NOTE: this documentation was automatically generated using pandoc.
This page provides information on how to use the Faust libraries.
The /libraries folder contains the different Faust libraries. If you wish to add your own functions to this library collection, you can refer to the “Contributing” section providing a set of coding conventions.
WARNING: These libraries replace the “old” Faust libraries. They are still being beta tested so you might encounter bugs while using them. If your codes still use the “old” Faust libraries, you might want to try to use Bart Brouns’ script that automatically makes an old Faust code compatible with the new libraries: https://github.com/magnetophon/faustCompressors/blob/master/newlib.sh. If you find a bug, please report it at rmichon_at_ccrma_dot_stanford_dot_edu. Thanks ;)!
The easiest and most standard way to use the Faust libraries is to import stdfaust.lib in your Faust code:
import("stdfaust.lib");This will give you access to all the Faust libraries through a series of environments:
sf: all.liban: analyzers.libba: basics.libco: compressors.libde: delays.libdm: demos.libdx: dx7.liben: envelopes.libfi: filters.libho: hoa.libma: maths.libef: misceffects.libos: oscillators.libno: noises.libpf: phaflangers.libpm: physmodels.libre: reverbs.libro: routes.libsi: signals.libsp: spats.libsy: synths.libve: vaeffects.libEnvironments can then be used as follows in your Faust code:
import("stdfaust.lib");
process = os.osc(440);In this case, we’re calling the osc function from oscillators.lib.
You can also access all the functions of all the libraries directly using the sf environment:
import("stdfaust.lib");
process = sf.osc(440);Alternatively, environments can be created by hand:
os = library("oscillators.lib");
process = os.osc(440);Finally, libraries can be simply imported in the Faust code (not recommended):
import("oscillators.lib");
process = osc(440);If you wish to add a function to any of these libraries or if you plan to add a new library, make sure that you follow the following conventions:
//-----------------functionName--------------------
// Description
//
// #### Usage
//
// ```
// Usage Example
// ```
//
// Where:
//
// * argument1: argument 1 description
//-------------------------------------------------make doclib.os.osc) should be used when calling a function declared in another library (see the section on Using the Faust Libraries).stdfaust.lib with its own environment (2 letters - see stdfaust.lib).generateDoc.declare a name and a version.//############### libraryName ##################
// Description
//
// * Section Name 1
// * Section Name 2
// * ...
//
// It should be used using the `[...]` environment:
//
// ```
// [...] = library("libraryName");
// process = [...].functionCall;
// ```
//
// Another option is to import `stdfaust.lib` which already contains the `[...]`
// environment:
//
// ```
// import("stdfaust.lib");
// process = [...].functionCall;
// ```
//##############################################
//================= Section Name ===============
// Description
//==============================================Only the libraries that are considered to be “standard” are documented:
analyzers.libbasics.libcompressors.libdelays.libdemos.libdx7.libenvelopes.libfilters.libhoa.libmaths.libmisceffects.liboscillators.libnoises.libphaflangers.libphysmodels.libreverbs.libroutes.libsignals.libspats.libsynths.libtonestacks.lib (not documented but example in /examples/misc)tubes.lib (not documented but example in /examples/misc)vaeffects.libOther deprecated libraries such as music.lib, etc. are present but are not documented to not confuse new users.
The doumentation of each library can be found in /documentation/library.html or in /documentation/library.pdf.
The /examples directory contains all the examples from the /examples folder of the Faust distribution as well as new ones. Most of them were updated to reflect the coding conventions described in the next section. Examples are organized by types in different folders. The /old folder contains examples that are fully deprecated, probably because they were integrated to the libraries and fully rewritten (see freeverb.dsp for example). Examples using deprecated libraries were integrated to the general tree but a warning comment was added at their beginning to point readers to the right library and function.
In order to have a uniformized library system, we established the following conventions (that hopefully will be followed by others when making modifications to them :-) ).
faust2md “standards” for each library: //### for main title (library name - equivalent to # in markdown), //=== for section declarations (equivalent to ## in markdown) and //--- for function declarations (equivalent to #### in markdown - see basics.lib for an example).#### markdown title.basics.lib).To prevent cross-references between libraries we generalized the use of the library("") system for function calls in all the libraries. This means that everytime a function declared in another library is called, the environment corresponding to this library needs to be called too. To make things easier, a stdfaust.lib library was created and is imported by all the libraries:
an = library("analyzers.lib");
ba = library("basics.lib");
co = library("compressors.lib");
de = library("delays.lib");
dm = library("demos.lib");
dx = library("dx7.lib");
en = library("envelopes.lib");
fi = library("filters.lib");
ho = library("hoa.lib");
ma = library("maths.lib");
ef = library("misceffects.lib");
os = library("oscillators.lib");
no = library("noises.lib");
pf = library("phaflangers.lib");
pm = library("physmodels.lib");
re = library("reverbs.lib");
ro = library("routes.lib");
sp = library("spats.lib");
si = library("signals.lib");
sy = library("synths.lib");
ve = library("vaeffects.lib");For example, if we wanted to use the smooth function which is now declared in signals.lib, we would do the following:
import("stdfaust.lib");
process = si.smooth(0.999);This standard is only used within the libraries: nothing prevents coders to still import signals.lib directly and call smooth without ro., etc.
“Demo” functions are placed in demos.lib and have a built-in user interface (UI). Their name ends with the _demo suffix. Each of these function have a .dsp file associated to them in the /examples folder.
Any function containing UI elements should be placed in this library and respect these standards.
“Standard” functions are here to simplify the life of new (or not so new) Faust coders. They are declared in /libraries/doc/standardFunctions.md and allow to point programmers to preferred functions to carry out a specific task. For example, there are many different types of lowpass filters declared in filters.lib and only one of them is considered to be standard, etc.
Now that Faust libraries are less author specific, each function will normally have its own copyright-and-license line in the library source (the .lib file, such as analyzers.lib). If not, see if the function is defined within a section of the .lib file stating the license in source-code comments. If not, then the copyright and license given at the beginning of the .lib file may be assumed, when present. If not, run git blame on the .lib file and ask the person who last edited the function!
Note that it is presently possible for a library function released under one license to utilize another library function having some different license. There is presently no indication of this situation in the Faust compiler output, but such notice is planned. For now, library contributors should strive to use only library functions having compatible licenses, and concerned end-users must manually determine the union of licenses applicable to the library functions they are using.
Dozens of functions are implemented in the Faust libraries and many of them are very specialized and not useful to beginners or to people who only need to use Faust for basic applications. This section offers an index organized by categories of the “standard Faust functions” (basic filters, effects, synthesizers, etc.). This index only contains functions without a user interface (UI). Faust functions with a built-in UI can be found in demos.lib.
| Function Type | Function Name | Description |
|---|---|---|
| Amplitude Follower | an.amp_follower |
Classic analog audio envelope follower |
| Octave Analyzers | an.mth_octave_analyzer[N] |
Octave analyzers |
| Function Type | Function Name | Description |
|---|---|---|
| Beats | ba.beat |
Pulses at a specific tempo |
| Block | si.block |
Terminate n signals |
| Break Point Function | ba.bpf |
Beak Point Function (BPF) |
| Bus | si.bus |
Bus of n signals |
| Bypass (Mono) | ba.bypass1 |
Mono bypass |
| Bypass (Stereo) | ba.bypass2 |
Stereo bypass |
| Count Elements | ba.count |
Count elements in a list |
| Count Down | ba.countdown |
Samples count down |
| Count Up | ba.countup |
Samples count up |
| Delay (Integer) | de.delay |
Integer delay |
| Delay (Float) | de.fdelay |
Fractional delay |
| Down Sample | ba.downSample |
Down sample a signal |
| Impulsify | ba.impulsify |
Turns a signal into an impulse |
| Sample and Hold | ba.sAndH |
Sample and hold |
| Signal Crossing | ro.cross |
Cross n signals |
| Smoother (Default) | si.smoo |
Exponential smoothing |
| Smoother | si.smooth |
Exponential smoothing with controllable pole |
| Take Element | ba.take |
Take en element from a list |
| Time | ba.time |
A simple timer |
| Function Type | Function Name | Description |
|---|---|---|
| dB to Linear | ba.db2linear |
Converts dB to linear values |
| Linear to dB | ba.linear2db |
Converts linear values to dB |
| MIDI Key to Hz | ba.midikey2hz |
Converts a MIDI key number into a frequency |
| Hz to MIDI Key | ba.hz2midikey |
Converts a frequency into MIDI key number |
| Pole to T60 | ba.pole2tau |
Converts a pole into a time constant (t60) |
| Samples to Seconds | ba.samp2sec |
Converts samples to seconds |
| Seconds to Samples | ba.sec2samp |
Converts seconds to samples |
| T60 to Pole | ba.tau2pole |
Converts a time constant (t60) into a pole |
| Function Type | Function Name | Description |
|---|---|---|
| Auto Wah | ve.autowah |
Auto-Wah effect |
| Compressor | co.compressor_mono |
Dynamic range compressor |
| Distortion | ef.cubicnl |
Cubic nonlinearity distortion |
| Crybaby | ve.crybaby |
Crybaby wah pedal |
| Echo | ef.echo |
Simple echo |
| Flanger | pf.flanger_stereo |
Flanging effect |
| Gate | ef.gate_mono |
Mono signal gate |
| Limiter | co.limiter_1176_R4_mono |
Limiter |
| Phaser | pf.phaser2_stereo |
Phaser effect |
| Reverb (FDN) | re.fdnrev0 |
Feedback delay network reverberator |
| Reverb (Freeverb) | re.mono_freeverb |
Most “famous” Schroeder reverberator |
| Reverb (Simple) | re.jcrev |
Simple Schroeder reverberator |
| Reverb (Zita) | re.zita_rev1_stereo |
High quality FDN reverberator |
| Panner | sp.panner |
Linear stereo panner |
| Pitch Shift | ef.transpose |
Simple pitch shifter |
| Panner | sp.spat |
N outputs spatializer |
| Speaker Simulator | ef.speakerbp |
Simple speaker simulator |
| Stereo Width | ef.stereo_width |
Stereo width effect |
| Vocoder | ve.vocoder |
Simple vocoder |
| Wah | ve.wah4 |
Wah effect |
| Function Type | Function Name | Description |
|---|---|---|
| ADSR | en.adsr |
Attack/Decay/Sustain/Release envelope generator |
| AR | en.ar |
Attack/Release envelope generator |
| ASR | en.asr |
Attack/Sustain/Release envelope generator |
| Exponential | en.smoothEnvelope |
Exponential envelope generator |
| Function Type | Function Name | Description |
|---|---|---|
| Bandpass (Butterworth) | fi.bandpass |
Generic butterworth bandpass |
| Bandpass (Resonant) | fi.resonbp |
Virtual analog resonant bandpass |
| Bandstop (Butterworth) | fi.bandstop |
Generic butterworth bandstop |
| Biquad | fi.tf2 |
“Standard” biquad filter |
| Comb (Allpass) | fi.allpass_fcomb |
Schroeder allpass comb filter |
| Comb (Feedback) | fi.fb_fcomb |
Feedback comb filter |
| Comb (Feedforward) | fi.ff_fcomb |
Feed-forward comb filter. |
| DC Blocker | fi.dcblocker |
Default dc blocker |
| Filterbank | fi.filterbank |
Generic filter bank |
| FIR (Arbitrary Order) | fi.fir |
Nth-order FIR filter |
| High Shelf | fi.high_shelf |
High shelf |
| Highpass (Butterworth) | fi.highpass |
Nth-order Butterworth highpass |
| Highpass (Resonant) | fi.resonhp |
Virtual analog resonant highpass |
| IIR (Arbitrary Order) | fi.iir |
Nth-order IIR filter |
| Level Filter | fi.levelfilter |
Dynamic level lowpass |
| Low Shelf | fi.low_shelf |
Low shelf |
| Lowpass (Butterworth) | fi.lowpass |
Nth-order Butterworth lowpass |
| Lowpass (Resonant) | fi.resonlp |
Virtual analog resonant lowpass |
| Notch Filter | fi.notchw |
Simple notch filter |
| Peak Equalizer | fi.peak_eq |
Peaking equalizer section |
| Function Type | Function Name | Description |
|---|---|---|
| Impulse | os.impulse |
Generate an impulse on start-up |
| Impulse Train | os.imptrain |
Band-limited impulse train |
| Phasor | os.phasor |
Simple phasor |
| Pink Noise | no.pink_noise |
Pink noise generator |
| Pulse Train | os.pulsetrain |
Band-limited pulse train |
| Pulse Train (Low Frequency) | os.lf_imptrain |
Low-frequency pulse train |
| Sawtooth | os.sawtooth |
Band-limited sawtooth wave |
| Sawtooth (Low Frequency) | os.lf_saw |
Low-frequency sawtooth wave |
| Sine (Filter-Based) | os.oscs |
Sine oscillator (filter-based) |
| Sine (Table-Based) | os.osc |
Sine oscillator (table-based) |
| Square | os.square |
Band-limited square wave |
| Square (Low Frequency) | os.lf_squarewave |
Low-frequency square wave |
| Triangle | os.triangle |
Band-limited triangle wave |
| Triangle (Low Frequency) | os.lf_triangle |
Low-frequency triangle wave |
| White Noise | no.noise |
White noise generator |
| Function Type | Function Name | Description |
|---|---|---|
| Additive Drum | sy.additiveDrum |
Additive synthesis drum |
| Bandpassed Sawtooth | sy.dubDub |
Sawtooth through resonant bandpass |
| Comb String | sy.combString |
String model based on a comb filter |
| FM | sy.fm |
Frequency modulation synthesizer |
| Lowpassed Sawtooth | sy.sawTrombone |
“Trombone” based on a filtered sawtooth |
| Popping Filter | sy.popFilterPerc |
Popping filter percussion instrument |
buttonCreates a button in the user interface. The button is a primitive circuit with one output and no input. The signal produced by the button is 0 when not pressed and 1 while pressed.
button("play") : _;Where "play" is the name of the button in the interface.
checkboxCreates a checkbox in the user interface. The checkbox is a primitive circuit with one output and no input. The signal produced by the checkbox is 0 when not checked and 1 when checked.
checkbox("play") : _;Where "play" is the name of the checkbox in the interface.
hsliderCreates a horizontal slider in the user interface. The hslider is a primitive circuit with one output and no input. hslider produces a signal between a minimum and a maximum value based on the position of the slider cursor.
hslider("volume",-10,-70,12,0.1) : _;Where volume is the name of the slider in the interface, -10 the default value of the slider when the program starts, -70 the minimum value, 12 the maximum value, and 0.1 the step the determines the precision of the control.
nentryCreates a numerical entry in the user interface. The nentry is a primitive circuit with one output and no input. nentry produces a signal between a minimum and a maximum value based on the user input.
nentry("volume",-10,-70,12,0.1) : _;Where volume is the name of the numerical entry in the interface, -10 the default value of the entry when the program starts, -70 the minimum value, 12 the maximum value, and 0.1 the step the determines the precision of the control.
vsliderCreates a vertical slider in the user interface. The vslider is a primitive circuit with one output and no input. vslider produces a signal between a minimum and a maximum value based on the position of the slider cursor.
vslider("volume",-10,-70,12,0.1) : _;Where volume is the name of the slider in the interface, -10 the default value of the slider when the program starts, -70 the minimum value, 12 the maximum value, and 0.1 the step the determines the precision of the control.
Faust Analyzers library. Its official prefix is an.
(an.)amp_followerClassic analog audio envelope follower with infinitely fast rise and exponential decay. The amplitude envelope instantaneously follows the absolute value going up, but then floats down exponentially. amp_follower is a standard Faust function.
_ : amp_follower(rel) : _Where:
rel: release time = amplitude-envelope time-constant (sec) going down(an.)amp_follower_udEnvelope follower with different up and down time-constants (also called a “peak detector”).
_ : amp_follower_ud(att,rel) : _Where:
att: attack time = amplitude-envelope time constant (sec) going uprel: release time = amplitude-envelope time constant (sec) going downWe assume rel >> att. Otherwise, consider rel ~ max(rel,att). For audio, att is normally faster (smaller) than rel (e.g., 0.001 and 0.01). Use amp_follower_ar below to remove this restriction.
(an.)amp_follower_arEnvelope follower with independent attack and release times. The release can be shorter than the attack (unlike in amp_follower_ud above).
_ : amp_follower_ar(att,rel) : _;Spectrum-analyzers split the input signal into a bank of parallel signals, one for each spectral band. They are related to the Mth-Octave Filter-Banks in filters.lib. The documentation of this library contains more details about the implementation. The parameters are:
M: number of band-slices per octave (>1)N: total number of bands (>2)ftop = upper bandlimit of the Mth-octave bands (<SR/2)In addition to the Mth-octave output signals, there is a highpass signal containing frequencies from ftop to SR/2, and a “dc band” lowpass signal containing frequencies from 0 (dc) up to the start of the Mth-octave bands. Thus, the N output signals are
highpass(ftop), MthOctaveBands(M,N-2,ftop), dcBand(ftop*2^(-M*(N-1)))A Spectrum-Analyzer is defined here as any band-split whose bands span the relevant spectrum, but whose band-signals do not necessarily sum to the original signal, either exactly or to within an allpass filtering. Spectrum analyzer outputs are normally at least nearly “power complementary”, i.e., the power spectra of the individual bands sum to the original power spectrum (to within some negligible tolerance).
Go to higher filter orders - see Regalia et al. or Vaidyanathan (cited below) regarding the construction of more aggressive recursive filter-banks using elliptic or Chebyshev prototype filters.
(an.)mth_octave_analyzerOctave analyzer. mth_octave_analyzer[N] are standard Faust functions.
_ : mth_octave_analyzer(O,M,ftop,N) : par(i,N,_); // Oth-order Butterworth
_ : mth_octave_analyzer6e(M,ftop,N) : par(i,N,_); // 6th-order ellipticAlso for convenience:
_ : mth_octave_analyzer3(M,ftop,N) : par(i,N,_); // 3d-order Butterworth
_ : mth_octave_analyzer5(M,ftop,N) : par(i,N,_); // 5th-roder Butterworth
mth_octave_analyzer_default = mth_octave_analyzer6e;Where:
O: order of filter used to split each frequency band into twoM: number of band-slices per octaveftop: highest band-split crossover frequency (e.g., 20 kHz)N: total number of bands (including dc and Nyquist)Spectral Level: Display (in bar graphs) the average signal level in each spectral band.
(an.)mth_octave_spectral_level6eSpectral level display.
_ : mth_octave_spectral_level6e(M,ftop,NBands,tau,dB_offset) : _;Where:
M: bands per octaveftop: lower edge frequency of top bandNBands: number of passbands (including highpass and dc bands),tau: spectral display averaging-time (time constant) in seconds,dB_offset: constant dB offset in all band level meters.Also for convenience:
mth_octave_spectral_level_default = mth_octave_spectral_level6e;
spectral_level = mth_octave_spectral_level(2,10000,20);(an.)[third|half]_octave_[analyzer|filterbank]A bunch of special cases based on the different analyzer functions described above:
third_octave_analyzer(N) = mth_octave_analyzer_default(3,10000,N);
third_octave_filterbank(N) = mth_octave_filterbank_default(3,10000,N);
half_octave_analyzer(N) = mth_octave_analyzer_default(2,10000,N);
half_octave_filterbank(N) = mth_octave_filterbank_default(2,10000,N);
octave_filterbank(N) = mth_octave_filterbank_default(1,10000,N);
octave_analyzer(N) = mth_octave_analyzer_default(1,10000,N);See mth_octave_spectral_level_demo in demos.lib.
These are similar to the Mth-octave analyzers above, except that the band-split frequencies are passed explicitly as arguments.
(an.)analyzerAnalyzer.
_ : analyzer(O,freqs) : par(i,N,_); // No delay equalizerWhere:
O: band-split filter order (ODD integer required for filterbank[i])freqs: (fc1,fc2,…,fcNs) [in numerically ascending order], where Ns=N-1 is the number of octave band-splits (total number of bands N=Ns+1).If frequencies are listed explicitly as arguments, enclose them in parens:
_ : analyzer(3,(fc1,fc2)) : _,_,_Sliding FFTs that compute a rectangularly windowed FFT each sample
(an.)fftFast Fourier Transform (FFT)
si.cbus(N) : fft(N) : si.cbus(N);Where:
si.cbus(N) is a bus of N complex signals, each specified by real and imaginary parts: (r0,i0), (r1,i1), (r2,i2), …N is the FFT size (must be a power of 2: 2,4,8,16,…)fft(N) performs a length N FFT for complex signals (radix 2)FFTs of Real Signals:
process = signal : an.rtocv(N) : an.fft(N);where an.rtocv converts a real (scalar) signal to a complex vector signal having a zero imaginary part.
See an.rfft_analyzer_c (in analyzers.lib) and related functions for more detailed usage examples.
Use an.rfft_spectral_level(N,tau,dB_offset) to display the power spectrum of a real signal.
See dm.fft_spectral_level_demo(N) in demos.lib for an example GUI driving an.rfft_spectral_level().
(an.)ifftInverse Fast Fourier Transform (IFFT)
si.cbus(N) : ifft(N) : si.cbus(N);Where:
A library of basic elements. Its official prefix is ba.
(ba.)samp2secConverts a number of samples to a duration in seconds. samp2sec is a standard Faust function.
samp2sec(n) : _Where:
n: number of samples(ba.)sec2sampConverts a duration in seconds to a number of samples. samp2sec is a standard Faust function.
sec2samp(d) : _Where:
d: duration in seconds(ba.)db2linearConverts a loudness in dB to a linear gain (0-1). db2linear is a standard Faust function.
db2linear(l) : _Where:
l: loudness in dB(ba.)linear2dbConverts a linear gain (0-1) to a loudness in dB. linear2db is a standard Faust function.
linear2db(g) : _Where:
g: a linear gain(ba.)lin2LogGainConverts a linear gain (0-1) to a log gain (0-1).
_ : lin2LogGain : _(ba.)log2LinGainConverts a log gain (0-1) to a linear gain (0-1).
