseg.xml opcodes/tablew.xml \ opcodes/tablewa.xml opcodes/tablewkt.xml opcodes/tablexkt.xml \ opcodes/tablexseg.xml opcodes/tambourine.xml opcodes/tan.xml \ opcodes/tanh.xml opcodes/taninv.xml opcodes/taninv2.xml \ opcodes/tbvcf.xml opcodes/tempest.xml opcodes/tempo.xml \ opcodes/tempoval.xml opcodes/tigoto.xml opcodes/timeinstk.xml \ opcodes/timeinsts.xml opcodes/timek.xml opcodes/times.xml \ opcodes/timout.xml opcodes/tival.xml opcodes/tlineto.xml \ opcodes/tone.xml opcodes/tonek.xml opcodes/tonex.xml \ opcodes/top.xml opcodes/tradsyn.xml opcodes/transeg.xml \ opcodes/trigger.xml opcodes/trigseq.xml opcodes/trirand.xml \ opcodes/turnoff.xml opcodes/turnon.xml opcodes/undef.xml \ opcodes/unirand.xml opcodes/upsamp.xml opcodes/urd.xml \ opcodes/valpass.xml opcodes/vbap16.xml opcodes/vbap16move.xml \ opcodes/vbap4.xml opcodes/vbap4move.xml opcodes/vbap8.xml \ opcodes/vbap8move.xml opcodes/vbaplsinit.xml opcodes/vbapz.xml \ opcodes/vbapzmove.xml opcodes/vco.xml opcodes/vco2.xml \ opcodes/vco2ft.xml opcodes/vco2ift.xml opcodes/vco2init.xml \ opcodes/vcomb.xml opcodes/vdelay.xml opcodes/vdelay3.xml \ opcodes/vdelayx.xml opcodes/vdelayxq.xml opcodes/vdelayxs.xml \ opcodes/vdelayxw.xml opcodes/vdelayxwq.xml opcodes/vdelayxws.xml \ opcodes/veloc.xml opcodes/vibes.xml opcodes/vibr.xml \ opcodes/vibrato.xml opcodes/vincr.xml opcodes/vlowres.xml \ opcodes/voice.xml opcodes/vpvoc.xml opcodes/waveset.xml \ opcodes/weibull.xml opcodes/wgbow.xml opcodes/wgbowedbar.xml \ opcodes/wgbrass.xml opcodes/wgclar.xml opcodes/wgflute.xml \ opcodes/wgpluck.xml opcodes/wgpluck2.xml opcodes/wguide1.xml \ opcodes/wguide2.xml opcodes/wrap.xml opcodes/wterrain.xml \ opcodes/xadsr.xml opcodes/xin.xml opcodes/xout.xml \ opcodes/xscanmap.xml opcodes/xscans.xml opcodes/xscansmap.xml \ opcodes/xscanu.xml opcodes/xtratim.xml opcodes/xyin.xml \ opcodes/zacl.xml opcodes/zakinit.xml opcodes/zamod.xml \ opcodes/zar.xml opcodes/zarg.xml opcodes/zaw.xml \ opcodes/zawm.xml opcodes/zfilter2.xml opcodes/zir.xml \ opcodes/ziw.xml opcodes/ziwm.xml opcodes/zkcl.xml \ opcodes/zkmod.xml opcodes/zkr.xml opcodes/zkw.xml \ opcodes/zkwm.xml opcodes/timedseq.xml opcodes/barmodel.xml \ opcodes/expcurve.xml opcodes/gainslider.xml opcodes/logcurve.xml \ opcodes/scale.xml opcodes/harmon234.xml opcodes/remoteport.xml \ opcodes/insremot.xml opcodes/insglobal.xml opcodes/midremot.xml \ opcodes/midglobal.xml opcodes/system.xml opcodes/pan2.xml \ opcodes/pvsbandp.xml opcodes/pvsbandr.xml opcodes/hrtfstat.xml \ opcodes/hrtfmove.xml opcodes/hrtfmove2.xml ORCH = orch/dirfiles.xml orch/express.xml orch/header.xml orch/macros.xml\ orch/iblock.xml orch/kvar.xml orch/nomen.xml \ orch/statemnt.xml orch/top.xml SCORE = score/eval.xml score/macros.xml score/mult.xml \ score/nextp.xml score/preproc.xml score/ramp.xml \ score/statemnt.xml score/top.xml scoregens/a.xml \ scoregens/b.xml scoregens/e.xml scoregens/f.xml \ scoregens/gen01.xml scoregens/gen02.xml scoregens/gen03.xml \ scoregens/gen04.xml scoregens/gen05.xml scoregens/gen06.xml \ scoregens/gen07.xml scoregens/gen08.xml scoregens/gen09.xml \ scoregens/gen10.xml scoregens/gen11.xml scoregens/gen12.xml \ scoregens/gen13.xml scoregens/gen14.xml scoregens/gen15.xml \ scoregens/gen16.xml scoregens/gen17.xml scoregens/gen18.xml \ scoregens/gen19.xml scoregens/gen20.xml scoregens/gen21.xml \ scoregens/gen22.xml scoregens/gen23.xml scoregens/gen24.xml \ scoregens/gen25.xml scoregens/gen27.xml scoregens/gen28.xml \ scoregens/gen30.xml scoregens/gen31.xml scoregens/gen32.xml \ scoregens/gen33.xml scoregens/gen34.xml scoregens/gen40.xml \ scoregens/gen41.xml scoregens/gen42.xml scoregens/i.xml \ scoregens/m.xml scoregens/n.xml scoregens/q.xml \ scoregens/r.xml scoregens/s.xml scoregens/t.xml \ scoregens/top.xml scoregens/v.xml scoregens/x.xml OPCODES5 = csound5/acknowledgments.xml csound5/cs5opcodes.xml \ csound5/extending.xml \ csound5/opcodes/OSC.xml csound5/opcodes/OSCinit.xml \ csound5/opcodes/OSClisten.xml csound5/opcodes/OSCsend.xml \ csound5/opcodes/chani.xml csound5/opcodes/chano.xml \ csound5/opcodes/delayk.xml csound5/opcodes/denorm.xml \ csound5/opcodes/diskin2.xml csound5/opcodes/fluid.xml \ csound5/opcodes/fluidAllOut.xml csound5/opcodes/fluidCCi.xml \ csound5/opcodes/fluidCCk.xml csound5/opcodes/fluidControl.xml \ csound5/opcodes/fluidEngine.xml csound5/opcodes/fluidLoad.xml \ csound5/opcodes/fluidNote.xml csound5/opcodes/fluidOut.xml \ csound5/opcodes/fluidProgramSelect.xml csound5/opcodes/fluidSetInterpMethod.xml \ csound5/opcodes/freeverb.xml \ csound5/opcodes/ftconv.xml csound5/opcodes/loris.xml \ csound5/opcodes/lorismorph.xml csound5/opcodes/lorisplay.xml \ csound5/opcodes/lorisread.xml csound5/opcodes/mixer.xml \ csound5/opcodes/mixerclear.xml csound5/opcodes/mixergetlevel.xml \ csound5/opcodes/mixerreceive.xml csound5/opcodes/mixersend.xml \ csound5/opcodes/mixersetlevel.xml csound5/opcodes/pconvolve.xml \ csound5/opcodes/puts.xml csound5/opcodes/py.xml \ csound5/opcodes/pyassign.xml csound5/opcodes/pycall.xml \ csound5/opcodes/pyeval.xml csound5/opcodes/pyexec.xml \ csound5/opcodes/pyinit.xml csound5/opcodes/pyrun.xml \ csound5/opcodes/rbjeq.xml csound5/opcodes/reverbsc.xml \ csound5/opcodes/sprintf.xml csound5/opcodes/strcat.xml \ csound5/opcodes/strcmp.xml csound5/opcodes/strcpy.xml \ csound5/opcodes/strget.xml csound5/opcodes/stringconvert.xml \ csound5/opcodes/strtod.xml csound5/opcodes/strtodk.xml \ csound5/opcodes/strtol.xml csound5/opcodes/strtolk.xml \ csound5/opcodes/top.xml csound5/opcodes/turnoff2.xml \ csound5/opcodes/vst4cs.xml csound5/opcodes/vstaudio.xml \ csound5/opcodes/vstbankload.xml csound5/opcodes/vstedit.xml \ csound5/opcodes/vstinfo.xml csound5/opcodes/vstinit.xml \ csound5/opcodes/vstmidiout.xml csound5/opcodes/vstnote.xml \ csound5/opcodes/vstparamset.xml csound5/opcodes/vstprogset.xml \ csound5/opcodes/event_i.xml csound5/opcodes/ftfree.xml \ csound5/preface.xml csound5/todo.xml \ csound5/dssi4cs/dssiactivate.xml csound5/dssi4cs/dssiaudio.xml \ csound5/dssi4cs/dssictls.xml csound5/dssi4cs/dssiinit.xml \ csound5/dssi4cs/dssilist.xml csound5/dssi4cs/top.xml \ csound5/usersguide.xml csound5/opcodes/loop.xml \ csound5/opcodes/chnexport.xml csound5/opcodes/chnget.xml \ csound5/opcodes/chnparams.xml csound5/opcodes/chnset.xml \ csound5/opcodes/chnmix.xml csound5/opcodes/chnclear.xml \ csound5/opcodes/chn.xml csound5/opcodes/soundouts.xml \ csound5/opcodes/ftgentmp.xml csound5/opcodes/printf.xml \ csound5/opcodes/bformenc.xml csound5/opcodes/bformdec.xml \ csound5/opcodes/bformenc1.xml csound5/opcodes/bformdec1.xml SIGS = siggen/additive.xml siggen/basic.xml siggen/dynamic.xml \ siggen/fmsynth.xml siggen/granular.xml siggen/lineexp.xml \ siggen/models.xml siggen/phasors.xml \ siggen/random.xml siggen/sample.xml siggen/scantop.xml \ siggen/tableacc.xml siggen/top.xml \ siggen/waveterr.xml siggen/wavguide.xml sigio/fileio.xml \ sigio/input.xml sigio/output.xml sigio/pdisplay.xml \ sigio/queries.xml sigio/top.xml sigmod/ampmod.xml \ sigmod/conmorph.xml sigmod/delayops.xml \ sigmod/panspatl.xml sigmod/reverbtn.xml sigmod/sample.xml \ sigmod/siglimit.xml sigmod/speciale.xml sigmod/speciali.xml \ sigmod/standard.xml sigmod/top.xml sigmod/wavguide.xml UTILS = utility/analysis.xml utility/convert.xml utility/cvanal.xml \ utility/dnoise.xml utility/hetro.xml utility/lpanal.xml \ utility/pvanal.xml utility/pvlook.xml utility/queries.xml \ utility/sdif2ad.xml utility/sndinfo.xml utility/srconv.xml \ utility/top.xml utility/scale.xml utility/makecsd.xml \ utility/cs.xml utility/mixer.xml utility/extractor.xml \ utility/atsa.xml SRCS = $(COMMAND) $(CONTROL) $(XMLS) $(MIDI) $(OPCODES) $(ORCH) $(SCORE) \ $(SIGS) $(UTILS) # Build rules. all: html $(XSL_HTML) $(XSL_HTMLHELP) $(XSL_PRINT) $(XSL_HTML_ONECHUNK): %: %.in [ -d "$(XSL_BASE_PATH)" ] || ( \ echo "The XSL_BASE_PATH variable must be set to the XSL stylesheets installation directory" ; \ false ) sed -e 's|@xsl_base_path@|$(XSL_BASE_PATH)|' $@.in > $@ html: $(XSL_HTML) manual.xml $(SRCS) Makefile rm -rf html echo "Remember to use the html-dist target for distribution!" -mkdir -p ${HTML_DIR} xsltproc --xinclude -o ${HTML_DIR}/ ${XSL_HTML} manual.xml cp -R images ${HTML_DIR}/ cp -R examples ${HTML_DIR}/ cp copying.txt ${HTML_DIR}/ cp csound.css ${HTML_DIR}/ rm -rf ${HTML_DIR}/images/CVS rm -rf ${HTML_DIR}/images/callouts/CVS rm -rf ${HTML_DIR}/examples/CVS echo "Remember to use the html-dist target for distribution!" html-dist: python quickref.py $(MAKE) html python makeframes.py html-bw: $(XSL_HTML) manual.xml $(SRCS) Makefile -mkdir -p ${HTML_DIR} xsltproc --xinclude -o ${HTML_DIR}/ ${XSL_HTML} manual.xml cp -R images ${HTML_DIR}/ cp -R examples ${HTML_DIR}/ cp copying.txt ${HTML_DIR}/ cp csound-bw.css ${HTML_DIR}/ rm -rf ${HTML_DIR}/images/CVS rm -rf ${HTML_DIR}/images/callouts/CVS rm -rf ${HTML_DIR}/examples/CVS html-onechunk: $(XSL_HTML_ONECHUNK) manual.xml $(SRCS) Makefile -mkdir -p ${HTML_ONE_DIR} xsltproc --xinclude -o ${HTML_ONE_DIR}/ ${XSL_HTML_ONECHUNK} manual.xml cp -R images ${HTML_ONE_DIR}/ cp -R examples ${HTML_ONE_DIR}/ cp copying.txt ${HTML_ONE_DIR}/ cp csound.css ${HTML_ONE_DIR}/ rm -rf ${HTML_ONE_DIR}/images/CVS rm -rf ${HTML_ONE_DIR}/images/callouts/CVS rm -rf ${HTML_ONE_DIR}/examples/CVS pdf: $(XSL_PRINT) manual.xml $(SRCS) Makefile xsltproc --xinclude -o $(BASENAME)$(VERSION)_manual.fo ${XSL_PRINT} manual.xml fop.sh -fo $(BASENAME)$(VERSION)_manual.fo -pdf $(BASENAME)$(VERSION)_manual.pdf rm $(BASENAME)$(VERSION)_manual.fo pdfA4: $(XSL_PRINT) manual.xml $(SRCS) Makefile xsltproc --stringparam paper.type A4 --xinclude \ -o $(BASENAME)$(VERSION)_manual_A4.fo ${XSL_PRINT} manual.xml fop.sh -fo $(BASENAME)$(VERSION)_manual_A4.fo -pdf $(BASENAME)$(VERSION)_manual_A4.pdf rm $(BASENAME)$(VERSION)_manual_A4.fo htmlhelp: ${XSL_HTMLHELP} manual.xml $(SRCS) -mkdir -p ${HTMLHELP_DIR} xsltproc --xinclude -o ${HTMLHELP_DIR}/ ${XSL_HTMLHELP} manual.xml cp images -R ${HTMLHELP_DIR}/ cp examples -R ${HTMLHELP_DIR}/ cp csound.css ${HTMLHELP_DIR}/ rm -rf ${HTMLHELP_DIR}/images/CVS rm -rf ${HTMLHELP_DIR}/images/callouts/CVS rm -rf ${HTMLHELP_DIR}/examples/CVS hhc.exe ${HTMLHELP_DIR}/htmlhelp.hhp mv ${HTMLHELP_DIR}/htmlhelp.chm csound_manual.chm manpages: $(XSL_MANPAGES) manual.xml $(SRCS) Makefile -mkdir -p ${MANPAGES_DIR} xsltproc --xinclude -o ${MANPAGES_DIR}/ ${XSL_MANPAGES} manpages.xml html-clean: rm -rf html rm -f examples/*.csd~ pdf-clean: rm -f *.pdf *.fo htmlhelp-clean: rm -rf *.chm rm -rf htmlhelp manpages-clean: rm -rf manpages clean: html-clean pdf-clean htmlhelp-clean manpages-clean rm -f $(XSL_HTML) $(XSL_HTML_ONECHUNK) $(XSL_PRINT) $(XSL_HTMLHELP) distclean: clean rm -f *.gz rm -f *.zip distall: all # TEST VALIDITY OF XML test-xml: xmllint --valid --xinclude --dtdattr --noent manual.xml html-package: html-dist #tar -czf $(BASENAME)-html-$(VERSION).tar.gz csound-$(VERSION)/manual zip -r /tmp/$(BASENAME)$(VERSION)_manual_html.zip html mv /tmp/$(BASENAME)$(VERSION)_manual_html.zip ./ html-onechunk-package: #tar -czf $(BASENAME)-html-$(VERSION).tar.gz csound-$(VERSION)/manual zip -r /tmp/$(BASENAME)$(VERSION)_manual_html_single.zip singleHTML mv /tmp/$(BASENAME)$(VERSION)_manual_html_single.zip ./ pdf-package: #tar -czf $(BASENAME)-html-$(VERSION).tar.gz csound-$(VERSION)/manual zip -r /tmp/$(BASENAME)$(VERSION)_manual_pdf.zip $(BASENAME)$(VERSION)_manual.pdf mv /tmp/$(BASENAME)$(VERSION)_manual_pdf.zip ./ pdfA4-package: #tar -czf $(BASENAME)-html-$(VERSION).tar.gz csound-$(VERSION)/manual zip -r /tmp/$(BASENAME)$(VERSION)_manual_A4_pdf.zip $(BASENAME)$(VERSION)_manual_A4.pdf mv /tmp/$(BASENAME)$(VERSION)_manual_A4_pdf.zip ./ htmlhelp-package: zip -r /tmp/$(BASENAME)$(VERSION)_manual_winhelp_chm.zip csound_manual.chm mv /tmp/$(BASENAME)$(VERSION)_manual_winhelp_chm.zip ./ src-package: cvs -z3 -d:pserver:anonymous@csound.cvs.sourceforge.net:/cvsroot/csound export -rCsound-5_08 manual tar -czf manual.tar.gz manual/ text-package: clean $(TEXT_FILE) mkdir -p /tmp/csound-$(VERSION)/manual cp $(TEXT_FILE) /tmp/csound-$(VERSION)/manual cp *.txt /tmp/csound-$(VERSION)/manual cp -a ./examples /tmp/csound-$(VERSION)/manual tar -C /tmp -czf /tmp/$(BASENAME)-text-$(VERSION).tar.gz csound-$(VERSION)/manual mv /tmp/$(BASENAME)-text-$(VERSION).tar.gz . docbook-package: clean mkdir -p /tmp/csound-$(VERSION)/manual cp -a . /tmp/csound-$(VERSION)/manual tar -C /tmp -czf /tmp/$(BASENAME)-docbook-$(VERSION).tar.gz csound-$(VERSION)/manual mv /tmp/$(BASENAME)-docbook-$(VERSION).tar.gz . docbook-floppy: clean mkdir -p /tmp/csound-$(VERSION)/manual cp -a . /tmp/csound-$(VERSION)/manual tar -C /tmp -cf /tmp/$(BASENAME)-docbook-$(VERSION).tar csound-$(VERSION)/manual bzip2 -9 /tmp/$(BASENAME)-docbook-$(VERSION).tar mount /mnt/floppy mv /tmp/$(BASENAME)-docbook-$(VERSION).tar.bz2 /mnt/floppy umount /mnt/floppy dist: clean html-dist html-package html-onechunk html-onechunk-package pdf pdf-package pdfA4 pdfA4-package .PHONY: $(XSL_HTML) $(XSL_HTML_ONECHUNK) $(XSL_PRINT) $(XSL_HTMLHELP) manual_src~dfsg/opcode_list.txt0000644000000000000000000006116712262561504016025 0ustar rootrootATSadd ATSaddnz ATSbufread ATScross ATSinfo ATSinterpread ATSpartialtap ATSread ATSreadnz ATSsinnoi FLbox FLbutBank FLbutton FLcloseButton FLcolor FLcolor2 FLcount FLexecButton FLgetsnap FLgroup FLgroupEnd FLgroup_end FLhide FLhvsBox FLhvsBoxSetValue FLjoy FLkeyIn FLknob FLlabel FLloadsnap FLmouse FLpack FLpackEnd FLpack_end FLpanel FLpanelEnd FLpanel_end FLprintk FLprintk2 FLroller FLrun FLsavesnap FLscroll FLscrollEnd FLscroll_end FLsetAlign FLsetBox FLsetColor FLsetColor2 FLsetFont FLsetPosition FLsetSize FLsetSnapGroup FLsetText FLsetTextColor FLsetTextSize FLsetTextType FLsetVal FLsetVal_i FLsetVali FLsetsnap FLshow FLslidBnk FLslidBnk2 FLslidBnk2Set FLslidBnk2Setk FLslidBnkGetHandle FLslidBnkSet FLslidBnkSetk FLslider FLtabs FLtabsEnd FLtabs_end FLtext FLupdate FLvalue FLvkeybd FLvslidBnk FLvslidBnk2 FLxyin JackoAudioIn JackoAudioInConnect JackoAudioOut JackoAudioOutConnect JackoFreewheel JackoInfo JackoInit JackoMidiInConnect JackoMidiOut JackoMidiOutConnect JackoNoteOut JackoOn JackoTransport MixerClear MixerGetLevel MixerReceive MixerSend MixerSetLevel MixerSetLevel_i OSCinit OSClisten OSCsend a abs active adsr adsyn adsynt adsynt2 aftouch alpass alwayson ampdb ampdbfs ampmidi ampmidid areson aresonk array atone atonek atonex babo balance bamboo barmodel bbcutm bbcuts betarand bexprnd bformdec bformdec1 bformenc bformenc1 binit biquad biquada birnd bqrez butbp butbr buthp butlp butterbp butterbr butterhp butterlp button buzz cabasa cauchy cauchyi ceil cell cent centroid cggoto chanctrl changed chani chano chebyshevpoly checkbox chn_S chn_a chn_k chnclear chnexport chnget chnmix chnparams chnset chuap cigoto cingoto ckgoto clear clfilt clip clockoff clockon cngoto comb compileorc compilestr compress connect control convle convolve copy2ftab copy2ttab copya2ftab copyf2array cos cosh cosinv cosseg cossegb cossegr cps2pch cpsmidi cpsmidib cpsmidinn cpsoct cpspch cpstmid cpstun cpstuni cpsxpch cpumeter cpuprc cross2 crossfm crossfmi crossfmpm crossfmpmi crosspm crosspmi crunch ctlchn ctrl14 ctrl21 ctrl7 ctrlinit cuserrnd dam date dates db dbamp dbfsamp dcblock dcblock2 dconv delay delay1 delayk delayr delayw deltap deltap3 deltapi deltapn deltapx deltapxw denorm diff diskgrain diskin diskin2 dispfft display distort distort1 divz doppler downsamp dripwater dssiactivate dssiaudio dssictls dssiinit dssilist dumpk dumpk2 dumpk3 dumpk4 duserrnd dust dust2 endin endop envlpx envlpxr ephasor eqfil evalstr event event_i exitnow exp expcurve expon exprand exprandi expseg expsega expsegb expsegba expsegr fareylen fareyleni ficlose filebit filelen filenchnls filepeak filesr filevalid fillarray filter2 fin fini fink fiopen flanger flashtxt flooper flooper2 floor fluidAllOut fluidCCi fluidCCk fluidControl fluidEngine fluidLoad fluidNote fluidOut fluidProgramSelect fluidSetInterpMethod fmb3 fmbell fmmetal fmpercfl fmrhode fmvoice fmwurlie fof fof2 fofilter fog fold follow follow2 foscil foscili fout fouti foutir foutk fprintks fprints frac fractalnoise freeverb ftchnls ftconv ftcps ftfree ftgen ftgenonce ftgentmp ftlen ftload ftloadk ftlptim ftmorf ftresize ftresizei ftsave ftsavek ftsr gain gainslider gauss gaussi gausstrig gbuzz genarray genarray_i gendy gendyc gendyx getcfg gogobel goto grain grain2 grain3 granule guiro harmon harmon2 harmon3 harmon4 hilbert hrtfearly hrtfer hrtfmove hrtfmove2 hrtfreverb hrtfstat hsboscil hvs1 hvs2 hvs3 i igoto ihold imagecreate imagefree imagegetpixel imageload imagesave imagesetpixel imagesize in in32 inch inh init initc14 initc21 initc7 inleta inletf inletk inletkid ino inq inrg ins insglobal insremot instr int integ interp invalue inx inz jacktransport jitter jitter2 joystick jspline k kgoto ktableseg la_i_add_mc la_i_add_mr la_i_add_vc la_i_add_vr la_i_assign_mc la_i_assign_mr la_i_assign_t la_i_assign_vc la_i_assign_vr la_i_conjugate_mc la_i_conjugate_mr la_i_conjugate_vc la_i_conjugate_vr la_i_distance_vc la_i_distance_vr la_i_divide_mc la_i_divide_mr la_i_divide_vc la_i_divide_vr la_i_dot_mc la_i_dot_mc_vc la_i_dot_mr la_i_dot_mr_vr la_i_dot_vc la_i_dot_vr la_i_get_mc la_i_get_mr la_i_get_vc la_i_get_vr la_i_invert_mc la_i_invert_mr la_i_lower_solve_mc la_i_lower_solve_mr la_i_lu_det_mc la_i_lu_det_mr la_i_lu_factor_mc la_i_lu_factor_mr la_i_lu_solve_mc la_i_lu_solve_mr la_i_mc_create la_i_mc_set la_i_mr_create la_i_mr_set la_i_multiply_mc la_i_multiply_mr la_i_multiply_vc la_i_multiply_vr la_i_norm1_mc la_i_norm1_mr la_i_norm1_vc la_i_norm1_vr la_i_norm_euclid_mc la_i_norm_euclid_mr la_i_norm_euclid_vc la_i_norm_euclid_vr la_i_norm_inf_mc la_i_norm_inf_mr la_i_norm_inf_vc la_i_norm_inf_vr la_i_norm_max_mc la_i_norm_max_mr la_i_print_mc la_i_print_mr la_i_print_vc la_i_print_vr la_i_qr_eigen_mc la_i_qr_eigen_mr la_i_qr_factor_mc la_i_qr_factor_mr la_i_qr_sym_eigen_mc la_i_qr_sym_eigen_mr la_i_random_mc la_i_random_mr la_i_random_vc la_i_random_vr la_i_size_mc la_i_size_mr la_i_size_vc la_i_size_vr la_i_subtract_mc la_i_subtract_mr la_i_subtract_vc la_i_subtract_vr la_i_t_assign la_i_trace_mc la_i_trace_mr la_i_transpose_mc la_i_transpose_mr la_i_upper_solve_mc la_i_upper_solve_mr la_i_vc_create la_i_vc_set la_i_vr_create la_i_vr_set la_k_a_assign la_k_add_mc la_k_add_mr la_k_add_vc la_k_add_vr la_k_assign_a la_k_assign_f la_k_assign_mc la_k_assign_mr la_k_assign_t la_k_assign_vc la_k_assign_vr la_k_conjugate_mc la_k_conjugate_mr la_k_conjugate_vc la_k_conjugate_vr la_k_current_f la_k_current_vr la_k_distance_vc la_k_distance_vr la_k_divide_mc la_k_divide_mr la_k_divide_vc la_k_divide_vr la_k_dot_mc la_k_dot_mc_vc la_k_dot_mr la_k_dot_mr_vr la_k_dot_vc la_k_dot_vr la_k_f_assign la_k_get_mc la_k_get_mr la_k_get_vc la_k_get_vr la_k_invert_mc la_k_invert_mr la_k_lower_solve_mc la_k_lower_solve_mr la_k_lu_det_mc la_k_lu_det_mr la_k_lu_factor_mc la_k_lu_factor_mr la_k_lu_solve_mc la_k_lu_solve_mr la_k_mc_set la_k_mr_set la_k_multiply_mc la_k_multiply_mr la_k_multiply_vc la_k_multiply_vr la_k_norm1_mc la_k_norm1_mr la_k_norm1_vc la_k_norm1_vr la_k_norm_euclid_mc la_k_norm_euclid_mr la_k_norm_euclid_vc la_k_norm_euclid_vr la_k_norm_inf_mc la_k_norm_inf_mr la_k_norm_inf_vc la_k_norm_inf_vr la_k_norm_max_mc la_k_norm_max_mr la_k_qr_eigen_mc la_k_qr_eigen_mr la_k_qr_factor_mc la_k_qr_factor_mr la_k_qr_sym_eigen_mc la_k_qr_sym_eigen_mr la_k_random_mc la_k_random_mr la_k_random_vc la_k_random_vr la_k_subtract_mc la_k_subtract_mr la_k_subtract_vc la_k_subtract_vr la_k_t_assign la_k_trace_mc la_k_trace_mr la_k_upper_solve_mc la_k_upper_solve_mr la_k_vc_set la_k_vr_set lenarray lentab lfo limit line linen linenr lineto linrand linseg linsegb linsegr locsend locsig log log10 log2 logbtwo logcurve loop_ge loop_gt loop_le loop_lt loopseg loopsegp looptseg loopxseg lorenz loscil loscil3 loscilx lowpass2 lowres lowresx lpf18 lpform lpfreson lphasor lpinterp lposcil lposcil3 lposcila lposcilsa lposcilsa2 lpread lpreson lpshold lpsholdp lpslot lua_exec lua_iaopcall lua_iaopcall_off lua_ikopcall lua_ikopcall_off lua_iopcall lua_iopcall_off lua_opdef mac maca madsr mandel mandol maparray maparray_i marimba massign max max_k maxabs maxabsaccum maxaccum maxalloc maxarray maxtab mclock mdelay median mediank metro midglobal midic14 midic21 midic7 midichannelaftertouch midichn midicontrolchange midictrl mididefault midifilestatus midiin midinoteoff midinoteoncps midinoteonkey midinoteonoct midinoteonpch midion midion2 midiout midipgm midipitchbend midipolyaftertouch midiprogramchange miditempo midremot min minabs minabsaccum minaccum minarray mincer mintab mirror mode modmatrix monitor moog moogladder moogvcf moogvcf2 moscil mp3in mp3len mpulse mrtmsg multitap mute mxadsr nestedap nlalp nlfilt nlfilt2 noise noteoff noteon noteondur noteondur2 notnum nreverb nrpn nsamp nstrnum ntrpol octave octcps octmidi octmidib octmidinn octpch opcode oscbnk oscil oscil1 oscil1i oscil3 oscili oscilikt osciliktp oscilikts osciln oscils oscilx out out32 outc outch outh outiat outic outic14 outipat outipb outipc outkat outkc outkc14 outkpat outkpb outkpc outleta outletf outletk outletkid outo outq outq1 outq2 outq3 outq4 outrg outs outs1 outs2 outvalue outx outz p pan pan2 pareq partials partikkel partikkelget partikkelset partikkelsync passign pcauchy pchbend pchmidi pchmidib pchmidinn pchoct pconvolve pcount pdclip pdhalf pdhalfy peak pgmassign pgmchn phaser1 phaser2 phasor phasorbnk pindex pinkish pitch pitchac pitchamdf planet platerev plltrack pluck poisson polyaft polynomial pop pop_f port portk poscil poscil3 pow powershape powoftwo prealloc prepiano print printf printf_i printk printk2 printks prints product pset ptable ptable3 ptablei ptableiw ptablew ptrack push push_f puts pvadd pvbufread pvcross pvinterp pvoc pvread pvs2tab pvsadsyn pvsanal pvsarp pvsbandp pvsbandr pvsbin pvsblur pvsbuffer pvsbufread pvsbufread2 pvscale pvscent pvscross pvsdemix pvsdiskin pvsdisp pvsenvftw pvsfilter pvsfread pvsfreeze pvsftr pvsftw pvsfwrite pvsgain pvsgendy pvshift pvsifd pvsin pvsinfo pvsinit pvslock pvsmaska pvsmix pvsmooth pvsmorph pvsosc pvsout pvspitch pvstanal pvstencil pvsvoc pvswarp pvsynth pwd pyassign pyassigni pyassignt pycall pycall1 pycall1i pycall1t pycall2 pycall2i pycall2t pycall3 pycall3i pycall3t pycall4 pycall4i pycall4t pycall5 pycall5i pycall5t pycall6 pycall6i pycall6t pycall7 pycall7i pycall7t pycall8 pycall8i pycall8t pycalli pycalln pycallni pycallt pyeval pyevali pyevalt pyexec pyexeci pyexect pyinit pylassign pylassigni pylassignt pylcall pylcall1 pylcall1i pylcall1t pylcall2 pylcall2i pylcall2t pylcall3 pylcall3i pylcall3t pylcall4 pylcall4i pylcall4t pylcall5 pylcall5i pylcall5t pylcall6 pylcall6i pylcall6t pylcall7 pylcall7i pylcall7t pylcall8 pylcall8i pylcall8t pylcalli pylcalln pylcallni pylcallt pyleval pylevali pylevalt pylexec pylexeci pylexect pylrun pylruni pylrunt pyrun pyruni pyrunt qinf qnan rand randh randi random randomh randomi rbjeq readclock readf readfi readk readk2 readk3 readk4 readks readscore readscratch reinit release remoteport remove repluck reson resonk resonr resonx resonxk resony resonz resyn return reverb reverb2 reverbsc rewindscore rezzy rigoto rireturn rms rnd rnd31 round rspline rtclock s16b14 s32b14 samphold sandpaper scale scalearray scalet scanhammer scans scantable scanu schedkwhen schedkwhennamed schedule schedwhen scoreline scoreline_i seed sekere semitone sense sensekey seqtime seqtime2 serialBegin serialEnd serialFlush serialPrint serialRead serialWrite serialWrite_i setctrl setksmps setscorepos sfilist sfinstr sfinstr3 sfinstr3m sfinstrm sfload sflooper sfpassign sfplay sfplay3 sfplay3m sfplaym sfplist sfpreset shaker signum sin sinh sininv sinsyn sleighbells slicearray slider16 slider16f slider16table slider16tablef slider32 slider32f slider32table slider32tablef slider64 slider64f slider64table slider64tablef slider8 slider8f slider8table slider8tablef sliderKawai sndload sndloop sndwarp sndwarpst sockrecv sockrecvs socksend socksend_k socksends soundin soundout soundouts space spat3d spat3di spat3dt spdist specaddm specdiff specdisp specfilt spechist specptrk specscal specsum spectrum splitrig sprintf sprintfk spsend sqrt stack statevar stix strcat strcatk strchar strchark strcmp strcmpk strcpy strcpyk strecv streson strget strindex strindexk strlen strlenk strlower strlowerk strrindex strrindexk strset strsub strsubk strtod strtodk strtol strtolk strupper strupperk stsend subinstr subinstrinit sum sumarray sumtab svfilter syncgrain syncloop syncphasor system system_i tab tab2pvs tab_i tabgen table table3 table3kt tablecopy tablefilter tablefilteri tablegpw tablei tableicopy tableigpw tableikt tableimix tableiw tablekt tablemix tableng tablera tableseg tableshuffle tableshufflei tablew tablewa tablewkt tablexkt tablexseg tabmap tabmap_i tabmorph tabmorpha tabmorphak tabmorphi tabplay tabrec tabslice tabsum tabw tabw_i tambourine tan tanh taninv taninv2 tb0 tb0_init tb1 tb10 tb10_init tb11 tb11_init tb12 tb12_init tb13 tb13_init tb14 tb14_init tb15 tb15_init tb1_init tb2 tb2_init tb3 tb3_init tb4 tb4_init tb5 tb5_init tb6 tb6_init tb7 tb7_init tb8 tb8_init tb9 tb9_init tbvcf tempest tempo temposcal tempoval tigoto timedseq timeinstk timeinsts timek times timout tival tlineto tone tonek tonex tradsyn trandom transeg transegb transegr trcross trfilter trhighest trigger trigseq trirand trlowest trmix trscale trshift trsplit turnoff turnoff2 turnon unirand upsamp urandom urd vadd vadd_i vaddv vaddv_i vaget valpass vaset vbap vbap16 vbap1move vbap4 vbap4move vbap8 vbap8move vbapg vbaplsinit vbapmove vbapz vbapzmove vcella vco vco2 vco2ft vco2ift vco2init vcomb vcopy vcopy_i vdel_k vdelay vdelay3 vdelayk vdelayx vdelayxq vdelayxs vdelayxw vdelayxwq vdelayxws vdivv vdivv_i vecdelay veloc vexp vexp_i vexpseg vexpv vexpv_i vibes vibr vibrato vincr vlimit vlinseg vlowres vmap vmirror vmult vmult_i vmultv vmultv_i voice vosim vphaseseg vport vpow vpow_i vpowv vpowv_i vpvoc vrandh vrandi vsubv vsubv_i vtaba vtabi vtabk vtable1k vtablea vtablei vtablek vtablewa vtablewi vtablewk vtabwa vtabwi vtabwk vwrap waveset weibull wgbow wgbowedbar wgbrass wgclar wgflute wgpluck wgpluck2 wguide1 wguide2 wiiconnect wiidata wiirange wiisend wrap writescratch wterrain xadsr xin xout xscanmap xscans xscansmap xscanu xtratim xyin zacl zakinit zamod zar zarg zaw zawm zfilter2 zir ziw ziwm zkcl zkmod zkr zkw zkwm manual_src~dfsg/siggen/0000755000000000000000000000000012262561504014221 5ustar rootrootmanual_src~dfsg/siggen/additive.xml0000644000000000000000000000144312262561504016536 0ustar rootroot
Additive Synthesis/Resynthesis The opcodes for additive synthesis and resynthesis are: adsyn adsynt adsynt2 hsboscil See the section Spectral processing for more information and further additive/resynthesis opcodes.
