Faust Libraries

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 ;)!

Using the Faust Libraries

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.lib
  • an: analyzers.lib
  • ba: basics.lib
  • co: compressors.lib
  • de: delays.lib
  • dm: demos.lib
  • dx: dx7.lib
  • en: envelopes.lib
  • fi: filters.lib
  • ho: hoa.lib
  • ma: maths.lib
  • ef: misceffects.lib
  • os: oscillators.lib
  • no: noises.lib
  • pf: phaflangers.lib
  • pm: physmodels.lib
  • re: reverbs.lib
  • ro: routes.lib
  • si: signals.lib
  • sp: spats.lib
  • sy: synths.lib
  • ve: vaeffects.lib

Environments can then be used as follows in your Faust code:

import("stdfaust.lib");
process = os.osc(440);

In this case, we're calling the osc function from oscillators.lib.

You can also access all the functions of all the libraries directly using the sf environment:

import("stdfaust.lib");
process = sf.osc(440);

Alternatively, environments can be created by hand:

os = library("oscillators.lib");
process = os.osc(440);

Finally, libraries can be simply imported in the Faust code (not recommended):

import("oscillators.lib");
process = osc(440);

Contributing

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:

New Functions

  • All functions must be preceded by a markdown documentation header respecting the following format (open the source code of any of the libraries for an example):
//-----------------functionName--------------------
// Description
//
// #### Usage
//
// ```
// Usage Example
// ```
//
// Where:
//
// * argument1: argument 1 description
//-------------------------------------------------
  • Every time a new function is added, the documentation should be updated simply by running make doclib.
  • The environment system (e.g. os.osc) should be used when calling a function declared in another library (see the section on Using the Faust Libraries).
  • Try to reuse exisiting functions as much as possible.
  • If you have any question, send an e-mail to rmichon_at_ccrma_dot_stanford_dot_edu.

New Libraries

  • Any new "standard" library should be declared in stdfaust.lib with its own environment (2 letters - see stdfaust.lib).
  • Any new "standard" library must be added to generateDoc.
  • Functions must be organized by sections.
  • Any new library should at least declare a name and a version.
  • The comment based markdown documentation of each library must respect the following format (open the source code of any of the libraries for an example):
//############### 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
//==============================================
  • If you have any question, send an e-mail to rmichon_at_ccrma_dot_stanford_dot_edu.

General Organization

Only the libraries that are considered to be "standard" are documented:

  • analyzers.lib
  • basics.lib
  • compressors.lib
  • delays.lib
  • demos.lib
  • dx7.lib
  • envelopes.lib
  • filters.lib
  • hoa.lib
  • maths.lib
  • misceffects.lib
  • oscillators.lib
  • noises.lib
  • phaflangers.lib
  • physmodels.lib
  • reverbs.lib
  • routes.lib
  • signals.lib
  • spats.lib
  • synths.lib
  • tonestacks.lib (not documented but example in /examples/misc)
  • tubes.lib (not documented but example in /examples/misc)
  • vaeffects.lib

Other 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.

Coding Conventions

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 :-) ).

Documentation

  • All the functions that we want to be "public" are documented.
  • We used the faust2md "standards" for each library: //### for main title (library name - equivalent to # in markdown), //=== for section declarations (equivalent to ## in markdown) and //--- for function declarations (equivalent to #### in markdown - see basics.lib for an example).
  • Sections in function documentation should be declared as #### markdown title.
  • Each function documentation provides a "Usage" section (see basics.lib).

Library Import

To prevent cross-references between libraries we generalized the use of the library("") system for function calls in all the libraries. This means that everytime a function declared in another library is called, the environment corresponding to this library needs to be called too. To make things easier, a stdfaust.lib library was created and is imported by all the libraries:

an = library("analyzers.lib");
ba = library("basics.lib");
co = library("compressors.lib");
de = library("delays.lib");
dm = library("demos.lib");
dx = library("dx7.lib");
en = library("envelopes.lib");
fi = library("filters.lib");
ho = library("hoa.lib");
ma = library("maths.lib");
ef = library("misceffects.lib");
os = library("oscillators.lib");
no = library("noises.lib");
pf = library("phaflangers.lib");
pm = library("physmodels.lib");
re = library("reverbs.lib");
ro = library("routes.lib");
sp = library("spats.lib");
si = library("signals.lib");
sy = library("synths.lib");
ve = library("vaeffects.lib");

For example, if we wanted to use the smooth function which is now declared in signals.lib, we would do the following:

import("stdfaust.lib");

process = si.smooth(0.999);

This standard is only used within the libraries: nothing prevents coders to still import signals.lib directly and call smooth without ro., etc.