_ : log2LinGain : _(ba.)tau2poleReturns a real pole giving exponential decay. Note that t60 (time to decay 60 dB) is ~6.91 time constants. tau2pole is a standard Faust function.
_ : smooth(tau2pole(tau)) : _Where:
tau: time-constant in seconds(ba.)pole2tauReturns the time-constant, in seconds, corresponding to the given real, positive pole in (0,1). pole2tau is a standard Faust function.
pole2tau(pole) : _Where:
pole: the pole(ba.)midikey2hzConverts a MIDI key number to a frequency in Hz (MIDI key 69 = A440). midikey2hz is a standard Faust function.
midikey2hz(mk) : _Where:
mk: the MIDI key number(ba.)hz2midikeyConverts a frequency in Hz to a MIDI key number (MIDI key 69 = A440). hz2midikey is a standard Faust function.
hz2midikey(f) : _Where:
f: frequency in Hz(ba.)pianokey2hzConverts a piano key number to a frequency in Hz (piano key 49 = A440).
pianokey2hz(pk) : _Where:
pk: the piano key number(ba.)hz2pianokeyConverts a frequency in Hz to a piano key number (piano key 49 = A440).
hz2pianokey(f) : _Where:
f: frequency in Hz(ba.)countdownStarts counting down from n included to 0. While trig is 1 the output is n. The countdown starts with the transition of trig from 1 to 0. At the end of the countdown the output value will remain at 0 until the next trig. countdown is a standard Faust function.
countdown(n,trig) : _Where:
count: the starting point of the countdowntrig: the trigger signal (1: start at n; 0: decrease until 0)(ba.)countupStarts counting up from 0 to n included. While trig is 1 the output is 0. The countup starts with the transition of trig from 1 to 0. At the end of the countup the output value will remain at n until the next trig. countup is a standard Faust function.
countup(n,trig) : _Where:
count: the maximum count valuetrig: the trigger signal (1: start at 0; 0: increase until n)(ba.)sweepCounts from 0 to period samples repeatedly, while run is 1. Outsputs zero while run is 0.
sweep(period,run) : _(ba.)timeA simple timer that counts every samples from the beginning of the process. time is a standard Faust function.
time : _(ba.)tempoConverts a tempo in BPM into a number of samples.
tempo(t) : _Where:
t: tempo in BPM(ba.)periodBasic sawtooth wave of period p.
period(p) : _Where:
p: period as a number of samples(ba.)pulsePulses (10000) generated at period p.
pulse(p) : _Where:
p: period as a number of samples(ba.)pulsenPulses (11110000) of length n generated at period p.
pulsen(n,p) : _Where:
n: the length of the pulse as a number of samplesp: period as a number of samples(ba.)cycleSplit nonzero input values into n cycles.
_ : cycle(n) <:Where:
n: the number of cycles/output signals(ba.)beatPulses at tempo t. beat is a standard Faust function.
beat(t) : _Where:
t: tempo in BPM(ba.)pulse_countupStarts counting up pulses. While trig is 1 the output is counting up, while trig is 0 the counter is reset to 0.
_ : pulse_countup(trig) : _Where:
trig: the trigger signal (1: start at next pulse; 0: reset to 0)(ba.)pulse_countdownStarts counting down pulses. While trig is 1 the output is counting down, while trig is 0 the counter is reset to 0.
_ : pulse_countdown(trig) : _Where:
trig: the trigger signal (1: start at next pulse; 0: reset to 0)(ba.)pulse_countup_loopStarts counting up pulses from 0 to n included. While trig is 1 the output is counting up, while trig is 0 the counter is reset to 0. At the end of the countup (n) the output value will be reset to 0.
_ : pulse_countup_loop(n,trig) : _Where:
n: the highest number of the countup (included) before reset to 0.trig: the trigger signal (1: start at next pulse; 0: reset to 0)(ba.)resetCtrFunction that lets through the mth impulse out of each consecutive group of n impulses.
_ : resetCtr(n,m) : _Where:
n: the total number of impulses being splitm: index of impulse to allow to be output(ba.)pulse_countdown_loopStarts counting down pulses from 0 to n included. While trig is 1 the output is counting down, while trig is 0 the counter is reset to 0. At the end of the countdown (n) the output value will be reset to 0.
_ : pulse_coundown_loop(n,trig) : _Where:
n: the highest number of the countup (included) before reset to 0.trig: the trigger signal (1: start at next pulse; 0: reset to 0)(ba.)countCount the number of elements of list l. count is a standard Faust function.
count(l)
count ((10,20,30,40)) -> 4Where:
l: list of elements(ba.)takeTake an element from a list. take is a standard Faust function.
take(e,l)
take(3,(10,20,30,40)) -> 30Where:
p: position (starting at 1)l: list of elements(ba.)subseqExtract a part of a list.
subseq(l, p, n)
subseq((10,20,30,40,50,60), 1, 3) -> (20,30,40)
subseq((10,20,30,40,50,60), 4, 1) -> 50Where:
l: listp: start point (0: begin of list)n: number of elementsFaust doesn’t have proper lists. Lists are simulated with parallel compositions and there is no empty list
(ba.)ifif-then-else implemented with a select2.
if(c, t, e) : _Where:
c: conditiont: signal selected while c is truee: signal selected while c is false(ba.)selectorSelects the ith input among n at compile time.
selector(i,n)
_,_,_,_ : selector(2,4) : _ // selects the 3rd input among 4Where:
i: input to select (int, numbered from 0, known at compile time)n: number of inputs (int, known at compile time, n > i)There is also cselector for selecting among complex input signals of the form (real,imag).
(ba.)selectnSelects the ith input among N at run time.
selectn(N,i)
_,_,_,_ : selectn(4,2) : _ // selects the 3rd input among 4Where:
N: number of inputs (int, known at compile time, N > 0)i: input to select (int, numbered from 0)N=64;
process = par(n,N, (par(i,N,i) : selectn(N,n)));(ba.)select2stereoSelect between 2 stereo signals.
_,_,_,_ : select2stereo(bpc) : _,_,_,_Where:
bpc: the selector switch (0/1)(ba.)latchLatch input on positive-going transition of “clock” (“sample-and-hold”).
_ : latch(clocksig) : _Where:
clocksig: hold trigger (0 for hold, 1 for bypass)(ba.)sAndHSample And Hold. sAndH is a standard Faust function.
_ : sAndH(t) : _Where:
t: hold trigger (0 for hold, 1 for bypass)(ba.)downSampleDown sample a signal. WARNING: this function doesn’t change the rate of a signal, it just holds samples… downSample is a standard Faust function.
_ : downSample(freq) : _Where:
freq: new rate in Hz(ba.)peakholdOutputs current max value above zero.
_ : peakhold(mode) : _;Where:
mode means: 0 - Pass through. A single sample 0 trigger will work as a reset. 1 - Track and hold max value.
(ba.)peakholderTracks abs peak and holds peak for ‘holdtime’ samples.
_ : peakholder(holdtime) : _;(ba.)impulsifyTurns the signal from a button into an impulse (1,0,0,… when button turns on). impulsify is a standard Faust function.
button("gate") : impulsify ;(ba.)automatRecord and replay to the values the input signal in a loop.
hslider(...) : automat(bps, size, init) : _(ba.)bpfbpf is an environment (a group of related definitions) that can be used to create break-point functions. It contains three functions :
start(x,y) to start a break-point functionend(x,y) to end a break-point functionpoint(x,y) to add intermediate points to a break-point functionA minimal break-point function must contain at least a start and an end point :
f = bpf.start(x0,y0) : bpf.end(x1,y1);A more involved break-point function can contains any number of intermediate points:
f = bpf.start(x0,y0) : bpf.point(x1,y1) : bpf.point(x2,y2) : bpf.end(x3,y3);In any case the x_{i} must be in increasing order (for all i, x_{i} < x_{i+1}). For example the following definition :
f = bpf.start(x0,y0) : ... : bpf.point(xi,yi) : ... : bpf.end(xn,yn);implements a break-point function f such that :
f(x) = y_{0} when x < x_{0}f(x) = y_{n} when x > x_{n}f(x) = y_{i} + (y_{i+1}-y_{i})*(x-x_{i})/(x_{i+1}-x_{i}) when x_{i} <= x and x < x_{i+1}bpf is a standard Faust function.
(ba.)listInterpLinearly interpolates between the elements of a list.
foo = listInterp((800,400,350,450,325),index);
i = 1.69; // range is 0-4
process = foo(i);Where:
index: the index (float) to interpolate between the different values. The range of index depends on the size of the list.(ba.)bypass1Takes a mono input signal, route it to e and bypass it if bpc = 1. bypass1 is a standard Faust function.
_ : bypass1(bpc,e) : _Where:
bpc: bypass switch (0/1)e: a mono effect(ba.)bypass2Takes a stereo input signal, route it to e and bypass it if bpc = 1. bypass2 is a standard Faust function.
_,_ : bypass2(bpc,e) : _,_Where:
bpc: bypass switch (0/1)e: a stereo effect(ba.)bypass1to2Bypass switch for effect e having mono input signal and stereo output. Effect e is bypassed if bpc = 1. bypass1to2 is a standard Faust function.
_ : bypass1(bpc,e) : _,_Where:
bpc: bypass switch (0/1)e: a mono-to-stereo effect(ba.)toggleTriggered by the change of 0 to 1, it toggles the output value between 0 and 1.
_ : toggle : _button("toggle") : toggle : vbargraph("output", 0, 1)
(an.amp_follower(0.1) > 0.01) : toggle : vbargraph("output", 0, 1) // takes audio input(ba.)on_and_offThe first channel set the output to 1, the second channel to 0.
_ , _ : on_and_off : _button("on"), button("off") : on_and_off : vbargraph("output", 0, 1)(ba.)selectoutnRoute input to the output among N at run time.
_ : selectoutn(n, s) : _,_,...nWhere:
n: number of outputs (int, known at compile time, N > 0)s: output number to route to (int, numbered from 0) (i.e. slider)process = 1 : selectoutn(3, sel) : par(i,3,bar) ;
sel = hslider("volume",0,0,2,1) : int;
bar = vbargraph("v.bargraph", 0, 1);Provides various operations on the last N samples using a high order `slidingReduce(op,N,maxN,disabledVal,x)`` fold-like function :
slidingSumN(n,maxn): the sliding sum of the last n input samplesslidingMaxN(n,maxn): the sliding max of the last n input samplesslidingMinN(n,maxn): the sliding min of the last n input samplesslidingMeanN(n,maxn): the sliding mean of the last n input samplesslidingRMSn(n,maxn): the sliding RMS of the last n input samplesIf we want the maximum of the last 8 values, we can do that as:
simpleMax(x) =
(
(
max(x@0,x@1),
max(x@2,x@3)
) :max
),
(
(
max(x@4,x@5),
max(x@6,x@7)
) :max
)
:max;max(x@2,x@3) is the same as max(x@0,x@1)@2 but the latter re-uses a value we already computed,so is more efficient. Using the same trick for values 4 trough 7, we can write:
efficientMax(x)=
(
(
max(x@0,x@1),
max(x@0,x@1)@2
) :max
),
(
(
max(x@0,x@1),
max(x@0,x@1)@2
) :max@4
)
:max;We can rewrite it recursively, so it becomes possible to get the maximum at have any number of values, as long as it’s a power of 2.
recursiveMax =
case {
(1,x) => x;
(N,x) => max(recursiveMax(N/2,x) , recursiveMax(N/2,x)@(N/2));
};What if we want to look at a number of values that’s not a power of 2? For each value, we will have to decide whether to use it or not. If N is bigger than the index of the value, we use it, otherwise we replace it with (0-(ma.INFINITY)):
variableMax(N,x) =
max(
max(
(
(x@0 : useVal(0)),
(x@1 : useVal(1))
):max,
(
(x@2 : useVal(2)),
(x@3 : useVal(3))
):max
),
max(
(
(x@4 : useVal(4)),
(x@5 : useVal(5))
):max,
(
(x@6 : useVal(6)),
(x@7 : useVal(7))
):max
)
)
with{
useVal(i) = select2( (N>=i) , (0-(ma.INFINITY)),_);
};Now it becomes impossible to re-use any values. To fix that let’s first look at how we’d implement it using recursiveMax, but with a fixed N that is not a power of 2. For example, this is how you’d do it with N=3:
binaryMaxThree(x) =
(
recursiveMax(1,x)@0, // the first x
recursiveMax(2,x)@1 // the second and third x
):max;N=6
binaryMaxSix(x) =
(
recursiveMax(2,x)@0, // first two
recursiveMax(4,x)@2 // third trough sixt
):max;Note that recursiveMax(2,x) is used at a different delay then in binaryMaxThree, since it represents 1 and 2, not 2 and 3. Each block is delayed the combined size of the previous blocks.
N=7
binaryMaxSeven(x) =
(
(
recursiveMax(1,x)@0, // first x
recursiveMax(2,x)@1 // second and third
):max,
(
recursiveMax(4,x)@3 // fourth trough seventh
)
):max;To make a variable version, we need to know which powers of two are used, and at which delay time.
Then it becomes a matter of:
par() statementsumOfPrevBlockSizes()useVal()combine()In faust, we can only do that for a fixed maximum number of values: maxN
variableBinaryMaxN(N,maxN,x) =
par(i,maxNrBits,recursiveMax(pow2(i),x)@sumOfPrevBlockSizes(N,maxN,i) : useVal(i)) : combine(maxNrBits) with {
// The sum of all the sizes of the previous blocks
sumOfPrevBlockSizes(N,maxN,0) = 0;
sumOfPrevBlockSizes(N,maxN,i) = (ba.subseq((allBlockSizes(N,maxN)),0,i):>_);
allBlockSizes(N,maxN) = par(i, maxNrBits, pow2(i) * isUsed(i) );
maxNrBits = int2nrOfBits(maxN);
// get the maximum of all blocks
combine(2) = max;
combine(N) = max(combine(N-1),_);
// Decide wether or not to use a certain value, based on N
useVal(i) = select2( isUsed(i), (0-(ma.INFINITY)),_);
isUsed(i) = ba.take(i+1,(int2bin(N,maxN)));
};(ba.)slidingReduceFold-like high order function. Apply a commutative binary operation <op> to the last <n> consecutive samples of a signal <x>. For example : slidingReduce(max,128,128,-(ma.INFINITY)) will compute the maximum of the last 128 samples. The output is updated each sample, unlike reduce, where the output is constant for the duration of a block
_ : slidingReduce(op,N,maxN,disabledVal) : _Where:
N: the number of values to processmaxN: the maximum number of values to process, needs to be a power of 2op: the operator. Needs to be a commutative one.disabledVal: the value to use when we want to ignore a value.In other words, op(x,disabledVal) should equal to x. For example, +(x,0) equals x and min(x,ma.INFINITY) equals x. So if we want to calculate the sum, we need to give 0 as disabledVal, and if we want the minimum, we need to give ma.INFINITY as disabledVal.
(ba.)slidingSumNThe sliding sum of the last n input samples.
_ : slidingSumN(N,maxN) : _Where:
N: the number of values to processmaxN: the maximum number of values to process, needs to be a power of 2(ba.)slidingMaxNThe sliding maximum of the last n input samples.
_ : slidingMaxN(N,maxN) : _Where:
N: the number of values to processmaxN: the maximum number of values to process, needs to be a power of 2(ba.)slidingSumNThe sliding minimum of the last n input samples.
_ : slidingMinN(N,maxN) : _Where:
N: the number of values to processmaxN: the maximum number of values to process, needs to be a power of 2(ba.)slidingMeanNThe sliding mean of the last n input samples.
_ : slidingMeanN(N,maxN) : _Where:
N: the number of values to processmaxN: the maximum number of values to process, needs to be a power of 2(ba.)slidingRMSnThe root mean square of the last n input samples.
_ : slidingRMSn(N,maxN) : _Where:
N: the number of values to processmaxN: the maximum number of values to process, needs to be a power of 2A library of compressor effects. Its official prefix is co.
(co.)compressor_monoMono dynamic range compressors. compressor_mono is a standard Faust function
_ : compressor_mono(ratio,thresh,att,rel) : _Where:
ratio: compression ratio (1 = no compression, >1 means compression)thresh: dB level threshold above which compression kicks in (0 dB = max level)att: attack time = time constant (sec) when level & compression going uprel: release time = time constant (sec) coming out of compression(co.)compressor_stereoStereo dynamic range compressors.
_,_ : compressor_stereo(ratio,thresh,att,rel) : _,_Where:
ratio: compression ratio (1 = no compression, >1 means compression)thresh: dB level threshold above which compression kicks in (0 dB = max level)att: attack time = time constant (sec) when level & compression going uprel: release time = time constant (sec) coming out of compression(co.)limiter_1176_R4_monoA limiter guards against hard-clipping. It can be can be implemented as a compressor having a high threshold (near the clipping level), fast attack and release, and high ratio. Since the ratio is so high, some knee smoothing is desirable (“soft limiting”). This example is intended to get you started using compressor_* as a limiter, so all parameters are hardwired to nominal values here. Ratios: 4 (moderate compression), 8 (severe compression), 12 (mild limiting), or 20 to 1 (hard limiting) Att: 20-800 MICROseconds (Note: scaled by ratio in the 1176) Rel: 50-1100 ms (Note: scaled by ratio in the 1176) Mike Shipley likes 4:1 (Grammy-winning mixer for Queen, Tom Petty, etc.) Faster attack gives “more bite” (e.g. on vocals) He hears a bright, clear eq effect as well (not implemented here) limiter_1176_R4_mono is a standard Faust function.
_ : limiter_1176_R4_mono : _;http://en.wikipedia.org/wiki/1176_Peak_Limiter
(co.)limiter_1176_R4_stereoA limiter guards against hard-clipping. It can be can be implemented as a compressor having a high threshold (near the clipping level), fast attack and release, and high ratio. Since the ratio is so high, some knee smoothing is desirable (“soft limiting”). This example is intended to get you started using compressor_* as a limiter, so all parameters are hardwired to nominal values here. Ratios: 4 (moderate compression), 8 (severe compression), 12 (mild limiting), or 20 to 1 (hard limiting) Att: 20-800 MICROseconds (Note: scaled by ratio in the 1176) Rel: 50-1100 ms (Note: scaled by ratio in the 1176) Mike Shipley likes 4:1 (Grammy-winning mixer for Queen, Tom Petty, etc.) Faster attack gives “more bite” (e.g. on vocals) He hears a bright, clear eq effect as well (not implemented here)
_,_ : limiter_1176_R4_stereo : _,_;http://en.wikipedia.org/wiki/1176_Peak_Limiter
This library contains a collection of delay functions. Its official prefix is de.
(de.)delaySimple d samples delay where n is the maximum delay length as a number of samples. Unlike the @ delay operator, here the delay signal d is explicitely bounded to the interval [0..n]. The consequence is that delay will compile even if the interval of d can’t be computed by the compiler. delay is a standard Faust function.
_ : delay(n,d) : _Where:
n: the max delay length (in samples)d: the delay length as a number of samples (integer)(de.)fdelaySimple d samples fractional delay based on 2 interpolated delay lines where n is the maximum delay length as a number of samples.
(de.)sdelays(mooth)delay: a mono delay that doesn’t click and doesn’t transpose when the delay time is changed.
_ : sdelay(N,it,dt) : _Where :
N: maximal delay in samplesit: interpolation time (in samples) for example 1024dt: delay time (in samples)(de.)fdelaylti and (de.)fdelayltvFractional delay line using Lagrange interpolation.
_ : fdelaylt[i|v](order, maxdelay, delay, inputsignal) : _Where order=1,2,3,... is the order of the Lagrange interpolation polynomial.
fdelaylti is most efficient, but designed for constant/slowly-varying delay. fdelayltv is more expensive and more robust when the delay varies rapidly.
NOTE: The requested delay should not be less than (N-1)/2.
(de.)fdelay[n]For convenience, fdelay1, fdelay2, fdelay3, fdelay4, fdelay5 are also available where n is the order of the interpolation.
Thiran Allpass Interpolation
https://ccrma.stanford.edu/~jos/pasp/Thiran_Allpass_Interpolators.html
(de.)fdelay[n]aDelay lines interpolated using Thiran allpass interpolation.
_ : fdelay[N]a(maxdelay, delay, inputsignal) : _(exactly like fdelay)
Where:
N=1,2,3, or 4 is the order of the Thiran interpolation filter, and the delay argument is at least N - 1/2.The interpolated delay should not be less than N - 1/2. (The allpass delay ranges from N - 1/2 to N + 1/2.) This constraint can be alleviated by altering the code, but be aware that allpass filters approach zero delay by means of pole-zero cancellations. The delay range [N-1/2,N+1/2] is not optimal. What is?
Delay arguments too small will produce an UNSTABLE allpass!
Because allpass interpolation is recursive, it is not as robust as Lagrange interpolation under time-varying conditions. (You may hear clicks when changing the delay rapidly.)
First-order allpass interpolation, delay d in [0.5,1.5]
This library contains a set of demo functions based on examples located in the /examples folder. Its official prefix is dm.
(dm.)mth_octave_spectral_level_demoDemonstrate mth_octave_spectral_level in a standalone GUI.
_ : mth_octave_spectral_level_demo(BandsPerOctave);
_ : spectral_level_demo : _; // 2/3 octave(dm.)parametric_eq_demoA parametric equalizer application.
_ : parametric_eq_demo : _ ;(dm.)spectral_tilt_demoA spectral tilt application.
_ : spectral_tilt_demo(N) : _ ;Where:
N: filter order (integer)All other parameters interactive
(dm.)mth_octave_filterbank_demo and (dm.)filterbank_demoGraphic Equalizer: Each filter-bank output signal routes through a fader.
_ : mth_octave_filterbank_demo(M) : _
_ : filterbank_demo : _Where:
N: number of bands per octave(dm.)cubicnl_demoDistortion demo application.
_ : cubicnl_demo : _;(dm.)gate_demoGate demo application.
_,_ : gate_demo : _,_;(dm.)compressor_demoCompressor demo application.
_,_ : compressor_demo : _,_;(dm.)moog_vcf_demoIllustrate and compare all three Moog VCF implementations above.
_ : moog_vcf_demo : _;(dm.)wah4_demoWah pedal application.
