Table of Contents
Table of Contents
by Barry L. Vercoe, MIT Media Lab
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-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.
Copyright (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 Piché of the Faculté de musique de l'Université de Montréal. A Print Edition, in Adobe Acrobat format, was then maintained by David M. 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.
In addition to the core code developed by Barry L. Vercoe at M.I.T., a large part of the Csound code was modified, developed and extended by an independent group of programmers, composers and scientists. Copyright to this code is held by the respective authors:
Table 1. Contributors
Mike Berry |
Eli Breder |
Andres Cabrera |
Michael Casey |
Michael Clark |
Perry Cook |
Sean Costello |
Rasmus Ekman |
Richard Dobson |
Mark Dolson |
Dan Ellis |
Tom Erbe |
John ffitch |
Bill Gardner |
Michael Gogins |
Matt Ingalls |
Richard Karpen |
Victor Lazzarini |
Allan Lee |
David Macintyre |
Gabriel Maldonado |
Max Mathews |
Hans Mikelson |
Peter Neubäcker |
Peter Nix |
Jean Piché |
Ville Pulkki |
John Ramsdell |
Marc Resibois |
Rob Shaw |
Paris Smaragdis |
Greg Sullivan |
Istvan Varga |
Bill Verplank |
Robin Whittle |
Steven Yi |
The official manual was compiled from the canonical Csound Manual sources maintained by John ffitch, Richard Boulanger, Jean Piché, Peter Nix, and David M. Boothe. The Alternative Csound Reference Manual was maintained by Kevin Conder. The Canonical Csound Reference Manual is maintained by the Csound community.
This manual is a product of the Csound community. The current version of the manual is based on the Alternative Csound Reference Manual, developed by Kevin Conder using DocBook/SGML. This was in itself based on the Official Csound Reference Manual (last known address: http://www.lakewoodsound.com/csound/hypertext/manual.htm), which was maintained by David M. Boothe.
In the winter of 2004, the manual was converted to DocBook/XML by Steven Yi to allow for more people to be able to compile and maintain the manual. 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.
Written by Steven Yi, January 2005.
Csound is copyright 1991-2005 by Barry Vercoe and John ffitch.
CsoundVST is copyright 2001-2005 by Michael Gogins.
Csound and 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 CsoundVST 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 CsoundVST; if not, write to the Free Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
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 doc/manual/copying.txt file.
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 2003 by Kevin Conder for modifications made to 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 David M. 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.
The Csound and CsoundVST Manual's network location is http://sourceforge.net/projects/csound.
Virtual Synthesis Technology (VST) PlugIn interface technology by Steinberg Soft- und Hardware GmbH.
CsoundVST source code contains modified versions of source code files from the VST SDK distributed by Steinberg. These files are to be used only for building CsoundVST. You are not licensed to use these files for any other purpose. If you make a derived product based on CsoundVST or the modified VST source files herein, you must apply to Steinberg for your own license to use the VST SDK.
Table of Contents
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.
To make music with Csound:
CsoundVST is an extended version of Csound that adds a graphical user interface, C++ and Python APIs, Python scripting, a library of Python extension modules for algorithmic composition, a VST plugin interface, and a Mathematica interface.
In addition to this "canonical" version of Csound and CsoundVST, there are other versions of Csound and other front ends for Csound, many of which can be found at http://csounds.com.
In the time since Barry Vercoe wrote the original Preface to this manual, printed above, many further contributions have been made to Csound. The current stable version of Csound is 4.23. Csound 5 is the next version, currently in early beta status. CsoundVST is an extended version of both Csound 4 and Csound 5.
Csound 5 begins a new major version of Csound that includes the following new features:
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.
CsoundVST is an extended version of Csound that runs both as a shared library (as a VST plugin or as an embedded synthesizer) and as a standalone program. Its main purposes are (a) to make it easier to extend Csound (e.g. the Loris plugin opcodes with their Python scripting), and (b) to streamline the actual use of Csound in composing, particularly for algorithmic composition, by integrating more tightly with other languages and other software.
Csound's "home page" is maintained by Richard Boulanger at http://csounds.com.
The Csound source code is maintained by John ffitch at http://www.sourceforge.net/projects/csound. Precompiled packages for some platforms also can be downloaded from that site.
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. Posts sent to csound@lists.bath.ac.uk go to all subscribed members of the list.
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.
Suspected bugs in the code may be entered using the bug tracking system at http://www.cs.bath.ac.uk/cgi-bin/csound.
Csound can either be built from source code, or installed from a packaged archive or installer program. This section describes how to install from an archive or installer program.
Archives and installers containing Csound binaries can be found at http://sourceforge.net/projects/csound, on the Files page, in various packages.
The most complete distribution can be found in the csoundvst package, which contains pre-built binaries for Csound 5 and CsoundVST on Windows, made with the MinGW compiler, as well as complete sources and SCons build system for all platforms. To install from the archive, unzip it into a csound5 directory on your computer, and configure it as explained below.
The csoundvst package also contains a Windows installer with the same Windows binaries, examples, and documentation. To install using the installer, simply execute it and follow the instructions provided by the installer.
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.
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 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.
On Windows, make sure the directory or directories (normally the csound5 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 the binaries archive is unpacked in C:\ you can use (otherwise set the paths accordingly):
set OPCODEDIR=C:\csound5\plugins set OPCODEDIR64=C:\csound5\plugins64 set PATH=%PATH%;C:\csound5\bin
If you get a pop-up about the missing Python library (python24.dll) and don't need the python opcodes, just delete csound5\plugins\py.dll and csound5\plugins64\py.dll, and the pop-up about the missing Python library should be gone.
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.
CsoundVST requires some additional configuration. On all platforms, CsoundVST requires that you have Python installed on your computer. The directory containing the _CsoundVST shared library and the CsoundVST.py file must be in your PYTHONPATH environment variable, so that the Python runtime knows how to load these files.
Assuming that you have installed and configured the software, Csound and CsoundVST can be operated in a variety of modes and configurations. The .csd and .py files in the examples directory demonstrate a few of these modes of operation. Some of these scores are simple, others are moderately complex.
You may need to edit the SoundFont file paths in instrument definitions that use the fluid SoundFont 2 player opcode to match your environment.
For real-time audio output, with or without MIDI control, you will probably want to tune the kr and ksmps orchestra statements, and the -b and -B command-line options, to give you the shortest possible latency that does not cause clicks or stutters in Csound's audio output.
In general, -b (Csound's audio buffer) should be set to a small power of 2 (such as 64 or 128), and -B (the audio driver's buffer) should be set to 2 or 4 times that.
Currently, with -B set to 512, audio output latency is about 12 milliseconds, fast enough for reasonably responsive keyboad playing.
Even shorter latencies, as low as 3 milliseconds on some systems, are feasible.
If your sound card does not have an ASIO driver, you can still use Csound with ASIO by downloading and installing the asio4all adapter from http://www.asio4all.com.
To enable the JACK plugin, use this command line option:
-+rtaudio=jack
Additionally, there are some command line options specific to JACK:
JACK Command-line Flags
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.
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)
As of Csound version 5.01, this option is deprecated and ignored.
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).
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.
The original method for running Csound was as a console program. This, of course, still works. Running csound without any arguments prints out a list of command-line options, which are more fully explained below. Normally, the user executes something like csound -W -omysoundfile myorchestra.orc myscore.sco or, to use the single-file Csound structured data (.csd) format, csound myscore.csd.
Csound can read and write soundfiles (off-line rendering), read and write digital audio using a sound card (real-time rendering), read and write MIDI files, and read and write MIDI using a MIDI interface and controller (real-time control).
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, or as a VST plugin in hosts such as the Cubase audio sequencer. CsoundVST provides both a C++ and a Python API to Csound, and to a set of classes for algorithmic composition.
CsoundVST contains a built-in Python interpreter. With Python, the user can generate a score, import a MIDI file, process notes, load and run a Csound orchestra, and in general do anything that can be done either with Csound or in Python.
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.
You can use CsoundVST as a Python extension module. In fact, you can do this either in a standard Python interpreter, such as Python command line or the Idle Python GUI, or in CsoundVST itself in Python mode.
To use CsoundVST in a standard Python interpreter, import CsoundVST.
import CsoundVST
The CsoundVST 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 CsoundVST >>> 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 >>>
To use CsoundVST itself as your Python interpreter, click on the CsoundVST Settings tab, and select the Python check box in the Csound performance mode box. Do not create a new CppSound object; you must use the builtin csound object in the CsoundVST module.
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()
To run your script in Csound VST, click on the Perform button.
The following instructions are for Cubase SX. You would follow roughly similar procedures in other hosts.
Use the Devices menu, Plug-In Information dialog, VST Plug-Ins tab, Shared VST Plug-ins Folder text field to add your csound5 directory to Cubase's plugin path. You can have multiple directories separated by semicolons.
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/CsoundVST/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 temp.orc temp.sco
Click on the VST Instruments dialog's on/off button to turn it on. This should compile the Csound orchestra. Note: If you don't compile the orchestra, you won't be able to assign the plugin to a track.
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. However, you can't have more than one CsoundVST plugin in the same song!
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://sourceforge.net/projects/csound.
The latest Csound source code is available through the Concurrent Versions System (CVS)(http://www.cvshome.org). To download Csound sources using CVS, run the following commands:
cvs -d:pserver:anonymous@cvs.sourceforge.net:/cvsroot/csound login cvs -z3 -d:pserver:anonymous@cvs.sourceforge.net:/cvsroot/csound co csound5
Information about accessing the CVS repository may be found in the SourceForge document Basic Introduction to CVS and SourceForge.net (SF.net) Project CVS Services.
If you wish to become a Csound developer, first obtain a SourceForge login, and then apply to John ffitch at the http://www.sourceforge.net/projects/csound site.
The procedure for building Csound 5 is briefly and incompletely outlined here.
The manual is built using make. Scripts are used for a few other tasks. However, this section focuses on the main Csound build system, which uses SCons, a Python program that replaces make for cross-platform configuration and building.
(Alternatively, for building a minimal version of Csound 5 (API library compiled as DLL, plugin libraries, and command line frontend) on Windows with MinGW/MSYS, you may alternatively edit and use Makefile-win32, eliminating the dependencies on Python and SCons.)
All Csound 5 SCons builds require the following:
Optional configurations can include the following. In most cases it is best to install the most recent stable versions.
Execute scons -h to discover the current configuration options.
Modify custom.py as required for your installation (usually required on Windows, may not be required on Linux).
Execute scons with the options you desire.
Set the environment variable OPCODEDIR to the directory where plugin libraries are installed; in the case of a double precision build, OPCODEDIR64 should be set instead. The NSIS installer performs this step.
To install on Linux, execute ./install.py or scons install.
To create a Windows installer, build Csound for double precision samples and including the Loris, STK, py, vst4cs, and Fluidsynth opcodes, build the manual, install the NSIS installer from http://nsis.sourceforge.net, and run csound5/installer/windows/csound.nsi.
This is a “to do” list, not necessarily complete, and in no particular order of priority or time, for Csound and CsoundVST:
Table of Contents
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.
dssi4cs enables the use of DSSI and LADSPA plugin effects and synthesizers within Csound on Linux. The following opcodes are available:
See the entry for dssiinit for a usage example.
vst4cs enables the use of VST plugin effects and synthesizers within Csound. The following opcodes are available:
vst4cs currently works for Windows only.
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 lorisout, 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.
Note that a version of Loris with a Python interface is packaged as part of the CsoundVST distribution, so it is possible to perform both analysis and synthesis with Loris in Csound 5.
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
.Example 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, 3.0 ; 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
Example 2.
; 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
Example 3.
; 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) ktcl linseg 0, ionset, .2, itmorph, 2.0, idecay, 2.1 ; clarinet time function, morph from .2 to 2.0 seconds ktfl linseg 0, ionset, .5, itmorph, 2.1, idecay, 2.3 ; flute time function, morph from .5 to 2.1 seconds 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
Example 4.
; 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
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.
OSC enables interaction between different audio processes, and in particular between Csound and other synthesis engines. The following opcodes are available:
Using the Python opcode family, you can interact with a Python interpreter embedded in Csound in five ways:
and you can do any of these things:
...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 prefix]
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
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 for controlling (at krate) 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.
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.
![]() | Note |
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String variables and related opcodes are not available in Csound versions older than 5.00. |
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
These opcodes perform operations on string variables:
Table of Contents
Public Csound is available for download from :
ftp://ftp.cs.bath.ac.uk/pub/dream/newest/
This Hypertext Edition of the manual, as well as the Print Edition, in Adobe Acrobat format (.pdf) are available for browser download from:
Detailed instructions for installing and configuring Csound on a Linux system may be obtained from:
http://www.csounds.com/secondprinting/cdroms/installing/linux/
Detailed instructions for installing and configuring Csound on Macintosh systems may be obtained from:
Detailed instructions for installing and configuring Csound on a MS-DOS or Windows 95/NT system may be obtained from:
Detailed instructions for installing and configuring Csound on a Windows 95, Windows 98, or Windows 2000 system may be obtained from:
For information on availability of Csound for other platforms, see The Csound FrontPage:
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
Posts sent to csound@lists.bath.ac.uk go to all subscribed members of the list.
Suspected bugs in the code may be entered using the bug tracking system at http://www.cs.bath.ac.uk/cgi-bin/csound.
Csound is a command for passing anorchestra file andscore file to Csound to generate a soundfile. 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.
With some recent additions to Csound, there are now three places (and in some cases four) 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 last assignment of an option will override any earlier ones. As of version 5.01, sample and control rate overrides (-r and -k flags) specified anywhere override sr, kr, and ksmps in the orchestra header.
Flags 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 (-L, -M 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.
Many flags are generic Csound command-line flags. Various platform implementations may not react the same way to different flags!
Listed first are the traditional flags present in Csound 4. Look further down for Csound 5 specific flags.
The format of a command is either:
csound [-flags] [orchname] [scorename]
or
csound [-flags] [csdfilename]
where the arguments are of 2 types: flags arguments (beginning with a “-”), and name arguments (such as filenames). Certain flag arguments take a following name or numeric argument.
Command-line Flags
Provide an extended command-line in file “FILE”
Use 24-bit audio samples.
Use 8-bit unsigned character audio samples.
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.
Write an AIFF format soundfile. Use with the -c, -s, -l, or -f flags.
Use a-law audio samples.
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.
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.
Use Cscore processing of the scorefile.
Use 8-bit signed character audio samples.
Defer GEN01 soundfile loads until performance time.
Suppress all displays.
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:
Note that this optimization (due to technical reasons) is not performed on i-rate temporary variables.
![]() | Warning |
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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. |
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.
Use single-format float audio samples (not playable on some systems, but can be read by -i, soundin and GEN01
Suppress graphics, use PostScript displays instead.
Suppress graphics, use ASCII displays instead.
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.
No header on output soundfile. Don't write a file header, just binary samples.
Display on-line help message.
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.
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.
Write an IRCAM format soundfile.
Currently disabled. Use database FILE for messages to print to console during performance. In Csound 5.00 and later versions, the localization of messages is controlled by two environment variables, both of which are optional. CSSTRNGS points to a directory containing .xmg files, and CS_LANG selects a language.
Do not generate any PEAK chunks.
Override the control rate (KR) supplied by the orchestra.
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.
Use long integer audio samples.
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.
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
32 = dB, no colors
64 = dB, out of range highlighted with red
96 = dB, all colors
The default is 135 (128+4+2+1), which means all messages, raw amplitude values, and printing elapsed time at the end of performance.
Save MIDI output to a file (Csound 5.00 and later only).
Notify (ring the bell) when score or MIDI track is done.
No sound. Do all processing, but bypass writing of sound to disk. This flag does not change the execution in any other way.
Log output to file FILE.
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 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.
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.
Continually rewrite the header while writing the soundfile (WAV/AIFF).
Override the sampling rate (SR) supplied by the orchestra.
Use short integer audio samples.
Linux only. Use real-time scheduling and lock memory. (Also requires -d and either -o dac or -o devaudio). See also --sched=N below for Csound 5.00 and later.
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).
Terminate the performance when the end of MIDI file is reached.
Prevents Csound from deleting the sorted score file, score.srt, upon exit.
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.
Invoke the utility program UTILITY. Use any invalid name to list the available utilities.
Use u-law audio samples.
Verbose translate and run. Prints details of orch translation and performance, enabling errors to be more clearly located.
Write a WAV format soundfile.
Extract a portion of the sorted score, score.srt, using the extract file FILE (see Extract).
Switch on dithering of audio conversion from internal floating point to 32, 16 and 8-bit formats.
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>
Csound5 Command-line Flags
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).
Enables displays, reverting the effect of any previous -d flag.
Disables expression optimization.
(max. length = 200 characters) Artist tag in output soundfile (no spaces)
(max. length = 200 characters) Comment tag in output soundfile (no spaces)
(max. length = 200 characters) Copyright tag in output soundfile (no spaces)
(max. length = 200 characters) Date tag in output soundfile (no spaces)
(max. length = 200 characters) Software tag in output soundfile (no spaces)
(max. length = 200 characters) Title tag in output soundfile (no spaces)
(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.
Enable message attributes (colors etc.); might need to be disabled on some terminals which print strange characters instead of modifying text attributes. default: true.
(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).
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.
(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.
(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).
(min: 0) Start playback at the specified time (in seconds), skipping earlier events in the score and MIDI file.
Set orchestra macro XXX to value YYY
Set score macro XXX to value YYY
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.
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.
The Unified 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.
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.
The Csound Element is used to alert the csound compiler to the .csd format. The file must begin with the start tag <CsoundSynthesizer>. The last line of the file must be the end tag </CsoundSynthesizer>. The remaining elements are defined below.
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.
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>.
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. New in Csound version 4.07. 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. New in Csound version 4.07. Using this tag is not recommended; use <CsFileB> instead.
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:
#.#
See example below. New in Csound version 4.09.
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>
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.
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 is also optional. The default values for missing i, f or t are:
all instruments, beginning of score, end of score.
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
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.
Comments are optional and are for the purpose of letting the user document his orchestra code. Comments always begin with a semicolon (;) and extend to the end of the line.
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)
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.
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 last 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, then SSDIR, then SFDIR.
Analysis control files for reading are sought in the current directory, then SADIR.
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.
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."
An orchestra program in Csound is comprised of orchestra header statements which set various global parameters, followed by a number of instrument blocks representing different instrument types. An instrument block, in turn, is comprised of ordinary statements that set values, control the logical flow, or invoke the various signal processing subroutines that lead to audio output.
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.
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.
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.).
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 eight forms of local and global variables:
Table 1. 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 | 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 |
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.
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.
Statements that are normally placed in an orchestra header are ctrlinit, ftgen, kr, ksmps, massign, nchnls, pgmassign, pset, seed, sr, and strset.
Statements that define an instrument block are endin and instr.
Statements that define a user defined opcode block are endop and opcode.
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.
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 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
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 |
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|
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.
; ---- 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
Written by Gabriel Maldonado (http://csounds.com/maldonado)
Widgets allow the design of a custom Graphical User Interface to control an orchestra in real-time. They are derived from the open-source library FLTK (Fast Light Tool Kit). Such 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
Valuators
Other widgets
Containers are widgets that contain other widgets such as panels, windows, etc. Csound provides the following container objects:
Panels
Scroll areas
Pack
Tabs
Groups
The most useful objects are named 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
There are other widgets that are not valuators nor containers:
Buttons
Button banks
Labels
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.
At last, there are three important opcodes that allow the following actions:
Running the widget thread.
Loading snapshots containing the status of all valuators of an orchestra.
Saving snapshots containing the status of all valuators of an orchestra.
Here is an example preview of Csound code for a window containing a valuator. 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, externally to 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.
Here is an example of creating a window:
;******************************* 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 ;*******************************
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:
;******************************* 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 ;put some synthesis code here ; gk1 and gk2 variables that contain the output of valuator ; widgets previously defined, can be used inside any instrument endin ;*******************************
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.
Since FLTK toolkit is still in evolution process, opcode syntax provided in Csound could be modified in future version. This could cause some incompatibilities between orchestras of a determinate version. However it should not be hard to modify early orchestras in order to make them compatible with later versions.
For more information, see the following sections.
The opcodes for FTLK containers are FLgroup, FLgroupEnd, FLpack, FLpackEnd, FLpanel, FLpanelEnd, FLscroll, FLscrollEnd, FLtabs, and FLtabsEnd.
Other FLTK widget opcodes are FLbox, FLbutBank, FLbutton, FLprintk, FLprintk2, and FLvalue,
Opcodes one can use to modify FLTK widget appearance are FLcolor2, FLcolor, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor2, FLsetColor, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal, FLsetVal_i, and FLshow.
The general FLTK widget-related opcodes are FLgetsnap, FLloadsnap, FLrun, FLsavesnap, FLsetsnap, and FLupdate.
The opcode for the FLTK slider bank is FLslidBnk.
The opcodes one can use to create score events from within a orchestra are event, event_i, schedule, schedwhen, and schedkwhen.
The opcodes one can use to create, call, or undefine macros are #define, $NAME, #ifdef, #end, #include, and #undef.
The opcodes to manipulate which orchestra statements are executed are cggoto, cigoto, ckgoto, cngoto, elseif, else, endif, goto, if, igoto, kgoto, loop_ge, loop_gt, loop_le, loop_lt, tigoto, and timout.
Opcodes that read from signals or on-screen controls are button, checkbox, control, follow, follow2, peak, pitch, pitchamdf, sense, sensekey, setctrl, tempest, tempo, tempoval, setime, trigger, trigseq, and xyin.
These opcodes let one define and use a sub-instrument: subinstr, and subinstrinit.
Opcodes that perform mathematical functions are abs, exp, frac, int, log, log10, logbtwo, powoftwo, and sqrt.
The following are further mathematical functions available in Csound5: ceil, floor, and round.
Opocodes that accept MIDI input are aftouch, chanctrl, ctrl7, ctrl14, ctrl21, initc7, initc14, initc21, midic7, midic14, midic21, midichannelaftertouch, midichn, midicontrolchange, mididefault, midinoteoff, midinoteoncps, midinoteonkey, midinoteonoct, midinoteonpch, midipitchbend, midipolyaftertouch, midiprogramchange, and polyaft.
Opcodes that convert MIDI values are ampmidi, cpsmidi, cpsmidib, cpstmid, midictrl, notnum, octmidi, octmidib, pchbend, pchmidi, pchmidib, and veloc.
Opcodes to turn MIDI notes on or off are midion, midion2, moscil, noteoff, noteon, noteondur, and noteondur2.
The FM synthesis opcodes are fmb3, fmbell, fmmetal, fmpercfl, fmrhode, fmvoice, fmwurlie, foscil, and foscili,
The granular synthesis opcodes are fof, fof2, fog, grain, grain2, grain3, granule, sndwarp, sndwarpst, and syncgrain.
The opcodes that generate linear or exponential curves or segments are adsr, expon, expseg, expsega, expsegr, jspline, line, linseg, linsegr, loopseg, lpshold, madsr, mxadsr, rspline, transeg, and xadsr.
The linear predictive coding resynthesis opcodes are lpfreson, lpinterp, lpread, lpreson, and lpslot.
The opcodes that model or emulate the sounds of other instruments are bamboo, cabasa, crunch, dripwater, gogobel, guiro, lorenz, mandol, marimba, moog, planet, sandpaper, sekere, shaker, sleighbells, stix, tambourine, vibes, and voice.
Opcodes that generate random numbers are betarnd, bexprnd, cauchy, cuserrnd, duserrnd, exprand, gauss, linrand, noise, pcauchy, pinkish, poisson, rand, randh, randi, rnd31, random, randomh, randomi, trirand, unirand, urd, and weibull.
Opcodes that implement sample playback are bbcutm, bbcuts, loscil, loscil3, lphasor, lposcil, lposcil3, sfilist, sfinstr, sfinstr3, sfinstr3m, sfinstrm, sfload, sfpassign, sfplay, sfplay3, sfplay3m, sfplaym, sfplist, sfpreset, and waveset.
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, and xscanu.
![]() | Use of PVOC-EX files with the old Csound pvoc opcodes |
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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 ktableseg, pvadd, pvbufread, pvcross, pvinterp, pvoc, pvread, tableseg, tablexseg, and vpvoc.
The opcode that uses wave terrain synthesis is wterrain.
The opcodes for file input and output are clear, dumpk, dumpk2, dumpk3, dumpk4, fiopen, fin, fini, fink, fout, fouti, foutir, foutk, readk, readk2, readk3, readk4, and vincr.
The opcodes that receive audio signals are: chani, chnget, diskin, diskin2, in, in32, inch, inh, ino, inq, ins, invalue, inx, inz, and soundin.
The opcodes that write audio signals are: chano, chnset, fout, out, out32, outc, outch, outh, outo, outq, outq1, outq2, outq3, outq4, outs, outs1, outs2, outvalue, outx, outz, soundout, and soundouts.
Opcodes for printing and displaying values are dispfft, display, flashtxt, print, printk, printk2, and printks.
The opcodes that query information about files are filelen, filenchnls, filepeak, and filesr.
The opcodes that convolve and morph signals are convle, convolve, cross2, dconv, ftconv, ftmorf, and pconvolve.
The opcodes that implement delay are delay, delay1, delayr, delayw, deltap, deltap3, deltapi, deltapn, deltapx, deltapw, multitap, vdelay, vdelay3, vdelayx, vdelayxs, vdelayxq, vdelayxw, vdelayxwq, and vdelayxws.
The opcodes that one can use for panning and spatialization are hrtfer, locsend, locsig, pan, space, spat3d, spat3di, spat3dt, spdist, spsend, vbap16, vbap16move, vbap4, vbap4move, vbap8, vbap8move, vbaplsinit, vbapz, and vbapzmove.
The opcodes one can use for reverberation are alpass, babo, comb, freeverb, nestedap, nreverb, reverb2, reverb, reverbsc, valpass, and vcomb.
The opcodes one may use to modify signals are a, diff, downsamp, fold, i, integ, interp, ntrpol, samphold, and upsamp.
Opcodes that generate special effects are distort1, flanger, harmon, jitter, jitter2, phaser1, phaser2, vibr, and vibrato.
The opcodes for standard filters are areson, aresonk, atone, atonek, atonex, biquad, biquada, butbp, butbr, buthp, butlp, butterbp, butterbr, butterhp, butterlp, clfilt, filter2, fofilter, hilbert, lineto, lowpass2, lowres, lowresx, lpf18, moogvcf, moogladder, port, portk, reson, resonk, resonr, resonx, resony, resonz, rezzy, statevar, svfilter, tbvcf, tlineto, tone, tonek, tonex, vlowres, and zfilter.
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.
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 pvsdemix, pvscale, pvshift, pvsmix, pvsfilter, pvsblur, pvstencil, pvsarp, pvsvoc
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
Vectorial opocodes support read/write access to arrays of vectors (or arrays of arrays).
vtablei, vtablek, vtablea, vtablewi, vtablewk, vtablewa, vtabi, vtabk, vtaba, vtabwi, vtabwk and vtabwa.
Author: Gabriel Maldonado
Originally available on CsoundAV.
Added to csound5.
These opcodes 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.
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 and vexp.
Author: Gabriel Maldonado
Originally available on CsoundAV.
Added to csound5.
These opcodes perform operations between 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 vector that overrides the old values of ifn1.
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 bmscan, vcella, adsynt, adsynt2 etc.
Operations Between two Vectorial Signals: vaddv, vsubv, vmultv, vdivv, vpowv, vexpv, vcopy, vcopy_i and vmap.
Author: Gabriel Maldonado
Originally available on CsoundAV.
Added to csound5.
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.
Operations Between a Vectorial and a Scalar Signal: vlinseg and vexpseg.
Author: Gabriel Maldonado Originally available on CsoundAV. Added to csound5.
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 work at k-rate.
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 two Vectorial Signals: vlimit, vwrap and vmirror.
Author: Gabriel Maldonado
Originally available on CsoundAV.
Added to csound5.
Vectorial Control-rate Delay Paths: vdelayk, vport and vecdelay.
Author: Gabriel Maldonado
Originally available on CsoundAV.
Added to csound5.
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.
Author: Gabriel Maldonado
Originally available on CsoundAV.
Added to csound5.
Cellular automata vectors:
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.
Opcodes for the zak patch system are zacl, zakinit, zamod, zar, zarg, zaw, zawm, zir, ziw, ziwm, zkcl, zkmod, zkr, zkw, and zkwm.
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.
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.
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.
The Carry feature should be used liberally. Its use, especially in large scores, can greatly reduce input typing and will simplify later changes.
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 ....
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.
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.
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.
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.
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
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. The symbols { and } to define an exponential ramp have been deprecated. Using the symbol ˜ 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.
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.
#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.
$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.
# 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.
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
Example 1. 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
Example 2. 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 $ARG(3.0)
which expands to
i1 0 1 8.00 1000 2.345 1.03 2.0 234.9 i1 + 1 8.01 1200 2.345 1.03 3.0 234.9
It is sometimes convenient to have the score in more than one file. This use is supported by the #include facility which is part of the macro system. A line containing the text
#include "filename"
where the character " can be replaced by any suitable character. For most uses the double quote symbol will probably be the most convenient. The file name can include a full path.
This takes input from the named file until it ends, when input reverts to the previous input. There is currently a limit of 20 on the depth of included files and macros.
A suggested use of #include would be to define a set of macros which are part of the composer's style. It could also be used to provide repeated sections.
s #include :section1: ;; Repeat that s #include :section1:
Alternative methods of doing repeats, use the r statement, m statement, and n statement.
In earlier versions of Csound the numbers presented in a score were used as given. There are occasions when some simple evaluation would be easier. This need is increased when there are macros. To assist in this area the syntax of an arithmetic expressions within square brackets [ ] has been introduced. Expressions built from the operations +, -, *, /, %, and ^ are allowed, together with grouping with ( ). The expressions can include numbers, and naturally macros whose values are numeric or arithmetic strings. All calculations are made in floating point numbers. Note that unary minus is not yet supported.
New in Csound version 3.56 are @x (next power-of-two greater than or equal to x) and @@x (next power-of-two-plus-one greater than or equal to x).
r3 CNT i1 0 [0.3*$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.
The GEN routines that generate sine or cosine values are GEN09, GEN10, GEN11, GEN19, GEN30, GEN33, and GEN34.
GEN routines that generate tables with linear or exponential segments are GEN05, GEN06, GEN07, GEN08, GEN16, GEN25, and GEN27.
The GEN routine for window functions is GEN20.
Table of Contents
In the above conditionals, a and b are first compared. If the indicated relation is true (a greater than b, a less than b, a greater than or equal to b, a less than or equal to b, a equal to b, 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. (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.
Here is an example of the != opcode. It uses the files notequal.orc and notequal.sco.
Example 1. Example of the != opcode.
/* notequal.orc */ ; 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 not 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 /* notequal.orc */
/* notequal.sco */ ; 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 /* notequal.sco */
Its output should include lines like this:
k1 = 2.000000, k2 = 1.000000 k1 = 3.000000, k2 = 0.000000 k1 = 4.000000, k2 = 1.000000
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.
# 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.
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.
Here is a simple example of the defining a macro. It uses the files define.orc and define.sco.
Example 2. Simple example of the define macro.
/* define.orc */ ; 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.orc */
/* define.sco */ ; 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 /* define.sco */
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 files define_args.orc and define_args.sco.
Example 3. Example of the define macro with arguments.
/* define_args.orc */ ; 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_args.orc */
/* define_args.sco */ ; 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 /* define_args.sco */
Its output should include lines like this:
Macro definition for OSCMACRO
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. There is currently 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.
Here is an example of the include opcode. It uses the files include.orc, include.sco, and table1.inc.
Example 4. Example of the include opcode.
/* include.orc */ 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.orc */
/* table1.inc */ ; Table #1, a sine wave. f 1 0 16384 10 1 /* table1.inc */
/* include.sco */ ; Include the file for Table #1. #include "table1.inc" ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* include.sco */
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.
# 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.
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.
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.
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.
# 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.
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.
Here is an example of the calling a macro. It uses the files define.orc and define.sco.
Example 6. An example of the calling a macro.
/* define.orc */ ; 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.orc */
/* define.sco */ ; 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 /* define.sco */
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 files define_args.orc and define_args.sco.
Example 7. An example of the calling a macro with arguments.
/* define_args.orc */ ; 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_args.orc */
/* define_args.sco */ ; 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 /* define_args.sco */
Its output should include a line like this:
Macro definition for OSCMACRO
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 i
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 that of the absolute value of b, by repeated subtraction. This is the same as modulus function in integers. New in Csound version 3.50.
Here is an example of the % operator. It uses the files modulus.orc and modulus.sco.
Example 8. Example of the % operator.
/* modulus.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = 5 % 3 print i1 endin /* modulus.orc */
/* modulus.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* modulus.sco */
Its output should include a line like this:
instr 1: i1 = 2.000
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 i
is taken as
(a - b) - c
Parentheses may be used as above to force particular groupings.
In the above conditionals, a and b are first compared. If the indicated relation is true (a greater than b, a less than b, a greater than or equal to b, a less than or equal to b, a equal to b, 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. (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.
Here is an example of the > opcode. It uses the files greaterthan.orc and greaterthan.sco.
Example 9. Example of the > opcode.
/* greaterthan.orc */ ; 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 /* greaterthan.orc */
/* greaterthan.sco */ ; 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 /* greaterthan.sco */
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
In the above conditionals, a and b are first compared. If the indicated relation is true (a greater than b, a less than b, a greater than or equal to b, a less than or equal to b, a equal to b, 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. (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.
Here is an example of the >= opcode. It uses the files greaterequal.orc and greaterequal.sco.
Example 10. Example of the >= opcode.
/* greaterequal.orc */ ; 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 /* greaterequal.orc */
/* greaterequal.sco */ ; 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 /* greaterequal.sco */
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
In the above conditionals, a and b are first compared. If the indicated relation is true (a greater than b, a less than b, a greater than or equal to b, a less than or equal to b, a equal to b, 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. (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.
Here is an example of the < opcode. It uses the files lessthan.orc and lessthan.sco.
Example 11. Example of the < opcode.
/* lessthan.orc */ ; 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 less 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 /* lessthan.orc */
/* lessthan.sco */ ; 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 /* lessthan.sco */
Its output should include lines like this:
k1 = 2.000000, k2 = 1.000000 k1 = 3.000000, k2 = 0.000000 k1 = 4.000000, k2 = 0.000000
In the above conditionals, a and b are first compared. If the indicated relation is true (a greater than b, a less than b, a greater than or equal to b, a less than or equal to b, a equal to b, 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. (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.
Here is an example of the <= opcode. It uses the files lessequal.orc and lessequal.sco.
Example 12. Example of the <= opcode.
/* lessequal.orc */ ; 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 less 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 /* lessequal.orc */
/* lessequal.sco */ ; 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 /* lessequal.sco */
Its output should include lines like this:
k1 = 2.000000, k2 = 1.000000 k1 = 3.000000, k2 = 1.000000 k1 = 4.000000, k2 = 0.000000
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 i
is taken as
(a - b) - c
Parentheses may be used as above to force particular groupings.
Here is an example of the * operator. It uses the files multiplies.orc and multiplies.sco.
Example 13. Example of the * operator.
/* multiplies.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = 24 * 8 print i1 endin /* multiplies.orc */
/* multiplies.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* multiplies.sco */
Its output should include a line like this:
instr 1: i1 = 192.000
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 i
is taken as
(a - b) - c
Parentheses may be used as above to force particular groupings.
Here is an example of the + operator. It uses the files adds.orc and adds.sco.
Example 14. Example of the + operator.
/* adds.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = 24 + 8 print i1 endin /* adds.orc */
/* adds.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* adds.sco */
Its output should include lines like:
instr 1: i1 = 32.000
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 i
is taken as
(a - b) - c
Parentheses may be used as above to force particular groupings.
Here is an example of the - operator. It uses the files subtracts.orc and subtracts.sco.
Example 15. Example of the - operator.
/* subtracts.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = 24 - 8 print i1 endin /* subtracts.orc */
/* subtracts.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* subtracts.sco */
Its output should include lines like this:
instr 1: i1 = 16.000
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 i
is taken as
(a - b) - c
Parentheses may be used as above to force particular groupings.
Here is an example of the / operator. It uses the files divides.orc and divides.sco.
Example 16. Example of the / operator.
/* divides.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = 24 / 8 print i1 endin /* divides.orc */
/* divides.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* divides.sco */
Its output should include lines like this:
instr 1: i1 = 3.000
= (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.
Here is an example of the assign opcode. It uses the files assign.orc and assign.sco.
Example 17. Example of the assign opcode.
/* assign.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Assign a value to the variable i1. i1 = 1234 ; Print the value of the i1 variable. print i1 endin /* assign.orc */
/* assign.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* assign.sco */
Its output should include a line like this:
instr 1: i1 = 1234.000
In the above conditionals, a and b are first compared. If the indicated relation is true (a greater than b, a less than b, a greater than or equal to b, a less than or equal to b, a equal to b, 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. (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.
Here is an example of the == opcode. It uses the files equal.orc and equal.sco.
Example 18. Example of the == opcode.
/* equal.orc */ ; 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 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 /* equal.orc */
/* equal.sco */ ; 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 /* equal.sco */
Its output should include lines like this:
k1 = 2.000000, k2 = 0.000000 k1 = 3.000000, k2 = 1.000000 k1 = 4.000000, k2 = 0.000000
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 i
is taken as
(a - b) - c
Parentheses may be used as above to force particular groupings.
The operator ^ 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.)
Here is an example of the ^ operator. It uses the files raises.orc and raises.sco.
Example 19. Example of the ^ operator.
/* raises.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = 2 ^ 12 print i1 endin /* raises.orc */
/* raises.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* raises.sco */
Its output should include a line like this:
instr 1: i1 = 4096.000
round — Returns the integer value nearest to x ; if the fractional part of x is exactly 0.5, the direction of rounding is undefined.
The integer value nearest to x ; if the fractional part of x is exactly 0.5, the direction of rounding is undefined.
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 i
is taken as
(a - b) - c
Parentheses may be used as above to force particular groupings.
The default is 32767, so all existing orcs should work.
These calls should all work:
ipeak = 0dbfs
asig oscil 0dbfs,freq,1 out asig * 0.3 * 0dbfs
and so on.
As for documentation: the usage should be obvious - the main thing is for people to start to code 0dbfs-relatively (and use the ampdb() opcodes a lot more!), rather than use explicit sample values.
Floats written to a file, when 0dbfs = 1, will in effect go through no range translation at all. So the nunbers in the file are exactly what the orc says they are.
![]() | BIG NB |
---|---|
All the main sample formats are supported, but I haven't got around to dealing with the char formats. Probably it's straight-forward... I have tried to cover the main utils - adsyn,lpanal etc. But there are bound to be things missing, sorry. Some of the parsing code is a bit grungy because I have a variable with a leading digit! |
Here is an example of the 0dbfs opcode. It uses the files 0dbfs.orc and 0dbfs.sco.
Example 20. Example of the 0dbfs opcode.
/* 0dbfs.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Set the 0dbfs to the 16-bit maximum. 0dbfs = 32767 ; Instrument #1. instr 1 ; Linearly increase the amplitude value "kamp" from ; 0 to 1 over the duration defined by p3. kamp line 0, p3, 1 ; Generate a basic tone using our amplitude value. a1 oscil kamp, 440, 1 ; Multiply the basic tone (with its amplitude between ; 0 and 1) by the full-scale 0dbfs value. out a1 * 0dbfs endin /* 0dbfs.orc */
/* 0dbfs.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 e /* 0dbfs.sco */
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.
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.
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.
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.
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.
Here is an example of the a opcode. It uses the files a.orc and a.sco.
Example 21. Example of the a opcode.
/* a.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create a sine wave at k-rate. kwave oscil 20000, 440, 1 ; Convert the k-rate sine wave to the audio-rate. awave = a(kwave) ; Output the audio-rate version of sine wave. out awave endin /* a.orc */
/* a.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* a.sco */
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.
Here is an example of the abs opcode. It uses the files abs.orc and abs.sco.
Example 22. Example of the abs opcode.
/* abs.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = -6 i2 = abs(i1) print i2 endin /* abs.orc */
/* abs.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* abs.sco */
Its output should include lines like:
instr 1: i2 = 6.000
kinsnum -- number of the instrument to be reported
active returns the number of active instances of instrument number insnum/kinsnum. As of Csound4.17 the output is updated at k-rate (if input arg is k-rate), to allow running count of instr instances.
Here is a simple example of the active opcode. It uses the files active.orc and active.sco.
Example 23. Simple example of the active opcode.
/* active.orc */ ; 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 /* active.orc */
/* active.sco */ ; 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 /* active.sco */
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 files active_k.orc and active_k.sco.
Example 24. Example of the active opcode at k-rate.
/* active_k.orc */ ; 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 /* active_k.orc */
/* active_k.sco */ ; 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 /* active_k.sco */
Its output should include lines like:
i2 1.00000 i2 2.00000
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
The envelope is the range 0 to 1 and may need to be scaled further. The envelope may be described as:
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.
Here is an example of the adsr opcode. It uses the files adsr.orc and adsr.sco.
Example 25. Example of the adsr opcode.
/* adsr.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a simple instrument. instr 1 ; Set the amplitude. kamp init 20000 ; Get the frequency from the fourth p-field. kcps = cpspch(p4) a1 vco kamp, kcps, 1 out a1 endin ; Instrument #2 - instrument with an ADSR envelope. instr 2 iatt = 0.05 idec = 0.5 islev = 0.08 irel = 0.008 ; Create an amplitude envelope. kenv adsr iatt, idec, islev, irel kamp = kenv * 20000 ; Get the frequency from the fourth p-field. kcps = cpspch(p4) a1 vco kamp, kcps, 1 out a1 endin /* adsr.orc */
/* adsr.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Set the tempo to 120 beats per minute. t 0 120 ; Play a melody with Instrument #1. ; p4 = frequency in pitch-class notation. i 1 0 1 8.04 i 1 1 1 8.04 i 1 2 1 8.05 i 1 3 1 8.07 i 1 4 1 8.07 i 1 5 1 8.05 i 1 6 1 8.04 i 1 7 1 8.02 i 1 8 1 8.00 i 1 9 1 8.00 i 1 10 1 8.02 i 1 11 1 8.04 i 1 12 2 8.04 i 1 14 2 8.02 ; Repeat the melody with Instrument #2. ; p4 = frequency in pitch-class notation. i 2 16 1 8.04 i 2 17 1 8.04 i 2 18 1 8.05 i 2 19 1 8.07 i 2 20 1 8.07 i 2 21 1 8.05 i 2 22 1 8.04 i 2 23 1 8.02 i 2 24 1 8.00 i 2 25 1 8.00 i 2 26 1 8.02 i 2 27 1 8.04 i 2 28 2 8.04 i 2 30 2 8.02 e /* adsr.sco */
adsyn — Output is an additive set of individually controlled sinusoids, using an oscillator bank.
Output is an additive set of individually controlled sinusoids, using an oscillator bank.
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).
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.
Here is an example of the adsyn opcode. It uses the files adsyn.orc, adsyn.sco, and kickroll.het. The file “kickroll.het” was created by using the hetro utility with the audio file kickroll.wav.
Example 26. Example of the adsyn opcode.
/* adsyn.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; If the modulation amounts are set to 1, adsyn ; will not perform any special modulation. kamod init 1 kfmod init 1 ksmod init 1 ; Re-synthesizes the file "kickroll.het". a1 adsyn kamod, kfmod, ksmod, "kickroll.het" out a1 * 32768 endin /* adsyn.orc */
/* adsyn.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* adsyn.sco */
adsynt — Performs additive synthesis with an arbitrary number of partials, not necessarily harmonic.
Performs additive synthesis with an arbitrary number of partials, not necessarily harmonic.
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.
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.
Here is an example of the adsynt opcode. It uses the files adsynt.orc and adsynt.sco. 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 27. Example of the adsynt opcode.
/* adsynt.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 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 5000, 150, giwave, gifrqs, giamps, icnt out 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 5000, 150, giwave, gifrqs, giamps, icnt out asig endin /* adsynt.orc */
/* adsynt.sco */ ; 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 /* adsynt.sco */
adsynt2 — Performs additive synthesis with an arbitrary number of partials -not necessarily harmonic- with interpolation.
Performs additive synthesis with an arbitrary number of partials, not necessarily harmonic. (see adsynt for detailed manual)
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.
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).
imin (optional, default=0) -- minimum limit on values obtained.
imax (optional, default=127) -- maximum limit on values obtained.
Get the current after-touch 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.
Here is an example of the aftouch opcode. It uses the files aftouch.orc and aftouch.sco.
Example 28. Example of the aftouch opcode.
/* aftouch.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 k1 aftouch printk2 k1 endin /* aftouch.orc */
/* aftouch.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* aftouch.sco */
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.
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 the 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 will begin to appear immediately.
Here is an example of the alpass opcode. It uses the files alpass.orc and alpass.sco.
Example 29. Example of the alpass opcode.
/* alpass.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the audio mixer. gamix init 0 ; Instrument #1. instr 1 ; Generate a source signal. a1 oscili 30000, cpspch(p4), 1 ; Output the direct sound. out a1 ; Add the source signal to the audio mixer. gamix = gamix + a1 endin ; Instrument #99 (highest instr number executed last) instr 99 krvt = 1.5 ilpt = 0.1 ; Filter the mixed signal. a99 alpass gamix, krvt, ilpt ; Output the result. out a99 ; Empty the mixer for the next pass. gamix = 0 endin /* alpass.orc */
/* alpass.sco */ ; Table #1, a sine wave. f 1 0 128 10 1 ; p4 = frequency (in a pitch-class) ; Play Instrument #1 for a tenth of a second, p4=7.00 i 1 0 0.1 7.00 ; Play Instrument #1 for a tenth of a second, p4=7.02 i 1 1 0.1 7.02 ; Play Instrument #1 for a tenth of a second, p4=7.04 i 1 2 0.1 7.04 ; Play Instrument #1 for a tenth of a second, p4=7.06 i 1 3 0.1 7.06 ; Make sure the filter remains active. i 99 0 5 e /* alpass.sco */
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
Here is an example of the ampdb opcode. It uses the files ampdb.orc and ampdb.sco.
Example 30. Example of the ampdb opcode.
/* ampdb.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 idb = 90 iamp = ampdb(idb) print iamp endin /* ampdb.orc */
/* ampdb.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* ampdb.sco */
Its output should include lines like:
instr 1: iamp = 31622.764
ampdbfs — Returns the amplitude equivalent of the decibel value x, which is relative to full scale amplitude.
Returns the amplitude equivalent of the decibel value x, which is relative to full scale amplitude. Full scale is assumed to be 16 bit. New is Csound version 4.10.
Here is an example of the ampdbfs opcode. It uses the files ampdbfs.orc and ampdbfs.sco.
Example 31. Example of the ampdbfs opcode.
/* ampdbfs.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 idb = -1 iamp = ampdbfs(idb) print iamp endin /* ampdbfs.orc */
/* ampdbfs.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* ampdbfs.sco */
Its output should include lines like:
instr 1: iamp = 29203.621
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.
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.
Here is an example of the ampmidi opcode. It uses the files ampmidi.orc and ampmidi.sco.
Example 32. Example of the ampmidi opcode.
/* ampmidi.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Scale the amplitude between 0 and 1. i1 ampmidi 1 print i1 endin /* ampmidi.orc */
/* ampmidi.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* ampmidi.sco */
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.
ares -- the output signal at audio rate.
asig -- the input signal at audio 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).
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.
Here is an example of the areson opcode. It uses the files areson.orc and areson.sco.
Example 33. Example of the areson opcode.
/* areson.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1 - an unfiltered noise waveform. instr 1 ; Generate a white noise signal. asig rand 20000 out asig endin ; Instrument #2 - a filtered noise waveform. instr 2 ; Generate a white noise signal. asig rand 20000 ; Filter it using the areson opcode. kcf init 1000 kbw init 100 afilt areson asig, kcf, kbw ; Clip the filtered signal's amplitude to 85 dB. a1 clip afilt, 2, ampdb(85) out a1 endin /* areson.orc */
/* areson.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 2 2 e /* areson.sco */
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.
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 reson. 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.
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.
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.
Here is an example of the atone opcode. It uses the files atone.orc and atone.sco.
Example 34. Example of the atone opcode.
/* atone.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1 - an unfiltered noise waveform. instr 1 ; Generate a white noise signal. asig rand 20000 out asig endin ; Instrument #2 - a filtered noise waveform. instr 2 ; Generate a white noise signal. asig rand 20000 ; Filter it using the atone opcode. khp init 2000 afilt atone asig, khp ; Clip the filtered signal's amplitude to 85 dB. a1 clip afilt, 2, ampdb(85) out a1 endin /* atone.orc */
/* atone.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 2 2 e /* atone.sco */
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.
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.
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.
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.
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.
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)
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.
Here is a simple example of the babo opcode. It uses the files babo.orc, babo.sco, and beats.wav.
Example 35. A simple example of the babo opcode.
/* babo.orc */ /* Written by Nicola Bernardini */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 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 /* babo.orc */
/* babo.sco */ /* 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 10 6 4 3 14.39 11.86 10 ; ^^^^^^^ ^^^^^^^^^^^^^^ ; ||||||| ++++++++++++++: optimal room dims according to ; ||||||| Milner and Bernard JASA 85(2), 1989 ; +++++++++: source position e /* babo.sco */
Here is an advanced example of the babo opcode. It uses the files babo_expert.orc, babo_expert.sco, and beats.wav.
Example 36. An advanced example of the babo opcode.
/* babo_expert.orc */ /* Written by Nicola Bernardini */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 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 /* babo_expert.orc */
/* babo_expert.sco */ /* 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 /* babo_expert.sco */
The rms power of asig can be interrogated, set, or adjusted to match that of a comparator signal.
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.
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.
Here is an example of the balance opcode. It uses the files balance.orc and balance.sco.
Example 37. Example of the balance opcode.
/* balance.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a band-limited pulse train. asrc buzz 30000, 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 out afin endin /* balance.orc */
/* balance.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* balance.sco */
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.
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.
kamp -- Amplitude of output. Note: As these instruments are stochastic, this is only an approximation.
Here is an example of the bamboo opcode. It uses the files bamboo.orc and bamboo.sco.
Example 38. Example of the bamboo opcode.
/* bamboo.orc */ sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 01 ;example of bamboo a1 bamboo p4, 0.01 out a1 endin /* bamboo.orc */
/* bamboo.sco */ i1 0 1 20000 e /* bamboo.sco */
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.
a1 bbcutm asource, ibps, isubdiv, ibarlength, iphrasebars, inumrepeats [, istutterspeed] [, istutterchance] [, ienvchoice ]
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.
asource -- The audio signal to be cut up. This version runs in real-time without knowledge of future audio.
Here is a simple example of the bbcutm opcode. It uses the files bbcutm.orc, bbcutm.sco, and beats.wav.