"Demo" Functions

"Demo" functions are placed in demos.lib and have a built-in user interface (UI). Their name ends with the _demo suffix. Each of these function have a .dsp file associated to them in the /examples folder.

Any function containing UI elements should be placed in this library and respect these standards.

"Standard" Functions

"Standard" functions are here to simplify the life of new (or not so new) Faust coders. They are declared in /libraries/doc/standardFunctions.md and allow to point programmers to preferred functions to carry out a specific task. For example, there are many different types of lowpass filters declared in filters.lib and only one of them is considered to be standard, etc.

Now that Faust libraries are less author specific, each function will normally have its own copyright-and-license line in the library source (the .lib file, such as analyzers.lib). If not, see if the function is defined within a section of the .lib file stating the license in source-code comments. If not, then the copyright and license given at the beginning of the .lib file may be assumed, when present. If not, run git blame on the .lib file and ask the person who last edited the function!

Note that it is presently possible for a library function released under one license to utilize another library function having some different license. There is presently no indication of this situation in the Faust compiler output, but such notice is planned. For now, library contributors should strive to use only library functions having compatible licenses, and concerned end-users must manually determine the union of licenses applicable to the library functions they are using.

Standard Functions

Dozens of functions are implemented in the Faust libraries and many of them are very specialized and not useful to beginners or to people who only need to use Faust for basic applications. This section offers an index organized by categories of the "standard Faust functions" (basic filters, effects, synthesizers, etc.). This index only contains functions without a user interface (UI). Faust functions with a built-in UI can be found in demos.lib.

Analysis Tools

Function Type Function Name Description
Amplitude Follower an.amp_follower Classic analog audio envelope follower
Octave Analyzers an.mth_octave_analyzer[N] Octave analyzers

Basic Elements

Function Type Function Name Description
Beats ba.beat Pulses at a specific tempo
Block si.block Terminate n signals
Break Point Function ba.bpf Beak Point Function (BPF)
Bus si.bus Bus of n signals
Bypass (Mono) ba.bypass1 Mono bypass
Bypass (Stereo) ba.bypass2 Stereo bypass
Count Elements ba.count Count elements in a list
Count Down ba.countdown Samples count down
Count Up ba.countup Samples count up
Delay (Integer) de.delay Integer delay
Delay (Float) de.fdelay Fractional delay
Down Sample ba.downSample Down sample a signal
Impulsify ba.impulsify Turns a signal into an impulse
Sample and Hold ba.sAndH Sample and hold
Signal Crossing ro.cross Cross n signals
Smoother (Default) si.smoo Exponential smoothing
Smoother si.smooth Exponential smoothing with controllable pole
Take Element ba.take Take en element from a list
Time ba.time A simple timer

Conversion

Function Type Function Name Description
dB to Linear ba.db2linear Converts dB to linear values
Linear to dB ba.linear2db Converts linear values to dB
MIDI Key to Hz ba.midikey2hz Converts a MIDI key number into a frequency
Hz to MIDI Key ba.hz2midikey Converts a frequency into MIDI key number
Pole to T60 ba.pole2tau Converts a pole into a time constant (t60)
Samples to Seconds ba.samp2sec Converts samples to seconds
Seconds to Samples ba.sec2samp Converts seconds to samples
T60 to Pole ba.tau2pole Converts a time constant (t60) into a pole

Effects

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

Envelope Generators

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

Filters

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

Oscillators/Sound Generators

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

Synths

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

analyzers.lib

Faust Analyzers library. Its official prefix is an.

Amplitude Tracking

amp_follower

Classic 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.