_ : wah4_demo : _;(dm.)crybaby_demoCrybaby effect application.
_ : crybaby_demo : _ ;(dm.)flanger_demoFlanger effect application.
_,_ : flanger_demo : _,_;(dm.)phaser2_demoPhaser effect demo application.
_,_ : phaser2_demo : _,_;(dm.)freeverb_demoFreeverb demo application.
_,_ : freeverb_demo : _,_;(dm.)stereo_reverb_testerHandy test inputs for reverberator demos below.
_ : stereo_reverb_tester : _(dm.)fdnrev0_demoA reverb application using fdnrev0.
_,_ : fdnrev0_demo(N,NB,BBSO) : _,_Where:
n: Feedback Delay Network (FDN) order / number of delay lines used = order of feedback matrix / 2, 4, 8, or 16 [extend primes array below for 32, 64, …]nb: Number of frequency bands / Number of (nearly) independent T60 controls / Integer 3 or greaterbbso = Butterworth band-split order / order of lowpass/highpass bandsplit used at each crossover freq / odd positive integer(dm.)zita_rev_fdn_demoReverb demo application based on zita_rev_fdn.
si.bus(8) : zita_rev_fdn_demo : si.bus(8)(dm.)zita_lightLight version of dm.zita_rev1 with only 2 UI elements.
_,_ : zita_light : _,_(dm.)zita_rev1Example GUI for zita_rev1_stereo (mostly following the Linux zita-rev1 GUI).
Only the dry/wet and output level parameters are “dezippered” here. If parameters are to be varied in real time, use smooth(0.999) or the like in the same way.
_,_ : zita_rev1 : _,_http://www.kokkinizita.net/linuxaudio/zita-rev1-doc/quickguide.html
(dm.)sawtooth_demoAn application demonstrating the different sawtooth oscillators of Faust.
sawtooth_demo : _(dm.)virtual_analog_oscillator_demoVirtual analog oscillator demo application.
virtual_analog_oscillator_demo : _(dm.)oscrs_demoSimple application demoing filter based oscillators.
oscrs_demo : _(dm.)velvet_noise_demoListen to velvet_noise!
velvet_noise_demo : _(dm.)latch_demoIllustrate latch operation
echo 'import("stdfaust.lib");' > latch_demo.dsp
echo 'process = dm.latch_demo;' >> latch_demo.dsp
faust2octave latch_demo.dsp
Octave:1> plot(faustout);(dm.)envelopes_demoIllustrate various envelopes overlaid, including their gate * 1.1
echo 'import("stdfaust.lib");' > envelopes_demo.dsp
echo 'process = dm.envelopes_demo;' >> envelopes_demo.dsp
faust2octave envelopes_demo.dsp
Octave:1> plot(faustout);(dm.)exciterPsychoacoustic harmonic exciter, with GUI.
_ : exciter : _(dm.)vocoder_demoUse example of the vocoder function where an impulse train is used as excitation.
_ : vocoder_demo : _;Yamaha DX7 emulation library. Its official prefix is dx.
(dx.)dx7_ampfDX7 amplitude conversion function. 3 versions of this function are available:
dx7_amp_bpf: BPF version (same as in the CSOUND toolkit)dx7_amp_func: estimated mathematical equivalent of dx7_amp_bpfdx7_ampf: default (sugar for dx7_amp_func)dx7AmpPreset : dx7_ampf_bpf : _Where:
dx7AmpPreset: DX7 amplitude value (0-99)(dx.)dx7_egraterisefDX7 envelope generator rise conversion function. 3 versions of this function are available:
dx7_egraterise_bpf: BPF version (same as in the CSOUND toolkit)dx7_egraterise_func: estimated mathematical equivalent of dx7_egraterise_bpfdx7_egraterisef: default (sugar for dx7_egraterise_func)dx7envelopeRise : dx7_egraterisef : _Where:
dx7envelopeRise: DX7 envelope rise value (0-99)(dx.)dx7_egraterisepercfDX7 envelope generator percussive rise conversion function. 3 versions of this function are available:
dx7_egrateriseperc_bpf: BPF version (same as in the CSOUND toolkit)dx7_egrateriseperc_func: estimated mathematical equivalent of dx7_egrateriseperc_bpfdx7_egraterisepercf: default (sugar for dx7_egrateriseperc_func)dx7envelopePercRise : dx7_egraterisepercf : _Where:
dx7envelopePercRise: DX7 envelope percussive rise value (0-99)(dx.)dx7_egratedecayfDX7 envelope generator decay conversion function. 3 versions of this function are available:
dx7_egratedecay_bpf: BPF version (same as in the CSOUND toolkit)dx7_egratedecay_func: estimated mathematical equivalent of dx7_egratedecay_bpfdx7_egratedecayf: default (sugar for dx7_egratedecay_func)dx7envelopeDecay : dx7_egratedecayf : _Where:
dx7envelopeDecay: DX7 envelope decay value (0-99)(dx.)dx7_egratedecaypercfDX7 envelope generator percussive decay conversion function. 3 versions of this function are available:
dx7_egratedecayperc_bpf: BPF version (same as in the CSOUND toolkit)dx7_egratedecayperc_func: estimated mathematical equivalent of dx7_egratedecayperc_bpfdx7_egratedecaypercf: default (sugar for dx7_egratedecayperc_func)dx7envelopePercDecay : dx7_egratedecaypercf : _Where:
dx7envelopePercDecay: DX7 envelope decay value (0-99)(dx.)dx7_eglv2peakfDX7 envelope level to peak conversion function. 3 versions of this function are available:
dx7_eglv2peak_bpf: BPF version (same as in the CSOUND toolkit)dx7_eglv2peak_func: estimated mathematical equivalent of dx7_eglv2peak_bpfdx7_eglv2peakf: default (sugar for dx7_eglv2peak_func)dx7Level : dx7_eglv2peakf : _Where:
dx7Level: DX7 level value (0-99)(dx.)dx7_velsensfDX7 velocity sensitivity conversion function.
dx7Velocity : dx7_velsensf : _Where:
dx7Velocity: DX7 level value (0-8)(dx.)dx7_fdbkscalefDX7 feedback scaling conversion function.
dx7Feedback : dx7_fdbkscalef : _Where:
dx7Feedback: DX7 feedback value(dx.)dx7_opDX7 Operator. Implements a phase-modulable sine wave oscillator connected to a DX7 envelope generator.
dx7_op(freq,phaseMod,outLev,R1,R2,R3,R4,L1,L2,L3,L4,keyVel,rateScale,type,gain,gate) : _Where:
freq: frequency of the oscillatorphaseMod: phase deviation (-1 - 1)outLev: preset output level (0-99)R1: preset envelope rate 1 (0-99)R2: preset envelope rate 2 (0-99)R3: preset envelope rate 3 (0-99)R4: preset envelope rate 4 (0-99)L1: preset envelope level 1 (0-99)L2: preset envelope level 2 (0-99)L3: preset envelope level 3 (0-99)L4: preset envelope level 4 (0-99)keyVel: preset key velocity sensitivity (0-99)rateScale: preset envelope rate scaletype: preset operator typegain: general gaingate: trigger signal(dx.)dx7_algoDX7 algorithms. Implements the 32 DX7 algorithms (a quick Google search should give your more details on this). Each algorithm uses 6 operators
dx7_algo(algN,egR1,egR2,egR3,egR4,egL1,egL2,egL3,egL4,outLevel,keyVelSens,ampModSens,opMode,opFreq,opDetune,opRateScale,feedback,lfoDelay,lfoDepth,lfoSpeed,freq,gain,gate) : _Where:
algN: algorithm number (0-31, should be an int…)egR1: preset envelope rates 1 (a list of 6 values between 0-99)egR2: preset envelope rates 2 (a list of 6 values between 0-99)egR3: preset envelope rates 3 (a list of 6 values between 0-99)egR4: preset envelope rates 4 (a list of 6 values between 0-99)egL1: preset envelope levels 1 (a list of 6 values between 0-99)egL2: preset envelope levels 2 (a list of 6 values between 0-99)egL3: preset envelope levels 3 (a list of 6 values between 0-99)egL4: preset envelope levels 4 (a list of 6 values between 0-99)outLev: preset output levels (a list of 6 values between 0-99)keyVel: preset key velocity sensitivities (a list of 6 values between 0-99)ampModSens: preset amplitude sensitivities (a list of 6 values between 0-99)opMode: preset operator mode (a list of 6 values between 0-1)opFreq: preset operator frequencies (a list of 6 values between 0-99)opDetune: preset operator detuning (a list of 6 values between 0-99)opRateScale: preset operator rate scale (a list of 6 values between 0-99)feedback: preset operator feedback (a list of 6 values between 0-99)lfoDelay: preset LFO delay (a list of 6 values between 0-99)lfoDepth: preset LFO depth (a list of 6 values between 0-99)lfoSpeed: preset LFO speed (a list of 6 values between 0-99)freq: fundamental frequencygain: general gaingate: trigger signal(dx.)dx7_uiGeneric DX7 function where all parameters are controllable using UI elements. The master-with-mute branch must be used for this function to work… This function is MIDI-compatible.
dx7_ui : _This library contains a collection of envelope generators. Its official prefix is en.
(en.)smoothEnvelopeAn envelope with an exponential attack and release. smoothEnvelope is a standard Faust function.
smoothEnvelope(ar,t) : _ar: attack and release duration (s)t: trigger signal (attack is triggered when t>0, release is triggered when t=0)(en.)arAR (Attack, Release) envelope generator (useful to create percussion envelopes). ar is a standard Faust function.
ar(a,r,t) : _Where:
a: attack (sec)r: release (sec)t: trigger signal (attack is triggered when t>0, release is triggered when t=0)(en.)arfeARFE (Attack and Release-to-Final-value Exponentially) envelope generator. Approximately equal to smoothEnvelope(Attack/6.91) when Attack == Release.
arfe(a,r,f,t) : _Where:
a, r: attack (sec), release (sec)f: final value to approach upon release (such as 0)t: trigger signal (attack is triggered when t>0, release is triggered when t=0)(en.)areARE (Attack, Release) envelope generator with Exponential segments. Approximately equal to smoothEnvelope(Attack/6.91) when Attack == Release.
are(a,r,t) : _Where:
a: attack (sec)r: release (sec)t: trigger signal (attack is triggered when t>0, release is triggered when t=0)(en.)asrASR (Attack, Sustain, Release) envelope generator. asr is a standard Faust function.
asr(a,s,r,t) : _Where:
a: attack (sec)s: sustain (percentage of t: 0-1)r: release (sec)t: trigger signal (attack is triggered when t>0, release is triggered when t=0)(en.)adsrADSR (Attack, Decay, Sustain, Release) envelope generator. adsr is a standard Faust function.
adsr(a,d,s,r,t) : _Where:
a: attack (sec)d: decay (sec)s: sustain (percentage of t: 0-1)r: release (sec)t: trigger signal (attack is triggered when t>0, release is triggered when t=0)(en.)adsreADSRE (Attack, Decay, Sustain, Release) envelope generator with Exponential segments.
adsre(a,d,s,r,g) : _Where:
a: attack (sec)d: decay (sec)s: sustain (percentage of t: 0-1)r: release (sec)t: trigger signal (attack is triggered when t>0, release is triggered when t=0)(en.)dx7envelopeDX7 operator envelope generator with 4 independent rates and levels. It is essentially a 4 points BPF.
dx7_envelope(R1,R2,R3,R4,L1,L2,L3,L4,t) : _Where:
RN: rates in secondsLN: levels (0-1)t: trigger signalFaust Filters library; Its official prefix is fi.
The Filters library is organized into 18 sections:
For more information, see ../documentation/library.pdf
(fi.)zeroOne zero filter. Difference equation: \(y(n) = x(x) - zx(n-1)\).
_ : zero(z) : _Where:
z: location of zero along real axis in z-planehttps://ccrma.stanford.edu/~jos/filters/One_Zero.html
(fi.)poleOne pole filter. Could also be called a “leaky integrator”. Difference equation: \(y(n) = x(n) + py(n-1)\).
_ : pole(p) : _Where:
p: pole location = feedback coefficienthttps://ccrma.stanford.edu/~jos/filters/One_Pole.html
(fi.)integratorSame as pole(1) [implemented separately for block-diagram clarity].
(fi.)dcblockeratDC blocker with configurable break frequency. The amplitude response is substantially flat above \(fb\), and sloped at about +6 dB/octave below \(fb\). Derived from the analog transfer function \(H(s) = \frac{s}{(s + 2 \pi fb)}\) by the low-frequency-matching bilinear transform method (i.e., the standard frequency-scaling constant 2*SR).
_ : dcblockerat(fb) : _Where:
fb: “break frequency” in Hz, i.e., -3 dB gain frequency.https://ccrma.stanford.edu/~jos/pasp/Bilinear_Transformation.html
(fi.)dcblockerDC blocker. Default dc blocker has -3dB point near 35 Hz (at 44.1 kHz) and high-frequency gain near 1.0025 (due to no scaling). dcblocker is as standard Faust function.
_ : dcblocker : _(fi.)ff_combFeed-Forward Comb Filter. Note that ff_comb requires integer delays (uses delay internally). ff_comb is a standard Faust function.
_ : ff_comb(maxdel,intdel,b0,bM) : _Where:
maxdel: maximum delay (a power of 2)intdel: current (integer) comb-filter delay between 0 and maxdeldel: current (float) comb-filter delay between 0 and maxdelb0: gain applied to delay-line inputbM: gain applied to delay-line output and then summed with inputhttps://ccrma.stanford.edu/~jos/pasp/Feedforward_Comb_Filters.html
(fi.)ff_fcombFeed-Forward Comb Filter. Note that ff_fcomb takes floating-point delays (uses fdelay internally). ff_fcomb is a standard Faust function.
_ : ff_fcomb(maxdel,del,b0,bM) : _Where:
maxdel: maximum delay (a power of 2)intdel: current (integer) comb-filter delay between 0 and maxdeldel: current (float) comb-filter delay between 0 and maxdelb0: gain applied to delay-line inputbM: gain applied to delay-line output and then summed with inputhttps://ccrma.stanford.edu/~jos/pasp/Feedforward_Comb_Filters.html
(fi.)ffcombfilterTypical special case of ff_comb() where: b0 = 1.
(fi.)fb_combFeed-Back Comb Filter (integer delay).
_ : fb_comb(maxdel,intdel,b0,aN) : _Where:
maxdel: maximum delay (a power of 2)intdel: current (integer) comb-filter delay between 0 and maxdeldel: current (float) comb-filter delay between 0 and maxdelb0: gain applied to delay-line input and forwarded to outputaN: minus the gain applied to delay-line output before summing with the input and feeding to the delay linehttps://ccrma.stanford.edu/~jos/pasp/Feedback_Comb_Filters.html
(fi.)fb_fcombFeed-Back Comb Filter (floating point delay).
_ : fb_fcomb(maxdel,del,b0,aN) : _Where:
maxdel: maximum delay (a power of 2)intdel: current (integer) comb-filter delay between 0 and maxdeldel: current (float) comb-filter delay between 0 and maxdelb0: gain applied to delay-line input and forwarded to outputaN: minus the gain applied to delay-line output before summing with the input and feeding to the delay linehttps://ccrma.stanford.edu/~jos/pasp/Feedback_Comb_Filters.html
(fi.)rev1Special case of fb_comb (rev1(maxdel,N,g)). The “rev1 section” dates back to the 1960s in computer-music reverberation. See the jcrev and brassrev in reverbs.lib for usage examples.
(fi.)fbcombfilter and (fi.)ffbcombfilterOther special cases of Feed-Back Comb Filter.
_ : fbcombfilter(maxdel,intdel,g) : _
_ : ffbcombfilter(maxdel,del,g) : _Where:
maxdel: maximum delay (a power of 2)intdel: current (integer) comb-filter delay between 0 and maxdeldel: current (float) comb-filter delay between 0 and maxdelg: feedback gainhttps://ccrma.stanford.edu/~jos/pasp/Feedback_Comb_Filters.html
(fi.)allpass_combSchroeder Allpass Comb Filter. Note that
allpass_comb(maxlen,len,aN) = ff_comb(maxlen,len,aN,1) : fb_comb(maxlen,len-1,1,aN);which is a direct-form-1 implementation, requiring two delay lines. The implementation here is direct-form-2 requiring only one delay line.
_ : allpass_comb (maxdel,intdel,aN) : _Where:
maxdel: maximum delay (a power of 2)intdel: current (integer) comb-filter delay between 0 and maxdeldel: current (float) comb-filter delay between 0 and maxdelaN: minus the feedback gain(fi.)allpass_fcombSchroeder Allpass Comb Filter. Note that
allpass_comb(maxlen,len,aN) = ff_comb(maxlen,len,aN,1) : fb_comb(maxlen,len-1,1,aN);which is a direct-form-1 implementation, requiring two delay lines. The implementation here is direct-form-2 requiring only one delay line.
allpass_fcomb is a standard Faust library.
_ : allpass_comb (maxdel,intdel,aN) : _
_ : allpass_fcomb(maxdel,del,aN) : _Where:
maxdel: maximum delay (a power of 2)intdel: current (float) comb-filter delay between 0 and maxdeldel: current (float) comb-filter delay between 0 and maxdelaN: minus the feedback gain(fi.)rev2Special case of allpass_comb (rev2(maxlen,len,g)). The “rev2 section” dates back to the 1960s in computer-music reverberation. See the jcrev and brassrev in reverbs.lib for usage examples.
(fi.)allpass_fcomb5 and (fi.)allpass_fcomb1aSame as allpass_fcomb but use fdelay5 and fdelay1a internally (Interpolation helps - look at an fft of faust2octave on
`1-1' <: allpass_fcomb(1024,10.5,0.95), allpass_fcomb5(1024,10.5,0.95);`).(fi.)iirNth-order Infinite-Impulse-Response (IIR) digital filter, implemented in terms of the Transfer-Function (TF) coefficients. Such filter structures are termed “direct form”.
iir is a standard Faust function.
_ : iir(bcoeffs,acoeffs) : _Where:
order: filter order (int) = max(#poles,#zeros)bcoeffs: (b0,b1,…,b_order) = TF numerator coefficientsacoeffs: (a1,…,a_order) = TF denominator coeffs (a0=1)https://ccrma.stanford.edu/~jos/filters/Four_Direct_Forms.html
(fi.)firFIR filter (convolution of FIR filter coefficients with a signal)
_ : fir(bv) : _fir is standard Faust function.
Where:
bv = b0,b1,…,bn is a parallel bank of coefficient signals.bv is processed using pattern-matching at compile time, so it must have this normal form (parallel signals).
Smoothing white noise with a five-point moving average:
bv = .2,.2,.2,.2,.2;
process = noise : fir(bv);Equivalent (note double parens):
process = noise : fir((.2,.2,.2,.2,.2));(fi.)conv and (fi.)convNConvolution of input signal with given coefficients.
_ : conv((k1,k2,k3,...,kN)) : _; // Argument = one signal bank
_ : convN(N,(k1,k2,k3,...)) : _; // Useful when N < count((k1,...))(fi.)tf1, (fi.)tf2 and (fi.)tf3tfN = N’th-order direct-form digital filter.
_ : tf1(b0,b1,a1) : _
_ : tf2(b0,b1,b2,a1,a2) : _
_ : tf3(b0,b1,b2,b3,a1,a2,a3) : _Where:
a: the polesb: the zeroshttps://ccrma.stanford.edu/~jos/fp/Direct_Form_I.html
(fi.)notchwSimple notch filter based on a biquad (tf2). notchw is a standard Faust function.
_ : notchw(width,freq) : _Where:
width: “notch width” in Hz (approximate)freq: “notch frequency” in Hzhttps://ccrma.stanford.edu/~jos/pasp/Phasing_2nd_Order_Allpass_Filters.html
Direct-Form Second-Order Biquad Sections
https://ccrma.stanford.edu/~jos/filters/Four_Direct_Forms.html
(fi.)tf21, (fi.)tf22, (fi.)tf22t and (fi.)tf21ttfN = N’th-order direct-form digital filter where:
tf21 is tf2, direct-form 1tf22 is tf2, direct-form 2tf22t is tf2, direct-form 2 transposedtf21t is tf2, direct-form 1 transposed_ : tf21(b0,b1,b2,a1,a2) : _
_ : tf22(b0,b1,b2,a1,a2) : _
_ : tf22t(b0,b1,b2,a1,a2) : _
_ : tf21t(b0,b1,b2,a1,a2) : _Where:
a: the polesb: the zeroshttps://ccrma.stanford.edu/~jos/fp/Direct_Form_I.html
Ladder and lattice digital filters generally have superior numerical properties relative to direct-form digital filters. They can be derived from digital waveguide filters, which gives them a physical interpretation.
(fi.)av2svCompute reflection coefficients sv from transfer-function denominator av.
sv = av2sv(av)Where:
av: parallel signal bank a1,...,aNsv: parallel signal bank s1,...,sNwhere ro = ith reflection coefficient, and ai = coefficient of z^(-i) in the filter transfer-function denominator A(z).
https://ccrma.stanford.edu/~jos/filters/Step_Down_Procedure.html (where reflection coefficients are denoted by k rather than s).
(fi.)bvav2nuvCompute lattice tap coefficients from transfer-function coefficients.
nuv = bvav2nuv(bv,av)Where:
av: parallel signal bank a1,...,aNbv: parallel signal bank b0,b1,...,aNnuv: parallel signal bank nu1,...,nuNwhere nui is the i’th tap coefficient, bi is the coefficient of z^(-i) in the filter numerator, ai is the coefficient of z^(-i) in the filter denominator
(fi.)iir_lat2Two-multiply latice IIR filter of arbitrary order.
_ : iir_lat2(bv,av) : _Where:
(fi.)allpassntTwo-multiply lattice allpass (nested order-1 direct-form-ii allpasses).
_ : allpassnt(n,sv) : _Where:
n: the order of the filtersv: the reflection coefficients (-1 1)(fi.)iir_klKelly-Lochbaum ladder IIR filter of arbitrary order.
_ : iir_kl(bv,av) : _Where:
(fi.)allpassnkltKelly-Lochbaum ladder allpass.