Example 39. A simple example of the bbcutm opcode.
/* bbcutm.orc */ ; 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 /* bbcutm.orc */
/* bbcutm.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 3 2 e /* bbcutm.sco */
Here are some more advanced examples...
Example 40. 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>
Example 41. Multiple simultaneous synchronised breaks
<CsoundSynthesizer> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 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>
Example 42. Cutting up any old audio- much more interesting noises than this should be possible!
<CsoundSynthesizer> <CsInstruments> sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 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>
Example 43. 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 = 1 instr 1 asource diskin "break7.wav",1,0,1 ;16th note cuts- but cut size 2 over half a beat. ;each half beat will eiather 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>
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.
a1,a2 bbcuts asource1, asource2, ibps, isubdiv, ibarlength, iphrasebars, inumrepeats [, istutterspeed] [, istutterchance] [, ienvchoice]
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.
Beta distribution random number generator (positive values only). This is an x-class noise generator.
ares betarand krange, kalpha, kbeta
ires betarand krange, kalpha, kbeta
kres betarand krange, kalpha, kbeta
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.
Here is an example of the betarand opcode. It uses the files betarand.orc and betarand.sco.
Example 44. Example of the betarand opcode.
/* betarand.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a number between 0 and 1 with a ; uniform distribution. ; krange = 1 ; kalpha = 1 ; kbeta = 1 i1 betarand 1, 1, 1 print i1 endin /* betarand.orc */
/* betarand.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* betarand.sco */
Its output should include lines like:
instr 1: i1 = 24583.412
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.
Here is an example of the bexprnd opcode. It uses the files bexprnd.orc and bexprnd.sco.
Example 45. Example of the bexprnd opcode.
/* bexprnd.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a random number between -1 and 1. ; krange = 1 i1 bexprnd 1 print i1 endin /* bexprnd.orc */
/* bexprnd.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* bexprnd.sco */
Its output should include lines like:
instr 1: i1 = 1.141
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
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.
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 out a1, a2, a3, a4, a5, a6, a7, a8 endin
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]]]
isetup –- loudspeaker setup. There are five supported setups: 1 denotes stereo setup. There must be two output cells with loudspeaker positions (azimuth angle clockwise/altitude angle) 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).
aw, ax, ay, ... -- input signal in the B format.
ao1 .. ao8 -– loudspeaker specific output signals.
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 out a1, a2, a3, a4, a5, a6, a7, a8 endin
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.
fsig -- output pv stream in PVS_AMP_FREQ format
fin -- input pv stream in TRACKS format
isize -- FFT size of output (N).
Example 46. Example
ain inch 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 out aout
The example above shows partial tracking of an ifd-analysis signal, conversion to bin frames and overlap-add resynthesis.
iskip (optional, default=0) -- if non-zero, intialization will be skipped. Default value 0. (New in Csound version 3.50)
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.
Here is an example of the biquad opcode. It uses the files biquad.orc and biquad.sco.
Example 47. Example of the biquad opcode.
/* biquad.orc */ ; 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 /* biquad.orc */
/* biquad.sco */ ; 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 /* biquad.sco */
iskip (optional, default=0) -- if non-zero, intialization will be skipped. Default value 0. (New in Csound version 3.50)
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.
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.
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.
Here is an example of the birnd opcode. It uses the files birnd.orc and birnd.sco.
Example 48. Example of the birnd opcode.
/* birnd.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a random number from -1 to 1. i1 = birnd(1) print i1 endin /* birnd.orc */
/* birnd.sco */ ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #1 for one second. i 1 1 1 e /* birnd.sco */
Its output should include lines like:
instr 1: i1 = 0.947 instr 1: i1 = -0.721
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)
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.
Here is an example of the bqrez opcode. It uses the files bqrez.orc and bqrez.sco.
Example 49. Example of the bqrez opcode borrowed from the “rezzy” opcode in Kevin Conder's manual.
/* bqrez.orc */ /* Written by Matt Gerassimof from example by Kevin Conder */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a nice sawtooth waveform. asig vco 16000, 220, 1 ; Vary the filter-cutoff frequency from .2 to 2 KHz. kfco line 200, p3, 2000 ; Set the resonance amount. kres init 0.99 a1 bqrez asig, kfco, kres out a1 endin /* bqrez.orc */
/* bqrez.sco */ /* Written by Matt Gerassimof from example by Kevin Conder */ ; Table #1, a sine wave for the vco opcode. f 1 0 16384 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 e /* bqrez.sco */
Implementation of a second-order band-pass Butterworth filter. This opcode can also be written as butbp.
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 -- Cutoff or center frequency for each of the filters.
kband -- Bandwidth of the bandpass and bandreject filters.
Here is an example of the butterbp opcode. It uses the files butterbp.orc and butterbp.sco.
Example 50. Example of the butterbp opcode.
/* butterbp.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1 - an unfiltered noise waveform. instr 1 ; White noise signal asig rand 22050 out asig endin ; Instrument #2 - a filtered noise waveform. instr 2 ; White noise signal asig rand 22050 ; Filter it, passing only 1950 to 2050 Hz. abp butterbp asig, 2000, 100 out abp endin /* butterbp.orc */
/* butterbp.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 2 2 e /* butterbp.sco */
Implementation of a second-order band-reject Butterworth filter. This opcode can also be written as butbr.
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 -- Cutoff or center frequency for each of the filters.
kband -- Bandwidth of the bandpass and bandreject filters.
Here is an example of the butterbr opcode. It uses the files butterbr.orc and butterbr.sco.
Example 51. Example of the butterbr opcode.
/* butterbr.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1 - an unfiltered noise waveform. instr 1 ; White noise signal asig rand 22050 out asig endin ; Instrument #2 - a filtered noise waveform. instr 2 ; White noise signal asig rand 22050 ; Filter it, cutting 2000 to 6000 Hz. abr butterbr asig, 4000, 2000 out abr endin /* butterbr.orc */
/* butterbr.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 2 2 e /* butterbr.sco */
Implementation of second-order high-pass Butterworth filter. This opcode can also be written as buthp.
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 -- Cutoff or center frequency for each of the filters.
Here is an example of the butterhp opcode. It uses the files butterhp.orc and butterhp.sco.
Example 52. Example of the butterhp opcode.
/* butterhp.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1 - an unfiltered noise waveform. instr 1 ; White noise signal asig rand 22050 out asig endin ; Instrument #2 - a filtered noise waveform. instr 2 ; White noise signal asig rand 22050 ; Filter it, passing frequencies above 250 Hz. ahp butterhp asig, 250 out ahp endin /* butterhp.orc */
/* butterhp.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 2 2 e /* butterhp.sco */
Implementation of a second-order low-pass Butterworth filter. This opcode can also be written as butlp.
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 -- Cutoff or center frequency for each of the filters.
Here is an example of the butterlp opcode. It uses the files butterlp.orc and butterlp.sco.
Example 53. Example of the butterlp opcode.
/* butterlp.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1 - an unfiltered noise waveform. instr 1 ; White noise signal asig rand 22050 out asig endin ; Instrument #2 - a filtered noise waveform. instr 2 ; White noise signal asig rand 22050 ; Filter it, cutting frequencies above 1 KHz. alp butterlp asig, 1000 out alp endin /* butterlp.orc */
/* butterlp.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 2 2 e /* butterlp.sco */
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
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.
Here is an example of the buzz opcode. It uses the files buzz.orc and buzz.sco.
Example 54. Example of the buzz opcode.
/* buzz.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 20000 kcps = 440 knh = 3 ifn = 1 a1 buzz kamp, kcps, knh, ifn out a1 endin /* buzz.orc */
/* buzz.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* buzz.sco */
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.
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.
Here is an example of the cabasa opcode. It uses the files cabasa.orc and cabasa.sco.
Example 55. Example of the cabasa opcode.
/* cabasa.orc */ ;orchestra --------------- sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 01 ;an example of a cabasa a1 cabasa p4, 0.01 out a1 endin /* cabasa.orc */
/* cabasa.sco */ ;score ------------------- i1 0 1 26000 e /* cabasa.sco */
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.
Here is an example of the cauchy opcode. It uses the files cauchy.orc and cauchy.sco.
Example 56. Example of the cauchy opcode.
/* cauchy.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a random number, spread from 10. ; kalpha = 10 i1 cauchy 10 print i1 endin /* cauchy.orc */
/* cauchy.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cauchy.sco */
Its output should include lines like:
instr 1: i1 = -0.106
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.
Here is an example of the cent opcode. It uses the files cent.orc and cent.sco.
Example 57. Example of the cent opcode.
/* cent.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; The root note is A above middle-C (440 Hz) iroot = 440 ; Raise the root note by 300 cents to C. icents = 300 ; Calculate the new note. ifactor = cent(icents) inew = iroot * ifactor ; Print out of all of the values. print iroot print ifactor print inew endin /* cent.orc */
/* cent.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cent.sco */
Its output should include lines like:
instr 1: iroot = 440.000 instr 1: ifactor = 1.189 instr 1: inew = 523.229
cggoto condition, label
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).
Here is an example of the cggoto opcode. It uses the files cggoto.orc and cggoto.sco.
Example 58. Example of the cggoto opcode.
/* cggoto.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = 1 ; 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 /* cggoto.orc */
/* cggoto.sco */ ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* cggoto.sco */
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.
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.
Here is an example of the changed opcode. It uses the file changed.csd.
Example 59. Example of the changed opcode.
<CsoundSynthesizer> <CsOptions> -odac -B441 -b441 </CsOptions> <CsInstruments> sr = 44100 kr = 100 ksmps = 441 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 </CsScore> </CsoundSynthesizer>
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. 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.
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 chani 1 a1 oscil p4, p5, 100 a2 lowpass2 a1, kc, 200 out a2 endin
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. 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.
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.
Here is a simple example of the checkbox opcode. It uses the files checkbox.orc and checkbox.sco.
Example 60. Simple example of the checkbox opcode.
/* checkbox.orc */ ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 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 out a1 endin /* checkbox.orc */
/* checkbox.sco */ ; Just generate a nice, ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for ten seconds. i 1 0 10 e /* checkbox.sco */
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.
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.
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.
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 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
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.
gival chnexport Sname, imode[, itype, idflt, imin, imax]
gkval chnexport Sname, imode[, itype, idflt, imin, imax]
gaval chnexport Sname, imode
gSval chnexport Sname, imode
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.
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.
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 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
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).
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.
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 chnget "cutoff" a1 oscil p4, p5, 100 a2 lowpass2 a1, kc, 200 out a2 endin
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.
Write to a channel of the named software bus. Implies declaring the channel with imode=2 (see also chn_k, chn_a, and chn_S).
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.
During the i-time pass only, unconditionally transfer control to the statement labeled by label.
cigoto condition, label
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).
Here is an example of the cigoto opcode. It uses the files cigoto.orc and cigoto.sco.
Example 61. Example of the cigoto opcode.
/* cigoto.orc */ ; 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 /* cigoto.orc */
/* cigoto.sco */ ; 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 /* cigoto.sco */
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
During the p-time passes only, unconditionally transfer control to the statement labeled by label.
ckgoto condition, label
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).
Here is an example of the ckgoto opcode. It uses the files ckgoto.orc and ckgoto.sco.
Example 62. Example of the ckgoto opcode.
/* ckgoto.orc */ ; 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 /* ckgoto.orc */
/* ckgoto.sco */ ; Table: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* ckgoto.sco */
Its output should include lines like:
kval = 0.000000, kfreq = 440.000000 kval = 0.999732, kfreq = 440.000000 kval = 1.999639, kfreq = 880.000000
avar1, avar2, avar3, ... -- signals to be zeroed
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 variable can be used for output signal by means of fout opcode. After the disk writing operation, the accumulator variable should be set to zero by means of clear opcode (or it will explode).
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.
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.
asig -- The input audio signal.
kfreq -- The corner frequency for low-pass or high-pass.
Here is an example of the clfilt opcode as a low-pass filter. It uses the files clfilt_lowpass.orc and clfilt_lowpass.sco.
Example 63. Example of the clfilt opcode as a low-pass filter.
/* clfilt_lowpass.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1 - an unfiltered noise waveform. instr 1 ; White noise signal asig rand 22050 out asig endin ; Instrument #2 - a filtered noise waveform. instr 2 ; White noise signal asig rand 22050 ; Lowpass filter signal asig with a ; 10-pole Butterworth at 500 Hz. a1 clfilt asig, 500, 0, 10 out a1 endin /* clfilt_lowpass.orc */
/* clfilt_lowpass.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 2 2 e /* clfilt_lowpass.sco */
Here is an example of the clfilt opcode as a high-pass filter. It uses the files clfilt_highpass.orc and clfilt_highpass.sco.
Example 64. Example of the clfilt opcode as a high-pass filter.
/* clfilt_highpass.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1 - an unfiltered noise waveform. instr 1 ; White noise signal asig rand 22050 out asig endin ; Instrument #2 - a filtered noise waveform. instr 2 ; White noise signal asig rand 22050 ; 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 out a1 endin /* clfilt_highpass.orc */
/* clfilt_highpass.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 2 2 e /* clfilt_highpass.sco */
Clips an a-rate signal to a predefined limit, in a “soft” manner, using one of three methods.
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.
asig -- a-rate input signal
The Bram de Jong method (imeth = 0) applies the algorithm:
|x| > a: f(x) = sin(x) * (a+(x-a)/(1+((x-a)/(1-a))2 |x| > 1: f(x) = sin(x) * (a+1)/2
This method requires that asig be normalized to 1.
The second method (imeth = 1) is the sine clip:
|x| < limit: f(x) = limit * sin(π*x/(2*limit)) f(x) = limit * sin(x)
The third method (imeth = 0) is the tanh clip:
|x| < limit: f(x) = limit * tanh(x/limit)/tanh(1) f(x) = limit * sin(x)
![]() | Note |
---|---|
Method 1 appears to be non-functional at release of Csound version 4.07. |
Here is an example of the clip opcode. It uses the files clip.orc and clip.sco.
Example 65. Example of the clip opcode.
/* clip.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a noisy waveform. arnd rand 44100 ; Clip the noisy waveform's amplitude to 20,000 a1 clip arnd, 2, 20000 out a1 endin /* clip.orc */
/* clip.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* clip.sco */
inum -- the number of a clock. There are 32 clocks numbered 0 through 31. All other values are mapped to clock number 32.
inum -- the number of a clock. There are 32 clocks numbered 0 through 31. All other values are mapped to clock number 32.
cngoto condition, label
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).
Here is an example of the cngoto opcode. It uses the files cngoto.orc and cngoto.sco.
Example 66. Example of the cngoto opcode.
/* cngoto.orc */ ; 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 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 /* cngoto.orc */
/* cngoto.sco */ ; Table: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* cngoto.sco */
Its output should include lines like:
kval = 0.000000, kfreq = 880.000000 kval = 0.999732, kfreq = 880.000000 kval = 1.999639, kfreq = 440.000000
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 4n*sr 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.
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.
Here is an example of the comb opcode. It uses the files comb.orc and comb.sco.
Example 67. Example of the comb opcode.
/* comb.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the audio mixer. gamix init 0 ; Instrument #1. instr 1 ; Generate a source signal. a1 oscili 30000, cpspch(p4), 1 ; Output the direct sound. out a1 ; Add the source signal to the audio mixer. gamix = gamix + a1 endin ; Instrument #99 (highest instr number executed last) instr 99 krvt = 1.5 ilpt = 0.1 ; Comb-filter the mixed signal. a99 comb gamix, krvt, ilpt ; Output the result. out a99 ; Empty the mixer for the next pass. gamix = 0 endin /* comb.orc */
/* comb.sco */ ; Table #1, a sine wave. f 1 0 128 10 1 ; p4 = frequency (in a pitch-class) ; Play Instrument #1 for a tenth of a second, p4=7.00 i 1 0 0.1 7.00 ; Play Instrument #1 for a tenth of a second, p4=7.02 i 1 1 0.1 7.02 ; Play Instrument #1 for a tenth of a second, p4=7.04 i 1 2 0.1 7.04 ; Play Instrument #1 for a tenth of a second, p4=7.06 i 1 3 0.1 7.06 ; Make sure the comb-filter remains active. i 99 0 5 e /* comb.sco */
Configurable slider controls for realtime user input. Requires Winsound or TCL/TK. control reads a slider's value.
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.
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.
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.
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.
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
Create .orc file, e.g.:
; Simple demonstration of CONVOLVE operator, to apply reverb. sr = 44100 kr = 441 ksmps = 100 nchnls = 2 instr 1 imix = 0.22 ; Wet/dry mix. Vary as desired. ; 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. ivol = 0.9 ; Overall volume level of reverb. May need to adjust ; when wet/dry mix is changed, to avoid clipping. idel = 1.32 ; Required delay to align dry audio with output of convolve. ; This can be automatically calculated within the orc file, ; if desired. adry soundin "anechoic.wav" ; input (dry) audio awet1,awet2 convolve adry,"l1_44.cv" ; stereo convolved (wet) audio 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, and output endin
Here is an example of the cos opcode. It uses the files cos.orc and cos.sco.
Example 68. Example of the cos opcode.
/* cos.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 25 i1 = cos(irad) print i1 endin /* cos.orc */
/* cos.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cos.sco */
Its output should include lines like this:
instr 1: i1 = 0.991
Here is an example of the cosh opcode. It uses the files cosh.orc and cosh.sco.
Example 69. Example of the cosh opcode.
/* cosh.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 1 i1 = cosh(irad) print i1 endin /* cosh.orc */
/* cosh.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cosh.sco */
Its output should include lines like this:
instr 1: i1 = 1.543
Here is an example of the cosinv opcode. It uses the files cosinv.orc and cosinv.sco.
Example 70. Example of the cosinv opcode.
/* cosinv.orc */ ; 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 /* cosinv.orc */
/* cosinv.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cosinv.sco */
Its output should include lines like this:
instr 1: i1 = 1.047
cps2pch — Converts a pitch-class value into cycles-per-second for equal divisions of the octave.
Converts a pitch-class value into cycles-per-second (Hz) for equal divisions of the octave.
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.
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Here is an example of the cps2pch opcode. It uses the files cps2pch.orc and cps2pch.sco.
Example 71. Example of the cps2pch opcode.
/* cps2pch.orc */ ; 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 /* cps2pch.orc */
/* cps2pch.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cps2pch.sco */
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 files cps2pch_ftable.orc and cps2pch_ftable.sco.
Example 72. Example of the cps2pch opcode using a table of frequency multipliers.
/* cps2pch_ftable.orc */ ; 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 /* cps2pch_ftable.orc */
/* cps2pch_ftable.sco */ ; 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 /* cps2pch_ftable.sco */
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 files cps2pch_19et.orc and cps2pch_19et.sco.
Example 73. Example of the cps2pch opcode using a 19ET scale.
/* cps2pch_19et.orc */ ; 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 /* cps2pch_19et.orc */
/* cps2pch_19et.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cps2pch_19et.sco */
Its output should include lines like this:
instr 1: icps = 281.429
Get the note number of the current MIDI event, expressed in cycles-per-second units, for local processing.
Here is an example of the cpsmidi opcode. It uses the files cpsmidi.orc and cpsmidi.sco.
Example 74. Example of the cpsmidi opcode.
/* cpsmidi.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 cpsmidi print i1 endin /* cpsmidi.orc */
/* cpsmidi.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* cpsmidi.sco */
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.
Get the note number of the current MIDI event and modify it by the current pitch-bend value, express it in cycles-per-second.
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.
Here is an example of the cpsmidib opcode. It uses the files cpsmidib.orc and cpsmidib.sco.
Example 75. Example of the cpsmidib opcode.
/* cpsmidib.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 cpsmidib print i1 endin /* cpsmidib.orc */
/* cpsmidib.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* cpsmidib.sco */
cpsoct (oct) (no rate restriction)
where the argument within the parentheses may be a further expression.
These 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:
Table 1. Pitch and Frequency Values
Name | Abbreviation |
---|---|
octave point pitch-class (8ve.pc) | pch |
octave point decimal | oct |
cycles per second | cps |
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. Thus A440 can be represented alternatively by 440 (cps),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.
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The conversion from pch or oct 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. |
Here is an example of the cpsoct opcode. It uses the files cpsoct.orc and cpsoct.sco.
Example 76. Example of the cpsoct opcode.
/* cpsoct.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Convert an octave-point-decimal value into a ; cycles-per-second value. ioct = 8.75 icps = cpsoct(ioct) print icps endin /* cpsoct.orc */
/* cpsoct.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cpsoct.sco */
Its output should include lines like this:
instr 1: icps = 440.000
cpspch (pch) (init- or control-rate args only)
where the argument within the parentheses may be a further expression.
These 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:
Table 2. Pitch and Frequency Values
Name | Abbreviation |
---|---|
octave point pitch-class (8ve.pc) | pch |
octave point decimal | oct |
cycles per second | cps |
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. Thus A440 can be represented alternatively by 440 (cps),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.
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The conversion from pch or oct 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. |
Here is an example of the cpspch opcode. It uses the files cpspch.orc and cpspch.sco.
Example 77. Example of the cpspch opcode.
/* cpspch.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Convert a pitch-class value into a ; cycles-per-second value. ipch = 8.09 icps = cpspch(ipch) print icps endin /* cpspch.orc */
/* cpspch.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cpspch.sco */
Its output should include lines like this:
instr 1: icps = 440.000
ifn -- function table containing the parameters (numgrades, interval, basefreq, basekeymidi) and the tuning ratios.
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...
Here is an example of the cpstmid opcode. It uses the files cpstmid.orc and cpstmid.sco.
Example 78. Example of the cpstmid opcode.
/* cpstmid.orc */ ; 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 i1 cpstmid ifn print i1 endin /* cpstmid.orc */
/* cpstmid.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* cpstmid.sco */
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...
Here is an example of the cpstun opcode. It uses the files cpstun.orc and cpstun.sco.
Example 79. Example of the cpstun opcode.
/* cpstun.orc */ ; 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 /* cpstun.orc */
/* cpstun.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cpstun.sco */
Its output should include lines like this:
i1 440.11044
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.
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...
Here is an example of the cpstuni opcode. It uses the files cpstuni.orc and cpstuni.sco.
Example 80. Example of the cpstuni opcode.
/* cpstuni.orc */ ; 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 /* cpstuni.orc */
/* cpstuni.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cpstuni.sco */
Its output should include lines like this:
instr 1: i1 = 440.110
cpsxpch — Converts a pitch-class value into cycles-per-second (Hz) for equal divisions of any interval.
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.
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
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Here is an example of the cpsxpch opcode. It uses the files cpsxpch.orc and cpsxpch.sco.
Example 81. Example of the cpsxpch opcode.
/* cpsxpch.orc */ ; 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 /* cpsxpch.orc */
/* cpsxpch.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cpsxpch.sco */
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 files cpsxpch_105et.orc and cpsxpch_105et.sco.
Example 82. Example of the cpsxpch opcode using a 10.5 ET scale.
/* cpsxpch_105et.orc */ ; 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 /* cpsxpch_105et.orc */
/* cpsxpch_105et.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cpsxpch_105et.sco */
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 files cpsxpch_pierce.orc and cpsxpch_pierce.sco.
Example 83. Example of the cpsxpch opcode using a Pierce scale centered on middle A.
/* cpsxpch_pierce.orc */ ; 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 /* cpsxpch_pierce.orc */
/* cpsxpch_pierce.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* cpsxpch_pierce.sco */
Its output should include lines like this:
instr 1: icps = 2827.762
cpuprc — Control allocation of cpu resources on a per-instrument basis, to optimize realtime output.
Control allocation of cpu resources on a per-instrument basis, to optimize realtime output.
insnum -- instrument number
ipercent -- percent of cpu processing-time to assign. Can also be expressed as a fractional value.
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.
Here is an example of the cpuprc opcode. It uses the files cpuprc.orc and cpuprc.sco.
Example 84. Example of the cpuprc opcode.
/* cpuprc.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Limit Instrument #1 to 5% of the CPU processing time. cpuprc 1, 5 ; Instrument #1 instr 1 a1 oscil 10000, 440, 1 out a1 endin /* cpuprc.orc */
/* cpuprc.sco */ ; Just generate a nice, ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* cpuprc.sco */
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.
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.
Here is an example of the cross2 opcode. It uses the files cross2.orc, cross2.sco and beats.wav.
Example 85. Example of the cross2 opcode.
/* cross2.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - Play an audio file. instr 1 ; Use the "beats.wav" audio file. aout soundin "beats.wav" out aout endin ; Instrument #2 - Cross-synthesize! instr 2 ; Use the "ahh" sound stored in Table #1. ain1 loscil 30000, 1, 1, 1 ; Use the "beats.wav" audio file. ain2 soundin "beats.wav" isize = 4096 ioverlap = 2 iwin = 2 kbias init 1 aout cross2 ain1, ain2, isize, ioverlap, iwin, kbias out aout endin /* cross2.orc */
/* cross2.sco */ ; Table #1: An audio file. f 1 0 128 1 "ahh.aiff" 0 4 0 ; Table #2: A windowing function. f 2 0 2048 20 2 ; Play Instrument #1 for 2 seconds. i 1 0 2 ; Play Instrument #2 for 2 seconds. i 2 2 2 e /* cross2.sco */
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.
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.
Here is an example of the crunch opcode. It uses the files crunch.orc and crunch.sco.
Example 86. Example of the crunch opcode.
/* crunch.orc */ ;orchestra --------------- sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 01 ;an example of a crunch a1 crunch p4, 0.01 out a1 endin /* crunch.orc */
/* crunch.sco */ ;score ------------------- i1 0 1 26000 e /* crunch.sco */
ctrl14 — 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]
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.
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 becase 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.
ctrl21 — 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]
idest -- output signal
ichan -- MIDI channel number (1-16)
ictlno -- MIDI controller number (0-127)
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 imax and imin val.
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.
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.
kdest -- 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.
aout cuserrnd kmin, kmax, ktableNum
iout cuserrnd imin, imax, itableNum
kout cuserrnd kmin, kmax, ktableNum
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
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.
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.
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.
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).
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 files dam.orc, dam.sco, and beats.wav.
Example 87. An example of the dam opcode compressing an audio signal.
/* dam.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1, uncompressed signal. instr 1 ; Use the "beats.wav" audio file. asig soundin "beats.wav" out asig endin ; Instrument #2, compressed signal. instr 2 ; Use the "beats.wav" audio file. asig soundin "beats.wav" ; Compress the audio signal. kthreshold init 25000 icomp1 = 0.5 icomp2 = 0.763 irtime = 0.1 iftime = 0.1 a1 dam asig, kthreshold, icomp1, icomp2, irtime, iftime out a1 endin /* dam.orc */
/* dam.sco */ ; Play Instrument #1 for 2 seconds. i 1 0 2 ; Play Instrument #2 for 2 seconds. i 2 2 2 e /* dam.sco */
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 files dam_expanded.orc, dam_expanded.sco, and mary.wav.
Example 88. An example of the dam opcode expanding an audio signal.
/* dam_expanded.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1, normal audio signal. instr 1 ; Use the "mary.wav" audio file. asig soundin "mary.wav" out asig endin ; Instrument #2, expanded audio signal. instr 2 ; Use the "mary.wav" audio file. asig soundin "mary.wav" ; Expand the audio signal. kthreshold init 7500 icomp1 = 2.25 icomp2 = 2.25 irtime = 0.1 iftime = 0.6 a1 dam asig, kthreshold, icomp1, icomp2, irtime, iftime out a1 endin /* dam_expanded.orc */
/* dam_expanded.sco */ ; Play Instrument #1. i 1 0.0 3.5 ; Play Instrument #2. i 2 3.5 3.5 e /* dam_expanded.sco */
This example expands the audio file “mary.wav”. You should hear a melody repeat twice. The second time, the sound should be louder (expanded) than the first.
Returns the amplitude equivalent for a given decibel amount. This opcode is the same as db.
Here is an example of the db opcode. It uses the files db.orc and db.sco.
Example 89. Example of the db opcode.
/* db.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Calculate the amplitude of 40 decibels. idecibels = 40 iamp = db(idecibels) print iamp endin /* db.orc */
/* db.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* db.sco */
Its output should include lines like:
instr 1: iamp = 100.000
Here is an example of the dbamp opcode. It uses the files dbamp.orc and dbamp.sco.
Example 90. Example of the dbamp opcode.
/* dbamp.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 iamp = 30000 idb = dbamp(iamp) print idb endin /* dbamp.orc */
/* dbamp.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* dbamp.sco */
Its output should include lines like this:
instr 1: idb = 89.542
dbfsamp — Returns the decibel equivalent of the raw amplitude x, relative to full scale amplitude.
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.
Here is an example of the dbfsamp opcode. It uses the files dbfsamp.orc and dbfsamp.sco.
Example 91. Example of the dbfsamp opcode.
/* dbfsamp.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 iamp = 30000 idb = dbfsamp(iamp) print idb endin /* dbfsamp.orc */
/* dbfsamp.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* dbfsamp.sco */
Its output should include lines like this:
instr 1: idb = -0.767
Implements the DC blocking filter
Y[i] = X[i] - X[i-1] + (igain * Y[i-1])
Based on work by Perry Cook.
Here is an example of the dcblock opcode. It uses the files dcblock.orc, dcblock.sco, and beats.wav.
Example 92. Example of the dcblock opcode.
/* dcblock.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 -- normal audio signal. instr 1 asig soundin "beats.wav" out asig endin ; Instrument #2 -- dcblock-ed audio signal. instr 2 asig soundin "beats.wav" igain = 0.75 a1 dcblock asig, igain out a1 endin /* dcblock.orc */
/* dcblock.sco */ ; Play Instrument #1 for 2 seconds. i 1 0 2 ; Play Instrument #2 for 2 seconds. i 2 2 2 e /* dcblock.sco */
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.
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.
Here is an example of the dconv opcode. It uses the files dconv.orc and dconv.sco.
Example 93. Example of the dconv opcode.
/* dconv.orc */ 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 /* dconv.orc */
/* dconv.sco */ f1 0 16 10 1 i1 0 10 e /* dconv.sco */
A signal can be read from or written into a delay path, or it can be automatically delayed by some time interval.
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.
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.
Here is an example of the delay opcode. It uses the files delay.orc and delay.sco.
Example 94. Example of the delay opcode.
/* delay.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Instrument #1 -- Delayed beeps. instr 1 ; Make a basic sound. abeep vco 20000, 440, 1 ; Delay the beep by .1 seconds. idlt = 0.1 adel delay abeep, idlt ; Send the beep to the left speaker and ; the delayed beep to the right speaker. outs abeep, adel endin /* delay.orc */
/* delay.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Keep the score running for 2 seconds. f 0 2 ; Play Instrument #1. i 1 0.0 0.2 i 1 0.5 0.2 e /* delay.sco */
iskip (optional, default=0) -- initial disposition of delay-loop data space (see reson). The default value is 0.
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.
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.
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.
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.
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).
Here is an example of the delayw opcode. It uses the files delayw.orc and delayw.sco.
Example 95. Example of the delayw opcode.
/* delayw.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Instrument #1 -- Delayed beeps. instr 1 ; Make a basic sound. abeep vco 20000, 440, 1 ; Set up a delay line. idlt = 0.1 adel delayr idlt ; Write the beep to the delay line. delayw abeep ; Send the beep to the left speaker and ; the delayed beep to the right speaker. outs abeep, adel endin /* delayw.orc */
/* delayw.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Keep the score running for 2 seconds. f 0 2 ; Play Instrument #1. i 1 0.0 0.2 i 1 0.5 0.2 e /* delayw.sco */
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 John ffitch).
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.
Example 96. 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
Example 97. 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
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 John ffitch).
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.
Example 98. 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
Example 99. 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
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 John ffitch).
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.
Example 100. 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
Example 101. 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
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 (Hans Mikelson).
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 John ffitch).
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.
Example 102. 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
Example 103. 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
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.
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.
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)
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Window sizes for opcodes other than deltapx are: deltap, deltapn: 1, deltapi: 2 (linear), deltap3: 4 (cubic) |
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.
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.
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)
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Window sizes for opcodes other than deltapx are: deltap, deltapn: 1, deltapi: 2 (linear), deltap3: 4 (cubic) |
iskip (optional) -- initial disposition of internal save space (see reson). The default value is 0.
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 * sin(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.
Here is an example of the diff opcode. It uses the files diff.orc and diff.sco.
Example 104. Example of the diff opcode.
/* diff.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 -- a normal instrument. instr 1 ; Generate a band-limited pulse train. asrc buzz 20000, 440, 20, 1 out asrc endin ; Instrument #2 -- a differentiated instrument. instr 2 ; Generate a band-limited pulse train. asrc buzz 20000, 440, 20, 1 ; Emphasize the highs. a1 diff asrc out a1 endin /* diff.orc */
/* diff.sco */ ; 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 1 1 e /* diff.sco */
ar1 [, ar2 [, ar3 [, ... ar24]]] diskin
ifilcod, kpitch [, iskiptim] [, iwraparound] [, iformat] [, iskipinit]
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)
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.
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.
diskin is identical to soundin except that it can alter the pitch of the sound that is being read.
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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:
|
Here is an example of the diskin opcode. It uses the files diskin.orc, diskin.sco, beats.wav.
Example 105. Example of the diskin opcode.
/* diskin.orc */ ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1 - play an audio file. instr 1 ; Play the audio file backwards. asig diskin "beats.wav", -1 out asig endin /* diskin.orc */
/* diskin.sco */ ; Play Instrument #1, the audio file, for three seconds. i 1 0 3 e /* diskin.sco */
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.
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. diskin2 allows more control and higher sound quality than diskin, but there is also the disadvantage of higher CPU usage.
a1[, a2[, ... a24]] diskin2 ifilcod, kpitch[, iskiptim[, iwrap[, iformat [, iwsize[, ibufsize[, iskipinit]]]]]]
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 2^31-1 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.
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.
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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.
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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.
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.
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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.
<CsoundSynthesizer> <CsOptions> ; set this to a directory where beats.aiff can be found --env:SSDIR+=/Csound/Documentation/manual/examples </CsOptions> <CsInstruments> sr = 48000 ksmps = 32 nchnls = 2 instr 1 ktrans linseg 1, 5, 2, 10, -2 a1 diskin2 "beats.aiff", ktrans, 0, 1, 0, 32 outs a1, a1 endin </CsInstruments> <CsScore> i 1 0 15 e </CsScore> </CsoundSynthesizer>
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.
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).
dispfft -- displays the Fourier Transform of an audio or control signal (asig or ksig) every iprd seconds using the Fast Fourier Transform method.
Here is an example of the dispfft opcode. It uses the files dispfft.orc, dispfft.sco and beats.wav.
Example 106. Example of the dispfft opcode.
/* dispfft.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 asig soundin "beats.wav" dispfft asig, 1, 512 out asig endin /* dispfft.orc */
/* dispfft.sco */ ; Play Instrument #1 for three seconds. i 1 0 3 e /* dispfft.sco */
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.
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 (optional, default=1) -- 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).
display -- displays the audio or control signal xsig every iprd seconds, as an amplitude vs. time graph.
Here is an example of the display opcode. It uses the files display.orc and display.sco.
Example 107. Example of the display opcode.
/* display.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Go from 1000 to 0 linearly, over the period defined by p3. klin line 1000, p3, 0 ; Create a new display each second, wait for the user. display klin, 1, 1, 1 endin /* display.orc */
/* display.sco */ ; Play Instrument #1 for 5 seconds. i 1 0 5 e /* display.sco */
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)
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).
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.
Here is an example of the distort1 opcode. It uses the files distort1.orc and distort1.sco.
Example 108. Example of the distort1 opcode.
/* distort1.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 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 outs aout, aout gadist = 0 endin /* distort1.orc */
/* distort1.sco */ ; Sta Dur Amp Pitch i1 0.0 3.0 10000 6.00 i1 0.5 2.5 10000 7.00 i1 1.0 2.0 10000 7.07 i1 1.5 1.5 10000 8.00 ; Sta Dur PreGain PostGain Shape1 Shape2 i50 0 3 2 1 0 0 e /* distort1.sco */
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.
Here is an example of the divz opcode. It uses the files divz.orc and divz.sco.
Example 109. Example of the divz opcode.
/* divz.orc */ ; 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 /* divz.orc */
/* divz.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* divz.sco */
Its output should include lines like:
200.000000 / 0.000000 = -1.000000 200.000000 / 19.999887 = 10.000056 200.000000 / 40.000027 = 4.999997
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.
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.
Here is an example of the downsamp opcode. It uses the files downsamp.orc and downsamp.sco.
Example 110. Example of the downsamp opcode.
/* downsamp.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create a noise signal at a-rate. anoise noise 20000, 0.2 ; Downsample the noise signal to k-rate. knoise downsamp anoise ; Use the noise signal at k-rate. a1 oscil 30000, knoise, 1 out anoise endin /* downsamp.orc */
/* downsamp.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* downsamp.sco */
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.
ares dripwater kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] [, ifreq1] [, ifreq2]
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. Rasmus Ekman 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.
kamp -- Amplitude of output. Note: As these instruments are stochastic, this is only an approximation.
Here is an example of the dripwater opcode. It uses the files dripwater.orc and dripwater.sco.
Example 111. Example of the dripwater opcode.
/* dripwater.orc */ sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 01 ;example of a water drip a1 line 5, p3, 5 ;preset an amplitude boost a2 dripwater p4, 0.01, 0, .9 ;dripwater needs a little amplitude help at these values a3 product a1, a2 ;increase amplitude out a3 endin /* dripwater.orc */
/* dripwater.sco */ i1 0 1 20000 e /* dripwater.sco */
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.
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.
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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. |
ihandle - handle for the plugin returned by dssiinit
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.
ihandle - handle for the plugin returned by dssiinit
iport - control port number
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.
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.
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.
Here is an example of the dssinit opcode. It uses the file dssi4cs.csd.
Example 112. Example of the dssiinit opcode. (Remember to change the Library name)
<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>
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.
Periodically writes an orchestra control-signal value to a named external file in a specific format.
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 lsat two are binary. The output file contains no header information.
iprd -- the period of ksig output i 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.
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.
Periodically writes two orchestra control-signal values to a named external file in a specific format.
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 lsat two are binary. The output file contains no header information.
iprd -- the period of ksig output i 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.
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.
Periodically writes three orchestra control-signal values to a named external file in a specific format.
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 lsat two are binary. The output file contains no header information.
iprd -- the period of ksig output i 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.
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.
Periodically writes four orchestra control-signal values to a named external file in a specific format.
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 lsat two are binary. The output file contains no header information.
iprd -- the period of ksig output i 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.
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.
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
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.
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.
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).
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.
Any block of code that begins with an "if...then" statement must end with an endif statement.
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).
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There may be any number of instrument blocks in an orchestra. |
Here is an example of the endin opcode. It uses the files endin.orc and endin.sco.
Example 113. Example of the endin opcode.
/* endin.orc */ ; 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 /* endin.orc */
/* endin.sco */ ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* endin.sco */
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]
envlpx -- apply an envelope consisting of 3 segments:
stored function rise shape
modified exponential pseudo steady state
exponential decay
ares envlpx xamp, irise, idur, idec, ifn, iatss, iatdec [, ixmod]
kres envlpx kamp, irise, idur, idec, ifn, iatss, iatdec [, ixmod]
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).
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 the rise and decay periods overlap then that will cause a truncated decay. If the overall duration idur is exceeded in performance, the final decay will continue on in the same direction, tending asymptotically to zero.
Here is an example of the envlpx opcode. It uses the files envlpx.orc and envlpx.sco.
Example 114. Example of the envlpx opcode.
/* envlpx.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a simple instrument. instr 1 ; Set the amplitude. kamp init 20000 ; Get the frequency from the fourth p-field. kcps = cpspch(p4) a1 vco kamp, kcps, 1 out a1 endin ; Instrument #2 - instrument with an amplitude envelope. instr 2 kamp = 20000 irise = 0.05 idur = p3 - .01 idec = 0.5 ifn = 2 iatss = 1 iatdec = 0.01 ; Create an amplitude envelope. kenv envlpx kamp, irise, idur, idec, ifn, iatss, iatdec ; Get the frequency from the fourth p-field. kcps = cpspch(p4) a1 vco kenv, kcps, 1 out a1 endin /* envlpx.orc */
/* envlpx.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Table #2, a rising envelope. f 2 0 129 -7 0 128 1 ; Set the tempo to 120 beats per minute. t 0 120 ; Make sure the score plays for 33 seconds. f 0 33 ; Play a melody with Instrument #1. ; p4 = frequency in pitch-class notation. i 1 0 1 8.04 i 1 1 1 8.04 i 1 2 1 8.05 i 1 3 1 8.07 i 1 4 1 8.07 i 1 5 1 8.05 i 1 6 1 8.04 i 1 7 1 8.02 i 1 8 1 8.00 i 1 9 1 8.00 i 1 10 1 8.02 i 1 11 1 8.04 i 1 12 2 8.04 i 1 14 2 8.02 ; Repeat the melody with Instrument #2. ; p4 = frequency in pitch-class notation. i 2 16 1 8.04 i 2 17 1 8.04 i 2 18 1 8.05 i 2 19 1 8.07 i 2 20 1 8.07 i 2 21 1 8.05 i 2 22 1 8.04 i 2 23 1 8.02 i 2 24 1 8.00 i 2 25 1 8.00 i 2 26 1 8.02 i 2 27 1 8.04 i 2 28 2 8.04 i 2 30 2 8.02 e /* envlpx.sco */
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.
ares envlpxr xamp, irise, idec, ifn, iatss, iatdec [, ixmod] [,irind]
kres envlpxr kamp, irise, idec, ifn, iatss, iatdec [, ixmod] [,irind]
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 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).
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.
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.
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 “r” 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.
event "scorechar", kinsnum, kdelay, kdur, [, kp4] [, kp5] [, ...]
event "scorechar", "insname", kdelay, kdur, [, kp4] [, kp5] [, ...]
“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.
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.
Here is an example of the event opcode. It uses the files event.orc and event.sco.
Example 115. Example of the event opcode.
/* event.orc */ ; 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 /* event.orc */
/* event.sco */ ; Make sure the score plays for two seconds. f 0 2 ; Play Instrument #1 for a half-second. i 1 0 0.5 e /* event.sco */
Here is an example of the event opcode using a named instrument. It uses the files event_named.orc and event_named.sco.
Example 116. Example of the event opcode using a named instrument.
/* event_named.orc */ ; 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 /* event_named.orc */
/* event_named.sco */ ; Make sure the score plays for two seconds. f 0 2 ; Play Instrument #1 for a half-second. i 1 0 0.5 e /* event_named.sco */
event_i "scorechar", iinsnum, idelay, idur, [, ip4] [, ip5] [, ...]
event "scorechar", "insname", idelay, idur, [, ip4] [, ip5] [, ...]
“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.
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.
Here is an example of the exp opcode. It uses the files exp.orc and exp.sco.
Example 117. Example of the exp opcode.
/* exp.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = exp(8) print i1 endin /* exp.orc */
/* exp.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* exp.sco */
Its output should include a line like this:
instr 1: i1 = 2980.958
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.
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.
Here is an example of the expon opcode. It uses the files expon.orc and expon.sco.
Example 118. Example of the expon opcode.
/* expon.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Define kcps as a frequency value that exponentially declines ; from 880 to 220. It declines over the period set by p3. kcps expon 880, p3, 220 a1 oscil 20000, kcps, 1 out a1 endin /* expon.orc */
/* expon.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* expon.sco */
Exponential distribution random number generator (positive values only). This is an x-class noise generator.
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.
Here is an example of the exprand opcode. It uses the files exprand.orc and exprand.sco.
Example 119. Example of the exprand opcode.
/* exprand.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a random between 0 and 1. ; krange = 1 i1 exprand 1 print i1 endin /* exprand.orc */
/* exprand.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* exprand.sco */
Its output should include a line like this:
instr 1: i1 = 0.174
ares expseg ia, idur1, ib [, idur2] [, ic] [...]
kres expseg ia, idur1, ib [, idur2] [, ic] [...]
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.
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.
Here is an example of the expseg opcode. It uses the files expseg.orc and expseg.sco.
Example 120. Example of the expseg opcode.
/* expseg.orc */ ; 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 /* expseg.orc */
/* expseg.sco */ ; 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 /* expseg.sco */
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.
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.
These units generate control or 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.
Here is an example of the expsega opcode. It uses the files expsega.orc and expsega.sco.
Example 121. Example of the expsega opcode.
/* expsega.orc */ ; 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 /* expsega.orc */
/* expsega.sco */ ; 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 /* expsega.sco */
expsegr — Trace a series of exponential segments between specified points including a release segment.
Trace a series of exponential segments between specified points including a release segment.
ares expsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz
kres expsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz
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.
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.
Here is an example of the expsegr opcode. It uses the files expsegr.orc and expsegr.sco.
Example 122. Example of the expsegr opcode.
/* expsegr.orc */ ; 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) ; Use an amplitude envelope with second-long release. kenv expsegr 0.01, p3/2, 1, p3/2, 0.01, 1, 1 kamp = kenv * 30000 a1 oscil kamp, kcps, 1 out a1 endin /* expsegr.orc */
/* expsegr.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Make sure the score lasts for four seconds. f 0 4 ; p4 = frequency (in pitch-class notation). ; 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 /* expsegr.sco */
Here is an example of the filelen opcode. It uses the files filelen.orc, filelen.sco, and mary.wav.
Example 123. Example of the filelen opcode.
/* filelen.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the length of the audio file ; "mary.wav" in seconds. ilen filelen "mary.wav" print ilen endin /* filelen.orc */
/* filelen.sco */ ; Play Instrument #1 for 1 second. i 1 0 1 e /* filelen.sco */
The audio file “mary.wav” is 3.5 seconds long. So filelen's output should include a line like this:
instr 1: ilen = 3.501
Here is an example of the filenchnls opcode. It uses the files filenchnls.orc, filenchnls.sco, and mary.wav.
Example 124. Example of the filenchnls opcode.
/* filenchnls.orc */ ; 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". ichnls filenchnls "mary.wav" print ichnls endin /* filenchnls.orc */
/* filenchnls.sco */ ; Play Instrument #1 for 1 second. i 1 0 1 e /* filenchnls.sco */
The audio file “mary.wav” is monoaural (1 channel). So filenchnls's output should include a line like this:
instr 1: ichnls = 1.000
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
filepeak returns the peak absolute value of the sound file ifilcod. Currently, filepeak supports only AIFF-C float files.
Here is an example of the filepeak opcode. It uses the files filepeak.orc, filepeak.sco, and mary.wav.
Example 125. Example of the filepeak opcode.
/* filepeak.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the peak absolute value of the ; audio file "mary.wav". ipeak filepeak "mary.wav" print ipeak endin /* filepeak.orc */
/* filepeak.sco */ ; Play Instrument #1 for 1 second. i 1 0 1 e /* filepeak.sco */
The peak absolute value of the audio file “mary.wav” is 0.306902. So filepeak's output should include a line like this:
instr 1: ipeak = 0.307
Here is an example of the filesr opcode. It uses the files filesr.orc, filesr.sco, and mary.wav.
Example 126. Example of the filesr opcode.
/* filesr.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the sampling rate of the ; audio file "mary.wav". isr filesr "mary.wav" print isr endin /* filesr.orc */
/* filesr.sco */ ; Play Instrument #1 for 1 second. i 1 0 1 e /* filesr.sco */
The audio file “mary.wav” was sampled at 44.1 KHz. So filesr's output should include a line like this:
instr 1: isr = 44100.000
filter2 — Performs filtering using a transposed form-II digital filter lattice with no time-varying control.
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
ares filter2 asig, iM, iN, ib0, ib1, ..., ibM, ia1, ia2, ..., iaN
kres filter2 ksig, iM, iN, ib0, ib1, ..., ibM, ia1, ia2, ..., iaN
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.
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.
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.
0 - 32 bit floating points without header
1 - 16 bit integers without header
fin (file input) is the complement of fout: it reads a multichannel file to generate audio rate signals. At the present time no header is supported for the file format. The user must be sure that the number of channels of the input file is the same as the number of ainX arguments.
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.
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)
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.
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.
0 - 32 bit floating points without header
1 - 16 bit integers without header
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
fiopen opens a file to be used by the fout family of opcodes. It must be defined in the header section, external to any instruments. It returns a number, ihandle, which unequivocally refers to the opened 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.
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 connecting adel to an oscillator output. Also 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.
Here is an example of the flanger opcode. It uses the files flanger.orc, flanger.sco, and beats.wav.
Example 127. Example of the flanger opcode.
/* flanger.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use the "beat.wav" audio file. asig soundin "beats.wav" ; Vary the delay amount from 0 to 0.01 seconds. adel line 0, p3, 0.01 kfeedback = 0.7 ; Apply flange to the input signal. aflang flanger asig, adel, kfeedback ; It can get loud, so clip its amplitude to 30,000. a1 clip aflang, 1, 30000 out a1 endin /* flanger.orc */
/* flanger.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 e /* flanger.sco */
Allows text to be displayed from instruments like sliders etc. (only on Unix and Windows at present)
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.
Here is an example of the flashtxt opcode. It uses the files flashtxt.orc and flashtxt.sco.
Example 128. Example of the flashtxt opcode.
/* flashtxt.orc */ ; 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 /* flashtxt.orc */
/* flashtxt.sco */ ; Table 1: an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 e /* flashtxt.sco */
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.
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. |
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.
Here is an example of the FLbox opcode. It uses the files FLbox.orc and FLbox.sco.
Example 129. Example of the FLbox opcode.
/* flbox.orc */ 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 /* flbox.orc */
/* flbox.sco */ ; Real-time performance for 1 hour. f 0 3600 e /* flbox.sco */
kout, ihandle FLbutBank itype, inumx, inumy, iwidth, iheight, ix, iy, iopcode [, kp1] [, kp2] [, kp3] [, kp4] [, kp5] [....] [, kpN]
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. Its meaning is different for different types of widget.
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.
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 12, 8, 8, 380, 180, 50, 350, 0, 7, 0, 0, 5000, 6000
will create the this bank:
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).
FLbutBank is very useful to retrieve snapshots.
kout, ihandle FLbutton "label", ion, ioff, itype, iwidth, iheight, ix, iy, iopcode [, kp1] [, kp2] [, kp3] [, kp4] [, kp5] [....] [, kpN]
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.
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
This is the appearance of the buttons:
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.
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.
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).
Here is an example of the FLbutton opcode. It uses the files FLbutton.orc, FLbutton.sco, and beats.wav.
Example 130. Example of the FLbutton opcode.
