NOTE: this documentation was automatically generated.
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: analyzer.libba: basic.libco: compressor.libde: delay.libdm: demo.liben: envelope.libfi: filter.libho: hoa.libma: math.libef: misceffect.libos: miscoscillator.libno: noise.libpf: phafla.libpm: pm.libre: reverb.libro: route.libsi: signal.libsp: spat.libsy: synth.libve: vaeffect.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 miscoscillator.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("miscoscillator.lib");
process = os.osc(440);
Finally, libraries can be simply imported in the Faust code (not recommended):
import("miscoscillator.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:
analyzer.libbasic.libcompressor.libdelay.libdemo.libenvelope.libfilter.libhoa.libmath.libmisceffect.libmiscoscillator.libnoise.libphafla.libpm.libreverb.libroute.libsignal.libspat.libsynth.libtonestack.lib (not documented but example in /examples/misc)tube.lib (not documented but example in /examples/misc)vaeffect.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 basic.lib for an example).#### markdown title.basic.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("analyzer.lib");
ba = library("basic.lib");
co = library("compressor.lib");
de = library("delay.lib");
dm = library("demo.lib");
en = library("envelope.lib");
fi = library("filter.lib");
ho = library("hoa.lib");
ma = library("math.lib");
ef = library("misceffect.lib");
os = library("miscoscillator.lib");
no = library("noise.lib");
pf = library("phafla.lib");
pm = library("pm.lib");
re = library("reverb.lib");
ro = library("route.lib");
sp = library("spat.lib");
si = library("signal.lib");
sy = library("synth.lib");
ve = library("vaeffect.lib");
For example, if we wanted to use the smooth function which is now declared in signal.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 signal.lib directly and call smooth without ro., etc.
"Demo" functions are placed in demo.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 filter.lib and only one of them is considered to be standard, etc.
Now that Faust libraries are not author specific, each function will be able to have its own licence/author declaration. This means that some libraries wont have a global licence/author/copyright declaration like it used to be the case.
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 demo.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 |
| 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 |
| 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.osc |
Sine oscillator (filter-based) |
| Sine (Table-Based) | os.oscsin |
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 |
This library contains a collection of tools to analyze signals.
It should be used using the an environment:
an = library("analyzer.lib");
process = an.functionCall;
Another option is to import stdfaust.lib which already contains the an environment:
import("stdfaust.lib");
process = an.functionCall;
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 downamp_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.
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 filter.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.
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 elliptic
Also 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.
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);
[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.
These are similar to the Mth-octave analyzers above, except that the band-split frequencies are passed explicitly as arguments.
analyzerAnalyzer.
_ : analyzer(O,freqs) : par(i,N,_); // No delay equalizer
Where:
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)) : _,_,_
A library of basic elements for Faust organized in 5 sections:
It should be used using the ba environment:
ba = library("basic.lib");
process = ba.functionCall;
Another option is to import stdfaust.lib which already contains the ba environment:
import("stdfaust.lib");
process = ba.functionCall;
samp2secConverts a number of samples to a duration in seconds. samp2sec is a standard Faust function.
samp2sec(n) : _
Where:
n: number of samplessec2sampConverts a duration in seconds to a number of samples. samp2sec is a standard Faust function.
sec2samp(d) : _
Where:
d: duration in secondsdb2linearConverts a loudness in dB to a linear gain (0-1). db2linear is a standard Faust function.
db2linear(l) : _
Where:
l: loudness in dBlinear2dbConverts a linear gain (0-1) to a loudness in dB. linear2db is a standard Faust function.
linear2db(g) : _
Where:
g: a linear gainlin2LogGainConverts a linear gain (0-1) to a log gain (0-1).
_ : lin2LogGain : _
log2LinGainConverts a log gain (0-1) to a linear gain (0-1).