_ : allpassklt(n,sv) : _Where:
n: the order of the filtersv: the reflection coefficients (-1 1)(fi.)iir_lat1One-multiply latice IIR filter of arbitrary order.
_ : iir_lat1(bv,av) : _Where:
(fi.)allpassn1mtOne-multiply lattice allpass with tap lines.
_ : allpassn1mt(n,sv) : _Where:
n: the order of the filtersv: the reflection coefficients (-1 1)(fi.)iir_nlNormalized ladder filter of arbitrary order.
_ : iir_nl(bv,av) : _Where:
(fi.)allpassnnltNormalized ladder allpass filter of arbitrary order.
_ : allpassnnlt(n,sv) : _Where:
n: the order of the filtersv: the reflection coefficients (-1,1)(fi.)tf2npBiquad based on a stable second-order Normalized Ladder Filter (more robust to modulation than tf2 and protected against instability).
_ : tf2np(b0,b1,b2,a1,a2) : _Where:
a: the polesb: the zeros(fi.)wgrSecond-order transformer-normalized digital waveguide resonator.
_ : wgr(f,r) : _Where:
f: resonance frequency (Hz)r: loss factor for exponential decay (set to 1 to make a numerically stable oscillator)(fi.)nlf2Second order normalized digital waveguide resonator.
_ : nlf2(f,r) : _Where:
f: resonance frequency (Hz)r: loss factor for exponential decay (set to 1 to make a sinusoidal oscillator)https://ccrma.stanford.edu/~jos/pasp/Power_Normalized_Waveguide_Filters.html
(fi.)apnlPassive Nonlinear Allpass based on Pierce switching springs idea. Switch between allpass coefficient a1 and a2 at signal zero crossings.
_ : apnl(a1,a2) : _Where:
a1 and a2: allpass coefficientsAn allpass filter has gain 1 at every frequency, but variable phase. Ladder/lattice allpass filters are specified by reflection coefficients. They are defined here as nested allpass filters, hence the names allpassn*.
(fi.)allpassnTwo-multiply lattice - each section is two multiply-adds.
_ : allpassn(n,sv) : _n: the order of the filtersv: the reflection coefficients (-1 1)(fi.)allpassnnNormalized form - four multiplies and two adds per section, but coefficients can be time varying and nonlinear without “parametric amplification” (modulation of signal energy).
_ : allpassnn(n,tv) : _Where:
n: the order of the filtertv: the reflection coefficients (-PI PI)(fi.)allpassklKelly-Lochbaum form - four multiplies and two adds per section, but all signals have an immediate physical interpretation as traveling pressure waves, etc.
_ : allpassnkl(n,sv) : _Where:
n: the order of the filtersv: the reflection coefficients (-1 1)(fi.)allpass1mOne-multiply form - one multiply and three adds per section. Normally the most efficient in special-purpose hardware.
_ : allpassn1m(n,sv) : _Where:
n: the order of the filtersv: the reflection coefficients (-1 1)(fi.)tf2s and (fi.)tf2snpSecond-order direct-form digital filter, specified by ANALOG transfer-function polynomials B(s)/A(s), and a frequency-scaling parameter. Digitization via the bilinear transform is built in.
_ : tf2s(b2,b1,b0,a1,a0,w1) : _Where:
b2 s^2 + b1 s + b0
H(s) = --------------------
s^2 + a1 s + a0and w1 is the desired digital frequency (in radians/second) corresponding to analog frequency 1 rad/sec (i.e., s = j).
A second-order ANALOG Butterworth lowpass filter, normalized to have cutoff frequency at 1 rad/sec, has transfer function
1
H(s) = -----------------
s^2 + a1 s + 1where a1 = sqrt(2). Therefore, a DIGITAL Butterworth lowpass cutting off at SR/4 is specified as tf2s(0,0,1,sqrt(2),1,PI*SR/2);
Bilinear transform scaled for exact mapping of w1.
https://ccrma.stanford.edu/~jos/pasp/Bilinear_Transformation.html
(fi.)tf3slfAnalogous to tf2s above, but third order, and using the typical low-frequency-matching bilinear-transform constant 2/T (“lf” series) instead of the specific-frequency-matching value used in tf2s and tf1s. Note the lack of a “w1” argument.
_ : tf3slf(b3,b2,b1,b0,a3,a2,a1,a0) : _(fi.)tf1sFirst-order direct-form digital filter, specified by ANALOG transfer-function polynomials B(s)/A(s), and a frequency-scaling parameter.
tf1s(b1,b0,a0,w1)Where:
b1 s + b0H(s) = ———- s + a0
and w1 is the desired digital frequency (in radians/second) corresponding to analog frequency 1 rad/sec (i.e., s = j).
A first-order ANALOG Butterworth lowpass filter, normalized to have cutoff frequency at 1 rad/sec, has transfer function
1H(s) = ——- s + 1
so b0 = a0 = 1 and b1 = 0. Therefore, a DIGITAL first-order Butterworth lowpass with gain -3dB at SR/4 is specified as
tf1s(0,1,1,PI*SR/2); // digital half-band order 1 ButterworthBilinear transform scaled for exact mapping of w1.
https://ccrma.stanford.edu/~jos/pasp/Bilinear_Transformation.html
(fi.)tf2sbBandpass mapping of tf2s: In addition to a frequency-scaling parameter w1 (set to HALF the desired passband width in rad/sec), there is a desired center-frequency parameter wc (also in rad/s). Thus, tf2sb implements a fourth-order digital bandpass filter section specified by the coefficients of a second-order analog lowpass prototpe section. Such sections can be combined in series for higher orders. The order of mappings is (1) frequency scaling (to set lowpass cutoff w1), (2) bandpass mapping to wc, then (3) the bilinear transform, with the usual scale parameter 2*SR. Algebra carried out in maxima and pasted here.
_ : tf2sb(b2,b1,b0,a1,a0,w1,wc) : _(fi.)tf1sbFirst-to-second-order lowpass-to-bandpass section mapping, analogous to tf2sb above.
_ : tf1sb(b1,b0,a0,w1,wc) : _(fi.)resonlpSimple resonant lowpass filter based on tf2s (virtual analog). resonlp is a standard Faust function.
_ : resonlp(fc,Q,gain) : _
_ : resonhp(fc,Q,gain) : _
_ : resonbp(fc,Q,gain) : _
Where:
fc: center frequency (Hz)Q: qgain: gain (0-1)(fi.)resonhpSimple resonant highpass filters based on tf2s (virtual analog). resonhp is a standard Faust function.
_ : resonlp(fc,Q,gain) : _
_ : resonhp(fc,Q,gain) : _
_ : resonbp(fc,Q,gain) : _
Where:
fc: center frequency (Hz)Q: qgain: gain (0-1)(fi.)resonbpSimple resonant bandpass filters based on tf2s (virtual analog). resonbp is a standard Faust function.
_ : resonlp(fc,Q,gain) : _
_ : resonhp(fc,Q,gain) : _
_ : resonbp(fc,Q,gain) : _
Where:
fc: center frequency (Hz)Q: qgain: gain (0-1)(fi.)lowpassNth-order Butterworth lowpass filter. lowpass is a standard Faust function.
_ : lowpass(N,fc) : _Where:
N: filter order (number of poles) [nonnegative constant integer]fc: desired cut-off frequency (-3dB frequency) in Hzbutter function in Octave ("[z,p,g] = butter(N,1,'s');")(fi.)highpassNth-order Butterworth highpass filters. highpass is a standard Faust function.
_ : highpass(N,fc) : _Where:
N: filter order (number of poles) [nonnegative constant integer]fc: desired cut-off frequency (-3dB frequency) in Hzbutter function in Octave ("[z,p,g] = butter(N,1,'s');")(fi.)lowpass0_highpass1These special allpass filters are needed by filterbank et al. below. They are equivalent to (lowpass(N,fc) +|- highpass(N,fc))/2, but with canceling pole-zero pairs removed (which occurs for odd N).
(fi.)lowpass_plus|minus_highpassElliptic (Cauer) Lowpass Filters
ncauer and ellip in Octave(fi.)lowpass3eThird-order Elliptic (Cauer) lowpass filter.
_ : lowpass3e(fc) : _Where:
fc: -3dB frequency in HzFor spectral band-slice level display (see octave_analyzer3e):
[z,p,g] = ncauer(Rp,Rs,3); % analog zeros, poles, and gain, where
Rp = 60 % dB ripple in stopband
Rs = 0.2 % dB ripple in passband(fi.)lowpass6eSixth-order Elliptic/Cauer lowpass filter.
_ : lowpass6e(fc) : _Where:
fc: -3dB frequency in HzFor spectral band-slice level display (see octave_analyzer6e):
[z,p,g] = ncauer(Rp,Rs,6); % analog zeros, poles, and gain, where
Rp = 80 % dB ripple in stopband
Rs = 0.2 % dB ripple in passband(fi.)highpass3eThird-order Elliptic (Cauer) highpass filter. Inversion of lowpass3e wrt unit circle in s plane (s <- 1/s)
_ : highpass3e(fc) : _Where:
fc: -3dB frequency in Hz(fi.)highpass6eSixth-order Elliptic/Cauer highpass filter. Inversion of lowpass3e wrt unit circle in s plane (s <- 1/s)
_ : highpass6e(fc) : _Where:
fc: -3dB frequency in Hz(fi.)bandpassOrder 2*Nh Butterworth bandpass filter made using the transformation s <- s + wc^2/s on lowpass(Nh), where wc is the desired bandpass center frequency. The lowpass(Nh) cutoff w1 is half the desired bandpass width. bandpass is a standard Faust function.
_ : bandpass(Nh,fl,fu) : _Where:
Nh: HALF the desired bandpass order (which is therefore even)fl: lower -3dB frequency in Hzfu: upper -3dB frequency in Hz Thus, the passband width is fu-fl, and its center frequency is (fl+fu)/2.http://cnx.org/content/m16913/latest/
(fi.)bandstopOrder 2*Nh Butterworth bandstop filter made using the transformation s <- s + wc^2/s on highpass(Nh), where wc is the desired bandpass center frequency. The highpass(Nh) cutoff w1 is half the desired bandpass width. bandstop is a standard Faust function.
_ : bandstop(Nh,fl,fu) : _Where:
Nh: HALF the desired bandstop order (which is therefore even)fl: lower -3dB frequency in Hzfu: upper -3dB frequency in Hz Thus, the passband (stopband) width is fu-fl, and its center frequency is (fl+fu)/2.http://cnx.org/content/m16913/latest/
(fi.)bandpass6eOrder 12 elliptic bandpass filter analogous to bandpass(6).
(fi.)bandpass12eOrder 24 elliptic bandpass filter analogous to bandpass(6).
Parametric Equalizers (Shelf, Peaking)
(fi.)low_shelfFirst-order “low shelf” filter (gain boost|cut between dc and some frequency) low_shelf is a standard Faust function.
_ : lowshelf(N,L0,fx) : _
_ : low_shelf(L0,fx) : _ // default case (order 3)
_ : lowshelf_other_freq(N,L0,fx) : _Where: * N: filter order 1, 3, 5, … (odd only). (default should be 3) * L0: desired level (dB) between dc and fx (boost L0>0 or cut L0<0) * fx: -3dB frequency of lowpass band (L0>0) or upper band (L0<0) (see “SHELF SHAPE” below).
The gain at SR/2 is constrained to be 1. The generalization to arbitrary odd orders is based on the well known fact that odd-order Butterworth band-splits are allpass-complementary (see filterbank documentation below for references).
The magnitude frequency response is approximately piecewise-linear on a log-log plot (“BODE PLOT”). The Bode “stick diagram” approximation L(lf) is easy to state in dB versus dB-frequency lf = dB(f):
See lowshelf_other_freq.
(fi.)high_shelfFirst-order “high shelf” filter (gain boost|cut above some frequency). high_shelf is a standard Faust function.
_ : highshelf(N,Lpi,fx) : _
_ : high_shelf(L0,fx) : _ // default case (order 3)
_ : highshelf_other_freq(N,Lpi,fx) : _Where:
N: filter order 1, 3, 5, … (odd only).Lpi: desired level (dB) between fx and SR/2 (boost Lpi>0 or cut Lpi<0)fx: -3dB frequency of highpass band (L0>0) or lower band (L0<0) (Use highshelf_other_freq() below to find the other one.)The gain at dc is constrained to be 1. See lowshelf documentation above for more details on shelf shape.
(fi.)peak_eqSecond order “peaking equalizer” section (gain boost or cut near some frequency) Also called a “parametric equalizer” section. peak_eq is a standard Faust function.
_ : peak_eq(Lfx,fx,B) : _;Where:
Lfx: level (dB) at fx (boost Lfx>0 or cut Lfx<0)fx: peak frequency (Hz)B: bandwidth (B) of peak in Hz(fi.)peak_eq_cqConstant-Q second order peaking equalizer section.
_ : peak_eq_cq(Lfx,fx,Q) : _;Where:
Lfx: level (dB) at fxfx: boost or cut frequency (Hz)Q: “Quality factor” = fx/B where B = bandwidth of peak in Hz(fi.)peak_eq_rmRegalia-Mitra second order peaking equalizer section
_ : peak_eq_rm(Lfx,fx,tanPiBT) : _;Where:
Lfx: level (dB) at fxfx: boost or cut frequency (Hz)tanPiBT: tan(PI*B/SR), where B = -3dB bandwidth (Hz) when 10^(Lfx/20) = 0 ~ PI*B/SR for narrow bandwidths BP.A. Regalia, S.K. Mitra, and P.P. Vaidyanathan, “The Digital All-Pass Filter: A Versatile Signal Processing Building Block” Proceedings of the IEEE, 76(1):19-37, Jan. 1988. (See pp. 29-30.)
(fi.)spectral_tiltSpectral tilt filter, providing an arbitrary spectral rolloff factor alpha in (-1,1), where -1 corresponds to one pole (-6 dB per octave), and +1 corresponds to one zero (+6 dB per octave). In other words, alpha is the slope of the ln magnitude versus ln frequency. For a “pinking filter” (e.g., to generate 1/f noise from white noise), set alpha to -1/2.
_ : spectral_tilt(N,f0,bw,alpha) : _Where:
N: desired integer filter order (fixed at compile time)f0: lower frequency limit for desired roll-off band > 0bw: bandwidth of desired roll-off bandalpha: slope of roll-off desired in nepers per neper, between -1 and 1 (ln mag / ln radian freq)See spectral_tilt_demo.
J.O. Smith and H.F. Smith, “Closed Form Fractional Integration and Differentiation via Real Exponentially Spaced Pole-Zero Pairs”, arXiv.org publication arXiv:1606.06154 [cs.CE], June 7, 2016, http://arxiv.org/abs/1606.06154
(fi.)levelfilterDynamic level lowpass filter. levelfilter is a standard Faust function.
_ : levelfilter(L,freq) : _Where:
L: desired level (in dB) at Nyquist limit (SR/2), e.g., -60freq: corner frequency (-3dB point) usually set to fundamental freqN: Number of filters in series where L = L/Nhttps://ccrma.stanford.edu/realsimple/faust_strings/Dynamic_Level_Lowpass_Filter.html
(fi.)levelfilterNDynamic level lowpass filter.
_ : levelfilterN(N,freq,L) : _Where:
L: desired level (in dB) at Nyquist limit (SR/2), e.g., -60freq: corner frequency (-3dB point) usually set to fundamental freqN: Number of filters in series where L = L/Nhttps://ccrma.stanford.edu/realsimple/faust_strings/Dynamic_Level_Lowpass_Filter.html
Mth-octave filter-banks split the input signal into a bank of parallel signals, one for each spectral band. They are related to the Mth-Octave Spectrum-Analyzers in analysis.lib. The documentation of this library contains more details about the implementation. The parameters are:
M: number of band-slices per octave (>1)N: total number of bands (>2)ftop: upper bandlimit of the Mth-octave bands (<SR/2)In addition to the Mth-octave output signals, there is a highpass signal containing frequencies from ftop to SR/2, and a “dc band” lowpass signal containing frequencies from 0 (dc) up to the start of the Mth-octave bands. Thus, the N output signals are
highpass(ftop), MthOctaveBands(M,N-2,ftop), dcBand(ftop*2^(-M*(N-1)))A Filter-Bank is defined here as a signal bandsplitter having the property that summing its output signals gives an allpass-filtered version of the filter-bank input signal. A more conventional term for this is an “allpass-complementary filter bank”. If the allpass filter is a pure delay (and possible scaling), the filter bank is said to be a “perfect-reconstruction filter bank” (see Vaidyanathan-1993 cited below for details). A “graphic equalizer”, in which band signals are scaled by gains and summed, should be based on a filter bank.
The filter-banks below are implemented as Butterworth or Elliptic spectrum-analyzers followed by delay equalizers that make them allpass-complementary.
Go to higher filter orders - see Regalia et al. or Vaidyanathan (cited below) regarding the construction of more aggressive recursive filter-banks using elliptic or Chebyshev prototype filters.
(fi.)mth_octave_filterbank[n]Allpass-complementary filter banks based on Butterworth band-splitting. For Butterworth band-splits, the needed delay equalizer is easily found.
_ : mth_octave_filterbank(O,M,ftop,N) : par(i,N,_); // Oth-order
_ : mth_octave_filterbank_alt(O,M,ftop,N) : par(i,N,_); // dc-inverted versionAlso for convenience:
_ : mth_octave_filterbank3(M,ftop,N) : par(i,N,_); // 3rd-order Butterworth
_ : mth_octave_filterbank5(M,ftop,N) : par(i,N,_); // 5th-order Butterworth
mth_octave_filterbank_default = mth_octave_filterbank5;Where:
O: order of filter used to split each frequency band into twoM: number of band-slices per octaveftop: highest band-split crossover frequency (e.g., 20 kHz)N: total number of bands (including dc and Nyquist)These are similar to the Mth-octave analyzers above, except that the band-split frequencies are passed explicitly as arguments.
(fi.)filterbankFilter bank. filterbank is a standard Faust function.
_ : filterbank (O,freqs) : par(i,N,_); // Butterworth band-splitsWhere:
O: band-split filter order (ODD integer required for filterbank[i])freqs: (fc1,fc2,…,fcNs) [in numerically ascending order], where Ns=N-1 is the number of octave band-splits (total number of bands N=Ns+1).If frequencies are listed explicitly as arguments, enclose them in parens:
_ : filterbank(3,(fc1,fc2)) : _,_,_(fi.)filterbankiInverted-dc filter bank.
_ : filterbanki(O,freqs) : par(i,N,_); // Inverted-dc versionWhere:
O: band-split filter order (ODD integer required for filterbank[i])freqs: (fc1,fc2,…,fcNs) [in numerically ascending order], where Ns=N-1 is the number of octave band-splits (total number of bands N=Ns+1).If frequencies are listed explicitly as arguments, enclose them in parens:
_ : filterbanki(3,(fc1,fc2)) : _,_,_Faust library for high order ambisonic. Its official prefix is ho.
(ho.)encoderAmbisonic encoder. Encodes a signal in the circular harmonics domain depending on an order of decomposition and an angle.
encoder(n, x, a) : _Where:
n: the orderx: the signala: the angle(ho.)decoderDecodes an ambisonics sound field for a circular array of loudspeakers.
_ : decoder(n, p) : _Where:
n: the orderp: the number of speakersNumber of loudspeakers must be greater or equal to 2n+1. It’s preferable to use 2n+2 loudspeakers.
(ho.)decoderStereoDecodes an ambisonic sound field for stereophonic configuration. An “home made” ambisonic decoder for stereophonic restitution (30° - 330°) : Sound field lose energy around 180°. You should use inPhase optimization with ponctual sources. #### Usage
_ : decoderStereo(n) : _Where:
n: the orderFunctions to weight the circular harmonics signals depending to the ambisonics optimization. It can be basic for no optimization, maxRe or inPhase.
(ho.)optimBasicThe basic optimization has no effect and should be used for a perfect circle of loudspeakers with one listener at the perfect center loudspeakers array.
_ : optimBasic(n) : _Where:
n: the order(ho.)optimMaxReThe maxRe optimization optimize energy vector. It should be used for an auditory confined in the center of the loudspeakers array.
_ : optimMaxRe(n) : _Where:
n: the order(ho.)optimInPhaseThe inPhase Optimization optimize energy vector and put all loudspeakers signals n phase. It should be used for an auditory.
here:
n: the order
(ho.)widerCan be used to wide the diffusion of a localized sound. The order depending signals are weighted and appear in a logarithmic way to have linear changes.
_ : wider(n,w) : _Where:
n: the orderw: the width value between 0 - 1(ho.)mapIt simulate the distance of the source by applying a gain on the signal and a wider processing on the soundfield.
map(n, x, r, a)Where:
n: the orderx: the signalr: the radiusa: the angle in radian(ho.)rotateRotates the sound field.
_ : rotate(n, a) : _Where:
n: the ordera: the angle in radianMathematic library for Faust. Its official prefix is ma.
(ma.)SRCurrent sampling rate (between 1000Hz and 192000Hz). Constant during program execution.
SR : _(ma.)BSCurrent block-size. Can change during the execution.
BS : _(ma.)PIConstant PI in double precision.
PI : _(ma.)INFINITYConstant INFINITY inherited from math.h.
INFINITY : _(ma.)FTZFlush to zero: force samples under the “maximum subnormal number” to be zero. Usually not needed in C++ because the architecture file take care of this, but can be useful in javascript for instance.
_ : ftz : _See : http://docs.oracle.com/cd/E19957-01/806-3568/ncg_math.html
(ma.)negInvert the sign (-x) of a signal.
_ : neg : _(ma.)sub(x,y)Subtract x and y.
(ma.)invCompute the inverse (1/x) of the input signal.
_ : inv : _(ma.)cbrtComputes the cube root of of the input signal.
_ : cbrt : _(ma.)hypotComputes the euclidian distance of the two input signals sqrt(xx+yy) without undue overflow or underflow.
_,_ : hypot : _(ma.)ldexpTakes two input signals: x and n, and multiplies x by 2 to the power n.
_,_ : ldexp : _(ma.)scalbTakes two input signals: x and n, and multiplies x by 2 to the power n.