/* flbutton.orc */ ; Using fl-buttons 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 = 1 FLpanel "Buttons", 320, 120, 100, 100 ion = 0 ioff = 0 itype = 1 iwidth = 50 iheight = 50 ix = 50 iy = 35 iopcode = 0 istarttim = 0 idur = -1 ; 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, 1, istarttim, idur, 0 ; Double speed backwards gkrew, ihb2 FLbutton "@<<", ion, ioff, itype, iwidth, iheight, ix+110, iy, iopcode, 1, istarttim, idur, -2 ; Double speed forwardS gkff, ihb2 FLbutton "@>>", ion, ioff, itype, iwidth, iheight, ix+165, iy, iopcode, 1, istarttim, idur, 2 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 diskin "beats.wav", p4, 0, 1 out asig endin /* flbutton.orc */
/* flbutton.sco */ ; A sine wave f 1 0 131072 10 1 ; Real-time performance for 1 hour. f 0 3600 e /* flbutton.sco */
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
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 without 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.
FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
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
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 without 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.
FLcolor, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
Allows the user to increase/decrease a value with mouse clicks on a corresponding arrow button.
kout, ihandle FLcount "label", imin, imax, istep1, istep2, itype, iwidth, iheight, ix, iy, iopcode [, kp1] [, kp2] [, kp3] [...] [, kpN]
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 coarse adjustments.
istep2 -- a floating-point number indicating the increment of valuator value corresponding to of each mouse click. istep2 is for fine 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.
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.
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.
Here is an example of the FLcount opcode. It uses the files FLcount.orc and FLcount.sco.
Example 131. Example of the FLcount opcode.
/* flcount.orc */ ; 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 /* flcount.orc */
/* flcount.sco */ ; 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 /* flcount.sco */
Retrieves a previously stored snapshot (in memory), i.e. sets all valuator to the corresponding values stored in that snaphot.
inumsnap -- current number of snapshots.
index -- a number referring unequivocally to a snapshot. Several snapshots can be stored in the same bank.
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).
“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. |
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.
FLgroupEnd, FLpack, FLpackEnd, FLpanel, FLpanelEnd, FLscroll, FLscrollEnd, FLtabs, FLtabsEnd
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.
Marks the end of a group of FLTK child widgets. This is another name for FLgroupEnd provides for compatibility. See FLgroupEnd
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.)
FLjoy is a squared area that allows the user to modify two output values at the same time. It acts like a joystick.
koutx, kouty, ihandlex, ihandley FLjoy "label", iminx, imaxx, iminy, imaxy, iexpx, iexpy, idispx, idispy, iwidth, iheight, ix, iy
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).
Here is an example of the FLjoy opcode. It uses the files FLjoy.orc and FLjoy.sco.
Example 132. Example of the FLjoy opcode.
/* fljoy.orc */ ; 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 /* fljoy.orc */
/* fljoy.sco */ ; 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 /* fljoy.sco */
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 pie knob.
A clock 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.
Here is an example of the FLknob opcode. It uses the files FLknob.orc and FLknob.sco.
Example 133. Example of the FLknob opcode.
/* flknob.orc */ ; 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 ; Height of the knob in pixels iheight = 70 ; Distance of the left edge of the knob ; from the left edge of the panel ix = 125 gkfreq, ihandle FLknob "Frequency", imin, imax, iexp, itype, idisp, iwidth, iheight, ix ; End of panel contents FLpanelEnd ; Run the widget thread! FLrun instr 1 iamp = 15000 ifn = 1 asig oscili iamp, gkfreq, ifn out asig endin /* flknob.orc */
/* flknob.sco */ ; 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 /* flknob.sco */
Modifies a set of parameters related to the text label appearence of a widget (i.e. size, font, alignment and color of corresponding text).
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
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.
FLcolor, FLcolor2, FLhide, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
FLloadsnap loads all the snapshots contained in a file into the memory bank of the current orchestra.
"filename" -- a double-quoted string corresponding to a file to load a bank of snapshots.
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.
istart -- loop start pos in seconds
idur -- loop duration in seconds
ifad -- crossfade duration in seconds
ifn -- function table number, generally created using GEN01
asig -- output sig
kon -- amplitude control
kpitch -- pitch control (transposition ratio); negative values play the loop back in reverse
Example 134. Example
aout flooper 16000, 1, 1, 4, 0.05 ; 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 the table 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).
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
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.
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.
FLgroup, FLgroupEnd, FLpackEnd, FLpanel, FLpanelEnd, FLscroll, FLscrollEnd, FLtabs, FLtabsEnd
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.
Marks the end of a group of compressed or aligned FLTK widgets. This is another name for FLpanelEnd provided for compatibility. See FLpanel_end
“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
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.
Here is an example of the FLpanel opcode. It uses the files FLpanel.orc and FLpanel.sco.
Example 135. Example of the FLpanel opcode.
/* flpanel.orc */ ; 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 /* flpanel.orc */
/* flpanel.sco */ ; 'Dummy' score event of 1 hour. f 0 3600 e /* flpanel.sco */
FLgroup, FLgroupEnd, FLpack, FLpackEnd, FLpanelEnd, FLscroll, FLscrollEnd, FLtabs, FLtabsEnd
FLpanelEnd — Marks the end of a group of FLTK widgets contained inside of a window (panel).
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_end — Marks the end of a group of FLTK widgets contained inside of a window (panel).
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
FLprintk is similar to printk but shows values of a k-rate signal in a text field instead of on the console.
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.
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.
FLprintk2 — A FLTK opcode that prints a new value every time a control-rate variable changes.
FLprintk2 is similar to FLprintk but shows a k-rate variable's value only when it changes.
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.
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.
kout, ihandle FLroller "label", imin, imax, istep, iexp, itype, idisp, iwidth, iheight, ix, iy
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).
Here is an example of the FLroller opcode. It uses the files FLroller.orc and FLroller.sco.
Example 136. Example of the FLroller opcode.
/* flroller.orc */ ; 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 /* flroller.orc */
/* flroller.sco */ ; 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 /* flroller.sco */
FLsavesnap saves all snapshots currently created (i.e. the entire memory bank) into a file.
“filename” -- a double-quoted string corresponding to a file to store a bank of snapshots.
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
"---------------------------"
.
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).
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.
Here is an example of the FLscroll opcode. It uses the files FLscroll.orc and FLscroll.sco.
Example 137. Example of the FLscroll opcode.
/* flscroll.orc */ ; 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 /* flscroll.orc */
/* flscroll.sco */ ; 'Dummy' score event of 1 hour. f 0 3600 e /* flscroll.sco */
FLgroup, FLgroupEnd, FLpack, FLpackEnd, FLpanel, FLpanelEnd, FLscrollEnd, FLtabs, FLtabsEnd
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.
A FLTK opcode that marks the end of an area with scrollbars. This is another name for FLscrollEnd provided for compatibility. See FLscrollEnd
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.
FLcolor, FLcolor2, FLhide, FLlabel, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
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.
FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
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.
Here is an example of the FLsetcolor opcode. It uses the files FLsetcolor.orc and FLsetcolor.sco.
Example 138. Example of the FLsetcolor opcode.
/* flsetcolor.orc */ ; 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 /* flsetcolor.orc */
/* flsetcolor.sco */ ; 'Dummy' score event for 1 hour. f 0 3600 e /* flsetcolor.sco */
FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
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.
FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
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.
FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
FLsetPosition sets the position of the target widget according to the ix and iy arguments.
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.
FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
FLsetSize resizes the target widget (not the size of its text) according to the iwidth and iheight arguments.
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.
FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
FLsetsnap stores the current status of all valuators present in the orchestra into a snapshot location (in memory).
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.
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.
FLsetText sets the label of the target widget to the double-quoted text string provided with the itext argument.
“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.
FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
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.
FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextSize, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
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.
FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextType, FLsetVal_i, FLsetVal, FLshow
FLsetTextType sets some attributes related to the fonts of the text label of the target widget.
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.
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.
FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetVal_i, FLsetVal, FLshow
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.
FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal, FLshow
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.
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.
FLcolor, FLcolor2, FLhide, FLlabel, FLsetAlign, FLsetBox, FLsetColor, FLsetColor2, FLsetFont, FLsetPosition, FLsetSize, FLsetText, FLsetTextColor, FLsetTextSize, FLsetTextType, FLsetVal_i, FLshow
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.
FLslidBnk "names", inumsliders [, ioutable] [, iwidth] [, iheight] [, ix] [, iy] [, itypetable] [, iexptable] [, istart_index] [, iminmaxtable]
“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).
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.
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.
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.
imax -- maximum value of output range.
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. 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 - 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 engraved slider (itype=3).
FLslider - a horizontal nice slider (itype=5).
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).
Here is an example of the FLslider opcode. It uses the files FLslider.orc and FLslider.sco.
Example 139. Example of the FLslider opcode.
/* flslider.orc */ ; 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 instr 1 iamp = 15000 ifn = 1 asig oscili iamp, gkfreq, ifn out asig endin /* flslider.orc */
/* flslider.sco */ ; 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 /* flslider.sco */
FLtabs is the “file card tabs” interface that allows 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.
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.
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.
It must be used together with FLgroup, another FLTK container opcode that groups child widgets.
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, rollers tab.
FLtabs example, joysticks tab.
(Each picture shows a different tab selection inside the same window.)
Here is an example of the FLtabs opcode. It uses the files FLtabs.orc and FLtabs.sco.
Example 140. Example of the FLtabs opcode.
/* fltabs.orc */ ; 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 /* fltabs.orc */
/* fltabs.sco */ ; 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 /* fltabs.sco */
FLgroup, FLgroupEnd, FLpack, FLpackEnd, FLpanel, FLpanelEnd, FLscroll, FLscrollEnd, FLtabsEnd
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.
Marks the end of a tabbed FLTK interface. This is another name for FLtabsEnd provided for compatibility. See FLtabsEnd
FLtext allows the user to modify a parameter value by directly typing it into a text field.
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 of each mouse click. The istep argument allows the user to arbitrarily slow roller's 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).
kout -- output value
FLtext allows the user to modify a parameter value by directly typing it into a text field:
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.
Here is an example of the FLtext opcode. It uses the files FLtext.orc and FLtext.sco.
Example 141. Example of the FLtext opcode.
/* fltext.orc */ ; 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 /* fltext.orc */
/* fltext.sco */ ; 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 /* fltext.sco */
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.
sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 1, 1, 1 ; Bright Steinway, program 1, channel 1 fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 2, 3, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidload "63.3mg The Sound Site Album Bank V1.0.SF2", 50, 2, 1 ; General MIDI, program 50, channel 2 ; 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 = 3 ikey = p4 ivelocity = p5 istatus = 144 fluidcontrol istatus, 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
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.
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)
This opcode is useful for setting controller values at init time. For continous changes, use fluidCCk.
sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 1, 1, 1 ; Bright Steinway, program 1, channel 1 fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 2, 3, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidload "63.3mg The Sound Site Album Bank V1.0.SF2", 50, 2, 1 ; General MIDI, program 50, channel 2 ; 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 = 3 ikey = p4 ivelocity = p5 istatus = 144 fluidcontrol istatus, 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 fluidout outs aleft * iamplitude, aright * iamplitude endin
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.
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
sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 1, 1, 1 ; Bright Steinway, program 1, channel 1 fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 2, 3, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidload "63.3mg The Sound Site Album Bank V1.0.SF2", 50, 2, 1 ; General MIDI, program 50, channel 2 ; 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 = 3 ikey = p4 ivelocity = p5 istatus = 144 fluidcontrol istatus, 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 fluidout outs aleft * iamplitude, aright * iamplitude endin
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.
ienginenum -- engine number assigned from fluidEngine
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. Note off messages need not be specified, as one is automatically generated when each Csound note expires or is released.
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.
sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 1, 1, 1 ; Bright Steinway, program 1, channel 1 fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 2, 3, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidload "63.3mg The Sound Site Album Bank V1.0.SF2", 50, 2, 1 ; General MIDI, program 50, channel 2 ; 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 = 3 ikey = p4 ivelocity = p5 istatus = 144 fluidControl istatus, 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 fluidout outs aleft * iamplitude, aright * iamplitude endin
Instantiates a fluidsynth engine, returning a number to identify the engine. ienginenum is used in conjunction with other opcodes for loading and playing SoundFonts and gathering the generated sound.
sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 1, 1, 1 ; Bright Steinway, program 1, channel 1 fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 2, 3, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidload "63.3mg The Sound Site Album Bank V1.0.SF2", 50, 2, 1 ; General MIDI, program 50, channel 2 ; 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 = 3 ikey = p4 ivelocity = p5 istatus = 144 fluidcontrol istatus, 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 fluidEngine outs aleft * iamplitude, aright * iamplitude endin
Loads a SoundFont into an instance of a fluidEngine, optionally listing banks and presets for SoundFont.
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.
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.
sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS fluidLoad "Piano Steinway Grand Model C (21,738KB).sf2", 1, 1, 1 ; Bright Steinway, program 1, channel 1 fluidLoad "Piano Steinway Grand Model C (21,738KB).sf2", 2, 3, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidLoad "63.3mg The Sound Site Album Bank V1.0.SF2", 50, 2, 1 ; General MIDI, program 50, channel 2 ; 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 = 3 ikey = p4 ivelocity = p5 istatus = 144 fluidcontrol istatus, 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 fluidout outs aleft * iamplitude, aright * iamplitude endin
Plays a note at imidikey pitch and imidivel velocity on ichannelnum channel of number ienginenum fluidEngine.
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)
sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 1, 1, 1 ; Bright Steinway, program 1, channel 1 fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 2, 3, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidload "63.3mg The Sound Site Album Bank V1.0.SF2", 50, 2, 1 ; General MIDI, program 50, channel 2 ; 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 = 3 ikey = p4 ivelocity = p5 istatus = 144 fluidcontrol istatus, 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 fluidNote outs aleft * iamplitude, aright * iamplitude endin
Outputs the sound from a fluidEngine.
Invoke fluidOut 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.
sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 1, 1, 1 ; Bright Steinway, program 1, channel 1 fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 2, 3, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidload "63.3mg The Sound Site Album Bank V1.0.SF2", 50, 2, 1 ; General MIDI, program 50, channel 2 ; 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 = 3 ikey = p4 ivelocity = p5 istatus = 144 fluidcontrol istatus, 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 fluidOut outs aleft * iamplitude, aright * iamplitude endin
fluidProgramSelect — Assigns a preset from a SoundFont to a channel on a fluidEngine.
sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 0dbfs = 32767 ; LOAD SOUNDFONTS fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 1, 1, 1 ; Bright Steinway, program 1, channel 1 fluidload "Piano Steinway Grand Model C (21,738KB).sf2", 2, 3, 1 ; Concert Steinway with reverb, program 2, channel 3 fluidload "63.3mg The Sound Site Album Bank V1.0.SF2", 50, 2, 1 ; General MIDI, program 50, channel 2 ; 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 = 3 ikey = p4 ivelocity = p5 istatus = 144 fluidcontrol istatus, 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 fluidProgramSelect outs aleft * iamplitude, aright * iamplitude endin
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).
Note that FLvalue is not a valuator and its value is fixed. Its value cannot be modified.
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.
Here is an example of the FLvalue opcode. It uses the files FLvalue.orc and FLvalue.sco.
Example 142. Example of the FLvalue opcode.
/* flvalue.orc */ ; 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 /* flvalue.orc */
/* flvalue.sco */ ; 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 /* flvalue.sco */
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.
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.
The initial waves should be:
ifn1 -- sine wave
ifn2 -- sine wave
ifn3 -- sine wave
ifn4 -- sine wave
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
Here is an example of the fmb3 opcode. It uses the files fmb3.orc and fmb3.sco.
Example 143. Example of the fmb3 opcode.
/* fmb3.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 15000 kfreq = 440 kc1 = 5 kc2 = 5 kvdepth = 0.005 kvrate = 6 ifn1 = 1 ifn2 = 1 ifn3 = 1 ifn4 = 1 ivfn = 1 a1 fmb3 kamp, kfreq, kc1, kc2, kvdepth, kvrate, \ ifn1, ifn2, ifn3, ifn4, ivfn out a1 endin /* fmb3.orc */
/* fmb3.sco */ ; Table #1, a sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* fmb3.sco */
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.
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 -- sine wave
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
Here is an example of the fmbell opcode. It uses the files fmbell.orc and fmbell.sco.
Example 144. Example of the fmbell opcode.
/* fmbell.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 10000 kfreq = 880 kc1 = 5 kc2 = 5 kvdepth = 0.005 kvrate = 6 ifn1 = 1 ifn2 = 1 ifn3 = 1 ifn4 = 1 ivfn = 1 a1 fmbell kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn out a1 endin /* fmbell.orc */
/* fmbell.sco */ ; Table #1, a sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 e /* fmbell.sco */
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.
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
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The file “twopeaks.aiff” is also available at ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. |
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
Here is an example of the fmmetal opcode. It uses the files fmmetal.orc, fmmetal.sco, and twopeaks.aiff.
Example 145. Example of the fmmetal opcode.
/* fmmetal.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 10000 kfreq = 440 kc1 = 6 kc2 = 5 kvdepth = 0 kvrate = 0 ifn1 = 1 ifn2 = 2 ifn3 = 2 ifn4 = 1 ivfn = 1 a1 fmmetal kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn out a1 endin /* fmmetal.orc */
/* fmmetal.sco */ ; Table #1, a normal sine wave. f 1 0 32768 10 1 ; Table #2, the "twopeaks.aiff" audio file. f 2 0 256 1 "twopeaks.aiff" 0 0 0 ; Play Instrument #1 for one second. i 1 0 1 e /* fmmetal.sco */
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.
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 -- sine wave
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
Here is an example of the fmpercfl opcode. It uses the files fmpercfl.orc and fmpercfl.sco.
Example 146. Example of the fmpercfl opcode.
/* fmpercfl.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 kfreq = 220 kc1 = 5 kc2 = 5 kvdepth = 0.005 kvrate = 6 ifn1 = 1 ifn2 = 1 ifn3 = 1 ifn4 = 1 ivfn = 1 a1 fmpercfl kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn out a1 endin /* fmpercfl.orc */
/* fmpercfl.sco */ ; Table #1, a sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* fmpercfl.sco */
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.
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
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The file “fwavblnk.aiff” is also available at ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. |
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
Here is an example of the fmrhode opcode. It uses the files fmrhode.orc, fmrhode.sco, and fwavblnk.aiff.
Example 147. Example of the fmrhode opcode.
/* fmrhode.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 kfreq = 220 kc1 = 6 kc2 = 0 kvdepth = 0.01 kvrate = 3 ifn1 = 1 ifn2 = 1 ifn3 = 1 ifn4 = 2 ivfn = 1 a1 fmrhode kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn out a1 endin /* fmrhode.orc */
/* fmrhode.sco */ ; Table #1, a sine wave. f 1 0 32768 10 1 ; Table #2, the "fwavblnk.aiff" audio file. f 2 0 256 1 "fwavblnk.aiff" 0 0 0 ; Play Instrument #1 for two seconds. i 1 0 2 e /* fmrhode.sco */
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
Here is an example of the fmvoice opcode. It uses the files fmvoice.orc and fmvoice.sco.
Example 148. Example of the fmvoice opcode.
/* fmvoice.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 kfreq = 110 ; Use the fourth p-field for the vowel. kvowel = p4 ktilt = 0 kvibamt = 0.005 kvibrate = 6 ifn1 = 1 ifn2 = 1 ifn3 = 1 ifn4 = 1 ivibfn = 1 a1 fmvoice kamp, kfreq, kvowel, ktilt, kvibamt, kvibrate, ifn1, ifn2, ifn3, ifn4, ivibfn out a1 endin /* fmvoice.orc */
/* fmvoice.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; p4 = vowel (a value from 0 to 64) ; Play Instrument #1 for one second, vowel=1. i 1 0 1 1 ; Play Instrument #1 for one second, vowel=2. i 1 1 1 2 ; Play Instrument #1 for one second, vowel=3. i 1 2 1 3 ; Play Instrument #1 for one second, vowel=4. i 1 3 1 4 ; Play Instrument #1 for one second, vowel=5. i 1 4 1 5 e /* fmvoice.sco */
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.
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
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The file “fwavblnk.aiff” is also available at ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. |
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
Here is an example of the fmwurlie opcode. It uses the files fmwurlie.orc, fmwurlie.sco, and fwavblnk.aiff.
Example 149. Example of the fmwurlie opcode.
/* fmwurlie.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 kfreq = 440 kc1 = 6 kc2 = 1 kvdepth = 0.005 kvrate = 6 ifn1 = 1 ifn2 = 1 ifn3 = 1 ifn4 = 2 ivfn = 1 a1 fmwurlie kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn out a1 endin /* fmwurlie.orc */
/* fmwurlie.sco */ ; Table #1, a sine wave. f 1 0 32768 10 1 ; Table #2, the "fwavblnk.aiff" audio file. f 2 0 256 1 "fwavblnk.aiff" 0 0 0 ; Play Instrument #1 for two seconds. i 1 0 2 e /* fmwurlie.sco */
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.
ares fof xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur [, iphs] [, ifmode] [, iskip]
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).
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.
Here is an example of the fof opcode. It uses the files fof.orc and fof.sco.
Example 150. Example of the fof opcode.
/* fof.orc */ /* Adapted from 1401.orc by Michael Clarke */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Combine five formants together to create ; an alto-"a" 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 = 14850 ifna = 1 ifnb = 2 itotdur = p3 ; First formant. k1amp = ampdb(0) k1form init 800 k1band init 80 ; Second formant. k2amp = ampdb(-4) k2form init 1150 k2band init 90 ; Third formant. k3amp = ampdb(-20) k3form init 2800 k3band init 120 ; Fourth formant. k4amp = ampdb(-36) k4form init 3500 k4band init 130 ; Fifth formant. k5amp = ampdb(-60) k5form init 4950 k5band init 140 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. out (a1+a2+a3+a4+a5) * 16384 endin /* fof.orc */
/* fof.sco */ /* Adapted from 1401.sco by Michael Clarke */ ; Table #1, a sine wave. f 1 0 4096 10 1 ; Table #2. f 2 0 1024 19 0.5 0.5 270 0.5 ; Play Instrument #1 for three seconds. i 1 0 3 e /* fof.sco */
The formant values for the alto-"a" sound were taken from the Formant Values Appendix.
fof2 — Produces sinusoid bursts including k-rate incremental indexing with each successive burst.
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.
ares fof2 xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss [, iskip]
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).
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 for 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 = -5/3 has the grain ending a perfect major sixth below. Note: There are no optional parameters in fof2
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.
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(?).
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.
asig -- input signal.
kcf -- filter centre frequency
kris -- impulse response attack time (secs).
kdec -- impulse response decay time (secs).
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.
ares fog xamp, xdens, xtrans, aspd, koct, kband, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur [, iphs] [, itmode] [, iskip]
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 iolaps.
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).
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 -- speed. The rate at which successive grains advance through the stored function table ifna. aspd is in the form of an index (0 to 1) to ifna. This 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.
The Csound fog generator is by Michael Clarke, extending his earlier work based on IRCAM's fof algorithm.
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.
Here is an example of the fold opcode. It uses the files fold.orc and fold.sco.
Example 152. Example of the fold opcode.
/* fold.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use an ordinary sine wave. asig oscils 30000, 100, 1 ; Vary the fold-over amount from 1 to 200. kincr line 1, p3, 200 a1 fold asig, kincr out a1 endin /* fold.orc */
/* fold.sco */ ; Play Instrument #1 for four seconds. i 1 0 4 e /* fold.sco */
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 )
Here is an example of the follow opcode. It uses the files follow.orc, follow.sco, and beats.wav.
Example 153. Example of the follow opcode.
/* follow.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - play a WAV file. instr 1 a1 soundin "beats.wav" out a1 endin ; Instrument #2 - have another waveform follow the WAV file. instr 2 ; Follow the WAV file. as soundin "beats.wav" af follow as, 0.01 ; Use a sine waveform. as oscil 4000, 440, 1 ; Have it use the amplitude of the followed WAV file. a1 balance as, af out a1 endin /* follow.orc */
/* follow.sco */ ; Just generate a nice, ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 2 2 e /* follow.sco */
To avoid zipper noise, by discontinuities produced from complex envelope tracking, a lowpass filter could be used, to smooth the estimated envelope.
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.
Here is an example of the follow2 opcode. It uses the files follow2.orc, follow2.sco, and beats.wav.
Example 154. Example of the follow2 opcode.
/* follow2.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - play a WAV file. instr 1 a1 soundin "beats.wav" out a1 endin ; Instrument #2 - have another waveform follow the WAV file. instr 2 ; Follow the WAV file. as soundin "beats.wav" af follow2 as, 0.01, 0.1 ; Use a noise waveform. ar rand 44100 ; Have it use the amplitude of the followed WAV file. a1 balance ar, af out a1 endin /* follow2.orc */
/* follow2.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for two seconds. i 2 2 2 e /* follow2.sco */
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.
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 = 1,1,2,... 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.
Here is an example of the foscil opcode. It uses the files foscil.orc and foscil.sco.
Example 155. Example of the foscil opcode.
/* foscil.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a basic FM waveform. instr 1 kamp = 10000 kcps = 440 kcar = 600 kmod = 210 kndx = 2 ifn = 1 a1 foscil kamp, kcps, kcar, kmod, kndx, ifn out a1 endin /* foscil.orc */
/* foscil.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* foscil.sco */
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.
xamp -- the amplitude of the output signal.
kcps -- the frequency of the output signal measured in cycles per second.
xcar -- the carrier frequency.
xmod -- 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.
Here is an example of the foscili opcode. It uses the files foscili.orc and foscili.sco.
Example 156. Example of the foscili opcode.
/* foscili.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a basic FM waveform. instr 1 kamp = 10000 kcps = 440 kcar = 600 kmod = 210 kndx = 2 ifn = 1 a1 foscil kamp, kcps, kcar, kmod, kndx, ifn out a1 endin ; Instrument #2 - the basic FM waveform with extra interpolation. instr 2 kamp = 10000 kcps = 440 kcar = 600 kmod = 210 kndx = 2 ifn = 1 a1 foscili kamp, kcps, kcar, kmod, kndx, ifn out a1 endin /* foscili.orc */
/* foscili.sco */ ; Table #1, a sine wave table with a small amount of data. f 1 0 4096 10 1 ; Play Instrument #1, the basic FM instrument, for ; two seconds. This should sound relatively rough. i 1 0 2 ; Play Instrument #2, the interpolated FM instrument, for ; two seconds. This should sound relatively smooth. i 2 2 2 e /* foscili.sco */
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.
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.
Here is a simple example of the fout opcode. It uses the files fout.orc and fout.sco.
Example 157. Example of the fout opcode.
/* fout.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 iamp = 10000 icps = 440 iphs = 0 ; Create an audio signal. asig oscils iamp, icps, iphs ; Write the audio signal to a headerless audio file ; called "fout.raw". fout "fout.raw", 1, asig endin /* fout.orc */
/* fout.sco */ ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* fout.sco */
Here is an example of the fout opcode with a polyphonic score. It uses the files fout_poly.orc, fout_poly.sco and beats.wav.
Example 158. Example of the fout opcode with a polyphonic score.
/* fout_poly.orc */ ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Initialize the global audio signal. gaudio init 0 ; Instrument #1 - Play an audio file. instr 1 ; Generate an audio signal using ; the audio file "beats.wav". asig soundin "beats.wav" ; Add this audio signal to the global one. vincr gaudio, 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 ; Add this audio signal to the global one. vincr gaudio, asig endin ; Instrument #99 - Save the global signal to a file. instr 99 ; Write the global audio signal to a headerless ; audio file called "fout_poly.raw". fout "fout_poly.raw", 1, gaudio ; Clear the global audio signal, preparing it ; for the next round. clear gaudio endin /* fout_poly.orc */
/* fout_poly.sco */ ; 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 /* fout_poly.sco */
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
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.
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
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.
foutk — Outputs k-rate signals of an arbitrary number of channels to a specified file, in raw (headerless) format.
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
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.
"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.
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.
Here is an example of the fprintks opcode. It uses the files fprintks.orc and fprintks.sco.
Example 159. Example of the fprintks opcode.
/* fprintks.orc */ /* 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 /* fprintks.orc */
/* fprintks.sco */ /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Play Instrument #1. i 1 0 0.001 /* fprintks.sco */
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
"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.
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.
Here is an example of the fprints opcode. It uses the files fprints.orc and fprints.sco.
Example 160. Example of the fprints opcode.
/* fprints.orc */ /* 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 /* fprints.orc */
/* fprints.sco */ /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Play Instrument #1. i 1 0 0.001 /* fprints.sco */
This example will generate a file called “my.sco”. It should contain a line like this:
;Generated score by ma++
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.
Here is an example of the frac opcode. It uses the files frac.orc and frac.sco.
Example 161. Example of the frac opcode.
/* frac.orc */ ; 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 /* frac.orc */
/* frac.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* frac.sco */
Its output should include a line like this:
instr 1: i2 = 0.200
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.
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.
ainL, ainR -- input signals; usually both are the same, but different inputs can be used for special effect
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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.
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.
Here is an example of the ftchnls opcode. It uses the files ftchnls.orc, ftchnls.sco, and mary.wav.
Example 162. Example of the ftchnls opcode.
/* ftchnls.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the number of channels in Table #1. ichnls = ftchnls(1) print ichnls endin /* ftchnls.orc */
/* ftchnls.sco */ ; Table #1: Use an audio file, Csound will determine its size. f 1 0 0 1 "mary.wav" 0 0 0 ; Play Instrument #1 for 1 second. i 1 0 1 e /* ftchnls.sco */
Since the audio file “mary.wav” is monophonic (1 channel), its output should include a line like this:
instr 1: ichnls = 1.000
ftconv — Low latency multichannel convolution, using a function table as impulse response source.
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.
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.
<CsoundSynthesizer> <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>
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.
This is equivalent to table generation in the score with the f statement.
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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. |
Here is an example of the ftgen opcode. It uses the files ftgen.orc and ftgen.sco.
Example 163. Example of the ftgen opcode.
/* ftgen.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Table #1, a sine wave using the GEN10 routine. gitemp ftgen 1, 0, 16384, 10, 1 ; Instrument #1 - a basic oscillator. instr 1 kamp = 10000 kcps = 440 ; Use Table #1. ifn = 1 a1 oscil kamp, kcps, ifn out a1 endin /* ftgen.orc */
/* ftgen.sco */ ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* ftgen.sco */
ftgentmp — Generate a score function table from within the orchestra, which is deleted at the end of the note.
Generate a score function table from within the orchestra, which is optionally deleted at the end of the note.
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.
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).
Here is an example of the ftlen opcode. It uses the files ftlen.orc, ftlen.sco, and mary.wav.
Example 164. Example of the ftlen opcode.
/* ftlen.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the size of Table #1. ; The size will be the number of points excluding the guard point. ilen = ftlen(1) print ilen endin /* ftlen.orc */
/* ftlen.sco */ ; Table #1: Use an audio file, Csound will determine its size. f 1 0 0 1 "mary.wav" 0 0 0 ; Play Instrument #1 for 1 second. i 1 0 1 e /* ftlen.sco */
The audio file “mary.wav” is 154390 samples long. The ftlen opcode reports it as 154389 samples long because it reserves 1 point for the guard point. Its output should include a line like this:
instr 1: ilen = 154389.000
"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.
"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.
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.
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The file's format is not compatible with a WAV-file and is not endian-safe. |
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.
Here is an example of the ftlptim opcode. It uses the files ftlptim.orc, ftlptim.sco, and mary.wav.
Example 165. Example of the ftlptim opcode.
/* ftlptim.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the loop-segment start time in Table #1. itim = ftlptim(1) print itim endin /* ftlptim.orc */
/* ftlptim.sco */ ; Table #1: Use an audio file, Csound will determine its size. f 1 0 0 1 "mary.wav" 0 0 0 ; Play Instrument #1 for 1 second. i 1 0 1 e /* ftlptim.sco */
Since the audio file “mary.wav” is non-looping, its output should include lines like this:
WARNING: non-looping sample instr 1: itim = 0.000
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.
iftfn -- The ftable function. The list of values are expected to be pre-existing ftable numbers.
iresfn -- Table number of the morphed function
The length of all the tables in iftfn must equal the length of iresfn.
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.
Here is an example of the ftmorf opcode. It uses the files ftmorf.orc and ftmorf.sco.
Example 166. Example of the ftmorf opcode.
/* ftmorf.orc */ sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 kndx line 0, p3, 7 ftmorf kndx, 1, 2 asig oscili 30000, 440, 2 out asig endin /* ftmorf.orc */
/* ftmorf.sco */ 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 10 e /* ftmorf.sco */
"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.
ftsave saves a list of tables to a file. The file's format can be binary or text.
![]() | Warning |
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The file's format is not compatible with a WAV-file and is not endian-safe. |
Here is an example of the ftsave opcode. It uses the files ftsave.orc and ftsave.sco.
Example 167. Example of the ftsave opcode.
/* ftsave.orc */ ; 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 /* ftsave.orc */
/* ftsave.sco */ ; Play Instrument #1 for 1 second. i 1 0 1 ; Play Instrument #2 for 1 second. i 2 2 1 e /* ftsave.sco */
"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.
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 |
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The file's format is not compatible with a WAV-file and is not endian-safe. |
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.
Here is an example of the ftsr opcode. It uses the files ftsr.orc, ftsr.sco, and mary.wav.
Example 168. Example of the ftsr opcode.
/* ftsr.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the sampling rate of Table #1. isr = ftsr(1) print isr endin /* ftsr.orc */
/* ftsr.sco */ ; Table #1: Use an audio file. f 1 0 262144 1 "mary.wav" 0 0 0 ; Play Instrument #1 for 1 second. i 1 0 1 e /* ftsr.sco */
Since the audio file “mary.wav” uses a 44.1 Khz sampling rate, its output should a line like this:
instr 1: isr = 44100.000
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.
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.
krange -- the range of the random numbers (-krange to +krange). 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.
Here is an example of the gauss opcode. It uses the files gauss.orc and gauss.sco.
Example 169. Example of the gauss opcode.
/* gauss.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a random number between -1 and 1. ; krange = 1 i1 gauss 1 print i1 endin /* gauss.orc */
/* gauss.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* gauss.sco */
Its output should include a line like this:
instr 1: i1 = 0.252
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
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.
Here is an example of the gbuzz opcode. It uses the files gbuzz.orc and gbuzz.sco.
Example 170. Example of the gbuzz opcode.
/* gbuzz.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 20000 kcps = 440 knh = 3 klh = 2 kmul = 0.7 ifn = 1 a1 gbuzz kamp, kcps, knh, klh, kmul, ifn out a1 endin /* gbuzz.orc */
/* gbuzz.sco */ ; Table #1, a simple cosine waveform. f 1 0 16384 11 1 ; Play Instrument #1 for one second. i 1 0 1 e /* gbuzz.sco */
Audio output is a tone related to the striking of a cow bell or similar. The method is a physical model developed from Perry Cook, but re-coded for Csound.
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.
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
Here is an example of the gogobel opcode. It uses the files gogobel.orc, gogobel.sco, and marmstk1.wav,
Example 171. Example of the gogobel opcode.
/* gogobel.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; kamp = 31129.60 ; kfreq = 440 ; ihrd = 0.5 ; ipos = 0.561 ; imp = 1 ; kvibf = 6.0 ; kvamp = 0.3 ; ivfn = 2 a1 gogobel 31129.60, 440, 0.5, 0.561, 1, 6.0, 0.3, 2 out a1 endin /* gogobel.orc */
/* gogobel.sco */ ; 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 e /* gogobel.sco */
goto label
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).
Here is an example of the goto opcode. It uses the files goto.orc and goto.sco.
Example 172. Example of the goto opcode.
/* goto.orc */ ; 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 /* goto.orc */
/* goto.sco */ ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* goto.sco */
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 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.
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 kamp. This means that the maximum amplitude a grain can have is kamp + kampoff and the minimum is kamp. If kampoff is set to zero then there is no random amplitude for each grain.
kpitchoff -- Maximum pitch deviation from kpitch 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.
This example generates a texture with gradually shorter grains and wider amp and pitch spread. It uses the files grain.orc, grain.sco, and mary.wav.
Example 173. Example of the grain opcode.
/* grain.orc */ ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 instr 1 insnd = 10 ibasfrq = 44100 / ftlen(insnd) ; Use original sample rate of insnd file kamp expseg 220, p3/2, 600, p3/2, 220 kpitch line ibasfrq, p3, ibasfrq * .8 kdens line 600, p3, 200 kaoff line 0, p3, 5000 kpoff line 0, p3, ibasfrq * .5 kgdur line .4, p3, .1 imaxgdur = .5 ar grain kamp, kpitch, kdens, kaoff, kpoff, kgdur, insnd, 5, imaxgdur, 0.0 out ar endin /* grain.orc */
/* grain.sco */ f5 0 512 20 2 ; Hanning window f10 0 262144 1 "mary.wav" 0 0 0 i1 0 6 e /* grain.sco */
Generate granular synthesis textures. grain2 is simpler to use, but grain3 offers more control.
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) ^ ((1 / irpow) - 1)
; for negative irpow values, it is
(1 - abs(x)) ^ ((-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.
iseed (optional, default=0) -- seed value for random number generator (positive integer in the range 1 to 2147483646 (2 ^ 31 - 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.
A diagram showing grains with a start time less than zero in red.
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 |
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grain2 internally uses the same random number generator as rnd31. So reading its documentation is also recommended. |
Here is an example of the grain2 opcode. It uses the files grain2.orc and grain2.sco.
Example 174. Example of the grain2 opcode.
/* grain2.orc */ 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 /* grain2.orc */
/* grain2.sco */ 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 /* grain2.sco */
Generate granular synthesis textures. grain2 is simpler to use but grain3 offers more control.
ares grain3 kcps, kphs, kfmd, kpmd, kgdur, kdens, imaxovr, kfn, iwfn, kfrpow, kprpow [, iseed] [, imode]
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).
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 ^ 31 - 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.
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 -- distribution of random frequency variation (see irpow).
kprpow -- distribution of random phase variation (see irpow). 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 |
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grain3 internally uses the same random number generator as rnd31. So reading its documentation is also recommended. |
Here is an example of the grain3 opcode. It uses the files grain3.orc and grain3.sco.
Example 175. Example of the grain3 opcode.
/* grain3.orc */ 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 /* grain3.orc */
/* grain3.sco */ t 0 60 i 1 0 3 i 2 4 3 i 3 8 3 e /* grain3.sco */
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.
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]
xamp -- amplitude.
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.
kgap -- gap between grains in sec.
igap_os -- gap random offset in % of the gap size, 0 gives no offset.
kgsize -- grain size in sec.
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.
Here is an example of the granule opcode. It uses the files granule.orc, granule.sco, and mary.wav.
Example 176. Example of the granule opcode.
/* granule.orc */ 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 /* granule.orc */
/* granule.sco */ ; f statement read sound file sine.aiff 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 /* granule.sco */
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. |
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.
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.
ifreq1 (optional) -- the first resonant frequency.
kamp -- Amplitude of output. Note: As these instruments are stochastic, this is only an approximation.
Here is an example of the guiro opcode. It uses the files guiro.orc and guiro.sco.
Example 177. Example of the guiro opcode.
/* guiro.orc */ sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 01 ;example of a guiro a1 guiro p4, 0.01 out a1 endin /* guiro.orc */
/* guiro.sco */ i1 0 1 20000 e /* guiro.sco */
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.
kestfrq -- estimated frequency of the input.
kmaxvar -- the maximum variance.
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.
Here is an example of the harmon opcode. It uses the files harmon.orc and harmon.sco.
Example 178. Example of the harmon opcode.
/* harmon.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; The frequency of the base note. inote = 440 ; Generate the base note. avco vco 20000, inote, 1 kestfrq = inote kmaxvar = 200 ; Calculate frequencies 3 semitones above and ; below the base note. kgenfreq1 = inote * semitone(3) kgenfreq2 = inote * semitone(-3) imode = 1 iminfrq = inote - 200 iprd = 0.1 ; Generate the harmony notes. a1 harmon avco, kestfrq, kmaxvar, kgenfreq1, kgenfreq2, \ imode, iminfrq, iprd out a1 endin /* harmon.orc */
/* harmon.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* harmon.sco */
asig -- input signal
ar1 -- cosine output of asig
ar2 -- sine 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.
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 files hilbert.orc, hilbert.sco, and mary.wav.
Example 179. Example of the hilbert opcode implementing frequency shifting.
/* hilbert.orc */ sr = 44100 kr = 4410 ksmps = 10 nchnls = 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 soundin "mary.wav" ; 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. out aupshift endin /* hilbert.orc */
/* hilbert.sco */ ; Sine table for quadrature oscillator. f 1 0 16384 10 1 ; Starting with no shift, ending with all ; frequencies shifted up by 200 Hz. i 1 0 2 0 200 ; Starting with no shift, ending with all ; frequencies shifted down by 200 Hz. i 1 2 2 0 -200 e /* hilbert.sco */
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 files hilbert_barberpole.orc, hilbert_barberpole.sco, and mary.wav.
Example 180. Example of the hilbert opcode sounding like a “barberpole phaser”.
/* hilbert_barberpole.orc */ ; 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 /* hilbert_barberpole.orc */
/* hilbert_barberpole.sco */ ; 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 /* hilbert_barberpole.sco */
The use of phase-difference networks in frequency shifters was pioneered by Harald Bode.1 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 analysis.3 A recent paper by Scott Wardle4 describes a digital implementation of a frequency shifter, as well as some unique applications.
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/.
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 above 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
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.
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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. |
Here is an example of the hrtfer opcode. It uses the files hrtfer.orc, hrtfer.sco, HRTFcompact, and beats.wav.
Example 181. Example of the hrtfer opcode.
/* hrtfer.orc */ ; 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 /* hrtfer.orc */
/* hrtfer.sco */ i 1 0 2 e /* hrtfer.sco */
An oscillator which takes tonality and brightness as arguments, relative to a base frequency.
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.
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.
Here is an example of the hsboscil opcode. It uses the files hsboscil.orc and hsboscil.sco.
Example 182. Example of the hsboscil opcode.
/* hsboscil.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 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 ; Instrument #1 - produces Risset's glissando. instr 1 kamp = 10000 kbrite = 0.5 ibasfreq = 200 ioctcnt = 5 ; Change ktone linearly from 0 to 1, ; over the period defined by p3. ktone line 0, p3, 1 a1 hsboscil kamp, ktone, kbrite, ibasfreq, giwave, giblend, ioctcnt out a1 endin /* hsboscil.orc */
/* hsboscil.sco */ ; Play Instrument #1 for ten seconds. i 1 0 10 e /* hsboscil.sco */
Here is an example of the hsboscil opcode in a MIDI instrument. It uses the files hsboscil_midi.orc and hsboscil_midi.sco.
Example 183. Example of the hsboscil opcode in a MIDI instrument.
/* hsboscil_midi.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 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 ; Instrument #1 - use hsboscil in a MIDI instrument. instr 1 ibase = cpsoct(6) ioctcnt = 5 ; all octaves sound alike. itona octmidi ; velocity is mapped to brightness ibrite ampmidi 3 ; Map an exponential envelope for the amplitude. kenv expon 20000, 1, 100 asig hsboscil kenv, itona, ibrite, ibase, giwave, giblend, ioctcnt out asig endin /* hsboscil_midi.orc */
/* hsboscil_midi.sco */ ; Play Instrument #1 for ten minutes i 1 0 6000 e /* hsboscil_midi.sco */
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.
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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. |
if ia R ib igoto label
if ka R kb kgoto label
if ia R ib 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).
Here is an example of the if...igoto combination. It uses the files igoto.orc and igoto.sco.
Example 184. Example of the if...igoto combination.
/* igoto.orc */ ; 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 /* igoto.orc */
/* igoto.sco */ ; 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 /* igoto.sco */
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 files kgoto.orc and kgoto.sco.
Example 185. Example of the if...kgoto combination.
/* kgoto.orc */ ; 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 /* kgoto.orc */
/* kgoto.sco */ ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* kgoto.sco */
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
Here is an example of the if...then combo. It uses the files ifthen.orc and ifthen.sco.
Example 186. Example of the if...then combo.
/* ifthen.orc */ 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 /* ifthen.orc */
/* ifthen.sco */ ; 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 /* ifthen.sco */
During the i-time pass only, unconditionally transfer control to the statement labeled by label.
igoto label
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).
Here is an example of the igoto opcode. It uses the files igoto.orc and igoto.sco.
Example 187. Example of the igoto opcode.
/* igoto.orc */ ; 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 /* igoto.orc */
/* igoto.sco */ ; 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 /* igoto.sco */
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
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.
Here is an example of the ihold opcode. It uses the files ihold.orc and ihold.sco.
Example 188. Example of the ihold opcode.
/* ihold.orc */ ; 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 /* ihold.orc */
/* ihold.sco */ ; 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 /* ihold.sco */
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.
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
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.
inch reads from a numbered channel determined by ksig1 into a1. 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.
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.
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)
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)
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)
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)
ichan -- MIDI channel (1-16)
ictlno -- controller number (0-127)
ivalue -- floating point value (must be within 0 to 1)
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)
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.
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.
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.
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.
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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).
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
If an instrument is defined with a name, you simply call it directly like an opcode:
asig MyOscil iamp, ipitch, iftable
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.
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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. |
Here is an example of the instr opcode. It uses the files instr.orc and instr.sco.
Example 189. Example of the instr opcode.
/* instr.orc */ ; 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 /* instr.orc */
/* instr.sco */ ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* instr.sco */
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.
Here is an example of the int opcode. It uses the files int.orc and int.sco.
Example 190. Example of the int opcode.
/* int.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = 16 / 5 i2 = int(i1) print i2 endin /* int.orc */
/* int.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* int.sco */
Its output should include a line like this:
instr 1: i2 = 3.000
iskip (optional) -- initial disposition of internal save space (see reson). The default value is 0.
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 * sin(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.
Here is an example of the integ opcode. It uses the files integ.orc and integ.sco.
Example 191. Example of the integ opcode.
/* integ.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 -- a differentiated signal. instr 1 ; Generate a band-limited pulse train. asrc buzz 20000, 440, 20, 1 ; Differentiate the signal. adiff diff asrc out adiff endin ; Instrument #2 -- a re-integrated signal. instr 2 ; Generate a band-limited pulse train. asrc buzz 20000, 440, 20, 1 ; Differentiate the signal. adiff diff asrc ; Re-integrate the previously differentiated signal. a1 integ adiff out a1 endin /* integ.orc */
/* integ.sco */ ; 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 1 1 e /* integ.sco */
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:
ksig -- input k-rate signal.
interp converts a control signal to an audio signal. It uses linear interpolation between successive kvals.
Here is an example of the interp opcode. It uses the files interp.orc and interp.sco.
Example 195. Example of the interp opcode.
/* interp.orc */ ; 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 /* interp.orc */
/* interp.sco */ ; 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 /* interp.sco */
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.
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.
Here is an example of the jitter opcode. It uses the files jitter.orc and jitter.sco.
Example 196. Example of the jitter opcode.
/* jitter.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Instrument #1 -- plain instrument. instr 1 aplain vco 20000, 220, 2, 0.83 outs aplain, aplain endin ; Instrument #2 -- instrument with jitter. instr 2 ; Create a signal modulated the jitter opcode. kamp init 2 kcpsmin init 4 kcpsmax init 6 kj jitter kamp, kcpsmin, kcpsmax aplain vco 20000, 220, 2, 0.83 ajitter vco 20000, 220+kj, 2, 0.83 outs aplain, ajitter endin /* jitter.orc */
/* jitter.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 3 seconds. i 1 0 3 ; Play Instrument #2 for 3 seconds. i 2 3 3 e /* jitter.sco */
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.
Here is an example of the jitter2 opcode. It uses the files jitter2.orc and jitter2.sco.
Example 197. Example of the jitter2 opcode.
/* jitter2.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 ; Instrument #1 -- plain instrument. instr 1 aplain vco 20000, 220, 2, 0.83 outs aplain, aplain endin ; Instrument #2 -- instrument with jitter. instr 2 ; Create a signal modulated with the jitter2 opcode. ktotamp init 2 kamp1 init 0.66 kcps1 init 3 kamp2 init 0.66 kcps2 init 3 kamp3 init 0.66 kcps3 init 3 kj jitter2 ktotamp, kamp1, kcps1, kamp2, kcps2, \ kamp3, kcps3 aplain vco 20000, 220, 2, 0.83 ajitter vco 20000, 220+kj, 2, 0.83 outs aplain, ajitter endin /* jitter2.orc */
/* jitter2.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 3 seconds. i 1 0 3 ; Play Instrument #2 for 3 seconds. i 2 3 3 e /* jitter2.sco */
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.
Converts an i-rate value to control rate, for example to be used with rnd() and birnd() to generate random numbers at k-rate.
During the p-time passes only, unconditionally transfer control to the statement labeled by label.
kgoto label
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).
Here is an example of the kgoto opcode. It uses the files kgoto.orc and kgoto.sco.
Example 198. Example of the kgoto opcode.
/* kgoto.orc */ ; 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 /* kgoto.orc */
/* kgoto.sco */ ; Table #1: a simple sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* kgoto.sco */
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
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.
kr = (optional) -- set control rate to iarg samples per second. The default value is 1000.
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 attempt to calculate the omitted value from the specified values, but it should evaluate to an integer.
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.
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 values, but it should evaluate to an integer.
![]() | Warning |
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ksmps must be an integer value. |
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.
Here is an example of the lfo opcode. It uses the files lfo.orc and lfo.sco.
Example 199. Example of the lfo opcode.
/* lfo.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 10 kcps = 5 itype = 4 k1 lfo kamp, kcps, itype ar oscil p4, p5+k1, 1 out ar endin /* lfo.orc */
/* lfo.sco */ ; Table #1: an ordinary sine wave. f 1 0 32768 10 1 ; p4 = amplitude of the output signal. ; p5 = frequency (in cycles per second) of the output signal. ; Play Instrument #1 for two seconds. i 1 0 2 10000 220 e /* lfo.sco */
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.
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.
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.
Here is an example of the line opcode. It uses the files line.orc and line.sco.
Example 200. Example of the line opcode.
/* line.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Define kcps as a frequency value that linearly declines ; from 880 to 220. It declines over the period set by p3. kcps line 880, p3, 220 a1 oscil 20000, kcps, 1 out a1 endin /* line.orc */
/* line.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* line.sco */
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.
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.
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.
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.
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.
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.
linenr 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 made independent by irind. Then it will begin a decay from wherever it was at the time.
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 “r” 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.
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.
Linear distribution random number generator (positive values only). This is an x-class noise generator.
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.
Here is an example of the linrand opcode. It uses the files linrand.orc and linrand.sco.
Example 201. Example of the linrand opcode.
/* linrand.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a random number between 0 and 1. ; krange = 1 i1 linrand 1 print i1 endin /* linrand.orc */
/* linrand.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* linrand.sco */
Its output should include a line like this:
instr 1: i1 = 0.394
ares linseg ia, idur1, ib [, idur2] [, ic] [...]
kres linseg ia, idur1, ib [, idur2] [, ic] [...]
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.
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.
Here is an example of the linseg opcode. It uses the files linseg.orc and linseg.sco.
Example 202. Example of the linseg opcode.