Usage

_ : amp_follower(rel) : _

Where:

  • rel: release time = amplitude-envelope time-constant (sec) going down

Reference

  • Musical Engineer's Handbook, Bernie Hutchins, Ithaca NY, 1975 Electronotes Newsletter, Bernie Hutchins

amp_follower_ud

Envelope follower with different up and down time-constants (also called a "peak detector").

Usage

   _ : amp_follower_ud(att,rel) : _

Where:

  • att: attack time = amplitude-envelope time constant (sec) going up
  • rel: release time = amplitude-envelope time constant (sec) going down

Note

We 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.

Reference


Spectrum-Analyzers

Spectrum-analyzers split the input signal into a bank of parallel signals, one for each spectral band. They are related to the Mth-Octave Filter-Banks in filters.lib. The documentation of this library contains more details about the implementation. The parameters are:

  • M: number of band-slices per octave (>1)
  • N: total number of bands (>2)
  • ftop = upper bandlimit of the Mth-octave bands (<SR/2)

In addition to the Mth-octave output signals, there is a highpass signal containing frequencies from ftop to SR/2, and a "dc band" lowpass signal containing frequencies from 0 (dc) up to the start of the Mth-octave bands. Thus, the N output signals are

highpass(ftop), MthOctaveBands(M,N-2,ftop), dcBand(ftop*2^(-M*(N-1)))

A Spectrum-Analyzer is defined here as any band-split whose bands span the relevant spectrum, but whose band-signals do not necessarily sum to the original signal, either exactly or to within an allpass filtering. Spectrum analyzer outputs are normally at least nearly "power complementary", i.e., the power spectra of the individual bands sum to the original power spectrum (to within some negligible tolerance).

Increasing Channel Isolation

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.

References

  • "Tree-structured complementary filter banks using all-pass sections", Regalia et al., IEEE Trans. Circuits & Systems, CAS-34:1470-1484, Dec. 1987
  • "Multirate Systems and Filter Banks", P. Vaidyanathan, Prentice-Hall, 1993
  • Elementary filter theory: https://ccrma.stanford.edu/~jos/filters/

mth_octave_analyzer

Octave analyzer. mth_octave_analyzer[N] are standard Faust functions.

Usage

_ : 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 two
  • M: number of band-slices per octave
  • ftop: highest band-split crossover frequency (e.g., 20 kHz)
  • N: total number of bands (including dc and Nyquist)

Mth-Octave Spectral Level

Spectral Level: Display (in bar graphs) the average signal level in each spectral band.

mth_octave_spectral_level6e

Spectral level display.

Usage:

_ : mth_octave_spectral_level6e(M,ftop,NBands,tau,dB_offset) : _;

Where:

  • M: bands per octave
  • ftop: lower edge frequency of top band
  • NBands: 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);

Usage

See mth_octave_spectral_level_demo in demos.lib.


Arbritary-Crossover Filter-Banks and Spectrum Analyzers

These are similar to the Mth-octave analyzers above, except that the band-split frequencies are passed explicitly as arguments.

analyzer

Analyzer.

Usage

_ : 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)) : _,_,_

Fast Fourier Transform (fft) and its Inverse (ifft)

Sliding FFTs that compute a rectangularly windowed FFT each sample

fft

Fast Fourier Transform (FFT)

Usage

si.cbus(N) : fft(N) : si.cbus(N);

Where:

  • si.cbus(N) is a bus of N complex signals, each specified by real and imaginary parts: (r0,i0), (r1,i1), (r2,i2), ...
  • N is the FFT size (must be a power of 2: 2,4,8,16,...)
  • fft(N) performs a length N FFT for complex signals (radix 2)
  • The output is a bank of N complex signals containing the complex spectrum over time: (R0, I0), (R1,I1), ...
  • The dc component is (R0,I0), where I0=0 for real input signals.

FFTs of Real Signals:

  • To perform a sliding FFT over a real input signal, you can say

    process = signal : an.rtocv(N) : an.fft(N);

    where an.rtocv converts a real (scalar) signal to a complex vector signal having a zero imaginary part.

  • See an.rfft_analyzer_c (in analyzers.lib) and related functions for more detailed usage examples.

  • Use an.rfft_spectral_level(N,tau,dB_offset) to display the power spectrum of a real signal.