_ : log2LinGain : _
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 secondspole2tauReturns 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 polemidikey2hzConverts 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 numberpianokey2hzConverts a piano key number to a frequency in Hz (piano key 49 = A440).
pianokey2hz(pk) : _
Where:
pk: the piano key numberhz2pianokeyConverts a frequency in Hz to a piano key number (piano key 49 = A440).
hz2pianokey(f) : _
Where:
f: frequency in HzcountdownStarts 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)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 starting point of the countuptrig: the trigger signal (1: start at 0; 0: increase until n)sweepCounts from 0 to period samples repeatedly, while run is 1. Outsputs zero while run is 0.
sweep(period,run) : _
timeA simple timer that counts every samples from the beginning of the process. time is a standard Faust function.
time : _
tempoConverts a tempo in BPM into a number of samples.
tempo(t) : _
Where:
t: tempo in BPMperiodBasic sawtooth wave of period p.
period(p) : _
Where:
p: period as a number of samplespulsePulses (10000) generated at period p.
pulse(p) : _
Where:
p: period as a number of samplespulsenPulses (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 samplesbeatPulses at tempo t. beat is a standard Faust function.
beat(t) : _
Where:
t: tempo in BPMpulse_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)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)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)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)countCount the number of elements of list l. count is a standard Faust function.
count(l)
count ((10,20,30,40)) -> 4
Where:
l: list of elementstakeTake an element from a list. take is a standard Faust function.
take(e,l)
take(3,(10,20,30,40)) -> 30
Where:
p: position (starting at 1)l: list of elementssubseqExtract 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) -> 50
Where:
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
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 falseselectorSelects the ith input among n at compile time.
selector(i,n)
_,_,_,_ : selector(2,4) : _ // selects the 3rd input among 4
Where:
i: input to select (int, numbered from 0, known at compile time)n: number of inputs (int, known at compile time, n > i)selectnSelects the ith input among N at run time.
selectn(N,i)
_,_,_,_ : selectn(4,2) : _ // selects the 3rd input among 4
Where:
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)));
select2stereoSelect between 2 stereo signals.
_,_,_,_ : select2stereo(bpc) : _,_,_,_
Where:
bpc: the selector switch (0/1)latchLatch input on positive-going transition of "clock" ("sample-and-hold").
_ : latch(clocksig) : _
Where:
clocksig: hold trigger (0 for hold, 1 for bypass)sAndHSample And Hold. sAndH is a standard Faust function.
_ : sAndH(t) : _
Where:
t: hold trigger (0 for hold, 1 for bypass)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.
peakholderTracks abs peak and holds peak for 'holdtime' samples.
_ : peakholder(holdtime) : _;
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 ;
automatRecord and replay to the values the input signal in a loop.
hslider(...) : automat(bps, size, init) : _
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.
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 effectbypass2Takes 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 effecttoggleTriggered 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
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)
selectoutnRoute input to the output among N at run time.
_ : selectoutn(n, s) : _,_,...n
Where:
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);
A library of compressor effects.
It should be used using the co environment:
co = library("compressor.lib");
process = co.functionCall;
Another option is to import stdfaust.lib which already contains the co environment:
import("stdfaust.lib");
process = co.functionCall;
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 compressioncompressor_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 compressionlimiter_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
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.
It should be used using the de environment:
de = library("delay.lib");
process = de.functionCall;
Another option is to import stdfaust.lib which already contains the de environment:
import("stdfaust.lib");
process = de.functionCall;
delaySimple d samples delay where n is the maximum delay length as a number of samples (it needs to be a power of 2). Unlike the @ delay operator, this function allows to preallocate memory which means that d can be changed dynamically at run time as long as it remains smaller than n. delay is a standard Faust function.
_ : delay(n,d) : _
Where:
n: the max delay length as a power of 2d: the delay length as a number of samples (integer)fdelaySimple d samples fractional delay based on 2 interpolated delay lines where n is the maximum delay length as a number of samples (it needs to be a power of 2 - see delay()). fdelay is a standard Faust function.
_ : fdelay(n,d) : _
Where:
n: the max delay length as a power of 2d: the delay length as a number of samples (float)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 samples (must be a constant power of 2, for example 65536)it: interpolation time (in samples) for example 1024dt: delay time (in samples)fdelaylti and 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.
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
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.
It should be used using the dm environment:
dm = library("demo.lib");
process = dm.functionCall;
Another option is to import stdfaust.lib which already contains the dm environment:
import("stdfaust.lib");
process = dm.functionCall;
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
parametric_eq_demoA parametric equalizer application.