manual_src~dfsg/siggen/basic.xml0000644000000000000000000000443412262561504016031 0ustar rootroot
Basic Oscillators The basic oscillator opcodes are: (note that opcodes that end with 'i' implement linear interpolation and those that end with '3' implement cubic interpolation) Oscillator Banks: oscbnk Simple table oscillators: oscil, oscil3 and oscili. Simple, fast sine oscilator: oscils Precision oscilators: poscil and poscil3. More flexible oscillators: oscilikt, osciliktp, oscilikts and osciln (also called oscilx). Oscillators can also be constructed from generic table read opcodes. See the Table Read/Write operations section. LFOs lfo vibr vibrato See the section Table access for other table reading opcodes that can be used as oscillators. Also see the section Dynamic spectrum Oscillators.
manual_src~dfsg/siggen/sampleXO.xml0000644000000000000000000001461612262561504016503 0ustar rootroot
Sample Playback Opcodes that implement sample playback and looping are: bbcutm bbcuts flooper flooper2 loscil loscil3 loscilx lphasor lposcil lposcil3 lposcila lposcilsa lposcilsa2 sndloop waveset See also the Signal Input section for other ways to input sound.
Soundfonts Soundfont opcodes These opcodes can also use soundfonts to generate sound. sfilist sfinstr sfinstr3 sfinstr3m sfinstrm sfload sfpassign sfplay sfplay3 sfplay3m sfplaym sflooper sfplist sfpreset
manual_src~dfsg/siggen/fmsynth.xml0000644000000000000000000000326312262561504016437 0ustar rootroot
FM Synthesis The FM synthesis opcodes are: foscil foscili crossfm, crossfmi, crosspm, crosspmi, crossfmpm, and crossfmpmi. FM instrument models fmb3 fmbell fmmetal fmpercfl fmrhode fmvoice fmwurlie
manual_src~dfsg/siggen/phasors.xml0000644000000000000000000000122412262561504016421 0ustar rootroot
Phasors The opcodes that generate a moving phase value: phasor phasorbnk syncphasor These opcodes are useful in combination with the Table access opcodes.
manual_src~dfsg/siggen/dynamic.xml0000644000000000000000000000235612262561504016375 0ustar rootroot
Dynamic Spectrum Oscillators The opcodes that generate dynamic spectra are: Harmonic spectra: buzz and gbuzz Impulse generator: mpulse Band limited oscillators (analog modelled): vco and vco2 The following opcodes can be used to generate band-limited waveforms for use with vco2 and other oscillators: vco2init vco2ft vco2ift
manual_src~dfsg/siggen/wavguide.xml0000644000000000000000000000246312262561504016563 0ustar rootroot
Waveguide Physical Modeling The opcodes that implement waveguide physical modeling are: pluck repluck wgbow wgbowedbar wgbrass wgclar wgflute wgpluck wgpluck2 wguide1 wguide2
manual_src~dfsg/siggen/lineexp.xml0000644000000000000000000000374512262561504016420 0ustar rootroot
Linear and Exponential Generators The opcodes that generate linear or exponential curves or segments are: expon expcurve expseg expsega expsegr gainslider jspline line linseg linsegr logcurve loopseg loopsegp lpshold lpsholdp rspline scale transeg
manual_src~dfsg/siggen/envelope.xml0000644000000000000000000000226112262561504016561 0ustar rootroot
Envelope Generators The following envelope generators are available: adsr madsr mxadsr xadsr linen linenr envlpx envlpxr Consult the Linear and exponential generators section for additional methods to create envelopes.
manual_src~dfsg/siggen/hvsXO.xml0000644000000000000000000000142712262561504016016 0ustar rootroot
Hyper Vectorial Synthesis vphaseseg hvs1 hvs2 hvs3
manual_src~dfsg/siggen/top.xml0000644000000000000000000000055012262561504015545 0ustar rootroot Signal Generators &siggenadditive; &siggenbasic; &siggendynamic; &siggenfmsynth; &siggengranular; &siggenhvs; &siggenlineexp; &siggenenvelope; &siggenmodels; &siggenphasors; &siggenrandom; &siggensample; &siggenscantop; &siggentableacc; &siggenwaveterr; &siggenwavguide; manual_src~dfsg/siggen/scantop.xml0000644000000000000000000001410512262561504016413 0ustar rootroot
Scanned Synthesis Scanned synthesis is a variant of physical modeling, where a network of masses connected by springs is used to generate a dynamic waveform. The opcode scanu defines the mass/spring network and sets it in motion. The opcode scans follows a predefined path (trajectory) around the network and outputs the detected waveform. Several scans instances may follow different paths around the same network. These are highly efficient mechanical modelling algorithms for both synthesis and sonic animation via algorithmic processing. They should run in real-time. Thus, the output is useful either directly as audio, or as controller values for other parameters. The Csound implementation adds support for a scanning path or matrix. Essentially, this offers the possibility of reconnecting the masses in different orders, causing the signal to propagate quite differently. They do not necessarily need to be connected to their direct neighbors. Essentially, the matrix has the effect of molding this surface into a radically different shape. To produce the matrices, the table format is straightforward. For example, for 4 masses we have the following grid describing the possible connections:   1 2 3 4 1         2         3         4         Whenever two masses are connected, the point they define is 1. If two masses are not connected, then the point they define is 0. For example, a unidirectional string has the following connections: (1,2), (2,3), (3,4). If it is bidirectional, it also has (2,1), (3,2), (4,3)). For the unidirectional string, the matrix appears:   1 2 3 4 1 0 1 0 0 2 0 0 1 0 3 0 0 0 1 4 0 0 0 0 The above table format of the connection matrix is for conceptual convenience only. The actual values shown in te table are obtained by scans from an ASCII file using GEN23. The actual ASCII file is created from the table model row by row. Therefore the ASCII file for the example table shown above becomes: 0100001000010000 This matrix example is very small and simple. In practice, most scanned synthesis instruments will use many more masses than four, so their matrices will be much larger and more complex. See the example in the scans documentation. Please note that the generated dynamic wavetables are very unstable. Certain values for masses, centering, and damping can cause the system to blow up and the most interesting sounds to emerge from your loudspeakers! The supplement to this manual contains a tutorial on scanned synthesis. The tutorial, examples, and other information on scanned synthesis is available from the Scanned Synthesis page at cSounds.com. Scanned synthesis developed by Bill Verplank, Max Mathews and Rob Shaw at Interval Research between 1998 and 2000. Opcodes that implement scanned synthesis are: scanhammer scans scantable scanu xscanmap xscans xscansmap xscanu
manual_src~dfsg/siggen/sample.xml0000644000000000000000000001522412262561504016230 0ustar rootroot
Sample Playback Opcodes that implement sample playback and looping are: bbcutm bbcuts flooper flooper2 loscil loscil3 loscilx lphasor lposcil lposcil3 lposcila lposcilsa lposcilsa2 sndloop waveset See also the Signal Input section for other ways to input sound.
Soundfonts Fluid Opcodes The fluid family of opcodes wraps Peter Hannape's SoundFont 2 player, FluidSynth: fluidEngine for instantiating a FluidSynth engine, fluidSetInterpMethod for setting interpolation method for a channel in a FluidSynth engine, fluidLoad for loading SoundFonts, fluidProgramSelect for assigning presets from a SoundFont to a FluidSynth engine's MIDI channel, fluidNote for playing a note on a FluidSynth engine's MIDI channel, fluidCCi for sending a controller message at i-time to a FluidSynth engine's MIDI channel, fluidCCk for sending a controller message at k-rate to a FluidSynth engine's MIDI channel. fluidControl for playing and controlling loaded Soundfonts (using 'raw' MIDI messages), fluidOut for receiving audio from a single FluidSynth engine, and fluidAllOut for receiving audio from all FluidSynth engines. fluidAllOut fluidCCi fluidCCk fluidControl fluidEngine fluidLoad fluidNote fluidOut fluidProgramSelect fluidSetInterpMethod "Old" Soundfont opcodes These opcodes can also use soundfonts to generate sound. sfplay etc. were created for one purpose -- to use the samples in SoundFonts. The fluid opcodes were created for another purpose -- to use SoundFonts more or less the way they were designed to be used, i.e. using keyboard mappings, layers, internal processing, etc. sfilist sfinstr sfinstr3 sfinstr3m sfinstrm sfload sfpassign sfplay sfplay3 sfplay3m sfplaym sflooper sfplist sfpreset
manual_src~dfsg/siggen/random.xml0000644000000000000000000001011512262561504016221 0ustar rootroot
Random (Noise) Generators Opcodes that generate random numbers are: betarnd bexprnd cauchy cuserrnd duserrnd dust dust2 exprand fractalnoise gauss gausstrig linrand noise pcauchy pinkish poisson rand randh randi rnd31 random randomh randomi trirand unirand urd weibull jitter jitter2 trandom See seed which sets the global seed value for all x-class noise generators, as well as other opcodes that use a random call, such as grain. rand, randh, randi, rnd(x) and birnd(x) are not affected by seed. See also functions which generate random numbers in the section Random Functions.
manual_src~dfsg/siggen/waveterr.xml0000644000000000000000000000033312262561504016601 0ustar rootroot
Wave Terrain Synthesis The opcode that uses wave terrain synthesis is wterrain.
manual_src~dfsg/siggen/hvs.xml0000644000000000000000000000142012262561504015540 0ustar rootroot
Hyper Vectorial Synthesis vphaseseg hvs1 hvs2 hvs3 The opcode FLhvsBox can be used to display the phase position for 2-dimensional Hyper Vectorial Synthesis.
manual_src~dfsg/siggen/granular.xml0000644000000000000000000000347212262561504016564 0ustar rootroot
Granular Synthesis The granular synthesis opcodes are: diskgrain fof fof2 fog grain grain2 grain3 granule partikkel partikkelsync sndwarp sndwarpst syncgrain syncloop vosim
manual_src~dfsg/siggen/models.xml0000644000000000000000000000615512262561504016235 0ustar rootroot
Models and Emulations The following opcodes model or emulate the sounds of other instruments (some based on the STK toolkit by Perry Cook): bamboo barmodel cabasa crunch dripwater gogobel guiro mandol marimba moog sandpaper sekere shaker sleighbells stix tambourine vibes voice Other models and emulations lorenz planet prepiano Fractal Number (Mandelbrot set) generator: mandel chuap gendy gendyc gendyx
manual_src~dfsg/siggen/tableacc.xml0000644000000000000000000000403012262561504016476 0ustar rootroot
Table Access The opcodes that access tables are: oscil1 oscil1i osciln oscilx table table3 tablei Opcodes ending in 'i' implement linear interpolation and opcodes ending in '3' implement cubic interpolation. The following opcodes implement fast table reading/writing without boundary checks: tab tab_i tabw tabw_i See the sections Table Queries, Read/Write Operationsand Table Reading with Dynamic Selection for other table operations. Although tables with a size which is not a power of two can be created using a negative size (see f score statement), some opcodes will not accept them.
manual_src~dfsg/csound5/0000755000000000000000000000000012262561504014325 5ustar rootrootmanual_src~dfsg/csound5/preface.xml0000644000000000000000000000715012262561504016457 0ustar rootroot BarryVercoe Realizing music by digital computer involves synthesizing audio signals with discrete points or samples representative of continuous waveforms. There are many ways to do this, each affording a different manner of control. Direct synthesis generates waveforms by sampling a stored function representing a single cycle; additive synthesis generates the many partials of a complex tone, each with its own loudness envelope; subtractive synthesis begins with a complex tone and filters it. Non-linear synthesis uses frequency modulation and waveshaping to give simple signals complex characteristics, while sampling and storage of a natural sound allows it to be used at will. Since comprehensive moment-by-mmoment specification of sound can be tedious, control is gained in two ways: 1) from the instruments in an orchestra, and 2) from the events within a score. An orchestra is really a computer program that can produce sound, while a score is a body of data which that program can react to. Whether a rise-time characteristic is a fixed constant in an instrument, or a variable of each note in the score, depends on how the user wants to control it. The instruments in a Csound orchestra are defined in a simple syntax that invokes complex audio processing routines. A score passed to this orchestra contains numerically coded pitch and control information, in standard numeric score format. Although many users are content with this format, higher level score processing languages are often convenient. The programs making up the Csound system have a long history of development, beginning with the Music 4 program written at Bell Telephone Laboratories in the early 1960s by Max Mathews. That initiated the stored table concept and much of the terminology that has since enabled computer music researchers to communicate. Valuable additions were made at Princeton by the late Godfrey Winham in Music 4B; my own Music 360 (1968) was very indebted to his work. With Music 11 (1973) I took a different tack: the two distinct networks of control and audio signal processing stemmed from my intensive involvement in the preceding years in hardware synthesizer concepts and design. This division has been retained in Csound. Because it is written entirely in C, Csound is easily installed on any machine running Unix or C. At MIT it runs on VAX/DECstations under Ultrix 4.2, on SUNs under OS 4.1, SGIs under 5.0, on IBM PCs under DOS 6.2 and Windows 3.1, and on the AppleMacintosh under ThinkC 5.0. With this single language for defining the audio signal processing, and portable audio formats like AIFF andWAV, users can move easily from machine to machine. The 1991 version added phase vocoder, FOF, and spectral data types. 1992 saw MIDI converter and control units, enabling Csound to be run from MIDI score-files and external keyboards. In 1994 the sound analysis programs (lpc, pvoc) were integrated into the main load module, enabling all Csound processing to be run from a single executable, and Cscore could pass scores directly to the orchestra for iterative performance. The 1995 release introduced an expanded MIDI set with MIDI-based linseg, butterworth filters, granular synthesis, and an improved spectral-based pitch tracker. Of special importance was the addition of run-time event generating tools (Cscore and MIDI) allowing run-time sensing and response setups that enable interactive composition and experiment. It appeared that real-time software synthesis was now showing some real promise. manual_src~dfsg/csound5/cs5opcodes.xml0000644000000000000000000000023212262561504017113 0ustar rootroot Csound5 Opcodes &dssi4cs; &vst4cs; &loris; &OSC; &lua; &py; &mixer; &stringconvert; manual_src~dfsg/csound5/usersguide.xml0000644000000000000000000000000212262561504017216 0ustar rootroot manual_src~dfsg/csound5/acknowledgments.xml0000644000000000000000000000356512262561504020245 0ustar rootroot Acknowledgements Csound contains contributions from musicians, scientists, and programmers from around the world. They include (but are not limited to): Contributors Allan LeeAndres CabreraBarry VercoeBill Gardner Bill Verplank Dan Ellis David Macintyre Eli Breder Gabriel Maldonado Greg Sullivan Hans Mikelson Istvan Varga Jean Piche John ffitch John Ramsdell Marc Resibois Mark Dolson Matt Ingalls Max Mathews Michael Casey Michael Clark Michael Gogins Mike Berry Paris Smaragdis Perry Cook Peter Neubacker Peter Nix Rasmus Ekman Richard Dobson Richard Karpen Rob Shaw Robin Whittle Sean Costello Steven Yi Tom Erbe Victor Lazzarini Ville Pulkki
manual_src~dfsg/csound5/opcodes/0000755000000000000000000000000012262561504015761 5ustar rootrootmanual_src~dfsg/csound5/opcodes/py.xml0000644000000000000000000000671312262561504017142 0ustar rootroot Python Opcodes
Introduction Using the Python opcode family, you can interact with a Python interpreter embedded in Csound in five ways: Initialize the Python interpreter (the pyinit opcodes), Run a statement (the pyrun opcodes), Execute a script (the pyexec opcodes), Invoke a callable and pass arguments (the pycall opcodes), Evaluate an expression (the pyeval opcodes), or Change the value of a Python object, possibly creating a new Python object (the pyassign opcodes); and you can do any of these things: At i-time or at k-time, In the global Python namespace, or in a namespace specific to an individual instance of a Csound instrument (local or "l" context), And can you can retrieve from 0 to 8 return values from callables that accept N parameters. ...this means that there are many Python-related opcodes. But all of these opcodes share the same py prefix, and have a regular naming scheme: "py" + [optional context prefix] + [action name] + [optional x-time suffix]
Orchestra Syntax Blocks of Python code, and indeed entire scripts, can be embedded in Csound orchestras using the {{ and }} directives to enclose the script, as follows: sr=44100 kr=4410 ksmps=10 nchnls=1 pyinit giSinusoid ftgen 0, 0, 8192, 10, 1 pyruni {{ import random pool = [(1 + i/10.0) ** 1.2 for i in range(100)] def get_number_from_pool(n, p): if random.random() < p: i = int(random.random() * len(pool)) pool[i] = n return random.choice(pool) }} instr 1 k1 oscil 1, 3, giSinusoid k2 pycall1 "get_number_from_pool", k1 + 2, p4 printk 0.01, k2 endin Credits Copyright (c) 2002 by Maurizio Umberto Puxeddu. All rights reserved. Portions copyright (c) 2004 and 2005 by Michael Gogins.
manual_src~dfsg/csound5/opcodes/Jacko.xml0000644000000000000000000001034412262561504017534 0ustar rootroot Jacko Opcodes These opcodes enable the use of Jack ports from within Csound orchestras and instruments. Ports can receive or send audio or MIDI data, and send note data. The Jacko opcodes do not replace the Jack driver and Jack command-line options for Csound, nor do the Jacko opcodes work with them (hence the name "Jacko" instead of "Jack"). The Jacko opcodes are an independent facility that offers greater flexibility in signal routing. In addition, the Jacko opcodes can work with the Jack system in "freewheeling" mode, which enables the use of Jack-enabled external synthesizers, such as Aeolus or Pianoteq, to render Csound pieces either faster or, even more importantly, slower than real time. This is extremely useful for rendering complex pieces without dropouts using instruments, such as Aeolus, that may not be available except through Jack. The Jacko opcodes include: JackoInit, for initializing the current instance of Csound as a Jack client. JackoInfo, for printing information about the Jack daemon, its clients, their ports, and their connections. JackoFreewheel, for turning Jack's freewheeling mode on or off. JackoAudioInConnect, for creating a connection from an external Jack audio output port to a Jack port in Csound. JackoAudioOutConnect, for creating a connection from a Jack port in Csound to an external Jack audio input port. JackoMidiInConnect, for creating a connection from an external Jack MIDI port. MIDI events from Jack are received by Csound's regular MIDI opcodes and MIDI interop system. JackoMidiOutConnect, for creating a connection from a Jack port in Csound to an external Jack MIDI input port. JackoOn, for turning Jack ports in Csound on or off. JackoAudioIn, for receiving audio from a Jack input port in Csound, which in turn has received the audio from its connected external port. JackoAudioOut, for sending audio to a Jack output port in Csound, which in turn will send the audio on to its connected external port. JackoMidiOut, for sending MIDI channel messages to a Jack output port in Csound, which in turn will send the MIDI on to its connected external port. JackoNoteOut, for sending a note (with duration) to a Jack output port in Csound, which in turn will send the note on to its connected external port. JackoTransport, for controlling the Jack transport. A typical scenario for the use of the Jacko opcodes would be something like this. Example Here is an example of the Jacko opcodes. It uses the file jacko.csd. Example of the Jacko opcodes. Credits By: &namemichael; 2010 manual_src~dfsg/csound5/opcodes/signalflowgraph.xml0000644000000000000000000001170612262561504021677 0ustar rootroot Signal Flow Graph Opcodes These opcodes enable the use of signal flow graphs (AKA asynchronous data flow graphs) in Csound orchestras. Signals flow from the outlets of source instruments and are summed in the inlets of sink instruments. Signals may be krate, arate, or frate. Any number of outlets may be connected to any number of inlets. When a new instance of an instrument is instantiated during performance, the declared connections also are automatically instantiated. Signal flow graphs simplify the construction of complex mixers, signal processing chains, and the like. They also simplify the re-use of "plug and play" instrument definitions and even entire sub-orchestras, which can simply be #included and then "plugged in" to existing orchestras. Note that inlets and outlets are defined in instruments without reference to how they are connected. Connections are defined in the orchestra header. It is this separation that enables plug-in instruments. Inlets must be named. Instruments may be named or numbered, but in either case each source instrument must be defined in the orchestra before any of its sinks. Naming instruments makes it easier to connect outlets and inlets in any higher-level orchestra to inlets and outlets in any lower-level #included orchestra. The signal flow graph opcodes include: outleta, for sending an arate signal from any instrument out a named port. outletk, for sending a krate signal from any instrument out a named port. outletkid, similar to outletk, but receiving a krate signal only from an identified instance of a port. outletf, for sending an frate signal from any instrument out a named port. inleta, for receiving an arate signal through a named port. inletk, for receiving a krate signal through a named port. inletkid, similiar to inletk, but transmitting a signal only between inlet and outlet opcodes . inletf, for receiving an frate signal through a named port. connect, for routing the signal from a named outlet in a source instrument to a named inlet in a sink instrument. alwayson for permanently activating an instrument from the orchestra header, without need of a score statement, e.g. for use as an effect processor receiving inputs from a number of sources. ftgenonce for instantiating function tables from within instrument definitions, without need for f-statements in the score or ftgen opcodes in the orchestra header. A typical scenario for the use of these opcodes would be something like this. A set of instruments would be defined, each in its own orchestra file, and each instrument would define inlet ports, outlet ports, and function tables within itself. Such instruments are completely self-contained. Then, a set of effects processors, such as equalizers, reverbs, compressors, and so on, would also be defined, each in its own file. Then, a customized master orchestra would #include the instruments and effects to be used, route the outputs of some instruments into one equalizer and the outputs of other effects into another equalizer, then route the outputs of both equalizers into a reverb, the output of the reverb into a compressor, and the output of the compressor into a stereo output soundfile. Example Here is an example of the signal flow graph opcodes. It uses the file signalflowgraph.csd. Example of the signal flow graph opcodes. manual_src~dfsg/csound5/opcodes/fluid.xml0000644000000000000000000000255412262561504017614 0ustar rootroot SoundFont Opcodes The fluid family of opcodes wraps Peter Hannape's SoundFont 2 player, FluidSynth: fluidEngine for instantiating a FluidSynth engine, fluidLoad for loading SoundFonts, fluidProgramSelect for assigning presets from a SoundFont to a FluidSynth engine's MIDI channel, fluidNote for playing a note on a FluidSynth engine's MIDI channel, fluidCCi for sending a controller message at i-time to a FluidSynth engine's MIDI channel, fluidNote for sending a controller message at k-rate to a FluidSynth engine's MIDI channel fluidControl for playing and controlling loaded Soundfonts, fluidOut for receiving audio from a FluidSynth engine, and fluidAllOut for receiving audio from all FluidSynth engines. manual_src~dfsg/csound5/opcodes/top.xml0000644000000000000000000000016612262561504017310 0ustar rootroot Csound5 Orchestra Opcodes and Operators manual_src~dfsg/csound5/opcodes/lua.xml0000644000000000000000000001147012262561504017267 0ustar rootroot Lua Opcodes The purposes of the Lua opcodes are: Make it possible to write Csound code in a user-friendly, high-level language with full lexical scoping, structures and classes, and support for functional programming, using LuaJIT (the Lua programming language, implemented with a just-in-time compiler and foreign function interface). Require the installation of no third party software packages, or at least a minimum installation; also, require no build system or external compilation. Run really fast; typically, almost as fast as compiled C code, and several times faster than user-defined opcodes. Using the Lua opcode family, you can interact with the Lua interpreter and just-in-time compiler (luajit) embedded in Csound as follows: Execute any arbitrary block of Lua code (the lua_exec opcode), Define an opcode in Lua taking any number or type of parameters, and returning any number or type of parameters (the lua_opdef opcode), Call a Lua opcode at i-rate (the lua_iopcall opcode), Call a Lua opcode at i-rate and k-rate (the lua_ikopcall opcods), or Call a Lua opcode at i-rate and a-rate (the lua_iaopcall opcode). Lua is Portuguese for "moon." And Lua (http://www.lua.org) is a lightweight, efficient dynamic programming language, designed for embedding in C/C++ and extending with C/C++. Lua has a stack-based calling mechanism and provides a toolkit of features (tables, metatables, anonymous functions, and closures) with which many styles of object-oriented and functional programming may be implemented. Lua's syntax is only slightly harder than Python's. Lua is already one of the fastest dynamic languages; yet LuaJIT by Mike Pall (http://luajit.org) goes much further, giving Lua a just-in-time optimizing trace compiler for Intel architectures. LuaJIT includes an efficient foreign function interface (FFI) with the ability to define C arrays, structures, and other types in Lua. The speed of LuaJIT/FFI ranges from several times as fast as Lua, to faster (in some contexts) than optimized C. Example Here is an example of a Lua opcode, implementing a Moog ladder filter. For purposes of comparison, a user-defined opcode and the native Csound opcode that compute the same sound using the same algorithm also are shown, and timed.. The example uses the file luamoog.csd. Example of a Lua opcode. Credits Copyright (c) 2011 by Michael Gogins. All rights reserved. manual_src~dfsg/csound5/opcodes/mixer.xml0000644000000000000000000000215712262561504017634 0ustar rootroot Mixer Opcodes The Mixer family of opcodes provides a global mixer for Csound. The Mixer opcodes include MixerSend for sending (that is, mixing in) an arate signal from any instrument to a channel of a mixer buss, MixerReceive for receiving an arate signal from a channel of any mixer buss in any instrument, MixerSetLevel (krate) and MixerSetLevel_i (irate) for controlling the level of the signal sent from a particular send to a particular buss, MixerGetLevel for reading (at krate) the level for sending a signal from a particular send to a particular buss, and MixerClear for resetting the busses to zero before the next kperiod of a performance. manual_src~dfsg/csound5/extending.xml0000644000000000000000000005416112262561504017043 0ustar rootroot Extending Csound
Adding Unit Generators If the existing Csound unit generators do not suit your needs, it is relatively easy to extend Csound by writing new unit generators in C or C++. The translator, loader, and run-time monitor will treat your module just like any other, provided you follow some conventions. Historically, this has been done with builtin unit generators, that is, with code that is statically linked with the rest of the Csound executable. Today, the preferred method is to create plugin unit generators. These are dynamic link libraries (DLLs) on Windows, and loadable modules (shared libraries that are dlopened) on Linux. Csound searches for and loads these plugins at run time on the path defined in OPCODEDIR. You can also load plugin opcodes from the command line using the --opcode-lib flag. The advantage of this method, of course, is that plugins created by any developer at any time can be used with already existing versions of Csound.