  • See dm.fft_spectral_level_demo(N) in demos.lib for an example GUI driving an.rfft_spectral_level().

Reference


ifft

Inverse Fast Fourier Transform (IFFT)

Usage

si.cbus(N) : ifft(N) : si.cbus(N);

Where:

  • N is the IFFT size (power of 2)
  • Input is a complex spectrum represented as interleaved real and imaginary parts: (R0, I0), (R1,I1), (R2,I2), ...
  • Output is a bank of N complex signals giving the complex signal in the time domain: (r0, i0), (r1,i1), (r2,i2), ...

amp_follower_ar

Envelope follower with independent attack and release times. The release can be shorter than the attack (unlike in amp_follower_ud above).

Usage

_ : amp_follower_ar(att,rel) : _;
  • Author Jonatan Liljedahl, revised by RM

basics.lib

A library of basic elements. Its official prefix is ba.

Conversion Tools

samp2sec

Converts a number of samples to a duration in seconds. samp2sec is a standard Faust function.

Usage

samp2sec(n) : _

Where:

  • n: number of samples

sec2samp

Converts a duration in seconds to a number of samples. samp2sec is a standard Faust function.

Usage

sec2samp(d) : _

Where:

  • d: duration in seconds

db2linear

Converts a loudness in dB to a linear gain (0-1). db2linear is a standard Faust function.

Usage

db2linear(l) : _

Where:

  • l: loudness in dB

linear2db

Converts a linear gain (0-1) to a loudness in dB. linear2db is a standard Faust function.

Usage

linear2db(g) : _

Where:

  • g: a linear gain

lin2LogGain

Converts a linear gain (0-1) to a log gain (0-1).

Usage

_ : lin2LogGain : _

log2LinGain

Converts a log gain (0-1) to a linear gain (0-1).

Usage

_ : log2LinGain : _

tau2pole

Returns 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.

Usage

_ : smooth(tau2pole(tau)) : _

Where:

  • tau: time-constant in seconds

pole2tau

Returns the time-constant, in seconds, corresponding to the given real, positive pole in (0,1). pole2tau is a standard Faust function.

Usage

pole2tau(pole) : _

Where:

  • pole: the pole

midikey2hz

Converts a MIDI key number to a frequency in Hz (MIDI key 69 = A440). midikey2hz is a standard Faust function.

Usage

midikey2hz(mk) : _

Where:

  • mk: the MIDI key number

hz2midikey

Converts a frequency in Hz to a MIDI key number (MIDI key 69 = A440). hz2midikey is a standard Faust function.

Usage

hz2midikey(f) : _

Where:

  • f: frequency in Hz

pianokey2hz

Converts a piano key number to a frequency in Hz (piano key 49 = A440).

Usage

pianokey2hz(pk) : _

Where:

  • pk: the piano key number

hz2pianokey

Converts a frequency in Hz to a piano key number (piano key 49 = A440).

Usage

hz2pianokey(f) : _

Where:

  • f: frequency in Hz

Counters and Time/Tempo Tools

countdown

Starts 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.

Usage

countdown(n,trig) : _

Where:

  • count: the starting point of the countdown
  • trig: the trigger signal (1: start at n; 0: decrease until 0)

countup

Starts 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.

Usage

countup(n,trig) : _

Where:

  • count: the maximum count value
  • trig: the trigger signal (1: start at 0; 0: increase until n)

sweep

Counts from 0 to period samples repeatedly, while run is 1. Outsputs zero while run is 0.

Usage

sweep(period,run) : _

time

A simple timer that counts every samples from the beginning of the process. time is a standard Faust function.

Usage

time : _

tempo

Converts a tempo in BPM into a number of samples.

Usage

tempo(t) : _

Where:

  • t: tempo in BPM

period

Basic sawtooth wave of period p.

Usage

period(p) : _

Where:

  • p: period as a number of samples

pulse

Pulses (10000) generated at period p.

Usage

pulse(p) : _

Where:

  • p: period as a number of samples

pulsen

Pulses (11110000) of length n generated at period p.