_ : parametric_eq_demo : _ ;
spectral_tilt_demoA spectral tilt application.
_ : spectral_tilt_demo(N) : _ ;
Where:
N: filter order (integer)All other parameters interactive
mth_octave_filterbank_demo and 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 octavecubicnl_demoDistortion demo application.
_ : cubicnl_demo : _;
gate_demoGate demo application.
_,_ : gate_demo : _,_;
compressor_demoCompressor demo application.
_,_ : compressor_demo : _,_;
exciterPsychoacoustic harmonic exciter, with GUI.
_ : exciter : _
moog_vcf_demoIllustrate and compare all three Moog VCF implementations above.
_ : moog_vcf_demo : _;
wah4_demoWah pedal application.
_ : wah4_demo : _;
crybaby_demoCrybaby effect application.
_ : crybaby_demo : _ ;
vocoder_demoUse example of the vocoder function where an impulse train is used as excitation.
_ : vocoder_demo : _;
flanger_demoFlanger effect application.
_,_ : flanger_demo : _,_;
phaser2_demoPhaser effect demo application.
_,_ : phaser2_demo : _,_;
freeverb_demoFreeverb demo application.
_,_ : freeverb_demo : _,_;
stereo_reverb_testerHandy test inputs for reverberator demos below.
_ : stereo_reverb_tester : _
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 integerzita_rev_fdn_demoReverb demo application based on zita_rev_fdn.
si.bus(8) : zita_rev_fdn_demo : si.bus(8)
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
sawtooth_demoAn application demonstrating the different sawtooth oscillators of Faust.
sawtooth_demo : _
virtual_analog_oscillator_demoVirtual analog oscillator demo application.
virtual_analog_oscillator_demo : _
oscrs_demoSimple application demoing filter based oscillators.
oscrs_demo : _
This library contains a collection of envelope generators.
It should be used using the en environment:
en = library("envelope.lib");
process = en.functionCall;
Another option is to import stdfaust.lib which already contains the en environment:
import("stdfaust.lib");
process = en.functionCall;
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 (0-1)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 (0 or 1)asrASR (Attack, Sustain, Release) envelope generator. asr is a standard Faust function.
asr(a,s,r,t) : _
Where:
a, s, r: attack (sec), sustain (percentage of t), release (sec)t: trigger signal ( >0 for attack, then release is when t back to 0)adsrADSR (Attack, Decay, Sustain, Release) envelope generator. adsr is a standard Faust function.
adsr(a,d,s,r,t) : _
Where:
a, d, s, r: attack (sec), decay (sec), sustain (percentage of t), release (sec)t: trigger signal ( >0 for attack, then release is when t back to 0)A library of filters and of more advanced filter-based sound processor organized in 18 sections:
It should be used using the fi environment:
fi = library("filter.lib");
process = fi.functionCall;
Another option is to import stdfaust.lib which already contains the fi environment:
import("stdfaust.lib");
process = fi.functionCall;
zeroOne zero filter. Difference equation: y(n) = x(n) - z * x(n-1).
_ : zero(z) : _
Where:
z: location of zero along real axis in z-planehttps://ccrma.stanford.edu/~jos/filters/One_Zero.html
poleOne pole filter. Could also be called a "leaky integrator". Difference equation: y(n) = x(n) + p * y(n-1).
_ : pole(z) : _
Where:
p: pole location = feedback coefficienthttps://ccrma.stanford.edu/~jos/filters/One_Pole.html
integratorSame as pole(1) [implemented separately for block-diagram clarity].
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) = s / (s + 2PIfb) 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
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 : _
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
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
ffcombfilterTypical special case of ff_comb() where: b0 = 1.
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
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
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 reverb.lib for usage examples.
fbcombfilter and 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
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 gainallpass_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 gainrev2Special 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 reverb.lib for usage examples.
allpass_fcomb5 and 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);`).
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
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));
conv and 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,...))
tf1, tf2 and 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
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
tf21, tf22, tf22t and 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.
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).
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
iir_lat2Two-multiply latice IIR filter or arbitrary order.
_ : iir_lat2(bv,av) : _
Where:
allpassntTwo-multiply lattice allpass (nested order-1 direct-form-ii allpasses).