Creating a Builtin Unit Generator You need a structure defining the inputs, outputs and workspace, plus some initialization code and some perf-time code. Let's put an example of these in two new files, newgen.h and newgen.c. The examples given are for Csound 5. For earlier versions, all opcode functions omit the first parameter (CSOUND *csound). /* newgen.h - define a structure */ /* Declares Csound structures and functions. */ #include "csoundCore.h" typedef struct { OPDS h; /* required header */ MYFLT *result, *istrt, *incr, *itime, *icontin; /* addr outarg, inargs */ MYFLT curval, vincr; /* private dataspace */ long countdown; /* ditto */ } RMP; /* newgen.c - init and perf code */ /* Declares Csound structures and functions. */ #include "csoundCore.h" /* Declares RMP structure. */ #include "newgen.h" int rampset (CSOUND *csound, RMP * p) /* at note initialization: */ { if (*p->icontin == FL(0.0)) p->curval = *p->istrt; /* optionally get new start value */ p->vincr = *p->incr / csound->esr; /* set s-rate increment per sec. */ p->countdown = *p->itime * csound->esr; /* counter for itime seconds */ return OK; } int ramp (CSOUND *csound, RMP * p) /* during note performance: */ { MYFLT *rsltp = p->result; /* init an output array pointer */ int nn = csound->ksmps; /* array size from orchestra */ do { *rsltp++ = p->curval; /* copy current value to output */ if (--p->countdown > 0) /* for the first itime seconds, */ p->curval += p->vincr; /* ramp the value */ } while (--nn); return OK; } Now we add this module to the translator table in entry1.c, under the opcode name rampt: #include "newgen.h" int rampset(CSOUND *, RMP *), ramp(CSOUND *, RMP *); /* opname dsblksiz thread outypes intypes iopadr kopadr aopadr */ { "rampt", S(RMP), 5, "a", "iiio", (SUBR) rampset, (SUBR) NULL, (SUBR) ramp }, Finally you must relink Csound with the new module. Add the name of the C file to the libCsoundSources list in the SConstruct file: libCsoundSources = Split(''' Engine/auxfd.c ... OOps/newgen.c ... Top/utility.c ''') Run scons just as you would for any other Csound build, and the new module will be built into your Csound. The above actions have added a new generator to the Csound language. It is an audio-rate linear ramp function which modifies an input value at a user-defined slope for some period. A ramp can optionally continue from the previous note's last value. The Csound manual entry would look like: ar rampt istart, islope, itime [, icontin] istart -- beginning value of an audio-rate linear ramp. Optionally overridden by a continue flag. islope -- slope of ramp, expressed as the y-interval change per second. itime -- ramp time in seconds, after which the value is held for the remainder of the note. icontin (optional) -- continue flag. If zero, ramping will proceed from input istart . If non-zero, ramping will proceed from the last value of the previous note. The default value is zero. The file newgen.h includes a one-line list of output and input parameters. These are the ports through which the new generator will communicate with the other generators in an instrument. Communication is by address, not value, and this is a list of pointers to values of type MYFLT (which is double if the macro USE_DOUBLE is defined, and float otherwise). There are no restrictions on names, but the input-output argument types are further defined by character strings in entry1.c (inargs, outargs). Inarg types are commonly x, a, k, and i, in the normal Csound manual conventions; also available are o (optional, defaulting to 0), p (optional, defaulting to 1). Outarg types include a, k, i and s (asig or ksig). It is important that all listed argument names be assigned a corresponding argument type in entry1.c. Also, i-type args are valid only at initialization time, and other-type args are available only at perf time. Subsequent lines in the RMP structure declare the work space needed to keep the code re-entrant. These enable the module to be used multiple times in multiple instrument copies while preserving all data. The file newgen.c contains two subroutines, each called with a pointer to the Csound instance and a pointer to the uniquely allocated RMP structure and its data. The subroutines can be of three types: note initialization, k-rate signal generation, a-rate signal generation. A module normally requires two of these: initialization, and either k-rate or a-rate subroutines which become inserted in various threaded lists of runnable tasks when an instrument is activated. The thread-types appear in entry1.c in two forms: isub, ksub and asub names; and a threading index which is the sum of isub=1, ksub=2, asub=4. The code itself may reference (but should only read) public members of the CSOUND structure defined in csoundCore.h, the most useful of which are: OPARMS *oparms MYFLT esr user-defined sampling rate MYFLT ekr user-defined control rate int ksmps user-defined ksmps int nchnls user-defined nchnls int oparms->odebug command-line -v flag int oparms->msglevel command-line -m level MYFLT tpidsr 2 * PI / esr
Function tables To access tablesstored function stored function tables, special help is available. The newly defined structure should include a pointer FUNC *ftp; initialized by the statement ftp = csound->FTFind(csound, p->ifuncno); where MYFLT *ifuncno is an i-type input argument containing the ftable number. The stored table is then at ftp->ftable, and other data such as length, phase masks, cps-to-incr converters, are also accessed from this pointer. See the FUNC structure in csoundCore.h, the csoundFTFind() code in fgens.c, and the code for oscset() and koscil() in OOps/ugens2.c.
Additional Space Sometimes the space requirement of a module is too large to be part of a structure (upper limit 65279 bytes, due to the unsigned short dsblksiz parameter and reserved codes >= 0xFF00), or it is dependent on an i-arg value which is not known until initialization. Additional space can be dynamically allocated and properly managed by including the line AUXCH auxch; in the defined structure (*p), then using the following style of code in the init module: csound->AuxAlloc(csound, npoints * sizeof(MYFLT), &p->auxch); The address of this auxiliary space is kept in a chain of such spaces belonging to this instrument, and is automatically managed while the instrument is being duplicated or garbage-collected during performance. The assignment void *auxp = p->auxch.auxp; will find the allocated space for init-time and perf-time use. See the LINSEG structure in ugens1.h and the code for lsgset() and klnseg() in OOps/ugens1.c.
File Sharing When accessing an external file often, or doing it from multiple places, it is often efficient to read the entire file into memory. This is accomplished by including the line MEMFIL *mfp; in the defined structure (*p), then using the following style of code in the init module: p->mfp = csound->ldmemfile(csound, filname); where char *filname is a string name of the file requested. The data read will be found between (char *) p->mfp->beginp; and (char *) p->mfp->endp; Loaded files do not belong to a particular instrument, but are automatically shared for multiple access. See the ADSYN structure in ugens3.h and the code for adset() and adsyn() in OOps/ugens3.c.
String arguments To permit a string input argument (MYFLT *ifilnam, say) in our defined structure (*p), assign it the argtype S in entry1.c, and include the following code in the init module: strcpy(filename, (char*) p->ifilnam); See the code for adset() in OOps/ugens3.c, lprdset() in OOps/ugens5.c, and pvset() in OOps/ugens8.c.
Adding a Plugin Unit Generator The procedure for creating a plugin unit generator is very similar to the procedure for creating a builtin. The actual unit generator code would normally be identical. The differences are as follows. Again supposing that your unit generator is named newgen, perform the following steps: Write your newgen.c and newgen.h file as you would for a builtin unit generator. Put these files in the csound5/Opcodes directory. #include "csdl.h" in your unit generator sources, instead of csoundCore.h. Add your OENTRY records and unit generator registration functions at the bottom of your C file. Example (but you can have as many unit generators in one plugin as you like): #define S sizeof static OENTRY localops[] = { { "rampt", S(RMP), 5, "a", "iiio", (SUBR) rampset, (SUBR) NULL, (SUBR)ramp }, }; /* * The following macro from csdl.h defines * the "csound_opcode_init()" opcode registration * function for the localops table. */ LINKAGE Add your plugin as a new target in the plugin opcodes section of the SConstruct build file: pluginEnvironment.SharedLibrary('newgen', Split('''Opcodes/newgen.c Opcodes/another_file_used_by_newgen.c Opcodes/yet_another_file_used_by_newgen.c''')) Run the Csound 5 build in the regular way.
<literal>OENTRY</literal> Reference The OENTRY structure (see H/csoundCore.h, Engine/entry1.c, and Engine/rdorch.c) contains the following public fields: opname, dsblksiz, thread, outypes, intypes, iopadr, kopadr, aopadr dsblksizThere are two types of opcodes, polymorphic and non-polymorphic. For non-polymorphic opcodes, the dsblksiz flag specifies the size of the opcode structure in bytes, and arguments are always passed to the opcode at the same rate. Polymorphic opcodes can accept arguments at different rates, and those arguments are actually dispatched to other opcodes as determined by the dsblksiz flag and the following naming convention (note: the following list is not complete, see Engine/entry1.c for all possible special dsblksiz codes): 0xffffThe type of the first output argument determines which unit generator function is actually called: XXX -> XXX.a, XXX.i, or XXX.k. 0xfffeThe types of the first two input arguments determine which unit generator function is actually called: XXX -> XXX.aa, XXX.ak, XXX.ka, or XXX.kk, as in the oscil unit generator. 0xfffdRefers to one input argument of type a or k, as in the peak unit generator. threadSpecifies the rate(s) at which the unit generator's functions are called, as follows: Rate at which ugens are called according to thread parameter 0i-rate or k-rate (B out only) 1i-rate 2k-rate 3i-rate and k-rate 4a-rate 5i-rate and a-rate 7i-rate and (k-rate or a-rate)
outypesLists the return values of the unit generator functions, if any. The types allowed are (note: the following list is not complete, see Engine/entry1.c for all possible output types): List of out types for ugens ii-rate scalar kk-rate scalar aa-rate vector xk-rate scalar or a-rate vector ff-rate streaming pvoc fsig type mmultiple a-rate output arguments
intypesLists the arguments the unit generator functions take, if any. The types allowed are (note: the following list is not complete, see Engine/entry1.c for all possible input types): List of in types ofr ugens ii-rate scalar kk-rate scalar aa-rate vector xa-rate scalar or a-rate vector ff-rate streaming pvoc fsig type SString B l mBegins an indefinite list of i-rate arguments (any count) MBegins an indefinite list of arguments (any rate, any count) NBegins an indefinite list of (optional a-, k-, i-, or S-rate)-rate arguments (any odd count) nBegins an indefinite list of i-rate arguments (any odd count) OOptional k-rate, defaulting to 0 oOptional i-rate, defaulting to 0 pOptional i-rate, defaulting to 1 qOptional i-rate, defaulting to 10 VOptional k-rate, defaulting to 0.5 vOptional i-rate, defaulting to 0.5 jOptional i-rate, defaulting to -1 hOptional i-rate, defaulting to 127 yBegins an indefinite list of a-rate arguments (any count) zBegins an indefinite list of k-rate arguments (any count) ZBegins an indefinite list of alternating k-rate and a-rate arguments (kaka...) (any count)
iopadrThe address of the unit generator function (of type int (*SUBR)(CSOUND *, void *)) that is called at i-time, or NULL for no function. kopadrThe address of the unit generator function (of type int (*SUBR)(CSOUND *, void *)) that is called at k-rate, or NULL for no function. aopadrThe address of the unit generator function (of type int (*SUBR)(CSOUND *, void *)) that is called at a-rate, or NULL for no function.
manual_src~dfsg/csd2docbook2.py0000644000000000000000000002560212262561504015602 0ustar rootroot# # Copyright (C) 2007 Francois Pinot # # This program is free software; you can redistribute it and/or # modify it under the terms of the GNU Lesser General Public License # as published by the Free Software Foundation; either version 2.1 # of the License, or (at your option) any later version. # # This program is distributed in the hope that it will be useful, # but WITHOUT ANY WARRANTY; without even the implied warranty of # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. # See the GNU Lesser General Public License for more details. # # You should have received a copy of the GNU Lesser General Public # License along with this program; if not, write to the # Free Software Foundation, Inc., # 59 Temple Place - Suite 330, Boston, MA 02111, USA # or connect to: # http://www.fsf.org/licensing/licenses/lgpl.html # #import csnd import getopt, sys, os, glob, re from xml.dom import minidom class OrchestraDict(dict): def __init__(self): self.initOpcodes() self.initHeaderSymbols() self.initBlockDelim() self.initOperators() self.initSeqCtrl() self.initMacros() self.labels = [] def initOpcodes(self): '''Read the list of csound opcodes from csound 5 API. Thus we're sure that the opcode list is up to date, however it will be complete only if csound has been built with all the plugins. It is needed to compile an orchestra and score before calling CsoundOpcodeList() in order to get the opcode list. So we use a dummy csd file for that purpose (tricky isn't it?) ''' #csound = csnd.CppSound() #dummy = ''' # #-dodac # # #instr 1 #print p3 #endin # # #i1 0 1 #e # #''' f = open("opcode_list.txt", 'r') file = f.read() f.close() #csound.Compile("dummy.csd") lo = file.split() #n = lo.Count() for i in xrange(len(lo)): self[lo[i]] = 'opc' #lo.Clear() #csound.Reset() #os.remove("dummy.csd") def initHeaderSymbols(self): self.update({ 'sr':'ohdr', 'kr':'ohdr', 'ksmps':'ohdr', 'nchnls':'ohdr', '0dbfs':'ohdr', 'ctrlinit':'ohdr', 'ftgen':'ohdr', 'massign':'ohdr', 'pgmassign':'ohdr', 'seed':'ohdr', 'strset':'ohdr'}) def initBlockDelim(self): self.update({ 'instr':'oblock', 'endin':'oblock', 'opcode':'oblock', 'endop':'oblock'}) def initOperators(self): self.update({ '==':'op', '!=':'op', '?':'op', '>':'op', '>=':'op', 'LeSsThAn':'op', 'LeSsThAn=':'op', '=':'op', '+':'op', '-':'op', '*':'op', '/':'op', '%':'op', '^':'op', 'AmPeRsAnDAmPeRsAnD':'op', '||':'op', 'AmPeRsAnD':'op', '|':'op', '~':'op', ':':'op'}) def initSeqCtrl(self): self.update({ 'cggoto':'octrl', 'cigoto':'octrl', 'ckgoto':'octrl', 'cngoto':'octrl', 'elseif':'octrl', 'else':'octrl', 'endif':'octrl', 'goto':'octrl', 'if':'octrl', 'igoto':'octrl', 'kgoto':'octrl', 'tigoto':'octrl', 'timout':'octrl', 'loop_ge':'octrl', 'loop_gt':'octrl', 'loop_le':'octrl', 'loop_lt':'octrl'}) def initMacros(self): self.update({ '#define':'omacro', '#ifdef':'omacro', '#ifndef':'omacro', '#end':'omacro', '#else':'omacro', '#include':'omacro', '#undef':'omacro'}) def addLabel(self, key): self[key] = 'olabel' self.labels.append(key) def clearLabels(self): for key in self.labels: if self.has_key(key): self.pop(key) self.labels = [] def get(self, key, x=None): if self.has_key(key): return '' + key + '' return key class OrchestraTransform(object): def __init__(self): motif = '''( \s+ | \( | == | != | LeSsThAn= | LeSsThAn | >= | > | \? | = | \+ | - | \* | / | % | ^ | AmPeRsAnDAmPeRsAnD | \|\| | AmPeRsAnD | \| | ~ | : )''' self.pattern = re.compile(motif, re.VERBOSE) self.labelPattern = re.compile('[a-zA-Z]\w*:') self.orcDict = OrchestraDict() def detectLabels(self, lines): for s in lines: c = self.labelPattern.match(s) if c: self.orcDict.addLabel(s[:c.end()-1]) def comment(self, s): return '' + s + '' def orcLine(self, line): inTokens = self.pattern.split(line) outTokens = [] for t in inTokens: if len(t) > 0: outTokens.append(self.orcDict.get(t, line)) return ''.join(outTokens) def transformLines(self, lines): self.orcDict.clearLabels() self.detectLabels(lines) outLines = [] for s in lines: i = s.find(';') if i == 0: outLines.append(self.comment(s)) elif i > 0: outLines.append(self.orcLine(s[:i]) + self.comment(s[i:])) else: outLines.append(self.orcLine(s)) return outLines def transform(self, filename): f = open(filename, 'r') s = f.read() f.close() s = s.replace('&', 'AmPeRsAnD') s = s.replace('<', 'LeSsThAn') s = s.replace('LeSsThAnCs', '\n') fout.write(s) fout.write('') fout.close() print filename class ScoreTransform(object): def __init__(self): self.pattern = re.compile('^\s*[abefimnqrstvx]') def comment(self, s): return '' + s + '' def scoLine(self, line): r = self.pattern.search(line) if r: s = line[r.end()-1] return line.replace(s, '' + s + '') return line def transformLines(self, lines): outLines = [] for s in lines: i = s.find(';') if i == 0: outLines.append(self.comment(s)) elif i > 0: outLines.append(self.scoLine(s[:i]) + self.comment(s[i:])) else: outLines.append(self.scoLine(s)) return outLines class CsdTransform(object): def __init__(self): self.orc_t = OrchestraTransform() self.sco_t = ScoreTransform() def text(self, node): s = node.childNodes[0].nodeValue if s[0] == '\n': start = 1 else: start = 0 if s[-1] == '\n': end = -1 else: end = None return s[start:end].split('\n') def optTransform(self, lines): comment = lambda s: '' + s + '' outLines = [] for s in lines: i = s.find(';') if i == 0: outLines.append(comment(s)) elif i > 0: outLines.append(s[:i] + comment(s[i:])) else: outLines.append(s) return outLines def transform(self, filename): tag = lambda s: '<' + s + '>' f = open(filename, 'r') s = f.read() f.close() # We convert & and < in "pseudo" entities because we don't want that # minidom resolves those entities s = s.replace('&', 'AmPeRsAnD') s = s.replace('<', 'LeSsThAn') s = s.replace('LeSsThAnCs', ' 0: options = options_elem[0] orchestra = csddoc.getElementsByTagName("CsInstruments")[0] score = csddoc.getElementsByTagName("CsScore")[0] outLines = [] outLines.append('') outLines.append(tag("CsoundSynthesizer")) if len(options_elem) > 0: outLines.append(tag("CsOptions")) outLines.extend(self.optTransform(self.text(options))) outLines.append(tag("/CsOptions")) outLines.append(tag("CsInstruments")) outLines.extend(self.orc_t.transformLines(self.text(orchestra))) outLines.append(tag("/CsInstruments")) outLines.append(tag("CsScore")) outLines.extend(self.sco_t.transformLines(self.text(score))) outLines.append(tag("/CsScore")) outLines.append(tag("/CsoundSynthesizer")) outLines.append('') csddoc.unlink() # We convert back our "pseudo" entities in actual entities s = '\n'.join(outLines).replace('AmPeRsAnD', '&') s = s.replace('LeSsThAn', '<') outfn = filename.replace('examples', 'examples-xml') fout = open(outfn + '.xml', 'w') fout.write(s) fout.close() print filename def main(): '''Usage: python csd2xml option [filename] where option is one of the following: -f filename or --file=filename transform examples/filename to examples-xml/filename.xml where filename is the name of a csd file or orc file -a or --all transform all the csd files of the example directory to xml files in the examples-xml directory.''' try: opts, args = getopt.getopt(sys.argv[1:], "f:a", ["file=", "all"]) except getopt.GetoptError: print main.__doc__ sys.exit(2) for o, a in opts: if o in ("-f", "--file") and a: if a.find('examples') < 0: a = 'examples' + os.sep + a if os.path.exists(a): suf = os.path.splitext(a)[1] if suf == '.csd': CsdTransform().transform(a) elif suf == '.orc': OrchestraTransform().transform(a) else: print a + ' is not a csd or orc filename' sys.exit(0) else: print a + " doesn't exists!" sys.exit(2) if o in ("-a", "--all"): ct = CsdTransform() infiles = glob.glob('examples/*.csd') for f in infiles: ct.transform(f) sys.exit(0) print main.__doc__ sys.exit(2) if __name__ == '__main__': main() manual_src~dfsg/sigio/0000755000000000000000000000000012262561504014057 5ustar rootrootmanual_src~dfsg/sigio/fileio.xml0000644000000000000000000000334212262561504016052 0ustar rootroot
File Input and Output The opcodes for file input and output are: File open/close: fiopen and ficlose. File output: dumpk, dumpk2, dumpk3, dumpk4, fout, fouti, foutir and foutk File input: readk, readk2, readk3, readk4, fin, fini and fink Utilities for use with the fout opcodes: clear, vincr Printing to a file: fprints and fprintks
manual_src~dfsg/sigio/pdisplay.xml0000644000000000000000000000225112262561504016426 0ustar rootroot
Printing and Display Opcodes for printing and displaying values are: dispfft display flashtxt print printf printf_i printk printk2 printks prints
manual_src~dfsg/sigio/softbus.xml0000644000000000000000000000214212262561504016265 0ustar rootroot
Software Bus Csound implements a software bus for internal routing or routing to external software calling the Csound API. The opcodes to use the software bus are: chn_k chn_a chn_S chnclear chnexport chnmix chnparams
manual_src~dfsg/sigio/output.xml0000644000000000000000000000362312262561504016145 0ustar rootroot
Signal Output The opcodes that write audio signals are: Synchronous output: out, out32, outc, outch, outh, outo, outrg, outq, outq1, outq2, outq3, outq4, outs,outs1, outs2 and outx Streaming output: soundout and soundouts User defined channel output: outvalue Direct from zak output: outz The opcode monitor can be used for monitoring the complete output of csound (the output spout frame). See the section Software Bus for input and output through the API.
manual_src~dfsg/sigio/outputXO.xml0000644000000000000000000000365712262561504016423 0ustar rootroot
Signal Output The opcodes that write audio signals are: Synchronous output: out, outch, outrg, outs,outs1, outs2 Streaming output: soundout and soundouts User defined channel output: outvalue Direct from zak output: outz The opcode monitor can be used for monitoring the complete output of csound (the output spout frame). See the section Software Bus for input and output through the API.
manual_src~dfsg/sigio/top.xml0000644000000000000000000000026212262561504015403 0ustar rootroot Signal Input and Output &sigiofileio; &sigioinput; &sigiooutput; &sigiosoftbus; &sigiopdisplay; &sigioqueries; manual_src~dfsg/sigio/inputXO.xml0000644000000000000000000000276312262561504016217 0ustar rootroot
Signal Input The opcodes that receive audio signals are: Synchronous input: in, inch, inrg, ins File streaming: diskin, diskin2 and soundin User defined channel input: invalue Streaming input: soundin See the section Software Bus for input and output through the API.
manual_src~dfsg/sigio/queries.xml0000644000000000000000000000113212262561504016253 0ustar rootroot
Sound File Queries The opcodes that query information about files are: filelen filenchnls filepeak filesr
manual_src~dfsg/sigio/input.xml0000644000000000000000000000322312262561504015740 0ustar rootroot
Signal Input The opcodes that receive audio signals are: Synchronous input: in, in32, inch, inh, ino, inq, inrg, ins and inx File streaming: diskin, diskin2 and soundin User defined channel input: invalue Streaming input: soundin Direct to zak input: inz See the section Software Bus for input and output through the API. The mp3in allows reading of mp3 files, which are currently not supported by ordinary reading methods inside Csound.