_,_ : scalb : _(ma.)log1pComputes log(1 + x) without undue loss of accuracy when x is nearly zero.
_ : log1p : _(ma.)logbReturn exponent of the input signal as a floating-point number.
_ : logb : _(ma.)ilogbReturn exponent of the input signal as an integer number.
_ : ilogb : _(ma.)log2Returns the base 2 logarithm of x.
_ : log2 : _(ma.)expm1Return exponent of the input signal minus 1 with better precision.
_ : expm1 : _(ma.)acoshComputes the principle value of the inverse hyperbolic cosine of the input signal.
_ : acosh : _(ma.)asinhComputes the inverse hyperbolic sine of the input signal.
_ : asinh : _(ma.)atanhComputes the inverse hyperbolic tangent of the input signal.
_ : atanh : _(ma.)sinhComputes the hyperbolic sine of the input signal.
_ : sinh : _(ma.)coshComputes the hyperbolic cosine of the input signal.
_ : cosh : _(ma.)tanhComputes the hyperbolic tangent of the input signal.
_ : tanh : _(ma.)erfComputes the error function of the input signal.
_ : erf : _(ma.)erfcComputes the complementary error function of the input signal.
_ : erfc : _(ma.)gammaComputes the gamma function of the input signal.
_ : gamma : _(ma.)lgammaCalculates the natural logorithm of the absolute value of the gamma function of the input signal.
_ : lgamma : _(ma.)J0Computes the Bessel function of the first kind of order 0 of the input signal.
_ : J0 : _(ma.)J1Computes the Bessel function of the first kind of order 1 of the input signal.
_ : J1 : _(ma.)JnComputes the Bessel function of the first kind of order n (first input signal) of the second input signal.
_,_ : Jn : _(ma.)Y0Computes the linearly independent Bessel function of the second kind of order 0 of the input signal.
_ : Y0 : _(ma.)Y1Computes the linearly independent Bessel function of the second kind of order 1 of the input signal.
_ : Y0 : _(ma.)YnComputes the linearly independent Bessel function of the second kind of order n (first input signal) of the second input signal.
_,_ : Yn : _(ma.)fabs, (ma.)fmax, (ma.)fminJust for compatibility…
fabs = abs
fmax = max
fmin = min(ma.)np2Gives the next power of 2 of x.
np2(n) : _Where:
n: an integer(ma.)fracGives the fractional part of n.
frac(n) : _Where:
n: a decimal number(ma.)moduloModulus operation.
modulo(x,N) : _Where:
x: the numeratorN: the denominator(ma.)isnanReturn non-zero if and only if x is a NaN.
isnan(x)
_ : isnan : _Where:
x: signal to analyse(ma.)chebychevChebychev transformation of order n.
_ : chebychev(n) : _Where:
n: the order of the polynomialT[0](x) = 1,
T[1](x) = x,
T[n](x) = 2x*T[n-1](x) - T[n-2](x)http://en.wikipedia.org/wiki/Chebyshev_polynomial
(ma.)chebychevpolyLinear combination of the first Chebyshev polynomials.
_ : chebychevpoly((c0,c1,...,cn)) : _Where:
cn: the different Chebychevs polynomials such that: chebychevpoly((c0,c1,…,cn)) = Sum of chebychev(i)*cihttp://www.csounds.com/manual/html/chebyshevpoly.html
(ma.)diffnNegated first-order difference.
_ : diffn : _(ma.)signumThe signum function signum(x) is defined as -1 for x<0, 0 for x==0, and 1 for x>0;
_ : signum : _This library contains a collection of audio effects. Its official prefix is ef.
(ef.)cubicnlCubic nonlinearity distortion. cubicnl is a standard Faust library.
_ : cubicnl(drive,offset) : _
_ : cubicnl_nodc(drive,offset) : _Where:
drive: distortion amount, between 0 and 1offset: constant added before nonlinearity to give even harmonics. Note: offset can introduce a nonzero mean - feed cubicnl output to dcblocker to remove this.(ef.)gate_monoMono signal gate. gate_mono is a standard Faust function.
_ : gate_mono(thresh,att,hold,rel) : _Where:
thresh: dB level threshold above which gate opens (e.g., -60 dB)att: attack time = time constant (sec) for gate to open (e.g., 0.0001 s = 0.1 ms)hold: hold time = time (sec) gate stays open after signal level < thresh (e.g., 0.1 s)rel: release time = time constant (sec) for gate to close (e.g., 0.020 s = 20 ms)(ef.)gate_stereoStereo signal gates. gate_stereo is a standard Faust function.
_,_ : gate_stereo(thresh,att,hold,rel) : _,_Where:
thresh: dB level threshold above which gate opens (e.g., -60 dB)att: attack time = time constant (sec) for gate to open (e.g., 0.0001 s = 0.1 ms)hold: hold time = time (sec) gate stays open after signal level < thresh (e.g., 0.1 s)rel: release time = time constant (sec) for gate to close (e.g., 0.020 s = 20 ms)(ef.)speakerbpDirt-simple speaker simulator (overall bandpass eq with observed roll-offs above and below the passband).
Low-frequency speaker model = +12 dB/octave slope breaking to flat near f1. Implemented using two dc blockers in series.
High-frequency model = -24 dB/octave slope implemented using a fourth-order Butterworth lowpass.
Example based on measured Celestion G12 (12" speaker):
speakerbp is a standard Faust function
speakerbp(f1,f2)
_ : speakerbp(130,5000) : _(ef.)piano_dispersion_filterPiano dispersion allpass filter in closed form.
piano_dispersion_filter(M,B,f0)
_ : piano_dispersion_filter(1,B,f0) : +(totalDelay),_ : fdelay(maxDelay) : _Where:
M: number of first-order allpass sections (compile-time only) Keep below 20. 8 is typical for medium-sized piano strings.B: string inharmonicity coefficient (0.0001 is typical)f0: fundamental frequency in Hzf0 of allpass chain in samples, provided in negative form to facilitate subtraction from delay-line length.(ef.)stereo_widthStereo Width effect using the Blumlein Shuffler technique. stereo_width is a standard Faust function.
_,_ : stereo_width(w) : _,_Where:
w: stereo width between 0 and 1At w=0, the output signal is mono ((left+right)/2 in both channels). At w=1, there is no effect (original stereo image). Thus, w between 0 and 1 varies stereo width from 0 to “original”.
(ef.)echoA simple echo effect.
echo is a standard Faust function
_ : echo(maxDuration,duration,feedback) : _Where:
maxDuration: the max echo duration in secondsduration: the echo duration in secondsfeedback: the feedback coefficient(ef.)transposeA simple pitch shifter based on 2 delay lines. transpose is a standard Faust function.
_ : transpose(w, x, s) : _Where:
w: the window length (samples)x: crossfade duration duration (samples)s: shift (semitones)(ef.)mesh_squareSquare Rectangular Digital Waveguide Mesh.
bus(4*N) : mesh_square(N) : bus(4*N);Where:
N: number of nodes along each edge - a power of two (1,2,4,8,…)https://ccrma.stanford.edu/~jos/pasp/Digital_Waveguide_Mesh.html
The mesh is constructed recursively using 2x2 embeddings. Thus, the top level of mesh_square(M) is a block 2x2 mesh, where each block is a mesh(M/2). Let these blocks be numbered 1,2,3,4 in the geometry NW,NE,SW,SE, i.e., as 1 2 3 4 Each block has four vector inputs and four vector outputs, where the length of each vector is M/2. Label the input vectors as Ni,Ei,Wi,Si, i.e., as the inputs from the North, East South, and West, and similarly for the outputs. Then, for example, the upper left input block of M/2 signals is labeled 1Ni. Most of the connections are internal, such as 1Eo -> 2Wi. The 8*(M/2) input signals are grouped in the order 1Ni 2Ni 3Si 4Si 1Wi 3Wi 2Ei 4Ei and the output signals are 1No 1Wo 2No 2Eo 3So 3Wo 4So 4Eo or
In: 1No 1Wo 2No 2Eo 3So 3Wo 4So 4Eo
Out: 1Ni 2Ni 3Si 4Si 1Wi 3Wi 2Ei 4Ei
Thus, the inputs are grouped by direction N,S,W,E, while the outputs are grouped by block number 1,2,3,4, which can also be interpreted as directions NW, NE, SW, SE. A simple program illustrating these orderings is process = mesh_square(2);.
Reflectively terminated mesh impulsed at one corner:
mesh_square_test(N,x) = mesh_square(N)~(busi(4*N,x)) // input to corner
with { busi(N,x) = bus(N) : par(i,N,*(-1)) : par(i,N-1,_), +(x); };
process = 1-1' : mesh_square_test(4); // all modes excited foreverIn this simple example, the mesh edges are connected as follows:
1No -> 1Ni, 1Wo -> 2Ni, 2No -> 3Si, 2Eo -> 4Si,
3So -> 1Wi, 3Wo -> 3Wi, 4So -> 2Ei, 4Eo -> 4Ei
A routing matrix can be used to obtain other connection geometries.
Faust Noise Generator Library. Its official prefix is no.
(no.)noiseWhite noise generator (outputs random number between -1 and 1). Noise is a standard Faust function.
noise : _(no.)multirandomGenerates multiple decorrelated random numbers in parallel.
multirandom(n) : si.bus(n)Where:
n: the number of decorrelated random numbers in parallel(no.)multinoiseGenerates multiple decorrelated noises in parallel.
multinoise(n) : si.bus(n)Where:
n: the number of decorrelated random numbers in parallel(no.)noisesTODO.
(no.)pink_noisePink noise (1/f noise) generator (third-order approximation) pink_noise is a standard Faust function.
pink_noise : _;https://ccrma.stanford.edu/~jos/sasp/Example_Synthesis_1_F_Noise.html
(no.)pink_noise_vmMulti pink noise generator.
pink_noise_vm(N) : _;Where:
N: number of latched white-noise processes to sum, not to exceed sizeof(int) in C++ (typically 32).(no.)lfnoise, (no.)lfnoise0 and (no.)lfnoiseNLow-frequency noise generators (Butterworth-filtered downsampled white noise)
lfnoise0(rate) : _; // new random number every int(SR/rate) samples or so
lfnoiseN(N,rate) : _; // same as "lfnoise0(rate) : lowpass(N,rate)" [see filters.lib]
lfnoise(rate) : _; // same as "lfnoise0(rate) : seq(i,5,lowpass(N,rate))" (no overshoot)(view waveforms in faust2octave):
rate = SR/100.0; // new random value every 100 samples (SR from music.lib)
process = lfnoise0(rate), // sampled/held noise (piecewise constant)
lfnoiseN(3,rate), // lfnoise0 smoothed by 3rd order Butterworth LPF
lfnoise(rate); // lfnoise0 smoothed with no overshoot(no.)sparse_noise_vmsparse noise generator.
sparse_noise(f0) : _;Where:
f0: average frequency of noise impulses per secondRandom impulses in the amplitude range -1 to 1 are generated at an average rate of f0 impulses per second.
(no.)velvet_noise_vmvelvet noise generator.
velvet_noise(amp,f0) : _;Where:
amp: amplitude of noise impulses (positive and negative)f0: average frequency of noise impulses per second(no.)gnoiseapproximate zero-mean, unit-variance Gaussian white noise generator
gnoise(N) : _;Where:
N: number of uniform random numbers added to approximate Gaussian white noiseThis library contains a collection of sound generators. Its official prefix is os.
(os.)sinwaveformSine waveform ready to use with a rdtable.
sinwaveform(tablesize) : _Where:
tablesize: the table size(os.)coswaveformCosine waveform ready to use with a rdtable.
coswaveform(tablesize) : _Where:
tablesize: the table size(os.)phasorA simple phasor to be used with a rdtable. phasor is a standard Faust function.
phasor(tablesize,freq) : _Where:
tablesize: the table sizefreq: the frequency of the wave (Hz)(os.)hs_phasorHardsyncing phasor to be used with an rdtable.
hs_phasor(ts,freq,c) : _Where:
ts: the tablesize for the related sine wavetablefreq: the fundamental frequency of the phasorc: a clock signal, c>0 resets phase to 0(os.)oscsinSine wave oscillator. oscsin is a standard Faust function.
oscsin(freq) : _Where:
freq: the frequency of the wave (Hz)(os.)hs_oscsinSin lookup table with hardsyncing phase.
hs_oscsin(freq,c) : _Where:
freq: the fundamental frequency of the phasorc: a clock signal, c>0 resets phase to 0(os.)osccosCosine wave oscillator.
osccos(freq) : _Where:
freq: the frequency of the wave (Hz)(os.)oscpA sine wave generator with controllable phase.
oscp(freq,p) : _Where:
freq: the frequency of the wave (Hz)p: the phase in radian(os.)osciInterpolated phase sine wave oscillator.
osci(freq) : _Where:
freq: the frequency of the wave (Hz)Low-Frequency Oscillators (LFOs) have prefix lf_ (no aliasing suppression, which is not audible at LF).
(os.)lf_imptrainUnit-amplitude low-frequency impulse train. lf_imptrain is a standard Faust function.
lf_imptrain(freq) : _Where:
freq: frequency in Hz(os.)lf_pulsetrainposUnit-amplitude nonnegative LF pulse train, duty cycle between 0 and 1
lf_pulsetrainpos(freq,duty) : _Where:
freq: frequency in Hzduty: duty cycle between 0 and 1(os.)lf_pulsetrainUnit-amplitude zero-mean LF pulse train, duty cycle between 0 and 1
lf_pulsetrain(freq,duty) : _Where:
freq: frequency in Hzduty: duty cycle between 0 and 1(os.)lf_squarewaveposPositive LF square wave in [0,1]
lf_squarewavepos(freq) : _Where:
freq: frequency in Hz(os.)lf_squarewaveZero-mean unit-amplitude LF square wave. lf_squarewave is a standard Faust function.
lf_squarewave(freq) : _Where:
freq: frequency in Hz(os.)lf_triangleposPositive unit-amplitude LF positive triangle wave
lf_trianglepos(freq) : _Where:
freq: frequency in Hz(os.)lf_trianglePositive unit-amplitude LF triangle wave lf_triangle is a standard Faust function.
lf_triangle(freq) : _Where:
freq: frequency in HzSawtooth waveform oscillators for virtual analog synthesis et al. The ‘simple’ versions (lf_rawsaw, lf_sawpos and saw1), are mere samplings of the ideal continuous-time (“analog”) waveforms. While simple, the aliasing due to sampling is quite audible. The differentiated polynomial waveform family (saw2, sawN, and derived functions) do some extra processing to suppress aliasing (not audible for very low fundamental frequencies). According to Lehtonen et al. (JASA 2012), the aliasing of saw2 should be inaudible at fundamental frequencies below 2 kHz or so, for a 44.1 kHz sampling rate and 60 dB SPL presentation level; fundamentals 415 and below required no aliasing suppression (i.e., saw1 is ok).
(os.)lf_rawsawSimple sawtooth waveform oscillator between 0 and period in samples.
lf_rawsaw(periodsamps)Where:
periodsamps: number of periods per samples(os.)lf_sawpos_phaseSimple sawtooth waveform oscillator between 0 and 1 with phase control.
lf_sawpos_phase(freq,phase)Where:
freq: frequencyphase: phase(os.)lf_sawposSimple sawtooth waveform oscillator between 0 and 1.
lf_sawpos(freq)Where:
freq: frequency(os.)lf_sawSimple sawtooth waveform. lf_saw is a standard Faust function.
lf_saw(freq)Where:
freq: frequency//——————(os.)sawN——————————– Bandlimited Sawtooth
sawN(N,freq), sawNp, saw2dpw(freq), saw2(freq), saw3(freq), saw4(freq), saw5(freq), saw6(freq), sawtooth(freq), saw2f2(freq) saw2f4(freq)
saw2)Polynomial Transition Regions (PTR) (for aliasing suppression)
sawN)Differentiated Polynomial Waves (DPW) (for aliasing suppression)
“Alias-Suppressed Oscillators based on Differentiated Polynomial Waveforms”, Vesa Valimaki, Juhan Nam, Julius Smith, and Jonathan Abel, IEEE Tr. Acoustics, Speech, and Language Processing (IEEE-ASLP), Vol. 18, no. 5, May 2010.
Correction-filtered versions of saw2: saw2f2, saw2f4 The correction filter compensates “droop” near half the sampling rate. See reference for sawN.
sawN(N,freq) : _
sawNp(N,freq,phase) : _
saw2dpw(freq) : _
saw2(freq) : _
saw3(freq) : _ // based on sawN
saw4(freq) : _ // based on sawN
saw5(freq) : _ // based on sawN
saw6(freq) : _ // based on sawN
sawtooth(freq) : _ // = saw2
saw2f2(freq) : _
saw2f4(freq) : _Where:
N: polynomial orderfreq: frequency in Hzphase: phase(os.)sawNpTODO: MarkDown doc in comments
(os.)saw2dpwTODO: MarkDown doc in comments
(os.)saw3TODO: MarkDown doc in comments
(os.)sawtoothAlias-free sawtooth wave. 2nd order interpolation (based on saw2). sawtooth is a standard Faust function.
sawtooth(freq) : _Where:
freq: frequency(os.)saw2f2TODO: MarkDown doc in comments
(os.)saw2f4TODO: MarkDown doc in comments
Bandlimited Pulse, Square, and Impulse Trains
pulsetrainN, pulsetrain, squareN, square, imptrain, imptrainN, triangle, triangleN
All are zero-mean and meant to oscillate in the audio frequency range. Use simpler sample-rounded lf_* versions above for LFOs.
pulsetrainN(N,freq,duty) : _
pulsetrain(freq, duty) : _ // = pulsetrainN(2)
squareN(N, freq) : _
square : _ // = squareN(2)
imptrainN(N,freq) : _
imptrain : _ // = imptrainN(2)
triangleN(N,freq) : _
triangle : _ // = triangleN(2)Where:
N: polynomial orderfreq: frequency in Hz(os.)pulsetrainNTODO: MarkDown doc in comments
(os.)pulsetrainBandlimited pulse train oscillator. Based on pulsetrainN(2). pulsetrain is a standard Faust function.
pulsetrain(freq, duty) : _Where:
freq: frequencyduty: duty cycle between 0 and 1(os.)squareNTODO: MarkDown doc in comments
(os.)squareBandlimited square wave oscillator. Based on squareN(2). square is a standard Faust function.
square(freq) : _Where:
freq: frequency(os.)impulseOne-time impulse generated when the Faust process is started. impulse is a standard Faust function.
impulse : _(os.)imptrainNTODO: MarkDown doc in comments
(os.)imptrainBandlimited impulse train generator. Based on imptrainN(2). imptrain is a standard Faust function.
imptrain(freq) : _Where:
freq: frequency(os.)triangleNTODO: MarkDown doc in comments
(os.)triangleBandlimited triangle wave oscillator. Based on triangleN(2). triangle is a standard Faust function.
triangle(freq) : _Where:
freq: frequencyFilter-Based Oscillators
osc[b|r|rs|rc|s|w](f), where f = frequency in Hz.(os.)oscbSinusoidal oscillator based on the biquad.
oscb(freq) : _Where:
freq: frequency(os.)oscrqSinusoidal (sine and cosine) oscillator based on 2D vector rotation, = undamped “coupled-form” resonator = lossless 2nd-order normalized ladder filter.
oscrq(freq) : _,_Where:
freq: frequency(os.)oscrsSinusoidal (sine) oscillator based on 2D vector rotation, = undamped “coupled-form” resonator = lossless 2nd-order normalized ladder filter.
oscrs(freq) : _Where:
freq: frequency(os.)oscrcSinusoidal (cosine) oscillator based on 2D vector rotation, = undamped “coupled-form” resonator = lossless 2nd-order normalized ladder filter.
oscrc(freq) : _Where:
freq: frequency(os.)oscsSinusoidal oscillator based on the state variable filter = undamped “modified-coupled-form” resonator = “magic circle” algorithm used in graphics
(os.)oscDefault sine wave oscillator (same as oscsin). osc is a standard Faust function.
osc(freq) : _Where:
freq: the frequency of the wave (Hz)Sinusoidal oscillator based on the waveguide resonator wgr.
(os.)oscwSinusoidal oscillator based on the waveguide resonator wgr. Unit-amplitude cosine oscillator.
oscwc(freq) : _Where:
freq: frequency(os.)oscwsSinusoidal oscillator based on the waveguide resonator wgr. Unit-amplitude sine oscillator
oscws(freq) : _Where:
freq: frequency(os.)oscwqSinusoidal oscillator based on the waveguide resonator wgr. Unit-amplitude cosine and sine (quadrature) oscillator.
oscwq(freq) : _Where:
freq: frequency(os.)oscwSinusoidal oscillator based on the waveguide resonator wgr. Unit-amplitude cosine oscillator (default)
oscw(freq) : _Where:
freq: frequencyOscillators that mimics some of the Casio CZ oscillators.
(os.)CZsawOscillator that mimics the Casio CZ saw oscillator CZsaw is a standard Faust function.
CZsaw(fund,index) : _Where:
fund: a saw-tooth waveform between 0 and 1 that the oscillator slaves toindex: the brightness of the oscillator, 0 to 1. 0 = sine-wave, 1 = saw-wave(os.)CZsquareOscillator that mimics the Casio CZ square oscillator CZsquare is a standard Faust function.
CZsquare(fund,index) : _Where:
fund: a saw-tooth waveform between 0 and 1 that the oscillator slaves toindex: the brightness of the oscillator, 0 to 1. 0 = sine-wave, 1 = square-wave(os.)CZpulseOscillator that mimics the Casio CZ pulse oscillator CZpulse is a standard Faust function.
CZpulse(fund,index) : _Where:
fund: a saw-tooth waveform between 0 and 1 that the oscillator slaves toindex: the brightness of the oscillator, 0 gives a sine-wave, 1 is closer to a pulse(os.)CZsinePulseOscillator that mimics the Casio CZ sine/pulse oscillator CZsinePulse is a standard Faust function.
CZsinePulse(fund,index) : _Where:
fund: a saw-tooth waveform between 0 and 1 that the oscillator slaves toindex: the brightness of the oscillator, 0 gives a sine-wave, 1 is a sine minus a pulse(os.)CZhalfSineOscillator that mimics the Casio CZ half sine oscillator CZhalfSine is a standard Faust function.