_ : allpassnt(n,sv) : _
Where:
n: the order of the filtersv: the reflexion coefficients (-1 1)iir_klKelly-Lochbaum ladder IIR filter or arbitrary order.
_ : iir_kl(bv,av) : _
Where:
allpassnkltKelly-Lochbaum ladder allpass.
_ : allpassklt(n,sv) : _
Where:
n: the order of the filtersv: the reflexion coefficients (-1 1)iir_lat1One-multiply latice IIR filter or arbitrary order.
_ : iir_lat1(bv,av) : _
Where:
allpassn1mtOne-multiply lattice allpass with tap lines.
_ : allpassn1mt(n,sv) : _
Where:
n: the order of the filtersv: the reflexion coefficients (-1 1)iir_nlNormalized ladder filter of arbitrary order.
_ : iir_nl(bv,av) : _
Where:
allpassnnltNormalized ladder allpass filter of arbitrary order.
_ : allpassnnlt(n,sv) : _
Where:
n: the order of the filtersv: the reflexion coefficients (-1,1)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 zeroswgrSecond-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)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
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*.
allpassnTwo-multiply lattice - each section is two multiply-adds.
_ : allpassn(n,sv) : _
n: the order of the filtersv: the reflexion coefficients (-1 1)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 reflexion coefficients (-PI PI)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 reflexion coefficients (-1 1)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 reflexion coefficients (-1 1)tf2s and 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 + a0
and 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 + 1
where 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
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) : _
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 + b0
H(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
1
H(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 Butterworth
Bilinear transform scaled for exact mapping of w1.
https://ccrma.stanford.edu/~jos/pasp/Bilinear_Transformation.html
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) : _
tf1sbFirst-to-second-order lowpass-to-bandpass section mapping, analogous to tf2sb above.
_ : tf1sb(b1,b0,a0,w1,wc) : _
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)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)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)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');")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');")lowpass0_highpass1TODO
These 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).
lowpass_plus|minus_highpassTODO
Elliptic (Cauer) Lowpass Filters
ncauer and ellip in Octavelowpass3eThird-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
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
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 Hzhighpass6eSixth-order Elliptic/Cauer highpass filter. Inversion of lowpass3e wrt unit circle in s plane (s <- 1/s)
_ : highpass6e(fc) : _
Where:
fc: -3dB frequency in HzbandpassOrder 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/
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/
bandpass6eOrder 12 elliptic bandpass filter analogous to bandpass(6).
bandpass12eOrder 24 elliptic bandpass filter analogous to bandpass(6).
Parametric Equalizers (Shelf, Peaking)
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.
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.
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 Hzpeak_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 Hzpeak_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.)
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 bandbw: bandwidth of desired roll-off bandalpha: slope of roll-off desired in nepers per neper (ln mag / ln radian freq)See spectral_tilt_demo.
Link to appear here when write up is done
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
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.
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 version
Also for convenience:
_ : mth_octave_filterbank3(M,ftop,N) : par(i,N,_); // 3d-order Butterworth
_ : mth_octave_filterbank5(M,ftop,N) : par(i,N,_); // 5th-roder Butterworth
mth_octave_filterbank_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)These are similar to the Mth-octave analyzers above, except that the band-split frequencies are passed explicitly as arguments.
filterbankFilter bank. filterbank is a standard Faust function.
_ : filterbank (O,freqs) : par(i,N,_); // Butterworth band-splits
Where:
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)) : _,_,_
filterbankiInverted-dc filter bank.
_ : filterbanki(O,freqs) : par(i,N,_); // Inverted-dc version
Where:
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.
It should be used using the ho environment:
ho = library("ho.lib");
process = ho.functionCall;
Another option is to import stdfaust.lib which already contains the ho environment:
import("stdfaust.lib");
process = ho.functionCall;
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 angledecoderDecodes 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.
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.
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 orderoptimMaxReThe 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 orderoptimInPhaseThe inPhase Optimization optimize energy vector and put all loudspeakers signals n phase. It should be used for an auditory.
here:
n: the order
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 - 1mapIt 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 radianrotateRotates the sound field.
_ : rotate(n, a) : _
Where:
n: the ordera: the angle in radianMathematic library for Faust. Some functions are implemenented as Faust foreign functions of math.h functions that are not part of Faust's primitives. Defines also various constants and several utilities.