manual_src~dfsg/misc/0000755000000000000000000000000012262561504013700 5ustar rootrootmanual_src~dfsg/misc/amp.xml0000644000000000000000000000223212262561504015176 0ustar rootroot Sound Intensity Values Sound Intensity Values (for a 1000 Hz tone) Dynamics Intensity (W/m ˆ 2) Level (dB) pain 1 120 fff 10 ˆ -2 100 f 10 ˆ -4 80 p 10 ˆ -6 60 ppp 10 ˆ -8 40 threshold 10 ˆ -12 0
manual_src~dfsg/misc/windows.xml0000644000000000000000000001257612262561504016127 0ustar rootroot Window Functions Windowing functions are used for analysis, and as waveform envelopes, particularly in granular synthesis. Window functions are built in to some opcodes, but others require a function table to generate the window. GEN20 is used for this purpose. The diagram of each window below, is accompanied by the f statement used to generate the it. Hamming Hamming window function statement f81 0 8192 20 1 1 Hamming Window Function. Hamming Window Function. Hanning Hanning window function statement f82 0 8192 20 2 1 Hanning Window Function Hanning Window Function Bartlett Bartlett window function statement f83 0 8192 20 3 1 Bartlett Window Function Bartlett Window Function Blackman Blackman window function statement f84 0 8192 20 4 1 Blackman Window Function Blackman Window Function Blackman-Harris Blackman-Harris window function statement f85 0 8192 20 5 1 Blackman-Harris Window Function Blackman-Harris Window Function Gaussian Gaussian window function statement f86 0 8192 20 6 1 Gaussian Window Function Gaussian Window Function Rectangle Rectangle window function statement f88 0 8192 -20 8 .1 Note: Vertical scale is exaggerated in this diagram. Rectangle Window Function Rectangle Window Function Sync Sync window function statement f89 0 4096 -20 9 .75 Sync Window Function Sync Window Function manual_src~dfsg/misc/formants.xml0000644000000000000000000006211212262561504016255 0ustar rootroot Formant Values alto <quote>a</quote> Values f1 f2 f3 f4 f5 freq (Hz) 800 1150 2800 3500 4950 amp (dB) 0 -4 -20 -36 -60 bw (Hz) 80 90 120 130 140
alto <quote>e</quote> Values f1 f2 f3 f4 f5 freq (Hz) 400 1600 2700 3300 4950 amp (dB) 0 -24 -30 -35 -60 bw (Hz) 60 80 120 150 200
alto <quote>i</quote> Values f1 f2 f3 f4 f5 freq (Hz) 350 1700 2700 3700 4950 amp (dB) 0 -20 -30 -36 -60 bw (Hz) 50 100 120 150 200
alto <quote>o</quote> Values f1 f2 f3 f4 f5 freq (Hz) 450 800 2830 3500 4950 amp (dB) 0 -9 -16 -28 -55 bw (Hz) 70 80 100 130 135
alto <quote>u</quote> Values f1 f2 f3 f4 f5 freq (Hz) 325 700 2530 3500 4950 amp (dB) 0 -12 -30 -40 -64 bw (Hz) 50 60 170 180 200
bass <quote>a</quote> Values f1 f2 f3 f4 f5 freq (Hz) 600 1040 2250 2450 2750 amp (dB) 0 -7 -9 -9 -20 bw (Hz) 60 70 110 120 130
bass <quote>e</quote> Values f1 f2 f3 f4 f5 freq (Hz) 400 1620 2400 2800 3100 amp (dB) 0 -12 -9 -12 -18 bw (Hz) 40 80 100 120 120
bass <quote>i</quote> Values f1 f2 f3 f4 f5 freq (Hz) 250 1750 2600 3050 3340 amp (dB) 0 -30 -16 -22 -28 bw (Hz) 60 90 100 120 120
bass <quote>o</quote> Values f1 f2 f3 f4 f5 freq (Hz) 400 750 2400 2600 2900 amp (dB) 0 -11 -21 -20 -40 bw (Hz) 40 80 100 120 120
bass <quote>u</quote> Values f1 f2 f3 f4 f5 freq (Hz) 350 600 2400 2675 2950 amp (dB) 0 -20 -32 -28 -36 bw (Hz) 40 80 100 120 120
countertenor <quote>a</quote> Values f1 f2 f3 f4 f5 freq (Hz) 660 1120 2750 3000 3350 amp (dB) 0 -6 -23 -24 -38 bw (Hz) 80 90 120 130 140
countertenor <quote>e</quote> Values f1 f2 f3 f4 f5 freq (Hz) 440 1800 2700 3000 3300 amp (dB) 0 -14 -18 -20 -20 bw (Hz) 70 80 100 120 120
countertenor <quote>i</quote> Values f1 f2 f3 f4 f5 freq (Hz) 270 1850 2900 3350 3590 amp (dB) 0 -24 -24 -36 -36 bw (Hz) 40 90 100 120 120
countertenor <quote>o</quote> Values f1 f2 f3 f4 f5 freq (Hz) 430 820 2700 3000 3300 amp (dB) 0 -10 -26 -22 -34 bw (Hz) 40 80 100 120 120
countertenor <quote>u</quote> Values f1 f2 f3 f4 f5 freq (Hz) 370 630 2750 3000 3400 amp (dB) 0 -20 -23 -30 -34 bw (Hz) 40 60 100 120 120
soprano <quote>a</quote> Values f1 f2 f3 f4 f5 freq (Hz) 800 1150 2900 3900 4950 amp (dB) 0 -6 -32 -20 -50 bw (Hz) 80 90 120 130 140
soprano <quote>e</quote> Values f1 f2 f3 f4 f5 freq (Hz) 350 2000 2800 3600 4950 amp (dB) 0 -20 -15 -40 -56 bw (Hz) 60 100 120 150 200
soprano <quote>i</quote> Values f1 f2 f3 f4 f5 freq (Hz) 270 2140 2950 3900 4950 amp (dB) 0 -12 -26 -26 -44 bw (Hz) 60 90 100 120 120
soprano <quote>o</quote> Values f1 f2 f3 f4 f5 freq (Hz) 450 800 2830 3800 4950 amp (dB) 0 -11 -22 -22 -50 bw (Hz) 40 80 100 120 120
soprano <quote>u</quote> Values f1 f2 f3 f4 f5 freq (Hz) 325 700 2700 3800 4950 amp (dB) 0 -16 -35 -40 -60 bw (Hz) 50 60 170 180 200
tenor <quote>a</quote> Values f1 f2 f3 f4 f5 freq (Hz) 650 1080 2650 2900 3250 amp (dB) 0 -6 -7 -8 -22 bw (Hz) 80 90 120 130 140
tenor <quote>e</quote> Values f1 f2 f3 f4 f5 freq (Hz) 400 1700 2600 3200 3580 amp (dB) 0 -14 -12 -14 -20 bw (Hz) 70 80 100 120 120
tenor <quote>i</quote> Values f1 f2 f3 f4 f5 freq (Hz) 290 1870 2800 3250 3540 amp (dB) 0 -15 -18 -20 -30 bw (Hz) 40 90 100 120 120
tenor <quote>o</quote> Values f1 f2 f3 f4 f5 freq (Hz) 400 800 2600 2800 3000 amp (dB) 0 -10 -12 -12 -26 bw (Hz) 70 80 100 130 135
tenor <quote>u</quote> Values f1 f2 f3 f4 f5 freq (Hz) 350 600 2700 2900 3300 amp (dB) 0 -20 -17 -14 -26 bw (Hz) 40 60 100 120 120
manual_src~dfsg/misc/quickref.xml0000644000000000000000000056601112262561504016244 0ustar rootroot Opcode Quick Reference Opcode Quick Reference Orchestra Syntax:Header 0dbfs = iarg0dbfs kr = iarg ksmps = iarg nchnls = iarg nchnls_i = iarg sr = iarg Orchestra Syntax:Block Statements endin  endop  instr i, j, ... opcode name, outtypes, intypes Orchestra Syntax:Macros #define NAME # replacement text # #define NAME(a' b' c') # replacement text # $NAME  #ifdef NAME ....#else  ....#end  #ifndef NAME ....#else  ....#end  #include "filename" #undef NAME Signal Generators:Additive Synthesis/Resynthesis ares adsyn kamod, kfmod, ksmod, ifilcod ares adsynt kamp, kcps, iwfn, ifreqfn, iampfn, icnt [, iphs] ar adsynt2 kamp, kcps, iwfn, ifreqfn, iampfn, icnt [, iphs] ares hsboscil kamp, ktone, kbrite, ibasfreq, iwfn, ioctfn \ [, ioctcnt] [, iphs] Signal Generators:Basic Oscillators kres lfo kamp, kcps [, itype]ares lfo kamp, kcps [, itype] ares oscbnk kcps, kamd, kfmd, kpmd, iovrlap, iseed, kl1minf, kl1maxf, \ kl2minf, kl2maxf, ilfomode, keqminf, keqmaxf, keqminl, keqmaxl, \ keqminq, keqmaxq, ieqmode, kfn [, il1fn] [, il2fn] [, ieqffn] \ [, ieqlfn] [, ieqqfn] [, itabl] [, ioutfn] ares oscil xamp, xcps [, ifn, iphs]kres oscil kamp, kcps [, ifn, iphs] ares oscil3 xamp, xcps [, ifn, iphs]kres oscil3 kamp, kcps [, ifn, iphs] ares oscili xamp, xcps, ifn [, iphs]kres oscili kamp, kcps, ifn [, iphs] ares oscilikt xamp, xcps, kfn [, iphs] [, istor]kres oscilikt kamp, kcps, kfn [, iphs] [, istor] ares osciliktp kcps, kfn, kphs [, istor] ares oscilikts xamp, xcps, kfn, async, kphs [, istor] ares osciln kamp, ifrq, ifn, itimes ares oscils iamp, icps, iphs [, iflg] ares poscil aamp, acps [, ifn, iphs]ares poscil aamp, kcps [, ifn, iphs]ares poscil kamp, acps [, ifn, iphs]ares poscil kamp, kcps [, ifn, iphs]ires poscil kamp, kcps [, ifn, iphs]kres poscil kamp, kcps [, ifn, iphs] ares poscil3 aamp, acps [, ifn, iphs]ares poscil3 aamp, kcps [, ifn, iphs]ares poscil3 kamp, acps [, ifn, iphs]ares poscil3 kamp, kcps [, ifn, iphs]ires poscil3 kamp, kcps [, ifn, iphs]kres poscil3 kamp, kcps [, ifn, iphs] kout vibr kAverageAmp, kAverageFreq, ifn kout vibrato kAverageAmp, kAverageFreq, kRandAmountAmp, kRandAmountFreq, kAmpMinRate, kAmpMaxRate, kcpsMinRate, kcpsMaxRate, ifn [, iphs Signal Generators:Dynamic Spectrum Oscillators ares buzz xamp, xcps, knh, ifn [, iphs] ares gbuzz xamp, xcps, knh, klh, kmul, ifn [, iphs] ares mpulse kamp, kintvl [, ioffset] ares vco xamp, xcps, iwave, kpw [, ifn] [, imaxd] [, ileak] [, inyx] \ [, iphs] [, iskip] ares vco2 kamp, kcps [, imode] [, kpw] [, kphs] [, inyx] kfn vco2ft kcps, iwave [, inyx] ifn vco2ift icps, iwave [, inyx] ifn vco2init iwave [, ibasfn] [, ipmul] [, iminsiz] [, imaxsiz] [, isrcft] Signal Generators:FM Synthesis a1, a2 crossfm xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2]a1, a2 crossfmi xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2]a1, a2 crosspm xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2]a1, a2 crosspmi xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2]a1, a2 crossfmpm xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2]a1, a2 crossfmpmi xfrq1, xfrq2, xndx1, xndx2, kcps, ifn1, ifn2 [, iphs1] [, iphs2] ares fmb3 kamp, kfreq, kc1, kc2, kvdepth, kvrate[, ifn1, ifn2, ifn3, \ ifn4, ivfn] ares fmbell kamp, kfreq, kc1, kc2, kvdepth, kvrate[, ifn1, ifn2, ifn3, \ ifn4, ivfn, isus] ares fmmetal kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, \ ifn4, ivfn ares fmpercfl kamp, kfreq, kc1, kc2, kvdepth, kvrate[, ifn1, ifn2, \ ifn3, ifn4, ivfn] ares fmrhode kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, \ ifn3, ifn4, ivfn ares fmvoice kamp, kfreq, kvowel, ktilt, kvibamt, kvibrate[, ifn1, \ ifn2, ifn3, ifn4, ivibfn] ares fmwurlie kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, \ ifn4, ivfn ares foscil xamp, kcps, xcar, xmod, kndx, ifn [, iphs] ares foscili xamp, kcps, xcar, xmod, kndx, ifn [, iphs] Signal Generators:Granular Synthesis asig diskgrain Sfname, kamp, kfreq, kpitch, kgrsize, kprate, \ ifun, iolaps [,imaxgrsize , ioffset] ares fof xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, \ ifna, ifnb, itotdur [, iphs] [, ifmode] [, iskip] ares fof2 xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, \ ifna, ifnb, itotdur, kphs, kgliss [, iskip] ares fog xamp, xdens, xtrans, aspd, koct, kband, kris, kdur, kdec, \ iolaps, ifna, ifnb, itotdur [, iphs] [, itmode] [, iskip] ares grain xamp, xpitch, xdens, kampoff, kpitchoff, kgdur, igfn, \ iwfn, imgdur [, igrnd] ares grain2 kcps, kfmd, kgdur, iovrlp, kfn, iwfn [, irpow] \ [, iseed] [, imode] ares grain3 kcps, kphs, kfmd, kpmd, kgdur, kdens, imaxovr, kfn, iwfn, \ kfrpow, kprpow [, iseed] [, imode] ares granule xamp, ivoice, iratio, imode, ithd, ifn, ipshift, igskip, \ igskip_os, ilength, kgap, igap_os, kgsize, igsize_os, iatt, idec \ [, iseed] [, ipitch1] [, ipitch2] [, ipitch3] [, ipitch4] [, ifnenv] a1 [, a2, a3, a4, a5, a6, a7, a8] partikkel agrainfreq, \ kdistribution, idisttab, async, kenv2amt, ienv2tab, ienv_attack, \ ienv_decay, ksustain_amount, ka_d_ratio, kduration, kamp, igainmasks, \ kwavfreq, ksweepshape, iwavfreqstarttab, iwavfreqendtab, awavfm, \ ifmamptab, kfmenv, icosine, ktraincps, knumpartials, kchroma, \ ichannelmasks, krandommask, kwaveform1, kwaveform2, kwaveform3, \ kwaveform4, iwaveamptab, asamplepos1, asamplepos2, asamplepos3, \ asamplepos4, kwavekey1, kwavekey2, kwavekey3, kwavekey4, imax_grains \ [, iopcode_id] async [,aphase] partikkelsync iopcode_id ares [, ac] sndwarp xamp, xtimewarp, xresample, ifn1, ibeg, iwsize, \ irandw, ioverlap, ifn2, itimemode ar1, ar2 [,ac1] [, ac2] sndwarpst xamp, xtimewarp, xresample, ifn1, \ ibeg, iwsize, irandw, ioverlap, ifn2, itimemode asig syncgrain kamp, kfreq, kpitch, kgrsize, kprate, ifun1, \ ifun2, iolaps asig syncloop kamp, kfreq, kpitch, kgrsize, kprate, klstart, \ klend, ifun1, ifun2, iolaps[,istart, iskip] ar vosim kamp, kFund, kForm, kDecay, kPulseCount, kPulseFactor, ifn [, iskip] Signal Generators:Hyper Vectorial Synthesis hvs1 kx, inumParms, inumPointsX, iOutTab, iPositionsTab, iSnapTab [, iConfigTab] hvs2 kx, ky, inumParms, inumPointsX, inumPointsY, iOutTab, iPositionsTab, iSnapTab [, iConfigTab] hvs3 kx, ky, kz, inumParms, inumPointsX, inumPointsY, inumPointsZ, iOutTab, iPositionsTab, iSnapTab [, iConfigTab] Signal Generators:Linear and Exponential Generators ares cosseg ia, idur1, ib [, idur2] [, ic] [...]kres cosseg ia, idur1, ib [, idur2] [, ic] [...] ares cossegb ia, itim1, ib [, itim2] [, ic] [...]kres cossegb ia, itim1, ib [, itim2] [, ic] [...] ares cossegr ia, idur1, ib [, idur2] [, ic] [...], irel, izkres cossegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz kout expcurve kindex, ksteepness ares expon ia, idur, ibkres expon ia, idur, ib ares expseg ia, idur1, ib [, idur2] [, ic] [...]kres expseg ia, idur1, ib [, idur2] [, ic] [...] ares expsega ia, idur1, ib [, idur2] [, ic] [...] ares expsegb ia, itim1, ib [, itim2] [, ic] [...]kres expsegb ia, itim1, ib [, itim2] [, ic] [...] ares expsegba ia, itim1, ib [, itim2] [, ic] [...] ares expsegr ia, idur1, ib [, idur2] [, ic] [...], irel, izkres expsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz kout gainslider kindex ares jspline xamp, kcpsMin, kcpsMaxkres jspline kamp, kcpsMin, kcpsMax ares line ia, idur, ibkres line ia, idur, ib ares linseg ia, idur1, ib [, idur2] [, ic] [...]kres linseg ia, idur1, ib [, idur2] [, ic] [...] ares linsegb ia, itim1, ib [, itim2] [, ic] [...]kres linsegb ia, itim1, ib [, itim2] [, ic] [...] ares linsegr ia, idur1, ib [, idur2] [, ic] [...], irel, izkres linsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz kout logcurve kindex, ksteepness ksig loopseg kfreq, ktrig, iphase, kvalue0, ktime0 [, kvalue1] [, ktime1] \ [, kvalue2] [, ktime2][...] ksig loopsegp kphase, kvalue0, kdur0, kvalue1 \ [, kdur1, ... , kdurN-1, kvalueN] ksig looptseg kfreq, ktrig, iphase, kvalue0, ktype0, ktime0, [, kvalue1] [,ktype1] [, ktime1] \ [, kvalue2] [,ktype2] [, ktime2] [...] [, kvalueN] [,ktypeN] [, ktimeN] ksig loopxseg kfreq, ktrig, iphase, ktime0, kvalue0 [, ktime1] [, kvalue1] \ [, ktime2] [, kvalue2] [...] ksig lpshold kfreq, ktrig, iphase, ktime0, kvalue0 [, kvalue1] [, ktime1] [, kvalue2] [, ktime2] [...] ksig lpsholdp kphase, kvalue0, ktime0 [, kvalue1] [, ktime1] \ [, kvalue2] [, ktime2] [...] ares rspline xrangeMin, xrangeMax, kcpsMin, kcpsMaxkres rspline krangeMin, krangeMax, kcpsMin, kcpsMax kscl scale kinput, kmax, kmin ares transeg ia, idur, itype, ib [, idur2] [, itype] [, ic] ...kres transeg ia, idur, itype, ib [, idur2] [, itype] [, ic] ... ares transegb ia, itim, itype, ib [, itim2] [, itype] [, ic] ...kres transegb ia, itim, itype, ib [, itim2] [, itype] [, ic] ... ares transegr ia, idur, itype, ib [, idur2] [, itype] [, ic] ...kres transegr ia, idur, itype, ib [, idur2] [, itype] [, ic] ... Signal Generators:Envelope Generators ares adsr iatt, idec, islev, irel [, idel]kres adsr iatt, idec, islev, irel [, idel] ares envlpx xamp, irise, idur, idec, ifn, iatss, iatdec [, ixmod]kres envlpx kamp, irise, idur, idec, ifn, iatss, iatdec [, ixmod] ares envlpxr xamp, irise, idec, ifn, iatss, iatdec [, ixmod] [,irind]kres envlpxr kamp, irise, idec, ifn, iatss, iatdec [, ixmod] [,irind] ares linen xamp, irise, idur, ideckres linen kamp, irise, idur, idec ares linenr xamp, irise, idec, iatdeckres linenr kamp, irise, idec, iatdec ares madsr iatt, idec, islev, irel [, idel] [, ireltim]kres madsr iatt, idec, islev, irel [, idel] [, ireltim] ares mxadsr iatt, idec, islev, irel [, idel] [, ireltim]kres mxadsr iatt, idec, islev, irel [, idel] [, ireltim] ares xadsr iatt, idec, islev, irel [, idel]kres xadsr iatt, idec, islev, irel [, idel] Signal Generators:Models and Emulations ares bamboo kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ [, ifreq1] [, ifreq2] ares barmodel kbcL, kbcR, iK, ib, kscan, iT30, ipos, ivel, iwid ares cabasa iamp, idettack [, inum] [, idamp] [, imaxshake] aI3, aV2, aV1 chuap kL, kR0, kC1, kG, kGa, kGb, kE, kC2, iI3, iV2, iV1, ktime_step ares crunch iamp, idettack [, inum] [, idamp] [, imaxshake] ares dripwater kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ [, ifreq1] [, ifreq2] ares gendy kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl [, initcps] [, knum]kres gendy kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl [, initcps] [, knum] ares gendyc kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl [, initcps] [, knum]kres gendyc kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl [, initcps] [, knum] ares gendyx kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl, kcurveup, kcurvedown [, initcps] [, knum]kres gendyx kamp, kampdist, kdurdist, kadpar, kddpar, kminfreq, kmaxfreq, \ kampscl, kdurscl, kcurveup, kcurvedown [, initcps] [, knum] ares gogobel kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivfn ares guiro kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] [, ifreq1] ax, ay, az lorenz ksv, krv, kbv, kh, ix, iy, iz, iskip [, iskipinit] kiter, koutrig mandel ktrig, kx, ky, kmaxIter ares mandol kamp, kfreq, kpluck, kdetune, kgain, ksize, ifn [, iminfreq] ares marimba kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec \ [, idoubles] [, itriples] ares moog kamp, kfreq, kfiltq, kfiltrate, kvibf, kvamp, iafn, iwfn, ivfn ax, ay, az planet kmass1, kmass2, ksep, ix, iy, iz, ivx, ivy, ivz, idelta \ [, ifriction] [, iskip] ares prepiano ifreq, iNS, iD, iK, \ iT30,iB, kbcl, kbcr, imass, ifreq, iinit, ipos, ivel, isfreq, \ isspread[, irattles, irubbers]al,ar prepiano ifreq, iNS, iD, iK, \ iT30,iB, kbcl, kbcr, imass, ifreq, iinit, ipos, ivel, isfreq, \ isspread[, irattles, irubbers] ares sandpaper iamp, idettack [, inum] [, idamp] [, imaxshake] ares sekere iamp, idettack [, inum] [, idamp] [, imaxshake] ares shaker kamp, kfreq, kbeans, kdamp, ktimes [, idecay] ares sleighbells kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ [, ifreq1] [, ifreq2] ares stix iamp, idettack [, inum] [, idamp] [, imaxshake] ares tambourine kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] \ [, ifreq1] [, ifreq2] ares vibes kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec ares voice kamp, kfreq, kphoneme, kform, kvibf, kvamp, ifn, ivfn Signal Generators:Phasors ares phasor xcps [, iphs]kres phasor kcps [, iphs] ares phasorbnk xcps, kndx, icnt [, iphs]kres phasorbnk kcps, kndx, icnt [, iphs] aphase, asyncout syncphasor xcps, asyncin, [, iphs] Signal Generators:Random (Noise) Generators ares betarand krange, kalpha, kbetaires betarand krange, kalpha, kbetakres betarand krange, kalpha, kbeta ares bexprnd krangeires bexprnd krangekres bexprnd krange ares cauchy kalphaires cauchy kalphakres cauchy kalpha ares cauchyi klambda, xamp, xcpsires cauchyi klambda, xamp, xcpskres cauchyi klambda, xamp, xcps aout cuserrnd kmin, kmax, ktableNumiout cuserrnd imin, imax, itableNumkout cuserrnd kmin, kmax, ktableNum aout duserrnd ktableNumiout duserrnd itableNumkout duserrnd ktableNum ares dust kamp, kdensitykres dust kamp, kdensity ares dust2 kamp, kdensitykres dust2 kamp, kdensity ares exprand klambdaires exprand klambdakres exprand klambda ares exprandi klambda, xamp, xcpsires exprandi klambda, xamp, xcpskres exprandi klambda, xamp, xcps ares fractalnoise kamp, kbeta ares gauss krangeires gauss krangekres gauss krange ares gaussi krange, xamp, xcpsires gaussi krange, xamp, xcpskres gaussi krange, xamp, xcps ares gausstrig kamp, kcps, kdev [, imode]kres gausstrig kamp, kcps, kdev [, imode] kout jitter kamp, kcpsMin, kcpsMax kout jitter2 ktotamp, kamp1, kcps1, kamp2, kcps2, kamp3, kcps3 ares linrand krangeires linrand krangekres linrand krange ares noise xamp, kbeta ares pcauchy kalphaires pcauchy kalphakres pcauchy kalpha ares pinkish xin [, imethod] [, inumbands] [, iseed] [, iskip] ares poisson klambdaires poisson klambdakres poisson klambda ares rand xamp [, iseed] [, isel] [, ioffset]kres rand xamp [, iseed] [, isel] [, ioffset] ares randh xamp, xcps [, iseed] [, isize] [, ioffset]kres randh kamp, kcps [, iseed] [, isize] [, ioffset] ares randi xamp, xcps [, iseed] [, isize] [, ioffset]kres randi kamp, kcps [, iseed] [, isize] [, ioffset] ares random kmin, kmaxires random imin, imaxkres random kmin, kmax ares randomh kmin, kmax, xcps [,imode] [,ifirstval]kres randomh kmin, kmax, kcps [,imode] [,ifirstval] ares randomi kmin, kmax, xcps [,imode] [,ifirstval]kres randomi kmin, kmax, kcps [,imode] [,ifirstval] ax rnd31 kscl, krpow [, iseed]ix rnd31 iscl, irpow [, iseed]kx rnd31 kscl, krpow [, iseed] seed ival kout trandom ktrig, kmin, kmax ares trirand krangeires trirand krangekres trirand krange ares unirand krangeires unirand krangekres unirand krange ax urandom [imin, imax]ix urandom [imin, imax]kx urandom [imin, imax] aout = urd(ktableNum)iout = urd(itableNum)kout = urd(ktableNum) ares weibull ksigma, ktauires weibull ksigma, ktaukres weibull ksigma, ktau Signal Generators:Sample Playback a1 bbcutm asource, ibps, isubdiv, ibarlength, iphrasebars, inumrepeats \ [, istutterspeed] [, istutterchance] [, ienvchoice ] a1,a2 bbcuts asource1, asource2, ibps, isubdiv, ibarlength, iphrasebars, \ inumrepeats [, istutterspeed] [, istutterchance] [, ienvchoice] asig flooper kamp, kpitch, istart, idur, ifad, ifn asig flooper2 kamp, kpitch, kloopstart, kloopend, kcrossfade, ifn \ [, istart, imode, ifenv, iskip] aleft, aright fluidAllOut  fluidCCi iEngineNumber, iChannelNumber, iControllerNumber, iValue fluidCCk iEngineNumber, iChannelNumber, iControllerNumber, kValue fluidControl ienginenum, kstatus, kchannel, kdata1, kdata2 ienginenum fluidEngine [iReverbEnabled] [, iChorusEnabled] [,iNumChannels] [, iPolyphony] isfnum fluidLoad soundfont, ienginenum[, ilistpresets] fluidNote ienginenum, ichannelnum, imidikey, imidivel aleft, aright fluidOut ienginenum fluidProgramSelect ienginenum, ichannelnum, isfnum, ibanknum, ipresetnum fluidSetInterpMethod ienginenum, ichannelnum, iInterpMethod ar1 [,ar2] loscil xamp, kcps, ifn [, ibas] [, imod1] [, ibeg1] [, iend1] \ [, imod2] [, ibeg2] [, iend2] ar1 [,ar2] loscil3 xamp, kcps, ifn [, ibas] [, imod1] [, ibeg1] [, iend1] \ [, imod2] [, ibeg2] [, iend2] ar1 [, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, ar13, ar14, \ ar15, ar16] loscilx xamp, kcps, ifn \ [, iwsize, ibas, istrt, imod1, ibeg1, iend1] ares lphasor xtrns [, ilps] [, ilpe] [, imode] [, istrt] [, istor] ares lposcil kamp, kfreqratio, kloop, kend, ifn [, iphs] ares lposcil3 kamp, kfreqratio, kloop, kend, ifn [, iphs] ar lposcila aamp, kfreqratio, kloop, kend, ift [,iphs] ar1, ar2 lposcilsa aamp, kfreqratio, kloop, kend, ift [,iphs] ar1, ar2 lposcilsa2 aamp, kfreqratio, kloop, kend, ift [,iphs] sfilist ifilhandle ar1, ar2 sfinstr ivel, inotenum, xamp, xfreq, instrnum, ifilhandle \ [, iflag] [, ioffset] ar1, ar2 sfinstr3 ivel, inotenum, xamp, xfreq, instrnum, ifilhandle \ [, iflag] [, ioffset] ares sfinstr3m ivel, inotenum, xamp, xfreq, instrnum, ifilhandle \ [, iflag] [, ioffset] ares sfinstrm ivel, inotenum, xamp, xfreq, instrnum, ifilhandle \ [, iflag] [, ioffset] ir sfload "filename" ar1, ar2 sflooper ivel, inotenum, kamp, kpitch, ipreindex, kloopstart, kloopend, kcrossfade \ [, istart, imode, ifenv, iskip] sfpassign istartindex, ifilhandle[, imsgs] ar1, ar2 sfplay ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset] [, ienv] ar1, ar2 sfplay3 ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset] [, ienv] ares sfplay3m ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset] [, ienv] ares sfplaym ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset] [, ienv] sfplist ifilhandle ir sfpreset iprog, ibank, ifilhandle, ipreindex asig, krec sndloop ain, kpitch, ktrig, idur, ifad ares waveset ain, krep [, ilen] Signal Generators:Scanned Synthesis scanhammer isrc, idst, ipos, imult ares scans kamp, kfreq, ifn, id [, iorder] aout scantable kamp, kpch, ipos, imass, istiff, idamp, ivel scanu init, irate, ifnvel, ifnmass, ifnstif, ifncentr, ifndamp, kmass, \ kstif, kcentr, kdamp, ileft, iright, kpos, kstrngth, ain, idisp, id kpos, kvel xscanmap iscan, kamp, kvamp [, iwhich] ares xscans kamp, kfreq, ifntraj, id [, iorder] xscansmap kpos, kvel, iscan, kamp, kvamp [, iwhich] xscanu init, irate, ifnvel, ifnmass, ifnstif, ifncentr, ifndamp, kmass, \ kstif, kcentr, kdamp, ileft, iright, kpos, kstrngth, ain, idisp, id Signal Generators:STK Opcodes asignal STKBandedWG ifrequency, iamplitude, [kpress, kv1[, kmot, kv2[, klfo, kv3[, klfodepth, kv4[, kvel, kv5[, kstrk, kv6[, kinstr, kv7]]]]]]] asignal STKBeeThree ifrequency, iamplitude, [kop4, kv1[, kop3, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] asignal STKBlowBotl ifrequency, iamplitude, [knoise, kv1[, klfo, kv2[, klfodepth, kv3[, kvol, kv4]]]] asignal STKBlowHole ifrequency, iamplitude, [kreed, kv1[, knoise, kv2[, khole, kv3[, kreg, kv4[, kbreath, kv5]]]]] asignal STKBowed ifrequency, iamplitude, [kpress, kv1[, kpos, kv2[, klfo, kv3[, klfodepth, kv4[, kvol, kv5]]]]] asignal STKBrass ifrequency, iamplitude, [klip, kv1[, kslide, kv2[, klfo, kv3[, klfodepth, kv4[, kvol, kv5]]]]] asignal STKClarinet ifrequency, iamplitude, [kstiff, kv1[, knoise, kv2[, klfo, kv3[, klfodepth, kv4[, kbreath, kv5]]]]] asignal STKDrummer ifrequency, iamplitude asignal STKFMVoices ifrequency, iamplitude, [kvowel, kv1[, kspec, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] asignal STKFlute ifrequency, iamplitude, [kjet, kv1[, knoise, kv2[, klfo, kv3[, klfodepth, kv4[, kbreath, kv5]]]]] asignal STKHevyMetl ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] asignal STKMandolin ifrequency, iamplitude, [kbody, kv1[, kpos, kv2[, ksus, kv3[, kdetune, kv4[, kmic, kv5]]]]] asignal STKModalBar ifrequency, iamplitude, [khard, kv1[, kpos, kv2[, klfo, kv3[, klfodepth, kv4[, kmix, kv5[, kvol, kv6[, kinstr, kv7]]]]]]] asignal STKMoog ifrequency, iamplitude, [kq, kv1[, krate, kv2[, klfo, kv3[, klfodepth, kv4[, kvol, kv5]]]]] asignal STKPercFlut ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] asignal STKPlucked ifrequency, iamplitude asignal STKResonate ifrequency, iamplitude, [kfreq, kv1[, kpole, kv2[, knotch, kv3[, kzero, kv4[, kenv, kv5]]]]] asignal STKRhodey ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] asignal STKSaxofony ifrequency, iamplitude, [kstiff, kv1[, kapert, kv2[, kblow, kv3[, knoise, kv4[, klfo, kv5[, klfodepth, kv6[, kbreath, kv7]]]]]]] asignal STKShakers ifrequency, iamplitude, [kenerg, kv1[, kdecay, kv2[, kshake, kv3[, knum, kv4[, kres, kv5[, kinstr, kv6]]]]]] asignal STKSimple ifrequency, iamplitude, [kpos, kv1[, kcross, kv2[, kenv, kv3[, kgain, kv4]]]] asignal STKSitar ifrequency, iamplitude asignal STKStifKarp ifrequency, iamplitude, [kpos, kv1[, ksus, kv2[, kstretch, kv3]]] asignal STKTubeBell ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] asignal STKVoicForm ifrequency, iamplitude, [kmix, kv1[, ksel, kv2[, klfo, kv3[, klfodepth, kv4[, kloud, kv5]]]]] asignal STKWhistle ifrequency, iamplitude, [kmod, kv1[, knoise, kv2[, kfipfreq, kv3[, kfipgain, kv4[, kvol, kv5]]]]] asignal STKWurley ifrequency, iamplitude, [kmod, kv1[, kcross, kv2[, klfo, kv3[, klfodepth, kv4[, kadsr, kv5]]]]] Signal Generators:Table Access kres oscil1 idel, kamp, idur [, ifn] kres oscil1i idel, kamp, idur [, ifn] ares ptable andx, ifn [, ixmode] [, ixoff] [, iwrap]ires