/* linseg.orc */ ; 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 linseg 0, p3*0.25, 1, p3*0.75, 0 kamp = kenv * 30000 a1 oscil kamp, kcps, 1 out a1 endin /* linseg.orc */
/* linseg.sco */ ; 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 /* linseg.sco */
linsegr — Trace a series of line segments between specified points including a release segment.
ares linsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz
kres linsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz
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.
Please note that the release time cannot be longer than 32767/kr seconds.
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.
Here is an example of the linsegr opcode. It uses the files linsegr.orc and linsegr.sco.
Example 203. Example of the linsegr opcode.
/* linsegr.orc */ ; 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) ; Use an amplitude envelope with second-long release. kenv linsegr 1, p3, 0.25, 1, 0 kamp = kenv * 30000 a1 oscil kamp, kcps, 1 out a1 endin /* linsegr.orc */
/* linsegr.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Make sure the score lasts for four seconds. f 0 4 ; p4 = frequency (in pitch-class notation). ; 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 /* linsegr.sco */
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!).
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.
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
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.
a1, a2 locsig asig, kdegree, kdistance, kreverbsend
a1, a2, a3, a4 locsig asig, kdegree, kdistance, kreverbsend
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 (ie.: 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.
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.
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
Returns the natural log of x (x positive only).
The argument value is restricted for log, log10, and sqrt.
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.
Here is an example of the log opcode. It uses the files log.orc and log.sco.
Example 204. Example of the log opcode.
/* log.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = log(8) print i1 endin /* log.orc */
/* log.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* log.sco */
Its output should include a line like this:
instr 1: i1 = 2.079
Returns the base 10 log of x (x positive only).
The argument value is restricted for log, log10, and sqrt.
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.
Here is an example of the log10 opcode. It uses the files log10.orc and log10.sco.
Example 205. Example of the log10 opcode.
/* log10.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = log10(8) print i1 endin /* log10.orc */
/* log10.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* log10.sco */
Its output should include a line like this:
instr 1: i1 = 0.903
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.
Here is an example of the logbtwo opcode. It uses the files logbtwo.orc and logbtwo.sco.
Example 206. Example of the logbtwo opcode.
/* logbtwo.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = logbtwo(3) print i1 endin /* logbtwo.orc */
/* logbtwo.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* logbtwo.sco */
Its output should include a line like this:
instr 1: i1 = 1.585
loop_lt indx, incr, imax, label
loop_lt kndx, kncr, kmax, label
loop_le indx, incr, imax, label
loop_le kndx, kncr, kmax, label
loop_gt indx, idecr, imin, label
loop_gt kndx, kdecr, kmin, label
loop_ge indx, idecr, imin, label
loop_ge kndx, kdecr, kmin, label
indx -- i-rate variable to count loop.
incr -- value to increment the loop (loop_lt, loop_le)
idecr -- value to decrement the loop (loop_gt, loop_ge)
imax -- maximum value of loop index (loop_lt, loop_le)
imin -- minimum value of loop index (loop_gt, loop_ge)
The actions of loop_lt is equivalent to the code
indx = indx + incr if (indx < imax) igoto label
or
kndx = kndx + kncr if (kndx < kmax) kgoto label
The actions of loop_le is equivalent to the code
indx = indx + incr if (indx <= imax) igoto label
or
kndx = kndx + kncr if (kndx <= kmax) kgoto label
The actions of loop_gt is equivalent to the code
indx = indx - idecr if (indx > imin) igoto label
or
kndx = kndx - kdecr if (kndx > kmin) kgoto label
The actions of loop_ge is equivalent to the code
indx = indx - idecr if (indx >= imin) igoto label
or
kndx = kndx - kdecr if (kndx >= kmin) kgoto label
loopseg — Generate control signal consisting of linear segments delimited by two or more specified points.
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.
ksig loopseg kfreq, ktrig, ktime0, kvalue0 [, ktime1] [, kvalue1] [, ktime2] [, kvalue2] [...]
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.
ktime0...ktimeN -- Times of points; expressed in fraction of a cycle.
kvalue0...kvalueN -- Values of points
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.
Here is an example of the loopseg opcode. It uses the files loopseg.orc and loopseg.sco.
Example 207. Example of the loopseg opcode.
/* loopseg.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 instr 1 kfreq line 1, p3, 20 klp loopseg kfreq, 0, 0, 0, 0.5, 30000, 1, 0 a1 oscil klp, 440, 1 out a1 endin /* loopseg.orc */
/* loopseg.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for five seconds. i 1 0 5 e /* loopseg.sco */
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.
ksig - output signal
kphase - NO INFORMATION
kvalue0 ...kvalueN - values of points
ktime0 ...ktimeN - times 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 a 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.
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.
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)
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
Here is an example of the lorenz opcode. It uses the files lorenz.orc and lorenz.sco.
Example 208. Example of the lorenz opcode.
/* lorenz.orc */ ; 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 220 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 /* lorenz.orc */
/* lorenz.sco */ ; Table #1 a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 5 seconds. i 1 0 5 e /* lorenz.sco */
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
Read sampled sound (mono or stereo) from a table, with optional sustain and release looping.
ares [,ar2] loscil xamp, kcps, ifn [, ibas] [, imod1] [, ibeg1] [, iend1] [, imod2,] [, ibeg2] [, iend2]
ifn -- function table number, typically denoting an AIFF sampled sound segment with prescribed looping points. 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 AIFF 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).
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.
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.
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.
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This is mono loscil: a1 loscil 10000, 1, 1 ...and this is stereo loscil: a1, a2 loscil 10000, 1, 1
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Here is an example of the loscil opcode. It uses the files loscil.orc, loscil.sco, and beats.aiff.
Example 209. Example of the loscil opcode.
/* loscil.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 ; If you don't know the frequency of your audio file, ; set both the kcps and ibas parameters equal to 1. kcps = 1 ifn = 1 ibas = 1 a1 loscil kamp, kcps, ifn, ibas out a1 endin /* loscil.orc */
/* loscil.sco */ ; Table #1: an audio file. f 1 0 262144 1 "beats.aiff" 0 4 0 ; Play Instrument #1 for 6 seconds. ; This will loop the audio file several times. i 1 0 6 e /* loscil.sco */
ares [,ar2] loscil3 xamp, kcps, ifn [, ibas] [, imod1] [, ibeg1] [, iend1] [, imod2] [, ibeg2] [, iend2]
ifn -- function table number, typically denoting an AIFF sampled sound segment with prescribed looping points. 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 AIFF 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).
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.
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.
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 experimental. It is identical to loscil except that it uses cubic interpolation. New in Csound version 3.50.
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This is mono loscil3: a1 loscil3 10000, 1, 1 ...and this is stereo loscil3: a1, a2 loscil3 10000, 1, 1
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Here is an example of the loscil3 opcode. It uses the files loscil3.orc, loscil3.sco, and beats.aiff.
Example 210. Example of the loscil3 opcode.
/* loscil3.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 ; If you don't know the frequency of your audio file, ; set both the kcps and ibas parameters equal to 1. kcps = 1 ifn = 1 ibas = 1 a1 loscil3 kamp, kcps, ifn, ibas out a1 endin /* loscil3.orc */
/* loscil3.sco */ ; Table #1: an audio file. f 1 0 131072 1 "beats.aiff" 0 4 0 ; Play Instrument #1 for 6 seconds. ; This will loop the drum pattern several times. i 1 0 6 e /* loscil3.sco */
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.
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.
Here is an example of the lowpass2 opcode. It uses the files lowpass2.orc and lowpass2.sco.
Example 211. Example of the lowpass2 opcode.
/* lowpass.orc */ /* 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 asig * kenv endin /* lowpass.orc */
/* lowpass2.sco */ /* Written by Sean Costello */ f1 0 8192 9 1 1 .25 i1 0 5 100 1000 i1 5 5 200 1000 e /* lowpass2.sco */
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.
asig -- input signal
kcutoff -- filter cutoff frequency point
kresonance -- resonance amount
lowres is a resonant lowpass filter derived from a Hans Mikelson 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.
Here is an example of the lowres opcode. It uses the files lowres.orc, lowres.sco and beats.wav.
Example 212. Example of the lowres opcode.
/* lowres.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a nice sawtooth waveform. asig vco 5000, 440, 1 ; Vary the cutoff frequency from 30 to 300 Hz. kcutoff line 30, p3, 300 kresonance = 10 ; Apply the filter. a1 lowres asig, kcutoff, kresonance out a1 endin /* lowres.orc */
/* lowres.sco */ ; Table #1, a sine wave for the vco opcode. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* lowres.sco */
lowresx is equivalent to more layers of lowres with the same arguments serially connected.
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.
asig -- input signal
kcutoff -- filter cutoff frequency point
kresonance -- 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 Hans Mikelson
Here is an example of the lowresx opcode. It uses the files lowresx.orc, lowresx.sco, and beats.wav.
Example 213. Example of the lowresx opcode.
/* lowresx.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - play the sawtooth waveform through a ; stack of filters. instr 1 ; Use a nice sawtooth waveform. asig vco 5000, 440, 1 ; Vary the cutoff frequency from 30 to 300 Hz. kcutoff line 30, p3, 300 kresonance = 3 inumlayer = 2 alr lowresx asig, kcutoff, kresonance, inumlayer ; It gets loud, so clip the output amplitude to 30,000. a1 clip alr, 1, 30000 out a1 endin /* lowresx.orc */
/* lowresx.sco */ ; Table #1, a sine wave for the vco opcode. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* lowresx.sco */
kfco -- the filter cutoff frequency in Hz. Should be in the range 0 to sr/2.
kres -- the amount of resonance. Self-oscillation occurs when kres is approximately 1. Shoujld usually be in the range 0 to 1, however, values slightly greater than 1 are possible for more sustained oscillation and an “overdrive” effect.
kdist -- amount of distortion. kdist = 0 gives a clean output. kdist > 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.
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This filter requires that the input signal be normalized to one. |
Here is an example of the lpf18 opcode. It uses the files lpf18.orc and lpf18.sco.
Example 214. Example of the lpf18 opcode.
/* lpf18.orc */ ; 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 init 0.3 aout lpf18 asine, kfco, kres, kdist out aout * 30000 endin /* lpf18.orc */
/* lpf18.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for four seconds. i 1 0 4 e /* lpf18.sco */
lpfreson — Resynthesises a signal from the data passed internally by a previous lpread, applying formant shifting.
Resynthesises a signal from the data passed internally by a previous lpread, applying formant shifting.
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.
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.
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.
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.
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
lposcil, lposcil3 — Read sampled sound from a table with optional looping and high precision.
Read sampled sound (mono or stereo) from a table, with optional sustain and release looping, and high precision.
kamp -- amplitude
kfreqratio -- multiply factor of table frequency (for example: 1 = original frequency, 1.5 = a fifth up , .5 = an octave down)
kloop -- 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.
Read sampled sound (mono or stereo) from a table, with optional sustain and release looping, and high precision. lposcil3 uses cubic interpolation.
kamp -- amplitude
kfreqratio -- multiply factor of table frequency (for example: 1 = original frequency, 1.5 = a fifth up , .5 = an octave down)
kloop -- 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.
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.
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.
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.
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.
ksig lpshold kfreq, ktrig, ktime0, kvalue0 [, ktime1] [, kvalue1] [, ktime2] [, kvalue2] [...]
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.
ktime0...ktimeN -- Times of points; expressed in fraction of a cycle
kvalue0...kvalueN -- Values of points
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.
Here is an example of the lpshold opcode. It uses the files lpshold.orc and lpshold.sco.
Example 215. Example of the lpshold opcode.
/* lpshold.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 instr 1 kfreq line 1, p3, 20 klp lpshold kfreq, 0, 0, 0, p3*0.25, 20000, p3*0.75, 0 a1 oscil klp, 440, 1 out a1 endin /* lpshold.orc */
/* lpshold.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for five seconds. i 1 0 5 e /* lpshold.sco */
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.
ksig lpsholdp kphase, ktrig, ktime0, kvalue0 [, ktime1] [, kvalue1] [, ktime2] [, kvalue2] [...]
ksig - output signal
kphase -
kvalue0 ...kvalueN - values of points
ktime0 ...ktimeN - times of points expessed in fraction of a cycle
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.
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.
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
ares madsr iatt, idec, islev, irel [, idel] [, ireltim]
kres madsr iatt, idec, islev, irel [, idel] [, ireltim]
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.
The envelope is the range 0 to 1 and may need to be scaled further. The envelope may be described as:
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.
Here is an example of the madsr opcode. It uses the files madsr.orc and madsr.sco.
Example 216. Example of the madsr opcode.
/* madsr.orc */ /* 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 /* madsr.orc */
/* madsr.sco */ /* 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 /* madsr.sco */
Returns the number of iterations corresponding to a given point of complex plane by applying the Mandelbrot set formula.
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.
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.
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.
Here is an example of the mandol opcode. It uses the files mandol.orc, mandol.sco, and mandpluk.aiff.
Example 217. Example of the mandol opcode.
/* mandol.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; kamp = 30000 ; kfreq = 880 ; kpluck = 0.4 ; kdetune = 0.99 ; kgain = 0.99 ; ksize = 2 ; ifn = 1 ; ifreq = 220 a1 mandol 30000, 880, 0.4, 0.99, 0.99, 2, 1, 220 out a1 endin /* mandol.orc */
/* mandol.sco */ ; Table #1: the "mandpluk.aiff" audio file f 1 0 8192 1 "mandpluk.aiff" 0 0 0 ; Play Instrument #1 for one second. i 1 0 1 e /* mandol.sco */
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 Perry Cook but re-coded for Csound.
ares marimba kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec [, idoubles] [, itriples]
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%.
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
Here is an example of the marimba opcode. It uses the files marimba.orc, marimba.sco, and marmstk1.wav.
Example 218. Example of the marimba opcode.
/* marimba.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; kamp = 31129.60 ; kfreq = 440 ; ihrd = 0.5 ; ipos = 0.561 ; imp = 1 ; kvibf = 6.0 ; kvamp = 0.05 ; ivibfn = 2 ; idec = 0.1 a1 marimba 31129.60, 440, 0.5, 0.561, 1, 6.0, 0.05, 2, 0.1 out a1 endin /* marimba.orc */
/* marimba.sco */ ; 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 e /* marimba.sco */
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).
maxabs — Produces a signal that is the maximum of the absolute values of any number of input signals.
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).
amax maxabs ain1 [, ain2] [, ain3] [, ain4] [...]
kmax maxabs kin1 [, kin2] [, kin3] [, kin4] [...]
maxabsaccum compares two audio-rate variables and stores the maximum of their absolute values into the first.
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.
maxaccum compares two audio-rate variables and stores the maximum value between them into the first.
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.
All instances of maxalloc must be defined in the header section, not in the instrument body.
Here is an example of the maxalloc opcode. It uses the files maxalloc.orc and maxalloc.sco.
Example 219. Example of the maxalloc opcode.
/* maxalloc.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Limit Instrument #1 to three instances. maxalloc 1, 3 ; 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 /* maxalloc.orc */
/* maxalloc.sco */ ; 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 /* maxalloc.sco */
Its output should contain a message like this:
WARNING: cannot allocate last note because it exceeds instr maxalloc
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.
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
This opcode can be useful in several situations, for example to implement a vu-meter.
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.
Generate a metronomic signal to be used in any circumstance an isochronous trigger is needed.
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.
Here is an example of the metro opcode. It uses the file metro.csd
Example 220. Example of the metro opcode.
<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>
midic14 — 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]
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.
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.
midic21 — 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]
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.
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.
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.
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.
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.
ilow (optional) -- optional low value after rescaling, defaults to 0.
ihigh (optional) -- optional high value after rescaling, defaults to 127.
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. |
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To adapt an ordinary Csound instrument designed for score activation for score/MIDI interoperability:
|
Here is an example of the midichannelaftertouch opcode. It uses the files midichannelaftertouch.orc and midichannelaftertouch.sco.
Example 221. Example of the midichannelaftertouch opcode.
/* midichannelaftertouch.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kaft init 0 midichannelaftertouch kaft ; Display the aftertouch value when it changes. printk2 kaft endin /* midichannelaftertouch.orc */
/* midichannelaftertouch.sco */ ; Play Instrument #1 for ten seconds. i 1 0 10 e /* midichannelaftertouch.sco */
Its output should include lines like:
i1 127.00000 i1 20.00000 i1 44.00000
midichn returns the MIDI channel number (1 - 16) from which the note was activated. In the case of score notes, it returns 0.
ichn -- channel number. If the current note was activated from score, it is set to zero.
Here is a simple example of the midichn opcode. It uses the files midichn.orc and midichn.sco.
Example 222. Example of the midichn opcode.
/* midichn.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 midichn print i1 endin /* midichn.orc */
/* midichn.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* midichn.sco */
Here is an advanced example of the midichn opcode. It uses the files midichn_advanced.mid, midichn_advanced.orc, and midichn_advanced.sco.
Don't forget that you must include the -F flag when using an external MIDI file like “midichn_advanced.mid”.
Example 223. An advanced example of the midichn opcode.
/* midichn_advanced.orc - written by Istvan Varga */ sr = 44100 ksmps = 10 nchnls = 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: endin /* midichn_advanced.orc - written by Istvan Varga */
/* midichn_advanced.sco - written by Istvan Varga */ t 0 60 f 0 6 2 -2 0 i 1 1 0.5 i 1 4 0.5 e /* midichn_advanced.sco - written by Istvan Varga */
Its output should include lines like:
note 7 (time = 0.00) was activated from channel 4 note 8 (time = 0.00) was activated from channel 2
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.
ilow (optional) -- optional low value after rescaling, defaults to 0.
ihigh (optional) -- optional high value after rescaling, defaults to 127.
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. |
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To adapt an ordinary Csound instrument designed for score activation for score/MIDI interoperability:
|
inum -- MIDI controller number (0-127)
imin, imax -- set minimum and maximum limits on values obtained.
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.
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.
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. |
---|---|
To adapt an ordinary Csound instrument designed for score activation for score/MIDI interoperability:
|
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.
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.
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. |
---|---|
To adapt an ordinary Csound instrument designed for score activation for score/MIDI interoperability:
|
Here is an example of the midinoteoff opcode. It uses the files midinoteoff.orc and midinoteoff.sco.
Example 224. Example of the midinoteoff opcode.
/* midinoteoff.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kkey init 0 kvelocity init 0 midinoteoff kkey, kvelocity ; Display the key value when it changes. printk2 kkey endin /* midinoteoff.orc */
/* midinoteoff.sco */ ; Play Instrument #1 for ten seconds. i 1 0 10 e /* midinoteoff.sco */
Its output should include lines like:
i1 60.00000 i1 76.00000
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.
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. |
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To adapt an ordinary Csound instrument designed for score activation for score/MIDI interoperability:
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Here is an example of the midinoteoncps opcode. It uses the files midinoteoncps.orc and midinoteoncps.sco.
Example 225. Example of the midinoteoncps opcode.
/* midinoteoncps.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kcps init 0 kvelocity init 0 midinoteoncps kcps, kvelocity ; Display the cycles-per-second value when it changes. printk2 kcps endin /* midinoteoncps.orc */
/* midinoteoncps.sco */ ; Play Instrument #1 for ten seconds. i 1 0 10 e /* midinoteoncps.sco */
Its output should include lines like:
i1 261.62561 i1 440.00006
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.
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. |
---|---|
To adapt an ordinary Csound instrument designed for score activation for score/MIDI interoperability:
|
Here is an example of the midinoteonkey opcode. It uses the files midinoteonkey.orc and midinoteonkey.sco.
Example 226. Example of the midinoteonkey opcode.
/* midinoteonkey.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kkey init 0 kvelocity init 0 midinoteonkey kkey, kvelocity ; Display the key value when it changes. printk2 kkey endin /* midinoteonkey.orc */
/* midinoteonkey.sco */ ; Play Instrument #1 for ten seconds. i 1 0 10 e /* midinoteonkey.sco */
Its output should include lines like:
i1 60.00000 i1 69.00000
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.
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. |
---|---|
To adapt an ordinary Csound instrument designed for score activation for score/MIDI interoperability:
|
Here is an example of the midinoteonoct opcode. It uses the files midinoteonoct.orc and midinoteonoct.sco.
Example 227. Example of the midinoteonoct opcode.
/* midinoteonoct.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 koct init 0 kvelocity init 0 midinoteonoct koct, kvelocity ; Display the octave-point-decimal value when it changes. printk2 koct endin /* midinoteonoct.orc */
/* midinoteonoct.sco */ ; Play Instrument #1 for ten seconds. i 1 0 10 e /* midinoteonoct.sco */
Its output should include lines like:
i1 8.00000 i1 9.33333
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.
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. |
---|---|
To adapt an ordinary Csound instrument designed for score activation for score/MIDI interoperability:
|
Here is an example of the midinoteonpch opcode. It uses the files midinoteonpch.orc and midinoteonpch.sco.
Example 228. Example of the midinoteonpch opcode.
/* midinoteonpch.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kpch init 0 kvelocity init 0 midinoteonpch kpch, kvelocity ; Display the pitch-class value when it changes. printk2 kpch endin /* midinoteonpch.orc */
/* midinoteonpch.sco */ ; Play Instrument #1 for ten seconds. i 1 0 10 e /* midinoteonpch.sco */
Its output should include lines like:
i1 8.09000 i1 9.05000
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.
Sends noteon and noteoff messages to the MIDI OUT port when triggered by a value different than zero.
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.
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 |
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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. |
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.
ilow (optional) -- optional low value after rescaling, defaults to 0.
ihigh (optional) -- optional high value after rescaling, defaults to 127.
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. |
---|---|
To adapt an ordinary Csound instrument designed for score activation for score/MIDI interoperability:
|
Here is an example of the midipitchbend opcode. It uses the files midipitchbend.orc and midipitchbend.sco.
Example 229. Example of the midipitchbend opcode.
/* midipitchbend.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kpb init 0 midipitchbend kpb ; Display the pitch-bend value when it changes. printk2 kpb endin /* midipitchbend.orc */
/* midipitchbend.sco */ ; Play Instrument #1 for ten seconds. i 1 0 10 e /* midipitchbend.sco */
Its output should include lines like:
i1 0.12695 i1 0.00000 i1 -0.01562
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.
ilow (optional) -- optional low value after rescaling, defaults to 0.
ihigh (optional) -- optional high value after rescaling, defaults to 127.
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. |
---|---|
To adapt an ordinary Csound instrument designed for score activation for score/MIDI interoperability:
|
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.
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. |
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To adapt an ordinary Csound instrument designed for score activation for score/MIDI interoperability:
|
miditempo — Returns the current tempo at k-rate, of either the MIDI file (if available) or the score
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).
minabs — Produces a signal that is the minimum of the absolute values of any number of input signals.
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).
amin minabs ain1 [, ain2] [, ain3] [, ain4] [...]
kmin minabs kin1 [, kin2] [, kin3] [, kin4] [...]
minabsaccum compares two audio-rate variables and stores the minimum of their absolute values into the first.
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.
minaccum compares two audio-rate variables and stores the minimum value between them into the first.
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.
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.
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.
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.
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
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.
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.
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.
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.
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.
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
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.
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.
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
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.
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
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 |
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The files “mandpluk.aiff” and “impuls20.aiff” are also available at ftp://ftp.cs.bath.ac.uk/pub/dream/documentation/sounds/modelling/. |
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
Here is an example of the moog opcode. It uses the files moog.orc, moog.sco, mandpluk.aiff, and impuls20.aiff.
Example 230. Example of the moog opcode.
/* moog.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 kfreq = 220 kfiltq = 0.81 kfiltrate = 0 kvibf = 1.4 kvamp = 2.22 iafn = 1 iwfn = 2 ivfn = 3 am moog kamp, kfreq, kfiltq, kfiltrate, kvibf, kvamp, iafn, iwfn, ivfn ; It tends to get loud, so clip moog's amplitude at 30,000. a1 clip am, 2, 30000 out a1 endin /* moog.orc */
/* moog.sco */ ; Table #1: the "mandpluk.aiff" audio file f 1 0 8192 1 "mandpluk.aiff" 0 0 0 ; Table #2: the "impuls20.aiff" audio file f 2 0 256 1 "impuls20.aiff" 0 0 0 ; Table #3: a sine wave f 3 0 256 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 e /* moog.sco */
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.
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.
asig -- input signal.
kcf -- filter cutoff frequency
kres -- 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.
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)
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 Implemnetation” by Stilson and Smith (CCRMA). This version was originally coded in Csound by Josep Comajuncosas. Some modifications and conversion to C were done by Hans Mikelson
Note: This filter requires that the input signal be normalized to one.
Here is an example of the moogvcf opcode. It uses the files moogvcf.orc and moogvcf.sco.
Example 232. Example of the moogvcf opcode.
/* moogvcf.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a nice sawtooth waveform. asig vco 32000, 220, 1 ; Vary the filter-cutoff frequency from .2 to 2 KHz. kfco line 200, p3, 2000 ; Set the resonance amount to one. krez init 1 ; Scale the amplitude to 32768. iscale = 32768 a1 moogvcf asig, kfco, krez, iscale out a1 endin /* moogvcf.orc */
/* moogvcf.sco */ ; Table #1, a sine wave for the vco opcode. f 1 0 16384 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 e /* moogvcf.sco */
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.
Generates a set of impulses of amplitude kamp at frequency kfreq. The first impulse is after a delay of ioffset seconds. The value of kfreq is read only after an impulse, so it is the interval to the next impulse at the time of an impulse.
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.
kamp -- amplitude of the impulses generated
kfreq -- frequency of the impulse train
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 calculated. If kfreq is zero, there is an infinite wait to the next impulse. If kfreq is negative, the frequency is counted in samples rather than seconds.
Here is an example of the mpulse opcode. It uses the files mpulse.orc and mpulse.sco.
Example 233. Example of the mpulse opcode.
/* mpulse.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate an impulse every 1/10th of a second. kamp = 30000 kfreq = 0.1 a1 mpulse kamp, kfreq out a1 endin /* mpulse.orc */
/* mpulse.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* mpulse.sco */
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)
The arguments itime and igain set the position and gain of each tap.
The delay line is fed by asig.
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.
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).
Here is an example of the mute opcode. It uses the files mute.orc and mute.sco.
Example 234. Example of the mute opcode.
/* mute.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Mute Instrument #2. mute 2 ; Instrument #1. instr 1 a1 oscils 10000, 440, 0 out a1 endin ; Instrument #2. instr 2 a1 oscils 10000, 880, 0 out a1 endin /* mute.orc */
/* mute.sco */ ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e /* mute.sco */
ares mxadsr iatt, idec, islev, irel [, idel] [, ireltim]
kres mxadsr iatt, idec, islev, irel [, idel] [, ireltim]
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)
The envelope is the range 0 to 1 and may need to be scaled further. The envelope may be described as:
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.
mxadsr is new in Csound version 3.51.
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.
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.
ares nestedap asig, imode, imaxdel, idel1, igain1 [, idel2] [, igain2] [, idel3] [, igain3] [, istor]
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).
asig -- input signal
If imode = 1, the filter takes the form:
Picture of imode 1 filter.
If imode = 2, the filter takes the form:
Picture of imode 2 filter.
If imode = 3, the filter takes the form:
Picture of imode 3 filter.
Here is an example of the nestedap opcode. It uses the files nestedap.orc, nestedap.sco, and beats.wav.
Example 235. Example of the nestedap opcode.
/* nestedap.orc */ sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 5 insnd = p4 gasig diskin 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 /* nestedap.orc */
/* nestedap.sco */ 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 /* nestedap.sco */
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)
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 |
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The "useful" ranges of parameters are not yet mapped. |
ioffset -- 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.
xamp -- amplitude of final output
kbeta -- beta of the lowpass filter. Should be in the range of 0 to 1.
The filter equation is:
y_n = sqrt(1-beta^2) * x_n + beta Y_(n-1)
where x_n is white noise.
Here is an example of the noise opcode. It uses the files noise.orc and noise.sco.
Example 236. Example of the noise opcode.
/* noise.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 ; Change the beta value linearly from 0 to 1. kbeta line 0, p3, 1 a1 noise kamp, kbeta out a1 endin /* noise.orc */
/* noise.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* noise.sco */
ichn -- MIDI channel number (1-16)
inum -- note number (0-127)
ivel -- velocity (0-127)
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.
ichn -- MIDI channel number (1-16)
inum -- note number (0-127)
ivel -- velocity (0-127)
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.
noteondur — Sends a noteon and a noteoff MIDI message both with the same channel, number and velocity.
Sends a noteon and a noteoff MIDI message both with the same channel, number and velocity.
ichn -- MIDI channel number (1-16)
inum -- note number (0-127)
ivel -- velocity (0-127)
idur -- how long, in seconds, this note should last.
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.
noteondur2 — Sends a noteon and a noteoff MIDI message both with the same channel, number and velocity.
Sends a noteon and a noteoff MIDI message both with the same channel, number and velocity.
ichn -- MIDI channel number (1-16)
inum -- note number (0-127)
ivel -- velocity (0-127)
idur -- how long, in seconds, this note should last.
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.
Here is an example of the notnum opcode. It uses the files notnum.orc and notnum.sco.
Example 237. Example of the notnum opcode.
/* notnum.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 notnum print i1 endin /* notnum.orc */
/* notnum.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* notnum.sco */
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.
ares nreverb asig, ktime, khdif [, iskip] [,inumCombs] [, ifnCombs] [, inumAlpas] [, ifnAlpas]
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.
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.
Here is a simple example of the nreverb opcode. It uses the files nreverb.orc and nreverb.sco.
Example 238. Simple example of the nreverb opcode.
/* nreverb.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 a1 oscil 10000, 440, 1 a2 nreverb a1, 2.5, .3 out a1 + a2 * .2 endin /* nreverb.orc */
/* nreverb.sco */ ; Table 1: an ordinary sine wave. f 1 0 32768 10 1 i 1 0.0 0.5 i 1 1.0 0.5 i 1 2.0 0.5 i 1 3.0 0.5 i 1 4.0 0.5 e /* nreverb.sco */
Here is an example of the nreverb opcode using an ftable for filter constants. It uses the files nreverb_ftable.orc, nreverb_ftable.sco, and beats.wav.
Example 239. An example of the nreverb opcode using an ftable for filter constants.
/* nreverb_ftable.orc */ ; 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 /* nreverb_ftable.orc */
/* nreverb_ftable.sco */ ; 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 /* nreverb_ftable.sco */
Sends a NPRN (Non-Registered Parameter Number) message to the MIDI OUT port each time one of the input arguments changes.
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)
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.
Here is an example of the nsamp opcode. It uses the files nsamp.orc, nsamp.sco, and mary.wav.
Example 240. Example of the nsamp opcode.
/* nsamp.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the size (in samples) of Table #1. isz = nsamp(1) print isz endin /* nsamp.orc */
/* nsamp.sco */ ; Table #1: Use an audio file. f 1 0 262144 1 "mary.wav" 0 0 0 ; Play Instrument #1 for 1 second. i 1 0 1 e /* nsamp.sco */
Since the audio file “mary.wav” has 154390 samples, its output should include a line like this:
instr 1: isz = 154390.000
ares ntrpol asig1, asig2, kpoint [, imin] [, imax]
ires ntrpol isig1, isig2, ipoint [, imin] [, imax]
kres ntrpol ksig1, ksig2, kpoint [, imin] [, imax]
imin -- minimum xpoint value (optional, default 0)
imax -- maximum xpoint value (optional, default 1)
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.
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.
Here is an example of the octave opcode. It uses the files octave.orc and octave.sco.
Example 241. Example of the octave opcode.
/* octave.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; The root note is A above middle-C (440 Hz) iroot = 440 ; Raise the root note by two octaves. ioctaves = 2 ; Calculate the new note. ifactor = octave(ioctaves) inew = iroot * ifactor ; Print out of all of the values. print iroot print ifactor print inew endin /* octave.orc */
/* octave.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* octave.sco */
Its output should include lines like:
instr 1: iroot = 440.000 instr 1: ifactor = 4.000 instr 1: inew = 1760.149
octcps (cps) (init- or control-rate args only)
where the argument within the parentheses may be a further expression.
These 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:
Table 3. Pitch and Frequency Values
Name | Abbreviation |
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octave point pitch-class (8ve.pc) | pch |
octave point decimal | oct |
cycles per second | cps |
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. Thus A440 can be represented alternatively by 440 (cps),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 |
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The conversion from pch or oct 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. |
Here is an example of the octcps opcode. It uses the files octcps.orc and octcps.sco.
Example 242. Example of the octcps opcode.
/* octcps.orc */ ; 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 /* octcps.orc */
/* octcps.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* octcps.sco */
Its output should include a line like this:
instr 1: ioct = 8.750
Get the note number of the current MIDI event, expressed in octave-point-decimal units, for local processing.
Here is an example of the octmidi opcode. It uses the files octmidi.orc and octmidi.sco.
Example 243. Example of the octmidi opcode.
/* octmidi.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 octmidi print i1 endin /* octmidi.orc */
/* octmidi.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* octmidi.sco */
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.
Get the note number of the current MIDI event and modify it by the current pitch-bend value, express it in octave-point-decimal.
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.
Here is an example of the octmidib opcode. It uses the files octmidib.orc and octmidib.sco.
Example 244. Example of the octmidib opcode.
/* octmidib.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 octmidib print i1 endin /* octmidib.orc */
/* octmidib.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* octmidib.sco */
octpch (pch) (init- or control-rate args only)
where the argument within the parentheses may be a further expression.
These 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:
Table 4. Pitch and Frequency Values
Name | Abbreviation |
---|---|
octave point pitch-class (8ve.pc) | pch |
octave point decimal | oct |
cycles per second | cps |
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. Thus A440 can be represented alternatively by 440 (cps),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 |
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The conversion from pch or oct 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. |
Here is an example of the octpch opcode. It uses the files octpch.orc and octpch.sco.
Example 245. Example of the octpch opcode.
/* octpch.orc */ ; 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 /* octpch.orc */
/* octpch.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* octpch.sco */
Its output should include a line like this:
instr 1: ioct = 8.750
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:
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.
![]() | Notes |
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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, 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 |
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 |
The maximum allowed number of input arguments is 24.
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 24.
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 |
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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 ksmps |
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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:
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.
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]
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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. |
Here is an example of a user-defined opcode. It uses the files opcode_example.orc and opcode_example.sco.
Example 246. Example of a user-defined opcode.
/* ---- opcode_example.orc ---- */ sr = 44100 ksmps = 50 nchnls = 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 = 20000 ; 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 out a1 endin /* ---- opcode_example.orc ---- */
/* ---- opcode_example.sco ---- */ i 1 0 4 e /* ---- opcode_example.sco ---- */
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.
kwhen -- a message is sent whenebver 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.
The example shows a simple string of messages being sent just once to a computer called xenakis, on port 7770 to be read by a process that recognises /foo/bar as its address.
sr = 44100 ksmps = 100 nchnls = 2 instr 1 OSCsend 1, "xenakis.cs.bath.ac.uk",7770, "/foo/bar", "sis", "FOO", 42, "bar" endin
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.
The example shows a pair of floating point numbers being received on port 7770.
sr = 44100 ksmps = 100 nchnls = 2 gihandle OSClisten 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
On each k-cycle looks to see if an OSC message has been send to a given path of a given type.
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.
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.
The example shows a pair of floating point numbers being received on port 7770.
sr = 44100 ksmps = 100 nchnls = 2 gihandle OSClisten 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
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).
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]
iovrlap -- Number of oscillator units.
iseed -- Seed value for random number generator (positive integer in the range 1 to 2147483646 (2 ^ 31 - 2)). iseed <= seeds 0 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.
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.
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oscbnk uses the same random number generator as rnd31. So reading its documentation is also recommended. |
Here is an example of oscbnk opcode. It uses the files oscbnk.orc and oscbnk.sco.
Example 247. Example of the oscbnk opcode.
/* oscbnk.orc */ /* 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 /* oscbnk.orc */
/* oscbnk.sco */ /* 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 /* oscbnk.sco */
Table ifn is incrementally sampled modulo the table length and the value obtained is multiplied by amp.
ifn -- function table number. Requires a wrap-around guard point.
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.
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.
Here is an example of the oscil opcode. It uses the files oscil.orc and oscil.sco.
Example 248. Example of the oscil opcode.
/* oscil.orc */ ; Initialize the global variables. 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 /* oscil.orc */
/* oscil.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* oscil.sco */
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 -- function table number. tablei, oscil1i require the extended guard point.
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.
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 -- function table number. oscil1i requires the extended guard point.
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.
Table ifn is incrementally sampled modulo the table length and the value obtained is multiplied by amp.
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.
kamp, xamp -- amplitude
kcps, xcps -- frequency in cycles per second.
oscil3 is experimental, and is identical to oscili, except that it uses cubic interpolation. (New in Csound version 3.50.)
Here is an example of the oscil3 opcode. It uses the files oscil3.orc and oscil3.sco.
Example 249. Example of the oscil3 opcode.
/* oscil3.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a basic oscillator. instr 1 kamp = 10000 kcps = 220 ifn = 1 a1 oscil kamp, kcps, ifn out a1 endin ; Instrument #2 - the basic oscillator with cubic interpolation. instr 2 kamp = 10000 kcps = 220 ifn = 1 a1 oscil3 kamp, kcps, ifn out a1 endin /* oscil3.orc */
/* oscil3.sco */ ; Table #1, a sine wave table with a small amount of data. f 1 0 32 10 0 1 ; Play Instrument #1, the basic oscillator, for ; two seconds. This should sound relatively rough. i 1 0 2 ; Play Instrument #2, the cubic interpolated oscillator, for ; two seconds. This should sound relatively smooth. i 2 2 2 e /* oscil3.sco */
Table ifn is incrementally sampled modulo the table length and the value obtained is multiplied by amp.
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.
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.
Here is an example of the oscili opcode. It uses the files oscili.orc and oscili.sco.
Example 250. Example of the oscili opcode.
/* oscili.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a basic oscillator. instr 1 kamp = 10000 kcps = 220 ifn = 1 a1 oscil kamp, kcps, ifn out a1 endin ; Instrument #2 - the basic oscillator with extra interpolation. instr 2 kamp = 10000 kcps = 220 ifn = 1 a1 oscili kamp, kcps, ifn out a1 endin /* oscili.orc */
/* oscili.sco */ ; Table #1, a sine wave table with a small amount of data. f 1 0 32 10 0 1 ; Play Instrument #1, the basic oscillator, for ; two seconds. This should sound relatively rough. i 1 0 2 ; Play Instrument #2, the interpolated oscillator, for ; two seconds. This should sound relatively smooth. i 2 2 2 e /* oscili.sco */
oscilikt — A linearly interpolated oscillator that allows changing the table number at k-rate.
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.
ares oscilikt xamp, xcps, kfn [, iphs] [, istor]
kres oscilikt kamp, kcps, kfn [, iphs] [, istor]
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.
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).
Here is an example of the oscilikt opcode. It uses the files oscilikt.orc and oscilikt.sco.
Example 251. Example of the oscilikt opcode.
/* oscilikt.orc */ ; 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 /* oscilikt.orc */
/* oscilikt.sco */ ; 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 /* oscilikt.sco */
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.
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.
Here is an example of the osciliktp opcode. It uses the files osciliktp.orc and osciliktp.sco.
Example 252. Example of the osciliktp opcode.
/* osciliktp.orc */ ; 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 /* osciliktp.orc */
/* osciliktp.sco */ ; Table #1: Sawtooth wave f 1 0 3 -2 1 0 -1 ; Play Instrument #1 for four seconds. i 1 0 4 e /* osciliktp.sco */
oscilikts — A linearly interpolated oscillator with sync status that allows changing the table number at k-rate.
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.
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.
Here is an example of the oscilikts opcode. It uses the files oscilikts.orc and oscilikts.sco.
Example 253. Example of the oscilikts opcode.
/* oscilikts.orc */ ; 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 /* oscilikts.orc */
/* oscilikts.sco */ ; Table #1: Sawtooth wave f 1 0 3 -2 1 0 -1 ; Play Instrument #1 for four seconds. i 1 0 4 e /* oscilikts.sco */
Accesses table values at a user-defined frequency. This opcode can also be written as oscilx.
ifrq, itimes -- rate and number of times through the stored table.
ifn -- function table number.
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.
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.
Here is an example of the oscils opcode. It uses the files oscils.orc and oscils.sco.
Example 254. Example of the oscils opcode.
/* oscils.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a fast sine oscillator. instr 1 iamp = 10000 icps = 440 iphs = 0 a1 oscils iamp, icps, iphs out a1 endin /* oscils.orc */
/* oscils.sco */ ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* oscils.sco */
Sends mono 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.
outc outputs as many channels as provided. Any channels greater than nchnls are ignored. Zeros are added as necessary
outch — Writes multi-channel audio data, with user-controllable channels, to an external device or stream.
Writes multi-channel audio data, with user-controllable channels, to an external device or stream.
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.
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))
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.
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))
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.
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))
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.
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))
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.
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))
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.
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))
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
Here is an example of the p opcode. It uses the files p.orc and p.sco.
Example 255. Example of the p opcode.
/* p.orc */ ; 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 /* p.orc */
/* p.sco */ ; p4 = value to be printed. ; Play Instrument #1 for one second, p4 = 50.375. i 1 0 1 50.375 e /* p.sco */
Its output should include lines like:
instr 1: i1 = 50.375
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.
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.
instr 1 k1 phasor 1/p3 ; fraction of circle 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) a1 oscili 10000,440, 1 ; audio signal.. a1,a2,a3,a4 pan a1, k2/2, k3/2, 2, 1, 1 ; sent in a circle (f2=1st quad sin) outq a1, a2, a3, a4 endin
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
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)
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
Here is an example of the pareq opcode. It uses the files pareq.orc and pareq.sco.
Example 256. Example of the pareq opcode.
/* pareq.orc */ 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 /* pareq.orc */
/* pareq.sco */ ; 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 /* pareq.sco */
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.
ffr -- 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 257. Example
ain inch 1 ; input signal 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) out aout
The example above shows partial tracking of an ifd-analysis signal and cubic-phase additive resynthesis with pitch shifting.
Cauchy distribution random number generator (positive values only). This is an x-class noise generator.
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.
Here is an example of the pcauchy opcode. It uses the files pcauchy.orc and pcauchy.sco.
Example 258. Example of the pcauchy opcode.
/* pcauchy.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a random number between 0 and 1. ; kalpha = 1 i1 pcauchy 1 print i1 endin /* pcauchy.orc */
/* pcauchy.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* pcauchy.sco */
Its output should include a line like this:
instr 1: i1 = 0.012
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.
Here is an example of the pchbend opcode. It uses the files pchbend.orc and pchbend.sco.
Example 259. Example of the pchbend opcode.
/* pchbend.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 pchbend print i1 endin /* pchbend.orc */
/* pchbend.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* pchbend.sco */
Get the note number of the current MIDI event, expressed in pitch-class units for local processing.
Here is an example of the pchmidi opcode. It uses the files pchmidi.orc and pchmidi.sco.
Example 260. Example of the pchmidi opcode.
/* pchmidi.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 pchmidi print i1 endin /* pchmidi.orc */
/* pchmidi.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* pchmidi.sco */
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.
Get the note number of the current MIDI event and modify it by the current pitch-bend value, express it in pitch-class units.
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.
Here is an example of the pchmidib pchmidib. It uses the files pchmidib.orc and pchmidib.sco.
Example 261. Example of the pchmidib pchmidib.
/* pchmidib.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 pchmidib print i1 endin /* pchmidib.orc */
/* pchmidib.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* pchmidib.sco */
pchoct (oct) (init- or control-rate args only)
where the argument within the parentheses may be a further expression.
These 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:
Table 5. Pitch and Frequency Values
Name | Abbreviation |
---|---|
octave point pitch-class (8ve.pc) | pch |
octave point decimal | oct |
cycles per second | cps |
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. Thus A440 can be represented alternatively by 440 (cps),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.
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The conversion from pch or oct 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. |
Here is an example of the pchoct opcode. It uses the files pchoct.orc and pchoct.sco.
Example 262. Example of the pchoct opcode.
/* pchoct.orc */ ; 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 /* pchoct.orc */
/* pchoct.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* pchoct.sco */
Its output should include a line like this:
instr 1: ipch = 8.090
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
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.
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
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.
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You will need to download the impulse response files from noisevault.com or replace the filenames with your own impulse files |
sr = 44100 ksmps = 100 nchnls = 2 instr 1 kmix = .5 ; Wet/dry mix. Vary as desired. kvol = .5*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 awetl, awetr pconvolve kvol*(al+ar), "Mercedes-van.wav", ipartitionsize ; Delay dry signal, to align it with the convoled sig adryl delay (1-kmix)*al, idel adryr delay (1-kmix)*ar, idel outs adryl+awetl, adryr+awetr endin instr 2 imix = 0.5 ; Wet/dry mix. Vary as desired. ivol = .5*imix ; Overall volume level of reverb. May need to adjust ; when wet/dry mix is changed, to avoid clipping. ipartitionsize = 32768 ; 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: al, ar soundin "John_Cage_1.aif", 0 awetl, awetr pconvolve ivol*(al+ar),"FactoryHall.aif", ipartitionsize adryl delay (1-imix)*al,idel ; Delay dry signal, to align it with adryr delay (1-imix)*ar,idel ; kcount = kcount - 1 if kcount > 0 kgoto loop outs awetl+adryl, awetr+adryr endin
These opcodes maintain the output k-rate variable as the peak absolute level so far received.
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.
Here is an example of the peak opcode. It uses the files peak.orc, peak.sco, and beats.wav.
Example 263. Example of the peak opcode.
/* peak.orc */ ; 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 /* peak.orc */
/* peak.sco */ ; Play Instrument #1, the audio file, for three seconds. i 1 0 3 e /* peak.sco */
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
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.
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.
Here is an example of the pgmassign opcode. It uses the files pgmassign.orc and pgmassign.sco.
Example 264. Example of the pgmassign opcode.
/* pgmassign.orc */ ; 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 /* pgmassign.orc */
/* pgmassign.sco */ ; Play Instrument #10 for one second. i 10 0 1 e /* pgmassign.sco */
Here is an example of the pgmassign opcode that will ignore program change events. It uses the files pgmassign_ignore.orc and pgmassign_ignore.sco.
Example 265. Example of the pgmassign opcode that will ignore program change events.
/* pgmassign_ignore.orc */ ; 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 /* pgmassign_ignore.orc */
/* pgmassign_ignore.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* pgmassign_ignore.sco */
Here is an advanced example of the pgmassign opcode. It uses the files pgmassign_advanced.mid, pgmassign_advanced.orc, and pgmassign_advanced.sco.
Don't forget that you must include the -F flag when using an external MIDI file like “pgmassign_advanced.mid”.
Example 266. An advanced example of the pgmassign opcode.
/* pgmassign_advanced.orc - written by Istvan Varga */ sr = 44100 ksmps = 10 nchnls = 1 massign 1, 1 ; channels 1 to 4 use instr 1 by default massign 2, 1 massign 3, 1 massign 4, 1 ; 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 30000, kcps, 1 out asnd endin instr 2 /* band-limited sawtooth */ kcps cpsmidib 2 ; note frequency asnd oscili 30000, kcps, 2 out asnd endin instr 3 /* band-limited square */ kcps cpsmidib 2 ; note frequency asnd oscili 30000, kcps, 3 out asnd endin /* pgmassign_advanced.orc - written by Istvan Varga */
/* pgmassign_advanced.sco - written by Istvan Varga */ t 0 120 f 0 8.5 2 -2 0 e /* pgmassign_advanced.sco - written by Istvan Varga */
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.
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.
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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.
Here is an example of the phaser1 opcode. It uses the files phaser1.orc and phaser1.sco.
Example 267. Example of the phaser1 opcode.
/* phaser1.orc */ 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 /* phaser1.orc */
/* phaser1.sco */ ; 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 /* phaser1.sco */
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.
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.
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.
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.
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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 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).
Here is an example of the phaser2 opcode. It uses the files phaser2.orc and phaser2.sco.
Example 268. Example of the phaser2 opcode.
/* phaser2.orc */ 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 /* phaser2.orc */
/* phaser2.sco */ ; 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 /* phaser2.sco */
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.
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.
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.
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.
Here is an example of the phasor opcode. It uses the files phasor.orc and phasor.sco.
Example 269. Example of the phasor opcode.
/* phasor.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create an index that repeats once per second. kcps init 1 kndx phasor kcps ; 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 /* phasor.orc */
/* phasor.sco */ ; 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 /* phasor.sco */
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.
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.
Here is an example of the phasorbnk opcode. It uses the files phasorbnk.orc and phasorbnk.sco.
Example 270. Example of the phasorbnk opcode.
/* phasorbnk.orc */ ; 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 /* phasorbnk.orc */
/* phasorbnk.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 e /* phasorbnk.sco */
Generate multiple voices with independent partials. This example is better with adsynt. See also the example under adsynt, for k-rate use of phasorbnk.
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
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.
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.
Here is an example of the pinkish opcode. It uses the files pinkish.orc and pinkish.sco.
Example 271. Example of the pinkish opcode.
/* pinkish.orc */ ; 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 /* pinkish.orc */
/* pinkish.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* pinkish.sco */
Kellet-filtered noise for a tied note (iskip is non-zero).
Authors: Phil Burk and John ffitch |
University of Bath/Codemist Ltd. |
Bath, UK |
May 2000 |
New in Csound Version 4.07
Adapted for Csound by Rasmus Ekman
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 Rasmus Ekman on September 2002.
Using the same techniques as spectrum and specptrk, pitch tracks the pitch of the signal in octave point decimal form, and amplitude in dB.
koct, kamp pitch asig, iupdte, ilo, ihi, idbthresh [, ifrqs] [, iconf] [, istrt] [, iocts] [, iq] [, inptls] [, irolloff] [, iskip]
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.
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.
Here is an example of the pitch opcode. It uses the files pitch.orc, pitch.sco and mary.wav.
Example 272. Example of the pitch opcode.
/* pitch.orc */ ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1 - play an audio file without effects. instr 1 asig soundin "mary.wav" out asig endin ; Instrument #2 - track the pitch of an audio file. instr 2 iupdte = 0.01 ilo = 7 ihi = 9 idbthresh = 10 ifrqs = 12 iconf = 10 istrt = 8 asig soundin "mary.wav" ; Follow the audio file, get its pitch and amplitude. koct, kamp pitch asig, iupdte, ilo, ihi, idbthresh, ifrqs, iconf, istrt ; Re-synthesize the audio file with a different sounding waveform. kamp2 = kamp * 10 kcps = cpsoct(koct) a1 oscil kamp2, kcps, 1 out a1 endin /* pitch.orc */
/* pitch.sco */ ; Table #1: A different sounding waveform. f 1 0 32768 11 7 3 .7 ; Play Instrument #1, the audio file, for three seconds. i 1 0 3 ; Play Instrument #2, the "re-synthesized" waveform, for three seconds. i 2 3 3 e /* pitch.sco */
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.
kcps, krms pitchamdf asig, imincps, imaxcps [, icps] [, imedi] [, idowns] [, iexcps] [, irmsmedi]
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.