It should be used using the fi environment:
ma = library("math.lib");
process = ma.functionCall;
Another option is to import stdfaust.lib which already contains the ma environment:
import("stdfaust.lib");
process = ma.functionCall;
SRCurrent sampling rate (between 1Hz and 192000Hz). Constant during program execution.
SR : _
BSCurrent block-size. Can change during the execution.
BS : _
PIConstant PI in double precisio.n
PI : _
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
negInvert the sign (-x) of a signal.
_ : neg : _
sub(x,y)Subtract x and y.
invCompute the inverse (1/x) of the input signal.
_ : inv : _
cbrtComputes the cube root of of the input signal.
_ : cbrt : _
hypotComputes the euclidian distance of the two input signals sqrt(xx+yy) without undue overflow or underflow.
_,_ : hypot : _
ldexpTakes two input signals: x and n, and multiplies x by 2 to the power n.
_,_ : ldexp : _
scalbTakes two input signals: x and n, and multiplies x by 2 to the power n.
_,_ : scalb : _
log1pComputes log(1 + x) without undue loss of accuracy when x is nearly zero.
_ : log1p : _
logbReturn exponent of the input signal as a floating-point number.
_ : logb : _
ilogbReturn exponent of the input signal as an integer number.
_ : ilogb : _
log2Returns the base 2 logarithm of x.
_ : log2 : _
expm1Return exponent of the input signal minus 1 with better precision.
_ : expm1 : _
acoshComputes the principle value of the inverse hyperbolic cosine of the input signal.
_ : acosh : _
asinhComputes the inverse hyperbolic sine of the input signal.
_ : asinh : _
atanhComputes the inverse hyperbolic tangent of the input signal.
_ : atanh : _
sinhComputes the hyperbolic sine of the input signal.
_ : sinh : _
coshComputes the hyperbolic cosine of the input signal.
_ : cosh : _
tanhComputes the hyperbolic tangent of the input signal.
_ : tanh : _
erfComputes the error function of the input signal.
_ : erf : _
erfcComputes the complementary error function of the input signal.
_ : erfc : _
gammaComputes the gamma function of the input signal.
_ : gamma : _
lgammaCalculates the natural logorithm of the absolute value of the gamma function of the input signal.
_ : lgamma : _
J0Computes the Bessel function of the first kind of order 0 of the input signal.
_ : J0 : _
J1Computes the Bessel function of the first kind of order 1 of the input signal.
_ : J1 : _
JnComputes the Bessel function of the first kind of order n (first input signal) of the second input signal.
_,_ : Jn : _
Y0Computes the linearly independent Bessel function of the second kind of order 0 of the input signal.
_ : Y0 : _
Y1Computes the linearly independent Bessel function of the second kind of order 1 of the input signal.
_ : Y0 : _
YnComputes the linearly independent Bessel function of the second kind of order n (first input signal) of the second input signal.
_,_ : Yn : _
fabs, fmax, fminJust for compatibility...
fabs = abs
fmax = max
fmin = min
np2Gives the next power of 2 of x.
np2(n) : _
Where:
n: an integerfracGives the fractional part of n.
frac(n) : _
Where:
n: a decimal numberisnanReturn non-zero if and only if x is a NaN.
isnan(x)
_ : isnan : _
Where:
x: signal to analysechebychevChebychev 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
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
diffnNegated first-roder difference.
_ : diffn : _
This library contains a collection of audio effects.
It should be used using the ef environment:
ef = library("misceffect.lib");
process = ef.functionCall;
Another option is to import stdfaust.lib which already contains the ef environment:
import("stdfaust.lib");
process = ef.functionCall;
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.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)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)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) : _
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.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".
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 coefficienttransposeA 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)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 forever
In 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.
This library contains a collection of sound generators.