Usage

pulsen(n,p) : _

Where:

  • n: the length of the pulse as a number of samples
  • p: period as a number of samples

cycle

Split nonzero input values into n cycles.

Usage

_ : cycle(n) <:

Where:

  • n: the number of cycles/output signals

beat

Pulses at tempo t. beat is a standard Faust function.

Usage

beat(t) : _

Where:

  • t: tempo in BPM

pulse_countup

Starts counting up pulses. While trig is 1 the output is counting up, while trig is 0 the counter is reset to 0.

Usage

_ : pulse_countup(trig) : _

Where:

  • trig: the trigger signal (1: start at next pulse; 0: reset to 0)

pulse_countdown

Starts counting down pulses. While trig is 1 the output is counting down, while trig is 0 the counter is reset to 0.

Usage

_ : pulse_countdown(trig) : _

Where:

  • trig: the trigger signal (1: start at next pulse; 0: reset to 0)

pulse_countup_loop

Starts 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.

Usage

_ : 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)

resetCtr

Function that lets through the mth impulse out of each consecutive group of n impulses.

Usage

_ : resetCtr(n,m) : _

Where:

  • n: the total number of impulses being split
  • m: index of impulse to allow to be output

pulse_countdown_loop

Starts 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.

Usage

_ : 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)

Array Processing/Pattern Matching

count

Count the number of elements of list l. count is a standard Faust function.

Usage

count(l)
count ((10,20,30,40)) -> 4

Where:

  • l: list of elements

take

Take an element from a list. take is a standard Faust function.

Usage

take(e,l)
take(3,(10,20,30,40)) -> 30

Where:

  • p: position (starting at 1)
  • l: list of elements

subseq

Extract a part of a list.

Usage

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: list
  • p: start point (0: begin of list)
  • n: number of elements

Note:

Faust doesn't have proper lists. Lists are simulated with parallel compositions and there is no empty list


Selectors (Conditions)

if

if-then-else implemented with a select2.

Usage

  • if(c, t, e) : _

Where:

  • c: condition
  • t: signal selected while c is true
  • e: signal selected while c is false

selector

Selects the ith input among n at compile time.

Usage

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)

There is also cselector for selecting among complex input signals of the form (real,imag).


selectn

Selects the ith input among N at run time.

Usage

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)

Example test program

N=64;
process = par(n,N, (par(i,N,i) : selectn(N,n)));

select2stereo

Select between 2 stereo signals.

Usage

_,_,_,_ : select2stereo(bpc) : _,_,_,_

Where:

  • bpc: the selector switch (0/1)

Other

latch

Latch input on positive-going transition of "clock" ("sample-and-hold").

Usage

_ : latch(clocksig) : _

Where:

  • clocksig: hold trigger (0 for hold, 1 for bypass)

sAndH

Sample And Hold. sAndH is a standard Faust function.

Usage

_ : sAndH(t) : _

Where:

  • t: hold trigger (0 for hold, 1 for bypass)

downSample

Down sample a signal. WARNING: this function doesn't change the rate of a signal, it just holds samples... downSample is a standard Faust function.

Usage

_ : downSample(freq) : _

Where:

  • freq: new rate in Hz

peakhold

Outputs current max value above zero.

Usage

_ : peakhold(mode) : _;

Where:

mode means: 0 - Pass through. A single sample 0 trigger will work as a reset. 1 - Track and hold max value.


peakholder

Tracks abs peak and holds peak for 'holdtime' samples.

Usage

_ : peakholder(holdtime) : _;

impulsify

Turns the signal from a button into an impulse (1,0,0,... when button turns on). impulsify is a standard Faust function.

Usage

button("gate") : impulsify ;

automat

Record and replay to the values the input signal in a loop.

Usage

hslider(...) : automat(bps, size, init) : _

bpf

bpf 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 function
  • end(x,y) to end a break-point function
  • point(x,y) to add intermediate points to a break-point function

A 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.


listInterp

Linearly interpolates between the elements of a list.

Usage

foo = listInterp((800,400,350,450,325),index);
i = 1.69; // range is 0-4
process = foo(i);

Where:

  • index: the index (float) to interpolate between the different values. The range of index depends on the size of the list.

bypass1

Takes a mono input signal, route it to e and bypass it if bpc = 1. bypass1 is a standard Faust function.