ptable indx, ifn [, ixmode] [, ixoff] [, iwrap]kres ptable kndx, ifn [, ixmode] [, ixoff] [, iwrap] ares ptable3 andx, ifn [, ixmode] [, ixoff] [, iwrap]ires ptable3 indx, ifn [, ixmode] [, ixoff] [, iwrap]kres ptable3 kndx, ifn [, ixmode] [, ixoff] [, iwrap] ares ptablei andx, ifn [, ixmode] [, ixoff] [, iwrap]ires ptablei indx, ifn [, ixmode] [, ixoff] [, iwrap]kres ptablei kndx, ifn [, ixmode] [, ixoff] [, iwrap] ir tab_i indx, ifn[, ixmode]kr tab kndx, ifn[, ixmode]ar tab xndx, ifn[, ixmode]tabw_i isig, indx, ifn [,ixmode]tabw ksig, kndx, ifn [,ixmode]tabw asig, andx, ifn [,ixmode] ares table andx, ifn [, ixmode] [, ixoff] [, iwrap]ires table indx, ifn [, ixmode] [, ixoff] [, iwrap]kres table kndx, ifn [, ixmode] [, ixoff] [, iwrap] ares table3 andx, ifn [, ixmode] [, ixoff] [, iwrap]ires table3 indx, ifn [, ixmode] [, ixoff] [, iwrap]kres table3 kndx, ifn [, ixmode] [, ixoff] [, iwrap] ares tablei andx, ifn [, ixmode] [, ixoff] [, iwrap]ires tablei indx, ifn [, ixmode] [, ixoff] [, iwrap]kres tablei kndx, ifn [, ixmode] [, ixoff] [, iwrap] Signal Generators:Wave Terrain Synthesis aout wterrain kamp, kpch, k_xcenter, k_ycenter, k_xradius, k_yradius, \ itabx, itaby Signal Generators:Waveguide Physical Modeling ares pluck kamp, kcps, icps, ifn, imeth [, iparm1] [, iparm2] ares repluck iplk, kamp, icps, kpick, krefl, axcite ares streson asig, kfr, ifdbgain ares wgbow kamp, kfreq, kpres, krat, kvibf, kvamp, ifn [, iminfreq] ares wgbowedbar kamp, kfreq, kpos, kbowpres, kgain [, iconst] [, itvel] \ [, ibowpos] [, ilow] ares wgbrass kamp, kfreq, ktens, iatt, kvibf, kvamp, ifn [, iminfreq] ares wgclar kamp, kfreq, kstiff, iatt, idetk, kngain, kvibf, kvamp, ifn \ [, iminfreq] ares wgflute kamp, kfreq, kjet, iatt, idetk, kngain, kvibf, kvamp, ifn \ [, iminfreq] [, ijetrf] [, iendrf] ares wgpluck icps, iamp, kpick, iplk, idamp, ifilt, axcite ares wgpluck2 iplk, kamp, icps, kpick, krefl Signal I/O:File I/O dumpk ksig, ifilname, iformat, iprd dumpk2 ksig1, ksig2, ifilname, iformat, iprd dumpk3 ksig1, ksig2, ksig3, ifilname, iformat, iprd dumpk4 ksig1, ksig2, ksig3, ksig4, ifilname, iformat, iprd ficlose ihandleficlose Sfilename fin ifilename, iskipframes, iformat, ain1 [, ain2] [, ain3] [,...]fin ifilename, iskipframes, iformat, arr[] fini ifilename, iskipframes, iformat, in1 [, in2] [, in3] [, ...] fink ifilename, iskipframes, iformat, kin1 [, kin2] [, kin3] [,...] ihandle fiopen ifilename, imode fout ifilename, iformat, aout1 [, aout2, aout3,...,aoutN]fout ifilename, iformat, array[] fouti ihandle, iformat, iflag, iout1 [, iout2, iout3,....,ioutN] foutir ihandle, iformat, iflag, iout1 [, iout2, iout3,....,ioutN] foutk ifilename, iformat, kout1 [, kout2, kout3,....,koutN] fprintks "filename", "string", [, kval1] [, kval2] [...] fprints "filename", "string" [, ival1] [, ival2] [...] Sres, kline readf ifilname Sres, iline readfi ifilname kres readk ifilname, iformat, iprd kr1, kr2 readk2 ifilname, iformat, iprd kr1, kr2, kr3 readk3 ifilname, iformat, iprd kr1, kr2, kr3, kr4 readk4 ifilname, iformat, iprd Signal I/O:Signal Input ar1 [, ar2 [, ar3 [, ... arN]]] diskin ifilcod, kpitch [, iskiptim] \ [, iwraparound] [, iformat] [, iskipinit]ar1[] diskin2 ifilcod, kpitch[, iskiptim \ [, iwrap[, iformat [, iwsize[, ibufsize[, iskipinit]]]]]] a1[, a2[, ... aN]] diskin2 ifilcod, kpitch[, iskiptim \ [, iwrap[, iformat [, iwsize[, ibufsize[, iskipinit]]]]]]ar1[] diskin2 ifilcod, kpitch[, iskiptim \ [, iwrap[, iformat [, iwsize[, ibufsize[, iskipinit]]]]]] ar1 in aarray in  ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, ar13, ar14, \ ar15, ar16, ar17, ar18, ar19, ar20, ar21, ar22, ar23, ar24, ar25, ar26, \ ar27, ar28, ar29, ar30, ar31, ar32 in32  ain1[, ...] inch kchan1[,...] ar1, ar2, ar3, ar4, ar5, ar6 inh  ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8 ino  ar1, ar2, ar3, a4 inq  inrg kstart, ain1 [,ain2, ain3, ..., ainN] ar1, ar2 ins  kvalue invalue "channel name"Sname invalue "channel name" ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, \ ar13, ar14, ar15, ar16 inx  inz ksig1 ar1, ar2 mp3in ifilcod[, iskptim, iformat, iskipinit, ibufsize] ar1[, ar2[, ar3[, ... a24]]] soundin ifilcod [, iskptim] [, iformat] \ [, iskipinit] [, ibufsize] Signal I/O:Signal Output mdelay kstatus, kchan, kd1, kd2, kdelay aout1 [,aout2 ... aoutX] monitor  out asig1[, asig2,....]out aarray out32 asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8, asig10, \ asig11, asig12, asig13, asig14, asig15, asig16, asig17, asig18, \ asig19, asig20, asig21, asig22, asig23, asig24, asig25, asig26, \ asig27, asig28, asig29, asig30, asig31, asig32 outc asig1 [, asig2] [...] outch kchan1, asig1 [, kchan2] [, asig2] [...] outh asig1, asig2, asig3, asig4, asig5, asig6 outo asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8 outq asig1, asig2, asig3, asig4 outq1 asig outq2 asig outq3 asig outq4 asig outrg kstart, aout1 [,aout2, aout3, ..., aoutN] outs asig1, asig2 outs1 asig outs2 asig outvalue "channel name", kvalueoutvalue "channel name", "string" outx asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8, \ asig9, asig10, asig11, asig12, asig13, asig14, asig15, asig16 outz ksig1 soundout asig1, ifilcod [, iformat] soundouts asigl, asigr, ifilcod [, iformat] Signal I/O:Software Bus kval chani kchanaval chani kchan chano kval, kchanchano aval, kchan chn_k Sname, imode[, itype, idflt, imin, ima, ix, iy, iwidth, iheight, Sattributes] chn_a Sname, imode chn_S Sname, imode chnclear Sname gival chnexport Sname, imode[, itype, idflt, imin, imax]gkval chnexport Sname, imode[, itype, idflt, imin, imax]gaval chnexport Sname, imodegSval chnexport Sname, imode ival chnget Snamekval chnget Snameaval chnget SnameSval chnget Sname chnmix aval, Sname itype, imode, ictltype, idflt, imin, imax chnparams  ival chnrecv Snamekval chnrecv Snameaval chnrecv SnameSval chnrecv Sname chnsend ival, Snamechnsend kval, Snamechnsend aval, Snamechnsend Sval, Sname chnset ival, Snamechnset kval, Snamechnset aval, Snamechnset Sval, Sname setksmps iksmps xinarg1 [, xinarg2] ... [xinargN] xin  xout xoutarg1 [, xoutarg2] ... [, xoutargN] Signal I/O:Printing and Display dispfft xsig, iprd, iwsiz [, iwtyp] [, idbout] [, iwtflg] display xsig, iprd [, inprds] [, iwtflg] flashtxt iwhich, String print iarg [, iarg1] [, iarg2] [...] printf_i Sfmt, itrig, [iarg1[, iarg2[, ... ]]]printf Sfmt, ktrig, [xarg1[, xarg2[, ... ]]] printk itime, kval [, ispace] printk2 kvar [, inumspaces] printks "string", itime [, kval1] [, kval2] [...] prints "string" [, kval1] [, kval2] [...] Signal I/O:Soundfile Queries ir filebit ifilcod [, iallowraw] ir filelen ifilcod, [iallowraw] ir filenchnls ifilcod [, iallowraw] ir filepeak ifilcod [, ichnl] ir filesr ifilcod [, iallowraw] ir filevalid ifilcod ir mp3len ifilcod Signal Modifiers:Amplitude Modifiers ares balance asig, acomp [, ihp] [, iskip] ares clip asig, imeth, ilimit [, iarg] ar compress aasig, acsig, kthresh, kloknee, khiknee, kratio, katt, krel, ilook ares dam asig, kthreshold, icomp1, icomp2, irtime, iftime ares gain asig, krms [, ihp] [, iskip] Signal Modifiers:Convolution and Morphing ar1 [, ar2] [, ar3] [, ar4] convolve ain, ifilcod [, ichannel] ares cross2 ain1, ain2, isize, ioverlap, iwin, kbias ares dconv asig, isize, ifn a1[, a2[, a3[, ... a8]]] ftconv ain, ift, iplen[, iskipsamples \ [, iirlen[, iskipinit]]] ftmorf kftndx, iftfn, iresfn ar1 [, ar2] [, ar3] [, ar4] pconvolve ain, ifilcod [, ipartitionsize, ichannel] Signal Modifiers:Delay ares delay asig, idlt [, iskip] ares delay1 asig [, iskip] kr delayk ksig, idel[, imode]kr vdel_k ksig, kdel, imdel[, imode] ares delayr idlt [, iskip] delayw asig ares deltap kdlt ares deltap3 xdlt ares deltapi xdlt ares deltapn xnumsamps aout deltapx adel, iwsize deltapxw ain, adel, iwsize ares multitap asig [, itime1, igain1] [, itime2, igain2] [...] outarg1, outarg2 newopcodename inarg1, inarg2 ares vdelay asig, adel, imaxdel [, iskip] ares vdelay3 asig, adel, imaxdel [, iskip] aout vdelayx ain, adl, imd, iws [, ist] aout1, aout2, aout3, aout4 vdelayxq ain1, ain2, ain3, ain4, adl, imd, iws [, ist] aout1, aout2 vdelayxs ain1, ain2, adl, imd, iws [, ist] aout vdelayxw ain, adl, imd, iws [, ist] aout1, aout2, aout3, aout4 vdelayxwq ain1, ain2, ain3, ain4, adl, \ imd, iws [, ist] aout1, aout2 vdelayxws ain1, ain2, adl, imd, iws [, ist] Signal Modifiers:Panning and Spatialization ao1, ao2 bformdec isetup, aw, ax, ay, az [, ar, as, at, au, av \ [, abk, al, am, an, ao, ap, aq]]ao1, ao2, ao3, ao4 bformdec isetup, aw, ax, ay, az [, ar, as, at, \ au, av [, abk, al, am, an, ao, ap, aq]]ao1, ao2, ao3, ao4, ao5 bformdec isetup, aw, ax, ay, az [, ar, as, \ at, au, av [, abk, al, am, an, ao, ap, aq]]ao1, ao2, ao3, ao4, ao5, ao6, ao7, ao8 bformdec isetup, aw, ax, ay, az \ [, ar, as, at, au, av [, abk, al, am, an, ao, ap, aq]]] ao1, ao2 bformdec1 isetup, aw, ax, ay, az [, ar, as, at, au, av \ [, abk, al, am, an, ao, ap, aq]]ao1, ao2, ao3, ao4 bformdec1 isetup, aw, ax, ay, az [, ar, as, at, \ au, av [, abk, al, am, an, ao, ap, aq]]ao1, ao2, ao3, ao4, ao5 bformdec1 isetup, aw, ax, ay, az [, ar, as, \ at, au, av [, abk, al, am, an, ao, ap, aq]]ao1, ao2, ao3, ao4, ao5, ao6, ao7, ao8 bformdec1 isetup, aw, ax, ay, az \ [, ar, as, at, au, av [, abk, al, am, an, ao, ap, aq]]]aout[] bformdec1 isetup, abform[] aw, ax, ay, az bformenc asig, kalpha, kbeta, kord0, kord1aw, ax, ay, az, ar, as, at, au, av bformenc asig, kalpha, kbeta, \ kord0, kord1 , kord2aw, ax, ay, az, ar, as, at, au, av, ak, al, am, an, ao, ap, aq bformenc \ asig, kalpha, kbeta, kord0, kord1, kord2, kord3 aw, ax, ay, az bformenc1 asig, kalpha, kbetaaw, ax, ay, az, ar, as, at, au, av bformenc1 asig, kalpha, kbetaaw, ax, ay, az, ar, as, at, au, av, ak, al, am, an, ao, ap, aq bformenc1 \ asig, kalpha, kbetaaarray[] bformenc1 asig, kalpha, kbeta aleft, aright, irt60low, irt60high, imfp hrtfearly asrc, ksrcx, ksrcy, ksrcz, klstnrx, klstnry, klstnrz, \ ifilel, ifiler, idefroom [,ifade, isr, iorder, ithreed, kheadrot, iroomx, iroomy, iroomz, iwallhigh, \ iwalllow, iwallgain1, iwallgain2, iwallgain3, ifloorhigh, ifloorlow, ifloorgain1, ifloorgain2, \ ifloorgain3, iceilinghigh, iceilinglow, iceilinggain1, iceilinggain2, iceilinggain3] aleft, aright hrtfer asig, kaz, kelev, HRTFcompact aleft, aright hrtfmove asrc, kAz, kElev, ifilel, ifiler [, imode, ifade, isr] aleft, aright hrtfmove2 asrc, kAz, kElev, ifilel, ifiler [,ioverlap, iradius, isr] aleft, aright, idel hrtfreverb asrc, ilowrt60, ihighrt60, ifilel, ifiler [,isr, imfp, iorder] aleft, aright hrtfstat asrc, iAz, iElev, ifilel, ifiler [,iradius, isr] a1, a2 locsend a1, a2, a3, a4 locsend  a1, a2 locsig asig, kdegree, kdistance, kreverbsenda1, a2, a3, a4 locsig asig, kdegree, kdistance, kreverbsend a1, a2, a3, a4 pan asig, kx, ky, ifn [, imode] [, ioffset] a1, a2 pan2 asig, xp [, imode] a1, a2, a3, a4 space asig, ifn, ktime, kreverbsend, kx, ky aW, aX, aY, aZ spat3d ain, kX, kY, kZ, idist, ift, imode, imdel, iovr [, istor] aW, aX, aY, aZ spat3di ain, iX, iY, iZ, idist, ift, imode [, istor] spat3dt ioutft, iX, iY, iZ, idist, ift, imode, irlen [, iftnocl] k1 spdist ifn, ktime, kx, ky a1, a2, a3, a4 spsend  ar1[, ar2...] vbap asig, kazim [, kelev] [, kspread] [, ilayout]array[] vbap asig, kazim [, kelev] [, kspread] [, ilayout] ar1, ..., ar16 vbap16 asig, kazim [, kelev] [, kspread] ar1, ..., ar16 vbap16move asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] ar1, ar2, ar3, ar4 vbap4 asig, kazim [, kelev] [, kspread] ar1, ar2, ar3, ar4 vbap4move asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] ar1, ..., ar8 vbap8 asig, kazim [, kelev] [, kspread] ar1, ..., ar8 vbap8move asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] k1[, k2...] vbapg kazim [,kelev] [, kspread] [, ilayout]karray[] vbapg kazim [,kelev] [, kspread] [, ilayout] kr1[, kr2...] vbapgmove idur, ispread, ifldnum, ifld1 \ [, ifld2] [...]karray[] vbapgmove idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] vbaplsinit idim, ilsnum [, idir1] [, idir2] [...] [, idir32] ar1[, ar2...] vbapmove asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...]aarray[] vbapmove asig, idur, ispread, ifldnum, ifld1 \ [, ifld2] [...] vbapz inumchnls, istartndx, asig, kazim [, kelev] [, kspread] vbapzmove inumchnls, istartndx, asig, idur, ispread, ifldnum, ifld1, \ ifld2, [...] Signal Modifiers:Reverberation ares alpass asig, xrvt, ilpt [, iskip] [, insmps] a1, a2 babo asig, ksrcx, ksrcy, ksrcz, irx, iry, irz [, idiff] [, ifno] ares comb asig, krvt, ilpt [, iskip] [, insmps] ares combinv asig, krvt, ilpt [, iskip] [, insmps] aoutL, aoutR freeverb ainL, ainR, kRoomSize, kHFDamp[, iSRate[, iSkip]] ares nestedap asig, imode, imaxdel, idel1, igain1 [, idel2] [, igain2] \ [, idel3] [, igain3] [, istor] ares nreverb asig, ktime, khdif [, iskip] [,inumCombs] [, ifnCombs] \ [, inumAlpas] [, ifnAlpas] a1[, a2, ...] platerev itabexcite. itabouts, kbndry, iaspect, istiff, idecay, iloss, aexcite1[, aexcite2, ...] ares reverb asig, krvt [, iskip] ares reverb2 asig, ktime, khdif [, iskip] [,inumCombs] \ [, ifnCombs] [, inumAlpas] [, ifnAlpas] aoutL, aoutR reverbsc ainL, ainR, kfblvl, kfco[, israte[, ipitchm[, iskip]]] ares valpass asig, krvt, xlpt, imaxlpt [, iskip] [, insmps] ares vcomb asig, krvt, xlpt, imaxlpt [, iskip] [, insmps] Signal Modifiers:Sample Level Operators denorm a1[, a2[, a3[, ... ]]] ares diff asig [, iskip]kres diff ksig [, iskip] kres downsamp asig [, iwlen] ares fold asig, kincr ares integ asig [, iskip]kres integ ksig [, iskip] ares interp ksig [, iskip] [, imode] ares ntrpol asig1, asig2, kpoint [, imin] [, imax]ires ntrpol isig1, isig2, ipoint [, imin] [, imax]kres ntrpol ksig1, ksig2, kpoint [, imin] [, imax] a(x) (control-rate args only) i(x) (control-rate or init-rate arg) k(x) (i-rate args only)k(x) (a-rate args only) ares samphold asig, agate [, ival] [, ivstor]kres samphold ksig, kgate [, ival] [, ivstor] ares upsamp ksig kval vaget kndx, avar vaset kval, kndx, avar Signal Modifiers:Signal Limiters ares limit asig, klow, khighires limit isig, ilow, ihighkres limit ksig, klow, khigh ares mirror asig, klow, khighires mirror isig, ilow, ihighkres mirror ksig, klow, khigh ares wrap asig, klow, khighires wrap isig, ilow, ihighkres wrap ksig, klow, khigh Signal Modifiers:Special Effects ar distort asig, kdist, ifn[, ihp, istor] ares distort1 asig, kpregain, kpostgain, kshape1, kshape2[, imode] ares flanger asig, adel, kfeedback [, imaxd] ares harmon asig, kestfrq, kmaxvar, kgenfreq1, kgenfreq2, imode, \ iminfrq, iprd ares harmon2 asig, koct, kfrq1, kfrq2, icpsmode, ilowest[, ipolarity]ares harmon3 asig, koct, kfrq1, \ kfrq2, kfrq3, icpsmode, ilowest[, ipolarity]ares harmon4 asig, koct, kfrq1, \ kfrq2, kfrq3, kfrq4, icpsmode, ilowest[, ipolarity] ares phaser1 asig, kfreq, kord, kfeedback [, iskip] ares phaser2 asig, kfreq, kq, kord, kmode, ksep, kfeedback Signal Modifiers:Standard Filters ares atone asig, khp [, iskip] ares atonex asig, khp [, inumlayer] [, iskip]ares atonex asig, ahp [, inumlayer] [, iskip] ares biquad asig, kb0, kb1, kb2, ka0, ka1, ka2 [, iskip] ares biquada asig, ab0, ab1, ab2, aa0, aa1, aa2 [, iskip] ares butbp asig, kfreq, kband [, iskip] ares butbr asig, kfreq, kband [, iskip] ares buthp asig, kfreq [, iskip]ares buthp asig, afreq [, iskip] ares butlp asig, kfreq [, iskip]ares butlp asig, afreq [, iskip] ares butterbp asig, xfreq, xband [, iskip] ares butterbr asig, xfreq, xband [, iskip] ares butterhp asig, kfreq [, iskip]ares butterhp asig, afreq [, iskip] ares butterlp asig, kfreq [, iskip]ares butterlp asig, afreq [, iskip] ares clfilt asig, kfreq, itype, inpol [, ikind] [, ipbr] [, isba] [, iskip] ashifted doppler asource, ksourceposition, kmicposition [, isoundspeed, ifiltercutoff] ares median asig, ksize, imaxsize [, iskip] kres mediank kin, ksize, imaxsize [, iskip] aout mode ain, xfreq, xQ [, iskip] ares tone asig, khp [, iskip] ares tonex asig, khp [, inumlayer] [, iskip]ares tonex asig, ahp [, inumlayer] [, iskip] Signal Modifiers:Standard Filters:Resonant ares areson asig, kcf, kbw [, iscl] [, iskip]ares areson asig, acf, kbw [, iscl] [, iskip]ares areson asig, kcf, abw [, iscl] [, iskip]ares areson asig, acf, abw [, iscl] [, iskip] ares bqrez asig, xfco, xres [, imode] [, iskip] ares lowpass2 asig, kcf, kq [, iskip] ares lowres asig, kcutoff, kresonance [, iskip] ares lowresx asig, xcutoff, xresonance [, inumlayer] [, iskip] ares lpf18 asig, xfco, xres, xdist [, iskip] asig moogladder ain, kcf, kres[, istor]asig moogladder ain, acf, kres[, istor]asig moogladder ain, kcf, ares[, istor]asig moogladder ain, acf, ares[, istor] ares moogvcf asig, xfco, xres [,iscale, iskip] ares moogvcf2 asig, xfco, xres [,iscale, iskip] ares reson asig, xcf, xbw [, iscl] [, iskip] ares resonr asig, xcf, xbw [, iscl] [, iskip] ares resonx asig, xcf, xbw [, inumlayer] [, iscl] [, iskip] ares resony asig, kbf, kbw, inum, ksep [, isepmode] [, iscl] [, iskip] ares resonz asig, xcf, xbw [, iscl] [, iskip] ares rezzy asig, xfco, xres [, imode, iskip] ahp,alp,abp,abr statevar ain, xcf, xq [, iosamps, istor] alow, ahigh, aband svfilter asig, kcf, kq [, iscl] ares tbvcf asig, xfco, xres, kdist, kasym [, iskip] ares vlowres asig, kfco, kres, iord, ksep Signal Modifiers:Standard Filters:Control kres aresonk ksig, kcf, kbw [, iscl] [, iskip] kres atonek ksig, khp [, iskip] kres lineto ksig, ktime kres port ksig, ihtim [, isig] kres portk ksig, khtim [, isig] kres resonk ksig, kcf, kbw [, iscl] [, iskip] kres resonxk ksig, kcf, kbw[, inumlayer, iscl, istor] kres tlineto ksig, ktime, ktrig kres tonek ksig, khp [, iskip] Signal Modifiers:Specialized Filters ares dcblock ain [, igain] ares dcblock2 ain [, iorder] [, iskip] asig eqfil ain, kcf, kbw, kgain[, istor] ares filter2 asig, iM, iN, ib0, ib1, ..., ibM, ia1, ia2, ..., iaNkres filter2 ksig, iM, iN, ib0, ib1, ..., ibM, ia1, ia2, ..., iaN asig fofilter ain, xcf, xris, xdec[, istor] ar1, ar2 hilbert asig ares nlfilt ain, ka, kb, kd, kC, kL ares nlfilt2 ain, ka, kb, kd, kC, kL ares pareq asig, kc, kv, kq [, imode] [, iskip] ar rbjeq asig, kfco, klvl, kQ, kS[, imode] ares zfilter2 asig, kdamp, kfreq, iM, iN, ib0, ib1, ..., ibM, \ ia1,ia2, ..., iaN Signal Modifiers:Waveguides ares wguide1 asig, xfreq, kcutoff, kfeedback ares wguide2 asig, xfreq1, xfreq2, kcutoff1, kcutoff2, \ kfeedback1, kfeedback2 Signal Modifiers:Waveshaping aout chebyshevpoly ain, k0 [, k1 [, k2 [...]]] aout pdclip ain, kWidth, kCenter [, ibipolar [, ifullscale]] aout pdhalf ain, kShapeAmount [, ibipolar [, ifullscale]] aout pdhalfy ain, kShapeAmount [, ibipolar [, ifullscale]] aout powershape ain, kShapeAmount [, ifullscale] Signal Modifiers:Comparators and Accumulators amax max ain1, ain2 [, ain3] [, ain4] [...]kmax max kin1, kin2 [, kin3] [, kin4] [...] knumkout max_k asig, ktrig, itype amax maxabs ain1, ain2 [, ain3] [, ain4] [...]kmax maxabs kin1, kin2 [, kin3] [, kin4] [...] maxabsaccum aAccumulator, aInput maxaccum aAccumulator, aInput amin min ain1, ain2 [, ain3] [, ain4] [...]kmin min kin1, kin2 [, kin3] [, kin4] [...] amin minabs ain1, ain2 [, ain3] [, ain4] [...]kmin minabs kin1, kin2 [, kin3] [, kin4] [...] minabsaccum aAccumulator, aInput minaccum aAccumulator, aInput Instrument Control:Clock Control clockoff inum clockon inum Instrument Control:Conditional Values (a == b ? v1 : v2) (a >= b ? v1 : v2) (a > b ? v1 : v2) (a <= b ? v1 : v2) (a < b ? v1 : v2) (a != b ? v1 : v2) Instrument Control:Compilation ires compileorc Sfilename ires compilestr Sorch ires evalstr Scode kres evalstr Scode, ktrig return ival Instrument Control:Duration Control ihold  kill kinstance turnoff  turnoff2 kinsno, kmode, krelease turnon insnum [, itime] Instrument Control:Invocation event "scorechar", kinsnum, kdelay, kdur, [, kp4] [, kp5] [, ...]event "scorechar", "insname", kdelay, kdur, [, kp4] [, kp5] [, ...] event_i "scorechar", iinsnum, idelay, idur, [, ip4] [, ip5] [, ...]event_i "scorechar", "insname", idelay, idur, [, ip4] [, ip5] [, ...] iHandle instance insnum, iwhen, idur [, ip4] [, ip5] [...]iHandle instance "insname", iwhen, idur [, ip4] [, ip5] [...] mute insnum [, iswitch]mute "insname" [, iswitch] readscore Sin remove insnum schedkwhen ktrigger, kmintim, kmaxnum, kinsnum, kwhen, kdur \ [, ip4] [, ip5] [...]schedkwhen ktrigger, kmintim, kmaxnum, "insname", kwhen, kdur \ [, ip4] [, ip5] [...] schedkwhennamed ktrigger, kmintim, kmaxnum, "name", kwhen, kdur \ [, ip4] [, ip5] [...] schedule insnum, iwhen, idur [, ip4] [, ip5] [...]schedule "insname", iwhen, idur [, ip4] [, ip5] [...] schedwhen ktrigger, kinsnum, kwhen, kdur [, ip4] [, ip5] [...]schedwhen ktrigger, "insname", kwhen, kdur [, ip4] [, ip5] [...] scoreline Sin, ktrig scoreline_i Sin Instrument Control:Program Flow Control cggoto condition, label cigoto condition, label ckgoto condition, label cngoto condition, label else  elseif xa R xb then  endif  goto label if ia R ib igoto labelif ka R kb kgoto labelif xa R xb goto labelif xa R xb then  igoto label kgoto label loop_ge indx, idecr, imin, labelloop_ge kndx, kdecr, kmin, label loop_gt indx, idecr, imin, labelloop_gt kndx, kdecr, kmin, label loop_le indx, incr, imax, labelloop_le kndx, kncr, kmax, label loop_lt indx, incr, imax, labelloop_lt kndx, kncr, kmax, label tigoto label timout istrt, idur, label until condition do ... od  Instrument Control:Realtime Performance Control ir active insnum [,iopt]ir active Sinsname [,iopt]kres active kinsnum [,iopt] ktot[,kcpu1, kcpu2,...]cpumeter ifreq cpuprc insnum, ipercentcpuprc Sinsname, ipercent exitnow  jacktransport icommand [, ilocation] maxalloc insnum, icountmaxalloc Sinsname, icount prealloc insnum, icountprealloc "insname", icount Instrument Control:Initialization and Reinitialization ares = xargires = iargkres = kargires, ... = iarg, ...kres, ... = karg, ...table [ kval] = karg ares init iargires init iargkres init iargares, ... init iarg, ...ires, ... init iarg, ...kres, ... init iarg, ...tab init isize[, ival] insno nstrnum "name" p(x) ivar1, ... passign [istart] ares += xargires += iargkres += kargtable [ kval] += karg pset icon1 [, icon2] [...] reinit label rigoto label rireturn  ir tival  Instrument Control:Sensing and Control kres button knum ktrig changed kvar1 [, kvar2,..., kvarN] kres checkbox knum kres control knum ares follow asig, idt ares follow2 asig, katt, krel Svalue getcfg iopt kres joystick kdevice ktab ktrig metro kfreq [, initphase] ksig midifilestatus  ksig miditempo  p5gconnect  kres p5gdata kcontrol icount pcount  kres peak asigkres peak ksig ivalue pindex ipfieldIndex koct, kamp pitch asig, iupdte, ilo, ihi, idbthresh [, ifrqs] [, iconf] \ [, istrt] [, iocts] [, iq] [, inptls] [, irolloff] [, iskip] kcps, krms pitchamdf asig, imincps, imaxcps [, icps] [, imedi] \ [, idowns] [, iexcps] [, irmsmedi] acps, alock plltrack asig, kd [, kloopf, kloopq, klf, khf, kthresh] kcps, kamp ptrack asig, ihopsize[,ipeaks] ival readscratch[index] rewindscore  kres rms asig [, ihp] [, iskip] kres[, kkeydown] sensekey  ktrig_out seqtime ktime_unit, kstart, kloop, kinitndx, kfn_times ktrig_out seqtime2 ktrig_in, ktime_unit, kstart, kloop, kinitndx, kfn_times setctrl inum, ival, itype setscorepos ipos  splitrig ktrig, kndx, imaxtics, ifn, kout1 [,kout2,...,koutN] ktemp tempest kin, iprd, imindur, imemdur, ihp, ithresh, ihtim, ixfdbak, \ istartempo, ifn [, idisprd] [, itweek] tempo ktempo, istartempo kres tempoval  ktrig timedseq ktimpnt, ifn, kp1 [,kp2, kp3, ...,kpN] kout trigger ksig, kthreshold, kmode trigseq ktrig_in, kstart, kloop, kinitndx, kfn_values, kout1 [, kout2] [...] ires wiiconnect [itimeout, imaxnum] kres wiidata kcontrol[, knum] wiirange icontrol, iminimum, imaximum[, inum] kres wiisend kcontrol, kvalue[, knum] writescratchival[, index] kx, ky xyin iprd, ixmin, ixmax, iymin, iymax [, ixinit] [, iyinit] Instrument Control:Stacks xval1, [xval2, ... , xval31] pop ival1, [ival2, ... , ival31] pop  fsig pop_f  push xval1, [xval2, ... , xval31]push ival1, [ival2, ... , ival31] push_f fsig stack iStackSize Instrument Control:Subinstrument Control a1, [...] [, a8] subinstr instrnum [, p4] [, p5] [...]a1, [...] [, a8] subinstr "insname" [, p4] [, p5] [...] subinstrinit instrnum [, p4] [, p5] [...]subinstrinit "insname" [, p4] [, p5] [...] Instrument Control:Time Reading ir date  Sir dates [ itime] ir readclock inum ires rtclock kres rtclock  kres timeinstk  kres timeinsts  ires timek kres timek  ires times kres times  Jacko Opcodes asignal JackoAudioIn ScsoundPortName JackoAudioInConnect SexternalPortName, ScsoundPortName JackoAudioOut ScsoundPortName, asignal JackoAudioOutConnect ScsoundPortName, SexternalPortName JackoFreewheel [ienabled] JackoInfo   JackoInit SclientName, ServerName JackoMidiInConnect SexternalPortName, ScsoundPortName JackoMidiOut ScsoundPortName, kstatus, kchannel, kdata1[, kdata2] JackoMidiOutConnect ScsoundPortName, SexternalPortName JackoNoteOut ScsoundPortName, kstatus, kchannel, kdata1[, kdata2] JackoOn [iactive] JackoTransport kcommand, [kposition] Lua Opcodes lua_exec Sluacode lua_iopcall Sname, ...lua_ikopcall Sname, ...lua_iaopcall Sname, ...lua_iopcall_off Sname, ...lua_ikopcall_off Sname, ...lua_iaopcall_off Sname, ... lua_opdef Sname, Sluacode Serial I/O iPort serialBegin SPortName [, ibaudRate] serialEnd iPort serialFlush iPort serialPrint iPort kByte serialRead iPort serialWrite iPort, iByte serialWrite iPort, kByte serialWrite iPort, SBytes serialWrite_i iPort, iByte serialWrite_i iPort, SBytes Table Control ftfree ifno, iwhen gir ftgen ifn, itime, isize, igen, iarga [, iargb ] [...] ifno ftgentmp ip1, ip2dummy, isize, igen, iarga, iargb, ... sndload Sfname[, ifmt[, ichns[, isr[, ibas[, iamp[, istrt \ [, ilpmod[, ilps[, ilpe]]]]]]]]] Table Control:Table Queries karray[] array ival1, ival2,.....ivaln karray[] fillarray ival1, ival2,.....ivaln ftchnls(x) (init-rate args only) ftcps(x) (init-rate args only) ftlen(x) (init-rate args only) ftlptim(x) (init-rate args only) ftsr(x) (init-rate args only) karray genarray kstart, kens[, inc]iarray genarray istart, iens[, inc] karray genarray_i istart, iend [,inc] ir lenarray karraykr lenarray karray karray maparray kinarray, Stringkarray maparray_i kinarray, String nsamp(x) (init-rate args only) karray slicearray kinarray, istart, iend ires tableng ifnkres tableng kfn kr tabsum ifn[[, kmin] [, kmax]] tb0_init ifntb1_init ifntb2_init ifntb3_init ifntb4_init ifntb5_init ifntb6_init ifntb7_init ifntb8_init ifntb9_init ifntb10_init ifntb11_init ifntb12_init ifntb13_init ifntb14_init ifntb15_init ifniout = tb0(iIndex)kout = tb0(kIndex)iout = tb1(iIndex)kout = tb1(kIndex)iout = tb2(iIndex)kout = tb2(kIndex)iout = tb3(iIndex)kout = tb3(kIndex)iout = tb4(iIndex)kout = tb4(kIndex)iout = tb5(iIndex)kout = tb5(kIndex)iout = tb6(iIndex)kout = tb6(kIndex)iout = tb7(iIndex)kout = tb7(kIndex)iout = tb8(iIndex)kout = tb8(kIndex)iout = tb9(iIndex)kout = tb9(kIndex)iout = tb10(iIndex)kout = tb10(kIndex)iout = tb11(iIndex)kout = tb11(kIndex)iout = tb12(iIndex)kout = tb12(kIndex)iout = tb13(iIndex)kout = tb13(kIndex)iout = tb14(iIndex)kout = tb14(kIndex)iout = tb15(iIndex)kout = tb15(kIndex) Table Control:Dynamic Selection ares tableikt xndx, kfn [, ixmode] [, ixoff] [, iwrap]kres tableikt kndx, kfn [, ixmode] [, ixoff] [, iwrap] ares tablekt xndx, kfn [, ixmode] [, ixoff] [, iwrap]kres tablekt kndx, kfn [, ixmode] [, ixoff] [, iwrap] ares tablexkt xndx, kfn, kwarp, iwsize [, ixmode] [, ixoff] [, iwrap] Table Control:Read/Write Opreations ftload "filename", iflag, ifn1 [, ifn2] [...] ftloadk "filename", ktrig, iflag, ifn1 [, ifn2] [...] ftsave "filename", iflag, ifn1 [, ifn2] [...] ftsavek "filename", ktrig, iflag, ifn1 [, ifn2] [...] ptablew asig, andx, ifn [, ixmode] [, ixoff] [, iwgmode]ptablew isig, indx, ifn [, ixmode] [, ixoff] [, iwgmode]ptablew ksig, kndx, ifn [, ixmode] [, ixoff] [, iwgmode] tablecopy kdft, ksft knumpassed tablefilter kouttable, kintatble, kmode, kparam inumpassed tablefilteri iouttable, iintatble, imode, iparam tablegpw kfn tableicopy idft, isft tableigpw ifn tableimix idft, idoff, ilen, is1ft, is1off, is1g, is2ft, is2off, is2g tableiw isig, indx, ifn [, ixmode] [, ixoff] [, iwgmode] tablemix kdft, kdoff, klen, ks1ft, ks1off, ks1g, ks2ft, ks2off, ks2g ares tablera kfn, kstart, koff tablew asig, andx, ifn [, ixmode] [, ixoff] [, iwgmode]tablew isig, indx, ifn [, ixmode] [, ixoff] [, iwgmode]tablew ksig, kndx, ifn [, ixmode] [, ixoff] [, iwgmode] kstart tablewa kfn, asig, koff tablewkt asig, andx, kfn [, ixmode] [, ixoff] [, iwgmode]tablewkt ksig, kndx, kfn [, ixmode] [, ixoff] [, iwgmode] kout tabmorph kindex, kweightpoint, ktabnum1, ktabnum2, \ ifn1, ifn2 [, ifn3, ifn4, ...,ifnN] aout tabmorpha aindex, aweightpoint, atabnum1, atabnum2, \ ifn1, ifn2 [, ifn3, ifn4, ... ifnN] aout tabmorphak aindex, kweightpoint, ktabnum1, ktabnum2, \ ifn1, ifn2 [, ifn3, ifn4, ... ifnN] kout tabmorphi kindex, kweightpoint, ktabnum1, ktabnum2, \ ifn1, ifn2 [, ifn3, ifn4, ..., ifnN] tabplay ktrig, knumtics, kfn, kout1 [,kout2,..., koutN] tabrec ktrig_start, ktrig_stop, knumtics, kfn, kin1 [,kin2,...,kinN] FLTK:Containers FLgroup "label", iwidth, iheight, ix, iy [, iborder] [, image] FLgroupEnd  FLpack iwidth, iheight, ix, iy, itype, ispace, iborder FLpackEnd  FLpanel "label", iwidth, iheight [, ix] [, iy] [, iborder] [, ikbdcapture] [, iclose] FLpanelEnd  FLscroll iwidth, iheight [, ix] [, iy] FLscrollEnd  FLtabs iwidth, iheight, ix, iy FLtabsEnd  FLTK:Valuators kout, ihandle FLcount "label", imin, imax, istep1, istep2, itype, \ iwidth, iheight, ix, iy, iopcode [, kp1] [, kp2] [, kp3] [...] [, kpN] koutx, kouty, ihandlex, ihandley FLjoy "label", iminx, imaxx, iminy, \ imaxy, iexpx, iexpy, idispx, idispy, iwidth, iheight, ix, iy kout, ihandle FLknob "label", imin, imax, iexp, itype, idisp, iwidth, \ ix, iy [, icursorsize] kout, ihandle FLroller "label", imin, imax, istep, iexp, itype, idisp, \ iwidth, iheight, ix, iy kout, ihandle FLslider "label", imin, imax, iexp, itype, idisp, iwidth, \ iheight, ix, iy kout, ihandle FLtext "label", imin, imax, istep, itype, iwidth, \ iheight, ix, iy FLTK:Other ihandle FLbox "label", itype, ifont, isize, iwidth, iheight, ix, iy [, image] kout, ihandle FLbutBank itype, inumx, inumy, iwidth, iheight, ix, iy, \ iopcode [, kp1] [, kp2] [, kp3] [, kp4] [, kp5] [....] [, kpN] kout, ihandle FLbutton "label", ion, ioff, itype, iwidth, iheight, ix, \ iy, iopcode [, kp1] [, kp2] [, kp3] [, kp4] [, kp5] [....] [, kpN] ihandle FLcloseButton "label", iwidth, iheight, ix, iy ihandle FLexecButton "command", iwidth, iheight, ix, iy inumsnap FLgetsnap index [, igroup] ihandle FLhvsBox inumlinesX, inumlinesY, iwidth, iheight, ix, iy [, image] FLhvsBox kx, ky, ihandle kascii FLkeyIn [ifn] FLloadsnap "filename" [, igroup] kx, ky, kb1, kb2, kb3 FLmouse [imode] FLprintk itime, kval, idisp FLprintk2 kval, idisp FLrun  FLsavesnap "filename" [, igroup] inumsnap, inumval FLsetsnap index [, ifn, igroup] FLsetSnapGroup igroup FLsetVal ktrig, kvalue, ihandle FLsetVal_i ivalue, ihandle FLslidBnk "names", inumsliders [, ioutable] [, iwidth] [, iheight] [, ix] \ [, iy] [, itypetable] [, iexptable] [, istart_index] [, iminmaxtable] FLslidBnk2 "names", inumsliders, ioutable, iconfigtable [,iwidth, iheight, ix, iy, istart_index] FLslidBnk2Set ihandle, ifn [, istartIndex, istartSlid, inumSlid] FLslidBnk2Setk ktrig, ihandle, ifn [, istartIndex, istartSlid, inumSlid] ihandle FLslidBnkGetHandle  FLslidBnkSet ihandle, ifn [, istartIndex, istartSlid, inumSlid] FLslidBnkSetk ktrig, ihandle, ifn [, istartIndex, istartSlid, inumSlid] FLupdate  ihandle FLvalue "label", iwidth, iheight, ix, iy FLvkeybd "keyboard.map", iwidth, iheight, ix, iy FLvslidBnk "names", inumsliders [, ioutable] [, iwidth] [, iheight] [, ix] \ [, iy] [, itypetable] [, iexptable] [, istart_index] [, iminmaxtable] FLvslidBnk2 "names", inumsliders, ioutable, iconfigtable [,iwidth, iheight, ix, iy, istart_index] koutx, kouty, kinside FLxyin ioutx_min, ioutx_max, iouty_min, iouty_max, \ iwindx_min, iwindx_max, iwindy_min, iwindy_max [, iexpx, iexpy, ioutx, iouty] vphaseseg kphase, ioutab, ielems, itab1,idist1,itab2 \ [,idist2,itab3, ... ,idistN-1,itabN] FLTK:Appearance FLcolor ired, igreen, iblue [, ired2, igreen2, iblue2] FLcolor2 ired, igreen, iblue FLhide ihandle FLlabel isize, ifont, ialign, ired, igreen, iblue FLsetAlign ialign, ihandle FLsetBox itype, ihandle FLsetColor ired, igreen, iblue, ihandle FLsetColor2 ired, igreen, iblue, ihandle FLsetFont ifont, ihandle FLsetPosition ix, iy, ihandle FLsetSize iwidth, iheight, ihandle FLsetText "itext", ihandle FLsetTextColor ired, iblue, igreen, ihandle FLsetTextSize isize, ihandle FLsetTextType itype, ihandle FLshow ihandle Mathematical Operations:Arithmetic and Logic Operations a + b (no rate restriction) a / b (no rate restriction) a % b (no rate restriction) a * b (no rate restriction) a && b (logical AND; not audio-rate) a & b (bitwise AND) ~ a (bitwise NOT) a | b (bitwise OR) a << b (bitshift left) a >> b (bitshift left) a # b (bitwise NON EQUIVALENCE) a || b (logical OR; not audio-rate) a ˆ b (b not audio-rate) a − b (no rate restriction) Mathematical Operations:Comparators and Accumulators clear avar1 [, avar2] [, avar3] [...] vincr accum, aincr Mathematical Operations:Mathematical Functions abs(x) (no rate restriction) ceil(x) (init-, control-, or audio-rate arg allowed) exp(x) (no rate restriction) floor(x) (init-, control-, or audio-rate arg allowed) frac(x) (init-rate or control-rate args; also works at audio rate in Csound5) int(x) (init-rate or control-rate; also works at audio rate in Csound5) log(x) (no rate restriction) log10(x) (no rate restriction) log2(x) (no rate restriction) logbtwo(x) (init-rate or control-rate args only) powoftwo(x) (init-rate or control-rate args only) qinf(x) (no rate restriction) qnan(x) (no rate restriction) round(x) (init-, control-, or audio-rate arg allowed) sqrt(x) (no rate restriction) Mathematical Operations:Trigonometric Functions cos(x) (no rate restriction) cosh(x) (no rate restriction) cosinv(x) (no rate restriction) signum(x) (no rate restriction) sin(x) (no rate restriction) sinh(x) (no rate restriction) sininv(x) (no rate restriction) tan(x) (no rate restriction) tanh(x) (no rate restriction) taninv(x) (no rate restriction) Mathematical Operations:Amplitude Functions ampdb(x) (no rate restriction) ampdbfs(x) (no rate restriction) db(x) dbamp(x) (init-rate or control-rate args only) dbfsamp(x) (init-rate or control-rate args only) Mathematical Operations:Random Functions birnd(x) (init- or control-rate only) rnd(x) (init- or control-rate only) Mathematical Operations:Opcode Equivalents of Functions ares divz xa, xb, ksubstires divz ia, ib, isubstkres divz ka, kb, ksubst...divz(ka, kb, ksubst)... (no rate restriction) ares mac ksig1, asig1 [, ksig2] [, asig2] [, ksig3] [, asig3] [...] ares maca asig1 , asig2 [, asig3] [, asig4] [, asig5] [...] aout polynomial ain, k0 [, k1 [, k2 [...]]] ares pow aarg, kpow [, inorm]ires pow iarg, ipow [, inorm]kres pow karg, kpow [, inorm] ares product asig1, asig2 [, asig3] [...] ares sum asig1 [, asig2] [, asig3] [...] ares taninv2 ay, axires taninv2 iy, ixkres taninv2 ky, kx...taninv2(ky, kx)... (no rate restriction) Pitch Converters:Functions cent(x) cpsmidinn (MidiNoteNumber) (init- or control-rate args only) cpsoct (oct) (no rate restriction) cpspch (pch) (init- or control-rate args only) octave(x) octcps (cps) (init- or control-rate args only) octmidinn (MidiNoteNumber) (init- or control-rate args only) octpch (pch) (init- or control-rate args only) pchmidinn (MidiNoteNumber) (init- or control-rate args only) pchoct (oct) (init- or control-rate args only) semitone(x) Pitch Converters:Tuning Opcodes icps cps2pch ipch, iequal kcps cpstun ktrig, kindex, kfn icps cpstuni index, ifn icps cpsxpch ipch, iequal, irepeat, ibase Real-time MIDI:Input kaft aftouch [imin] [, imax] ival chanctrl ichnl, ictlno [, ilow] [, ihigh]kval chanctrl ichnl, ictlno [, ilow] [, ihigh] idest ctrl14 ichan, ictlno1, ictlno2, imin, imax [, ifn]kdest ctrl14 ichan, ictlno1, ictlno2, kmin, kmax [, ifn] idest ctrl21 ichan, ictlno1, ictlno2, ictlno3, imin, imax [, ifn]kdest ctrl21 ichan, ictlno1, ictlno2, ictlno3, kmin, kmax [, ifn] idest ctrl7 ichan, ictlno, imin, imax [, ifn]kdest ctrl7 ichan, ictlno, kmin, kmax [, ifn]adest ctrl7 ichan, ictlno, kmin, kmax [, ifn] [, icutoff] ctrlinit ichnl, ictlno1, ival1 [, ictlno2] [, ival2] [, ictlno3] \ [, ival3] [,...ival32] initc14 ichan, ictlno1, ictlno2, ivalue initc21 ichan, ictlno1, ictlno2, ictlno3, ivalue initc7 ichan, ictlno, ivalue massign ichnl, insnum[, ireset]massign ichnl, "insname"[, ireset] idest midic14 ictlno1, ictlno2, imin, imax [, ifn]kdest midic14 ictlno1, ictlno2, kmin, kmax [, ifn] idest midic21 ictlno1, ictlno2, ictlno3, imin, imax [, ifn]kdest midic21 ictlno1, ictlno2, ictlno3, kmin, kmax [, ifn] idest midic7 ictlno, imin, imax [, ifn]kdest midic7 ictlno, kmin, kmax [, ifn] ival midictrl inum [, imin] [, imax]kval midictrl inum [, imin] [, imax] ival notnum  ibend pchbend [imin] [, imax]kbend pchbend [imin] [, imax] pgmassign ipgm, inst[, ichn]pgmassign ipgm, "insname"[, ichn] ires polyaft inote [, ilow] [, ihigh]kres polyaft inote [, ilow] [, ihigh] ival veloc [ilow] [, ihigh] Real-time MIDI:Output nrpn kchan, kparmnum, kparmvalue outiat ichn, ivalue, imin, imax outic ichn, inum, ivalue, imin, imax outic14 ichn, imsb, ilsb, ivalue, imin, imax outipat ichn, inotenum, ivalue, imin, imax outipb ichn, ivalue, imin, imax outipc ichn, iprog, imin, imax outkat kchn, kvalue, kmin, kmax outkc kchn, knum, kvalue, kmin, kmax outkc14 kchn, kmsb, klsb, kvalue, kmin, kmax outkpat kchn, knotenum, kvalue, kmin, kmax outkpb kchn, kvalue, kmin, kmax outkpc kchn, kprog, kmin, kmax Real-time MIDI:Converters iamp ampmidi iscal [, ifn] iamplitude ampmidid ivelocity, idecibelskamplitude ampmidid kvelocity, idecibels icps cpsmidi  icps cpsmidib [irange]kcps cpsmidib [irange] icps cpstmid ifn ioct octmidi  ioct octmidib [irange]koct octmidib [irange] ipch pchmidi  ipch pchmidib [irange]kpch pchmidib [irange] Real-time MIDI:Generic I/O kstatus, kchan, kdata1, kdata2 midiin  midiout kstatus, kchan, kdata1, kdata2 Real-time MIDI:Event Extenders kflag release  xtratim iextradur Real-time MIDI:Note Output midion kchn, knum, kvel midion2 kchn, knum, kvel, ktrig moscil kchn, knum, kvel, kdur, kpause noteoff ichn, inum, ivel noteon ichn, inum, ivel noteondur ichn, inum, ivel, idur noteondur2 ichn, inum, ivel, idur Real-time MIDI:MIDI/Score Interoperability midichannelaftertouch xchannelaftertouch [, ilow] [, ihigh] ichn midichn  midicontrolchange xcontroller, xcontrollervalue [, ilow] [, ihigh] mididefault xdefault, xvalue midinoteoff xkey, xvelocity midinoteoncps xcps, xvelocity midinoteonkey xkey, xvelocity midinoteonoct xoct, xvelocity midinoteonpch xpch, xvelocity midipitchbend xpitchbend [, ilow] [, ihigh] midipolyaftertouch xpolyaftertouch, xcontrollervalue [, ilow] [, ihigh] midiprogramchange xprogram Real-time MIDI:System Realtime mclock ifreq mrtmsg imsgtype Real-time MIDI:Slider Banks i1,...,i16 s16b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, \ initvalue1, ifn1,..., ictlno_msb16, ictlno_lsb16, imin16, imax16, initvalue16, ifn16k1,...,k16 s16b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, \ initvalue1, ifn1,..., ictlno_msb16, ictlno_lsb16, imin16, imax16, initvalue16, ifn16 i1,...,i32 s32b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, \ initvalue1, ifn1,..., ictlno_msb32, ictlno_lsb32, imin32, imax32, initvalue32, ifn32k1,...,k32 s32b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, \ initvalue1, ifn1,..., ictlno_msb32, ictlno_lsb32, imin32, imax32, initvalue32, ifn32 i1,...,i16 slider16 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum16, imin16, imax16, init16, ifn16k1,...,k16 slider16 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum16, imin16, imax16, init16, ifn16 k1,...,k16 slider16f ichan, ictlnum1, imin1, imax1, init1, ifn1, \ icutoff1,..., ictlnum16, imin16, imax16, init16, ifn16, icutoff16 kflag slider16table ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, .... , ictlnum16, imin16, imax16, init16, ifn16 kflag slider16tablef ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, icutoff1, .... , ictlnum16, imin16, imax16, init16, ifn16, icutoff16 i1,...,i32 slider32 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum32, imin32, imax32, init32, ifn32k1,...,k32 slider32 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum32, imin32, imax32, init32, ifn32 k1,...,k32 slider32f ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1, \ ..., ictlnum32, imin32, imax32, init32, ifn32, icutoff32 kflag slider32table ichan, ioutTable, ioffset, ictlnum1, imin1, \ imax1, init1, ifn1, .... , ictlnum32, imin32, imax32, init32, ifn32 kflag slider32tablef ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, icutoff1, .... , ictlnum32, imin32, imax32, init32, ifn32, icutoff32 i1,...,i64 slider64 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum64, imin64, imax64, init64, ifn64k1,...,k64 slider64 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum64, imin64, imax64, init64, ifn64 k1,...,k64 slider64f ichan, ictlnum1, imin1, imax1, init1, ifn1, \ icutoff1,..., ictlnum64, imin64, imax64, init64, ifn64, icutoff64 kflag slider64table ichan, ioutTable, ioffset, ictlnum1, imin1, \ imax1, init1, ifn1, .... , ictlnum64, imin64, imax64, init64, ifn64 kflag slider64tablef ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, icutoff1, .... , ictlnum64, imin64, imax64, init64, ifn64, icutoff64 i1,...,i8 slider8 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum8, imin8, imax8, init8, ifn8k1,...,k8 slider8 ichan, ictlnum1, imin1, imax1, init1, ifn1,..., \ ictlnum8, imin8, imax8, init8, ifn8 k1,...,k8 slider8f ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1, \ ..., ictlnum8, imin8, imax8, init8, ifn8, icutoff8 kflag slider8table ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1,..., ictlnum8, imin8, imax8, init8, ifn8 kflag slider8tablef ichan, ioutTable, ioffset, ictlnum1, imin1, imax1, \ init1, ifn1, icutoff1, .... , ictlnum8, imin8, imax8, init8, ifn8, icutoff8 k1, k2, ...., k16 sliderKawai imin1, imax1, init1, ifn1, \ imin2, imax2, init2, ifn2, ..., imin16, imax16, init16, ifn16 Signal Flow Graph Opcodes alwayson Tinstrument [p4, ..., pn] connect Tsource1, Soutlet1, Tsink1, Sinlet1 ifno ftgenonce ip1, ip2dummy, isize, igen, iarga, iargb, ... asignal inleta Sname fsignal inletf Sname ksignal inletk Sname ksignal inletkid Sname, SinstanceID outleta Sname, asignal outletf Sname, fsignal outletk Sname, ksignal outletkid Sname, SinstanceID, ksignal Spectral Processing:STFT ktableseg ifn1, idur1, ifn2 [, idur2] [, ifn3] [...] ares pvadd ktimpnt, kfmod, ifilcod, ifn, ibins [, ibinoffset] \ [, ibinincr] [, iextractmode] [, ifreqlim] [, igatefn] pvbufread ktimpnt, ifile ares pvcross ktimpnt, kfmod, ifile, kampscale1, kampscale2 [, ispecwp] ares pvinterp ktimpnt, kfmod, ifile, kfreqscale1, kfreqscale2, \ kampscale1, kampscale2, kfreqinterp, kampinterp ares pvoc ktimpnt, kfmod, ifilcod [, ispecwp] [, iextractmode] \ [, ifreqlim] [, igatefn] kfreq, kamp pvread ktimpnt, ifile, ibin tableseg ifn1, idur1, ifn2 [, idur2] [, ifn3] [...] tablexseg ifn1, idur1, ifn2 [, idur2] [, ifn3] [...] ares vpvoc ktimpnt, kfmod, ifile [, ispecwp] [, ifn] Spectral Processing:LPC ares lpfreson asig, kfrqratio lpinterp islot1, islot2, kmix krmsr, krmso, kerr, kcps lpread ktimpnt, ifilcod [, inpoles] [, ifrmrate] ares lpreson asig lpslot islot Spectral Processing:Non-Standard wsig specaddm wsig1, wsig2 [, imul2] wsig specdiff wsigin specdisp wsig, iprd [, iwtflg] wsig specfilt wsigin, ifhtim wsig spechist wsigin koct, kamp specptrk wsig, kvar, ilo, ihi, istr, idbthresh, inptls, \ irolloff [, iodd] [, iconfs] [, interp] [, ifprd] [, iwtflg] wsig specscal wsigin, ifscale, ifthresh ksum specsum wsig [, interp] wsig spectrum xsig, iprd, iocts, ifrqa [, iq] [, ihann] [, idbout] \ [, idsprd] [, idsinrs] Spectral Processing:Streaming fsig binit fin, isize ftrks partials ffr, fphs, kthresh, kminpts, kmaxgap, imaxtracks kframe pvs2tab tvar, fsig ares pvsadsyn fsrc, inoscs, kfmod [, ibinoffset] [, ibinincr] [, iinit] fsig pvsanal ain, ifftsize, ioverlap, iwinsize, iwintype [, iformat] [, iinit] fsig pvsarp fsigin, kbin, kdepth, kgain fsig pvsbandp fsigin, xlowcut, xlowfull, \ xhighfull, xhighcut[, ktype] fsig pvsbandr fsigin, xlowcut, xlowfull, \ xhighfull, xhighcut[, ktype] kamp, kfr pvsbin fsig, kbin fsig pvsblur fsigin, kblurtime, imaxdel ihandle, ktime pvsbuffer fsig, ilen fsig pvsbufread ktime, khandle[, ilo, ihi, iclear] fsig pvsbufread2 ktime, khandle, ift1, ift2 fsig pvscale fsigin, kscal[, kkeepform, kgain, kcoefs] kcent pvscent fsig fsig pvscross fsrc, fdest, kamp1, kamp2 fsig pvsdemix fleft, fright, kpos, kwidth, ipoints fsig pvsdiskin SFname,ktscal,kgain[,ioffset, ichan] pvsdisp fsig[, ibins, iwtflg] fsig pvsfilter fsigin, fsigfil, kdepth[, igain] fsig pvsfread ktimpt, ifn [, ichan] fsig pvsfreeze fsigin, kfreeza, kfreezf pvsftr fsrc, ifna [, ifnf] kflag pvsftw fsrc, ifna [, ifnf] pvsfwrite fsig, ifile fsig pvsgain fsigin, kgain fsig pvshift fsigin, kshift, klowest[, kkeepform, igain, kcoefs] ffr,fphs pvsifd ain, ifftsize, ihopsize, iwintype[,iscal] fsig pvsin kchan[, isize, iolap, iwinsize, iwintype, iformat] ioverlap, inumbins, iwinsize, iformat pvsinfo fsrc fsig pvsinit isize[, iolap, iwinsize, iwintype, iformat] fsig pvslock fsigin, klock fsig pvsmaska fsrc, ifn, kdepth fsig pvsmix fsigin1, fsigin2 fsig pvsmooth fsigin, kacf, kfcf fsig pvsmorph fsig1, fsig2, kampint, kfrqint fsig pvsosc kamp, kfreq, ktype, isize [,ioverlap] [, iwinsize] [, iwintype] [, iformat] pvsout fsig, kchan kfr, kamp pvspitch fsig, kthresh fsig pvstanal ktimescal, kamp, kpitch, ktab, [kdetect, kwrap, ioffset,ifftsize, ihop, idbthresh] fsig pvstencil fsigin, kgain, klevel, iftable fsig pvsvoc famp, fexc, kdepth, kgain [,kcoefs] fsig pvswarp fsigin, kscal, kshift[, klowest, kmeth, kgain, kcoefs] ares pvsynth fsrc, [iinit] asig resyn fin, kscal, kpitch, kmaxtracks, ifn asig sinsyn fin, kscal, kmaxtracks, ifn fsig tab2pvs tvar[,ihopsize, iwinsize, iwintype] asig tradsyn fin, kscal, kpitch, kmaxtracks, ifn fsig trcross fin1, fin2, ksearch, kdepth [, kmode] fsig trfilter fin, kamnt, ifn fsig, kfr, kamp trhighest fin1, kscal fsig, kfr, kamp trlowest fin1, kscal fsig trmix fin1, fin2 fsig trscale fin, kpitch[, kgain] fsig trshift fin, kpshift[, kgain] fsiglow, fsighi trsplit fin, ksplit[, kgainlow, kgainhigh] Spectral Processing:ATS ar ATSadd ktimepnt, kfmod, iatsfile, ifn, ipartials[, ipartialoffset, \ ipartialincr, igatefn] ar ATSaddnz ktimepnt, iatsfile, ibands[, ibandoffset, ibandincr] ATSbufread ktimepnt, kfmod, iatsfile, ipartials[, ipartialoffset, \ ipartialincr] ar ATScross ktimepnt, kfmod, iatsfile, ifn, kmylev, kbuflev, ipartials \ [, ipartialoffset, ipartialincr] idata ATSinfo iatsfile, ilocation kamp ATSinterpread kfreq kfrq, kamp ATSpartialtap ipartialnum kfreq, kamp ATSread ktimepnt, iatsfile, ipartial kenergy ATSreadnz ktimepnt, iatsfile, iband ar ATSsinnoi ktimepnt, ksinlev, knzlev, kfmod, iatsfile, ipartials \ [, ipartialoffset, ipartialincr] Spectral Processing:Loris lorismorph isrcidx, itgtidx, istoreidx, kfreqmorphenv, kampmorphenv, kbwmorphenv ar lorisplay ireadidx, kfreqenv, kampenv, kbwenv lorisread ktimpnt, ifilcod, istoreidx, kfreqenv, kampenv, kbwenv[, ifadetime] Spectral Processing:Other kcent centroid asig, ktrig, ifftsize asig mincer atimpt, kamp, kpitch, ktab, klock[,ifftsize,idecim] asig temposcal ktimescal, kamp, kpitch, ktab, klock [,ifftsize, idecim, ithresh] Strings:Definition Sdst strfromurl StringURL Sdst strget indx strset iarg, istring Strings:Manipulation puts Sstr, ktrig[, inonl] Sdst sprintf Sfmt, xarg1[, xarg2[, ... ]] Sdst sprintfk Sfmt, xarg1[, xarg2[, ... ]] Sdst strcat Ssrc1, Ssrc2 Sdst strcatk Ssrc1, Ssrc2 ires strcmp S1, S2 kres strcmpk S1, S2 Sdst strcpy SsrcSdst = Ssrc Sdst strcpyk Ssrc ipos strindex S1, S2 kpos strindexk S1, S2 ilen strlen Sstr klen strlenk Sstr ipos strrindex S1, S2 kpos strrindexk S1, S2 Sdst strsub Ssrc[, istart[, iend]] Sdst strsubk Ssrc, kstart, kend Strings:Conversion ichr strchar Sstr[, ipos] kchr strchark Sstr[, kpos] Sdst strlower Ssrc Sdst strlowerk Ssrc ir strtod Sstrir strtod indx kr strtodk Sstrkr strtodk kndx ir strtol Sstrir strtol indx kr strtolk Sstrkr strtolk kndx Sdst strupper Ssrc Sdst strupperk Ssrc Vectorial:Tables vtaba andx, ifn, aout1 [, aout2, aout3, .... , aoutN ] vtabi indx, ifn, iout1 [, iout2, iout3, .... , ioutN ] vtabk kndx, ifn, kout1 [, kout2, kout3, .... , koutN ] vtable1k kfn,kout1 [, kout2, kout3, .... , koutN ] vtablea andx, kfn, kinterp, ixmode, aout1 [, aout2, aout3, .... , aoutN ] vtablei indx, ifn, interp, ixmode, iout1 [, iout2, iout3, .... , ioutN ] vtablek kndx, kfn, kinterp, ixmode, kout1 [, kout2, kout3, .... , koutN ] vtablewa andx, kfn, ixmode, ainarg1 [, ainarg2, ainarg3 , .... , ainargN ] vtablewi indx, ifn, ixmode, inarg1 [, inarg2, inarg3 , .... , inargN ] vtablewk kndx, kfn, ixmode, kinarg1 [, kinarg2, kinarg3 , .... , kinargN ] vtabwa andx, ifn, ainarg1 [, ainarg2, ainarg3 , .... , ainargN ] vtabwi indx, ifn, inarg1 [, inarg2, inarg3 , .... , inargN ] vtabwk kndx, ifn, kinarg1 [, kinarg2, kinarg3 , .... , kinargN ] Vectorial:Scalar operations vadd ifn, kval, kelements [, kdstoffset] [, kverbose] vadd_i ifn, ival, ielements [, idstoffset] vexp ifn, kval, kelements [, kdstoffset] [, kverbose] vexp_i ifn, ival, ielements[, idstoffset] vmult ifn, kval, kelements [, kdstoffset] [, kverbose] vmult_i ifn, ival, ielements [, idstoffset] vpow ifn, kval, kelements [, kdstoffset] [, kverbose] vpow_i ifn, ival, ielements [, idstoffset] Vectorial:Vectorial operations vaddv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] vaddv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] vcopy ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [, kverbose] vcopy_i ifn1, ifn2, ielements [,idstoffset, isrcoffset] vdivv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] vdivv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] vexpv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] vexpv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] vmap ifn1, ifn2, ielements [,idstoffset, isrcoffset] vmultv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] vmultv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] vpowv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] vpowv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] vsubv ifn1, ifn2, kelements [, kdstoffset] [, ksrcoffset] [,kverbose] vsubv_i ifn1, ifn2, ielements [, idstoffset] [, isrcoffset] Vectorial:Envelopes vexpseg ifnout, ielements, ifn1, idur1, ifn2 [, idur2, ifn3 [...]] vlinseg ifnout, ielements, ifn1, idur1, ifn2 [, idur2, ifn3 [...]] Vectorial:Limiting and Wrapping vlimit ifn, kmin, kmax, ielements vmirror ifn, kmin, kmax, ielements vwrap ifn, kmin, kmax, ielements Vectorial:Delay Paths kout vdelayk ksig, kdel, imaxdel [, iskip, imode] vecdelay ifn, ifnIn, ifnDel, ielements, imaxdel [, iskip] vport ifn, khtime, ielements [, ifnInit] Vectorial:Random vrandh ifn, krange, kcps, ielements [, idstoffset] [, iseed] \ [, isize] [, ioffset] vrandi ifn, krange, kcps, ielements [, idstoffset] [, iseed] \ [, isize] [, ioffset] Vectorial:Cellular Automata cell ktrig, kreinit, ioutFunc, initStateFunc, iRuleFunc, ielements vcella ktrig, kreinit, ioutFunc, initStateFunc, \ iRuleFunc, ielements, irulelen [, iradius] Zak Patch System zacl kfirst, klast zakinit isizea, isizek ares zamod asig, kzamod ares zar kndx ares zarg kndx, kgain zaw asig, kndx zawm asig, kndx [, imix] ir zir indx ziw isig, indx ziwm isig, indx [, imix] zkcl kfirst, klast kres zkmod ksig, kzkmod kres zkr kndx zkw ksig, kndx zkwm ksig, kndx [, imix] Plugin Hosting:DSSI and LADSPA dssiactivate ihandle, ktoggle [aout1, aout2, ..., aout9] dssiaudio ihandle, [ain1, ain2, ..., ain9] dssictls ihandle, iport, kvalue, ktrigger ihandle dssiinit ilibraryname, iplugindex [, iverbose] dssilist  Plugin Hosting:VST aout1,aout2 vstaudio instance, [ain1, ain2]aout1,aout2 vstaudiog instance, [ain1, ain2] vstbankload instance, ipath vstedit instance vstinfo instance instance vstinit ilibrarypath [,iverbose] vstmidiout instance, kstatus, kchan, kdata1, kdata2 vstnote instance, kchan, knote, kveloc, kdur vstparamset instance, kparam, kvaluekvalue vstparamget instance, kparam vstprogset instance, kprogram OSC ihandle OSCinit iport kans OSClisten ihandle, idest, itype [, xdata1, xdata2, ...] OSCsend kwhen, ihost, iport, idestination, itype [, kdata1, kdata2, ...] Faust Opcodes ihandle,a1[,a2,...] faustaudio ifac[,ain1,...] ihandle faustcompile Scode, Sargs faustctl idsp,Scontrol,kval ihandle,a1[,a2,...] faustgen SCode[,ain1,...] Network remoteport iportnum asig sockrecv iport, ilengthasigl, asigr sockrecvs iport, ilengthasig strecv Sipaddr, iport socksend asig, Sipaddr, iport, ilengthsocksends asigl, asigr, Sipaddr, iport, ilengthstsend asig, Sipaddr, iport Remote Opcodes