It should be used using the os environment:
os = library("miscoscillator.lib");
process = os.functionCall;
Another option is to import stdfaust.lib which already contains the os environment:
import("stdfaust.lib");
process = os.functionCall;
sinwaveformSine waveform ready to use with a rdtable.
sinwaveform(tablesize) : _
Where:
tablesize: the table sizecoswaveformCosine waveform ready to use with a rdtable.
coswaveform(tablesize) : _
Where:
tablesize: the table sizephasorA 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)oscsinSine wave oscillator. oscsin is a standard Faust function.
oscsin(freq) : _
Where:
freq: the frequency of the wave (Hz)oscosCosine wave oscillator.
osccos(freq) : _
Where:
freq: the frequency of the wave (Hz)oscpA sine wave generator with controllable phase.
oscp(freq,p) : _
Where:
freq: the frequency of the wave (Hz)p: the phase in radianosciInterpolated phase sine wave oscillator.
osci(freq) : _
Where:
freq: the frequency of the wave (Hz)Low-frequency oscillators have prefix lf_ (no aliasing suppression, signal-means not necessarily zero).
lf_imptrainUnit-amplitude low-frequency impulse train. lf_imptrain is a standard Faust function.
lf_imptrain(freq) : _
Where:
freq: frequency in Hzlf_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 1lf_squarewaveposPositive LF square wave in [0,1]
lf_squarewavepos(freq) : _
Where:
freq: frequency in Hzlf_squarewaveZero-mean unit-amplitude LF square wave. lf_squarewave is a standard Faust function.
lf_squarewave(freq) : _
Where:
freq: frequency in Hzlf_triangleposPositive unit-amplitude LF positive triangle wave
lf_trianglepos(freq) : _
Where:
freq: frequency in Hzlf_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).
lf_rawsawSimple sawtooth waveform oscillator between 0 and period in samples.
lf_rawsaw(periodsamps)
Where:
periodsamps: number of periods per sampleslf_sawposSimple sawtooth waveform oscillator between 0 and 1.
lf_sawpos(freq)
Where:
freq: frequencylf_sawSimple sawtooth waveform. lf_saw is a standard Faust function.
lf_saw(freq)
Where:
freq: frequencylf_sawpos_phaseSimple sawtooth waveform oscillator between 0 and 1 with phase control.
lf_sawpos_phase(freq,phase)
Where:
freq: frequencyphase: phaseBandlimited 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: phasesawNTODO: implemented but not documented. For now, you can look at the source code.
sawNpTODO: implemented but not documented. For now, you can look at the source code.
saw2dpwTODO: implemented but not documented. For now, you can look at the source code.
saw3TODO: implemented but not documented. For now, you can look at the source code.
sawtoothAlias-free sawtooth wave. 2nd order interpolation (based on saw2). sawtooth is a standard Faust function.
sawtooth(freq) : _
Where:
freq: frequencysaw2f2TODO: implemented but not documented. For now, you can look at the source code.
saw2f4TODO: implemented but not documented. For now, you can look at the source code.
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 HzpulsetrainNTODO: implemented but not documented. For now, you can look at the source code.
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 1squareNTODO: implemented but not documented. For now, you can look at the source code.
squareBandlimited square wave oscillator. Based on squareN(2). square is a standard Faust function.
square(freq) : _
Where:
freq: frequencyimpulseOne-time impulse generated when the Faust process is started. impulse is a standard Faust function.
impulse : _
imptrainNTODO: implemented but not documented. For now, you can look at the source code.
imptrainBandlimited impulse train generator. Based on imptrainN(2). imptrain is a standard Faust function.
imptrain(freq) : _
Where:
freq: frequencytriangleNTODO: implemented but not documented. For now, you can look at the source code.
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.
oscbSinusoidal oscillator based on the biquad.
oscb(freq) : _
Where:
freq: frequencyoscrqSinusoidal (sine and cosine) oscillator based on 2D vector rotation, = undamped "coupled-form" resonator = lossless 2nd-order normalized ladder filter.
oscrq(freq) : _,_
Where:
freq: frequencyoscrsSinusoidal (sine) oscillator based on 2D vector rotation, = undamped "coupled-form" resonator = lossless 2nd-order normalized ladder filter.
oscrs(freq) : _
Where:
freq: frequencyoscrcSinusoidal (cosine) oscillator based on 2D vector rotation, = undamped "coupled-form" resonator = lossless 2nd-order normalized ladder filter.
oscrc(freq) : _
Where:
freq: frequencyoscDefault sine wave oscillator (same as oscrs). osc is a standard Faust function.