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.
Here is an example of the pitchamdf opcode. It uses the files pitchamdf.orc, pitchamdf.sco and mary.wav.
Example 273. Example of the pitchamdf opcode.
/* pitchamdf.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; synth waveform giwave ftgen 2, 0, 1024, 10, 1, 1, 1, 1 ; Instrument #1 - play an audio file with no effects. instr 1 ; get input signal with original freq. asig soundin "mary.wav" out asig endin ; Instrument #2 - play the synth waveform using the ; same pitch and amplitude as the audio file. instr 2 ; get input signal with original freq. asig soundin "mary.wav" ; lowpass-filter asig tone asig, 1000 ; extract pitch and envelope kcps, krms pitchamdf asig, 150, 500, 200 ; "re-synthesize" with the synth waveform, giwave. asig1 oscil krms, kcps, giwave out asig1 endin /* pitchamdf.orc */
/* pitchamdf.sco */ ; Play Instrument #1, the audio file, for three seconds. i 1 0 3 ; Play Instrument #2, the "re-synthesized" waveform, for three seconds. i 2 3 3 e /* pitchamdf.sco */
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.
ax, ay, az planet kmass1, kmass2,
ksep, ix, iy, iz, ivx, ivy, ivz, idelta [, ifriction] [, iskip]
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 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)
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
Here is an example of the planet opcode. It uses the files planet.orc and planet.sco.
Example 274. Example of the planet opcode.
/* planet.orc */ ; 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 /* planet.orc */
/* planet.sco */ ; Table #1 a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 10 seconds. i 1 0 10 e /* planet.sco */
Audio output is a naturally decaying plucked string or drum sound based on the Karplus-Strong algorithms.
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 + iparm2must 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.
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.
Here is an example of the pluck opcode. It uses the files pluck.orc and pluck.sco.
Example 275. Example of the pluck opcode.
/* pluck.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 20000 kcps = 440 icps = 440 ifn = 0 imeth = 1 a1 pluck kamp, kcps, icps, ifn, imeth out a1 endin /* pluck.orc */
/* pluck.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 e /* pluck.sco */
Poisson distribution random number generator (positive values only). This is an x-class noise generator.
klambda -- the mean of the 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.
Here is an example of the poisson opcode. It uses the files poisson.orc and poisson.sco.
Example 276. Example of the poisson opcode.
/* poisson.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generates a random number in a poisson distribution. ; klambda = 1 i1 poisson 1 print i1 endin /* poisson.orc */
/* poisson.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* poisson.sco */
Its output should include a line like this:
instr 1: i1 = 1.000
polyaft returns the polyphonic pressure of the selected note number, optionally mapped to an user-specified range.
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
Here is an example of the polyaft opcode. It uses the files polyaft.mid, polyaft.orc and polyaft.sco.
Don't forget that you must include the -F flag when using an external MIDI file like “polyaft.mid”.
Example 277. Example of the polyaft opcode.
/* polyaft.orc - written by Istvan Varga */ 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 /* polyaft.orc - written by Istvan Varga */
/* polyaft.sco - written by Istvan Varga */ t 0 120 f 0 9 2 -2 0 e /* polyaft.sco - written by Istvan Varga */
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.
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.
isig (optional, default=0) -- initial (i.e. previous) value for internal feedback. The default value is 0.
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.
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]
ifn -- function table number
iphs (optional, default=0) -- initial phase (in samples)
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.
Here is an example of the poscil opcode. It uses the files poscil.orc and poscil.sco.
Example 278. Example of the poscil opcode.
/* poscil.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a basic oscillator. instr 1 kamp = 10000 kcps = 440 ifn = 1 a1 poscil kamp, kcps, ifn out a1 endin /* poscil.orc */
/* poscil.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* poscil.sco */
ifn -- function table number
iphs (optional, default=0) -- initial phase (in samples)
ares -- output signal
kamp -- the amplitude of the output signal.
kcps -- the frequency of the output signal in cycles per second.
poscil3 uses cubic interpolation.
Here is an example of the poscil3 opcode. It uses the files poscil3.orc and poscil3.sco.
Example 279. Example of the poscil3 opcode.
/* poscil3.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - a basic oscillator. instr 1 kamp = 10000 kcps = 440 ifn = 1 a1 poscil3 kamp, kcps, ifn out a1 endin /* poscil3.orc */
/* poscil3.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* poscil3.sco */
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!
aarg, iarg, karg -- the base.
ipow, kpow -- the exponent.
![]() | Note |
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Use ^ with caution in arithmetical statements, as the precedence may not be correct. New in Csound version 3.493. |
Here is an example of the pow opcode. It uses the files pow.orc and pow.sco.
Example 280. Example of the pow opcode.
/* pow.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; This could also be expressed as: i1 = 2 ^ 12 i1 pow 2, 12 print i1 endin /* pow.orc */
/* pow.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* pow.sco */
Its output should include a line like this:
instr 1: i1 = 4096.000
powoftwo() function returns 2 ^ x 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.
Here is an example of the powoftwo opcode. It uses the files powoftwo.orc and powoftwo.sco.
Example 281. Example of the powoftwo opcode.
/* powoftwo.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = powoftwo(12) print i1 endin /* powoftwo.orc */
/* powoftwo.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* powoftwo.sco */
Its output should include a line like this:
instr 1: i1 = 4096.000
insnum -- instrument number
icount -- number of instrument allocations
“insname” -- A string (in double-quotes) representing a named instrument.
All instances of prealloc must be defined in the header section, not in the instrument body.
Here is an example of the prealloc opcode. It uses the files prealloc.orc and prealloc.sco.
Example 282. Example of the prealloc opcode.
/* prealloc.orc */ ; 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 /* prealloc.orc */
/* prealloc.sco */ ; 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 /* prealloc.sco */
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.
print -- print the current value of the i-time arguments (or expressions) iarg at every i-pass through the instrument.
Here is an example of the print opcode. It uses the files print.orc and print.sco.
Example 283. Example of the print opcode.
/* print.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print the fourth p-field. print p4 endin /* print.orc */
/* print.sco */ ; p4 = value to be printed. ; Play Instrument #1 for one second, p4 = 50.375. i 1 0 1 50.375 ; Play Instrument #1 for one second, p4 = 300. i 1 1 1 300 ; Play Instrument #1 for one second, p4 = -999. i 1 2 1 -999 e /* print.sco */
Its output should include lines like this:
instr 1: p4 = 50.375 instr 1: p4 = 300.000 instr 1: p4 = -999.000
printf and printf_i write formatted output, similarly to the C function printf(). printf_i runs at i-time only, while printfk runs both at initialization and performance time.
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
itrig -- if greater than zero the opcode performs the printing; otherwise it is an null operation.
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.
itime -- time in seconds between printings.
ispace (optional, default=0) -- number of spaces to insert before printing. (default: 0, max: 130)
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.
Here is an example of the printk opcode. It uses the files printk.orc and printk.sco.
Example 284. Example of the printk opcode.
/* printk.orc */ ; 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 /* printk.orc */
/* printk.sco */ ; Play Instrument #1 for 5 seconds. i 1 0 5 e /* printk.sco */
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
inumspaces (optional, default=0) -- number of space characters printed before the value of kvar
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 |
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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. |
Here is an example of the printk2 opcode. It uses the files printk2.orc and printk2.sco.
Example 285. Example of the printk2 opcode.
/* printk2.orc */ ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1. instr 1 ; Change a value linearly from 0 to 10, ; over the period defined by p3. kval1 line 0, p3, 10 ; If kval1 is greater than or equal to 5, ; then kval=2, else kval=1. kval2 = (kval1 >= 5 ? 2 : 1) ; Print the value of kval2 when it changes. printk2 kval2 endin /* printk2.orc */
/* printk2.sco */ ; Play Instrument #1 for 5 seconds. i 1 0 5 e /* printk2.sco */
Its output should include a line like this:
i1 1.00000 i1 2.00000
"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.
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.
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 |
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\\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 |
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Prior to version 4.23, only the %f format code was supported. |
Here is an example of the printks opcode. It uses the files printks.orc and printks.sco.
Example 286. Example of the printks opcode.
/* printks.orc */ ; 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 /* printks.orc */
/* printks.sco */ ; Play Instrument #1 for 5 seconds. i 1 0 5 e /* printks.sco */
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
"string" -- the text string to be printed. Can be up to 8192 characters and must be in double quotes.
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.
Here is an example of the prints opcode. It uses the files prints.orc and prints.sco.
Example 287. Example of the prints opcode.
/* prints.orc */ /* 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 /* prints.orc */
/* prints.sco */ /* Written by Matt Ingalls, edited by Kevin Conder. */ ; Play instrument #1. i 1 0 0.004 /* prints.sco */
Its output should include a line like this:
1234.568 ;;;;;;semicolons!
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.
puts prints a string with an optional newline at the end whenever the trigger signal is positive and changes.
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
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.
ares pvadd ktimpnt, kfmod, ifilcod, ifn, ibins [, ibinoffset] [, ibinincr] [, iextractmode] [, ifreqlim] [, igatefn]
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.
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 one 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 |
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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. |
pvbufread — Reads from a phase vocoder analysis file and makes the retrieved data available.
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.
ifile -- the pvoc number (n in pvoc.n) or the name in quotes of the analysis file made using pvanal. (See pvoc.)
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.
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 .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*.15, 1, p3*.35, 0, p3*.25, 0, p3*.15, 1, p3*.1, 1 pvbufread ktime1, "oboe.pvoc" apv pvinterp ktime2,1,"clar.pvoc",1,1.065,1,.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 .001, p3, 1 pvbufread ktime1, "oboe.pvoc" apv pvcross ktime2, 1, "clar.pvoc", 1-kcross, kcross
pvcross — Applies the amplitudes from one phase vocoder analysis file to the data from a second file.
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.
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.
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.
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.
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 .001, p3, 1 pvbufread ktime1, "oboe.pvoc" apv pvcross ktime2, 1, "clar.pvoc", 1-kcross, kcross
pvinterp — Interpolates between the amplitudes and frequencies of two phase vocoder analysis files.
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.
ares pvinterp ktimpnt, kfmod, ifile, kfreqscale1, kfreqscale2, kampscale1, kampscale2, kfreqinterp, kampinterp
ifile -- the pvoc number (n in pvoc.n) or the name in quotes of the analysis file made using pvanal. (See pvoc.)
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=.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.
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 .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*.15, 1, p3*.35, 0, p3*.25, 0, p3*.15, 1, p3*.1, 1 pvbufread ktime1, "oboe.pvoc" apv pvinterp ktime2,1,"clar.pvoc",1,1.065,1,.75,1-kinterp,1-kinterp
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 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 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.
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.
pvread — Reads from a phase vocoder analysis file and returns the frequency and amplitude from a single analysis channel or bin.
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.
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.
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.
The example below shows the use pvread to synthesize a single component 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.
ktime line 0, p3, 3 kfreq, kamp pvread ktime, "pvoc.file", 7 ; read ;data from 7th analysis bin. asig oscili kamp, kfreq, 1 ; function 1 ;is a stored sine
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.
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.
pvsanal — Generate an fsig from a mono audio source ain, using phase vocoder overlap-add analysis.
Generate an fsig from a mono audio source ain, using phase vocoder overlap-add analysis.
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 eaxmple 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.
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.
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
Example 288. Example
asig in ; get the signal in 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, 0, 2 ; arpeggiate it (range 220.5 - 2425.5) atps pvsynth ftps ; synthesise it out atps
The example above shows a spectral arpeggiator working in the 220.5 - 2425.5 range (sr=44100). The LFO outputs a positive-only signal, so its ftable will be defined in the 0 - 1 range (a hanning window can be used, for instance).
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 are applied to fdest, using scale factors kamp1 and kamp2 respectively. 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.
Example 289. 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 */ a1 soundin "input.wav" fsig pvsanal a1, ifftsize, ifftsize/4, ifftsize, iwtype kcen pvscent fsig adm oscil 32000, kcent, 1 out adm
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.
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.
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 290. 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
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.
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.
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.
idur filelen "test.pvx" ; find dur of (stereo) analysis file kpos line 0,p3,idur ; to ensure we process whole file fsigr pvsfread kpos,"test.pvx",1 ; create fsig from R channel
(NB: as this example shows, the filelen opcode has been extended to accept both old and new analysis file formats).
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
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.
ifn ftgen 0,0,inbins,10,1 ; make ftable kflag pvsftw fsrc,ifn ; export amps to table, kamp init 0 if kflag==0 kgoto contin ; only proc when frame is ready ; kill lowest bins, for obvious effect tablew kamp,1,ifn 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
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
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.
ifn ftgen 0,0,inbins,10,1 ; make ftable kflag pvsftw fsrc,ifn ; export amps to table, kamp init 0 if kflag==0 kgoto contin ; only proc when frame is ready ; kill lowest bins, for obvious effect tablew kamp,1,ifn 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
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.
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).
Example 291. Example
ain inch 1 ; input signal fs1,fsi2 pvsifd ain,2048,512,1 ; pvsifd analysis fst partials fs1,fsi2,.003,1,3,500 ; partial tracking aout resyn fst, 1, 1.5, 500, 1 ; resynthesis (up a 5th) out aout
The example above shows the pvsifd analysis feeding into partial tracking and cubic-phase additive resynthesis with pitch shifting.
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.
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).
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.
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.
Example 293. Example (using score-supplied f-table, assuming fsig fftsize = 1024)
; 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 20000,199,50,3 ; pulsewave source fsig pvsanal asig,1024,256,1024,0 ; create fsig kmod linseg 0,p3/2,1,p3/2,0 ; simple control sig fsig pvsmaska fsig,2,kmod ; apply weird eq to fsig aout pvsynth fsig ; resynthesize, dispfft aout,0.1,1024 ; and view the effect
This also illustrates that the usual Csound behaviour applies to fsigs; the same name can be used for both input and output.
Example 294. Example (using score-supplied f-table, assuming fsig fftsize = 1024)
; 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 20000,199,50,3 ; pulsewave source fsig pvsanal asig,1024,256,1024,0 ; create fsig kmod linseg 0,p3/2,1,p3/2,0 ; simple control sig fsig pvsmaska fsig,2,kmod ; apply weird eq to fsig aout pvsynth fsig ; resynthesize, dispfft aout,0.1,1024 ; and view the effect
This also illustrates that the usual Csound behaviour applies to fsigs; the same name can be used for both input and output.
Scale the frequency components of a pv stream, resulting in pitch shift. Output amplitudes can be optionally modified in order to attempt formant preservation.
fsig -- output pv stream
fsigin -- input pv stream
kscal -- scaling ratio.
ikeepform -- attempt to keep input signal -- -- formants; 0: do not keep formants; 1: keep formants by imposing original amps; 2: keep formants by filtering using the original spec envelope (defaults to 0).
igain -- amplitude scaling (defaults to 1).
The quality of the pitch shift will be improved with the use of a Hanning window in the pvoc analysis. Formant preservation is only successful with strong-formant sounds, such as voices and certain instrumental sounds, but also can be used for intersting transformation effects.
Example 295. Example
asig in ; get the signal in fsig pvsanal asig, 1024, 256, 1024, 1 ; analyse it ftps pvscale fsig, 1.5, 1, 2 ; transpose it keeping formants atps pvsynth ftps ; synthesise it adp delayr .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.
pvshift — Shift the frequency components of a pv stream, stretching/compressing its spectrum.
fsig -- output pv stream
fsigin -- input pv stream
kshift -- shift amount (in Hz, positive or negative).
klowest -- lowest frequency to be shifted.
ikeepform -- attempt to keep input signal formants; 0: do not keep formants; 1: keep formants by imposing original amps; 2: keep formants by filtering using the original spec envelope (defaults to 0).
igain -- amplitude scaling (defaults to 1).
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 ikeepform flag can be used to try and preserve formants for possibly interesting and unusual spectral modifications.
Mix 'seamlessly' two pv signals. This opcode combines the most prominent components of two pvoc streams into a single mixed stream.
fsig -- output pv stream
fsigin1 -- input pv stream.
fsigin2 -- input pv stream, which must have same format as fsigin1.
pvsfilter — Multiply amplitudes of a pvoc stream by those of a second pvoc stream, with dynamic scaling.
Multiply amplitudes of a pvoc stream by those of a second pvoc stream, with dynamic scaling.
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.
Example 298. 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 fin pvsanal asig1,1024,256,1024,0 ; pvoc analysis fil pvsanal asig2,1024,256,1024,0 fout pvsfilter fin, fout, kdepth ; filter signal aout pvsynth fsigout ; 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.
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.
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.
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 PVOCEX 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'.
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.
pvsvoc — Combine the spectral envelope of one fsig with the excitation (frequencies) of another.
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.
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.
Example 301. Example
asig in ; get the signal in asyn oscili 16000, 150, 1 ; excitation signal 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 out atps
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.
pyassign — Assign the value of the given Csound variable to a Python variable possibly destroying its previous content.
pyassign "variable", kvalue
pyassigni "variable", ivalue
pylassign "variable", kvalue
pylassigni "variable", ivalue
pyassignt ktrigger, "variable", kvalue
pylassignt ktrigger, "variable", kvalue
pycall — 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, ...
This family of opcodes call 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.
They pass any number of parameters which are cast to float inside the Python interpreter.
The pycall/pycalli, pycall1/pycall1i ... pycall8/pycall8i opcodes can accomodate for a number of results ranging from 0 to 8 according to their numerical prefix (0 is omitted).
The pycalln/pycallni opcodes can accomodate for any number of results: the callable name is followed by the number of output arguments, then come the list of Csound output variable and the list of parameters to be passed.
The returning value of the callable must be None for pycall or pycalli, a float for pycall1i or pycall1i and a tuple (with proper size) of floats for the pycall2/pycall2i ... pycall8/pycall8i and pycalln/pycallni opcodes.
Example 302. Calling a C or Python function
Supposing we have previously defined or imported a function named get_number_from_pool as:
from random import random, choice # a pool of 100 numbers pool = [i ** 1.3 for i in range(100)] def get_number_from_pool(n, p): # substitute an old number with the new number? if random() < p: i = choice(range(len(pool))) pool[i] = n # return a random number from the pool return choice(pool)
then the following orchestra code
k2 pycall1 "get_number_from_pool", k1, p6
would set k2 randomly from a pool of numbers changing in time. You can pass new pools elements and control the change rate from the orchestra.
Example 303. Calling a Function Object
A more generic implementation of the previous example makes use of a simple function object:
from random import random, choice class GetNumberFromPool: def __init__(self, e, begin=0, end=100, step=1): self.pool = [i ** e for i in range(begin, end, step)] def __call__(self, n, p): # substitute an old number with the new number? if random() < p: i = choice(range(len(pool))) pool[i] = n # return a random number from the pool return choice(pool) get_number_from_pool1 = GetNumberFromPool(1.3) get_number_from_pool2 = GetNumberFromPool(1.5, 50, 250, 2)
Then the following orchestra code:
k2 pycall1 "get_number_from_pool1", k1, p6 k4 pycall1 "get_number_from_pool2", k3, p7
would set k2 and k3 randomly from a pool of numbers changing in time. You can pass new pools elements (here k1 and k3) and control the change rate (here p6 and p7) from the orchestra.
As you can see in the first snippet, you can customize the initialization of the pool as well as create several pools.
pyeval — 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"
These opcodes evaluate a generic Python expression and store the result in a Csound variable at k-time or i-time (i suffix).
The expression must evaluate in a float or an object that can be cast to a float.
They can be used effectively to trasfer data from a Python object into a Csound variable.
pyexec "filename"
pyexeci "filename"
pylexec "filename"
pylexeci "filename"
pyexect ktrigger, "filename"
plyexect ktrigger, "filename"
Execute a script from a file at k-time or i-time (i suffix).
This is not the same as calling the script with the system() call, since the code is executed by the embedded interpreter.
The code contained in the specified file is executed in the global environment for opcodes pyexec and pyexeci and in the private environment for the opcodes pylexec and pylexeci.
These opcodes perform no message passing. However, since the statement has access to the main namespace and the private namespace, it can interact with objects previously created in that environment.
The "local" version of the pyexec opcodes are useful when the code ran by different instances of an instrument should not interact.
Example 304. Orchestra (pyexec.orc)
sr=44100 kr=4410 ksmps=10 nchnls=1 ;If you're not running CsoundVST you need the following line ;to initialize the python interpreter ;pyinit pyruni "import random" pyexeci "pyexec1.py" instr 1 pyexec "pyexec2.py" pylexeci "pyexec3.py" pylexec "pyexec4.py" endin
Example 306. The pyexec1.py Script
import time, os print print "Welcome to Csound!" try: s = ', %s?' % os.getenv('USER') except: s = '?' print 'What sound do you want to hear today%s' % s answer = raw_input()
If I run this example on my machine I get something like:
Using ../../csound.xmg Csound Version 4.19 (Mar 23 2002) Embedded Python interpreter version 2.2 orchname: pyexec.orc scorename: pyexec.sco sorting score ... ... done orch compiler: 11 lines read instr 1 Csound Version 4.19 (Mar 23 2002) displays suppressed Welcome to Csound! What sound do you want to hear today, maurizio?then I answer
damn youthen Csound continues with the normal performance
your answer is "damn you" a private random number: 0.884006 new alloc for instr 1: your answer is "damn you" a private random number: 0.884006 your answer is "damn you" a private random number: 0.889868 your answer is "damn you" a private random number: 0.884006 your answer is "damn you" a private random number: 0.889868 your answer is "damn you" a private random number: 0.884006 your answer is "damn you" ...Embarassing.
In the same instrument a message is created in the private namespace and printed, appearing different for each instance.
In the command-line version of Csound, you must first invoke the pyinit opcode in the orchestra header to initialize the Python interpreter, before using any of the other Python opcodes.
But if you use the Python opcodes in the CsoundVST version of Csound, you need not invoke pyinit, because CsoundVST automatically initializes the Python interpreter for you. In addition, CsoundVST automatically creates a Python interface to the Csound API, in the form a global instance of the CsoundVST.CppSound class named csound. Therefore, Python code written in the Csound orchestra has access to the global csound object.
pyrun "statement"
pyruni "statement"
pylrun "statement"
pylruni "statement"
pyrunt ktrigger, "statement"
pylrunt ktrigger, "statement"
Execute the specified Python statement at k-time (pyrun and pylrun) or i-time (pyruni and pylruni).
The statement is executed in the global environment for pyrun and pyruni or the local environment for pylrun and pylruni.
These opcodes perform no message passing. However, since the statement have access to the main namespace and the private namespace, it can interact with objects previously created in that environment.
The "local" version of the pyrun opcodes are useful when the code ran by different instances of an instrument should not interact.
Example 310. Orchestra
sr=44100 kr=4410 ksmps=10 nchnls=1 ;If you're not running CsoundVST you need the following line ;to initialize the python interpreter ;pyinit pyruni "import random" instr 1 ; This message is stored in the main namespace ; and is the same for every instance pyruni "message = 'a global random number: %f' % random.random()" pyrun "print message" ; This message is stored in the private namespace ; and is different for different instances pylruni "message = 'a private random number: %f' % random.random()" pylrun "print message" endin
Running this score you should get intermixed pairs of messages from the two instances of instrument 1.
The first message of each pair is stored into the main namespace and so the second instance overwrites the message of the first instance. The result is that first message will be the same for both instances.
The second message is different for the two instances, being stored in the private namespace.
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.
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.
Here is an example of the rand opcode. It uses the files rand.orc and rand.sco.
Example 312. Example of the rand opcode.
/* rand.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Choose a random frequency between 4,100 and 44,100. kfreq rand 20000 kcps = kfreq + 24100 a1 oscil 30000, kcps, 1 out a1 endin /* rand.orc */
/* rand.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* rand.sco */
ares randh xamp, xcps [, iseed] [, isize] [, ioffset]
kres randh kamp, kcps [, iseed] [, isize] [, ioffset]
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 value greater than 1 will be used directly, without scaling. A negative value will cause seed re-initialization to be skipped. The default seed value is .5.
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.
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.
Here is an example of the randh opcode. It uses the files randh.orc and randh.sco.
Example 313. Example of the randh opcode.
/* randh.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Choose a random frequency between 4,100 and 44,100. ; Generate new random numbers at 220 Hz. ; kamp = 40000 ; kcps = 220 ; iseed = 0.5 ; isize = 0 ; ioffset = 4100 kcps randh 40000, 220, 0.5, 0, 4100 a1 oscil 30000, kcps, 1 out a1 endin /* randh.orc */
/* randh.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* randh.sco */
ares randi xamp, xcps [, iseed] [, isize] [, ioffset]
kres randi kamp, kcps [, iseed] [, isize] [, ioffset]
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 value greater than 1 will be used directly, without scaling. A negative value will cause seed re-initialization to be skipped. The default seed value is .5.
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.
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.
Here is an example of the randi opcode. It uses the files randi.orc and randi.sco.
Example 314. Example of the randi opcode.
/* randi.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Choose a random frequency between 4,100 and 44,100. ; Generate new random numbers at 10 Hz. ; kamp = 40000 ; kcps = 10 ; iseed = 0.5 ; isize = 0 ; ioffset = 4100 kcps randi 40000, 10, 0.5, 0, 4100 a1 oscil 30000, kcps, 1 out a1 endin /* randi.orc */
/* randi.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* randi.sco */
kmin -- minimum range limit
kmax -- maximum range limit
The random opcode is similar to linrand and trirand but sometimes I [Gabriel Maldonado] find it more convenient because allows the user to set arbitrary minimum and maximum values.
Here is an example of the random opcode. It uses the files random.orc and random.sco.
Example 315. Example of the random opcode.
/* random.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a random number between 220 and 440. kmin init 220 kmax init 440 k1 random kmin, kmax printks "k1 = %f\\n", 0.1, k1 endin /* random.orc */
/* random.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* random.sco */
Its output should include lines like:
k1 = 414.232056 k1 = 419.393402 k1 = 275.376373
randomh — Generates random numbers with a user-defined limit and holds them for a period of time.
kmin -- minimum range limit
kmax -- maximum range limit
kcps, acps -- rate of random break-point generation
The randomh opcode is similar to randh but allows the user to set arbitrary minimum and maximum values.
Here is an example of the randomh opcode. It uses the files randomh.orc and randomh.sco.
Example 316. Example of the randomh opcode.
/* randomh.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Choose a random frequency between 220 and 440 Hz. ; Generate new random numbers at 10 Hz. kmin = 220 kmax = 440 kcps = 10 k1 randomh kmin, kmax, kcps printks "k1 = %f\\n", 0.1, k1 endin /* randomh.orc */
/* randh.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* randh.sco */
Its output should include lines like:
k1 = 220.000000 k1 = 414.232056 k1 = 284.095184
randomi — Generates a user-controlled random number series with interpolation between each new number.
Generates a user-controlled random number series with interpolation between each new number.
kmin -- minimum range limit
kmax -- maximum range limit
kcps, acps -- rate of random break-point generation
The randomi opcode is similar to randi but allows the user to set arbitrary minimum and maximum values.
Here is an example of the randomi opcode. It uses the files randomi.orc and randomi.sco.
Example 317. Example of the randomi opcode.
/* randomi.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; 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 k1 randomi kmin, kmax, kcps printks "k1 = %f\\n", 0.1, k1 endin /* randomi.orc */
/* randomi.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* randomi.sco */
Its output should include lines like:
k1 = 220.000000 k1 = 414.226196 k1 = 284.101074
rbjeq — Parametric equalizer and filter opcode with 7 filter types, based on algorithm by Robert Bristow-Johnson.
Parametric equalizer and filter opcode with 7 filter types, based on algorithm by Robert Bristow-Johnson.
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.
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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).
ar -- the output signal.
asig -- the input signal
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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.
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
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.
Here is an example of the readclock opcode. It uses the files readclock.orc and readclock.sco.
Example 318. Example of the readclock opcode.
/* readclock.orc */ ; 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 /* readclock.orc */
/* readclock.sco */ ; 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 /* readclock.sco */
Its output should include lines like this:
instr 1: i1 = 0.000 instr 1: i1 = 90.000 instr 1: i1 = 180.000
Periodically reads an orchestra control-signal value to a named external file in a specific format.
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 lsat two are binary. The output file contains no header information.
iprd -- the period of ksig output i 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.
ipol -- if non-zero, and iprd implies more than one control period, interpolate the k- signals between the periodic reads from the external file. If the value is 0, repeat each signal between frames. Currently not supported.
kres -- a control-rate signal
This opcode allows a generated control signal value to be read from 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 readk opcodes in an instrument or orchestra and they may read from the same or different files.
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 lsat two are binary. The output file contains no header information.
iprd -- the period of ksig output i 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.
ipol -- if non-zero, and iprd implies more than one control period, interpolate the k- signals between the periodic reads from the external file. If the value is 0, repeat each signal between frames. Currently not supported.
kr1, kr2 -- control-rate signals
This opcode allows two generated control signal values to be read from 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 readk2 opcodes in an instrument or orchestra and they may read from the same or different files.
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 lsat two are binary. The output file contains no header information.
iprd -- the period of ksig output i 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.
ipol -- if non-zero, and iprd implies more than one control period, interpolate the k- signals between the periodic reads from the external file. If the value is 0, repeat each signal between frames. Currently not supported.
kr1, kr2, kr3 -- control-rate signals
This opcode allows three generated control signal values to be read from 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 readk3 opcodes in an instrument or orchestra and they may read from the same or different files.
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 lsat two are binary. The output file contains no header information.
iprd -- the period of ksig output i 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.
ipol -- if non-zero, and iprd implies more than one control period, interpolate the k- signals between the periodic reads from the external file. If the value is 0, repeat each signal between frames. Currently not supported.
kr1, kr2, kr3, kr4 -- control-rate signals.
This opcode allows four generated control signal values to be read from 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 readk4 opcodes in an instrument or orchestra and they may read from the same or different files.
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.
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 files reinit.orc and reinit.sco.
Example 319. Example of the reinit opcode.
/* reinit.orc */ 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 /* reinit.orc */
/* reinit.sco */ f1 0 4096 10 1 i1 0 10 e /* reinit.sco */
kflag -- indicates whether the note is in its “release” stage.
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.
instr 1 ;allows complex ADSR envelope with MIDI events 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 < .5) 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, 1 out a1 endin
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.
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.
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).
Here is an example of the repluck opcode. It uses the files repluck.orc and repluck.sco.
Example 320. Example of the repluck opcode.
/* repluck.orc */ ; 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 axcite oscil 1, 1, 1 apluck repluck iplk, kamp, icps, kpick, krefl, axcite out apluck endin /* repluck.orc */
/* repluck.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* repluck.sco */
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.
ares -- the output signal at audio rate.
asig -- the input signal at audio 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).
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).
Here is an example of the reson opcode. It uses the files reson.orc and reson.sco.
Example 321. Example of the reson opcode.
/* reson.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a sine waveform. asine buzz 15000, 440, 3, 1 ; Vary the cut-off frequency from 220 to 1280. kcf line 220, p3, 1320 kbw init 20 ; Run the sine through a resonant filter. ares reson asine, kcf, kbw ; Give the filtered signal the same amplitude ; as the original signal. a1 balance ares, asine out a1 endin /* reson.orc */
/* reson.sco */ ; Table #1, an ordinary sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 4 seconds. i 1 0 4 e /* reson.sco */
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.
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.
Implementations of a second-order, two-pole two-zero bandpass filter with variable frequency response.
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.
asig -- input signal to be filtered
kcf -- cutoff or resonant frequency of the filter, measured in Hz
kbw -- 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.
Here is an example of the resonr and resonz opcodes. It uses the files resonr.orc and resonr.sco.
Example 322. Example of the resonr and resonz opcodes.
/* resonr.orc */ /* 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 /* resonr.orc */
/* resonr.sco */ /* 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 /* resonr.sco */
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.
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.
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.
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.
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.
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.
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.
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.
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.
Here is an example of the resony opcode. It uses the files resony.orc, resony.sco, and beats.wav.
Example 323. Example of the resony opcode.
/* resony.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a nice sawtooth waveform. asig vco 32000, 220, 1 ; Vary the base frequency from 60 to 600 Hz. kbf line 60, p3, 600 kbw = 50 inum = 2 ksep = 1 isepmode = 0 iscl = 1 a1 resony asig, kbf, kbw, inum, ksep, isepmode, iscl out a1 endin /* resony.orc */
/* resony.sco */ ; Table #1, a sine wave for the vco opcode. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* resony.sco */
Implementations of a second-order, two-pole two-zero bandpass filter with variable frequency response.
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.
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
kcf -- cutoff or resonant frequency of the filter, measured in Hz
kbw -- bandwidth of the filter (the Hz difference between the upper and lower half-power points)
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.
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.
resyn — Streaming partial track additive synthesis with cubic phase interpolation with pitch control and support for timescale-modified input
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.
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)
Example 324. Example
ain inch 1 ; input signal 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) out aout
The example above shows partial tracking of an ifd-analysis signal and cubic-phase additive resynthesis with pitch shifting.
iskip (optional, default=0) -- initial disposition of delay-loop data space (cf. reson). The default value is 0.
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.
Here is an example of the reverb opcode. It uses the files reverb.orc and reverb.sco.
Example 325. Example of the reverb opcode.
/* reverb.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; init an audio receiver/mixer ga1 init 0 ; Instrument #1. (there may be many copies) instr 1 ; generate a source signal a1 oscili 7000, cpspch(p4), 1 ; output the direct sound out a1 ; and add to audio receiver ga1 = ga1 + a1 endin ; (highest instr number executed last) instr 99 ; reverberate whatever is in ga1 a3 reverb ga1, 1.5 ; and output the result out a3 ; empty the receiver for the next pass ga1 = 0 endin /* reverb.orc */
/* reverb.sco */ ; Table #1, a sine wave. f 1 0 128 10 1 ; p4 = frequency (in a pitch-class) ; Play Instrument #1 for a tenth of a second, p4=6.00 i 1 0 0.1 6.00 ; Play Instrument #1 for a tenth of a second, p4=6.02 i 1 1 0.1 6.02 ; Play Instrument #1 for a tenth of a second, p4=6.04 i 1 2 0.1 6.04 ; Play Instrument #1 for a tenth of a second, p4=6.06 i 1 3 0.1 6.06 ; Make sure the reverb remains active. i 99 0 6 e /* reverb.sco */
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.
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.
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.
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
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)
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 Hans Mikelson.
Here is an example of the rezzy opcode. It uses the files rezzy.orc and rezzy.sco.
Example 326. Example of the rezzy opcode.
/* rezzy.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a nice sawtooth waveform. asig vco 32000, 220, 1 ; Vary the filter-cutoff frequency from .2 to 2 KHz. kfco line 200, p3, 2000 ; Set the resonance amount. kres init 25 a1 rezzy asig, kfco, kres out a1 endin /* rezzy.orc */
/* rezzy.sco */ ; Table #1, a sine wave for the vco opcode. f 1 0 16384 10 1 ; Play Instrument #1 for three seconds. i 1 0 3 e /* rezzy.sco */
Terminates a reinit pass (i.e., no-op at standard i-time). This statement, or an endin, will cause normal performance to be resumed.
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 files reinit.orc and reinit.sco.
Example 327. Example of the rireturn opcode.
/* reinit.orc */ 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 /* reinit.orc */
/* reinit.sco */ f1 0 4096 10 1 i1 0 10 e /* reinit.sco */
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.
asig -- input audio 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.
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.
Here is an example of the rnd opcode. It uses the files rnd.orc and rnd.sco.
Example 328. Example of the rnd opcode.
/* rnd.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a random number from 0 to 1. i1 = rnd(1) print i1 endin /* rnd.orc */
/* rnd.sco */ ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #1 for one second. i 1 1 1 e /* rnd.sco */
Its output should be:
rnd at i-rate: 0.973500 rnd at k-rate: 0.139405 rnd at i-rate: 0.973500 rnd at k-rate: 0.040065 rnd at i-rate: 0.973500 rnd at k-rate: 0.412845 rnd at i-rate: 0.973500 rnd at k-rate: 0.440650 rnd at i-rate: 0.973500 rnd at k-rate: 0.663581 rnd at i-rate: 0.973500 rnd at k-rate: 0.876723 rnd at i-rate: 0.973500 rnd at k-rate: 0.302459 rnd at i-rate: 0.973500 rnd at k-rate: 0.398580 rnd at i-rate: 0.973500 rnd at k-rate: 0.448875 rnd at i-rate: 0.973500 rnd at k-rate: 0.907728
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.
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) ^ ((1 / irpow) - 1); for negative irpow values, it is (1 - abs(x)) ^ ((-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.
iseed (optional, default=0) -- seed value for random number generator (positive integer in the range 1 to 2147483646 (2 ^ 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.
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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.
Here is an example of the rnd31 opcode at a-rate. It uses the files rnd31.orc and rnd31.sco.
Example 329. An example of the rnd31 opcode at a-rate.
/* rnd31.orc */ ; 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 /* rnd31.orc */
/* rnd31.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* rnd31.sco */
Here is an example of the rnd31 opcode at k-rate. It uses the files rnd31_krate.orc and rnd31_krate.sco.
Example 330. An example of the rnd31 opcode at k-rate.
/* rnd31_krate.orc */ ; 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 /* rnd31_krate.orc */
/* rnd31_krate.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* rnd31_krate.sco */
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 files rnd31_seed7.orc and rnd31_seed7.sco.
Example 331. An example of the rnd31 opcode that uses the number 7 as a seed value.
/* rnd31_seed7.orc */ ; 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 /* rnd31_seed7.orc */
/* rnd31_seed7.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* rnd31_seed7.sco */
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 files rnd31_time.orc and rnd31_time.sco.
Example 332. An example of the rnd31 opcode that uses the current time as a seed value.
/* rnd31_time.orc */ ; 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 /* rnd31_time.orc */
/* rnd31_time.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* rnd31_time.sco */
Its output should include lines like this:
instr 1: i1 = -0.691 instr 1: i2 = -0.686 instr 1: i3 = -0.358
ares rspline xrangeMin, xrangeMax, kcpsMin, kcpsMax
kres rspline krangeMin, krangeMax, kcpsMin, kcpsMax
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.
xamp -- Amplitude factor
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.
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.
Here is an example of the rtclock opcode. It uses the files rtclock.orc and rtclock.sco.
Example 333. Example of the rtclock opcode.
/* rtclock.orc */ ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1 instr 1 ; Get the system time. k1 rtclock ; Print it once per second. printk 1, k1 endin /* rtclock.orc */
/* rtclock.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 e /* rtclock.sco */
Its output should include lines like this:
i 1 time 0.00002: 1018236096.00000 i 1 time 1.00002: 1018236224.00000
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
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
icutoff1 ... icutoff64 -- low-pass filter cutoff frequency for each controller
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.
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
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
icutoff1 ... icutoff64 -- low-pass filter cutoff frequency for each controller
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.
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)
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.
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
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.
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.
Here is an example of the sandpaper opcode. It uses the files sandpaper.orc and sandpaper.sco.
Example 334. Example of the sandpaper opcode.
/* sandpaper.orc */ ;orchestra --------------- sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 01 ;an example of sandpaper blocks a1 line 2, p3, 2 ;preset amplitude increase a2 sandpaper p4, 0.01 ;sandpaper needs a little amp help at these settings a3 product a1, a2 ;increase amplitude out a3 endin /* sandpaper.orc */
/* sandpaper.sco */ ;score ------------------- i1 0 1 26000 e /* sandpaper.sco */
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.
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.
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
Here is an example of the scanned synthesis. It uses the files scans.orc, scans.sco, and string-128.matrix.
Example 335. Example of the scans opcode.
/* scans.orc */ sr = 44100 kr = 4410 ksmps = 10 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 /* scans.orc */
/* scans.sco */ ; 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 /* scans.sco */
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.
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.
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.
Here is an example of the scantable opcode. It uses the files scantable.orc and scantable.sco.
Example 336. Example of the scantable opcode.
/* scantable.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Table #1 - initial position git1 ftgen 1, 0, 128, 7, 0, 64, 1, 64, 0 ; Table #2 - masses git2 ftgen 2, 0, 128, -7, 1, 128, 1 ; Table #3 - stiffness git3 ftgen 3, 0, 128, -7, 0, 64, 100, 64, 0 ; Table #4 - damping git4 ftgen 4, 0, 128, -7, 1, 128, 1 ; Table #5 - initial velocity git5 ftgen 5, 0, 128, -7, 0, 128, 0 ; Instrument #1. instr 1 kamp init 20000 kpch init 220 ipos = 1 imass = 2 istiff = 3 idamp = 4 ivel = 5 a1 scantable kamp, kpch, ipos, imass, istiff, idamp, ivel a2 dcblock a1 out a2 endin /* scantable.orc */
/* scantable.sco */ ; Play Instrument #1 for ten seconds. i 1 0 10 e /* scantable.sco */
scanu init, irate, ifnvel, ifnmass, ifnstif, ifncentr, ifndamp, kmass, kstif, kcentr, kdamp, ileft, iright, kpos, kstrngth, ain, idisp, id
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.
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.
schedkwhen ktrigger, kmintim, kmaxnum, kinsnum, kwhen, kdur [, ip4] [, ip5] [...]
schedkwhen ktrigger, kmintim, kmaxnum, "insname", kwhen, kdur [, ip4] [, ip5] [...]
“insname” -- A string (in double-quotes) representing a named instrument.
ip4, ip5, ... -- Equivalent to p4, p5, etc., in a score i statement
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 extant 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.)
Note: 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.
Here is an example of the schedkwhen opcode. It uses the files schedkwhen.orc and schedkwhen.sco.
Example 337. Example of the schedkwhen opcode.
/* schedkwhen.orc */ ; 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 /* schedkwhen.orc */
/* schedkwhen.sco */ ; 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 /* schedkwhen.sco */
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 extant 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.)
Note: 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.
schedule insnum, iwhen, idur [, ip4] [, ip5] [...]
schedule "insname", iwhen, idur [, ip4] [, ip5] [...]
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.
ktrigger -- trigger value for new event
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.
Here is an example of the schedule opcode. It uses the files schedule.orc and schedule.sco.
Example 338. Example of the schedule opcode.
/* schedule.orc */ ; 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 /* schedule.orc */
/* schedule.sco */ ; 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 /* schedule.sco */
schedwhen ktrigger, kinsnum, kwhen, kdur [, ip4] [, ip5] [...]
schedwhen ktrigger, "insname", kwhen, kdur [, ip4] [, ip5] [...]
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 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.
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Support for named instruments is broken in version 4.23 |
Here is an example of the schedwhen opcode. It uses the files schedwhen.orc and schedwhen.sco.
Example 339. Example of the schedwhen opcode.
/* schedwhen.orc */ ; 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 kinsnum = 2 kwhen = 0 kdur = p3 ; Play Instrument #2 at the same time, if the trigger is set. schedwhen ktrigger, 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 /* schedwhen.orc */
/* schedwhen.sco */ ; 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, trigger Instrument #2. i 1 0 0.5 1 ; Play Instrument #1 for half a second, no trigger. i 1 1 0.5 0 e /* schedwhen.sco */
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.
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.
Here is an example of the sekere opcode. It uses the files sekere.orc and sekere.sco.
Example 340. Example of the sekere opcode.
/* sekere.orc */ ;orchestra --------------- sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 01 ;an example of a sekere a1 sekere p4, 0.01 out a1 endin /* sekere.orc */
/* sekere.sco */ ;score ------------------- i1 0 1 26000 e /* sekere.sco */
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.
Here is an example of the semitone opcode. It uses the files semitone.orc and semitone.sco.
Example 341. Example of the semitone opcode.
/* semitone.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; The root note is A above middle-C (440 Hz) iroot = 440 ; Raise the root note by three semitones to C. isemitone = 3 ; Calculate the new note. ifactor = semitone(isemitone) inew = iroot * ifactor ; Print out all of the values. print iroot print ifactor print inew endin /* semitone.orc */
/* semitone.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* semitone.sco */
Its output should include lines like:
instr 1: iroot = 440.000 instr 1: ifactor = 1.189 instr 1: inew = 523.229
Returns the ASCII code of a key that has been pressed, or -1 if no key has been pressed.
At release, this has not been properly verified, and seems not to work at all on Windows.
![]() | Note |
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This opcode can also be written as sense. |
Here is an example of the sensekey opcode. It uses the files sensekey.orc and sensekey.sco.
Example 342. Example of the sensekey opcode.
/* sensekey.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 k1 sensekey printk2 k1 endin /* sensekey.orc */
/* sensekey.sco */ ; Play Instrument #1 for thirty seconds. i 1 0 30 e /* sensekey.sco */
Here is what the output should look like when the "q" button is pressed...
q i1 357967744.00000
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
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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.
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 seqtime opcode.
Example 343. Example of the seqtime opcode.
instr 1 icps cpsmidi iamp ampmidi 5000 ktrig seqtime 1, 1, 10, 0, 1 trigseq ktrig, 0, 10, 0, 2, kdur, kampratio, kfreqratio schedkwhen ktrig, -1, -1, 2, 0, kdur, kampratio*iamp, kfreqratio*icps endin instr 2 **** put here your intrument code ******* out a1 endin
ktrig_out -- output trigger signal
ktime_unit -- unit of measure of time, related to seconds.
ktime_in -- input trigger signal.
kstart -- start index of looped section
kloop -- end index of looped section
kinitndx -- initial index
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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.
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.
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)
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.
Here is an example of the setctrl opcode. It uses the files setctrl.orc and setctrl.sco.
Example 344. Example of the setctrl opcode.
/* setctrl.orc */ ; 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 /* setctrl.orc */
/* setsctrl.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for thirty seconds. i 1 0 30 e /* setsctrl.sco */
Its output should include lines like this:
i1 38.00000 i1 40.00000 i1 43.00000
Sets the local ksmps value in a user-defined opcode block.
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). setksmps should be used before any other opcodes (but allowed after xin), otherwise unpredictable results may occur.
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).
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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 ksmps |
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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:
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 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), otherwise unpredictable results may occur.
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]
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.
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.
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.
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.
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.
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.
sfinstr3 — Plays a SoundFont2 (SF2) sample instrument, generating a stereo sound with cubic interpolation.
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.
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.
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.
sfinstr3m — Plays a SoundFont2 (SF2) sample instrument, generating a mono sound with cubic interpolation.
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.
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.
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.
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.
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.
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 plays 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.
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.
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 typed within double-quotes. Use “/” to separate directories. This applies to DOS and Windows as well, where using a backslash will generate an error.
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.
sfpassign — Assigns all presets of a SoundFont2 (SF2) sample file to a sequence of progressive index numbers.
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.
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.
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.
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.
ivel -- velocity value
inotenum -- MIDI note number value
ipreindex -- preset index
iflag -- flag regarding the behavior of xfreq and inotenum
ioffset (optional) -- start playing at offset, in samples.
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.
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.
sfplay3 — Plays a SoundFont2 (SF2) sample preset, generating a stereo sound with cubic interpolation.
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.
ivel -- velocity value
inotenum -- MIDI note number value
ipreindex -- preset index
iflag -- flag regarding the behavior of xfreq and inotenum
ioffset (optional) -- start playing at offset, in samples.
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.
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.
sfplay3m — Plays a SoundFont2 (SF2) sample preset, generating a mono sound with cubic interpolation.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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
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.
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 Perry Cook, but re-coded for Csound.
idecay -- If present indicates for how long at the end of the note the shaker is to be damped. The default value is zero.
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 |
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The argument knum was redundant, so it was removed in version 3.49. |
Here is an example of the shaker opcode. It uses the files shaker.orc and shaker.sco.
Example 345. Example of the shaker opcode.
/* shaker.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1 instr 1 a1 shaker 10000, 440, 8, 0.999, 100, 0 out a1 endin /* shaker.orc */
/* shaker.sco */ i 1 0 1 e /* shaker.sco */
Here is an example of the sin opcode. It uses the files sin.orc and sin.sco.
Example 346. Example of the sin opcode.
/* sin.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 25 i1 = sin(irad) print i1 endin /* sin.orc */
/* sin.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* sin.sco */
Its output should include a line like this:
instr 1: i1 = -0.132
Here is an example of the sinh opcode. It uses the files sinh.orc and sinh.sco.
Example 347. Example of the sinh opcode.
/* sinh.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 1 i1 = sinh(irad) print i1 endin /* sinh.orc */
/* sinh.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* sinh.sco */
Its output should a line like this:
instr 1: i1 = 1.175
Here is an example of the sininv opcode. It uses the files sininv.orc and sininv.sco.
Example 348. Example of the sininv opcode.
/* sininv.orc */ ; 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 /* sininv.orc */
/* sininv.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* sininv.sco */
Its output should include a line like this:
instr 1: i1 = 0.524
The sinsyn opcode takes an input containg a TRACKS pv streaming signal (as generated, for instance by the partials opcode). It sinsynthesises the signal using linear amplitude and cubic phase interpolation to drive a bank of interpolating oscillators with amplitude and pitch scaling controls. 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.
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)
Example 349. Example
ain inch 1 ; input signal fs1,fsi2 pvsifd ain,2048,512,1 ; ifd analysis fst partials fs1,fsi2,.003,1,3,500 ; partial tracking aout sinsyn fst, 1, 1.5, 500, 1 ; resynthesis (up a 5th) out aout
The example above shows partial tracking of an ifd-analysis signal and cubic-phase additive resynthesis.
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.
ares sleighbells kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] [, ifreq1] [, ifreq2]
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.
kamp -- Amplitude of output. Note: As these instruments are stochastic, this is only an approximation.
Here is an example of the sleighbells opcode. It uses the files sleighbells.orc and sleighbells.sco.
Example 350. Example of the sleighbells opcode.
/* sleighbells.orc */ sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1: An example of sleighbells. instr 1 a1 sleighbells 20000, 0.01 out a1 endin /* sleighbells.orc */
/* sleighbells.sco */ i 1 0.00 0.25 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 0.25 i 1 1.80 0.25 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 /* sleighbells.sco */
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
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
icutoff1 ... icutoff64 -- low-pass filter cutoff frequency for each controller
k1 ... k64 -- 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.
slider16f — 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
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
k1 ... k64 -- 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.
![]() | Warning |
---|---|
slider16f does not output the required initial value immediately, but only after some k-cycles because the filter slightly delays the output. |
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
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
icutoff1 ... icutoff64 -- low-pass filter cutoff frequency for each controller
k1 ... k64 -- 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.
slider32f — 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
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
k1 ... k64 -- 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.
![]() | Warning |
---|---|
slider32f opcodes do not output the required initial value immediately, but only after some k-cycles because the filter slightly delays the output. |
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
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
icutoff1 ... icutoff64 -- low-pass filter cutoff frequency for each controller
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.
slider64f — 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
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
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.