Usage

_ : bypass1(bpc,e) : _

Where:

  • bpc: bypass switch (0/1)
  • e: a mono effect

bypass2

Takes a stereo input signal, route it to e and bypass it if bpc = 1. bypass2 is a standard Faust function.

Usage

_,_ : bypass2(bpc,e) : _,_

Where:

  • bpc: bypass switch (0/1)
  • e: a stereo effect

bypass1to2

Bypass switch for effect e having mono input signal and stereo output. Effect e is bypassed if bpc = 1. bypass1to2 is a standard Faust function.

Usage

_ : bypass1(bpc,e) : _,_

Where:

  • bpc: bypass switch (0/1)
  • e: a mono-to-stereo effect

toggle

Triggered by the change of 0 to 1, it toggles the output value between 0 and 1.

Usage

_ : toggle : _

Examples

button("toggle") : toggle : vbargraph("output", 0, 1)
(an.amp_follower(0.1) > 0.01) : toggle : vbargraph("output", 0, 1) // takes audio input

on_and_off

The first channel set the output to 1, the second channel to 0.

Usage

_ , _ : on_and_off : _

Example

button("on"), button("off") : on_and_off : vbargraph("output", 0, 1)

selectoutn

Route input to the output among N at run time.

Usage

_ : 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)

Example

process = 1 : selectoutn(3, sel) : par(i,3,bar) ;
sel = hslider("volume",0,0,2,1) : int;
bar = vbargraph("v.bargraph", 0, 1);

compressors.lib

A library of compressor effects. Its official prefix is co.

Functions Reference

compressor_mono

Mono dynamic range compressors. compressor_mono is a standard Faust function

Usage

_ : 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 up
  • rel: release time = time constant (sec) coming out of compression

References


compressor_stereo

Stereo dynamic range compressors.

Usage

_,_ : 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 up
  • rel: release time = time constant (sec) coming out of compression

References


limiter_1176_R4_mono

A 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.

Usage

 _ : limiter_1176_R4_mono : _;

Reference:

http://en.wikipedia.org/wiki/1176_Peak_Limiter


limiter_1176_R4_stereo

A 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)

Usage

 _,_ : limiter_1176_R4_stereo : _,_;

Reference:

http://en.wikipedia.org/wiki/1176_Peak_Limiter


delays.lib

This library contains a collection of delay functions. Its official prefix is de.

Basic Delay Functions

delay

Simple d samples delay where n is the maximum delay length as a number of samples. Unlike the @ delay operator, here the delay signal d is explicitely bounded to the interval [0..n]. The consequence is that delay will compile even if the interval of d can't be computed by the compiler. delay is a standard Faust function.

Usage

_ : delay(n,d) : _

Where:

  • n: the max delay length (in samples)
  • d: the delay length as a number of samples (integer)

fdelay

Simple d samples fractional delay based on 2 interpolated delay lines where n is the maximum delay length as a number of samples.

sdelay

s(mooth)delay: a mono delay that doesn't click and doesn't transpose when the delay time is changed.

Usage

_ : sdelay(N,it,dt) : _

Where :

  • N: maximal delay in samples
  • it: interpolation time (in samples) for example 1024
  • dt: delay time (in samples)

Lagrange Interpolation

fdelaylti and fdelayltv

Fractional delay line using Lagrange interpolation.

Usage

_ : 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.

References

  • https://ccrma.stanford.edu/~jos/pasp/Lagrange_Interpolation.html
  • (fixed-delay case)(https://ccrma.stanford.edu/~jos/Interpolation/Efficient_Time_Invariant_Lagrange_Interpolation.html)
  • (variable-delay case)(https://ccrma.stanford.edu/~jos/Interpolation/Time_Varying_Lagrange_Interpolation.html)
  • Timo I. Laakso et al., "Splitting the Unit Delay - Tools for Fractional Delay Filter Design", IEEE Signal Processing Magazine, vol. 13, no. 1, pp. 30-60, Jan 1996.
  • Philippe Depalle and Stephan Tassart, "Fractional Delay Lines using Lagrange Interpolators", ICMC Proceedings, pp. 341-343, 1996.

fdelay[n]

For convenience, fdelay1, fdelay2, fdelay3, fdelay4, fdelay5 are also available where n is the order of the interpolation.