osc(freq) : _
Where:
freq: the frequency of the wave (Hz)oscsSinusoidal oscillator based on the state variable filter = undamped "modified-coupled-form" resonator = "magic circle" algorithm used in graphics
Sinusoidal oscillator based on the waveguide resonator wgr.
oscwSinusoidal oscillator based on the waveguide resonator wgr. Unit-amplitude cosine oscillator.
oscwc(freq) : _
Where:
freq: frequencyoscwsSinusoidal oscillator based on the waveguide resonator wgr. Unit-amplitude sine oscillator
oscws(freq) : _
Where:
freq: frequencyoscwqSinusoidal oscillator based on the waveguide resonator wgr. Unit-amplitude cosine and sine (quadrature) oscillator.
oscwq(freq) : _
Where:
freq: frequencyoscwSinusoidal oscillator based on the waveguide resonator wgr. Unit-amplitude cosine oscillator (default)
oscw(freq) : _
Where:
freq: frequencyA library of noise generators.
It should be used using the no environment:
no = library("noise.lib");
process = no.functionCall;
Another option is to import stdfaust.lib which already contains the no environment:
import("stdfaust.lib");
process = no.functionCall;
noiseWhite noise generator (outputs random number between -1 and 1). Noise is a standard Faust function.
noise : _
multirandomGenerates multiple decorrelated random numbers in parallel.
multirandom(n) : _
Where:
n: the number of decorrelated random numbers in parallelmultinoiseGenerates multiple decorrelated noises in parallel.
multinoise(n) : _
Where:
n: the number of decorrelated random numbers in parallelnoisesTODO.
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
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).lfnoise, lfnoise0 and 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 filter.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
A library of compressor effects.
It should be used using the pf environment:
pf = library("phafla.lib");
process = pf.functionCall;
Another option is to import stdfaust.lib which already contains the pf environment:
import("stdfaust.lib");
process = pf.functionCall;
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
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
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:
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.
It should be used using the fi environment:
pm = library("pm.lib");
process = pm.functionCall;
Another option is to import stdfaust.lib which already contains the pm environment:
import("stdfaust.lib");
process = pm.functionCall;
Creates a chain of bidirectional blocks. Blocks must have 3 inputs and outputs. The first input/output correspond to the left going signal, the second input/output correspond to the right going signal and the third input/output is the mix of the main signal output. 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 the left and right going waves. ### Usage
rightGoingWaves,leftGoingWaves,mixedOutput : chain(A:B) : rightGoingWaves,leftGoingWaves,mixedOutput
with{
A = _,_,_;
B = _,_,_;
};
filter.lib (crossnn)
Adds a waveguide input anywhere between 2 blocks in a chain of blocks (see chain()). ### Usage
string(x) = chain(A:input(x):B)
Where x is the input signal to be added to the chain.
Adds a waveguide output anywhere between 2 blocks in a chain of blocks and sends it to the mix output channel (see chain()). ### Usage
chain(A:output:B)
Creates 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. ### Usage
rightGoingWaves,leftGoingWaves,mixedOutput : terminations(a,b,c) : rightGoingWaves,leftGoingWaves,mixedOutput
with{
a = *(-1); // left termination
b = chain(D:E:F); // bidirectional chain of blocks (D, E, F, etc.)
c = *(-1); // right termination
};
filter.lib (crossnn)
Same as terminations() but closes the inputs and outputs of the bidirectional chain (only the mixed output remains). ### Usage
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
};
filter.lib (crossnn)
Creates 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. ### Usage
rightGoingWaves,leftGoingWaves,mixedOutput : terminations(a,b) : rightGoingWaves,leftGoingWaves,mixedOutput
with{
a = *(-1); // left termination
b = chain(D:E:F); // bidirectional chain of blocks (D, E, F, etc.)