![]() | Warning |
---|---|
slider64f opcodes do not output the required initial value immediately, but only after some k-cycles because the filter slightly delays the output. |
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
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
icutoff1 ... icutoff64 -- low-pass filter cutoff frequency for each controller
k1 ... k64 -- 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.
slider8f — 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
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
k1 ... k64 -- 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.
![]() | Warning |
---|---|
slider8f opcodes do not output the required initial value immediately, but only after some k-cycles because the filter slightly delays the output. |
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.
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
kon --on signal: when 0, processing is bypassed. When switched on (kon >= 1), the opcode starts recording until the loop memory is full. It then plays the looped sound until it is switched off again (kon = 0). Another recording can start again with kon >= 1.
Example 351. Example
asig in ; get the signal in ktrig line 0, 1, 1 ; trigger signal aout,krec sndloop asig, 1, ktrig, 4, 0.05 ; rec starts at 1 sec, for 4 secs 0.05 crossfade printk 1, krec ; prints the recording signal out aout
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.
sndwarp — Reads a mono sound sample from a table and applies time-stretching and/or pitch modification.
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*.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.
ares [, ac] sndwarp xamp, xtimewarp, xresample, ifn1, ibeg, iwsize, irandw, ioverlap, ifn2, itimemode
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 a sine (i.e.: f1 0 16384 9 .5 1 0) which works quite well. Other shapes can also be used.
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 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, xitimewarp 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.
Here is an example of the sndwarp opcode. It uses the files sndwarp.orc, sndwarp.sco, and mary.wav.
Example 352. Example of the sndwarp opcode.
/* sndwarp.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - play an audio file. instr 1 ; Use the audio file defined in Table #1. a1 loscil 30000, 1, 1, 1 out a1 endin ; Instrument #2 - time-stretch an audio file. instr 2 kamp init 6500 ; Start at 1 second and end at 3.5 seconds. ktimewarp line 1, p3, 3.5 ; Playback at the normal speed. kresample init 1 ; Use the audio file defined in Table #1. ifn1 = 1 ibeg = 0 iwsize = 4410 irandw = 882 ioverlap = 15 ; Use Table #2 for the windowing function. ifn2 = 2 ; Use the ktimewarp parameter as a "time" pointer. itimemode = 1 a1 sndwarp kamp, ktimewarp, kresample, ifn1, ibeg, iwsize, irandw, ioverlap, ifn2, itimemode out a1 endin /* sndwarp.orc */
/* sndwarp.sco */ ; Table #1: an audio file. f 1 0 262144 1 "mary.wav" 0 0 0 ; Table #2: half of a sine wave. f 2 0 16384 9 0.5 1 0 ; Play Instrument #1 for 3.5 seconds. i 1 0 3.5 ; Play Instrument #2 for 7 seconds (time-stretched). i 2 3.5 10.5 e /* sndwarp.sco */
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 sndwarp 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 |
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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! |
sndwarpst — Reads a stereo sound sample from a table and applies time-stretching and/or pitch modification.
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.
ar1, ar2 [,ac1] [, ac2] sndwarpst xamp, xtimewarp, xresample, ifn1, ibeg, iwsize, irandw, ioverlap, ifn2, itimemode
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 a sine (i.e.: f1 0 16384 9 .5 1 0) which works quite well. Other shapes can also be used.
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, xitimewarp 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.
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 sndwarp 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 |
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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! |
ar1[, ar2[, ar3[, ... a24]]] soundin ifilcod [, iskptim] [, iformat] [, iskipinit] [, ibufsize]
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.
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 |
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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:
|
Here is an example of the soundin opcode. It uses the files soundin.orc, soundin.sco, beats.wav.
Example 353. Example of the soundin opcode.
/* soundin.orc */ ; Initialize the global variables. sr = 44100 kr = 44100 ksmps = 1 nchnls = 1 ; Instrument #1 - play an audio file. instr 1 asig soundin "beats.wav" out asig endin /* soundin.orc */
/* soundin.sco */ ; Play Instrument #1, the audio file, for three seconds. i 1 0 3 e /* soundin.sco */
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.
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.
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.
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 left 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 |
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If ifn is 0, then space will take its values for the xy coordinates from kx and ky. |
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.
instr 1 asig ;some audio signal ktime line 0, p3, p10 a1, a2, a3, a4 space asig,1, ktime, .1 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
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 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.
instr 1 ... a1, a2, a3, a4 space asig, 0, 0, .1, p4, p5 ar1, ar2, ar3, ar4 spsend ga1=ga1+ar1 ga2=ga2+ar2 outs a1, a2 endin instr 99 ; reverb.... .... endin
A few notes: p4 and p5 are the X and Y values
;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 e
The next example shows a simple intuitive use of the distance values returned by spdist to simulate Doppler shift.
ktime line 0, p3, 10 kdist spdist 1, ktime kfreq = (ifreq * 340) / (340 + kdist) asig oscili iamp, kfreq, 1 a1, a2, a3, a4 space asig, 1, ktime, .1 ar1, ar2, ar3, ar4 spsend
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!
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).
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(iX^2 + iY^2 + iZ^2)
In Mode 4, distance can be calculated as:
distance from left mic = sqrt((iX + idist/2)^2 + iY^2 + iZ^2)
distance from right mic = sqrt((iX - idist/2)^2 + iY^2 + iZ^2)
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).
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.
Here is a example of the spat3d opcode that outputs a stereo file. It uses the files spat3d_stereo.orc and spat3d_stereo.sco.
Example 354. Stereo example of the spat3d opcode.
/* spat3d_stereo.orc */ /* 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 /* spat3d_stereo.orc */
/* spat3d_stereo.sco */ /* Written by Istvan Varga */ i 1 0 10 e /* spat3d_stereo.sco */
Here is a example of the spat3d opcode that outputs a UHJ file. It uses the files spat3d_UHJ.orc and spat3d_UHJ.sco.
Example 355. UHJ example of the spat3d opcode.
/* spat3d_UHJ.orc */ /* 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 /* spat3d_UHJ.orc */
/* spat3d_UHJ.sco */ /* Written by Istvan Varga */ t 0 60 i 1 0.0 8.0 e /* spat3d_UHJ.sco */
Here is a example of the spat3d opcode that outputs a quadrophonic file. It uses the files spat3d_quad.orc and spat3d_quad.sco.
Example 356. Quadrophonic example of the spat3d opcode.
/* spat3d_quad.orc */ /* 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 /* spat3d_quad.orc */
/* spat3d_quad.sco */ /* Written by Istvan Varga */ t 0 60 i 1 0 10 e /* spat3d_quad.sco */
spat3di — Positions the input sound in a 3D space with the sound source position set at i-time.
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.
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(iX^2 + iY^2 + iZ^2)
In Mode 4, distance can be calculated as:
distance from left mic = sqrt((iX + idist/2)^2 + iY^2 + iZ^2)
distance from right mic = sqrt((iX - idist/2)^2 + iY^2 + iZ^2)
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).
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.
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.
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(iX^2 + iY^2 + iZ^2)
In Mode 4, distance can be calculated as:
distance from left mic = sqrt((iX + idist/2)^2 + iY^2 + iZ^2)
distance from right mic = sqrt((iX - idist/2)^2 + iY^2 + iZ^2)
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).
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.
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 left 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.
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.
instr 1 asig ;some audio signal ktime line 0, p3, p10 a1, a2, a3, a4 space asig,1, ktime, .1 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
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 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.
instr 1 ... a1, a2, a3, a4 space asig, 0, 0, .1, p4, p5 ar1, ar2, ar3, ar4 spsend ga1=ga1+ar1 ga2=ga2+ar2 outs a1, a2 endin instr 99 ; reverb.... .... endin
A few notes: p4 and p5 are the X and Y values
;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 e
The next example shows a simple intuitive use of the distance values returned by spdist to simulate Doppler shift.
ktime line 0, p3, 10 kdist spdist 1, ktime kfreq = (ifreq * 340) / (340 + kdist) asig oscili iamp, kfreq, 1 a1, a2, a3, a4 space asig, 1, ktime, .1 ar1, ar2, ar3, ar4 spsend
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!
imul2 (optional, default=0) -- if non-zero, scale the wsig2 magnitudes before adding. The default value is 0.
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).
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.
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).
wsig -- the input spectrum.
Displays the magnitude values of spectrum wsig every iprd seconds (rounded to some integral number of wsig's originating iprd).
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.
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.
koct, kamp specptrk wsig, kvar, ilo, ihi, istr, idbthresh, inptls, irolloff [, iodd] [, iconfs] [, interp] [, ifprd] [, iwtflg]
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).
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.)
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, .01, 7, 24, 15, 0, 3 ; get a 7-oct spectrum, 24 bibs/oct specdisp wsig, .2 ; display this and now estimate koct,ka spectrk wsig, kvar, 7.0, 10, 9, 20, 4, .7, 1, 5, 1, .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, .25, 20 ; & display the pitch track display ka, .25, 20 ; plus the summed root mag outs a1, aosc ; output 1 original and 1 new track
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)
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.
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).
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.
Generate a constant-Q, exponentially-spaced DFT across all octaves of a multiply-downsampled control or audio input signal.
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).
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.
splitrig splits a trigger signal (i.e. a timed sequence of control-rate impulses) into several channels following a structure designed by the user.
imaxtics - number of tics belonging to largest pattern
ifn - number of table containing channel-data structuring
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 cicle is repeated.
The scheme of patterns is defined by the user and is stored into ifn table according to the following format:
gi1 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,\
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 cannel, 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 cicle 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
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).
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.
instr 1 asig ;some audio signal ktime line 0, p3, p10 a1, a2, a3, a4 space asig,1, ktime, .1 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
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 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.
instr 1 ... a1, a2, a3, a4 space asig, 0, 0, .1, p4, p5 ar1, ar2, ar3, ar4 spsend ga1=ga1+ar1 ga2=ga2+ar2 outs a1, a2 endin instr 99 ; reverb.... .... endin
A few notes: p4 and p5 are the X and Y values
;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 e
The next example shows a simple intuitive use of the distance values returned by spdist to simulate Doppler shift.
ktime line 0, p3, 10 kdist spdist 1, ktime kfreq = (ifreq * 340) / (340 + kdist) asig oscili iamp, kfreq, 1 a1, a2, a3, a4 space asig, 1, ktime, .1 ar1, ar2, ar3, ar4 spsend
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!
sprintf and sprintfk write printf-style formatted output to a string variable, similarly to the C function sprintf(). sprintf runs at i-time only, while sprintfk runs both at initialization and performance time.
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.
Returns the square root of x (x non-negative).
The argument value is restricted for log, log10, and sqrt.
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.
Here is an example of the sqrt opcode. It uses the files sqrt.orc and sqrt.sco.
Example 357. Example of the sqrt opcode.
/* sqrt.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 = sqrt(64) print i1 endin /* sqrt.orc */
/* sqrt.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* sqrt.sco */
Its output should include lines like this:
instr 1: i1 = 8.000
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.
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. Csound will attempt to calculate the omitted value from the specified values, but it should evaluate to an integer.
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.
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.
ahp -- high-pass output signal.
alp -- low-pass output signal.
abp -- band-pass signal.
abr -- band-reject signal.
asig -- input signal.
kcf -- filter cutoff frequency
kq -- filter Q. 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).
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.
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.
Here is an example of the stix opcode. It uses the files stix.orc and stix.sco.
Example 359. Example of the stix opcode.
/* stix.orc */ ;orchestra --------------- sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 01 ;an example of stix a1 line 20, p3, 20 ;preset amplitude increase a2 stix p4, 0.01 ;stix needs a little amp help at these settings a3 product a1, a2 ;increase amplitude out a3 endin /* stix.orc */
/* stix.sco */ ;score ------------------- i1 0 1 26000 e /* stix.sco */
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.
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.
streson is an adaptation of the StringFlt object of the SndObj Sound Object Library developed by the author.
Here is an example of the streson opcode. It uses the files streson.orc and streson.sco.
Example 360. Example of the streson opcode.
/* streson.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a normal sine wave. asig oscils 8000, 440, 1 ; Vary the fundamental frequency of the string ; resonator linearly from 220 to 880 Hertz. kfr line 220, p3, 880 ifdbgain = 0.95 ; Run our sine wave through the string resonator. astres streson asig, kfr, ifdbgain ; The resonance can get quite loud. ; So we'll clip the signal at 30,000. a1 clip astres, 1, 30000 out a1 endin /* streson.orc */
/* streson.sco */ ; Play Instrument #1 for five seconds. i 1 0 5 e /* streson.sco */
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.
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.
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If a k-rate index variable is used, it should be valid at i-time as well. |
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 a floating point or integer number, an init or perf error occurs and the instrument is deactivated.
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.
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If a k-rate index variable is used, it should be valid at i-time as well. |
kr strtolk Sstr
kr strtolk kndx
strtolk can parse numbers in decimal, octal (prefixed by 0), and hexadecimal (with a prefix of 0x) format.
a1, [...] [, a8] subinstr instrnum [, p4] [, p5] [...]
a1, [...] [, a8] subinstr "insname" [, p4] [, p5] [...]
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.
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.
Here is an example of the subinstr opcode. It uses the files subinstr.orc and subinstr.sco.
Example 361. Example of the subinstr opcode.
/* subinstr.orc */ ; 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 /* subinstr.orc */
/* subinstr.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #2 for one second. i 2 0 1 e /* subinstr.sco */
Here is an example of the subinstr opcode using a named instrument. It uses the files subinstr_named.orc and subinstr_named.sco.
Example 362. Example of the subinstr opcode using a named instrument.
/* subinstr_named.orc */ ; 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 /* subinstr_named.orc */
/* subinstr_named.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* subinstr_named.sco */
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.
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.
svfilter — A resonant second order filter, with simultaneous lowpass, highpass and bandpass outputs.
Implementation of a resonant second order filter, with simultaneous lowpass, highpass and bandpass outputs.
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.
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).
Here is an example of the svfilter opcode. It uses the files svfilter.orc and svfilter.sco.
Example 363. Example of the svfilter opcode.
/* svfilter.orc */ ; 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 /* svfilter.orc */
/* svfilter.sco */ 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 /* svfilter.sco */
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 (with aif files).
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 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).
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.
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.
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.
An event-sequencer in which time can be controlled by a time-pointer. Sequence data are stored into a table.
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 positve 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 sequnces 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 schedk or schedkwhen to actually activate note events. Notice that kp1,...kpn data can be further processed (for example delayed with delayk, transposed, etc.) before feeding schedk or schedkwhen.
ktimepoint can be controlled by linear signal, for example:
ktimpnt line 0,p3,3 ; orignal sequence duration was 3 secs ktrig timedseq ktimpnt,1,kp1,kp2,kp3,kp4,kp5 schedk ktrig, 105, 2, 0, kp3,kp4,kp5
in this case the complete sequence (with orginal duration of 3 seconds) will be played in p3 seconds.
You can loop a sequence by contolling it with a phasor:
kphs phasor 1/3 ktimpnt = kphs * 3 ktrig timedseq ktimpnt,1,kp1,kp2,kp3,kp4,kp5 schedk 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: 1. 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 schedk mechanism eats one of them). 2. 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.
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.
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.
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)
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.
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).
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]
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
tabrec ktrig_start, ktrig_stop, knumtics, kfn, kin1 [,kin2,...,kinN]
tabplay ktrig, knumtics, kfn, kout1 [,kout2,..., koutN]
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.
ktrig -- starts playing when non-zero.
kout1,...,koutN -- playback output signals.
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 impluses have been received by ktrig_stop argument.
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.
ares table andx, ifn [, ixmode] [, ixoff] [, iwrap]
ires table indx, ifn [, ixmode] [, ixoff] [, iwrap]
kres table kndx, ifn [, ixmode] [, ixoff] [, iwrap]
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.
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.
Here is an example of the table opcode. It uses the files table.orc and table.sco.
Example 365. Example of the table opcode.
/* table.orc */ ; 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.orc */
/* table.sco */ ; 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 /* table.sco */
ares table3 andx, ifn [, ixmode] [, ixoff] [, iwrap]
ires table3 indx, ifn [, ixmode] [, ixoff] [, iwrap]
kres table3 kndx, ifn [, ixmode] [, ixoff] [, iwrap]
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.
table3 is experimental, and is identical to tablei, except that it uses cubic interpolation. (New in Csound version 3.50.)
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.
ares tablei andx, ifn [, ixmode] [, ixoff] [, iwrap]
ires tablei indx, ifn [, ixmode] [, ixoff] [, iwrap]
kres tablei kndx, ifn [, ixmode] [, ixoff] [, iwrap]
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.
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).
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.
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.
k-rate control over table numbers.
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.
ares tableikt xndx, kfn [, ixmode] [, ixoff] [, iwrap]
kres tableikt kndx, kfn [, ixmode] [, ixoff] [, iwrap]
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.
kndx -- Index into table, either a positive number range
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 |
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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. |
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.
tableimix -- 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.
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.
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.
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 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.
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.
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.
ares tablekt xndx, kfn [, ixmode] [, ixoff] [, iwrap]
kres tablekt kndx, kfn [, ixmode] [, ixoff] [, iwrap]
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.
kndx -- Index into table, either a positive number range
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 |
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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. |
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.
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.
Here is an example of the tableng opcode. It uses the files tableng.orc and tableng.sco.
Example 366. Example of the tableng opcode.
/* tableng.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Let's look at Table #1. ifn = 1 ilen tableng ifn print ilen endin /* tableng.orc */
/* tableng.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* tableng.sco */
The table is 16,384 samples long. So its output should include a line like this:
instr 1: ilen = 16384.000
These opcodes read and write tables in sequential locations to and from an a-rate variable. Some thought is required before using them. They have at least two major, and quite different, applications which are discussed below.
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, these are intended to be used in pairs, 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 |
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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.
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.
tableseg — Creates a new function table by making linear segments between values in stored function tables.
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.
Note: this opcode can also be written as ktableseg.
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. The valid combinations of variable types are shown by the first letter of the variable names.
tablew asig, andx, ifn [, ixmode] [, ixoff] [, iwgmode]
tablew isig, indx, ifn [, ixmode] [, ixoff] [, iwgmode]
tablew ksig, kndx, ifn [, ixmode] [, ixoff] [, iwgmode]
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.
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 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.
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.
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.
These opcodes read and write tables in sequential locations to and from an a-rate variable. Some thought is required before using them. They have at least two major, and quite different, applications which are discussed below.
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, these are intended to be used in pairs, 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 |
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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.
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.
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.
tablewkt asig, andx, kfn [, ixmode] [, ixoff] [, iwgmode]
tablewkt ksig, kndx, kfn [, ixmode] [, ixoff] [, iwgmode]
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.
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 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.
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.
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.
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)
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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).
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iwrap also applies to extra samples for interpolation. |
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.
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kwarp has no effect if it is less than, or equal to 1, or linear or cubic interpolation is used. |
tablexseg — Creates a new function table by making exponential segments between values in stored function tables.
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.
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.
ares tambourine kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] [, ifreq1] [, ifreq2]
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.
kamp -- Amplitude of output. Note: As these instruments are stochastic, this is only an approximation.
Here is an example of the tambourine opcode. It uses the files tambourine.orc and tambourine.sco.
Example 367. Example of the tambourine opcode.
/* tambourine.orc */ sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1: An example of a tambourine. instr 01 a1 tambourine 15000, 0.01 out a1 endin /* tambourine.orc */
/* tambourine.sco */ i 1 0 1 e /* tambourine.sco */
Here is an example of the tan opcode. It uses the files tan.orc and tan.sco.
Example 368. Example of the tan opcode.
/* tan.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 25 i1 = tan(irad) print i1 endin /* tan.orc */
/* tan.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* tan.sco */
Its output should include a line like this:
instr 1: i1 = -0.134
Here is an example of the tanh opcode. It uses the files tanh.orc and tanh.sco.
Example 369. Example of the tanh opcode.
/* tanh.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 irad = 1 i1 = tanh(irad) print i1 endin /* tanh.orc */
/* tanh.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* tanh.sco */
Its output should include a line like this:
instr 1: i1 = 0.762
Here is an example of the taninv opcode. It uses the files taninv.orc and taninv.sco.
Example 370. Example of the taninv opcode.
/* taninv.orc */ ; 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 /* taninv.orc */
/* taninv.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* taninv.sco */
Its output should include a line like this:
instr 1: i1 = 0.464
ares taninv2 ay, ax
ires taninv2 iy, ix
kres taninv2 ky, kx
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:
PI/2, if y is positive.
-PI/2, if y is negative.
0, if y is 0.
ky, kx -- control rate signals to be converted
ay, ax -- audio rate signals to be converted
Here is an example of the taninv2 opcode. It uses the files taninv2.orc and taninv2.sco.
Example 371. Example of the taninv2 opcode.
/* taninv2.orc */ ; 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 /* taninv2.orc */
/* taninv2.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* taninv2.sco */
Its output should include a line like this:
instr 1: i1 = 0.464
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.
iskip (optional, default=0) -- if non zero skip the initialisation of the filter. (New in Csound version 4.23f13 and 5.0)
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.
Here is an example of the tbvcf opcode. It uses the files tbvcf.orc and tbvcf.sco.
Example 372. Example of the tbvcf opcode.
/* tbvcf.orc */ ;--------------------------------------------------------- ; TBVCF Test ; Coded by Hans Mikelson December, 2000 ;--------------------------------------------------------- sr = 44100 ; Sample rate kr = 4410 ; Kontrol rate ksmps = 10 ; Samples/Kontrol period nchnls = 2 ; Normal stereo zakinit 50, 50 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, 1 ; 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 /* tbvcf.orc */
/* tbvcf.sco */ f1 0 65536 10 1 ; TeeBee Test ; Sta Dur Amp Pitch Pan Q Dist1 Asym i10 0 0.2 32767 7.00 .5 0.0 2.0 0.0 i10 0.3 0.2 32767 7.00 .5 0.8 2.0 0.0 i10 0.6 0.2 32767 7.00 .5 1.6 2.0 0.0 i10 0.9 0.2 32767 7.00 .5 2.4 2.0 0.0 i10 1.2 0.2 32767 7.00 .5 3.2 2.0 0.0 i10 1.5 0.2 32767 7.00 .5 4.0 2.0 0.0 i10 1.8 0.2 32767 7.00 .5 0.0 2.0 0.25 i10 2.1 0.2 32767 7.00 .5 0.8 2.0 0.25 i10 2.4 0.2 32767 7.00 .5 1.6 2.0 0.25 i10 2.7 0.2 32767 7.00 .5 2.4 2.0 0.25 i10 3.0 0.2 32767 7.00 .5 3.2 2.0 0.25 i10 3.3 0.2 32767 7.00 .5 4.0 2.0 0.25 i10 3.6 0.2 32767 7.00 .5 0.0 2.0 0.5 i10 3.9 0.2 32767 7.00 .5 0.8 2.0 0.5 i10 4.2 0.2 32767 7.00 .5 1.6 2.0 0.5 i10 4.5 0.2 32767 7.00 .5 2.4 2.0 0.5 i10 4.8 0.2 32767 7.00 .5 3.2 2.0 0.5 i10 5.1 0.2 32767 7.00 .5 4.0 2.0 0.5 i10 5.4 0.2 32767 7.00 .5 0.0 2.0 0.75 i10 5.7 0.2 32767 7.00 .5 0.8 2.0 0.75 i10 6.0 0.2 32767 7.00 .5 1.6 2.0 0.75 i10 6.3 0.2 32767 7.00 .5 2.4 2.0 0.75 i10 6.6 0.2 32767 7.00 .5 3.2 2.0 0.75 i10 6.9 0.2 32767 7.00 .5 4.0 2.0 0.75 i10 7.2 0.2 32767 7.00 .5 0.0 2.0 1.0 i10 7.5 0.2 32767 7.00 .5 0.8 2.0 1.0 i10 7.8 0.2 32767 7.00 .5 1.6 2.0 1.0 i10 8.1 0.2 32767 7.00 .5 2.4 2.0 1.0 i10 8.4 0.2 32767 7.00 .5 3.2 2.0 1.0 i10 8.7 0.2 32767 7.00 .5 4.0 2.0 1.0 e /* tbvcf.sco */
ktemp tempest kin, iprd, imindur, imemdur, ihp, ithresh, ihtim, ixfdbak, istartempo, ifn [, idisprd] [, itweek]
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).
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.
Here is an example of the tempest opcode. It uses the files tempest.orc, tempest.sco, and beats.wav.
Example 373. Example of the tempest opcode.
/* tempest.orc */ ; 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 /* tempest.orc */
/* tempest.sco */ ; Table #1, a declining line. f 1 0 128 16 1 128 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* tempest.sco */
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
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.
Here is an example of the tempo opcode. Remember, it only works if you use the -t flag with Csound. The example uses the files tempo.orc and tempo.sco.
Example 374. Example of the tempo opcode.
/* tempo.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; 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 /* tempo.orc */
/* tempo.sco */ ; 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 00.10 0 i 1 00.25 00.10 0 i 1 00.50 00.10 0 i 1 00.75 00.10 0 s ; Play Instrument #1 at a faster tempo, repeat 3 times. r3 i 1 00.00 00.10 1 i 1 00.25 00.10 1 i 1 00.50 00.10 1 i 1 00.75 00.10 1 s e /* tempo.sco */
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).
Here is an example of the tempoval opcode. Remember, it only works if you use the -t flag with Csound. It uses the files tempoval.orc and tempoval.sco.
Example 375. Example of the tempoval opcode.
/* tempoval.orc */ ; 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 /* tempoval.orc */
/* tempoval.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* tempoval.sco */
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
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, delay).
tigoto label
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).
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.
timeinstk 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.
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.
Here is an example of the timeinstk opcode. It uses the files timeinstk.orc and timeinstk.sco.
Example 376. Example of the timeinstk opcode.
/* timeinstk.orc */ ; 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 /* timeinstk.orc */
/* timeinstk.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 e /* timeinstk.sco */
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
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.
Here is an example of the timeinsts opcode. It uses the files timeinsts.orc and timeinsts.sco.
Example 377. Example of the timeinsts opcode.
/* timeinsts.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Print out the value from timeinsts every half-second. k1 timeinsts printks "k1 = %f seconds\\n", 0.5, k1 endin /* timeinsts.orc */
/* timeinsts.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 e /* timeinsts.sco */
Its output should include lines like this:
k1 = 0.000227 seconds k1 = 0.500000 seconds k1 = 1.000000 seconds k1 = 1.500000 seconds k1 = 2.000000 seconds
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.
Here is an example of the timek opcode. It uses the files timek.orc and timek.sco.
Example 378. Example of the timek opcode.
/* timek.orc */ ; 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 /* timek.orc */
/* timek.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 e /* timek.sco */
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
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.
Here is an example of the times opcode. It uses the files times.orc and times.sco.
Example 379. Example of the times opcode.
/* times.orc */ ; 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 /* times.orc */
/* times.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 e /* times.sco */
Its output should include lines like this:
k1 = 0.000227 seconds k1 = 0.500000 seconds k1 = 1.000000 seconds k1 = 1.500000 seconds k1 = 2.000000 seconds
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).
timout istrt, idur, label
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).
tival — Puts the value of the instrument's internal “tie-in” flag into the named i-rate variable.
Puts the value of the instrument's internal “tie-in” flag into the named i-rate variable.
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.)
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 useage is different.
A first-order recursive low-pass 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
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.
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.
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.
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.
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.
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.
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.
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)
Example 380. Example
ain inch 1 ; input signal fs1,fsi2 pvsifd ain,2048,512,1 ; ifd analysis fst partials fs1,fsi2,.003,1,3,500 ; partial tracking aout tradsyn fst, 1, 1.5, 500, 1 ; resynthesis (up a 5th) out aout
The example above shows partial tracking of an ifd-analysis signal and linear additive resynthesis with pitch shifting.
ares transeg ia, idur, itype, ib [, idur2] [, itype] [, ic] ...
kres transeg ia, idur, itype, ib [, idur2] [, itype] [, ic] ...
ia -- starting value.
ib, ic, etc. -- value after idur seconds.
idur, idur2, 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))
If itype > 0, there is a slowly rising, fast decaying (convex) curve, while if itype < 0, the curve is fast rising, slowly decaying (concave). See also GEN16.
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.
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.
Example 381. Example
ain inch 1 ; input signals ain inch 2 fs1,fsi2 pvsifd ain,2048,512,1 ; ifd analysis fst partials fs1,fsi2,.003,1,3,500 ; partial tracking fs11,fsi12 pvsifd ain,2048,512,1 ; ifd analysis (second input) fst1 partials fs11,fsi12,.003,1,3,500 ; partial tracking \(second input fcr trcross fst,fst1, 1.05, 1 ; cross-synthesis (mode 0) aout tradsyn fcr, 1, 1, 500, 1 ; resynthesis of tracks out aout
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.
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.
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.
Example 382. Example
gifn ftgen 2, 0, -22050, 5 1 1000 1 4000 0.000001 17050 0.000001 ; low-pass filter curve of 22050 points instr 1 ain inch 1 ; input signal fs1,fsi2 pvsifd ain,2048,512,1 ; ifd analysis fst partials fs1,fsi2,.003,1,3,500 ; partial tracking fscl trfilter fst, 1, gifn ; filtering using function table 2 aout tradsyn fscl, 1, 1, 500, 1 ; resynthesis out aout endin
The example above shows partial tracking of an ifd-analysis signal and linear additive resynthesis with low-pass filtering.
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.
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.
Example 383. Example
ain inch 1 ; input signal fs1,fsi2 pvsifd ain,2048,512,1 ; ifd analysis fst partials fs1,fsi2,.003,1,3,500 ; partial tracking fhi,kfr,kamp trhighest fst,1 ; highest freq-track aout tradsyn fhi, 1, 1, 1, 1 ; resynthesis of highest frequency out aout
The example above shows partial tracking of an ifd-analysis signal, extraction of the highest frequency and resynthesis.
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.
Here is an example of the trigger opcode. It uses the files trigger.orc and trigger.sco.
Example 384. Example of the trigger opcode.
/* trigger.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a square-wave low frequency oscillator as the trigger. klf lfo 1, 10, 3 ktr trigger klf, 1, 2 ; When the value of the trigger isn't equal to 0, print it out. if (ktr == 0) kgoto contin ; Print the value of the trigger and the time it occurred. ktm times printks "time = %f seconds, trigger = %f\\n", 0, ktm, ktr contin: ; Continue with processing. endin /* trigger.orc */
/* trigger.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* trigger.sco */
Its output should include lines like this:
time = 0.050340 seconds, trigger = 1.000000 time = 0.150340 seconds, trigger = 1.000000 time = 0.250340 seconds, trigger = 1.000000 time = 0.350340 seconds, trigger = 1.000000 time = 0.450340 seconds, trigger = 1.000000 time = 0.550340 seconds, trigger = 1.000000 time = 0.650340 seconds, trigger = 1.000000 time = 0.750340 seconds, trigger = 1.000000 time = 0.850340 seconds, trigger = 1.000000 time = 0.950340 seconds, trigger = 1.000000
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 -- numer 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 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 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.
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.
Here is an example of the trirand opcode. It uses the files trirand.orc and trirand.sco.
Example 385. Example of the trirand opcode.
/* trirand.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a random number between -1 and 1. ; krange = 1 i1 trirand 1 print i1 endin /* trirand.orc */
/* trirand.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* trirand.sco */
Its output should include lines like this:
instr 1: i1 = 7506.261
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.
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.
Example 386. Example
ain inch 1 ; input signal 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 out aout
The example above shows partial tracking of an ifd-analysis signal, extraction of the lowest frequency and resynthesis.
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.
fsig -- output pv stream in TRACKS format
fin1 -- first input pv stream in TRACKS format.
fin2 -- second input pv stream in TRACKS format
Example 387. Example
ain inch 1 ; input signal 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 fscl trscale fshi, 1.15 ; shift the upper tracks fmix trmix fslo,fscl ; mix the shifted and unshifted tracks aout tradsyn fmix, 1, 1, 500, 1 ; resynthesis of tracks out aout
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.
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.
fsig -- output pv stream in TRACKS format
fin -- input pv stream in TRACKS format
kpitch -- frequency scaling
kgain -- amplitude scaling (default 1)
Example 388. Example
ain inch 1 ; input signal fs1,fsi2 pvsifd ain,2048,512,1 ; ifd analysis fst partials fs1,fsi2,.003,1,3,500 ; partial tracking fscl trscale fst, 1.5 ; frequency scale (up a 5th) aout tradsyn fscl, 1, 1, 500, 1 ; resynthesis out aout
The example above shows partial tracking of an ifd-analysis signal and linear additive resynthesis with pitch shifting.
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.
fsig -- output pv stream in TRACKS format
fin -- input pv stream in TRACKS format
kshift -- frequency shift in Hz
kgain -- amplitude scaling (default 1)
Example 389. Example
ain inch 1 ; input signal fs1,fsi2 pvsifd ain,2048,512,1 ; ifd analysis fst partials fs1,fsi2,.003,1,3,500 ; partial tracking fscl trshift fst, 150 ; frequency shift (adds 150Hz to all tracks) aout tradsyn fscl, 1, 1, 500, 1 ; resynthesis out aout
The example above shows partial tracking of an ifd-analysis signal and linear additive resynthesis with frequency shifting.
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.
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).
Example 390. Example
ain inch 1 ; input signal 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 out aout
The example above shows partial tracking of an ifd-analysis signal and linear additive resynthesis of the upper part of the spectrum (from 1500Hz).
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.
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 files turnoff.orc and turnoff.sco.
Example 391. Example of the turnoff opcode.
/* turnoff.orc */ ; 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 /* turnoff.orc */
/* turnoff.sco */ ; Table #1: an ordinary sine wave. f 1 0 32768 10 1 ; Play Instrument #1 for 4 seconds. i 1 0 4 e /* turnoff.sco */
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)
insnum -- instrument number to be activated
itime (optional, default=0) -- delay, in seconds, after which instrument insnum will be activated. Default is 0.
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.)
Uniform distribution random number generator (positive values only). This is an x-class noise generator.
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.
Here is an example of the unirand opcode. It uses the files unirand.orc and unirand.sco.
Example 392. Example of the unirand opcode.
/* unirand.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a random number between 0 and 1. ; krange = 1 i1 unirand 1 print i1 endin /* unirand.orc */
/* unirand.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* unirand.sco */
Its output should include lines like this:
instr 1: i1 = 0.840
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.
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
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
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.
ifn - number of the table hosting the vectorial signal to be processed
ielements - number of elements of the vector
kval - scalar operand to be processed
vadd adds kval operand to each element of the vector contained in the table ifn.
These opcodes (vadd, vmult, vpow, 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.
In all these opcodes, the resulting vectors are stored in ifn, overriding the intial vectors. If you want to keep initial vector, use vcopy opcode 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.
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
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 opcode to copy it in another table.
This opcode works at k-rate.
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.
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.
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 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). Its output will begin to appear immediately.
iazim -- azimuth angle of the virtual source
ielev (optional) -- elevation angle of the virtual source
ispread (optional) -- 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.
asig -- audio signal to be panned
vbap16 takes an input signal, asig, and distribute it among 16 outputs, according to the controls iazim and ielev, and the configured loudspeaker placement. If idim = 2, ielev 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.
Example 393. 2-D panning example with stationary virtual sources
sr = 4100 kr = 441 ksmps = 100 nchnls = 4 vbaplsinit 2, 6, 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 endin
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.
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.
Example 394. 2-D panning example with stationary virtual sources
sr = 4100 kr = 441 ksmps = 100 nchnls = 4 vbaplsinit 2, 6, 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 endin
iazim -- azimuth angle of the virtual source
ielev (optional) -- elevation angle of the virtual source
ispread (optional) -- 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.
asig -- audio signal to be panned
vbap4 takes an input signal, asig and distributes it among 4 outputs, according to the controls iazim and ielev, and the configured loudspeaker placement. If idim = 2, ielev 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.
Example 395. 2-D panning example with stationary virtual sources
sr = 4100 kr = 441 ksmps = 100 nchnls = 4 vbaplsinit 2, 6, 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 endin
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).
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.
Example 396. 2-D panning example with stationary virtual sources
sr = 4100 kr = 441 ksmps = 100 nchnls = 4 vbaplsinit 2, 6, 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 endin
iazim -- azimuth angle of the virtual source
ielev (optional) -- elevation angle of the virtual source
ispread (optional) -- 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.
asig -- audio signal to be panned
vbap8 takes an input signal, asig, and distributes it among 8 outputs, according to the controls iazim and ielev, and the configured loudspeaker placement. If idim = 2, ielev 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.
Example 397. 2-D panning example with stationary virtual sources
sr = 4100 kr = 441 ksmps = 100 nchnls = 4 vbaplsinit 2, 6, 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 endin
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).
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.
Example 398. 2-D panning example with stationary virtual sources
sr = 4100 kr = 441 ksmps = 100 nchnls = 4 vbaplsinit 2, 6, 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 endin
idim -- dimensionality of loudspeaker array. Either 2 or 3.
ilsnum -- number of loudspeakers. In two dimensions, the number can vary from 2 to 16. In three dimensions, the number can vary from 3 and 16.
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.).
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.
Example 399. 2-D panning example with stationary virtual sources
sr = 4100 kr = 441 ksmps = 100 nchnls = 4 vbaplsinit 2, 6, 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 endin
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
iazim -- azimuth angle of the virtual source
ielev (optional) -- elevation angle of the virtual source
ispread (optional) -- 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.
asig -- audio signal to be panned
The opcode vbapz is the multiple channel analog of the opcodes like vbap4, working on inumchnls and using a ZAK array for output.
Example 400. 2-D panning example with stationary virtual sources
sr = 4100 kr = 441 ksmps = 100 nchnls = 4 vbaplsinit 2, 6, 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 endin
vbapzmove — Writes a multi-channel audio signal to a ZAK array with moving virtual sources.
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).
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.
Example 401. 2-D panning example with stationary virtual sources
sr = 4100 kr = 441 ksmps = 100 nchnls = 4 vbaplsinit 2, 6, 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 endin
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)
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).
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.
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)
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.
Here is an example of the vco opcode. It uses the files vco.orc and vco.sco.
Example 402. Example of the vco opcode.
/* vco.orc */ ; 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 /* vco.orc */
/* vco.sco */ ; 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 /* vco.sco */
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.
imode (optional, default=0) -- a sum of values representing the waveform and its control values.
One may use of 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.
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.
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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.
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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:
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Here is an example of the vco2 opcode. It uses the files vco2.orc and vco2.sco.
Example 403. Example of the vco2 opcode.
/* vco2.orc */ 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 /* vco2.orc */
/* vco2.sco */ 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 /* vco2.sco */
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).
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.
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).
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.
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.
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.
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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. |
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.
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 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 will appear only after ilpt seconds.
ifn - number of the table where the vectorial signal will be copied
ifn - number of the table hosting the vectorial signal to be copied
ielements - number of elements of the vector
vcopy copies ifn2 to ifn1. Useful to keep old vector values, by storing them in another table.
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
Here is an example of the vcopy opcode. It uses the files vcopy.csd.
Example 404. Example of the vcopy opcode.
<CsoundSynthesizer> <CsOptions> ;use appropriate realtime flags ;-+rtaudio=jack -odac:alsa_pcm:playback_ -B256 -b256 </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.aiff" 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>
ifn - number of the table where the vectorial signal will be copied
ifn - number of the table hosting the vectorial signal to be copied
ielements - number of elements of the vector
vcopy copies ifn2 to ifn1. 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.
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
Here is an example of the vcopy opcode. It uses the files vcopy.csd.
Example 405. Example of the vcopy opcode.
<CsoundSynthesizer> <CsOptions> ;use appropriate realtime flags ;-+rtaudio=jack -odac:alsa_pcm:playback_ -B256 -b256 </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.aiff" 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>
This is an interpolating variable time delay, it is not very different from the existing implementation (deltapi), it is only easier to use.
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
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.
f1 0 8192 10 1 ims = 100 ; Maximum delay time in msec a1 oscil 10000, 1737, 1 ; Make a signal a2 oscil ims/2, 1/p3, 1 ; Make an LFO a2 = a2 + ims/2 ; Offset the LFO so that it is positive a3 vdelay a1, a2, ims ; Use the LFO to control delay time out a3
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.
vdelay3 is experimental. It is the same as vdelay except that it uses cubic interpolation. (New in Version 3.50.)
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.
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.
f1 0 8192 10 1 ims = 100 ; Maximum delay time in msec a1 oscil 10000, 1737, 1 ; Make a signal a2 oscil ims/2, 1/p3, 1 ; Make an LFO a2 = a2 + ims/2 ; Offset the LFO so that it is positive a3 vdelay a1, a2, ims ; Use the LFO to control delay time out a3
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.
aout -- output audio signal
ain -- input audio signal
adl -- delay time in seconds
imd -- max. delay time (seconds)
iws -- interpolation window size (see below)
ist (optional) -- skip initialization if not zero
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.
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aout1, aout2, aout3, aout4 -- output audio signals.
ain1, ain2, ain3, ain4 -- input audio signals.
adl -- delay time in seconds
imd -- max. delay time (seconds)
iws -- interpolation window size (see below)
ist (optional) -- skip initialization if not zero
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.
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aout1, aout2 -- output audio signals
ain1, ain2 -- input audio signals
adl -- delay time in seconds
imd -- max. delay time (seconds)
iws -- interpolation window size (see below)
ist -- skip initialization if not zero
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.
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aout -- output audio signal
ain -- input audio signal
adl -- delay time in seconds
imd -- max. delay time (seconds)
iws -- interpolation window size (see below)
ist -- skip initialization if not zero
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.
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ain1, ain2, ain3, ain4 -- input audio signals
aout1, aout2, aout3, aout4 -- output audio signals
adl -- delay time in seconds
imd -- max. delay time (seconds)
iws -- interpolation window size (see below)
ist -- skip initialization if not zero
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.
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ain1, ain2 -- input audio signals
aout1, aout2 -- output audio signals
adl -- delay time in seconds
imd -- max. delay time (seconds)
iws -- interpolation window size (see below)
ist -- skip initialization if not zero
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. vdelayx) allow delaying 2 or 4 variables at once (stereo or quad signals); this is much more efficient than using separate opcodes for each channel.
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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
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 opcode to copy it in another table.
This opcode works at k-rate.
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.
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.
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.
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.
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.
Here is an example of the veloc opcode. It uses the files veloc.orc and veloc.sco.
Example 406. Example of the veloc opcode.
/* veloc.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 i1 veloc print i1 endin /* veloc.orc */
/* veloc.sco */ ; Play Instrument #1 for 12 seconds. i 1 0 12 e /* veloc.sco */
vexp — Performs power-of operations between a vectorial control signal and a scalar control signal
Performs power-of operations between a vectorial control signal and a scalar control signal
ifn - number of the table hosting the vectorial signal to be processed
ielements - number of elements of the vector
kval - scalar operand to be processed
vexp rises kval to each element contained in the table ifn.
These opcodes (vadd, vmult, vpow, 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.
In all these opcodes, the resulting vectors are stored in ifn, overriding the intial vectors. If you want to keep initial vector, use vcopy opcode 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.
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.
dur2, idur3, etc. - duration in seconds of subsequent segments.
ielements - number of elements of vectors.
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.
Here is an example of the vexpseg opcode. It uses the files vexpseg.csd.
Example 407. Example of the vexpseg opcode.
<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>
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
vexpv divides two vectorial control signals, that is, each element of ifn2 is risen 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 opcode to copy it in another table.
This opcode works at k-rate.
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.
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 Perry Cook, but re-coded for Csound.
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%.
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
Here is an example of the vibes opcode. It uses the files vibes.orc, vibes.sco, and marmstk1.wav.
Example 408. Example of the vibes opcode.
/* vibes.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; kamp = 20000 ; kfreq = 440 ; ihrd = 0.5 ; ipos = 0.561 ; imp = 1 ; kvibf = 6.0 ; kvamp = 0.05 ; ivibfn = 2 ; idec = 0.1 a1 vibes 20000, 440, 0.5, 0.561, 1, 6.0, 0.05, 2, 0.1 out a1 endin /* vibes.orc */
/* vibes.sco */ ; 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 e /* vibes.sco */
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.
Here is an example of the vibr opcode. It uses the files vibr.orc and vibr.sco.
Example 409. Example of the vibr opcode.
/* vibr.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create a vibrato waveform. kaverageamp init 7500 kaveragefreq init 5 ifn = 1 kvamp vibr kaverageamp, kaveragefreq, ifn ; Generate a tone including the vibrato. a1 oscili 10000+kvamp, 440, 2 out a1 endin /* vibr.orc */
/* vibr.sco */ ; Table #1, a sine wave for the vibrato. f 1 0 256 10 1 ; Table #1, a sine wave for the oscillator. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* vibr.sco */
kout vibrato kAverageAmp, kAverageFreq, kRandAmountAmp, kRandAmountFreq, kAmpMinRate, kAmpMaxRate, kcpsMinRate, kcpsMaxRate, ifn [, iphs]
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.
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.
Here is an example of the vibrato opcode. It uses the files vibrato.orc and vibrato.sco.
Example 410. Example of the vibrato opcode.
/* vibrato.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create a vibrato waveform. kaverageamp init 2500 kaveragefreq init 6 krandamountamp init 0.3 krandamountfreq init 0.5 kampminrate init 3 kampmaxrate init 5 kcpsminrate init 3 kcpsmaxrate init 5 ifn = 1 kvamp vibrato kaverageamp, kaveragefreq, krandamountamp, \ krandamountfreq, kampminrate, kampmaxrate, \ kcpsminrate, kcpsmaxrate, ifn ; Generate a tone including the vibrato. a1 oscili 10000+kvamp, 440, 2 out a1 endin /* vibrato.orc */
/* vibrato.sco */ ; Table #1, a sine wave for the vibrato. f 1 0 256 10 1 ; Table #1, a sine wave for the oscillator. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* vibrato.sco */
asig -- audio 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 variable can be used for output signal by means of fout opcode. After the disk writing operation, the accumulator variable should be set to zero by means of clear opcode (or it will explode).
ifn - number of the table hosting the vector to be processed
ielements - number of elements of the vector
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
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.
dur2, idur3, etc. - duration in seconds of subsequent segments.
ielements - number of elements of vectors.
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.
Here is an example of the vlinseg opcode. It uses the files vlinseg.csd.
Example 411. Example of the vlinseg opcode.
<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>
vlowres — A bank of filters in which the cutoff frequency can be separated under user control.
asig -- input signal
kfco -- frequency cutoff (not in Hz)
ksep -- frequency cutoff separation for each filter
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.
Here is an example of the vlowres opcode. It uses the files vlowres.orc, vlowres.sco, and beats.wav.
Example 412. Example of the vlowres opcode.
/* vlowres.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Use a nice sawtooth waveform. asig vco 32000, 220, 1 ; Vary the cutoff frequency from 30 to 300 Hz. kfco line 30, p3, 300 kres = 25 iord = 2 ksep = 20 ; Apply the filters. avlr vlowres asig, kfco, kres, iord, ksep ; It gets loud, so clip the output amplitude to 30,000. a1 clip avlr, 1, 30000 out a1 endin /* vlowres.orc */
/* vlowres.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* vlowres.sco */
Maps elements from a vectorial control signal onto itself according to the indeces of a second vectorial control signal
ifn - number of the table where the vectorial signal will be copied
ifn - number of the table hosting the vectorial signal to be copied
ielements - number of elements of the vector
vmap maps elements of ifn1 according to the values of table ifn2. 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 crash due to an illegal memory access error will occurr. Elements of ifn1 are treated as integers, so any fractional part will be truncated.
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
ifn - number of the table hosting the vector to be processed
ielements - number of elements of the vector
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
ifn - number of the table hosting the vectorial signal to be processed
ielements - number of elements of the vector
kval - scalar operand to be processed
vmult multiplies each elements of the vector contained in the table ifn by kval operand.
These opcodes (vadd, vmult, vpow, 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.
In all these opcodes, the resulting vectors are stored in ifn, overriding the intial vectors. If you want to keep initial vector, use vcopy opcode 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.
Here is an example of the vmult opcode. It uses the files vmult.csd.
Example 413. Example of the vmult opcode.
<CsoundSynthesizer> <CsOptions> -odac -B441 -b441 </CsOptions> <CsInstruments> sr=44100 kr=4410 ksmps=10 nchnls=2 instr 1 ;table playback ar lposcil 1, 1, 0, 262144, 1 out 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.aiff" 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>
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
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 opcode to copy it in another table.
This opcode works at k-rate.
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.
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/.
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
Here is an example of the voice opcode. It uses the files voice.orc, voice.sco, and impuls20.aiff.
Example 414. Example of the voice opcode.
/* voice.orc */ ; Initialize the global variables. sr = 22050 kr = 2205 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 3 kfreq = 0.8 kphoneme = 6 kform = 0.488 kvibf = 0.04 kvamp = 1 ifn = 1 ivfn = 2 av voice kamp, kfreq, kphoneme, kform, kvibf, kvamp, ifn, ivfn ; It tends to get loud, so clip voice's amplitude at 30,000. a1 clip av, 2, 30000 out a1 endin /* voice.orc */
/* voice.sco */ ; Table #1, an audio file for the carrier waveform. f 1 0 256 1 "impuls20.aiff" 0 0 0 ; Table #2, a sine wave for the vibrato waveform. f 2 0 256 10 1 ; Play Instrument #1 for a half-second. i 1 0 0.5 e /* voice.sco */
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 intial portamento values.
vport is similar to port, but operates with vectorial signals, istead of with scalar signals. Each vector element is treated as an indipendent control signal. Input vector 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.
vpow — Performs power-of operations between a vectorial control signal and a scalar control signal
Performs power-of operations between a vectorial control signal and a scalar control signal
ifn - number of the table hosting the vectorial signal to be processed
ielements - number of elements of the vector
kval - scalar operand to be processed
vpow elevates each element of the vector contained in the table ifn to the power of kval.
These opcodes (vadd, vmult, vpow, 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.
In all these opcodes, the resulting vectors are stored in ifn, overriding the intial vectors. If you want to keep initial vector, use vcopy opcode 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.
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
vpowv divides two vectorial control signals, that is, each element of ifn1 is risen 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 opcode to copy it in another table.
This opcode works at k-rate.
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.
vpvoc — Implements signal reconstruction using an fft-based phase vocoder and an extra envelope.
Implements signal reconstruction using an fft-based phase vocoder and an extra envelope.
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)
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.
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*.5, 2, p3*.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.
ifn - number of the table where the vectorial signal will be generated
ielements - number of elements of the vector
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 with vectors instead of with scalar values..
The output is a vector contained in ifn (that must be previously allocated).
All these operators are designed to be used together with other opocdes that operate with vector such as bmscan, adsynt etc.
Note: bmscan not yet available on Canonical Csound
ifn - number of the table where the vectorial signal will be generated
ielements - number of elements of the vector
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 with vectors instead of with scalar values..
The output is a vector contained in ifn (that must be previously allocated).
All these operators are designed to be used together with other opocdes that operate with vector such as bmscan, adsynt etc.