Thiran Allpass Interpolation

Thiran Allpass Interpolation

Reference

https://ccrma.stanford.edu/~jos/pasp/Thiran_Allpass_Interpolators.html

fdelay[n]a

Delay lines interpolated using Thiran allpass interpolation.

Usage

_ : 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.

Note

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]


demos.lib

This library contains a set of demo functions based on examples located in the /examples folder. Its official prefix is dm.

Analyzers

mth_octave_spectral_level_demo

Demonstrate mth_octave_spectral_level in a standalone GUI.

Usage

_ : mth_octave_spectral_level_demo(BandsPerOctave);
_ : spectral_level_demo : _; // 2/3 octave

Filters

parametric_eq_demo

A parametric equalizer application.

Usage:

_ : parametric_eq_demo : _ ;

spectral_tilt_demo

A spectral tilt application.

Usage

_ : spectral_tilt_demo(N) : _ ;

Where:

  • N: filter order (integer)

All other parameters interactive


mth_octave_filterbank_demo and filterbank_demo

Graphic Equalizer: Each filter-bank output signal routes through a fader.

Usage

_ : mth_octave_filterbank_demo(M) : _
_ : filterbank_demo : _

Where:

  • N: number of bands per octave

Effects

cubicnl_demo

Distortion demo application.

Usage:

_ : cubicnl_demo : _;

gate_demo

Gate demo application.

Usage

_,_ : gate_demo : _,_;

compressor_demo

Compressor demo application.

Usage

_,_ : compressor_demo : _,_;

moog_vcf_demo

Illustrate and compare all three Moog VCF implementations above.

Usage

_ : moog_vcf_demo : _;

wah4_demo

Wah pedal application.

Usage

_ : wah4_demo : _;

crybaby_demo

Crybaby effect application.

Usage

_ : crybaby_demo : _ ;

flanger_demo

Flanger effect application.

Usage

_,_ : flanger_demo : _,_;

phaser2_demo

Phaser effect demo application.

Usage

_,_ : phaser2_demo : _,_;

stereo_reverb_tester

Handy test inputs for reverberator demos below.

Usage

_ : stereo_reverb_tester : _

fdnrev0_demo

A reverb application using fdnrev0.

Usage

_,_ : 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 greater
  • bbso = Butterworth band-split order / order of lowpass/highpass bandsplit used at each crossover freq / odd positive integer

zita_rev_fdn_demo

Reverb demo application based on zita_rev_fdn.

Usage

si.bus(8) : zita_rev_fdn_demo : si.bus(8)

zita_rev1

Example 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.

Usage

_,_ : zita_rev1 : _,_

Reference

http://www.kokkinizita.net/linuxaudio/zita-rev1-doc/quickguide.html


Generators

sawtooth_demo

An application demonstrating the different sawtooth oscillators of Faust.

Usage

sawtooth_demo : _

virtual_analog_oscillator_demo

Virtual analog oscillator demo application.

Usage

virtual_analog_oscillator_demo : _

oscrs_demo

Simple application demoing filter based oscillators.

Usage

oscrs_demo : _

velvet_noise_demo

Listen to velvet_noise!

Usage

velvet_noise_demo : _

latch_demo

Illustrate latch operation

Usage

echo 'import("stdfaust.lib");' > latch_demo.dsp
echo 'process = dm.latch_demo;' >> latch_demo.dsp
faust2octave latch_demo.dsp
Octave:1> plot(faustout);

envelopes_demo

Illustrate various envelopes overlaid, including their gate * 1.1

Usage

echo 'import("stdfaust.lib");' > envelopes_demo.dsp
echo 'process = dm.envelopes_demo;' >> envelopes_demo.dsp
faust2octave envelopes_demo.dsp
Octave:1> plot(faustout);

exciter

Psychoacoustic harmonic exciter, with GUI.

Usage

_ : exciter : _

References


vocoder_demo

Use example of the vocoder function where an impulse train is used as excitation.