CZhalfSine(fund,index) : _Where:
fund: a saw-tooth waveform between 0 and 1 that the oscillator slaves toindex: the brightness of the oscillator, 0 gives a sine-wave, 1 is somewhere between a saw and a square(os.)CZresSawOscillator that mimics the Casio CZ resonant saw-tooth oscillator CZresSaw is a standard Faust function.
CZresSaw(fund,res) : _Where:
fund: a saw-tooth waveform between 0 and 1 that the oscillator slaves tores: the frequency of resonance as a factor of the fundamental pitch.(os.)CZresTriangleOscillator that mimics the Casio CZ resonant triangle oscillator CZresTriangle is a standard Faust function.
CZresTriangle(fund,res) : _Where:
fund: a saw-tooth waveform between 0 and 1 that the oscillator slaves tores: the frequency of resonance as a factor of the fundamental pitch.(os.)CZresTrapOscillator that mimics the Casio CZ resonant trapeze oscillator CZresTrap is a standard Faust function.
CZresTrap(fund,res) : _Where:
fund: a saw-tooth waveform between 0 and 1 that the oscillator slaves tores: the frequency of resonance as a factor of the fundamental pitch.A library of phasor and flanger effects. Its official prefix is pf.
(pf.)flanger_monoMono flanging effect.
_ : flanger_mono(dmax,curdel,depth,fb,invert) : _;Where:
dmax: maximum delay-line length (power of 2) - 10 ms typicalcurdel: current dynamic delay (not to exceed dmax)depth: effect strength between 0 and 1 (1 typical)fb: feedback gain between 0 and 1 (0 typical)invert: 0 for normal, 1 to invert sign of flanging sumhttps://ccrma.stanford.edu/~jos/pasp/Flanging.html
(pf.)flanger_stereoStereo flanging effect. flanger_stereo is a standard Faust function.
_,_ : flanger_stereo(dmax,curdel1,curdel2,depth,fb,invert) : _,_;Where:
dmax: maximum delay-line length (power of 2) - 10 ms typicalcurdel: current dynamic delay (not to exceed dmax)depth: effect strength between 0 and 1 (1 typical)fb: feedback gain between 0 and 1 (0 typical)invert: 0 for normal, 1 to invert sign of flanging sumhttps://ccrma.stanford.edu/~jos/pasp/Flanging.html
(pf.)phaser2_monoMono phasing effect.
_ : phaser2_mono(Notches,phase,width,frqmin,fratio,frqmax,speed,depth,fb,invert) : _;Where:
Notches: number of spectral notches (MACRO ARGUMENT - not a signal)phase: phase of the oscillator (0-1)width: approximate width of spectral notches in Hzfrqmin: approximate minimum frequency of first spectral notch in Hzfratio: ratio of adjacent notch frequenciesfrqmax: approximate maximum frequency of first spectral notch in Hzspeed: LFO frequency in Hz (rate of periodic notch sweep cycles)depth: effect strength between 0 and 1 (1 typical) (aka “intensity”) when depth=2, “vibrato mode” is obtained (pure allpass chain)fb: feedback gain between -1 and 1 (0 typical)invert: 0 for normal, 1 to invert sign of flanging sumReference:
(pf.)phaser2_stereoStereo phasing effect. phaser2_stereo is a standard Faust function.
_ : phaser2_stereo(Notches,phase,width,frqmin,fratio,frqmax,speed,depth,fb,invert) : _;Where:
Notches: number of spectral notches (MACRO ARGUMENT - not a signal)phase: phase of the oscillator (0-1)width: approximate width of spectral notches in Hzfrqmin: approximate minimum frequency of first spectral notch in Hzfratio: ratio of adjacent notch frequenciesfrqmax: approximate maximum frequency of first spectral notch in Hzspeed: LFO frequency in Hz (rate of periodic notch sweep cycles)depth: effect strength between 0 and 1 (1 typical) (aka “intensity”) when depth=2, “vibrato mode” is obtained (pure allpass chain)fb: feedback gain between -1 and 1 (0 typical)invert: 0 for normal, 1 to invert sign of flanging sumReference:
Faust physical modeling library; Its official prefix is pm.
This library provides an environment to facilitate physical modeling of musical instruments. It contains dozens of functions implementing low and high level elements going from a simple waveguide to fully operational models with built-in UI, etc.
It is organized as follows:
This library is part of the Faust Physical Modeling ToolKit. More information on how to use this library can be found on this page: https://ccrma.stanford.edu/~rmichon/pmFaust. Tutorials on how to make physical models of musical instruments using Faust can be found here as well.
Useful pre-defined variables for physical modeling.
(pm.)speedOfSoundSpeed of sound in meters per second (340m/s).
(pm.)maxLengthThe default maximum length (3) in meters of strings and tubes used in this library. This variable should be overriden to allow longer strings or tubes.
Useful conversion tools for physical modeling.
(pm.)f2lFrequency to length in meters.
f2l(freq) : distanceInMetersWhere:
freq: the frequency(pm.)l2fLength in meters to frequency.
l2f(length) : freqWhere:
length: length/distance in meters(pm.)l2sLength in meters to number of samples.
l2s(l) : numberOfSamplesWhere:
l: length in metersSet of fundamental functions to create bi-directional block diagrams in Faust. These elements are used as the basis of this library to connect high level elements (e.g., mouthpieces, strings, bridge, instrument body, etc.). Each block has 3 inputs and 3 outputs. The first input/output carry left going waves, the second input/output carry right going waves, and the third input/output is used to carry any potential output signal to the end of the algorithm.
(pm.)basicBlockEmpty bidirectional block to be used with chain: 3 signals ins and 3 signals out.
chain(basicBlock : basicBlock : etc.)(pm.)chainCreates a chain of bidirectional blocks. Blocks must have 3 inputs and outputs. The first input/output carry left going waves, the second input/output carry right going waves, and the third input/output is used to carry any potential output signal to the end of the algorithm. The implied one sample delay created by the ~ operator is generalized to the left and right going waves. Thus, n blocks in chain() will add an n samples delay to both left and right going waves.
leftGoingWaves,rightGoingWaves,mixedOutput : chain( A : B ) : leftGoingWaves,rightGoingWaves,mixedOutput
with{
A = _,_,_;
B = _,_,_;
};(pm.)inLeftWaveAdds a signal to left going waves anywhere in a chain of blocks.
model(x) = chain(A : inLeftWave(x) : B)Where A and B are bidirectional blocks and x is the signal added to left going waves in that chain.
(pm.)inRightWaveAdds a signal to right going waves anywhere in a chain of blocks.
model(x) = chain(A : inRightWave(x) : B)Where A and B are bidirectional blocks and x is the signal added to right going waves in that chain.
(pm.)inAdds a signal to left and right going waves anywhere in a chain of blocks.
model(x) = chain(A : in(x) : B)Where A and B are bidirectional blocks and x is the signal added to left and right going waves in that chain.
(pm.)outLeftWaveSends the signal of left going waves to the output channel of the chain.
chain(A : outLeftWave : B)Where A and B are bidirectional blocks.
(pm.)outRightWaveSends the signal of right going waves to the output channel of the chain.
chain(A : outRightWave : B)Where A and B are bidirectional blocks.
(pm.)outSends the signal of right and left going waves to the output channel of the chain.
chain(A : out : B)Where A and B are bidirectional blocks.
(pm.)terminationsCreates terminations on both sides of a chain without closing the inputs and outputs of the bidirectional signals chain. As for chain, this function adds a 1 sample delay to the bidirectional signal, both ways. Of courses, this function can be nested within a chain.
terminations(a,b,c)
with{
a = *(-1); // left termination
b = chain(D : E : F); // bidirectional chain of blocks (D, E, F, etc.)
c = *(-1); // right termination
};(pm.)lTerminationCreates a termination on the left side of a chain without closing the inputs and outputs of the bidirectional signals chain. This function adds a 1 sample delay near the termination and can be nested within another chain.
lTerminations(a,b)
with{
a = *(-1); // left termination
b = chain(D : E : F); // bidirectional chain of blocks (D, E, F, etc.)
};(pm.)rTerminationCreates a termination on the right side of a chain without closing the inputs and outputs of the bidirectional signals chain. This function adds a 1 sample delay near the termination and can be nested within another chain.
rTerminations(b,c)
with{
b = chain(D : E : F); // bidirectional chain of blocks (D, E, F, etc.)
c = *(-1); // right termination
};(pm.)closeInsCloses the inputs of a bidirectional chain in all directions.
closeIns : chain(...) : _,_,_(pm.)closeOutsCloses the outputs of a bidirectional chain in all directions except for the main signal output (3d output).
_,_,_ : chain(...) : _(pm.)endChainCloses the inputs and outputs of a bidirectional chain in all directions except for the main signal output (3d output).
endChain(chain(...)) : _Basic elements for physical modeling (e.g., waveguides, specific filters, etc.).
(pm.)waveguideNA series of waveguide functions based on various types of delays (see fdelay[n]).
waveguideUd: unit delay waveguidewaveguideFd: fractional delay waveguidewaveguideFd2: second order fractional delay waveguidewaveguideFd4: fourth order fractional delay waveguidechain(A : waveguideUd(nMax,n) : B)Where:
nMax: the maximum length of the delays in the waveguiden: the length of the delay lines in samples.(pm.)waveguideStandard pm.lib waveguide (based on waveguideFd4).
chain(A : waveguide(nMax,n) : B)Where:
nMax: the maximum length of the delays in the waveguiden: the length of the delay lines in samples.(pm.)bridgeFilterGeneric two zeros bridge FIR filter (as implemented in the STK) that can be used to implement the reflectance violin, guitar, etc. bridges.
_ : bridge(brightness,absorption) : _Where:
brightness: controls the damping of high frequencies (0-1)absorption: controls the absorption of the brige and thus the t60 of the string plugged to it (0-1) (1 = 20 seconds)(pm.)modeFilterResonant bandpass filter that can be used to implement a single resonance (mode).
_ : modeFilter(freq,t60,gain) : _Where:
freq: mode frequencyt60: mode resonance duration (in seconds)gain: mode gain (0-1)Low and high level string instruments parts. Most of the elements in this section can be used in a bidirectional chain.
(pm.)stringSegmentA string segment without terminations (just a simple waveguide).
chain(A : stringSegment(maxLength,length) : B)Where:
maxLength: the maximum length of the string in meters (should be static)length: the length of the string in meters(pm.)openStringA bidirectional block implementing a basic “generic” string with a selectable excitation position. Lowpass filters are built-in and allow to simulate the effect of dispersion on the sound and thus to change the “stiffness” of the string.
chain(... : openString(length,stiffness,pluckPosition,excitation) : ...)Where:
length: the length of the string in metersstiffness: the stiffness of the string (0-1) (1 for max stiffness)pluckPosition: excitation position (0-1) (1 is bottom)excitation: the excitation signal(pm.)nylonStringA bidirectional block implementing a basic nylon string with selectable excitation position. This element is based on openString and has a fix stiffness corresponding to that of a nylon string.
chain(... : nylonString(length,pluckPosition,excitation) : ...)Where:
length: the length of the string in meterspluckPosition: excitation position (0-1) (1 is bottom)excitation: the excitation signal(pm.)steelStringA bidirectional block implementing a basic steel string with selectable excitation position. This element is based on openString and has a fix stiffness corresponding to that of a steel string.
chain(... : steelString(length,pluckPosition,excitation) : ...)Where:
length: the length of the string in meterspluckPosition: excitation position (0-1) (1 is bottom)excitation: the excitation signal(pm.)openStringPickA bidirectional block implementing a “generic” string with selectable excitation position. It also has a built-in pickup whose position is the same as the excitation position. Thus, moving the excitation position will also move the pickup.
chain(... : openStringPick(length,stiffness,pluckPosition,excitation) : ...)Where:
length: the length of the string in metersstiffness: the stiffness of the string (0-1) (1 for max stiffness)pluckPosition: excitation position (0-1) (1 is bottom)excitation: the excitation signal(pm.)openStringPickUpA bidirectional block implementing a “generic” string with selectable excitation position and stiffness. It also has a built-in pickup whose position can be independenly selected. The only constraint is that the pickup has to be placed after the excitation position.
chain(... : openStringPickUp(length,stiffness,pluckPosition,excitation) : ...)Where:
length: the length of the string in metersstiffness: the stiffness of the string (0-1) (1 for max stiffness)pluckPosition: pluck position between the top of the string and the pickup (0-1) (1 for same as pickup position)pickupPosition: position of the pickup on the string (0-1) (1 is bottom)excitation: the excitation signal(pm.)openStringPickDownA bidirectional block implementing a “generic” string with selectable excitation position and stiffness. It also has a built-in pickup whose position can be independenly selected. The only constraint is that the pickup has to be placed before the excitation position.
chain(... : openStringPickDown(length,stiffness,pluckPosition,excitation) : ...)Where:
length: the length of the string in metersstiffness: the stiffness of the string (0-1) (1 for max stiffness)pluckPosition: pluck position on the string (0-1) (1 is bottom)pickupPosition: position of the pickup between the top of the string and the excitation position (0-1) (1 is excitation position)excitation: the excitation signal(pm.)ksReflexionFilterThe “typical” one-zero Karplus-strong feedforward reflexion filter. This filter will be typically used in a termination (see below).
terminations(_,chain(...),ksReflexionFilter)(pm.)rStringRigidTerminationBidirectional block implementing a right rigid string termination (no damping, just phase inversion).
chain(rStringRigidTermination : stringSegment : ...)(pm.)lStringRigidTerminationBidirectional block implementing a left rigid string termination (no damping, just phase inversion).
chain(... : stringSegment : lStringRigidTermination)(pm.)elecGuitarBridgeBidirectional block implementing a simple electric guitar bridge. This block is based on bridgeFilter. The bridge doesn’t implement transmittance since it is not meant to be connected to a body (unlike acoustic guitar). It also partially sets the resonance duration of the string with the nuts used on the other side.
chain(... : stringSegment : elecGuitarBridge)(pm.)elecGuitarNutsBidirectional block implementing a simple electric guitar nuts. This block is based on bridgeFilter and does essentially the same thing as elecGuitarBridge, but on the other side of the chain. It also partially sets the resonance duration of the string with the bridge used on the other side.
chain(elecGuitarNuts : stringSegment : ...)(pm.)guitarBridgeBidirectional block implementing a simple acoustic guitar bridge. This bridge damps more hight frequencies than elecGuitarBridge and implements a transmittance filter. It also partially sets the resonance duration of the string with the nuts used on the other side.
chain(... : stringSegment : guitarBridge)(pm.)guitarNutsBidirectional block implementing a simple acoustic guitar nuts. This nuts damps more hight frequencies than elecGuitarNuts and implements a transmittance filter. It also partially sets the resonance duration of the string with the bridge used on the other side.
chain(guitarNuts : stringSegment : ...)(pm.)idealStringAn “ideal” string with rigid terminations and where the plucking position and the pick-up position are the same. Since terminations are rigid, this string will ring forever.
1-1' : idealString(length,reflexion,xPosition,excitation)With: * length: the length of the string in meters * pluckPosition: the plucking position (0.001-0.999) * excitation: the input signal for the excitation
(pm.)ksA Karplus-Strong string (in that case, the string is implemented as a one dimension waveguide).
ks(length,damping,excitation) : _Where:
length: the length of the string in metersdamping: string damping (0-1)excitation: excitation signal(pm.)ks_ui_MIDIReady-to-use, MIDI-enabled Karplus-Strong string with buil-in UI.
ks_ui_MIDI : _(pm.)elecGuitarModelA simple electric guitar model (without audio effects, of course) with selectable pluck position. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function. Pitch is changed by changing the length of the string and not through a finger model.
elecGuitarModel(length,pluckPosition,mute,excitation) : _Where:
length: the length of the string in meterspluckPosition: pluck position (0-1) (1 is on the bridge)mute: mute coefficient (1 for no mute and 0 for instant mute)excitation: excitation signal(pm.)elecGuitarA simple electric guitar model with steel strings (based on elecGuitarModel) implementing an excitation model. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function.
elecGuitar(length,pluckPosition,trigger) : _Where:
length: the length of the string in meterspluckPosition: pluck position (0-1) (1 is on the bridge)mute: mute coefficient (1 for no mute and 0 for instant mute)gain: gain of the pluck (0-1)trigger: trigger signal (1 for on, 0 for off)(pm.)elecGuitar_ui_MIDIReady-to-use MIDI-enabled electric guitar physical model with built-in UI.
elecGuitar_ui_MIDI : _(pm.)guitarBodyWARNING: not implemented yet! Bidirectional block implementing a simple acoustic guitar body.
chain(... : guitarBody)(pm.)guitarModelA simple acoustic guitar model with steel strings and selectable excitation position. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function. Pitch is changed by changing the length of the string and not through a finger model. WARNING: this function doesn’t currently implement a body (just strings and bridge)
guitarModel(length,pluckPosition,excitation) : _Where:
length: the length of the string in meterspluckPosition: pluck position (0-1) (1 is on the bridge)excitation: excitation signal(pm.)guitarA simple acoustic guitar model with steel strings (based on guitarModel) implementing an excitation model. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function.
guitar(length,pluckPosition,trigger) : _Where:
length: the length of the string in meterspluckPosition: pluck position (0-1) (1 is on the bridge)gain: gain of the excitationtrigger: trigger signal (1 for on, 0 for off)(pm.)guitar_ui_MIDIReady-to-use MIDI-enabled steel strings acoustic guitar physical model with built-in UI.
guitar_ui_MIDI : _(pm.)nylonGuitarModelA simple acoustic guitar model with nylon strings and selectable excitation position. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function. Pitch is changed by changing the length of the string and not through a finger model. WARNING: this function doesn’t currently implement a body (just strings and bridge)
nylonGuitarModel(length,pluckPosition,excitation) : _Where:
length: the length of the string in meterspluckPosition: pluck position (0-1) (1 is on the bridge)excitation: excitation signal(pm.)nylonGuitarA simple acoustic guitar model with steel strings (based on nylonGuitarModel) implementing an excitation model. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function.
nylonGuitar(length,pluckPosition,trigger) : _Where:
length: the length of the string in meterspluckPosition: pluck position (0-1) (1 is on the bridge)gain: gain of the excitation (0-1)trigger: trigger signal (1 for on, 0 for off)(pm.)nylonGuitar_ui_MIDIReady-to-use MIDI-enabled nylon strings acoustic guitar physical model with built-in UI.
nylonGuitar_ui_MIDI : _(pm.)modeInterpResModular string instrument resonator based on IR measurements made on 3D printed models. The 2D space allowing for the control of the shape and the scale of the model is enabled by interpolating between modes parameters. More information about this technique/project can be found here: https://ccrma.stanford.edu/~rmichon/3dPrintingModeling/
_ : modeInterpRes(nModes,x,y) : _Where:
nModes: number of modeled modes (40 max)x: shape of the resonator (0: square, 1: square with rounded corners, 2: round)y: scale of the resonator (0: small, 1: medium, 2: large)(pm.)modularInterpBodyBidirectional block implementing a modular string instrument resonator (see modeInterpRes).
chain(... : modularInterpBody(nModes,shape,scale) : ...)Where:
nModes: number of modeled modes (40 max)shape: shape of the resonator (0: square, 1: square with rounded corners, 2: round)scale: scale of the resonator (0: small, 1: medium, 2: large)(pm.)modularInterpStringModelString instrument model with a modular body (see modeInterpRes and https://ccrma.stanford.edu/~rmichon/3dPrintingModeling/).
modularInterpStringModel(length,pluckPosition,shape,scale,bodyExcitation,stringExcitation) : _Where:
stringLength: the length of the string in meterspluckPosition: pluck position (0-1) (1 is on the bridge)shape: shape of the resonator (0: square, 1: square with rounded corners, 2: round)scale: scale of the resonator (0: small, 1: medium, 2: large)bodyExcitation: excitation signal for the bodystringExcitation: excitation signal for the string(pm.)modularInterpInstrString instrument with a modular body (see modeInterpRes and https://ccrma.stanford.edu/~rmichon/3dPrintingModeling/).
modularInterpInstr(stringLength,pluckPosition,shape,scale,gain,tapBody,triggerString) : _Where:
stringLength: the length of the string in meterspluckPosition: pluck position (0-1) (1 is on the bridge)shape: shape of the resonator (0: square, 1: square with rounded corners, 2: round)scale: scale of the resonator (0: small, 1: medium, 2: large)gain: of the string excitationtapBody: send an impulse in the body of the instrument where the string is connected (1 for on, 0 for off)triggerString: trigger signal for the string (1 for on, 0 for off)(pm.)modularInterpInstr_ui_MIDIReady-to-use MIDI-enabled string instrument with a modular body (see modeInterpRes and https://ccrma.stanford.edu/~rmichon/3dPrintingModeling/) with built-in UI.
modularInterpInstr_ui_MIDI : _Low and high level basic string instruments parts. Most of the elements in this section can be used in a bidirectional chain.