Note: bmscan not yet available on Canonical Csound
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.
instance - the number which identifies the plugin, to be passed to other vst4cs opcodes.
instance - the number which identifies the plugin, to be passed to other vst4cs opcodes.
ipath - the full pathname of the parameter bank (.fxb file).
vstedit opens the custom GUI editor widow for a VST plugin. Note that not all VST plugins have custom GUI editors.
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.
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.
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 paramters dynamically.
instance - the number which identifies the plugin, to be passed to other vst4cs opcodes.
instance - the number which identifies the plugin, to be passed to other vst4cs opcodes.
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.
Example 418.
/* 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 i4 1 1 57 32 i4 3 1 60 100 i4 5 1 62 100 i4 7 1 64 100 i4 9 1 65 100 i4 11 1 67 100 i4 13 1 69 100 i4 15 3 71 100 i4 18 3 72 100 e
instance - the number which identifies the plugin, to be passed to other vst4cs opcodes.
kchan - The midi channel to trasnmit the note on.
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.
Example 419.
/* orc */ sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 giHandle5 vstinit "c:/vstplugins/cheeze/cheeze machine.dll",1 instr 3 ain1 = 0 ga1, ga2 vstplugg gihandle5, ain1, ain1 endin instr 4 vstnote giHandle5, 1, p4, p5, p3 endin instr 10 outs ga1, ga2 endin
/* sco */ i 3 0 21 i 10 0 21 i4 1 3 57 55 i4 3 3 60 100 i4 5 3 62 100 i4 7 3 64 100 i4 9 2 65 100 i4 11 1 67 100 i4 13 1 69 100 i4 15 3 71 100 i4 18 3 72 100
vstparamset and vstparamget are used for parameter comunication to and from a VST plugin.
instance - the number which identifies the plugin, to be passed to other vst4cs opcodes.
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.
Example 420.
/* 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 kline line 0,p3,1 vstparamset gihandle1, 3, kline endin
/* sco */ i 3 0 21 i4 1 1 57 32 i4 3 1 60 100 i4 5 1 62 100 i4 7 1 64 100 i4 9 1 65 100 i4 11 1 67 100 i4 13 1 69 100 i4 15 3 71 100 i4 18 3 72 100 e
instance - the number which identifies the plugin, to be passed to other vst4cs opcodes.
kprogram - the number of the program to set.
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
vsubv 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 opcode to copy it in another table.
This opcode works at k-rate.
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.
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.
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 frame is automatically determined by the number of outN or inargN arguments, and must remain fixed for all indexes of each table.
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 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).
Here is an example of the vtablei opcode. It uses the files vtablei.csd
Example 422. Example of the vtablei opcode.
<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>
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
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 frame is automatically determined by the number of outN or inargN arguments, and must remain fixed for all indexes of each table.
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 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).
Here is an example of the vtablek opcode. It uses the files vtablek.csd.
Example 423. Example of the vtablek opcode.
<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>
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).
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 frame is automatically determined by the number of outN or inargN arguments, and must remain fixed for all indexes of each table.
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 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).
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 - 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 frame is automatically determined by the number of outN or inargN arguments, and must remain fixed for all indexes of each table.
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.
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).
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 - 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 frame is automatically determined by the number of outN or inargN arguments, and must remain fixed for all indexes of each table.
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 vtable, in order to correct eventual out-of-range values.
Here is an example of the vtablewk opcode. It uses the files vtablewk.csd.
Example 424. Example of the vtablewk opcode.
<CsoundSynthesizer> <CsOptions> -odac -b441 -B441 </CsOptions> <CsInstruments> sr=44100 kr=4410 ksmps=10 nchnls=2 instr 1 vcopy ar random 0, 1 vtablewa ar out ar,ar endin </CsInstruments> <CsScore> f1 0 262144 -1 "beats.aiff" 0 4 0 f2 0 262144 2 0 i1 0 4 i2 3 1 s i1 0 4 i3 3 1 s i1 0 4 </CsScore> </CsoundSynthesizer>
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).
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 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 frame is automatically determined by the number of outN or inargN arguments, and must remain fixed for all indexes of each table.
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.
Here is an example of the vtablewa opcode. It uses the files vtablewa.csd.
Example 425. Example of the vtablek opcode.
<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.aiff" 0 4 0 f2 0 262144 2 0 f3 0 1024 10 1 i1 0 10 </CsScore> </CsoundSynthesizer>
indx - Index into f-table, either a positive number range matching the table length
ifn - table number
iout1...ioutN - 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 frame is automatically determined by the number of outN or inargN arguments, and must remain fixed for all indexes of each table.
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 vtabi 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).
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.
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 frame is automatically determined by the number of outN or inargN arguments, and must remain fixed for all indexes of each table.
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 vtabk 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).
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.
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 frame is automatically determined by the number of outN or inargN arguments, and must remain fixed for all indexes of each table.
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.
Notice that vtaba 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).
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.
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
inarg1...inargN - 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 frame is automatically determined by the number of outN or inargN arguments, and must remain fixed for all indexes of each table.
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.
kndx - Index into f-table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0). kinarg1...kinargN - input 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 frame is automatically determined by the number of outN or inargN arguments, and must remain fixed for all indexes of each table.
vtabwk 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 vtabwk, in order to correct eventual out-of-range values.
andx - Index into f-table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0). ainarg1...ainargN - input 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 frame is automatically determined by the number of outN or inargN arguments, and must remain fixed for all indexes of each table.
vtabwa 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 vtabwa, in order to correct eventual out-of-range values.
ifn - number of the table hosting the vector to be processed
ielements - number of elements of the vector
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
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).
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.
Here is an example of the waveset opcode. It uses the files waveset.orc, waveset.sco, and beats.wav.
Example 426. Example of the waveset opcode.
/* waveset.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1 - play an audio file. instr 1 asig soundin "beats.wav" out asig endin ; Instrument #2 - stretch the audio file with waveset. instr 2 asig soundin "beats.wav" a1 waveset asig, 2 out a1 endin /* waveset.orc */
/* waveset.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 ; Play Instrument #2 for four seconds. i 2 3 4 e /* waveset.sco */
Weibull distribution random number generator (positive values only). This is an x-class noise generator
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.
Here is an example of the weibull opcode. It uses the files weibull.orc and weibull.sco.
Example 427. Example of the weibull opcode.
/* weibull.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Generate a random number in a Weibull distribution. ; ksigma = 1 ; ktau = 1 i1 weibull 1, 1 print i1 endin /* weibull.orc */
/* weibull.sco */ ; Play Instrument #1 for one second. i 1 0 1 e /* weibull.sco */
Its output should include lines like this:
instr 1: i1 = 1.834
Audio output is a tone similar to a bowed string, using a physical model developed from Perry Cook, but re-coded for Csound.
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.
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
Here is an example of the wgbow opcode. It uses the files wgbow.orc and wgbow.sco.
Example 428. Example of the wgbow opcode.
/* wgbow.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 31129.60 kfreq = 440 kpres = 3.0 krat = 0.127236 kvibf = 6.12723 ifn = 1 ; Create an amplitude envelope for the vibrato. kv linseg 0, 0.5, 0, 1, 1, p3-0.5, 1 kvamp = kv * 0.01 a1 wgbow kamp, kfreq, kpres, krat, kvibf, kvamp, ifn out a1 endin /* wgbow.orc */
/* wgbow.sco */ ; Table #1, a sine wave. f 1 0 128 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* wgbow.sco */
A physical model of a bowed bar, belonging to the Perry Cook family of waveguide instruments.
iconst (optional, default=0) -- an integration constant. Default is zero.
itvel (optional, default=0) -- either 0 or 1. When ktvel = 0, the bow velocity follows an ADSR style trajectory. When ktvel = 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
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.
Here is an example of the wgbowedbar opcode. It uses the files wgbowedbar.orc and wgbowedbar.sco.
Example 429. Example of the wgbowedbar opcode.
/* wgbowedbar.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 ; pos = [0, 1] ; bowpress = [1, 10] ; gain = [0.8, 1] ; intr = [0,1] ; trackvel = [0, 1] ; bowpos = [0, 1] kb line 0.5, p3, 0.1 kp line 0.6, p3, 0.7 kc line 1, p3, 1 a1 wgbowedbar p4, cpspch(p5), kb, kp, 0.995, p6, 0 out a1 endin /* wgbowedbar.orc */
/* wgbowedbar.sco */ i1 0 3 32000 7.00 0 e /* wgbowedbar.sco */
Audio output is a tone related to a brass instrument, using a physical model developed from Perry Cook, but re-coded for Csound.
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.
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
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This is rather poor, and at present uncontrolled. Needs revision, and possibly more parameters. |
Here is an example of the wgbrass opcode. It uses the files wgbrass.orc and wgbrass.sco.
Example 430. Example of the wgbrass opcode.
/* wgbrass.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 31129.60 kfreq = 440 ktens = 0.4 iatt = 0.1 kvibf = 6.137 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 /* wgbrass.orc */
/* wgbrass.sco */ ; Table #1, a sine wave. f 1 0 128 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* wgbrass.sco */
Audio output is a tone similar to a clarinet, using a physical model developed from Perry Cook, but re-coded for Csound.
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.
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
Here is an example of the wgclar opcode. It uses the files wgclar.orc and wgclar.sco.
Example 431. Example of the wgclar opcode.
/* wgclar.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp init 31129.60 kfreq = 440 kstiff = -0.3 iatt = 0.1 idetk = 0.1 kngain = 0.2 kvibf = 5.735 kvamp = 0.1 ifn = 1 a1 wgclar kamp, kfreq, kstiff, iatt, idetk, kngain, kvibf, kvamp, ifn out a1 endin /* wgclar.orc */
/* wgclar.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for one second. i 1 0 1 e /* wgclar.sco */
Audio output is a tone similar to a flute, using a physical model developed from Perry Cook, but re-coded for Csound.
ares wgflute kamp, kfreq, kjet, iatt, idetk, kngain, kvibf, kvamp, ifn [, iminfreq] [, ijetrf] [, iendrf]
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.
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
Here is an example of the wgflute opcode. It uses the files wgflute.orc and wgflute.sco.
Example 432. Example of the wgflute opcode.
/* wgflute.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 31129.60 kfreq = 440 kjet = 0.32 iatt = 0.1 idetk = 0.1 kngain = 0.15 kvibf = 5.925 kvamp = 0.05 ifn = 1 a1 wgflute kamp, kfreq, kjet, iatt, idetk, kngain, kvibf, kvamp, ifn out a1 endin /* wgflute.orc */
/* wgflute.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* wgflute.sco */
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 idamp1, 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.
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.
The following example produces a moderately long note with rapidly decaying upper partials. It uses the files wgpluck.orc and wgpluck.sco.
Example 433. An example of the wgpluck opcode.
/* wgpluck.orc */ ; 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 /* wgpluck.orc */
/* wgpluck.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* wgpluck.sco */
The following example produces a shorter, brighter note. It uses the files wgpluck_brighter.orc and wgpluck_brighter.sco.
Example 434. An example of the wgpluck opcode with a shorter, brighter note.
/* wgpluck_brighter.orc */ ; 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 /* wgpluck_brighter.orc */
/* wgpluck_brighter.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* wgpluck_brighter.sco */
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.
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.
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).
Here is an example of the wgpluck2 opcode. It uses the files wgpluck2.orc and wgpluck2.sco.
Example 435. Example of the wgpluck2 opcode.
/* wgpluck2.orc */ ; 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 /* wgpluck2.orc */
/* wgpluck2.sco */ ; Play Instrument #1 for two seconds. i 1 0 2 e /* wgpluck2.sco */
wguide1 — A simple waveguide model consisting of one delay-line and one first-order lowpass filter.
A simple waveguide model consisting of one delay-line and one first-order lowpass filter.
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.
Here is an example of the wguide1 opcode. It uses the files wguide1.orc and wguide1.sco.
Example 436. Example of the wguide1 opcode.
/* wguide1.orc */ ; 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 /* wguide1.orc */
/* wguide1.sco */ ; Play Instrument #1 for 2 seconds. i 1 0 2 ; Play Instrument #2 for 2 seconds. i 2 2 2 e /* wguide1.sco */
wguide2 — A model of beaten plate consisting of two parallel delay-lines and two first-order lowpass filters.
A model of beaten plate consisting of two parallel delay-lines and two first-order lowpass filters.
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.
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.
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.
Here is an example of the wterrain opcode. It uses the files wterrain.orc and wterrain.sco.
Example 437. Example of the wterrain opcode.
/* wterrain.orc */ ; 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 /* wterrain.orc */
/* wterrain.sco */ 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 /* wterrain.sco */
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
The envelope is the range 0 to 1 and may need to be scaled further. The envelope may be described as:
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 xadsr is identical to adsr except it uses exponential, rather than linear, line segments.
xadsr is new in Csound version 3.51.
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.
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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]
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.
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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]
iscan -- which scan process to read
iwhich (optional) -- which node to sense. The default is 0.
iscan -- which scan process to read
iwhich (optional) -- which node to sense. The default is 0.
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.
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.
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
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>
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.
xscanu init, irate, ifnvel, ifnmass, ifnstif, ifncentr, ifndamp, kmass, kstif, kcentr, kdamp, ileft, iright, kpos, kstrngth, ain, idisp, id
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.
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.
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>
Extend the duration of real-time generated events and handle their extra life (see also linenr).
xtratim extends current MIDI-activated note duration of iextradur seconds after the corresponding noteoff message has deactivated current note itself. This opcode has no output arguments.
This opcode is useful for implementing complex release-oriented envelopes.
instr 1 ;allows complex ADSR envelope with MIDI events 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 .5) 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, 1 out a1 endin
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.
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.
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.
Here is an example of the xyin opcode. It uses the files xyin.orc and xyin.sco.
Example 438. Example of the xyin opcode.
/* xyin.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; 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 oscil kamp, kcps, 1 out a1 endin /* xyin.orc */
/* xyin.sco */ ; Table #1, a sine wave. f 1 0 16384 10 1 ; Play Instrument #1 for 30 seconds. i 1 0 30 e /* xyin.sco */
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
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.
Here is an example of the zacl opcode. It uses the files zacl.orc and zacl.sco.
Example 439. Example of the zacl opcode.
/* zacl.orc */ ; 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 /* zacl.orc */
/* zacl.sco */ ; 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 /* zacl.sco */
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.
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.
Here is an example of the zakinit opcode. It uses the files zakinit.orc and zakinit.sco.
Example 440. Example of the zakinit opcode.
/* zakinit.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Initialize the ZAK space. ; Create 3 a-rate variables and 5 k-rate variables. zakinit 3, 5 ; 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, 3 endin /* zakinit.orc */
/* zakinit.sco */ ; 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 /* zakinit.sco */
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.
Here is an example of the zamod opcode. It uses the files zamod.orc and zamod.sco.
Example 441. Example of the zamod opcode.
/* zamod.orc */ ; 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 /* zamod.orc */
/* zamod.sco */ ; 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 /* zamod.sco */
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.
Here is an example of the zar opcode. It uses the files zar.orc and zar.sco.
Example 442. Example of the zar opcode.
/* zar.orc */ ; 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 /* zar.orc */
/* zar.sco */ ; 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 /* zar.sco */
kndx -- points to the za location to be read.
kgain -- multiplier for the a-rate signal.
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.
Here is an example of the zarg opcode. It uses the files zarg.orc and zarg.sco.
Example 443. Example of the zarg opcode.
/* zarg.orc */ ; 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 /* zarg.orc */
/* zarg.sco */ ; 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 /* zarg.sco */
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.
Here is an example of the zaw opcode. It uses the files zaw.orc and zaw.sco.
Example 444. Example of the zaw opcode.
/* zaw.orc */ ; 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 /* zaw.orc */
/* zaw.sco */ ; 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 /* zaw.sco */
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.
Here is an example of the zawm opcode. It uses the files zawm.orc and zawm.sco.
Example 445. Example of the zawm opcode.
/* zawm.orc */ ; 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 /* zawm.orc */
/* zawm.sco */ ; 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 /* zawm.sco */
zfilter2 — Performs filtering using a transposed form-II digital filter lattice with radial pole-shearing and angular pole-warping.
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
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.
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.
Here is an example of the zir opcode. It uses the files zir.orc and zir.sco.
Example 446. Example of the zir opcode.
/* zir.orc */ ; 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 /* zir.orc */
/* zir.sco */ ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e /* zir.sco */
isig -- initializes the value of the zk location.
indx -- points to the zk or za location to which to write.
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.
Here is an example of the ziw opcode. It uses the files ziw.orc and ziw.sco.
Example 447. Example of the ziw opcode.
/* ziw.orc */ ; 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 /* ziw.orc */
/* ziw.sco */ ; Play Instrument #1 for one second. i 1 0 1 ; Play Instrument #2 for one second. i 2 0 1 e /* ziw.sco */
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.
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.
Here is an example of the ziwm opcode. It uses the files ziwm.orc and ziwm.sco.
Example 448. Example of the ziwm opcode.
/* ziwm.orc */ ; 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 /* ziwm.orc */
/* ziwm.sco */ ; 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 /* ziwm.sco */
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.
Here is an example of the zkcl opcode. It uses the files zkcl.orc and zkcl.sco.
Example 449. Example of the zkcl opcode.
/* zkcl.orc */ ; 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 /* zkcl.orc */
/* zkcl.sco */ ; 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 /* zkcl.sco */
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.
Here is an example of the zkmod opcode. It uses the files zkmod.orc and zkmod.sco.
Example 450. Example of the zkmod opcode.
/* zkmod.orc */ ; 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 /* zkmod.orc */
/* zkmod.sco */ ; 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 /* zkmod.sco */
Here is an example of the zkr opcode. It uses the files zkr.orc and zkr.sco.
Example 451. Example of the zkr opcode.
/* zkr.orc */ ; 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 /* zkr.orc */
/* zkr.sco */ ; 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 /* zkr.sco */
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.
Here is an example of the zkw opcode. It uses the files zkw.orc and zkw.sco.
Example 452. Example of the zkw opcode.
/* zkw.orc */ ; 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 /* zkw.orc */
/* zkw.sco */ ; 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 /* zkw.sco */
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.
Here is an example of the zkwm opcode. It uses the files zkwm.orc and zkwm.sco.
Example 453. Example of the zkwm opcode.
/* zkwm.orc */ ; 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 /* zkwm.orc */
/* zkwm.sco */ ; 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 /* zkwm.sco */
Audio output is a tone similar to a stuck 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.
iK -- dimensionless siffness 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.
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.
Here is an example of the barmodel opcode. It uses the files barmodel.orc and barmodel.sco.
Example 454. Example of the barmodel opcode.
/* barmodel.orc */ ; 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 /* barmodel.orc */
/* barmodel.sco */ 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 */
This causes score time to be advanced by a specified amount without producing sound samples.
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 |
.
.
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.
p1 -- Specifies how the clock is to be set.
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.
e statement — This statement may be used to mark the end of the last section of the score.
All pfields are ignored.
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.
f Statement (or Function Table Statement) — Causes a GEN subroutine to place values in a stored function table.
This causes a GEN subroutine to place values in a stored function table for use by instruments.
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 (see below). Maximum table size is 16777216 (2**24) points.
p4 -- Number of the GEN routine to be called (see GEN ROUTINES). A negative value will cause rescaling to be omitted.
p5
p6 ... -- Parameters whose meaning is determined by the particular GEN routine.
Function tables are arrays of floating-point values. Arrays can be of any length in powers 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.
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.
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).
i — Makes an instrument active at a specific time and for a certain duration.
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.
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 zero value will invoke an initialization pass without performance (see also instr).
p4 ... -- Parameters whose significance is determined by the instrument.
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 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
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.
Here is an instrument which can find out whether it is tied to a previous note (tival returns 1), and whether it is held (negative p3). Attack and release are handled accordingly:
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 out ar tieskip: ; Skip some initialization on tied note endin
A simple score using three instances of the above instrument:
f1 0 8192 10 1 ; Sine i10.1 0 -1 7.00 10000 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
p1 -- Instrument number to mute/unmute.
p2 -- Action time of function generation (or destruction) 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.
p1 -- Number of times to repeat the section.
p2 -- Macro(name) to advance with each repetition (optional).
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.
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Because of serious problems of interaction with macro expansion, sections must start and end in the same file, and not in a macro. |
Here is an example of the r statement. It uses the files r.orc and r.sco.
Example 1. Example of the r statement.
/* r.orc */ ; 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 /* r.orc */
/* r.sco */ ; 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 /* r.sco */
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.
A section ending automatically invokes a Purge of inactive instrument and data spaces.
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This statement sets the tempo and specifies the accelerations and decelerations for the current section. This is done by converting beats into seconds.
p1 -- Must be zero.
p2 -- Initial tempo on beats per minute.
p3, p5, p7,... -- Times in beats per minute (in non-decreasing order).
p4, p6, p8,... -- Tempi for the referenced beat times.
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.
The v statement takes effect with the following i statement, and remains in effect until the next v statement, s statement, or e statement.
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 ;note1 v2 i1 1 . ;note2 i1 2 . ;note3 v1 i1 3 . ;note4 i1 4 . ;note5 e
In this example, note2 and note4 occur simultaneously, while note3 actually occurs before note2, that is, at its original place. Durations are unaffected.
i1 0 1 v2 i. + . i. . .
In this example, the v statement has no effect.
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.
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Here is a simple example of the GEN01 routine. It uses the files gen01.orc, gen01.sco, and beats.wav. It uses the audio file “beats.wav”, here is its diagram:
Diagram of the waveform generated by GEN01.
Example 2. A simple example of the GEN01 routine.
/* gen01.orc */ ; 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 /* gen01.orc */
/* gen01.sco */ ; Table #1: read an audio file (using GEN01). f 1 0 131072 1 "beats.wav" 0 4 0 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* gen01.sco */
Here is another example of the GEN01 routine. Csound will automatically compute the tablesize because we have set it to 0. This example uses the files gen01computed.orc, gen01computed.sco, and beats.wav.
Example 3. An example of the GEN01 routine with a computed tablesize.
/* gen01computed.orc */ ; 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 /* gen01computed.orc */
/* gen01computed.sco */ ; 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 /* gen01computed.sco */
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.
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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. |
Here is a simple example of the GEN02 routine. It uses the files gen02.orc and gen02.sco. It places 12 values plus an explicit wrap-around guard value into a table of size next-highest power of 2. Rescaling is inhibited. Here is its diagram:
Diagram of the waveform generated by GEN02.
Example 4. A simple example of the GEN02 routine.
/* gen02.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create an index over the length of our entire note. kcps init 1/p3 kndx phasor kcps ; Read Table #1 with our index. ifn = 1 ixmode = 1 kamp tablei kndx, ifn, ixmode ; Create a sine wave, use the Table #1 values to control ; the amplitude. This creates a sound with a long attack. a1 oscil kamp*30000, 440, 2 out a1 endin /* gen02.orc */
/* gen02.sco */ ; Table #1: an envelope with a long attack (using GEN02). f 1 0 16 2 0 1 2 3 4 5 6 7 8 9 10 11 0 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* gen02.sco */
This subroutine generates a stored function table by evaluating a polynomial in x over a fixed interval and with specified coefficients.
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.
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Here is a simple example of the GEN03 routine. It uses the files gen03.orc and gen03.sco. It fills a table with a 4th order polynomial function over the x-interval -1 to 1. The origin will be at the offset position 512. The function is post-normalized. Here is its diagram:
Diagram of the waveform generated by GEN03.
Example 5. A simple example of the GEN03 routine.
/* gen03.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create an index over the length of our entire note. kcps init 1/p3 kndx phasor kcps ; Read Table #1 with our index. ifn = 1 ixmode = 1 kamp table kndx, ifn, ixmode ; Create a sine wave, use the Table #1 values to control ; the amplitude. a1 oscil kamp*30000, 440, 2 out a1 endin /* gen03.orc */
/* gen03.sco */ ; Table #1: a polynomial function (using GEN03). f 1 0 1025 3 -1 1 5 4 3 2 2 1 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* gen03.sco */
This subroutine generates a normalizing function by examining the contents of an existing table.
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.
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f 2 0 512 4 1 1
This creates a normalizing function for use in connection with the GEN03 table 1 example. Midpoint bipolar offset is specified.
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 (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.
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Here is a simple example of the GEN05 routine. It uses the files gen05.orc and gen05.sco. It will create a nice percussive amplitude envelope. Here is its diagram:
Diagram of the waveform generated by GEN05.
Example 6. A simple example of the GEN05 routine.
/* gen05.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create an index over the length of our entire note. kcps init 1/p3 kndx phasor kcps ; Read Table #1 with our index. ifn = 1 ixmode = 1 kamp table kndx, ifn, ixmode ; Create a sine wave, use the Table #1 values to control ; the amplitude. This creates a nice percussive sound. a1 oscil kamp*30000, 440, 2 out a1 endin /* gen05.orc */
/* gen05.sco */ ; Table #1: a percussive envelope (using GEN05). f 1 0 64 5 1 2 120 60 1 1 0.001 1 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* gen05.sco */
This subroutine will generate a function comprised of segments of cubic polynomials, spanning specified points just three at a time.
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).
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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. |
Here is a simple example of the GEN06 routine. It uses the files gen06.orc and gen06.sco. It creates 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. Here is its diagram:
Diagram of the waveform generated by GEN06.
Example 7. A simple example of the GEN06 routine.
/* gen06.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create an index over the length of our entire note. kcps init 1/p3 kndx phasor kcps ; Read Table #1 with our index. ifn = 1 ixmode = 1 kval table kndx, ifn, ixmode ; Generate a sine waveform, use our Table #1 value to ; vary its frequency by 100 Hz from its base frequency. ibasefreq = 440 kfreq = kval * 100 a1 oscil 20000, ibasefreq + kfreq, 2 out a1 endin /* gen06.orc */
/* gen06.sco */ ; Table #1: a curve (using GEN06). f 1 0 65 6 0 16 0.5 16 1 16 0 16 -1 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* gen06.sco */
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.
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Here is a simple example of the GEN07 routine. It uses the files gen07.orc and gen07.sco. It will create a single-cycle sawtooth whose discontinuity is mid-way in the stored function. Here is its diagram:
Diagram of the waveform generated by GEN07.
Example 8. A simple example of the GEN07 routine.
/* gen07.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 kcps = 440 ifn = 1 ; Play the sine wave stored in Table #1. a1 oscil kamp, kcps, ifn out a1 endin /* gen07.orc */
/* gen07.sco */ ; Table #1: a sawtooth wave (using GEN07). f 1 0 256 7 0 128 1 0 -1 128 0 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* gen07.sco */
This subroutine will generate a piecewise cubic spline curve, the smoothest possible through all specified points.
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.
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Here is a simple example of the GEN08 routine. It uses the files gen08.orc and gen08.sco. It will create a curve with a smooth hump in the middle, going briefly negative outside the hump then flat at its ends. Here is its diagram:
Diagram of the waveform generated by GEN08.
Example 9. A simple example of the GEN08 routine.
/* gen08.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create an index over the length of our entire note. kcps init 1/p3 kndx phasor kcps ; Read Table #1 with our index. ifn = 1 ixmode = 1 kval table kndx, ifn, ixmode ; Generate a sine waveform, use our Table #1 value to ; vary its frequency by 100 Hz from its base frequency. ibasefreq = 440 kfreq = kval * 100 a1 oscil 20000, ibasefreq + kfreq, 2 out a1 endin /* gen08.orc */
/* gen08.sco */ ; Table #1: a curve with a smooth hump (using GEN08). f 1 0 65 8 0 16 0 16 1 16 0 16 0 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for two seconds. i 1 0 2 e /* gen08.sco */
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.
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).
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Here is a simple example of the GEN09 routine. It uses the files gen09.orc and gen09.sco. It will generate a cosine wave, a sine wave with an initial phase of 90 degrees. Here is its diagram:
Diagram of the waveform generated by GEN09.
Example 10. A simple example of the GEN09 routine.
/* gen09.orc */ ; 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 /* gen09.orc */
/* gen09.sco */ ; Table #1: a cosine wave (using GEN09). ; This is a sine wave with an initial phase of 90 degrees. f 1 0 16384 9 1 1 90 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* gen09.sco */
Here is another example of the GEN09 routine. It uses the files gen09square.orc and gen09square.sco. It combines partials l, 3 and 9 in the relative strengths in which they are found in a square wave, except that partial 9 is upside down. It will be rescaled, here is its diagram:
Diagram of the waveform generated by GEN09.
Example 11. A square wave generated by the GEN09 routine.
/* gen09square.orc */ ; 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 /* gen09square.orc */
/* gen09square.sco */ ; 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 /* gen09square.sco */
These subroutines generate composite waveforms made up of weighted sums of simple sinusoids. The specification of each contributing partial requires 1 pfield using GEN10.
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.
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Here is a simple example of the GEN10 routine. It uses the files gen10.orc and gen10.sco. It will generate a simple sine wave. Here is its diagram:
Diagram of the waveform generated by GEN10.
Example 12. A simple example of the GEN10 routine.
/* gen10.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 kcps = 440 ifn = 1 ; Play the sine wave stored in Table #1. a1 oscil kamp, kcps, ifn out a1 endin /* gen10.orc */
/* gen10.sco */ ; Table #1: a simple sine wave (using GEN10). f 1 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* gen10.sco */
This subroutine generates an additive set of cosine partials, in the manner of Csound generators buzz and gbuzz.
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.
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Here is a simple example of the GEN11 routine. It uses the files gen11.orc and gen11.sco. It will generate a simple cosine wave. Here is its diagram:
Diagram of the waveform generated by GEN11.
Example 13. A simple example of the GEN11 routine.
/* gen11.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 kamp = 30000 kcps = 440 ifn = 1 ; Play the cosine wave stored in Table #1. a1 oscil kamp, kcps, ifn out a1 endin /* gen11.orc */
/* gen11.sco */ ; Table #1: a simple cosine wave (using GEN11). f 1 0 16384 11 1 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* gen11.sco */
This generates the log of a modified Bessel function of the second kind, order 0, suitable for use in amplitude-modulated FM.
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.
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Here is a simple example of the GEN12 routine. It uses the files gen12.orc and gen12.sco. It generates the function ln(I0(x)) from 0 to 20. Here is its diagram:
Diagram of the waveform generated by GEN12.
Example 14. A simple example of the GEN12 routine.
/* gen12.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create an index over the length of our entire note. kcps init 1/p3 kndx phasor kcps ; Read Table #1 with our index. ifn = 1 ixmode = 1 kamp tablei kndx, ifn, ixmode ; Create a sine wave, use the Table #1 values to control ; the amplitude. This creates a sound with a long attack. a1 oscil kamp*30000, 440, 2 out a1 endin /* gen12.orc */
/* gen12.sco */ ; Table #1: a modified Bessel function (using GEN12). f 1 0 2049 12 20 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* gen12.sco */
GEN13 — Stores a polynomial whose coefficients derive from the Chebyshev polynomials of the first kind.
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.
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.
Here is a simple example of the GEN13 routine. It uses the files gen13.orc and gen13.sco. It creates a function which, under waveshaping, will split a sinusoid into 3 odd-harmonic partials of relative strength 5:3:1. Here is its diagram:
Diagram of the waveform generated by GEN13.
Example 15. A simple example of the GEN13 routine.
/* gen13.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create an index over the length of our entire note. kcps init 1/p3 kndx phasor kcps ; Read Table #1 with our index. ifn = 1 ixmode = 1 kval table kndx, ifn, ixmode ; Generate a sine waveform, use our Table #1 value to ; vary its frequency by 100 Hz from its base frequency. ibasefreq = 440 kfreq = kval * 100 a1 oscil 20000, ibasefreq + kfreq, 2 out a1 endin /* gen13.orc */
/* gen13.sco */ ; Table #1: a polynomial function (using GEN13). f 1 0 1025 13 1 1 0 5 0 3 0 1 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* gen13.sco */
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.
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.
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Here is a simple example of the GEN14 routine. It uses the files gen14.orc and gen14.sco. It creates a function which, under waveshaping, will split a sinusoid into 3 odd-harmonic partials of relative strength 5:3:1. Here is its diagram:
Diagram of the waveform generated by GEN14.
Example 16. A simple example of the GEN14 routine.
/* gen14.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create an index over the length of our entire note. kcps init 1/p3 kndx phasor kcps ; Read Table #1 with our index. ifn = 1 ixmode = 1 kval table kndx, ifn, ixmode ; Generate a sine waveform, use our Table #1 value to ; vary its frequency by 100 Hz from its base frequency. ibasefreq = 440 kfreq = kval * 100 a1 oscil 20000, ibasefreq + kfreq, 2 out a1 endin /* gen14.orc */
/* gen14.sco */ ; Table #1: a polynomial function (using GEN14). f 1 0 1025 14 1 1 0 5 0 3 0 1 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for 2 seconds. i 1 0 2 e /* gen14.sco */
This subroutine creates two tables of stored polynomial functions, suitable for use in phase quadrature operations.
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.
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GEN15 creates two tables of equal size, labeled f # and f # + 1. Table # will contain a Chebyshev function of the first kind, drawn using GEN03 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. |
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.
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
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If type > 0, there is a slowly rising, fast decaying (convex) curve, while if type < 0, the curve is fast rising, slowly decaying (concave). See also transeg. |
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.
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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). |
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.
Writes composite waveforms made up of pre-existing waveforms. Each contributing waveform requires 4 pfields and can overlap with other waveforms.
size -- number of points in the table. Must be a power-of-2 plus 1 (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.
f 1 0 4096 10 1 f 2 0 1025 18 1 1 0 512 1 1 513 1025
f2 consists of two copies of f1 written in to locations 0-512 and 513-1025.
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.
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).
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Here is a simple example of the GEN19 routine. It uses the files gen19.orc and gen19.sco. It will generate a nice bell curve, here is its diagram:
Diagram of the waveform generated by GEN19.
Example 17. A simple example of the GEN19 routine.
/* gen19.orc */ ; Initialize the global variables. sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 ; Instrument #1. instr 1 ; Create an index over the length of our entire note. kcps init 1/p3 kndx phasor kcps ; Read Table #1 with our index. ifn = 1 ixmode = 1 kval table kndx, ifn, ixmode ; Generate a sine waveform, use our Table #1 value to ; vary its frequency by 100 Hz from its base frequency. ibasefreq = 440 kfreq = kval * 100 a1 oscil 20000, ibasefreq + kfreq, 2 out a1 endin /* gen19.orc */
/* gen19.sco */ ; Table #1: a bell curve (using GEN19). f 1 0 16384 -19 1 1 260 1 ; Table #2, a sine wave. f 2 0 16384 10 1 ; Play Instrument #1 for 3 seconds. i 1 0 3 e /* gen19.sco */
This subroutine generates functions of different windows. These windows are usually used for spectrum analysis or for grain envelopes.
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 Kaiser window.
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.
For diagrams, see Window Functions
This generates tables of different random distributions. (See also betarand, bexprnd, cauchy, exprand, gauss, linrand, pcauchy, poisson, trirand, unirand, and weibull)
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.
f1 0 1024 21 1 ; Uniform (white noise) f1 0 1024 21 6 ; Gaussian 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.
"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)
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All characters following ';' (comment) are ignored until next line (numbers too). |
This subroutine reads numeric values from another allocated function-table and rescales them according to the max and min values given by the user.
#, 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.
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This GEN is useful, for example, to eliminate the starting offset in exponential segments allowing a real starting from zero. |
These subroutines are used to construct functions from segments of exponential curves in breakpoint fashion.
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.
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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. |
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.
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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. |
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.
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 soundfile 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.
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 left 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.
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).
This routine is similar to GEN09, but allows mixing any waveform specified in an existing table.
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).
This routine is similar to GEN31, but allows specifying source ftable for each partial. Tables can be resampled either with FFT, or linear interpolation.
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)
itmp ftgen 1, 0, 16384, 7, 1, 16384, -1 ; sawtooth itmp ftgen 2, 0, 8192, 10, 1 ; sine ; mix tables itmp ftgen 5, 0, 4096, -32, -2, 1.5, 1.0, 0.25, 1, 2, 0.5, 0, \ 1, 3, -0.25, 0.5 ; window itmp ftgen 6, 0, 16384, 20, 3, 1 ; 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 out a1 * 10000 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 out a1 * 2000 endin ---------- score: ---------- t 0 60 i 1 0 10 i 2 12 10 e
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).
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.
; partials 1, 4, 7, 10, 13, 16, etc. with base frequency of 400 Hz ibsfrq = 400 ; estimate number of partials inumh = int(1.5 + sr * 0.5 / (3 * ibsfrq)) ; source table length isrcln = int(0.5 + exp(log(2) * int(1.01 + log(inumh * 3) / log(2)))) ; create empty source table itmp ftgen 1, 0, isrcln, -2, 0 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 ; store output in ftable 2 (size = 262144) itmp ftgen 2, 0, 262144, -34, 1, inumh, 1, -1
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).
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.
; partials 1, 4, 7, 10, 13, 16, etc. with base frequency of 400 Hz ibsfrq = 400 ; estimate number of partials inumh = int(1.5 + sr * 0.5 / (3 * ibsfrq)) ; source table length isrcln = int(0.5 + exp(log(2) * int(1.01 + log(inumh * 3) / log(2)))) ; create empty source table itmp ftgen 1, 0, isrcln, -2, 0 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 ; store output in ftable 2 (size = 262144) itmp ftgen 2, 0, 262144, -34, 1, inumh, 1, -1
Generates a continuous random distribution function starting from the shape of a user-defined distribution histogram.
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).
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).
Generates a random distribution function of discrete ranges of values by giving a list of groups of three numbers.
The first number of each group is a the minimum value of the first 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. Even if any number can be assigned to the probability element of each group, it is suggested to give it a percent value, in order to make it clearer to the user.
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.
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.
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.
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if p4 is positive, the table will be post-normalised. A negative p4 will cause post-normalisation to be skipped. |
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).
GEN51 — This subroutine fills a table with a fully customized micro-tuning scale, in the manner of Csound opcodes cpstun, cpstuni and cpstmid.
This subroutine fills a table with a fully customized micro-tuning scale, in the manner of Csound opcodes cpstun, cpstuni and cpstmid.
f # time size -51 numgrades interval basefreq basekey tuningRatio1 tuningRatio2 .... tuningRationN
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 0 128 -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 0 128 -2 24 1.5 440 48 1 1.01 1.02 1.03 ..etc...>
GEN52 — Creates an interleaved multichannel table from the specified source tables, in the format expected by the "ftconv" opcode.
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.
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] [flags] [filenames]
utilname [flags] [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.
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.
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.
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).
As of Csound 4.08, hetro can write SDIF ouput files if the output file name ends with ".sdif". See the sdif2ad utility for more information about the Csound's SDIF support.
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.
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 ..........
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.
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.
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.
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.
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.
pvanal asound pvfile
will analyze the soundfile "asound" using the default frmsiz and windfact to produce the file "pvfile" suitable for use with pvoc.
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.
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.
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.
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.
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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. |
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.
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
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
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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. |
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)] ^ 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.
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.
enakis 259% ../csound -U pvlook test.pv Using csound.txt Csound Version 3.57 (Aug 3 1999) util PVLOOK: ; Bins in Analysis: 513 ; First Bin Shown: 1 ; Number of Bins Shown: 513 ; Frames in Analysis: 1184 ; First Frame Shown: 1 ; Number of Data Frames Shown: 1184 Bin 1 Freqs.0.000 87.891 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 0.000 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 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0.245 0.247 0.248 0.251 0.252 0.249 0.247 0.245 0.245 0.248 0.250 0.251 0.251 0.247 0.246 0.245 0.245 0.249 0.251 0.251 0.250 0.247 0.245 0.246 0.246 0.249 0.252 0.251 0.249 0.247 0.244 0.247 0.248 0.250 0.252 0.250 0.247 0.246 0.245 0.247 0.250 0.251 0.252 0.249 0.246 0.245 0.245 0.247 0.251 0.251 0.250 0.249 0.246 0.245 0.247 0.248 0.251 0.251 0.249 0.248 0.245 0.245 0.248 0.249 0.251 0.251 0.248 0.246 0.245 0.245 0.249 0.251 0.251 0.251 0.247 0.245 0.245 0.246 0.249 0.251 0.250 0.249 0.247 0.244 0.246 0.248 0.250 0.252 0.250 0.247 0.246 0.245 0.247 0.249 0.250 0.251 0.249 0.246 0.246 0.245 0.247 0.250 0.250 0.250 0.249 0.245 0.245 0.246 0.248 0.251 0.251 0.249 0.248 0.245 0.245 0.247 0.249 0.251 0.251 0.248 0.246 0.245 0.245 0.248 0.250 0.251 0.250 0.247 0.245 0.245 0.246 0.249 0.251 0.250 0.249 0.246 0.244 0.246 0.247 0.250 0.251 0.250 0.248 0.246 0.245 0.247 0.249 0.250 0.251 0.249 0.247 0.246 0.245 0.247 0.250 0.250 0.251 0.248 0.245 0.245 0.246 0.248 0.251 0.251 0.249 0.248 0.245 0.245 0.247 0.249 0.251 0.251 0.248 0.247 0.245 0.245 0.248 0.249 0.250 0.250 0.247 0.246 0.246 0.246 0.249 0.251 0.250 0.250 0.246 0.245 0.246 0.247 0.250 0.251 0.249 0.248 0.246 0.244 0.246 0.248 0.250 0.251 0.249 0.247 0.246 0.245 0.247 0.250 0.250 0.251 0.249 0.245 0.245 0.246 0.248 0.251 0.250 0.250 0.248 0.245 0.245 0.247 0.248 0.251 0.250 0.248 0.247 0.245 0.246 0.248 0.250 0.251 0.250 0.247 0.246 0.245 0.246 0.249 0.251 0.250 0.249 0.246 0.245 0.246 0.247 0.250 0.251 0.250 0.249 0.246 0.244 0.246 0.248 0.250 0.251 0.249 0.247 0.246 0.245 0.247 0.249 0.250 0.251 0.287 0.331 0.178 0.008 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 0.000 0.000 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3.292 3.297 3.292 3.293 3.294 3.289 3.292 3.294 3.291 3.296 3.293 3.291 3.294 3.291 3.292 3.296 3.292 3.294 3.295 3.289 3.292 3.292 3.291 3.296 3.294 3.292 3.295 3.290 3.290 3.293 3.291 3.295 3.296 3.291 3.294 3.291 3.289 3.294 3.292 3.293 3.295 3.291 3.292 3.293 3.290 3.294 3.295 3.292 3.294 3.291 3.289 3.293 3.291 3.293 3.296 3.292 3.293 3.293 3.288 3.292 3.293 3.292 3.296 3.293 3.291 3.294 3.289 3.292 3.295 3.291 3.294 3.293 3.289 3.292 3.291 3.290 3.295 3.293 3.292 3.294 3.289 3.291 3.293 3.290 3.295 3.294 3.290 3.293 3.290 3.289 3.294 3.291 3.293 3.295 3.290 3.292 3.292 3.289 3.293 3.293 3.292 3.295 3.291 3.289 3.292 3.290 3.292 3.295 3.291 3.293 3.292 3.288 3.292 3.291 3.291 3.295 3.291 3.291 3.292 3.289 3.291 3.294 3.291 3.294 3.292 3.289 3.292 3.290 3.290 3.295 3.292 3.293 3.294 3.289 3.291 3.292 3.290 3.294 3.293 3.291 3.293 3.289 3.290 3.293 3.291 3.294 3.295 3.290 3.292 3.291 3.289 3.294 3.293 3.292 3.294 3.290 3.290 3.292 3.289 3.293 3.294 3.291 3.293 3.291 3.289 3.292 3.291 3.291 3.295 3.291 3.291 3.292 3.288 3.292 3.293 3.291 3.295 3.292 3.290 3.292 3.289 3.291 3.294 3.291 3.293 3.292 3.288 3.291 3.291 3.290 3.295 3.292 3.291 3.293 3.289 3.290 3.292 3.290 3.294 3.293 3.290 3.292 3.290 3.289 3.293 3.291 3.292 3.294 3.290 3.290 3.291 3.289 3.293 3.293 3.291 3.293 3.290 3.288 3.291 3.290 3.292 3.294 3.290 3.292 3.291 3.288 3.291 3.291 3.291 3.294 3.291 3.290 3.291 3.288 3.291 3.293 3.291 3.293 3.292 3.288 3.291 3.290 3.290 3.294 3.291 3.291 3.292 3.288 3.290 3.291 3.290 3.294 3.293 3.290 3.292 3.289 3.289 3.293 3.290 3.292 3.293 3.289 3.291 3.290 3.289 3.293 3.292 3.291 3.293 3.289 3.289 3.291 3.289 3.292 3.293 3.290 3.292 3.290 3.288 3.292 3.291 3.291 3.294 3.290 3.290 3.291 3.288 3.291 3.292 3.291 3.293 3.291 3.288 3.291 3.289 3.290 3.293 3.290 3.292 3.292 3.288 3.291 3.291 3.290 3.293 3.291 3.290 3.292 3.288 3.289 3.292 3.290 3.292 3.293 3.289 3.291 3.289 3.288 3.293 3.291 3.291 3.292 3.288 3.289 3.290 3.288 3.292 3.293 3.290 3.292 3.289 3.288 3.291 3.290 3.291 3.293 3.289 3.290 3.290 3.287 3.291 3.291 3.290 3.293 3.290 3.288 3.290 3.288 3.290 3.293 3.291 3.292 3.291 3.288 3.290 3.289 3.289 3.293 3.290 3.290 3.291 3.287 3.289 3.291 3.289 3.292 3.291 3.288 3.290 3.288 3.288 3.292 3.290 3.291 3.292 3.288 3.289 3.290 3.288 3.292 3.292 3.290 3.292 3.289 3.288 3.291 3.289 3.291 3.293 3.289 3.291 3.290 3.287 3.291 3.290 3.290 3.293 3.289 3.289 3.290 3.287 3.290 3.292 3.290 3.292 3.290 3.287 3.290 3.289 3.289 3.292 3.290 3.290 3.291 3.287 3.289 3.290 3.289 3.292 3.291 3.289 3.291 3.288
etc...
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.
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
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.
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 = break file
-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.
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.
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)
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.
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.
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.
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.
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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. |
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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>.
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Quoted options that contain spaces are allowed. |
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
Cscore is a program for generating and manipulating numeric score files. It comprises a number of function subprograms, called into operation by a user-written control program, and can be invoked either as a standalone score preprocessor, or as part of the Csound run-time system:
Cscore [scorefilein] [scorefileout]
or
CSound [-C] [otherflags] [orchname] [scorename]
The available function programs augment the C language library functions; they can read either standard 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 also in C, and is compiled and linked to the function programs (or the entire Csound) 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.
An event in Cscore is equivalent to one statement of a standard numeric score or time-warped score (see any score.srt), stored internally in time-warped format. It is either created in-line, or read in from an existing score file (either format). Its main components are an opcode and an array of pfield values. It is stored somewhere in memory, organized by a structure that starts as follows:
typedef struct { CSHDR h; /* space-managing header */ long op; /* opcode-t, w, f, i, a, s or e */ long pcnt; /* number of pfields p1, p2, p3 ... */ long strlen; /* length of optional string argument */ char *strarg; /* address of optional string argument */ float p2orig; /* unwarped p2, p3 */ float p3orig; float offtim; /* storage used during performance */ float p[1]; /* array of pfields p0, p1, p2 ... */ } EVENT;
Any function subprogram 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 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 function subprograms enable scores to be created and manipulated in this way.
In the following summary of Cscore function calls, some simple naming conventions are used:
the symbols e, f are pointers to events (notes); the symbols a, b are pointers to lists (arrays) of such events; the letters ev at the end of a function name signify operation on an event; the letter l at the start of a function name signifies operation on a list. the symbol fp is a score input stream file pointer (FILE *); calling syntax description e = createv(n); create a blank event with n pfields int n; 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
The general format for a control program is:
#include "cscore.h" cscore() { /* VARIABLE DECLARATIONS */ /* PROGRAM BODY */ }
The include statement will define the event and list structures for the program. 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" cscore() { EVLIST *a; /* a is allowed to point to an event list */ a = lget(); /* read events in, return the list pointer */ lput(a); /* write these events out (unchanged) */ putstr("e"); /* write the string e to output */ }
After execution of lget(), 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 (lput) 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. 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" cscore() { EVENT *e; /* a pointer to an event */ EVLIST *a; a = lget(); /* 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 */ lput(a); /* write out the list of events */ putstr("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 a and e to examine each of the events in the list. Note that e has not preserved the numeral 4, but the contents of that slot. To inspect p5 of the previous listed event we need only redefine e with the assignment
e = a-e[3];
More generally, if we declare a new variable f to be a pointer to a pointer to an event, the statement
f = &a-e[4];
will set f to the address of the fourth event in the event list a, and *f will signify the contents of the slot, namely the event pointer itself. The expression
(*f)-p[5],
like e-p[5], signifies the fifth pfield of the selected event. However, we can advance to the next slot in the evlist by advancing the pointer f. In C this is denoted by f++.
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, 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.
By pointing the variable f to the first note-event and incrementing f inside a while block which iterates n times (the number of events in the list), one statement can be made to act upon the same pfield of each successive event.
#include "cscore.h" cscore() { EVENT *e,**f; /* declarations. see pp.8-9 in the */ EVLIST *a,*b; /* C language programming manual */ int n; a = lget(); /* read score into event list "a" */ b = lsepf(a); /* separate f statements */ lput(b); /* write f statements out to score */ lrelev(b); /* and release the spaces used */ e = defev("t 0 120"); /* define event for tempo statement */ putev(e); /* write tempo statement to score */ lput(a); /* write the notes */ putstr("s"); /* section end */ putev(e); /* write tempo statement again */ b = lcopyev(a); /* make a copy of the notes in "a" */ n = b-nevents; /* and get the number present */ f = &a-e[1]; while (n--) /* iterate the following line n times: */ (*f++)-p[5] *= .5; /* transpose pitch down one octave */ a = lcat(b,a); /* now add these notes to original pitches */ lput(a); putstr("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
Next we extend the above program by using the while statement 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" cscore() { EVENT *e,**f; EVLIST *a,*b; int n, dim; /* declare two integer variables */ a = lget(); b = lsepf(a); lput(b); lrelev(b); e = defev("t 0 120"); putev(e); lput(a); putstr("s"); putev(e); /* write out another tempo statement */ b = lcopyev(a); n = b-nevents; dim = 0; /* initialize dim to 0 */ f = &a-e[1]; while (n--){ (*f)-p[6] -= dim; /* subtract current value of dim */ (*f++)-p[5] *= .5; /* transpose, move f to next event */ dim += 2000; /* increase dim for each note */ } a = lcat(b,a); lput(a); putstr("e"); }
The increment of f in the above programs has depended on certain precedence rules of C. Although this keeps the code tight, the practice can be dangerous for beginners. Incrementing may alternately be written as a separate statement to make it more clear.
while (n--){ (*f)-p[6] -= dim; (*f)-p[5] *= .5; dim += 2000; f++; }
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 while block.