};
filter.lib (crossnn)
Creates 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. ### Usage
rightGoingWaves,leftGoingWaves,mixedOutput : terminations(b,c) : rightGoingWaves,leftGoingWaves,mixedOutput
with{
b = chain(D:E:F); // bidirectional chain of blocks (D, E, F, etc.)
c = *(-1); // right termination
};
filter.lib (crossnn)
A simple waveguide block based on a 4th order fractional delay. ### Usage
rightGoingWaves,leftGoingWaves,mixedOutput : waveguide(nMax,n) : rightGoingWaves,leftGoingWaves,mixedOutput
With: * nMax: the maximum length of the waveguide in samples * n the length of the waveguide in samples. ### Requires filter.lib (fdelay4)
An ideal string with rigid terminations and where the plucking position and the pick-up position are the same. ### Usage
1-1' : idealString(length,reflexion,xPosition,x)
With: * length: the length of the string in meters * reflexion: the coefficient of reflexion (0-0.99999999) * pluckPosition: the plucking position (0.001-0.999) * x: the input signal for the excitation ### Requires filter.lib (fdelay4,crossnn)
A library of reverb effects.
It should be used using the re environment:
re = library("reverb.lib");
process = re.functionCall;
Another option is to import stdfaust.lib which already contains the re environment:
import("stdfaust.lib");
process = re.functionCall;
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 : _,_,_,_
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 : _,_
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)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)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
zita_rev_fdnInternal 8x8 late-reverberation FDN used in the FOSS Linux reverb zita-rev1 by Fons Adriaensen . 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)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)
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)
A library of basic elements to handle signal routing in Faust.
It should be used using the si environment:
ro = library("route.lib");
process = ro.functionCall;
Another option is to import stdfaust.lib which already contains the si environment:
import("stdfaust.lib");
process = ro.functionCall;
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),_;
crossnnCross two bus(n)s.
_,_,... : crossmm(n) : _,_,...
Where:
n: the number of signals in the buscrossn1Cross bus(n) and bus(1).
_,_,... : crossn1(n) : _,_,...
Where:
n: the number of signals in the first businterleaveInterleave 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)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)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.
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.
It should be used using the si environment:
si = library("signal.lib");
process = si.functionCall;
Another option is to import stdfaust.lib which already contains the si environment:
import("stdfaust.lib");
process = si.functionCall;
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.blockBlock - terminate n signals. block is a standard Faust function.
_,_,... : block(n) : _,...
Where:
n: the number of signals to be blockedinterpolateLinear interpolation between two signals.
_,_ : interpolate(i) : _
Where:
i: interpolation control between 0 and 1 (0: first input; 1: second input)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
smooSmoothing function based on smooth ideal to smooth UI signals (sliders, etc.) down. smoo is a standard Faust function.
hslider(...) : smoo;
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 1bsmoothBlock smooth linear interpolation during a block of samples.
hslider(...) : bsmooth : _
lag_udLag filter with separate times for up and down.
_ : lag_ud(up, dn, signal) : _;
dotDot product for two vectors of size n.
_,_,_,_,_,_ : dot(n) : _
Where:
n: size of the vectors (int, must be known at compile time)This library contains a collection of tools for sound spatialization.
It should be used using the sp environment:
sp = library("spat.lib");
process = sp.functionCall;
Another option is to import stdfaust.lib which already contains the sp environment:
import("stdfaust.lib");
process = sp.functionCall;
pannerA simple linear stereo panner. panner is a standard Faust function.
_ : panner(g) : _,_
Where:
g: the panning (0-1)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)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 envelope generators.
It should be used using the sy environment:
sy = library("synth.lib");
process = sy.functionCall;
Another option is to import stdfaust.lib which already contains the sy environment:
import("stdfaust.lib");
process = sy.functionCall;
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)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)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)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)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)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.
It should be used using the ve environment:
ve = library("vaeffect.lib");
process = ve.functionCall;
Another option is to import stdfaust.lib which already contains the ve environment:
import("stdfaust.lib");
process = ve.functionCall;
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:
fr: corner-resonance frequency in Hz ( less than SR/6.3 or so )res: Normalized amount of corner-resonance between 0 and 1 (0 is no resonance, 1 is maximum)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:
fr: corner-resonance frequency in Hzres: Normalized amount of corner-resonance between 0 and 1 (0 is min resonance, 1 is maximum)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
autowahAuto-wah effect. autowah is a standard Faust function.
_ : autowah(level) : _;
Where:
level: amount of effect desired (0 to 1).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
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 signal