(pm.)bowTableExtremely basic bow table that can be used to implement a wide range of bow types for many different bowed string instruments (violin, cello, etc.)
excitation : bowTable(offeset,slope) : _Where:
excitation: an excitation signaloffset: table offsetslope: table slope(pm.)violinBowTableViolin bow table based on bowTable.
bowVelocity : violinBowTable(bowPressure) : _Where:
bowVelocity: velocity of the bow/excitation signal (0-1)bowPressure: bow pressure on the string (0-1)(pm.)bowInteractionBidirectional block implementing the interaction of a bow in a chain.
chain(... : stringSegment : bowInteraction(bowTable) : stringSegment : ...)Where:
bowTable: the bow table(pm.)violinBowBidirectional block implementing a violin bow and its interaction with a string.
chain(... : stringSegment : violinBow(bowPressure,bowVelocity) : stringSegment : ...)Where:
bowVelocity: velocity of the bow / excitation signal (0-1)bowPressure: bow pressure on the string (0-1)(pm.)violinBowedStringViolin bowed string bidirectional block with controllable bow position. Terminations are not implemented in this model.
chain(nuts : violinBowedString(stringLength,bowPressure,bowVelocity,bowPosition) : bridge)Where:
stringLength: the length of the string in metersbowVelocity: velocity of the bow / excitation signal (0-1)bowPressure: bow pressure on the string (0-1)bowPosition: the position of the bow on the string (0-1)(pm.)violinNutsBidirectional block implementing simple violin nuts. This function is based on bridgeFilter.
chain(violinNuts : stringSegment : ...)(pm.)violinBridgeBidirectional block implementing a simple violin bridge. This function is based on bridgeFilter.
chain(... : stringSegment : violinBridge(pm.)violinBodyBidirectional block implementing a simple violin body (just a simple resonant lowpass filter).
chain(... : stringSegment : violinBridge : violinBody)(pm.)violinModelReady-to-use simple violin physical model. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function. Pitch is changed by changing the length of the string (and not through a finger model).
violinModel(stringLength,bowPressure,bowVelocity,bridgeReflexion,
bridgeAbsorption,bowPosition) : _Where:
stringLength: the length of the string in metersbowVelocity: velocity of the bow / excitation signal (0-1)bowPressure: bow pressure on the string (0-1))bowPosition: the position of the bow on the string (0-1)(pm.)violin_uiReady-to-use violin physical model with built-in UI.
violinModel_ui : _(pm.)violin_ui_MIDIReady-to-use MIDI-enabled violin physical model with built-in UI.
violin_ui_MIDI : _Low and high level basic wind instruments parts. Most of the elements in this section can be used in a bidirectional chain.
(pm.)openTubeA tube segment without terminations (same as stringSegment).
chain(A : openTube(maxLength,length) : B)Where:
maxLength: the maximum length of the tube in meters (should be static)length: the length of the tube in meters(pm.)reedTableExtremely basic reed table that can be used to implement a wide range of single reed types for many different instruments (saxophone, clarinet, etc.).
excitation : reedTable(offeset,slope) : _Where:
excitation: an excitation signaloffset: table offsetslope: table slope(pm.)fluteJetTableExtremely basic flute jet table.
excitation : fluteJetTable : _Where:
excitation: an excitation signal(pm.)brassLipsTableSimple brass lips/mouthpiece table. Since this implementation is very basic and that the lips and tube of the instrument are coupled to each other, the length of that tube must be provided here.
excitation : brassLipsTable(tubeLength,lipsTension) : _Where:
excitation: an excitation signal (can be DC)tubeLength: length in meters of the tube connected to the mouthpiecelipsTension: tension of the lips (0-1) (default: 0.5)(pm.)clarinetReedClarinet reed based on reedTable with controllable stiffness.
excitation : clarinetReed(stiffness) : _Where:
excitation: an excitation signalstiffness: reed stiffness (0-1)(pm.)clarinetMouthPieceBidirectional block implementing a clarinet mouthpiece as well as the various interactions happening with traveling waves. This element is ready to be plugged to a tube…
chain(clarinetMouthPiece(reedStiffness,pressure) : tube : etc.)Where:
pressure: the pressure of the air flow (DC) created by the virtual performer (0-1). This can also be any kind of signal that will directly injected in the mouthpiece (e.g., breath noise, etc.)reedStiffness: reed stiffness (0-1)(pm.)brassLipsBidirectional block implementing a brass mouthpiece as well as the various interactions happening with traveling waves. This element is ready to be plugged to a tube…
chain(brassLips(tubeLength,lipsTension,pressure) : tube : etc.)Where:
tubeLength: length in meters of the tube connected to the mouthpiecelipsTension: tension of the lips (0-1) (default: 0.5)pressure: the pressure of the air flow (DC) created by the virtual performer (0-1). This can also be any kind of signal that will directly injected in the mouthpiece (e.g., breath noise, etc.)(pm.)fluteEmbouchureBidirectional block implementing a flute embouchure as well as the various interactions happening with traveling waves. This element is ready to be plugged between tubes segments…
chain(... : tube : fluteEmbouchure(pressure) : tube : etc.)Where:
pressure: the pressure of the air flow (DC) created by the virtual performer (0-1). This can also be any kind of signal that will directly injected in the mouthpiece (e.g., breath noise, etc.)(pm.)wBellGeneric wind instrument bell bidirectional block that should be placed at the end of a chain.
chain(... : wBell(opening))Where:
opening: the “opening” of bell (0-1)(pm.)fluteHeadSimple flute head implementing waves reflexion.
chain(fluteHead : tube : ...)(pm.)fluteFootSimple flute foot implementing waves reflexion and dispersion.
chain(... : tube : fluteFoot)(pm.)clarinetModelA simple clarinet physical model without tone holes (pitch is changed by changing the length of the tube of the instrument).
clarinetModel(length,pressure,reedStiffness,bellOpening) : _Where:
tubeLength: the length of the tube in meterspressure: the pressure of the air flow created by the virtual performer (0-1). This can also be any kind of signal that will directly injected in the mouthpiece (e.g., breath noise, etc.)reedStiffness: reed stiffness (0-1)bellOpening: the opening of bell (0-1)(pm.)clarinetModel_uiSame as clarinetModel but with a built-in UI. This function doesn’t implement a virtual “blower”, thus pressure remains an argument here.
clarinetModel_ui(pressure) : _Where:
pressure: the pressure of the air flow created by the virtual performer (0-1). This can also be any kind of signal that will be directly injected in the mouthpiece (e.g., breath noise, etc.)(pm.)clarinet_uiReady-to-use clarinet physical model with built-in UI based on clarinetModel.
clarinet_ui : _(pm.)clarinet_ui_MIDIReady-to-use MIDI compliant clarinet physical model with built-in UI.
clarinet_ui_MIDI : _(pm.)brassModelA simple generic brass instrument physical model without pistons (pitch is changed by changing the length of the tube of the instrument). This model is kind of hard to control and might not sound very good if bad parameters are given to it…
brassModel(tubeLength,lipsTension,mute,pressure) : _Where:
tubeLength: the length of the tube in meterslipsTension: tension of the lips (0-1) (default: 0.5)mute: mute opening at the end of the instrument (0-1) (default: 0.5)pressure: the pressure of the air flow created by the virtual performer (0-1). This can also be any kind of signal that will directly injected in the mouthpiece (e.g., breath noise, etc.)(pm.)brassModel_uiSame as brassModel but with a built-in UI. This function doesn’t implement a virtual “blower”, thus pressure remains an argument here.
brassModel_ui(pressure) : _Where:
pressure: the pressure of the air flow created by the virtual performer (0-1). This can also be any kind of signal that will be directly injected in the mouthpiece (e.g., breath noise, etc.)(pm.)brass_uiReady-to-use brass instrument physical model with built-in UI based on brassModel.
brass_ui : _(pm.)brass_ui_MIDIReady-to-use MIDI-controllable brass instrument physical model with built-in UI.
brass_ui_MIDI : _(pm.)fluteModelA simple generic brass instrument physical model without tone holes (pitch is changed by changing the length of the tube of the instrument).
fluteModel(tubeLength,lipsTension,mute,pressure) : _Where:
tubeLength: the length of the tube in metersmouthPosition: position of the mouth on the embouchure (0-1) (default: 0.5)pressure: the pressure of the air flow created by the virtual performer (0-1). This can also be any kind of signal that will directly injected in the mouthpiece (e.g., breath noise, etc.)(pm.)fluteModel_uiSame as fluteModel but with a built-in UI. This function doesn’t implement a virtual “blower”, thus pressure remains an argument here.
fluteModel_ui(pressure) : _Where:
pressure: the pressure of the air flow created by the virtual performer (0-1). This can also be any kind of signal that will be directly injected in the mouthpiece (e.g., breath noise, etc.)(pm.)flute_uiReady-to-use flute physical model with built-in UI based on fluteModel.
flute_ui : _(pm.)flute_ui_MIDIReady-to-use MIDI-controllable flute physical model with built-in UI.
brass_ui_MIDI : _Various kind of excitation signal generators.
(pm.)impulseExcitationCreates an impulse excitation of one sample.
gate = button('gate');
impulseExcitation(gate) : chain;Where:
gate: a gate button(pm.)strikeModelCreates a filtered noise excitation.
gate = button('gate');
strikeModel(LPcutoff,HPcutoff,sharpness,gain,gate) : chain;Where:
HPcutoff: highpass cutoff frequencyLPcutoff: lowpass cutoff frequencysharpness: sharpness of the attack and release (0-1)gain: gain of the excitationgate: a gate button/trigger signal (0/1)(pm.)strikeStrikes generator with controllable excitation position.
gate = button('gate');
strike(exPos,sharpness,gain,gate) : chain;Where:
exPos: excitation position wiht 0: for max low freqs and 1: for max high freqs. So, on membrane for example, 0 would be the middle and 1 the edgesharpness: sharpness of the attack and release (0-1)gain: gain of the excitationgate: a gate button/trigger signal (0/1)(pm.)pluckStringCreates a plucking excitation signal.
trigger = button('gate');
pluckString(stringLength,cutoff,maxFreq,sharpness,trigger)Where:
stringLength: length of the string to pluckcutoff: cutoff ratio (1 for default)maxFreq: max frequency ratio (1 for default)sharpness: sharpness of the attack and release (1 for default)gain: gain of the excitation (0-1)trigger: trigger signal (1 for on, 0 for off)(pm.)blowerA virtual blower creating a DC signal with some breath noise in it.
blower(pressure,breathGain,breathCutoff) : _Where:
pressure: pressure (0-1)breathGain: breath noise gain (0-1) (recommended: 0.005)breathCutoff: breath cuttoff frequency (Hz) (recommended: 2000)(pm.)blower_uiSame as blower but with a built-in UI.
blower : somethingToBeBlownHigh and low level functions for modal synthesis of percussion instruments.
(pm.)djembeModelDirt-simple djembe modal physical model. Mode parameters are empirically calculated and don’t correspond to any measurements or 3D model. They kind of sound good though :).
excitation : djembeModel(freq)Where:
excitation: excitation signalfreq: fundamental frequency of the bar(pm.)djembeDirt-simple djembe modal physical model. Mode parameters are empirically calculated and don’t correspond to any measurements or 3D model. They kind of sound good though :).
This model also implements a virtual “exciter”.
djembe(freq,strikePosition,strikeSharpness,gain,trigger)Where:
freq: fundamental frequency of the modelstrikePosition: strike position (0 for the middle of the membrane and 1 for the edge)strikeSharpness: sharpness of the strike (0-1, default: 0.5)gain: gain of the striketrigger: trigger signal (0: off, 1: on)(pm.)djembe_ui_MIDISimple MIDI controllable djembe physical model with built-in UI.
djembe_ui_MIDI : _(pm.)marimbaBarModelGeneric marimba tone bar modal model.
This model was generated using mesh2faust from a 3D CAD model of a marimba tone bar (libraries/modalmodels/marimbaBar). The corresponding CAD model is that of a C2 tone bar (original fundamental frequency: ~65Hz). While marimbaBarModel allows to translate the harmonic content of the generated sound by providing a frequency (freq), mode transposition has limits and the model will sound less and less like a marimba tone bar as it diverges from C2. To make an accurate model of a marimba, we’d want to have an independent model for each bar…
This model contains 5 excitation positions going linearly from the center bottom to the center top of the bar. Obviously, a model with more excitation position could be regenerated using mesh2faust.
excitation : marimbaBarModel(freq,exPos,t60,t60DecayRatio,t60DecaySlope)Where:
excitation: excitation signalfreq: fundamental frequency of the barexPos: excitation position (0-4)t60: T60 in seconds (recommended value: 0.1)t60DecayRatio: T60 decay ratio (recommended value: 1)t60DecaySlope: T60 decay slope (recommended value: 5)(pm.)marimbaResTubeSimple marimba resonance tube.
marimbaResTube(tubeLength,excitation)Where:
tubeLength: the length of the tube in metersexcitation: the excitation signal (audio in)(pm.)marimbaModelSimple marimba physical model implementing a single tone bar connected to tube. This model is scalable and can be adapted to any size of bar/tube (see marimbaBarModel to know more about the limitations of this type of system).
excitation : marimbaModel(freq,exPos) : _Where:
freq: the frequency of the bar/tube coupleexPos: excitation position (0-4)(pm.)marimbaSimple marimba physical model implementing a single tone bar connected to tube. This model is scalable and can be adapted to any size of bar/tube (see marimbaBarModel to know more about the limitations of this type of system).
This function also implement a virtual exciter to drive the model.
excitation : marimba(freq,strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _Where:
excitation: the excitation signalfreq: the frequency of the bar/tube couplestrikePosition: strike position (0-4)strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)strikeSharpness: shaarpness of the strike (recommened: ~0.25)gain: gain of the strike (0-1)trigger signal (0: off, 1: on)(pm.)marimba_ui_MIDISimple MIDI controllable marimba physical model with built-in UI implementing a single tone bar connected to tube. This model is scalable and can be adapted to any size of bar/tube (see marimbaBarModel to know more about the limitations of this type of system).
marimba_ui_MIDI : _(pm.)churchBellModelGeneric church bell modal model generated by mesh2faust from libraries/modalmodels/churchBell.
Modeled after T. Rossing and R. Perrin, Vibrations of Bells, Applied Acoustics 2, 1987.
Model height is 301 mm.
This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.
excitation : churchBellModel(nModes,exPos,t60,t60DecayRatio,t60DecaySlope)Where:
excitation: the excitation signalnModes: number of synthesized modes (max: 50)exPos: excitation position (0-6)t60: T60 in seconds (recommended value: 0.1)t60DecayRatio: T60 decay ratio (recommended value: 1)t60DecaySlope: T60 decay slope (recommended value: 5)(pm.)churchBellGeneric church bell modal model.
Modeled after T. Rossing and R. Perrin, Vibrations of Bells, Applied Acoustics 2, 1987.
Model height is 301 mm.
This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.
This function also implement a virtual exciter to drive the model.
excitation : churchBell(strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _Where:
excitation: the excitation signalstrikePosition: strike position (0-6)strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)strikeSharpness: shaarpness of the strike (recommened: ~0.25)gain: gain of the strike (0-1)trigger signal (0: off, 1: on)(pm.)churchBell_uiChurch bell physical model based on churchBell with built-in UI.
churchBell_ui : _(pm.)englishBellModelEnglish church bell modal model generated by mesh2faust from libraries/modalmodels/englishBell.
Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell’ Tone, Archives of Foundry Engineering, 2016.
Model height is 1 m.
This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.
excitation : englishBellModel(nModes,exPos,t60,t60DecayRatio,t60DecaySlope)Where:
excitation: the excitation signalnModes: number of synthesized modes (max: 50)exPos: excitation position (0-6)t60: T60 in seconds (recommended value: 0.1)t60DecayRatio: T60 decay ratio (recommended value: 1)t60DecaySlope: T60 decay slope (recommended value: 5)(pm.)englishBellEnglish church bell modal model.
Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell’ Tone, Archives of Foundry Engineering, 2016.
Model height is 1 m.
This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.
This function also implement a virtual exciter to drive the model.
excitation : englishBell(strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _Where:
excitation: the excitation signalstrikePosition: strike position (0-6)strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)strikeSharpness: shaarpness of the strike (recommened: ~0.25)gain: gain of the strike (0-1)trigger signal (0: off, 1: on)(pm.)englishBell_uiEnglish church bell physical model based on englishBell with built-in UI.
englishBell_ui : _(pm.)frenchBellModelFrench church bell modal model generated by mesh2faust from libraries/modalmodels/frenchBell.
Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell’ Tone, Archives of Foundry Engineering, 2016.
Model height is 1 m.
This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.
excitation : frenchBellModel(nModes,exPos,t60,t60DecayRatio,t60DecaySlope)Where:
excitation: the excitation signalnModes: number of synthesized modes (max: 50)exPos: excitation position (0-6)t60: T60 in seconds (recommended value: 0.1)t60DecayRatio: T60 decay ratio (recommended value: 1)t60DecaySlope: T60 decay slope (recommended value: 5)(pm.)frenchBellFrench church bell modal model.
Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell’ Tone, Archives of Foundry Engineering, 2016.
Model height is 1 m.
This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.
This function also implement a virtual exciter to drive the model.
excitation : frenchBell(strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _Where:
excitation: the excitation signalstrikePosition: strike position (0-6)strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)strikeSharpness: shaarpness of the strike (recommened: ~0.25)gain: gain of the strike (0-1)trigger signal (0: off, 1: on)(pm.)frenchBell_uiFrench church bell physical model based on frenchBell with built-in UI.
frenchBell_ui : _(pm.)germanBellModelGerman church bell modal model generated by mesh2faust from libraries/modalmodels/germanBell.
Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell’ Tone, Archives of Foundry Engineering, 2016.
Model height is 1 m.
This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.
excitation : germanBellModel(nModes,exPos,t60,t60DecayRatio,t60DecaySlope)Where:
excitation: the excitation signalnModes: number of synthesized modes (max: 50)exPos: excitation position (0-6)t60: T60 in seconds (recommended value: 0.1)t60DecayRatio: T60 decay ratio (recommended value: 1)t60DecaySlope: T60 decay slope (recommended value: 5)(pm.)germanBellGerman church bell modal model.
Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell’ Tone, Archives of Foundry Engineering, 2016.
Model height is 1 m.
This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.
This function also implement a virtual exciter to drive the model.
excitation : germanBell(strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _Where:
excitation: the excitation signalstrikePosition: strike position (0-6)strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)strikeSharpness: shaarpness of the strike (recommened: ~0.25)gain: gain of the strike (0-1)trigger signal (0: off, 1: on)(pm.)germanBell_uiGerman church bell physical model based on germanBell with built-in UI.
germanBell_ui : _(pm.)russianBellModelRussian church bell modal model generated by mesh2faust from libraries/modalmodels/russianBell.
Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell’ Tone, Archives of Foundry Engineering, 2016.
Model height is 2 m.
This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.
excitation : russianBellModel(nModes,exPos,t60,t60DecayRatio,t60DecaySlope)Where:
excitation: the excitation signalnModes: number of synthesized modes (max: 50)exPos: excitation position (0-6)t60: T60 in seconds (recommended value: 0.1)t60DecayRatio: T60 decay ratio (recommended value: 1)t60DecaySlope: T60 decay slope (recommended value: 5)(pm.)russianBellRussian church bell modal model.
Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell’ Tone, Archives of Foundry Engineering, 2016.
Model height is 2 m.
This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.
This function also implement a virtual exciter to drive the model.
excitation : russianBell(strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _Where:
excitation: the excitation signalstrikePosition: strike position (0-6)strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)strikeSharpness: shaarpness of the strike (recommened: ~0.25)gain: gain of the strike (0-1)trigger signal (0: off, 1: on)(pm.)russianBell_uiRussian church bell physical model based on russianBell with built-in UI.
russianBell_ui : _(pm.)standardBellModelStandard church bell modal model generated by mesh2faust from libraries/modalmodels/standardBell.
Modeled after T. Rossing and R. Perrin, Vibrations of Bells, Applied Acoustics 2, 1987.
Model height is 1.8 m.
This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.
excitation : standardBellModel(nModes,exPos,t60,t60DecayRatio,t60DecaySlope)Where:
excitation: the excitation signalnModes: number of synthesized modes (max: 50)exPos: excitation position (0-6)t60: T60 in seconds (recommended value: 0.1)t60DecayRatio: T60 decay ratio (recommended value: 1)t60DecaySlope: T60 decay slope (recommended value: 5)(pm.)standardBellStandard church bell modal model.
Modeled after T. Rossing and R. Perrin, Vibrations of Bells, Applied Acoustics 2, 1987.
Model height is 1.8 m.
This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.
This function also implement a virtual exciter to drive the model.
excitation : standardBell(strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _Where:
excitation: the excitation signalstrikePosition: strike position (0-6)strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)strikeSharpness: shaarpness of the strike (recommened: ~0.25)gain: gain of the strike (0-1)trigger signal (0: off, 1: on)(pm.)standardBell_uiStandard church bell physical model based on standardBell with built-in UI.
standardBell_ui : _Vocal synthesizer functions (source/filter, fof, etc.).
(pm.)formantValuesFormant data values.
The formant data used here come from the CSOUND manual http://www.csounds.com/manual/html/.
ba.take(j+1,formantValues.f(i)) : _
ba.take(j+1,formantValues.g(i)) : _
ba.take(j+1,formantValues.bw(i)) : _Where:
i: formant numberj: (voiceType*nFormants)+vowelvoiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)(pm.)voiceGenderCalculate the gender for the provided voiceType value. (0: male, 1: female)
voiceGender(voiceType) : _Where:
voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)(pm.)skirtWidthMultiplierCalculates value to multiply bandwidth to obtain skirtwidth for a Fof filter.
skirtWidthMultiplier(vowel,freq,gender) : _Where:
vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)freq: the fundamental frequency of the excitation signalgender: gender of the voice used in the fof filter (0: male, 1: female)(pm.)autobendFreqAutobends the center frequencies of formants 1 and 2 based on the fundamental frequency of the excitation signal and leaves all other formant frequencies unchanged. Ported from chant-lib. Reference: https://ccrma.stanford.edu/~rmichon/chantLib/
_ : autobendFreq(n,freq,voiceType) : _Where:
n: formant indexfreq: the fundamental frequency of the excitation signalvoiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)(pm.)vocalEffortChanges the gains of the formants based on the fundamental frequency of the excitation signal. Higher formants are reinforced for higher fundamental frequencies. Ported from chant-lib. Reference: https://ccrma.stanford.edu/~rmichon/chantLib/
_ : vocalEffort(freq,gender) : _Where:
freq: the fundamental frequency of the excitation signalgender: the gender of the voice type (0: male, 1: female)(pm.)fofFunction to generate a single Formant-Wave-Function. Reference: https://ccrma.stanford.edu/~mjolsen/pdfs/smc2016_MOlsenFOF.pdf
_ : fof(fc,bw,a,g) : _Where:
fc: formant center frequency,bw: formant bandwidth (Hz),sw: formant skirtwidth (Hz)g: linear scale factor (g=1 gives 0dB amplitude response at fc)(pm.)fofSHFOF with sample and hold used on bw and a parameter used in the filter-cycling FOF function fofCycle. Reference: https://ccrma.stanford.edu/~mjolsen/pdfs/smc2016_MOlsenFOF.pdf
_ : fofSH(fc,bw,a,g) : _Where: all parameters same as for fof
(pm.)fofCycleFOF implementation where time-varying filter parameter noise is mitigated by using a cycle of n sample and hold FOF filters. Reference: https://ccrma.stanford.edu/~mjolsen/pdfs/smc2016_MOlsenFOF.pdf
_ : fofCycle(fc,bw,a,g,n) : _Where:
n: the number of FOF filters to cycle throughfof(pm.)fofSmoothFOF implementation where time-varying filter parameter noise is mitigated by lowpass filtering the filter parameters bw and a with smooth.