#include "cscore.h" int cue[3]={0,10,17}; /* declare an array of 3 integers */ cscore() { EVENT *e, **f; EVLIST *a, *b; int n, dim, cuecount, holdn; /* declare new variables */ a = lget(); b = lsepf(a); lput(b); lrelev(b); e = defev("t 0 120"); putev(e); n = a-nevents; holdn = n; /* hold the value of "n" to reset below */ cuecount = 0; /* initialize cuecount to "0" */ dim = 0; while (cuecount <= 2) { /* count 3 iterations of inner "while" */ f = &a-e[1]; /* reset pointer to first event of list "a" */ n = holdn; /* reset value of "n" to original note count */ while (n--) { (*f)-p[6] -= dim; (*f)-p[2] += cue[cuecount]; /* add values of cue */ f++; } printf("; diagnostic: cue = %d\n", cue[cuecount]); cuecount++; dim += 2000; lput(a); } putstr("e"); }
Here the inner while block looks at the events of list a (the notes) and the outer while block looks at each repetition of the events of list a (the pitch group repetitions). 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 identical input file, the C program above will generate:
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;
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.
./.htmlinclude "cscore.h" /* CSCORE_SWITCH.C */ cscore() /* callable from either CSound or standalone cscore */ { EVLIST *a, *b; FILE *fp1, *fp2; /* declare two scorefile stream pointers */ fp1 = getcurfp(); /* this is the command-line score */ fp2 = filopen("score2.srt"); /* this is an additional score file */ a = lget(); /* read section from score 1 */ lput(a); /* write it out as is */ putstr("s"); setcurfp(fp2); b = lget(); /* read section from score 2 */ lput(b); /* write it out as is */ putstr("s"); lrelev(a); /* optional to reclaim space */ lrelev(b); setcurfp(fp1); a = lget(); /* read next section from score 1 */ lput(a); /* write it out */ putstr("s"); setcurfp(fp2); b = lget(); /* read next sect from score 2 */ lput(b); / * write it out */ putstr("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.
./.htmlinclude "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() /* callable from either CSound or standalone cscore */ { EVLIST *a, *b; register 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; register float *p = pulse16; register 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 = lget(); /* read sect from tempo-warped score */ b = lseptwf(a); /* separate warp & fn statements */ lplay(b); /* and send these to performance */ a = lappstrev(a, "s"); /* append a sect statement to note list */ lplay(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]; } } lplay(a); /* now play modified list */ }
As stated above, 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.
A list of events can be conveyed to a Csound orchestra using lplay. There may be any number of lplay 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 lsort). If there is no lplay in a Cscore module run from within Csound, all events written out (via putev, putstr or lput) constitute a new score, which will be sent initially to scsort then to the Csound orchestra for performance. These can be examined in the files “cscore.out” and “cscore.srt”.
A standalone cscore program will normally use the put commands to write into its output file. If a standalone Cscore program contains lplay, the events thus intended for performance will instead be printed on the console.
A note list sent by lplay for performance should be temporally distinct from subsequent note lists. No note-end should extend past the next list's start time, since lplay 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 lgetnext() or lgetuntil()to fetch and process score segments prior to performance.
A Cscore program can be invoked either as a Standalone program or as part of Csound:
cscore -U pvanal scorename outfilename
or
csound -C [otherflags] orchname scorename
To create a standalone program, write a cscore.c program as shown above and test compile it with 'cc cscore.c'. If the compiler cannot find "cscore.h", try using -I/usr/local/include, or just copy the cscore.h module from the Csound source directory into your own. There will still be unresolved references, so you must now link your program with certain Csound I/O modules. If your Csound installation has created a libcscore.a, you can type
cc -o cscore.c -lcscore
Else set an environment variable to a Csound directory containing the already compiled modules, and invoke them explicitly:
setenv CSOUND /ti/u/bv/Csound cc -o cscore cscore.c $CSOUND/cscoremain.o $CSOUND/cscorefns.o \ $CSOUND/rdscore.o $CSOUND/memalloc.o
The resulting executable can be applied to an input scorefilein by typing:
cscore scorefilein scorefileout
To operate from CSound, first proceed as above then link your program to a complete set of Csound modules. If your Csound installation has created a libcsound.a, you can do this by typing
cc -o mycsound cscore.o -lcsounc -lX11 -lm (X11 if your installation included it)
Else copy *.c, *.h and Makefile from the Csound source directory, replace cscore.c by your own, then run “make 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”. With no lplay, the subsequent stages of processing can be seen in the files “cscore.out” and “cscore.srt”.
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. 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.
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
To access 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.
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.
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.
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.
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'''))
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
Table 3.
i | i-rate scalar |
k | k-rate scalar |
a | a-rate vector |
x | k-rate vector or a-rate vector |
f | f-rate streaming pvoc fsig type |
S | String |
B | |
l | |
m | Begins an indefinite list of i-rate arguments (any count) |
M | Begins an indefinite list of arguments (any rate, any count) |
n | Begins an indefinite list of i-rate arguments (any odd count) |
o | Optional i-rate, defaulting to 0 |
p | Optional i-rate, defaulting to 1 |
q | Optional i-rate, defaulting to 10 |
v | Optional i-rate, defaulting to 0.5 |
j | Optional i-rate, defaulting to -1 |
h | Optional i-rate, defaulting to 127 |
y | Begins an indefinite list of a-rate arguments (any count) |
z | Begins an indefinite list of k-rate arguments (any count) |
Z | Begins an indefinite list of alternating k-rate and a-rate arguments (kaka...) (any count) |
Table A.1. 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 |
Table C.1. alto “a”
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 |
Table C.2. alto “e”
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 |
Table C.3. alto “i”
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 |
Table C.4. alto “o”
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 |
Table C.5. alto “u”
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 |
Table C.6. bass “a”
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 |
Table C.7. bass “e”
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 |
Table C.8. bass “i”
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 |
Table C.9. bass “o”
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 |
Table C.10. bass “u”
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 |
Table C.11. countertenor “a”
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 |
Table C.12. countertenor “e”
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 |
Table C.13. countertenor “i”
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 |
Table C.14. countertenor “o”
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 |
Table C.15. countertenor “u”
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 |
Table C.16. soprano “a”
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 |
Table C.17. soprano “e”
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 |
Table C.18. soprano “i”
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 |
Table C.19. soprano “o”
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 |
Table C.20. soprano “u”
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 |
Table C.21. tenor “a”
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 |
Table C.22. tenor “e”
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 |
Table C.23. tenor “i”
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 |
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.
Hanning.
Hanning Window Function
Bartlett.
Bartlett Window Function
Blackman.
Blackman Window Function
Blackman-Harris.
Blackman-Harris Window Function
Gaussian.
Gaussian Window Function
Rectangle.
Note: Vertical scale is exaggerated in this diagram.
Rectangle Window Function
Sync.
Sync Window Function
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.
Csound64 is a version of Csound that uses 64-bit DOUBLE's internally to do processing versus regular Csound's 32-bit FLOAT's. 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, Csound64 is typicaly used after a work is finished for a final production run.
Notes On Using Csound64.
hetro files generated for Csound will work with Csound64.
PVOC-EX analysis and pvanal files generated for Csound will not work with Csound64. For Csound64, use of pvanal and pvoc opcodes are not currently supported. If your work file uses pvoc, use Csound. (This is a temporary issue relating to older file formats and is currently being addressed and worked on.)
lpanal files generated for Csound will not work with Csound64. For Csound64, use of lpanal and lpc opcodes are not currently supported. If your work file uses lpc, use Csound. (This is a temporary issue relating to older file formats and is currently being addressed and worked on.)
cvanal files generated for Csound will not work with Csound64. To generated cv files usable by Csound64, use the following command line:
csound64 -U cvanal
instead of either of the following:
csound -U cvanal
cvanal
This will generate a 64-bit cv file. If you were working with 32-bit Csound and using a 32-bit cv file, the cv file will not work with Csound64. When you switch to Csound64, you will need to use a 64-bit generated cv file.
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)
(a != b ? v1 : v2)
#define NAME # replacement text #
#define NAME(a' b' c') # replacement text #
#include “filename”
#undef NAME
$NAME
a % b (no rate restriction)
a && b (logical AND; not audio-rate)
(a > b ? v1 : v2)
(a >= b ? v1 : v2)
(a < b ? v1 : v2)
(a <= b ? v1 : v2)
a * b (no rate restriction)
+ a (no rate restriction)
− a (no rate restriction)
a / b (no rate restriction)
ar = xarg
ir = iarg
kr = karg
(a == b ? v1 : v2)
a ^ b (b not audio-rate)
a || b (logical OR; not audio-rate)
0dbfs = iarg
a(x) (control-rate args only)
abs(x) (no rate restriction)
ir active insnum
kr active kinsnum
ar adsr iatt, idec, islev, irel [, idel]
kr adsr iatt, idec, islev, irel [, idel]
ar adsyn kamod, kfmod, ksmod, ifilcod
ar adsynt kamp, kcps, iwfn, ifreqfn, iampfn, icnt [, iphs]
kaft aftouch [imin] [, imax]
ar alpass asig, krvt, ilpt [, iskip] [, insmps]
ampdbfs(x) (no rate restriction)
ampdb(x) (no rate restriction)
iamp ampmidi iscal [, ifn]
kr aresonk ksig, kcf, kbw [, iscl] [, iskip]
ar areson asig, kcf, kbw [, iscl] [, iskip]
kr atonek ksig, khp [, iskip]
ar atone asig, khp [, iskip]
ar atonex asig, khp [, inumlayer] [, iskip]
a1, a2 babo asig, ksrcx, ksrcy, ksrcz, irx, iry, irz [, idiff] [, ifno]
ar balance asig, acomp [, ihp] [, iskip]
ar bamboo kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] [, ifreq1] [, ifreq2]
a1 bbcutm asource, ibps, isubdiv, ibarlength, iphrasebars, inumrepeats [, istutterspeed] [, istutterchance] [, ienvchoice ]
a1,a2 bbcuts asource1, asource2, ibps, isubdiv, ibarlength, iphrasebars, inumrepeats [, istutterspeed] [, istutterchance] [, ienvchoice]
ar betarand krange, kalpha, kbeta
ir betarand krange, kalpha, kbeta
kr betarand krange, kalpha, kbeta
ar bexprnd krange
ir bexprnd krange
kr bexprnd krange
ar biquada asig, ab0, ab1, ab2, aa0, aa1, aa2 [, iskip]
ar biquad asig, kb0, kb1, kb2, ka0, ka1, ka2 [, iskip]
birnd(x) (init- or control-rate only)
ar bqrez asig, xfco, xres [, imode]
ar butbp asig, kfreq, kband [, iskip]
ar butbr asig, kfreq, kband [, iskip]
ar buthp asig, kfreq [, iskip]
ar butlp asig, kfreq [, iskip]
ar butterbp asig, kfreq, kband [, iskip]
ar butterbr asig, kfreq, kband [, iskip]
ar butterhp asig, kfreq [, iskip]
ar butterlp asig, kfreq [, iskip]
kr button knum
ar buzz xamp, xcps, knh, ifn [, iphs]
ar cabasa iamp, idettack [, inum] [, idamp] [, imaxshake]
ar cauchy kalpha
ir cauchy kalpha
kr cauchy kalpha
cent(x)
cggoto condition, label
ival chanctrl ichnl, ictlno [, ilow] [, ihigh]
kval chanctrl ichnl, ictlno [, ilow] [, ihigh]
kr checkbox knum
cigoto condition, label
ckgoto condition, label
clear avar1 [, avar2] [, avar3] [...]
ar clfilt asig, kfreq, itype, inpol [, ikind] [, ipbr] [, isba] [, iskip]
ar clip asig, imeth, ilimit [, iarg]
clockoff inum
clockon inum
cngoto condition, label
ar comb asig, krvt, ilpt [, iskip] [, insmps]
kr control knum
ar1 [, ar2] [, ar3] [, ar4] convle ain, ifilcod [, ichannel]
ar1 [, ar2] [, ar3] [, ar4] convolve ain, ifilcod [, ichannel]
cosh(x) (no rate restriction)
cosinv(x) (no rate restriction)
cos(x) (no rate restriction)
icps cps2pch ipch, iequal
icps cpsmidib [irange]
kcps cpsmidib [irange]
icps cpsmidi
cpsoct (oct) (no rate restriction)
cpspch (pch) (init- or control-rate args only)
icps cpstmid ifn
icps cpstuni index, ifn
kcps cpstun ktrig, kindex, kfn
icps cpsxpch ipch, iequal, irepeat, ibase
cpuprc insnum, ipercent
ar cross2 ain1, ain2, isize, ioverlap, iwin, kbias
ar crunch iamp, idettack [, inum] [, idamp] [, imaxshake]
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]
ctrlinit ichnl, ictlno1, ival1 [, ictlno2] [, ival2] [, ictlno3] [, ival3] [,...ival32]
aout cuserrnd kmin, kmax, ktableNum
iout cuserrnd imin, imax, itableNum
kout cuserrnd kmin, kmax, ktableNum
ar dam asig, kthreshold, icomp1, icomp2, irtime, iftime
dbamp(x) (init-rate or control-rate args only)
dbfsamp(x) (init-rate or control-rate args only)
db(x)
ar dcblock ain [, igain]
ar dconv asig, isize, ifn
ar delay1 asig [, iskip]
ar delayr idlt [, iskip]
ar delay asig, idlt [, iskip]
delayw asig
ar deltap3 xdlt
ar deltapi xdlt
ar deltapn xnumsamps
ar deltap kdlt
aout deltapx adel, iwsize
deltapxw ain, adel, iwsize
ar diff asig [, iskip]
kr diff ksig [, iskip]
ar1 [,ar2] [, ar3] [, ar4] diskin ifilcod, kpitch [, iskiptim] [, iwraparound] [, iformat]
dispfft xsig, iprd, iwsiz [, iwtyp] [, idbout] [, iwtflg]
display xsig, iprd [, inprds] [, iwtflg]
ar distort1 asig, kpregain, kpostgain, kshape1, kshape2
ar divz xa, xb, ksubst
ir divz ia, ib, isubst
kr divz ka, kb, ksubst
kr downsamp asig [, iwlen]
ar dripwater kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] [, ifreq1] [, ifreq2]
dumpk2 ksig1, ksig2, ifilname, iformat, iprd
dumpk3 ksig1, ksig2, ksig3, ifilname, iformat, iprd
dumpk4 ksig1, ksig2, ksig3, ksig4, ifilname, iformat, iprd
dumpk ksig, ifilname, iformat, iprd
aout duserrnd ktableNum
iout duserrnd itableNum
kout duserrnd ktableNum
elseif xa R xb then
else
endif
endin
endop
ar envlpxr xamp, irise, idur, idec, ifn, iatss, iatdec [, ixmod] [,irind]
kr envlpxr kamp, irise, idur, idec, ifn, iatss, iatdec [, ixmod] [,irind]
ar envlpx xamp, irise, idur, idec, ifn, iatss, iatdec [, ixmod]
kr envlpx kamp, irise, idur, idec, ifn, iatss, iatdec [, ixmod]
event "scorechar", kinsnum, kdelay, kdur, [, kp4] [, kp5] [, ...]
event "scorechar", "insname", kdelay, kdur, [, kp4] [, kp5] [, ...]
ar expon ia, idur1, ib
kr expon ia, idur1, ib
ar exprand krange
ir exprand krange
kr exprand krange
ar expsega ia, idur1, ib [, idur2] [, ic] [...]
ar expsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz
kr expsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz
ar expseg ia, idur1, ib [, idur2] [, ic] [...]
kr expseg ia, idur1, ib [, idur2] [, ic] [...]
exp(x) (no rate restriction)
ir filelen ifilcod
ir filenchnls ifilcod
ir filepeak ifilcod [, ichnl]
ir filesr ifilcod
ar filter2 asig, iM, iN, ib0, ib1, ..., ibM, ia1, ia2, ..., iaN
kr filter2 ksig, iM, iN, ib0, ib1, ..., ibM, ia1, ia2, ..., iaN
fini ifilename, iskipframes, iformat, in1 [, in2] [, in3] [, ...]
fink ifilename, iskipframes, iformat, kin1 [, kin2] [, kin3] [,...]
fin ifilename, iskipframes, iformat, ain1 [, ain2] [, ain3] [,...]
ihandle fiopen ifilename, imode
ar flanger asig, adel, kfeedback [, imaxd]
flashtxt iwhich, String
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]
FLcolor2 ired, igreen, iblue
FLcolor ired, igreen, iblue
kout, ihandle FLcount "label", imin, imax, istep1, istep2, itype, iwidth, iheight, ix, iy, iopcode [, kp1] [, kp2] [, kp3] [...] [, kpN]
inumsnap FLgetsnap index
FLgroupEnd
FLgroup "label", iwidth, iheight, ix, iy [, iborder] [, image]
FLhide ihandle
koutx, kouty, ihandlex, ihandley FLjoy "label", iminx, imaxx, iminy, imaxy, iexpx, iexpy, idispx, idispy, iwidth, iheight, ix, iy
kout FLkeyb kparam1 [, kparam2] ... [, kparamN]
kout, ihandle FLknob "label", imin, imax, iexp, itype, idisp, iwidth, ix, iy [, icursorsize]
FLlabel isize, ifont, ialign, ired, igreen, iblue
FLloadsnap "filename"
FLpackEnd
FLpack iwidth, iheight, ix, iy, itype, ispace, iborder
FLpanelEnd
FLpanel "label", iwidth, iheight [, ix] [, iy] [, iborder]
FLprintk2 kval, idisp
FLprintk itime, kval, idisp
kout, ihandle FLroller "label", imin, imax, istep, iexp, itype, idisp, iwidth, iheight, ix, iy
FLrun
FLsavesnap "filename"
FLscrollEnd
FLscroll iwidth, iheight [, ix] [, iy]
FLsetAlign ialign, ihandle
FLsetBox itype, ihandle
FLsetColor2 ired, igreen, iblue, ihandle
FLsetColor ired, igreen, iblue, ihandle
FLsetFont ifont, ihandle
FLsetPosition ix, iy, ihandle
FLsetSize iwidth, iheight, ihandle
inumsnap, inumval FLsetsnap index [, ifn]
FLsetTextColor isize, ihandle
FLsetText "itext", ihandle
FLsetTextSize isize, ihandle
FLsetTextType itype, ihandle
FLsetVal_i kvalue, ihandle
FLsetVal ktrig, kvalue, ihandle
FLshow ihandle
FLslidBnk "names", inumsliders [, ioutable] [, iwidth] [, iheight] [, ix] [, iy] [, itypetable] [, iexptable] [, istart_index] [, iminmaxtable]
kout, ihandle FLslider "label", imin, imax, iexp, itype, idisp, iwidth, iheight, ix, iy
FLtabsEnd
FLtabs iwidth, iheight, ix, iy
kout, ihandle FLtext "label", imin, imax, istep, itype, iwidth, iheight, ix, iy
FLupdate
ihandle FLvalue "label", iwidth, iheight, ix, iy
ar fmb3 kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn
ar fmbell kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn
ar fmmetal kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn
ar fmpercfl kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn
ar fmrhode kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn
ar fmvoice kamp, kfreq, kvowel, ktilt, kvibamt, kvibrate, ifn1, ifn2, ifn3, ifn4, ivibfn
ar fmwurlie kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn
ar fof2 xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss [, iskip]
ar fof xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur [, iphs] [, ifmode] [, iskip]
ar fog xamp, xdens, xtrans, aspd, koct, kband, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur [, iphs] [, itmode] [, iskip]
ar fold asig, kincr
ar follow2 asig, katt, krel
ar follow asig, idt
ar foscili xamp, kcps, xcar, xmod, kndx, ifn [, iphs]
ar foscil xamp, kcps, xcar, xmod, kndx, ifn [, iphs]
foutir ihandle, iformat, iflag, iout1 [, iout2, iout3,....,ioutN]
fouti ihandle, iformat, iflag, iout1 [, iout2, iout3,....,ioutN]
foutk ifilename, iformat, kout1 [, kout2, kout3,....,koutN]
fout ifilename, iformat, aout1 [, aout2, aout3,...,aoutN]
fprintks "filename", "string", [, kval1] [, kval2] [...]
fprints "filename", "string" [, kval1] [, kval2] [...]
frac(x) (init-rate or control-rate args only)
ftchnls(x) (init-rate args only)
gir ftgen ifn, itime, isize, igen, iarga [, iargb ] [...]
ftlen(x) (init-rate args only)
ftloadk "filename", ktrig, iflag, ifn1 [, ifn2] [...]
ftload "filename", iflag, ifn1 [, ifn2] [...]
ftlptim(x) (init-rate args only)
ftmorf kftndx, iftfn, iresfn
ftsavek "filename", ktrig, iflag, ifn1 [, ifn2] [...]
ftsave "filename", iflag, ifn1 [, ifn2] [...]
ftsr(x) (init-rate args only)
ar gain asig, krms [, ihp] [, iskip]
ar gauss krange
ir gauss krange
kr gauss krange
ar gbuzz xamp, xcps, knh, klh, kmul, ifn [, iphs]
ar gogobel kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivfn
goto label
ar grain2 kcps, kfmd, kgdur, iovrlp, kfn, iwfn [, irpow] [, iseed] [, imode]
ar grain3 kcps, kphs, kfmd, kpmd, kgdur, kdens, imaxovr, kfn, iwfn, kfrpow, kprpow [, iseed] [, imode]
ar grain xamp, xpitch, xdens, kampoff, kpitchoff, kgdur, igfn, iwfn, imgdur [, igrnd]
ar 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]
ar guiro kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] [, ifreq1]
ar harmon asig, kestfrq, kmaxvar, kgenfreq1, kgenfreq2, imode, iminfrq, iprd
ar1, ar2 hilbert asig
aleft, aright hrtfer asig, kaz, kelev, “HRTFcompact”
ar hsboscil kamp, ktone, kbrite, ibasfreq, iwfn, ioctfn [, ioctcnt] [, iphs]
i(x) (control-rate args only)
if ia R ib igoto label
if ka R kb kgoto label
if ia R ib goto label
if xa R xb then
igoto label
ihold
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
ar1 inch ksig1
ar1, ar2, ar3, ar4, ar5, ar6 inh
initc14 ichan, ictlno1, ictlno2, ivalue
initc21 ichan, ictlno1, ictlno2, ictlno3, ivalue
initc7 ichan, ictlno, ivalue
ar init iarg
ir init iarg
kr init iarg
ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8 ino
ar1, ar2, ar3, a4 inq
ar1 in
ar1, ar2 ins
instr i, j, ...
ar integ asig [, iskip]
kr integ ksig [, iskip]
ar interp ksig [, iskip] [, imode]
int(x) (init-rate or control-rate args only)
kvalue invalue "channel name"
ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, ar13, ar14, ar15, ar16 inx
inz ksig1
kout jitter2 ktotamp, kamp1, kcps1, kamp2, kcps2, kamp3, kcps3
kout jitter kamp, kcpsMin, kcpsMax
ar jspline xamp, kcpsMin, kcpsMax
kr jspline kamp, kcpsMin, kcpsMax
kgoto label
kr = iarg
ksmps = iarg
ktableseg ifn1, idur1, ifn2 [, idur2] [, ifn3] [...]
kr lfo kamp, kcps [, itype]
ar lfo kamp, kcps [, itype]
ar limit asig, klow, khigh
ir limit isig, ilow, ihigh
kr limit ksig, klow, khigh
ar linenr xamp, irise, idec, iatdec
kr linenr kamp, irise, idec, iatdec
ar linen xamp, irise, idur, idec
kr linen kamp, irise, idur, idec
ar line ia, idur1, ib
kr line ia, idur1, ib
kr lineto ksig, ktime
ar linrand krange
ir linrand krange
kr linrand krange
ar linsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz
kr linsegr ia, idur1, ib [, idur2] [, ic] [...], irel, iz
ar linseg ia, idur1, ib [, idur2] [, ic] [...]
kr linseg ia, idur1, ib [, idur2] [, ic] [...]
a1, a2 locsend
a1, a2, a3, a4 locsend
a1, a2 locsig asig, kdegree, kdistance, kreverbsend
a1, a2, a3, a4 locsig asig, kdegree, kdistance, kreverbsend
log10(x) (no rate restriction)
logbtwo(x) (init-rate or control-rate args only)
log(x) (no rate restriction)
ksig loopseg kfreq, ktrig, ktime0, kvalue0 [, ktime1] [, kvalue1] [, ktime2] [, kvalue2] [...]
ax, ay, az lorenz ksv, krv, kbv, kh, ix, iy, iz, iskip
ar [,ar2] loscil3 xamp, kcps, ifn [, ibas] [, imod1] [, ibeg1] [, iend1] [, imod2] [, ibeg2] [, iend2]
ar [,ar2] loscil xamp, kcps, ifn [, ibas] [, imod1] [, ibeg1] [, iend1] [, imod2,] [, ibeg2] [, iend2]
ar lowpass2 asig, kcf, kq [, iskip]
ar lowres asig, kcutoff, kresonance [, iskip]
ar lowresx asig, kcutoff, kresonance [, inumlayer] [, iskip]
ar lpf18 asig, kfco, kres, kdist
ar lpfreson asig, kfrqratio
ar lphasor xtrns [, ilps] [, ilpe] [, imode] [, istrt] [, istor]
lpinterp islot1, islot2, kmix
ar lposcil3 kamp, kfreqratio, kloop, kend, ifn [, iphs]
ar lposcil kamp, kfreqratio, kloop, kend, ifn [, iphs]
krmsr, krmso, kerr, kcps lpread ktimpnt, ifilcod [, inpoles] [, ifrmrate]
ar lpreson asig
ksig lpshold kfreq, ktrig, ktime0, kvalue0 [, ktime1] [, kvalue1] [, ktime2] [, kvalue2] [...]
lpslot islot
ar maca asig1 [, asig2] [, asig3] [, asig4] [, asig5] [...]
ar mac asig1, ksig1 [, asig2] [, ksig2] [, asig3] [, ksig3] [...]
ar madsr iatt, idec, islev, irel [, idel] [, ireltim]
kr madsr iatt, idec, islev, irel [, idel] [, ireltim]
ar mandol kamp, kfreq, kpluck, kdetune, kgain, ksize, ifn [, iminfreq]
ar marimba kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec [, idoubles] [, itriples]
massign ichnl, insnum
massign ichnl, "insname"
maxalloc insnum, icount
mclock ifreq
mdelay kstatus, kchan, kd1, kd2, kdelay
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]
midichannelaftertouch xchannelaftertouch [, ilow] [, ihigh]
ichn midichn
midicontrolchange xcontroller, xcontrollervalue [, ilow] [, ihigh]
ival midictrl inum [, imin] [, imax]
kval midictrl inum [, imin] [, imax]
mididefault xdefault, xvalue
kstatus, kchan, kdata1, kdata2 midiin
midinoteoff xkey, xvelocity
midinoteoncps xcps, xvelocity
midinoteonkey xkey, xvelocity
midinoteonoct xoct, xvelocity
midinoteonpch xpch, xvelocity
midion2 kchn, knum, kvel, ktrig
midion kchn, knum, kvel
midiout kstatus, kchan, kdata1, kdata2
midipitchbend xpitchbend [, ilow] [, ihigh]
midipolyaftertouch xpolyaftertouch, xcontrollervalue [, ilow] [, ihigh]
midiprogramchange xprogram
ar mirror asig, klow, khigh
ir mirror isig, ilow, ihigh
kr mirror ksig, klow, khigh
ar moog kamp, kfreq, kfiltq, kfiltrate, kvibf, kvamp, iafn, iwfn, ivfn
ar moogvcf asig, xfco, xres [, iscale]
moscil kchn, knum, kvel, kdur, kpause
ar mpulse kamp, kfreq [, ioffset]
mrtmsg imsgtype
ar multitap asig [, itime1] [, igain1] [, itime2] [, igain2] [...]
mute insnum [, iswitch]
mute "insname" [, iswitch]
ar mxadsr iatt, idec, islev, irel [, idel] [, ireltim]
kr mxadsr iatt, idec, islev, irel [, idel] [, ireltim]
nchnls = iarg
ar nestedap asig, imode, imaxdel, idel1, igain1 [, idel2] [, igain2] [, idel3] [, igain3] [, istor]
ar nlfilt ain, ka, kb, kd, kC, kL
ar noise xamp, kbeta
noteoff ichn, inum, ivel
noteondur2 ichn, inum, ivel, idur
noteondur ichn, inum, ivel, idur
noteon ichn, inum, ivel
ival notnum
ar nreverb asig, ktime, khdif [, iskip] [,inumCombs] [, ifnCombs] [, inumAlpas] [, ifnAlpas]
nrpn kchan, kparmnum, kparmvalue
nsamp(x) (init-rate args only)
insno nstrnum "name"
ar ntrpol asig1, asig2, kpoint [, imin] [, imax]
ir ntrpol isig1, isig2, ipoint [, imin] [, imax]
kr ntrpol ksig1, ksig2, kpoint [, imin] [, imax]
octave(x)
octcps (cps) (init- or control-rate args only)
ioct octmidib [irange]
koct octmidib [irange]
ioct octmidi
octpch (pch) (init- or control-rate args only)
opcode name, outtypes, intypes
ar 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]
kr oscil1i idel, kamp, idur, ifn
kr oscil1 idel, kamp, idur, ifn
ar oscil3 xamp, xcps, ifn [, iphs]
kr oscil3 kamp, kcps, ifn [, iphs]
ar osciliktp kcps, kfn, kphs [, istor]
ar oscilikt xamp, xcps, kfn [, iphs] [, istor]
kr oscilikt kamp, kcps, kfn [, iphs] [, istor]
ar oscilikts xamp, xcps, kfn, async, kphs [, istor]
ar oscili xamp, xcps, ifn [, iphs]
kr oscili kamp, kcps, ifn [, iphs]
ar osciln kamp, ifrq, ifn, itimes
ar oscil xamp, xcps, ifn [, iphs]
kr oscil kamp, kcps, ifn [, iphs]
ar oscils iamp, icps, iphs [, iflg]
ar oscilx kamp, ifrq, ifn, itimes
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
outch ksig1, asig1 [, ksig2] [, asig2] [...]
outc asig1 [, asig2] [...]
outh asig1, asig2, asig3, asig4, asig5, asig6
outiat ichn, ivalue, imin, imax
outic14 ichn, imsb, ilsb, ivalue, imin, imax
outic ichn, inum, ivalue, imin, imax
outipat ichn, inotenum, ivalue, imin, imax
outipb ichn, ivalue, imin, imax
outipc ichn, iprog, imin, imax
outkat kchn, kvalue, kmin, kmax
outkc14 kchn, kmsb, klsb, kvalue, kmin, kmax
outkc kchn, knum, kvalue, kmin, kmax
outkpat kchn, knotenum, kvalue, kmin, kmax
outkpb kchn, kvalue, kmin, kmax
outkpc kchn, kprog, kmin, kmax
outo asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8
outq1 asig
outq2 asig
outq3 asig
outq4 asig
outq asig1, asig2, asig3, asig4
outs1 asig
outs2 asig
out asig
outs asig1, asig2
outvalue "channel name", kvalue
outx asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8, asig9, asig10, asig11, asig12, asig13, asig14, asig15, asig16
outz ksig1
a1, a2, a3, a4 pan asig, kx, ky, ifn [, imode] [, ioffset]
ar pareq asig, kc, kv, kq [, imode]
ar pcauchy kalpha
ir pcauchy kalpha
kr pcauchy kalpha
ibend pchbend [imin] [, imax]
kbend pchbend [imin] [, imax]
ipch pchmidib [irange]
kpch pchmidib [irange]
ipch pchmidi
pchoct (oct) (init- or control-rate args only)
kr peak asig
kr peak ksig
pgmassign ipgm, inst
pgmassign ipgm, "insname"
ar phaser1 asig, kfreq, kord, kfeedback [, iskip]
ar phaser2 asig, kfreq, kq, kord, kmode, ksep, kfeedback
ar phasorbnk xcps, kndx, icnt [, iphs]
kr phasorbnk kcps, kndx, icnt [, iphs]
ar phasor xcps [, iphs]
kr phasor kcps [, iphs]
ar pinkish xin [, imethod] [, inumbands] [, iseed] [, iskip]
kcps, krms pitchamdf asig, imincps, imaxcps [, icps] [, imedi] [, idowns] [, iexcps] [, irmsmedi]
koct, kamp pitch asig, iupdte, ilo, ihi, idbthresh [, ifrqs] [, iconf] [, istrt] [, iocts] [, iq] [, inptls] [, irolloff] [, iskip]
ax, ay, az planet kmass1, kmass2, ksep, ix, iy, iz, ivx, ivy, ivz, idelta [, ifriction]
ar pluck kamp, kcps, icps, ifn, imeth [, iparm1] [, iparm2]
ar poisson klambda
ir poisson klambda
kr poisson klambda
ir polyaft inote [, ilow] [, ihigh]
kr polyaft inote [, ilow] [, ihigh]
kr portk ksig, khtim [, isig]
kr port ksig, ihtim [, isig]
ar poscil3 kamp, kcps, ifn [, iphs]
kr poscil3 kamp, kcps, ifn [, iphs]
ar poscil aamp, acps, ifn [, iphs]
ar poscil aamp, kcps, ifn [, iphs]
ar poscil kamp, acps, ifn [, iphs]
ar poscil kamp, kcps, ifn [, iphs]
ir poscil kamp, kcps, ifn [, iphs]
kr poscil kamp, kcps, ifn [, iphs]
powoftwo(x) (init-rate or control-rate args only)
ar pow aarg, kpow [, inorm]
ir pow iarg, ipow [, inorm]
kr pow karg, kpow [, inorm]
prealloc insnum, icount
prealloc "insname", icount
printk2 kvar [, inumspaces]
printk itime, kval [, ispace]
printks "string", itime [, kval1] [, kval2] [...]
print iarg [, iarg1] [, iarg2] [...]
prints "string" [, kval1] [, kval2] [...]
ar product asig1, asig2 [, asig3] [...]
pset icon1 [, icon2] [...]
p(x)
ar pvadd ktimpnt, kfmod, ifilcod, ifn, ibins [, ibinoffset] [, ibinincr] [, iextractmode] [, ifreqlim] [, igatefn]
pvbufread ktimpnt, ifile
ar pvcross ktimpnt, kfmod, ifile, kampscale1, kampscale2 [, ispecwp]
ar pvinterp ktimpnt, kfmod, ifile, kfreqscale1, kfreqscale2, kampscale1, kampscale2, kfreqinterp, kampinterp
ar pvoc ktimpnt, kfmod, ifilcod [, ispecwp] [, iextractmode] [, ifreqlim] [, igatefn]
kfreq, kamp pvread ktimpnt, ifile, ibin
ar pvsadsyn fsrc, inoscs, kfmod [, ibinoffset] [, ibinincr] [, iinit]
fsig pvsanal ain, ifftsize, ioverlap, iwinsize, iwintype [, iformat] [, iinit]
fsig pvscross fsrc, fdest, kamp1, kamp2
fsig pvsfread ktimpt, ifn [, ichan]
pvsftr fsrc, ifna [, ifnf]
kflag pvsftw fsrc, ifna [, ifnf]
ioverlap, inumbins, iwinsize, iformat pvsinfo fsrc
fsig pvsmaska fsrc, ifn, kdepth
ar pvsynth fsrc, [iinit]
ar randh xamp, xcps [, iseed] [, isize] [, ioffset]
kr randh kamp, kcps [, iseed] [, isize] [, ioffset]
ar randi xamp, xcps [, iseed] [, isize] [, ioffset]
kr randi kamp, kcps [, iseed] [, isize] [, ioffset]
ar randomh kmin, kmax, acps
kr randomh kmin, kmax, kcps
ar randomi kmin, kmax, acps
kr randomi kmin, kmax, kcps
ar random kmin, kmax
ir random imin, imax
kr random kmin, kmax
ar rand xamp [, iseed] [, isel] [, ibase]
kr rand xamp [, iseed] [, isel] [, ibase]
ir readclock inum
kr1, kr2 readk2 ifilname, iformat, ipol [, interp]
kr1, kr2, kr3 readk3 ifilname, iformat, ipol [, interp]
kr1, kr2, kr3, kr4 readk4 ifilname, iformat, ipol [, interp]
kr readk ifilname, iformat, ipol [, interp]
reinit label
kflag release
ar repluck iplk, kamp, icps, kpick, krefl, axcite
kr resonk ksig, kcf, kbw [, iscl] [, iskip]
ar resonr asig, kcf, kbw [, iscl] [, iskip]
ar reson asig, kcf, kbw [, iscl] [, iskip]
ar resonx asig, kcf, kbw [, inumlayer] [, iscl] [, iskip]
ar resony asig, kbf, kbw, inum, ksep [, isepmode] [, iscl] [, iskip]
ar resonz asig, kcf, kbw [, iscl] [, iskip]
ar reverb2 asig, ktime, khdif [, iskip] [,inumCombs] [, ifnCombs] [, inumAlpas] [, ifnAlpas]
ar reverb asig, krvt [, iskip]
ar rezzy asig, xfco, xres [, imode]
rigoto label
rireturn
kr rms asig [, ihp] [, iskip]
ax rnd31 kscl, krpow [, iseed]
ix rnd31 iscl, irpow [, iseed]
kx rnd31 kscl, krpow [, iseed]
rnd(x) (init- or control-rate only)
ar rspline xrangeMin, xrangeMax, kcpsMin, kcpsMax
kr rspline krangeMin, krangeMax, kcpsMin, kcpsMax
ir rtclock
kr rtclock
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
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
ar samphold asig, agate [, ival] [, ivstor]
kr samphold ksig, kgate [, ival] [, ivstor]
ar sandpaper iamp, idettack [, inum] [, idamp] [, imaxshake]
scanhammer isrc, idst, ipos, imult
ar 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
schedkwhennamed ktrigger, kmintim, kmaxnum, "name", kwhen, kdur [, ip4] [, ip5] [...]
schedkwhen ktrigger, kmintim, kmaxnum, kinsnum, kwhen, kdur [, ip4] [, ip5] [...]
schedkwhen ktrigger, kmintim, kmaxnum, "insname", 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] [...]
seed ival
ar sekere iamp, idettack [, inum] [, idamp] [, imaxshake]
semitone(x)
kr sensekey
kr sense
ktrig_out seqtime ktime_unit, kstart, kloop, kinitndx, kfn_times
setctrl inum, ival, itype
setksmps iksmps
sfilist ifilhandle
ar sfinstr3m ivel, inotenum, xamp, xfreq, instrnum, ifilhandle [, iflag] [, ioffset]
ar1, ar2 sfinstr3 ivel, inotenum, xamp, xfreq, instrnum, ifilhandle [, iflag] [, ioffset]
ar sfinstrm ivel, inotenum, xamp, xfreq, instrnum, ifilhandle [, iflag] [, ioffset]
ar1, ar2 sfinstr ivel, inotenum, xamp, xfreq, instrnum, ifilhandle [, iflag] [, ioffset]
ir sfload "filename"
sfpassign istartindex, ifilhandle
ar sfplay3m ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset]
ar1, ar2 sfplay3 ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset]
ar sfplaym ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset]
ar1, ar2 sfplay ivel, inotenum, xamp, xfreq, ipreindex [, iflag] [, ioffset]
sfplist ifilhandle
ir sfpreset iprog, ibank, ifilhandle, ipreindex
ar shaker kamp, kfreq, kbeans, kdamp, ktimes [, idecay]
sinh(x) (no rate restriction)
sininv(x) (no rate restriction)
sin(x) (no rate restriction)
ar sleighbells kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] [, ifreq1] [, ifreq2]
k1,...,k16 slider16f ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1,..., ictlnum16, imin16, imax16, init16, ifn16, icutoff16
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
k1,...,k32 slider32f ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1,..., ictlnum32, imin32, imax32, init32, ifn32, icutoff32
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
k1,...,k64 slider64f ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1,..., ictlnum64, imin64, imax64, init64, ifn64, icutoff64
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
k1,...,k8 slider8f ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1,..., ictlnum8, imin8, imax8, init8, ifn8, icutoff8
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
ar [, 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
ar1 soundin ifilcod [, iskptim] [, iformat]
ar1, ar2 soundin ifilcod [, iskptim] [, iformat]
ar1, ar2, ar3 soundin ifilcod [, iskptim] [, iformat]
ar1, ar2, ar3, ar4 soundin ifilcod [, iskptim] [, iformat]
soundout asig1, ifilcod [, iformat]
a1, a2, a3, a4 space asig, ifn, ktime, kreverbsend, kx, ky
aW, aX, aY, aZ spat3di ain, iX, iY, iZ, idist, ift, imode [, istor]
aW, aX, aY, aZ spat3d ain, kX, kY, kZ, idist, ift, imode, imdel, iovr [, istor]
spat3dt ioutft, iX, iY, iZ, idist, ift, imode, irlen [, iftnocl]
k1 spdist ifn, ktime, kx, ky
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]
a1, a2, a3, a4 spsend
sqrt(x) (no rate restriction)
sr = iarg
ar stix iamp, idettack [, inum] [, idamp] [, imaxshake]
ar streson asig, kfr, ifdbgain
strset iarg, istring
subinstrinit instrnum [, p4] [, p5] [...]
subinstrinit "insname" [, p4] [, p5] [...]
a1, [...] [, a8] subinstr instrnum [, p4] [, p5] [...]
a1, [...] [, a8] subinstr "insname" [, p4] [, p5] [...]
ar sum asig1 [, asig2] [, asig3] [...]
alow, ahigh, aband svfilter asig, kcf, kq [, iscl]
ar table3 andx, ifn [, ixmode] [, ixoff] [, iwrap]
ir table3 indx, ifn [, ixmode] [, ixoff] [, iwrap]
kr table3 kndx, ifn [, ixmode] [, ixoff] [, iwrap]
tablecopy kdft, ksft
tablegpw kfn
tableicopy idft, isft
tableigpw ifn
ar tableikt xndx, kfn [, ixmode] [, ixoff] [, iwrap]
kr tableikt kndx, kfn [, ixmode] [, ixoff] [, iwrap]
tableimix idft, idoff, ilen, is1ft, is1off, is1g, is2ft, is2off, is2g
ar tablei andx, ifn [, ixmode] [, ixoff] [, iwrap]
ir tablei indx, ifn [, ixmode] [, ixoff] [, iwrap]
kr tablei kndx, ifn [, ixmode] [, ixoff] [, iwrap]
tableiw isig, indx, ifn [, ixmode] [, ixoff] [, iwgmode]
ar tablekt xndx, kfn [, ixmode] [, ixoff] [, iwrap]
kr tablekt kndx, kfn [, ixmode] [, ixoff] [, iwrap]
tablemix kdft, kdoff, klen, ks1ft, ks1off, ks1g, ks2ft, ks2off, ks2g
ir tableng ifn
kr tableng kfn
ar tablera kfn, kstart, koff
tableseg ifn1, idur1, ifn2 [, idur2] [, ifn3] [...]
ar table andx, ifn [, ixmode] [, ixoff] [, iwrap]
ir table indx, ifn [, ixmode] [, ixoff] [, iwrap]
kr table kndx, ifn [, ixmode] [, ixoff] [, iwrap]
kstart tablewa kfn, asig, koff
tablewkt asig, andx, kfn [, ixmode] [, ixoff] [, iwgmode]
tablewkt ksig, kndx, kfn [, ixmode] [, ixoff] [, iwgmode]
tablew asig, andx, ifn [, ixmode] [, ixoff] [, iwgmode]
tablew isig, indx, ifn [, ixmode] [, ixoff] [, iwgmode]
tablew ksig, kndx, ifn [, ixmode] [, ixoff] [, iwgmode]
ar tablexkt xndx, kfn, kwarp, iwsize [, ixmode] [, ixoff] [, iwrap]
tablexseg ifn1, idur1, ifn2 [, idur2] [, ifn3] [...]
ar tambourine kamp, idettack [, inum] [, idamp] [, imaxshake] [, ifreq] [, ifreq1] [, ifreq2]
tanh(x) (no rate restriction)
ar taninv2 ay, ax
ir taninv2 iy, ix
kr taninv2 ky, kx
taninv(x) (no rate restriction)
tan(x) (no rate restriction)
ar tbvcf asig, xfco, xres, kdist, kasym
ktemp tempest kin, iprd, imindur, imemdur, ihp, ithresh, ihtim, ixfdbak, istartempo, ifn [, idisprd] [, itweek]
tempo ktempo, istartempo
kr tempoval
tigoto label
kr timeinstk
kr timeinsts
kr timeinsts
ir timek
kr timek
ir times
kr times
timout istrt, idur, label
ir tival
kr tlineto ksig, ktime, ktrig
kr tonek ksig, khp [, iskip]
ar tone asig, khp [, iskip]
ar tonex asig, khp [, inumlayer] [, iskip]
ar transeg ia, idur, itype, ib [, idur2] [, itype] [, ic] ...
kr transeg ia, idur, itype, ib [, idur2] [, itype] [, ic] ...
kout trigger ksig, kthreshold, kmode
trigseq ktrig_in, kstart, kloop, kinitndx, kfn_values, kout1 [, kout2] [...]
ar trirand krange
ir trirand krange
kr trirand krange
turnoff
turnon insnum [, itime]
ar unirand krange
ir unirand krange
kr unirand krange
ar upsamp ksig
aout = urd(ktableNum)
iout = urd(itableNum)
kout = urd(ktableNum)
ar valpass asig, krvt, xlpt, imaxlpt [, iskip] [, insmps]
ar1, ..., ar16 vbap16move asig, ispread, ifldnum, ifld1 [, ifld2] [...]
ar1, ..., ar16 vbap16 asig, iazim [, ielev] [, ispread]
ar1, ar2, ar3, ar4 vbap4move asig, ispread, ifldnum, ifld1 [, ifld2] [...]
ar1, ar2, ar3, ar4 vbap4 asig, iazim [, ielev] [, ispread]
ar1, ..., ar8 vbap8move asig, ispread, ifldnum, ifld1 [, ifld2] [...]
ar1, ..., ar8 vbap8 asig, iazim [, ielev] [, ispread]
vbaplsinit idim, ilsnum [, idir1] [, idir2] [...] [, idir32]
vbapzmove inumchnls, istartndx, asig, idur, ispread, ifldnum, ifld1, ifld2, [...]
vbapz inumchnls, istartndx, asig, iazim [, ielev] [, ispread]
kfn vco2ft kcps, iwave [, inyx]
ifn vco2ift icps, iwave [, inyx]
ifn vco2init iwave [, ibasfn] [, ipmul] [, iminsiz] [, imaxsiz] [, isrcft]
ar vco2 kamp, kcps [, imode] [, kpw] [, kphs] [, inyx]
ar vcomb asig, krvt, xlpt, imaxlpt [, iskip] [, insmps]
ar vco xamp, xcps, iwave, kpw [, ifn] [, imaxd] [, ileak] [, inyx] [, iphs]
ar vdelay3 asig, adel, imaxdel [, iskip]
ar vdelay asig, adel, imaxdel [, iskip]
aout1, aout2, aout3, aout4 vdelayxq ain1, ain2, ain3, ain4, adl, imd, iws [, ist]
aout vdelayx ain, adl, imd, iws [, ist]
aout1, aout2 vdelayxs ain1, ain2, adl, imd, iws [, ist]
aout1, aout2, aout3, aout4 vdelayxwq ain1, ain2, ain3, ain4, adl, imd, iws [, ist]
aout vdelayxw ain, adl, imd, iws [, ist]
aout1, aout2 vdelayxws ain1, ain2, adl, imd, iws [, ist]
ival veloc [ilow] [, ihigh]
ar vibes kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec
kout vibrato kAverageAmp, kAverageFreq, kRandAmountAmp, kRandAmountFreq, kAmpMinRate, kAmpMaxRate, kcpsMinRate, kcpsMaxRate, ifn [, iphs]
kout vibr kAverageAmp, kAverageFreq, ifn
vincr asig, aincr
ar vlowres asig, kfco, kres, iord, ksep
ar voice kamp, kfreq, kphoneme, kform, kvibf, kvamp, ifn, ivfn
ar vpvoc ktimpnt, kfmod, ifile [, ispecwp] [, ifn]
ar waveset ain, krep [, ilen]
ar weibull ksigma, ktau
ir weibull ksigma, ktau
kr weibull ksigma, ktau
ar wgbowedbar kamp, kfreq, kpos, kbowpres, kgain [, iconst] [, itvel] [, ibowpos] [, ilow]
ar wgbow kamp, kfreq, kpres, krat, kvibf, kvamp, ifn [, iminfreq]
ar wgbrass kamp, kfreq, ktens, iatt, kvibf, kvamp, ifn [, iminfreq]
ar wgclar kamp, kfreq, kstiff, iatt, idetk, kngain, kvibf, kvamp, ifn [, iminfreq]
ar wgflute kamp, kfreq, kjet, iatt, idetk, kngain, kvibf, kvamp, ifn [, iminfreq] [, ijetrf] [, iendrf]
ar wgpluck2 iplk, kamp, icps, kpick, krefl
ar wgpluck icps, iamp, kpick, iplk, idamp, ifilt, axcite
ar wguide1 asig, xfreq, kcutoff, kfeedback
ar wguide2 asig, xfreq1, xfreq2, kcutoff1, kcutoff2, kfeedback1, kfeedback2
ar wrap asig, klow, khigh
ir wrap isig, ilow, ihigh
kr wrap ksig, klow, khigh
aout wterrain kamp, kpch, k_xcenter, k_ycenter, k_xradius, k_yradius, itabx, itaby
ar xadsr iatt, idec, islev, irel [, idel]
kr xadsr iatt, idec, islev, irel [, idel]
xinarg1 [, xinarg2] ... [xinargN] xin
xout xoutarg1 [, xoutarg2] ... [, xoutargN]
kpos, kvel xscanmap iscan, kamp, kvamp [, iwhich]
xscansmap kpos, kvel, iscan, kamp, kvamp [, iwhich]
ar xscans kamp, kfreq, ifntraj, id [, iorder]
xscanu init, irate, ifnvel, ifnmass, ifnstif, ifncentr, ifndamp, kmass, kstif, kcentr, kdamp, ileft, iright, kpos, kstrngth, ain, idisp, id
xtratim iextradur
kx, ky xyin iprd, ixmin, ixmax, iymin, iymax [, ixinit] [, iyinit]
zacl kfirst, klast
zakinit isizea, isizek
ar zamod asig, kzamod
ar zarg kndx, kgain
ar zar kndx
zawm asig, kndx [, imix]
zaw asig, kndx
ar zfilter2 asig, kdamp, kfreq, iM, iN, ib0, ib1, ..., ibM, ia1,ia2, ..., iaN
ir zir indx
ziwm isig, indx [, imix]
ziw isig, indx
zkcl kfirst, klast
kr zkmod ksig, kzkmod
kr zkr kndx
zkwm ksig, kndx [, imix]
zkw ksig, kndx