insglobal isource, instrnum [,instrnum...] insremot idestination, isource, instrnum [,instrnum...] midglobal isource, instrnum [,instrnum...] midremot idestination, isource, instrnum [,instrnum...] Mixer Opcodes MixerClear  kgain MixerGetLevel isend, ibuss asignal MixerReceive ibuss, ichannel MixerSend asignal, isend, ibuss, ichannel MixerSetLevel isend, ibuss, kgain MixerSetLevel_i isend, ibuss, igain Python Opcodes pyassign "variable", kvaluepyassigni "variable", ivaluepylassign "variable", kvaluepylassigni "variable", ivaluepyassignt ktrigger, "variable", kvaluepylassignt ktrigger, "variable", kvalue pycall "callable", karg1, ...kresult pycall1 "callable", karg1, ...kresult1, kresult2 pycall2 "callable", karg1, ...kr1, kr2, kr3 pycall3 "callable", karg1, ...kr1, kr2, kr3, kr4 pycall4 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5 pycall5 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6 pycall6 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7 pycall7 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7, kr8 pycall8 "callable", karg1, ... pycallt ktrigger, "callable", karg1, ...kresult pycall1t ktrigger, "callable", karg1, ...kresult1, kresult2 pycall2t ktrigger, "callable", karg1, ...kr1, kr2, kr3 pycall3t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4 pycall4t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5 pycall5t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6 pycall6t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7 pycall7t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7, kr8 pycall8t ktrigger, "callable", karg1, ... pycalli "callable", karg1, ...iresult pycall1i "callable", iarg1, ...iresult1, iresult2 pycall2i "callable", iarg1, ...ir1, ir2, ir3 pycall3i "callable", iarg1, ...ir1, ir2, ir3, ir4 pycall4i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5 pycall5i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5, ir6 pycall6i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5, ir6, ir7 pycall7i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5, ir6, ir7, ir8 pycall8i "callable", iarg1, ...pycalln "callable", nresults, kresult1, ..., kresultn, karg1, ...pycallni "callable", nresults, iresult1, ..., iresultn, iarg1, ... pylcall "callable", karg1, ...kresult pylcall1 "callable", karg1, ...kresult1, kresult2 pylcall2 "callable", karg1, ...kr1, kr2, kr3 pylcall3 "callable", karg1, ...kr1, kr2, kr3, kr4 pylcall4 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5 pylcall5 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6 pylcall6 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7 pylcall7 "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7, kr8 pylcall8 "callable", karg1, ... pylcallt ktrigger, "callable", karg1, ...kresult pylcall1t ktrigger, "callable", karg1, ...kresult1, kresult2 pylcall2t ktrigger, "callable", karg1, ...kr1, kr2, kr3 pylcall3t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4 pylcall4t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5 pylcall5t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6 pylcall6t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7 pylcall7t ktrigger, "callable", karg1, ...kr1, kr2, kr3, kr4, kr5, kr6, kr7, kr8 pylcall8t ktrigger, "callable", karg1, ... pylcalli "callable", karg1, ...iresult pylcall1i "callable", iarg1, ...iresult1, iresult2 pylcall2i "callable", iarg1, ...ir1, ir2, ir3 pylcall3i "callable", iarg1, ...ir1, ir2, ir3, ir4 pylcall4i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5 pylcall5i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5, ir6 pylcall6i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5, ir6, ir7 pylcall7i "callable", iarg1, ...ir1, ir2, ir3, ir4, ir5, ir6, ir7, ir8 pylcall8i "callable", iarg1, ...pylcalln "callable", nresults, kresult1, ..., kresultn, karg1, ...pylcallni "callable", nresults, iresult1, ..., iresultn, iarg1, ... kresult pyeval "expression"iresult pyevali "expression"kresult pyleval "expression"iresult pylevali "expression"kresult pyevalt ktrigger, "expression"kresult pylevalt ktrigger, "expression" pyexec "filename"pyexeci "filename"pylexec "filename"pylexeci "filename"pyexect ktrigger, "filename"plyexect ktrigger, "filename" pyinit  pyrun "statement"pyruni "statement"pylrun "statement"pylruni "statement"pyrunt ktrigger, "statement"pylrunt ktrigger, "statement" Image Processing Opcodes iimagenum imagecreate iwidth, iheight imagefree iimagenum ared, agreen, ablue imagegetpixel iimagenum, ax, aykred, kgreen, kblue imagegetpixel iimagenum, kx, ky iimagenum imageload filename imagesave iimagenum, filename imagesetpixel iimagenum, ax, ay, ared, agreen, ablueimagesetpixel iimagenum, kx, ky, kred, kgreen, kblue iwidth, iheight imagesize iimagenum Array Opcodes copya2ftab tab, kftbl copyf2array tab, kftbl kmax maxarray karray kmin minarray karray scalearray tarray, kmin, kmax[, kleft, kright] ksum sumarray karray Miscellaneous kfl fareylen kfn ifl fareyleni ifn modmatrix iresfn, isrcmodfn, isrcparmfn, imodscale, inum_mod, \\ inum_parm, kupdate Sres pwd  ires system_i itrig, Scmd, [inowait]kres system ktrig, Scmd, [knowait] tableshuffle ktablenumtableshufflei itablenum Utilities csound -U atsa [flags] infilename outfilename cs [-OPTIONS] <name> [CSOUND OPTIONS ... ] csb64enc [OPTIONS ... ] infile1 [ infile2 [ ... ]] csound -U cvanal [flags] infilename outfilenamecvanal [flags] infilename outfilename dnoise [flags] -i noise_ref_file -o output_soundfile input_soundfile envext [-flags] soundfilecsound -U envext [-flags] soundfile extractor [OPTIONS ... ] infile het_export het_file cstext_filecsound -U het_export het_file cstext_file het_import cstext_file het_filecsound -U het_import cstext_file het_file csound -U hetro [flags] infilename outfilenamehetro [flags] infilename outfilename csound -U lpanal [flags] infilename outfilenamelpanal [flags] infilename outfilename makecsd [OPTIONS ... ] infile1 [ infile2 [ ... ]] mixer [OPTIONS ... ] infile [[OPTIONS... ] infile] ... mkdb [-m] [base_directory] pv_export pv_file cstext_filecsound -U pv_export pv_file cstext_file pv_import cstext_file pv_filecsound -U pv_import cstext_file pv_file csound -U pvanal [flags] infilename outfilenamepvanal [flags] infilename outfilename csound -U pvlook [flags] infilenamepvlook [flags] infilename scale [OPTIONS ... ] infile sdif2ad [flags] infilename outfilename csound -U sndinfo [options] soundfilenames ...sndinfo [options] soundfilenames ... srconv [flags] infile manual_src~dfsg/misc/modalfreq.xml0000644000000000000000000001216312262561504016377 0ustar rootroot Modal Frequency Ratios Contributed by Scott Lindroth John Bower, a student of Scott Lindroth, compiled this list of modal frequencies for various objects and materials. Some modes work better than others, and most need to be in a particular frequency range to sound plausible. Caveat emptor. In general, wooden objects will not sound "wooden" unless a stochastic component is present in the sound (try banded waveguides). Nonetheless, some of the wooden objects make wonderful metallic instruments as well. This ratios can be useful together with opcodes like mode or streson. Modal Frequency Ratios Instrument Modal Frequency Ratios Dahina tabla [1, 2.89, 4.95, 6.99, 8.01, 9.02] Bayan tabla [1, 2.0, 3.01, 4.01, 4.69, 5.63] Red Cedar wood plate [1, 1.47, 2.09, 2.56] Redwood wood plate [1, 1.47, 2.11, 2.57] Douglas Fir wood plate [1, 1.42, 2.11, 2.47] uniform wooden bar [1, 2.572, 4.644, 6.984, 9.723, 12] uniform aluminum bar [1, 2.756, 5.423, 8.988, 13.448, 18.680] Xylophone [1, 3.932, 9.538, 16.688, 24.566, 31.147] Vibraphone 1 [1, 3.984, 10.668, 17.979, 23.679, 33.642] Vibraphone 2 [1, 3.997, 9.469, 15.566, 20.863, 29.440] Chalandi plates ([62, 107, 360, 460, 863] Hz +-2Hz) [1, 1.72581, 5.80645, 7.41935, 13.91935] ratios tibetan bowl (180mm) ( [221, 614, 1145, 1804, 2577, 3456, 4419] Hz) 934g, 180mm [1, 2.77828, 5.18099, 8.16289, 11.66063, 15.63801, 19.99 ratios tibetan bowl (152 mm) ([314, 836, 1519, 2360, 3341, 4462, 5696] Hz) 563g, 152mm [1, 2.66242, 4.83757, 7.51592, 10.64012, 14.21019, 18.14027] ratios tibetan bowl (140 mm) ([528, 1460, 2704, 4122, 5694] Hz) 557g, 140mm [1, 2.76515, 5.12121, 7.80681, 10.78409] ratios Wine Glass [1, 2.32, 4.25, 6.63, 9.38] small handbell ([1312.0, 1314.5, 2353.3, 2362.9, 3306.5, 3309.4, 3923.8, 3928.2, 4966.6, 4993.7, 5994.4, 6003.0, 6598.9, 6619.7, 7971.7, 7753.2, 8413.1, 8453.3, 9292.4, 9305.2, 9602.3, 9912.4] Hz) [ 1, 1.0019054878049, 1.7936737804878, 1.8009908536585, 2.5201981707317, 2.5224085365854, 2.9907012195122, 2.9940548780488, 3.7855182926829, 3.8061737804878, 4.5689024390244, 4.5754573170732, 5.0296493902439, 5.0455030487805, 6.0759908536585, 5.9094512195122, 6.4124237804878, 6.4430640243902, 7.0826219512195, 7.0923780487805, 7.3188262195122, 7.5551829268293 ] ratios spinel sphere with diameter of 3.6675mm ([977.25, 1003.16, 1390.13, 1414.93, 1432.84, 1465.34, 1748.48, 1834.20, 1919.90, 1933.64, 1987.20, 2096.48, 2107.10, 2202.08, 2238.40, 2280.10, 0 /*2290.53 calculated*/, 2400.88, 2435.85, 2507.80, 2546.30, 2608.55, 2652.35, 2691.70, 2708.00] Hz) [ 1, 1.026513174725, 1.4224916858532, 1.4478690202098, 1.4661959580455, 1.499452545408, 1.7891839345101, 1.8768994627782, 1.9645945254541, 1.9786543873113, 2.0334612432847, 2.1452852391916, 2.1561524686621, 2.2533435661294, 2.2905090816065, 2.3331798413917, 0, 2.4567715528268, 2.4925556408289, 2.5661806088514, 2.6055768738808, 2.6692760296751, 2.7140956766436, 2.7543617293425, 2.7710411870043 ] ratios pot lid [ 1, 3.2, 6.23, 6.27, 9.92, 14.15] ratios
manual_src~dfsg/misc/top.xml0000644000000000000000000000023712262561504015226 0ustar rootroot &miscquickref; &miscexamples; &miscpitch; &miscamp; &miscformants; &miscmodalfreq; &miscwindows; &miscsf2; &misccsound64; &miscglossary; manual_src~dfsg/misc/csound64.xml0000644000000000000000000001030412262561504016065 0ustar rootroot Csound Double (64-bit) vs. Float (32-bit) Csound can be built to use 64-bit DOUBLES internally to do processing versus regular Csound's 32-bit FLOATS. This larger resolution for processing internally yields a much "cleaner" sound but at the expense of extended processing time. Because it does require much longer to process, Csound compiled for doubles is typicaly used after a work is finished for a final production run. If you are using csound for realtime output, you should use the 32-bit (float) version, which provides faster output. For offline rendering, you can use either, but for the final master, the 64-bit version will produce higher quality output. The actual resolution should be the same as for the type of the audio sample variable. For "float" Csound, that is a 32-bit, single-precision floating point number. It has 24 bits of precision in the mantissa. For "double" Csound, that is a 64-bit, double-precision floating point number in the mantissa. It has 52 bits of precision. For each decimal digit, there is between 3 and 4 bits. So, there are 7 digits of precision for "float" and 16 digits of precision for "double." For each multiplication or division, depending on whether the operands are integers, rational numbers, repeating decimals, or irrational decimals, there may or may not be rounding error in the result. If there is rounding error, it is at most 1 bit of lost precision per operation (in addition to any rounding in the binary representation of rational or real numbers). For float samples, if the signal stays within the mantissa, the signal to noise ratio is 6.02 times 24, or 144 dB. At worst, each operation will create 6.02 dB of additional noise due to rounding error. Our ears have an effective dynamic range of 120 to 130 dB, but we like our music compressed to a dynamic range of AT MOST 60 dB (and usually much less, say 20 dB). That gives us (144 - 60) / 6.02 = about 10 worst-case operations before we could possibly hear any degradation. In practice, we have several times that many operations before we hear any degradation or noise. For double samples, if the signal stays within the mantissa, the signal to noise ratio is 6.02 times 52, or 313 dB. At worst, each operation will create 6.02 dB of additional noise due to rounding error. That gives us (313 - 60) / 6.02 = about 42, in practice several times that, of operations before there is any audible degradation or noise. But if you trace the number of arithmetic operations in typical Csound instruments or other software synthesizers, the very complex instruments definitely are pushing into the range of audible degradation on good monitors with float, so it is no surprise that in some cases blindfold, controlled ABX testing confirms occasional audible differences between music synthesized with "float" Csound versus the same music synthesized with "double" Csound. Equally, it is no surprise that there are easily audible differences between digital and analog implementations of the same synthesis algorithms. There is a good deal more "digital signal processing headroom" with double Csound, and therefore it should be used for all music rendered for critical listening. The float version should only be used where its speed advantage of about 15% is critical for real-time performance. Notes On Using Csound built for double precision hetro, PVOC-EX analysis and pvanal files generated for 32-bit Csound (float) will work with 64-bit Csound (double precision). All other analysis formats can be generated in a way for cross-use between 32 and 64 bit versions and big-endian and little-endian architectures, if the -X option is used in generation. The older format in csound5 is still generatable but is not machins independant. manual_src~dfsg/misc/pitch.xml0000644000000000000000000004622612262561504015543 0ustar rootroot Pitch Conversion Pitch Conversion Note Hz cpspch MIDI C-1 8.176 3.00 0 C#-1 8.662 3.01 1 D-1 9.177 3.02 2 D#-1 9.723 3.03 3 E-1 10.301 3.04 4 F-1 10.913 3.05 5 F#-1 11.562 3.06 6 G-1 12.250 3.07 7 G#-1 12.978 3.08 8 A-1 13.750 3.09 9 A#-1 14.568 3.10 10 B-1 15.434 3.11 11 C0 16.352 4.00 12 C#0 17.324 4.01 13 D0 18.354 4.02 14 D#0 19.445 4.03 15 E0 20.602 4.04 16 F0 21.827 4.05 17 F#0 23.125 4.06 18 G0 24.500 4.07 19 G#0 25.957 4.08 20 A0 27.500 4.09 21 A#0 29.135 4.10 22 B0 30.868 4.11 23 C1 32.703 5.00 24 C#1 34.648 5.01 25 D1 36.708 5.02 26 D#1 38.891 5.03 27 E1 41.203 5.04 28 F1 43.654 5.05 29 F#1 46.249 5.06 30 G1 48.999 5.07 31 G#1 51.913 5.08 32 A1 55.000 5.09 33 A#1 58.270 5.10 34 B1 61.735 5.11 35 C2 65.406 6.00 36 C#2 69.296 6.01 37 D2 73.416 6.02 38 D#2 77.782 6.03 39 E2 82.407 6.04 40 F2 87.307 6.05 41 F#2 92.499 6.06 42 G2 97.999 6.07 43 G#2 103.826 6.08 44 A2 110.000 6.09 45 A#2 116.541 6.10 46 B2 123.471 6.11 47 C3 130.813 7.00 48 C#3 138.591 7.01 49 D3 146.832 7.02 50 D#3 155.563 7.03 51 E3 164.814 7.04 52 F3 174.614 7.05 53 F#3 184.997 7.06 54 G3 195.998 7.07 55 G#3 207.652 7.08 56 A3 220.000 7.09 57 A#3 233.082 7.10 58 B3 246.942 7.11 59 C4 261.626 8.00 60 C#4 277.183 8.01 61 D4 293.665 8.02 62 D#4 311.127 8.03 63 E4 329.628 8.04 64 F4 349.228 8.05 65 F#4 369.994 8.06 66 G4 391.995 8.07 67 G#4 415.305 8.08 68 A4 440.000 8.09 69 A#4 466.164 8.10 70 B4 493.883 8.11 71 C5 523.251 9.00 72 C#5 554.365 9.01 73 D5 587.330 9.02 74 D#5 622.254 9.03 75 E5 659.255 9.04 76 F5 698.456 9.05 77 F#5 739.989 9.06 78 G5 783.991 9.07 79 G#5 830.609 9.08 80 A5 880.000 9.09 81 A#5 932.328 9.10 82 B5 987.767 9.11 83 C6 1046.502 10.00 84 C#6 1108.731 10.01 85 D6 1174.659 10.02 86 D#6 1244.508 10.03 87 E6 1318.510 10.04 88 F6 1396.913 10.05 89 F#6 1479.978 10.06 90 G6 1567.982 10.07 91 G#6 1661.219 10.08 92 A6 1760.000 10.09 93 A#6 1864.655 10.10 94 B6 1975.533 10.11 95 C7 2093.005 11.00 96 C#7 2217.461 11.01 97 D7 2349.318 11.02 98 D#7 2489.016 11.03 99 E7 2637.020 11.04 100 F7 2793.826 11.05 101 F#7 2959.955 11.06 102 G7 3135.963 11.07 103 G#7 3322.438 11.08 104 A7 3520.000 11.09 105 A#7 3729.310 11.10 106 B7 3951.066 11.11 107 C8 4186.009 12.00 108 C#8 4434.922 12.01 109 D8 4698.636 12.02 110 D#8 4978.032 12.03 111 E8 5274.041 12.04 112 F8 5587.652 12.05 113 F#8 5919.911 12.06 114 G8 6271.927 12.07 115 G#8 6644.875 12.08 116 A8 7040.000 12.09 117 A#8 7458.620 12.10 118 B8 7902.133 12.11 119 C9 8372.018 13.00 120 C#9 8869.844 13.01 121 D9 9397.273 13.02 122 D#9 9956.063 13.03 123 E9 10548.08 13.04 124 F9 11175.30 13.05 125 F#9 11839.82 13.06 126 G9 12543.85 13.07 127
manual_src~dfsg/misc/examples.xml0000644000000000000000000033434712262561504016256 0ustar rootroot List of examplesOrchestra Syntax:Header 0dbfs.csd nchnls.csd nchnls_i.csd sr.csd Orchestra Syntax:Block Statements endin.csd endop.csd instr.csd opcode_example.csd Orchestra Syntax:Macros define.csd define_args.csd define.csd define_args.csd include.csd Signal Generators:Additive Synthesis/Resynthesis adsyn.csd adsynt.csd adsynt2.csd hsboscil.csd hsboscil_midi.csd Signal Generators:Basic Oscillators lfo.csd oscbnk.csd oscil.csd oscil3.csd oscili.csd oscilikt.csd osciliktp.csd oscilikts.csd osciln.csd oscils.csd poscil.csd poscil3.csd poscil3-file.csd vibr.csd vibrato.csd Signal Generators:Dynamic Spectrum Oscillators buzz.csd gbuzz.csd mpulse.csd vco.csd vco2.csd vco2ift.csd vco2init.csd Signal Generators:FM Synthesis crossfm.csd fmb3.csd fmbell.csd fmmetal.csd fmpercfl.csd fmrhode.csd fmvoice.csd fmwurlie.csd foscil.csd foscili.csd Signal Generators:Granular Synthesis diskgrain.csd fof.csd fof2.csd fof2-2.csd fog.csd grain.csd grain2.csd grain3.csd granule.csd partikkel.csd partikkel-2.csd partikkelsync.csd sndwarp.csd sndwarpst.csd syncgrain.csd syncloop.csd vosim.csd Signal Generators:Hyper Vectorial Synthesis hvs1.csd hvs2.csd hvs2-2.csd hvs3.csd Signal Generators:Linear and Exponential Generators cosseg.csd cossegb.csd cossegr.csd expcurve.csd expon.csd expseg.csd expsega.csd expsegb.csd expsegba.csd expsegr.csd gainslider.csd jspline.csd line.csd linseg.csd linsegb.csd linsegr.csd logcurve.csd loopseg.csd loopsegp.csd looptseg.csd loopxseg.csd lpshold.csd rspline.csd scale.csd transeg.csd transegb.csd transegr.csd Signal Generators:Envelope Generators adsr.csd envlpx.csd envlpxr.csd linen.csd linenr.csd madsr.csd madsr-2.csd mxadsr.csd xadsr.csd Signal Generators:Models and Emulations bamboo.csd barmodel.csd cabasa.csd chuap.csd crunch.csd dripwater.csd gendy.csd gendy-2.csd gendyc.csd gendyx.csd gendyx-2.csd gogobel.csd guiro.csd lorenz.csd mandel.csd mandol.csd marimba.csd moog.csd planet.csd prepiano.csd sandpaper.csd sekere.csd shaker.csd sleighbells.csd stix.csd tambourine.csd vibes.csd voice.csd Signal Generators:Phasors phasor.csd phasorbnk.csd syncphasor.csd syncphasor-CZresonance.csd Signal Generators:Random (Noise) Generators betarand.csd bexprnd.csd cauchy.csd cauchyi.csd cuserrnd.csd duserrnd.csd dust.csd dust2.csd exprand.csd exprandi.csd fractalnoise.csd gauss.csd gaussi.csd gausstrig.csd gausstrig-2.csd jitter.csd jitter2.csd linrand.csd noise.csd noise-2.csd pcauchy.csd pinkish.csd poisson.csd rand.csd randh.csd randi.csd random.csd randomh.csd randomi.csd rnd31.csd rnd31_krate.csd rnd31_seed7.csd rnd31_time.csd seed.csd trandom.csd trirand.csd unirand.csd urandom.csd urandom_krate.csd urd.csd weibull.csd Signal Generators:Sample Playback bbcutm.csd bbcuts.csd flooper.csd flooper2.csd fluidAllOut.csd fluidCCi.csd fluidCCk.csd fluidcomplex.csd fluidEngine.csd fluidLoad.csd fluidNote.csd fluidOut.csd fluidProgramSelect.csd fluidcomplex.csd fluidSetInterpMethod.csd loscil.csd loscil3.csd lphasor.csd lposcil.csd lposcil3.csd lposcila.csd lposcilsa.csd lposcilsa2.csd sfilist.csd sfinstr.csd sfinstr3.csd sfinstr3m.csd sfload.csd sflooper.csd sfpassign.csd sfplay3.csd sfplay3m.csd sfplaym.csd sfplist.csd sfpreset.csd sndloop.csd waveset.csd Signal Generators:Scanned Synthesis scans.csd scans-2.csd scantable.csd xscanmap.csd xscans.csd xscanu.csd Signal Generators:STK Opcodes STKBandedWG.csd STKBeeThree.csd STKBlowBotl.csd STKBlowHole.csd STKBowed.csd STKBrass.csd STKClarinet.csd STKDrummer.csd STKFMVoices.csd STKFlute.csd STKHevyMetl.csd STKMandolin.csd STKModalBar.csd STKMoog.csd STKPercFlut.csd STKPlucked.csd STKResonate.csd STKRhodey.csd STKSaxofony.csd STKShakers.csd STKSimple.csd STKSitar.csd STKStifKarp.csd STKTubeBell.csd STKVoicForm.csd STKWhistle.csd STKWurley.csd Signal Generators:Table Access oscil1i.csd oscil1i.csd ptable.csd ptablei.csd tab.csd table.csd tablei.csd Signal Generators:Wave Terrain Synthesis wterrain.csd Signal Generators:Waveguide Physical Modeling pluck.csd repluck.csd streson.csd wgbow.csd wgbowedbar.csd wgbrass.csd wgclar.csd wgflute.csd wgpluck.csd wgpluck_brighter.csd wgpluck2.csd Signal I/O:File I/O dumpk.csd dumpk-2.csd dumpk2.csd dumpk3.csd dumpk4.csd ficlose.csd fin.csd fiopen.csd fout.csd fout_ftable.csd fouti.csd foutir.csd fprintks.csd fprintks-2.csd scogen.csd fprints.csd readf.csd readfi.csd readk.csd readk-2.csd readk2.csd readk3.csd readk4.csd Signal I/O:Signal Input diskin.csd diskin2.csd in.csd inch.csd inq.csd inrg.csd ins.csd invalue.csd mp3in.csd soundin.csd Signal I/O:Signal Output mdelay.csd monitor.csd out.csd outc.csd outch.csd outch-2.csd outq.csd outq1.csd outq2.csd outq3.csd outq4.csd outrg.csd outs.csd outs1.csd outs2.csd outvalue.csd Signal I/O:Software Bus chnclear.csd chnget.csd chnmix.csd chnset.csd Signal I/O:Printing and Display dispfft.csd display.csd flashtxt.csd print.csd printf.csd printk.csd printk2.csd printks.csd prints.csd Signal I/O:Soundfile Queries filebit.csd filelen.csd filenchnls.csd filepeak.csd filesr.csd filevalid.csd mp3len.csd Signal Modifiers:Amplitude Modifiers balance.csd clip.csd compress.csd dam.csd dam_expanded.csd gain.csd Signal Modifiers:Convolution and Morphing convolve.csd cross2.csd dconv.csd ftconv.csd ftmorf.csd pconvolve.csd Signal Modifiers:Delay delay.csd delay1.csd delayk.csd delayr.csd delayw.csd deltap.csd deltap3.csd deltapi.csd deltapn.csd deltapx.csd deltapxw.csd multitap.csd newopcodename.csd vdelay.csd vdelay3.csd vdelayx.csd vdelayxq.csd vdelayxs.csd vdelayxw.csd vdelayxwq.csd vdelayxws.csd Signal Modifiers:Panning and Spatialization bformenc.csd bformenc1.csd bformenc.csd bformenc1.csd hrtfearly.csd hrtfer.csd hrtfmove.csd hrtfmove2.csd hrtfstat.csd hrtfstat-2.csd locsend_stereo.csd locsig_quad.csd pan.csd pan2.csd space_quad.csd space_stereo.csd spat3d_stereo.csd spat3d_UHJ.csd spat3d_quad.csd spat3dt.csd spdist.csd spsend.csd vbap.csd vbap4.csd vbap4move.csd vbap8.csd vbap8move.csd vbapg.csd vbapgmove.csd vbaplsinit.csd vbapmove.csd Signal Modifiers:Reverberation alpass.csd babo.csd babo_expert.csd comb.csd combinv.csd freeverb.csd nestedap.csd nreverb.csd nreverb_ftable.csd plate.csd reverb.csd reverbsc.csd valpass.csd valpass-2.csd vcomb.csd Signal Modifiers:Sample Level Operators denorm.csd diff.csd downsamp.csd fold.csd integ.csd interp.csd ntrpol.csd opa.csd samphold.csd upsamp.csd vaget.csd vaset.csd Signal Modifiers:Signal Limiters limit.csd mirror.csd wrap.csd Signal Modifiers:Special Effects distort.csd distort1.csd flanger.csd harmon.csd harmon3.csd phaser1.csd phaser2.csd Signal Modifiers:Standard Filters atone.csd atonex.csd biquad.csd biquad-2.csd butterbp.csd butterbr.csd butterhp.csd butterlp.csd clfilt_lowpass.csd clfilt_highpass.csd doppler.csd median.csd mediank.csd mode.csd tone.csd tonex.csd Signal Modifiers:Standard Filters:Resonant areson.csd bqrez.csd lowpass2.csd lowres.csd lowresx.csd lpf18.csd moogladder.csd moogvcf.csd moogvcf2.csd reson.csd resonr.csd resonx.csd resony.csd resonr.csd rezzy.csd statevar.csd svfilter.csd tbvcf.csd vlowres.csd Signal Modifiers:Standard Filters:Control aresonk.csd atonek.csd lineto.csd port.csd portk.csd resonk.csd resonxk.csd tlineto.csd tonek.csd Signal Modifiers:Specialized Filters dcblock.csd dcblock2.csd eqfil.csd filter2.csd fofilter.csd hilbert.csd hilbert_barberpole.csd nlfilt.csd nlfilt2.csd pareq.csd rbjeq.csd Signal Modifiers:Waveguides wguide1.csd wguide2.csd Signal Modifiers:Waveshaping chebyshevpoly.csd pdclip.csd pdhalf.csd pdhalfy.csd powershape.csd Signal Modifiers:Comparators and Accumulators max.csd max_k.csd maxabs.csd maxabsaccum.csd maxaccum.csd min.csd minabs.csd minabsaccum.csd minaccum.csd Instrument Control:Clock Control clockoff.csd clockon.csd Instrument Control:Conditional Values equals.csd greaterequal.csd greaterthan.csd lessequal.csd lessthan.csd notequal.csd Instrument Control:Compilation compileorc.csd compilestr.csd Instrument Control:Duration Control ihold.csd kill.csd turnoff.csd Instrument Control:Invocation event.csd event_named.csd event_i.csd instance.csd mute.csd readscore.csd schedkwhen.csd schedkwhennamed.csd schedule.csd schedwhen.csd scoreline.csd scoreline_i.csd Instrument Control:Program Flow Control cggoto.csd cigoto.csd ckgoto.csd cngoto.csd else.csd elseif.csd endif.csd goto.csd igoto.csd kgoto.csd ifthen.csd igoto.csd kgoto.csd loop_le.csd loop_lt.csd tigoto.csd timout.csd until.csd Instrument Control:Realtime Performance Control active.csd active_k.csd active_scale.csd cpumeter.csd cpuprc.csd exitnow.csd jacktransport.csd maxalloc.csd prealloc.csd Instrument Control:Initialization and Reinitialization assign.csd init.csd p.csd passign.csd reverb.csd pset.csd pset-midi.csd reinit.csd reinit.csd tival.csd Instrument Control:Sensing and Control changed.csd checkbox.csd follow.csd follow2.csd getcfg.csd joystick.csd joystick-2.csd metro.csd metro-2.csd miditempo.csd p5g.csd pcount.csd peak.csd pindex.csd pindex-2.csd pitch.csd pitchamdf.csd plltrack.csd ptrack.csd readscratch.csd rewindscore.csd rms.csd sensekey.csd FLpanel-sensekey.csd FLpanel-sensekey2.csd seqtime.csd seqtime2.csd setctrl.csd setscorepos.csd tempest.csd tempo.csd tempoval.csd timedseq.csd trigger.csd trigseq.csd wii.csd readscratch.csd xyin.csd Instrument Control:Stacks pop.csd push.csd stack.csd Instrument Control:Subinstrument Control subinstr.csd subinstr_named.csd Instrument Control:Time Reading date.csd dates.csd readclock.csd rtclock.csd timeinstk.csd timeinsts.csd timek.csd times_complex.csd Jacko Opcodes JackoInfo.csd JackoInit.csd Lua Opcodes luaopcode.csd luamoog.csd Serial I/O serialBegin.csd serialRead.csd serialWrite.csd Table Control ftfree.csd ftgen.csd ftgen-2.csd ftgentmp.csd Table Control:Table Queries array.csd fillarray.csd ftchnls.csd ftcps.csd ftlen.csd ftlptim.csd ftsr.csd genarray.csd genarray_i.csd lenarray.csd maparray.csd nsamp.csd slicearray.csd tableng.csd tabsum.csd Table Control:Dynamic Selection tableikt.csd tablekt.csd tablexkt.csd Table Control:Read/Write Opreations ftsave.csd tablecopy.csd tablefilter.csd tablefilter.csd tableimix.csd