_ : fofSmooth(fc,bw,sw,g,tau) : _Where:
tau: the desired smoothing time constant in secondsfof(pm.)formantFilterFofCycleFormant filter based on a single FOF filter. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. A cycle of n fof filters with sample-and-hold is used so that the fof filter parameters can be varied in realtime. This technique is more robust but more computationally expensive than formantFilterFofSmooth.Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.
_ : formantFilterFofCycle(voiceType,vowel,nFormants,i,freq) : _Where:
voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)nFormants: number of formant regions in frequency domain, typically 5i: formant number (i.e. 0 - 4) used to index formant data value arraysfreq: fundamental frequency of excitation signal. Used to calculate rise time of envelope(pm.)formantFilterFofSmoothFormant filter based on a single FOF filter. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Fof filter parameters are lowpass filtered to mitigate possible noise from varying them in realtime. Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.
_ : formantFilterFofSmooth(voiceType,vowel,nFormants,i,freq) : _Where:
voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)nFormants: number of formant regions in frequency domain, typically 5i: formant number (i.e. 1 - 5) used to index formant data value arraysfreq: fundamental frequency of excitation signal. Used to calculate rise time of envelope(pm.)formantFilterBPFormant filter based on a single resonant bandpass filter. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.
_ : formantFilterBP(voiceType,vowel,nFormants,i,freq) : _Where:
voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)nFormants: number of formant regions in frequency domain, typically 5i: formant index used to index formant data value arraysfreq: fundamental frequency of excitation signal.(pm.)formantFilterbankFormant filterbank which can use different types of filterbank functions and different excitation signals. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.
_ : formantFilterbank(voiceType,vowel,formantGen,freq) : _Where:
voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)formantGen: the specific formant filterbank function (i.e. FormantFilterbankBP, FormantFilterbankFof,…)freq: fundamental frequency of excitation signal. Needed for FOF version to calculate rise time of envelope(pm.)formantFilterbankFofCycleFormant filterbank based on a bank of fof filters. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.
_ : formantFilterbankFofCycle(voiceType,vowel,freq) : _Where:
voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)freq: the fundamental frequency of the excitation signal. Needed to calculate the skirtwidth of the FOF envelopes and for the autobendFreq and vocalEffort functions(pm.)formantFilterbankFofSmoothFormant filterbank based on a bank of fof filters. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.
_ : formantFilterbankFofSmooth(voiceType,vowel,freq) : _Where:
voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)freq: the fundamental frequency of the excitation signal. Needed to calculate the skirtwidth of the FOF envelopes and for the autobendFreq and vocalEffort functions(pm.)formantFilterbankBPFormant filterbank based on a bank of resonant bandpass filters. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.
_ : formantFilterbankBP(voiceType,vowel) : _Where:
voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)freq: the fundamental frequency of the excitation signal. Needed for the autobendFreq and vocalEffort functions(pm.)SFFormantModelSimple formant/vocal synthesizer based on a source/filter model. The source and filterbank must be specified by the user. filterbank must take the same input parameters as formantFilterbank (BP/FofCycle /FofSmooth). Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the synthesized voice to be realistic.
SFFormantModel(voiceType,vowel,exType,freq,gain,source,filterbank,isFof) : _Where:
voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: uexType: voice vs. fricative sound ratio (0-1 where 1 is 100% fricative)freq: the fundamental frequency of the source signalgain: linear gain multiplier to multiply the source byisFof: whether model is FOF based (0: no, 1: yes)(pm.)SFFormantModelFofCycleSimple formant/vocal synthesizer based on a source/filter model. The source is just a periodic impulse and the “filter” is a bank of FOF filters. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the synthesized voice to be realistic. This model does not work with noise in the source signal so exType has been removed and model does not depend on SFFormantModel function.
SFFormantModelFofCycle(voiceType,vowel,freq,gain) : _Where:
voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: ufreq: the fundamental frequency of the source signalgain: linear gain multiplier to multiply the source by(pm.)SFFormantModelFofSmoothSimple formant/vocal synthesizer based on a source/filter model. The source is just a periodic impulse and the “filter” is a bank of FOF filters. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the synthesized voice to be realistic.
SFFormantModelFofSmooth(voiceType,vowel,freq,gain) : _Where:
voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: ufreq: the fundamental frequency of the source signalgain: linear gain multiplier to multiply the source by(pm.)SFFormantModelBPSimple formant/vocal synthesizer based on a source/filter model. The source is just a sawtooth wave and the “filter” is a bank of resonant bandpass filters. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the synthesized voice to be realistic.
The formant data used here come from the CSOUND manual http://www.csounds.com/manual/html/.
SFFormantModelBP(voiceType,vowel,exType,freq,gain) : _Where:
voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: uexType: voice vs. fricative sound ratio (0-1 where 1 is 100% fricative)freq: the fundamental frequency of the source signalgain: linear gain multiplier to multiply the source by(pm.)SFFormantModelFofCycle_uiReady-to-use source-filter vocal synthesizer with built-in user interface.
SFFormantModelFofCycle_ui : _(pm.)SFFormantModelFofSmooth_uiReady-to-use source-filter vocal synthesizer with built-in user interface.
SFFormantModelFofSmooth_ui : _(pm.)SFFormantModelBP_uiReady-to-use source-filter vocal synthesizer with built-in user interface.
SFFormantModelBP_ui : _(pm.)SFFormantModelFofCycle_ui_MIDIReady-to-use MIDI-controllable source-filter vocal synthesizer.
SFFormantModelFofCycle_ui_MIDI : _(pm.)SFFormantModelFofSmooth_ui_MIDIReady-to-use MIDI-controllable source-filter vocal synthesizer.
SFFormantModelFofSmooth_ui_MIDI : _(pm.)SFFormantModelBP_ui_MIDIReady-to-use MIDI-controllable source-filter vocal synthesizer.
SFFormantModelBP_ui_MIDI : _Various miscellaneous functions.
(pm.)allpassNLBidirectional block adding nonlinearities in both directions in a chain. Nonlinearities are created by modulating the coefficients of a passive allpass filter by the signal it is processing.
chain(... : allpassNL(nonlinearity) : ...)Where:
nonlinearity: amount of nonlinearity to be added (0-1)A library of reverb effects. Its official prefix is re.
(re.)jcrevThis artificial reverberator take a mono signal and output stereo (satrev) and quad (jcrev). They were implemented by John Chowning in the MUS10 computer-music language (descended from Music V by Max Mathews). They are Schroeder Reverberators, well tuned for their size. Nowadays, the more expensive freeverb is more commonly used (see the Faust examples directory).
jcrev reverb below was made from a listing of “RV”, dated April 14, 1972, which was recovered from an old SAIL DART backup tape. John Chowning thinks this might be the one that became the well known and often copied JCREV.
jcrev is a standard Faust function
_ : jcrev : _,_,_,_(re.)satrevThis artificial reverberator take a mono signal and output stereo (satrev) and quad (jcrev). They were implemented by John Chowning in the MUS10 computer-music language (descended from Music V by Max Mathews). They are Schroeder Reverberators, well tuned for their size. Nowadays, the more expensive freeverb is more commonly used (see the Faust examples directory).
satrev was made from a listing of “SATREV”, dated May 15, 1971, which was recovered from an old SAIL DART backup tape. John Chowning thinks this might be the one used on his often-heard brass canon sound examples, one of which can be found at https://ccrma.stanford.edu/~jos/wav/FM_BrassCanon2.wav
_ : satrev : _,_(re.)fdnrev0Pure Feedback Delay Network Reverberator (generalized for easy scaling). fdnrev0 is a standard Faust function.
<1,2,4,...,N signals> <:
fdnrev0(MAXDELAY,delays,BBSO,freqs,durs,loopgainmax,nonl) :>
<1,2,4,...,N signals>Where:
N: 2, 4, 8, … (power of 2)MAXDELAY: power of 2 at least as large as longest delay-line lengthdelays: N delay lines, N a power of 2, lengths perferably coprimeBBSO: odd positive integer = order of bandsplit desired at freqsfreqs: NB-1 crossover frequencies separating desired frequency bandsdurs: NB decay times (t60) desired for the various bandsloopgainmax: scalar gain between 0 and 1 used to “squelch” the reverbnonl: nonlinearity (0 to 0.999…, 0 being linear)https://ccrma.stanford.edu/~jos/pasp/FDN_Reverberation.html
(re.)zita_rev_fdnInternal 8x8 late-reverberation FDN used in the FOSS Linux reverb zita-rev1 by Fons Adriaensen fons@linuxaudio.org. This is an FDN reverb with allpass comb filters in each feedback delay in addition to the damping filters.
bus(8) : zita_rev_fdn(f1,f2,t60dc,t60m,fsmax) : bus(8)Where:
f1: crossover frequency (Hz) separating dc and midrange frequenciesf2: frequency (Hz) above f1 where T60 = t60m/2 (see below)t60dc: desired decay time (t60) at frequency 0 (sec)t60m: desired decay time (t60) at midrange frequencies (sec)fsmax: maximum sampling rate to be used (Hz)(re.)zita_rev1_stereoExtend zita_rev_fdn to include zita_rev1 input/output mapping in stereo mode. zita_rev1_stereo is a standard Faust function.
_,_ : zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax) : _,_Where:
rdel = delay (in ms) before reverberation begins (e.g., 0 to ~100 ms) (remaining args and refs as for zita_rev_fdn above)
(re.)zita_rev1_ambiExtend zita_rev_fdn to include zita_rev1 input/output mapping in “ambisonics mode”, as provided in the Linux C++ version.
_,_ : zita_rev1_ambi(rgxyz,rdel,f1,f2,t60dc,t60m,fsmax) : _,_,_,_Where:
rgxyz = relative gain of lanes 1,4,2 to lane 0 in output (e.g., -9 to 9) (remaining args and references as for zita_rev1_stereo above)
(re.)mono_freeverbA simple Schroeder reverberator primarily developed by “Jezar at Dreampoint” that is extensively used in the free-software world. It uses four Schroeder allpasses in series and eight parallel Schroeder-Moorer filtered-feedback comb-filters for each audio channel, and is said to be especially well tuned.
mono_freeverb is a standard Faust function.
_ : mono_freeverb(fb1, fb2, damp, spread) : _;Where:
fb1: coefficient of the lowpass comb filters (0-1)fb2: coefficient of the allpass comb filters (0-1)damp: damping of the lowpass comb filter (0-1)spread: spatial spread in number of samples (for stereo)While this version is licensed LGPL (with exception) along with other GRAME library functions, the file freeverb.dsp in the examples directory of older Faust distributions, such as faust-0.9.85, was released under the BSD license, which is less restrictive.
(re.)stereo_freeverbA simple Schroeder reverberator primarily developed by “Jezar at Dreampoint” that is extensively used in the free-software world. It uses four Schroeder allpasses in series and eight parallel Schroeder-Moorer filtered-feedback comb-filters for each audio channel, and is said to be especially well tuned.
_,_ : stereo_freeverb(fb1, fb2, damp, spread) : _,_;Where:
fb1: coefficient of the lowpass comb filters (0-1)fb2: coefficient of the allpass comb filters (0-1)damp: damping of the lowpass comb filter (0-1)spread: spatial spread in number of samples (for stereo)A library of basic elements to handle signal routing in Faust. Its official prefix is ro.
(ro.)crossCross n signals: (x1,x2,..,xn) -> (xn,..,x2,x1). cross is a standard Faust function.
cross(n)
_,_,_ : cross(3) : _,_,_Where:
n: number of signals (int, must be known at compile time)Special case: cross2:
cross2 = _,cross(2),_;(ro.)crossnnCross two bus(n)s.
_,_,... : crossmm(n) : _,_,...Where:
n: the number of signals in the bus(ro.)crossn1Cross bus(n) and bus(1).
_,_,... : crossn1(n) : _,_,...Where:
n: the number of signals in the first bus(ro.)interleaveInterleave rowcol cables from column order to row order. input : x(0), x(1), x(2) …, x(rowcol-1) output: x(0+0row), x(0+1row), x(0+2row), …, x(1+0row), x(1+1row), x(1+2row), …
_,_,_,_,_,_ : interleave(row,column) : _,_,_,_,_,_Where:
row: the number of row (int, known at compile time)column: the number of column (int, known at compile time)(ro.)butterflyAddition (first half) then substraction (second half) of interleaved signals.
_,_,_,_ : butterfly(n) : _,_,_,_Where:
n: size of the butterfly (n is int, even and known at compile time)(ro.)hadamardHadamard matrix function of size n = 2^k.
_,_,_,_ : hadamard(n) : _,_,_,_Where:
n: 2^k, size of the matrix (int, must be known at compile time)Implementation contributed by Remy Muller.
(ro.)recursivizeCreate a recursion from two arbitrary processors p and q.
_,_ : recursivize(p,q) : _,_
Where:
p: the forward arbitrary processorq: the feedback arbitrary processorA library of basic elements to handle signals in Faust. Its official prefix is si.
(si.)busn parallel cables. bus is a standard Faust function.
bus(n)
bus(4) : _,_,_,_Where:
n: is an integer known at compile time that indicates the number of parallel cables.(si.)blockBlock - terminate n signals. block is a standard Faust function.
_,_,... : block(n) : _,...Where:
n: the number of signals to be blocked(si.)interpolateLinear interpolation between two signals.
_,_ : interpolate(i) : _Where:
i: interpolation control between 0 and 1 (0: first input; 1: second input)(si.)smooSmoothing function based on smooth ideal to smooth UI signals (sliders, etc.) down. smoo is a standard Faust function.
hslider(...) : smoo;(si.)polySmoothA smoothing function based on smooth that doesn’t smooth when a trigger signal is given. This is very useful when making polyphonic synthesizer to make sure that the value of the parameter is the right one when the note is started.
hslider(...) : polysmooth(g,s,d) : _Where:
g: the gate/trigger signal used when making polyphonic synthss: the smoothness (see smooth)d: the number of samples to wait before the signal start being smoothed after g switched to 1(si.)smoothAndHA smoothing function based on smooth that holds its output signal when a trigger is sent to it. This feature is convenient when implementing polyphonic instruments to prevent some smoothed parameter to change when a note-off event is sent.
hslider(...) : smoothAndH(g,s) : _Where:
g: the hold signal (0 for hold, 1 for bypass)s: the smoothness (see smooth)(si.)bsmoothBlock smooth linear interpolation during a block of samples.
hslider(...) : bsmooth : _(si.)dotDot product for two vectors of size n.
_,_,_,_,_,_ : dot(n) : _Where:
n: size of the vectors (int, must be known at compile time)(si.)smoothExponential smoothing by a unity-dc-gain one-pole lowpass. smooth is a standard Faust function.
_ : smooth(tau2pole(tau)) : _Where:
tau: desired smoothing time constant in seconds, orhslider(...) : smooth(s) : _Where:
s: smoothness between 0 and 1. s=0 for no smoothing, s=0.999 is “very smooth”, s>1 is unstable, and s=1 yields the zero signal for all inputs. The exponential time-constant is approximately 1/(1-s) samples, when s is close to (but less than) 1.https://ccrma.stanford.edu/~jos/mdft/Convolution_Example_2_ADSR.html
(si.)cbusn parallel cables for complex signals. cbus is a standard Faust function.
cbus(n)
cbus(4) : (r0,i0), (r1,i1), (r2,i2), (r3,i3)Where:
n: is an integer known at compile time that indicates the number of parallel cables.(si.)cmulmultiply two complex signals pointwise. cmul is a standard Faust function.
(r1,i1) : cmul(r2,i2) : (_,_);Where:
(r1,i1) = real and imaginary parts of signal 1(r2,i2) = real and imaginary parts of signal 2(si.)lag_udLag filter with separate times for up and down.
_ : lag_ud(up, dn, signal) : _;This library contains a collection of tools for sound spatialization. Its official prefix is sp.
(sp.)pannerA simple linear stereo panner. panner is a standard Faust function.
_ : panner(g) : _,_Where:
g: the panning (0-1)(sp.)spatGMEM SPAT: n-outputs spatializer. spat is a standard Faust function.
_ : spat(n,r,d) : _,_,...Where:
n: number of outputsr: rotation (between 0 et 1)d: distance of the source (between 0 et 1)(sp.)stereoizeTransform an arbitrary processor p into a stereo processor with 2 inputs and 2 outputs.
_,_ : stereoize(p) : _,_Where:
p: the arbitrary processorThis library contains a collection of synthesizers. Its official prefix is sy.
(sy.)popFilterPercA simple percussion instrument based on a “popped” resonant bandpass filter. popFilterPerc is a standard Faust function.
popFilterDrum(freq,q,gate) : _;Where:
freq: the resonance frequency of the instrumentq: the q of the res filter (typically, 5 is a good value)gate: the trigger signal (0 or 1)(sy.)dubDubA simple synth based on a sawtooth wave filtered by a resonant lowpass. dubDub is a standard Faust function.
dubDub(freq,ctFreq,q,gate) : _;Where:
freq: frequency of the sawtoothctFreq: cutoff frequency of the filterq: Q of the filtergate: the trigger signal (0 or 1)(sy.)sawTromboneA simple trombone based on a lowpassed sawtooth wave. sawTrombone is a standard Faust function.
sawTrombone(att,freq,gain,gate) : _Where:
att: exponential attack duration in s (typically 0.01)freq: the frequencygain: the gain (0-1)gate: the gate (0 or 1)(sy.)combStringSimplest string physical model ever based on a comb filter. combString is a standard Faust function.
combString(freq,res,gate) : _;Where:
freq: the frequency of the stringres: string T60 (resonance time) in secondgate: trigger signal (0 or 1)(sy.)additiveDrumA simple drum using additive synthesis. additiveDrum is a standard Faust function.
additiveDrum(freq,freqRatio,gain,harmDec,att,rel,gate) : _Where:
freq: the resonance frequency of the drumfreqRatio: a list of ratio to choose the frequency of the mode in function of freq e.g.(1 1.2 1.5 …). The first element should always be one (fundamental).gain: the gain of each mode as a list (1 0.9 0.8 …). The first element is the gain of the fundamental.harmDec: harmonic decay ratio (0-1): configure the speed at which higher modes decay compare to lower modes.att: attack duration in secondrel: release duration in secondgate: trigger signal (0 or 1)(sy.)fmAn FM synthesizer with an arbitrary number of modulators connected as a sequence. fm is a standard Faust function.
freqs = (300,400,...);
indices = (20,...);
fm(freqs,indices) : _Where:
freqs: a list of frequencies where the first one is the frequency of the carrier and the others, the frequency of the modulator(s)indices: the indices of modulation (Nfreqs-1)A library of virtual analog filter effects. Its official prefix is ve.
(ve.)moog_vcfMoog “Voltage Controlled Filter” (VCF) in “analog” form. Moog VCF implemented using the same logical block diagram as the classic analog circuit. As such, it neglects the one-sample delay associated with the feedback path around the four one-poles. This extra delay alters the response, especially at high frequencies (see reference [1] for details). See moog_vcf_2b below for a more accurate implementation.
moog_vcf(res,fr)Where:
res: normalized amount of corner-resonance between 0 and 1 (0 is no resonance, 1 is maximum)fr: corner-resonance frequency in Hz (less than SR/6.3 or so)(ve.)moog_vcf_2b[n]Moog “Voltage Controlled Filter” (VCF) as two biquads. Implementation of the ideal Moog VCF transfer function factored into second-order sections. As a result, it is more accurate than moog_vcf above, but its coefficient formulas are more complex when one or both parameters are varied. Here, res is the fourth root of that in moog_vcf, so, as the sampling rate approaches infinity, moog_vcf(res,fr) becomes equivalent to moog_vcf_2b[n](res^4,fr) (when res and fr are constant). moog_vcf_2b uses two direct-form biquads (tf2). moog_vcf_2bn uses two protected normalized-ladder biquads (tf2np).
moog_vcf_2b(res,fr)
moog_vcf_2bn(res,fr)Where:
res: normalized amount of corner-resonance between 0 and 1 (0 is min resonance, 1 is maximum)fr: corner-resonance frequency in Hz(ve.)wah4Wah effect, 4th order. wah4 is a standard Faust function.
_ : wah4(fr) : _Where:
fr: resonance frequency in Hzhttps://ccrma.stanford.edu/~jos/pasp/vegf.html
(ve.)autowahAuto-wah effect. autowah is a standard Faust function.
_ : autowah(level) : _;Where:
level: amount of effect desired (0 to 1).(ve.)crybabyDigitized CryBaby wah pedal. crybaby is a standard Faust function.
_ : crybaby(wah) : _Where:
wah: “pedal angle” from 0 to 1https://ccrma.stanford.edu/~jos/pasp/vegf.html
(ve.)vocoderA very simple vocoder where the spectrum of the modulation signal is analyzed using a filter bank. vocoder is a standard Faust function.
_ : vocoder(nBands,att,rel,BWRatio,source,excitation) : _;Where:
nBands: Number of vocoder bandsatt: Attack time in secondsrel: Release time in secondsBWRatio: Coefficient to adjust the bandwidth of each band (0.1 - 2)source: Modulation signalexcitation: Excitation/Carrier signalPermission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files (the “Software”), to deal in the Software without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software.
Any person wishing to distribute modifications to the Software is asked to send the modifications to the original developer so that they can be incorporated into the canonical version. For software copyrighted by Julius O. Smith III, email your modifications to jos@ccrma.stanford.edu. This is, however, not a binding provision of this license.
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