tableiw.csd tablemix.csd tabmorph.csd tabmorpha.csd tabmorphak.csd tabmorphi.csd FLTK:Containers FLpanel.csd FLscroll.csd FLtabs.csd FLTK:Valuators FLcount.csd FLjoy.csd FLknob.csd FLknob-2.csd FLroller.csd FLslider.csd FLslider-2.csd FLtext.csd FLTK:Other FLbox.csd FLbutBank.csd FLbutton.csd FLexecButton.csd FLhvsBox.csd FLhvsBoxSetValue.csd FLkeyIn.csd FLmouse.csd FLsavesnap_simple.csd FLsavesnap.csd FLslidBnk.csd FLslidBnk2.csd FLslidBnk2Set.csd FLslidBnk2Setk.csd FLslidBnkGetHandle.csd FLslidBnkSet.csd FLslidBnkSetk.csd FLvalue.csd FLvslidBnk.csd FLvslidBnk2.csd FLxyin.csd FLxyin-2.csd vphaseseg.csd FLTK:Appearance FLsetcolor.csd FLsetText.csd Mathematical Operations:Arithmetic and Logic Operations adds.csd divides.csd modulus.csd multiplies.csd opand.csd bitwise.csd bitshift.csd logicOR.csd raises.csd subtracts.csd Mathematical Operations:Comparators and Accumulators clear.csd vincr.csd vincr-complex.csd Mathematical Operations:Mathematical Functions abs.csd ceil.csd exp.csd floor.csd frac.csd int.csd log.csd log10.csd log2.csd logbtwo.csd powoftwo.csd qinf.csd qnan.csd round.csd sqrt.csd Mathematical Operations:Trigonometric Functions cos.csd cosh.csd cosinv.csd signum.csd sin.csd sinh.csd sininv.csd tan.csd tanh.csd taninv.csd Mathematical Operations:Amplitude Functions ampdb.csd ampdbfs.csd db.csd dbamp.csd dbfsamp.csd Mathematical Operations:Random Functions birnd.csd rnd.csd Mathematical Operations:Opcode Equivalents of Functions divz.csd mac.csd maca.csd polynomial.csd pow.csd product.csd sum.csd taninv2.csd Pitch Converters:Functions cent.csd cpsmidinn.csd cpsmidinn2.csd cpsoct.csd cpspch.csd octave.csd octcps.csd cpsmidinn.csd octpch.csd cpsmidinn.csd pchoct.csd semitone.csd Pitch Converters:Tuning Opcodes cps2pch.csd cps2pch_ftable.csd cps2pch_19et.csd cpstun.csd cpstuni.csd cpsxpch.csd cpsxpch_105et.csd cpsxpch_pierce.csd Real-time MIDI:Input aftouch.csd chanctrl.csd ctrl7.csd initc7.csd massign.csd midic7.csd midictrl.csd notnum.csd pchbend.csd pgmassign.csd pgmassign_ignore.csd pgmassign_advanced.csd polyaft.csd veloc.csd Real-time MIDI:Output nrpn.csd outiat.csd outic.csd outipb.csd outipc.csd outkat.csd outkc.csd outkpb.csd outkpc.csd outkpc_fltk.csd Real-time MIDI:Converters ampmidi.csd ampmidid.csd cpsmidi.csd cpsmidib.csd cpstmid.csd octmidi.csd octmidib.csd pchmidi.csd pchmidib.csd Real-time MIDI:Generic I/O midiin.csd midiout.csd Real-time MIDI:Event Extenders xtratim.csd xtratim-2.csd Real-time MIDI:Note Output midion_simple.csd midion_scale.csd midion2.csd moscil.csd noteondur.csd noteondur2.csd Real-time MIDI:MIDI/Score Interoperability midichannelaftertouch.csd midichn.csd midichn_advanced.csd midicontrolchange.csd mididefault.csd midinoteoff.csd midinoteoncps.csd midinoteonkey.csd midinoteonoct.csd midinoteonpch.csd midipitchbend.csd midipolyaftertouch.csd midiprogramchange.csd Real-time MIDI:System Realtime mclock.csd Signal Flow Graph Opcodes alwayson.csd connect.csd ftgenonce.csd inleta.csd inletk.csd outleta.csd outletk.csd Spectral Processing:STFT pvadd.csd pvbufread.csd pvcross.csd pvinterp.csd pvoc.csd pvread.csd tableseg.csd tablexseg.csd vpvoc.csd Spectral Processing:LPC lpfreson.csd lpread.csd lpreson.csd lpreson-2.csd Spectral Processing:Streaming binit.csd partials.csd pvsadsyn.csd pvsanal.csd pvsarp.csd pvsarp2.csd pvsbandp.csd pvsbandr.csd pvsbin.csd pvsblur.csd pvsbufread.csd pvsbufread2.csd pvscale.csd pvscent.csd pvscross.csd pvsdiskin.csd pvsdisp.csd pvsfilter.csd pvsfread.csd pvsfreeze.csd pvsftr.csd pvsftw.csd pvsfwrite.csd pvsgain.csd pvshift.csd pvsifd.csd pvsinfo.csd pvslock.csd pvsmaska.csd pvsmix.csd pvsmooth.csd pvsmorph.csd pvsmorph2.csd pvsosc.csd pvspitch.csd pvstanal.csd pvsvoc.csd pvswarp.csd pvsynth.csd resyn.csd sinsyn.csd tradsyn.csd trcross.csd trfilter.csd trhighest.csd trlowest.csd trmix.csd trscale.csd trshift.csd trsplit.csd Spectral Processing:ATS ATSadd.csd ATSadd-2.csd ATSaddnz.csd ATSaddnz-2.csd ATSbufread.csd ATScross.csd ATSinfo.csd ATSinfo-2.csd ATSinterpread.csd ATSpartialtap.csd ATSread.csd ATSread-2.csd ATSreadnz.csd ATSsinnoi.csd ATSsinnoi-2.csd Spectral Processing:Loris lorismorph.csd lorisplay.csd lorisread.csd Spectral Processing:Other centroid.csd mincer.csd temposcal.csd Strings:Definition strfromurl.csd strget.csd strset.csd Strings:Manipulation puts.csd sprintf.csd sprintfk.csd strcat.csd strcatk.csd strcmp.csd strcpyk.csd strindexk.csd strrindex.csd strsub.csd Strings:Conversion strchar.csd strlower.csd strtod.csd strtodk.csd strtol.csd strtolk.csd Vectorial:Tables vtable1k.csd vtablei.csd vtablek.csd vtablewa.csd vtablewk.csd Vectorial:Scalar operations vadd.csd vadd_i.csd vexp.csd vexp_i.csd vmult-2.csd vmult.csd vmult_i.csd vpow.csd vpow-2.csd vpow_i.csd vpow_i-2.csd Vectorial:Vectorial operations vaddv.csd vcopy.csd vdivv.csd vexpv.csd vmap.csd vmultv.csd vpowv.csd vsubv.csd Vectorial:Envelopes vexpseg.csd vlinseg.csd Vectorial:Random vrandh.csd vrandi.csd Vectorial:Cellular Automata cell.csd vcella.csd Zak Patch System zacl.csd zakinit.csd zamod.csd zar.csd zarg.csd zaw.csd zawm.csd zir.csd ziw.csd ziwm.csd zkcl.csd zkmod.csd zkr.csd zkw.csd zkwm.csd Plugin Hosting:DSSI and LADSPA dssiactivate.csd dssiaudio.csd dssictls.csd dssiinit.csd dssilist.csd Plugin Hosting:VST vst4cs.csd vst4cs.csd vst4cs.csd vst4cs.csd vst4cs.csd vst4cs.csd OSC OSCmidisend.csd OSCmidircv.csd Remote Opcodes insremot.csd insremotM.csd midremot.csd Mixer Opcodes Mixer.csd Mixer.csd Mixer.csd Mixer.csd Mixer.csd Image Processing Opcodes imageopcodes.csd imageopcodes.csd imageopcodesdemo2.csd imageopcodes.csd imageopcodes.csd imageopcodes.csd imageopcodes.csd Array Opcodes copya2ftab.csd copyf2array.csd maxarray.csd minarray.csd scalearray.csd sumarray.csd Miscellaneous fareyleni.csd modmatrix.csd pwd.csd system.csd farey7shuffled.csd manual_src~dfsg/misc/glossary.xml0000644000000000000000000000337712262561504016277 0ustar rootroot Glossary G Guard Point A guard point is the last position on a function table. If the length is, say 1024, the table will have 1024+1 (1025) points: the extra point is the guard point. In any case, for a 1024-point table, the first point is index 0 and the last 1023; index 1024 is not really used) The reason for a guard-point is that some opcodes interpolate to obtain a table value, in which case, when the table index is say, 1023.5, we need the value of the 1024 pos in order to interpolate. There are two ways of filling this point (writing the value that goes in it): Default way: by copying the value of the 1st point in the table Extended Guard-Point: extending the contour of the table (continuing to calculate the table for one extra point) In general the first mode is used for wrap-around applications, such as an oscillator (which loops continuously reading the table). The second use is for one-shot readouts, such as envelopes, where the last point needs to be interpolated correctly following the table contour (we are not looping back to the beginning of the table) manual_src~dfsg/misc/sf2.xml0000644000000000000000000000373512262561504015124 0ustar rootroot SoundFont2 File Format Beginning with Csound Version 4.07, Csound supports the SoundFont2 sample file format. SoundFont2 (or SF2) is a widespread standard which allows encoding banks of wavetable-based sounds into a binary file. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format follows. The SF2 format is made by generator and modulator objects. All current Csound opcodes regarding SF2 support the generator function only. There are several levels of generators having a hierarchical structure. The most basic kind of generator object is a sample. Samples may or may not be be looped, and are associated with a MIDI note number, called the base-key. When a sample is associated with a range of MIDI note numbers, a range of velocities, a transposition (coarse and fine tuning), a scale tuning, and a level scaling factor, the sample and its associations make up a split. A set of splits, together with a name, make up an instrument. When an instrument is associated with a key range, a velocity range, a level scaling factor, and a transposition, the instrument and its associations make up a layer. A set of layers, together with a name, makes up a preset. Presets are normally the final sound-generating structures ready for the user. They generate sound according to the settings of their lower-level components. Both sample data and structure data is embedded in the same SF2 binary file. A single SF2 file can contain up to a maximum of 128 banks of 128 preset programs, for a total of 16384 presets in one SF2 file. The maximum number of layers, instruments, splits, and samples is not defined, and probably is only limited by the computer's memory. manual_src~dfsg/score/0000755000000000000000000000000012262561504014060 5ustar rootrootmanual_src~dfsg/score/nextp.xml0000644000000000000000000000625112262561504015744 0ustar rootroot
Next-P and Previous-P Symbols At the close of any of the operations Carry, Tempo, and Sort, three additional score features are interpreted during file writeout: next-p, previous-p, and ramping. np pp scorenext-p scoreprevious-p i statement pfields containing the symbols npx or ppx (where x is some integer) will be replaced by the appropriate pfield value found on the next i statement (or previous i statement) that has the same p1. For example, the symbol np7 will be replaced by the value found in p7 of the next note that is to be played by this instrument. np and pp symbols are recursive and can reference other np and pp symbols which can reference others, etc. References must eventually terminate in a real number or a ramp symbol. Closed loop references should be avoided. np and pp symbols are illegal in p1, p2 and p3 (although they may reference these). np and pp symbols may be Carried. np and pp references cannot cross a Section boundary. Any forward or backward reference to a non-existent note-statement will be given the value zero. E.g.: the statements i1 0 1 10 np4 pp5 i1 1 1 20 i1 1 1 30 will result in i1 0 1 10 20 0 i1 1 1 20 30 20 i1 2 1 30 0 30 np and pp symbols can provide an instrument with contextual knowledge of the score, enabling it to glissando or crescendo, for instance, toward the pitch or dynamic of some future event (which may or may not be immediately adjacent). Note that while the Carry feature will propagate np and pp through unsorted statements, the operation that interprets these symbols is acting on a time-warped and fully sorted version of the score.
manual_src~dfsg/score/mult.xml0000644000000000000000000000465512262561504015575 0ustar rootroot
Multiple File Score Description Using the score in more than one file. Syntax #include "filename" Performance It is sometimes convenient to have the score in more than one file. This use is supported by the #include facility which is part of the macro system. A line containing the text #includescore multiple file scores #include "filename" where the character " can be replaced by any suitable character. For most uses the double quote symbol will probably be the most convenient. The file name can include a full path. This takes input from the named file until it ends, when input reverts to the previous input. There is currently a limit of 20 on the depth of included files and macros. A suggested use of #include would be to define a set of macros which are part of the composer's style. It could also be used to provide repeated sections. s #include :section1: ;; Repeat that s #include :section1: Alternative methods of doing repeats, use the r statement, m statement, and n statement. Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April, 1998 (New in Csound version 3.48) Thanks to &nameluis; for pointing out the incorrect syntax in multiple file include statement.
manual_src~dfsg/score/ramp.xml0000644000000000000000000000510112262561504015536 0ustar rootroot
Ramping i statement pfields containing the symbol < will be replaced by values derived from linear interpolation of a time-based ramp. Ramps are anchored at each end by the first real number found in the same pfield of a preceding and following note played by the same instrument. E.g.: the statements scoreramping i1 0 1 100 i1 1 1 < i1 2 1 < i1 3 1 400 i1 4 1 < i1 5 1 0 will result in i1 0 1 100 i1 1 1 200 i1 2 1 300 i1 3 1 400 i1 4 1 200 i1 5 1 0 Ramps cannot cross a Section boundary. Ramps cannot be anchored by an np or pp symbol (although they may be referenced by these). Ramp symbols are illegal in p1, p2 and p3. Ramp symbols may be Carried. Note, however, that while the Carry feature will propagate ramp symbols through unsorted statements, the operation that interprets these symbols is acting on a time-warped and fully sorted version of the score. In fact, time-based linear interpolation is based on warped score-time, so that a ramp which spans a group of accelerating notes will remain linear with respect to strict chronological time. Starting with Csound version 3.52, using the symbols ( or ) will result in an exponential interpolation ramp, similar to expon. Using the symbol ˜ (a tilde) will result in uniform, random distribution between the first and last values of the ramp. Use of these functions must follow the same rules as the linear ramp function. ( ) ˜
manual_src~dfsg/score/statemnt.xml0000644000000000000000000000467512262561504016455 0ustar rootroot
Score Statements The statements used in scores are: a - Advance score time by a specified amount b - Resets the clock e - Marks the end of the last section of the score f - Causes a GEN subroutine to place values in a stored function table i - Makes an instrument active at a specific time and for a certain duration m - Sets a named mark in the score n - Repeats a section q - Used to quiet an instrument r - Starts a repeated section s - Marks the end of a section t - Sets the tempo v - Provides for locally variable time warping of score events x - Skip the rest of the current section { - Begins a non-sectional, nestable loop. } - Ends a non-sectional, nestable loop.
manual_src~dfsg/score/preproc.xml0000644000000000000000000001652012262561504016260 0ustar rootroot
Preprocessing of Standard Scores A Score (a collection of score statements) is divided into time-ordered sections by the s statement. Before being read by the orchestra, a score is preprocessed one section at a time. Each section is normally processed by 3 routines: Carry, Tempo, and Sort.
Carry Within a group of consecutive i statements whose p1 whole numbers correspond, any pfield left empty will take its value from the same pfield of the preceding statement. An empty pfield can be denoted by a single point (.) delimited by spaces. No point is required after the last nonempty pfield. The output of Carry preprocessing will show the carried values explicitly. The Carry Feature is not affected by intervening comments or blank lines; it is turned off only by a non- i statement or by an i statement with unlike p1 whole number. scorecarry Three additional features are available for p2 alone: +, ^+x, and ^-x. The symbol + in p2 will be given the value of p2 + p3 from the preceding i statement. This enables note action times to be automatically determined from the sum of preceding durations. The + symbol can itself be carried. It is legal only in p2. E.g.: the statements i1 0 .5 100 i . + i will result in i1 0 .5 100 i1 .5 .5 100 i1 1 .5 100 The symbols ^+x and ^-x determine the current p2 by adding or subtracting, respectively, the value of x from the preceding p2. These may be used in p2 only and are not carried like the + symbol. Note also that there should be no spaces following the ^, the +, or the - parts of these symbols -- the number must come directly after as in ^+2.3. If the example above had been i1 0 .5 100 i . ^+1 i . ^+1 the result would instead be i1 0 .5 100 i1 1 .5 100 i1 2 .5 100 The Carry feature should be used liberally. Its use, especially in large scores, can greatly reduce input typing and will simplify later changes. There can sometimes be circumstances where you do not want "missing" pfields after the last one entered to be implicitly carried. An example would be an instrument that is designed to take a variable number of pfields. Beginning with Csound 5.08, you can prevent the implicit carrying of pfields at the end of an i statement by using the symbol ! (called the "no-carry symbol"). The ! must appear at the end of an i statement and it cannot be used in p1, p2, or p3, since these pfields are required. Here is an example: i1 0 .5 100 i . + i . . . ! i This score would be interpreted as i1 0 .5 100 i1 .5 .5 100 i1 1 .5 ; no p4 i1 1.5 .5 ; only p1 to p3 are carried here
Tempo This operation time warps a score section according to the information in a t statement. The tempo operation converts p2 (and, for i statements, p3) from original beats into real seconds, since those are the units required by the orchestra. After time warping, score files will be seen to have orchestra-readable format demonstrated by the following: i p1 p2beats p2seconds p3beats p3seconds p4 p5 .... scoretempo
Sort This routine sorts all action-time statements into chronological order by p2 value. It also sorts coincident events into precedence order. Whenever an f statement and an i statement have the same p2 value, the f statement will precede. Whenever two or more i statements have the same p2 value, they will be sorted into ascending p1 value order. If they also have the same p1 value, they will be sorted into ascending p3 value order. Score sorting is done section by section (see s statement). Automatic sorting implies that score statements may appear in any order within a section. scoresort The operations Carry, Tempo and Sort are combined in a 3-phase single pass over a score file, to produce a new file in orchestra-readable format ( see the Tempo example). Processing can be invoked either explicitly by the Scsort command, or implicitly by Csound which processes the score before calling the orchestra. Source-format files and orchestra-readable files are both in ASCII character form, and may be either perused or further modified by standard text editors. User-written routines can be used to modify score files before or after the above processes, provided the final orchestra-readable statement format is not violated. Sections of different formats can be sequentially batched; and sections of like format can be merged for automatic sorting.
manual_src~dfsg/score/macros.xml0000644000000000000000000001574612262561504016103 0ustar rootroot
Score Macros Description Macros are textual replacements which are made in the score as it is being presented to the system. The macro system in Csound is a very simple one, and uses the characters # and $ to define and call macros. This can can allow for simpler score writing, and provide an elementary alternative to full score generation systems.The score macro system is similar to, but independent of, the macro system in the orchestra language. scoremacros macrosscore #define NAME -- defines a simple macro. The name of the macro must begin with a letter and can consist of any combination of letters and numbers. Case is significant. This form is limiting, in that the variable names are fixed. More flexibility can be obtained by using a macro with arguments, described below. #definescore $namescore #undefscore #define NAME(a' b' c') -- defines a macro with arguments. This can be used in more complex situations. The name of the macro must begin with a letter and can consist of any combination of letters and numbers. Within the replacement text, the arguments can be substituted by the form: $A. In fact, the implementation defines the arguments as simple macros. There may be up to 5 arguments, and the names may be any choice of letters. Remember that case is significant in macro names. $NAME. -- calls a defined macro. To use a macro, the name is used following a $ character. The name is terminated by the first character which is neither a letter nor a number. If it is necessary for the name not to terminate with a space, a period, which will be ignored, can be used to terminate the name. The string, $NAME., is replaced by the replacement text from the definition. The replacement text can also include macro calls. #undef NAME -- undefines a macro name. If a macro is no longer required, it can be undefined with #undef NAME. Syntax #define NAME # replacement text # #define NAME(a' b' c') # replacement text # $NAME. #undef NAME Initialization # replacement text # -- The replacement text is any character string (not containing a #) and can extend over mutliple lines. The replacement text is enclosed within the # characters, which ensure that additional characters are not inadvertently captured. Performance Some care is needed with textual replacement macros, as they can sometimes do strange things. They take no notice of any meaning, so spaces are significant. This is why, unlike the C programming language, the definition has the replacement text surrounded by # characters. Used carefully, this simple macro system is a powerful concept, but it can be abused. Another Use For Macros When writing a complex score it is sometimes all too easy to forget to what the various instrument numbers refer. One can use macros to give names to the numbers. For example #define Flute #i1# #define Whoop #i2# $Flute. 0 10 4000 440 $Whoop. 5 1 Examples Simple Macro A note-event has a set of p-fields which are repeated: #define ARGS # 1.01 2.33 138# i1 0 1 8.00 1000 $ARGS i1 0 1 8.01 1500 $ARGS i1 0 1 8.02 1200 $ARGS i1 0 1 8.03 1000 $ARGS This will get expanded before sorting into: i1 0 1 8.00 1000 1.01 2.33 138 i1 0 1 8.01 1500 1.01 2.33 138 i1 0 1 8.02 1200 1.01 2.33 138 i1 0 1 8.03 1000 1.01 2.33 138 This can save typing, and is makes revisions easier. If there were two sets of p-fields one could have a second macro (there is no real limit on the number of macros one can define). #define ARGS1 # 1.01 2.33 138# #define ARGS2 # 1.41 10.33 1.00# i1 0 1 8.00 1000 $ARGS1 i1 0 1 8.01 1500 $ARGS2 i1 0 1 8.02 1200 $ARGS1 i1 0 1 8.03 1000 $ARGS2 Macros with arguments #define ARG(A) # 2.345 1.03 $A 234.9# i1 0 1 8.00 1000 $ARG(2.0) i1 + 1 8.01 1200 $ARG(3.0) which expands to i1 0 1 8.00 1000 2.345 1.03 2.0 234.9 i1 + 1 8.01 1200 2.345 1.03 3.0 234.9 Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April, 1998 (New in Csound version 3.48)
manual_src~dfsg/score/top.xml0000644000000000000000000000201212262561504015377 0ustar rootroot The Standard Numeric Score The score section contains events that instatiate instruments from the orchestra. There are various score statements that enable complex score building within the csound language. Currently, the maximum length of the score is 231-1 control periods. For example, with kr=1500, you can run a score for a maximum of about 16.5 days before problems occur due to overflowing signed 32-bit integer variables. Note also that when using single precision floats (i.e. the 'f' installers rather than the 'd' ones), the accuracy of timing becomes worse after performing for a long time. &scorepreproc; &scorestatemnt; &scorenextp; &scoreramp; &scoremacros; &scoremult; &scoreeval; &scorestrings; manual_src~dfsg/score/strings.xml0000644000000000000000000000126112262561504016273 0ustar rootroot
Strings in p-fields You can pass a string as a p-field instead of a number, like this: i 1 0 10 "A4" The string can be received by the instrument and further processed using the string opcodes. Currently only one p-field can contain a string (i.e. no more than one string per line is allowed). You can overcome this using strset and strget.
manual_src~dfsg/score/eval.xml0000644000000000000000000001204712262561504015535 0ustar rootroot
Evaluation of Expressions In earlier versions of Csound the numbers presented in a score were used as given. There are occasions when some simple evaluation would be easier. This need is increased when there are macros. To assist in this area the syntax of arithmetic expressions within square brackets [ ] has been introduced. Expressions built from the operations +, -, *, /, % ("modulo"), and ^ ("power of") are allowed, together with grouping with ( ). Unary minus and plus are also supported. The expressions can include numbers, and naturally macros whose values are numeric or arithmetic strings. All calculations are made in floating point numbers. The usual precedence rules are followed when evaluating: expressions within parantheses ( ) are evaluated first and ^ is evaluated before *, /, and % which are evaluated before + and -. In addition to arithmetic operations, the following bitwise logical operators are also available: & (AND), | (OR), and # (XOR, exclusive-OR). These operators round their operands to the nearest (long) integer before evaluating. The logical operators have the same precedence as the *, /, and % arithmetic operators. Finally, the tilde symbol ˜ can be used in an expression wherever a number is permissible to use. Each ˜ will evaluate to a random value between zero (0) and one (1). Example r3 CNT i1 0 [0.3*$CNT.] i1 + [($CNT./3)+0.2] e As the three copies of the section have the macro $CNT. with the different values of 1, 2 and 3, this expands to s i1 0 0.3 i1 0.3 0.533333 s i1 0 0.6 i1 0.6 0.866667 s i1 0 0.9 i1 0.9 1.2 e This is an extreme form, but the evaluation system can be used to ensure that repeated sections are subtly different. Here are some simple examples of each operator: i1 0 1 [ 110 + 220 ] ; evaluates to 330 i1 + . [ 330 - 55 ] ; 275 i1 + . [ 44 * 10 ] ; 440 i1 + . [ 1100 / 2 ] ; 550 i1 + . [ 5 ^ 4 ] ; 625 i1 + . [ 5660 % 1000 ] ; 660 i1 + . [ 110 & 220 ] ; 76 i1 + . [ 110 | 220 ] ; 254 i1 + . [ 110 # 220 ] ; 178 i1 + . [~] ; random between 0-1 i1 + . [~ * 4 + 1] ; random between 1-5 i1 + . [~ * 95 + 5] ; random between 5-100 i1 + . [ 8 / 2 * 3 ] ; 12 i1 + . [ 4 + 3 - 2 + 1 ] ; 6 i1 + . [ 4 + 3 * 2 + 1 ] ; 11 i1 + . [(4 + 3)*(2 + 1)] ; 21 i1 + . [ 2 * 2 & 3 ] ; 4 i1 + . [ 3 & 2 * 2 ] ; 0 i1 + . [ 4 | 3 * 3 ] ; 13 The @ operator New in Csound version 3.56 are @x (next power-of-two greater than or equal to x) and @@x (next power-of-two-plus-one greater than or equal to x). @ @@ [ @ 11 ] will evaluate to 16 [ @@ 11 ] to 17 Credits Author: &namejohn; University of Bath/Codemist Ltd. Bath, UK April, 1998 (New in Csound version 3.48)

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