Usage

_ : vocoder_demo : _;

freeverb_demo

Freeverb demo application.

Usage

_,_ : freeverb_demo : _,_;

dx7.lib

Yamaha DX7 emulation library. Its official prefix is dx.

dx7_ampf

DX7 amplitude conversion function. 3 versions of this function are available:

  • dx7_amp_bpf: BPF version (same as in the CSOUND toolkit)
  • dx7_amp_func: estimated mathematical equivalent of dx7_amp_bpf
  • dx7_ampf: default (sugar for dx7_amp_func)

Usage:

dx7AmpPreset : dx7_ampf_bpf : _

Where:

  • dx7AmpPreset: DX7 amplitude value (0-99)

dx7_egraterisef

DX7 envelope generator rise conversion function. 3 versions of this function are available:

  • dx7_egraterise_bpf: BPF version (same as in the CSOUND toolkit)
  • dx7_egraterise_func: estimated mathematical equivalent of dx7_egraterise_bpf
  • dx7_egraterisef: default (sugar for dx7_egraterise_func)

Usage:

dx7envelopeRise : dx7_egraterisef : _

Where:

  • dx7envelopeRise: DX7 envelope rise value (0-99)

dx7_egraterisepercf

DX7 envelope generator percussive rise conversion function. 3 versions of this function are available:

  • dx7_egrateriseperc_bpf: BPF version (same as in the CSOUND toolkit)
  • dx7_egrateriseperc_func: estimated mathematical equivalent of dx7_egrateriseperc_bpf
  • dx7_egraterisepercf: default (sugar for dx7_egrateriseperc_func)

Usage:

dx7envelopePercRise : dx7_egraterisepercf : _

Where:

  • dx7envelopePercRise: DX7 envelope percussive rise value (0-99)

dx7_egratedecayf

DX7 envelope generator decay conversion function. 3 versions of this function are available:

  • dx7_egratedecay_bpf: BPF version (same as in the CSOUND toolkit)
  • dx7_egratedecay_func: estimated mathematical equivalent of dx7_egratedecay_bpf
  • dx7_egratedecayf: default (sugar for dx7_egratedecay_func)

Usage:

dx7envelopeDecay : dx7_egratedecayf : _

Where:

  • dx7envelopeDecay: DX7 envelope decay value (0-99)

dx7_egratedecaypercf

DX7 envelope generator percussive decay conversion function. 3 versions of this function are available:

  • dx7_egratedecayperc_bpf: BPF version (same as in the CSOUND toolkit)
  • dx7_egratedecayperc_func: estimated mathematical equivalent of dx7_egratedecayperc_bpf
  • dx7_egratedecaypercf: default (sugar for dx7_egratedecayperc_func)

Usage:

dx7envelopePercDecay : dx7_egratedecaypercf : _

Where:

  • dx7envelopePercDecay: DX7 envelope decay value (0-99)

dx7_eglv2peakf

DX7 envelope level to peak conversion function. 3 versions of this function are available:

  • dx7_eglv2peak_bpf: BPF version (same as in the CSOUND toolkit)
  • dx7_eglv2peak_func: estimated mathematical equivalent of dx7_eglv2peak_bpf
  • dx7_eglv2peakf: default (sugar for dx7_eglv2peak_func)

Usage:

dx7Level : dx7_eglv2peakf : _

Where:

  • dx7Level: DX7 level value (0-99)

dx7_velsensf

DX7 velocity sensitivity conversion function.

Usage:

dx7Velocity  : dx7_velsensf : _

Where:

  • dx7Velocity: DX7 level value (0-8)

dx7_fdbkscalef

DX7 feedback scaling conversion function.

Usage:

dx7Feedback  : dx7_fdbkscalef : _

Where:

  • dx7Feedback: DX7 feedback value

dx7_op

DX7 Operator. Implements a phase-modulable sine wave oscillator connected to a DX7 envelope generator.

Usage:

dx7_op(freq,phaseMod,outLev,R1,R2,R3,R4,L1,L2,L3,L4,keyVel,rateScale,type,gain,gate) : _

Where:

  • freq: frequency of the oscillator
  • phaseMod: phase deviation (-1 - 1)
  • outLev: preset output level (0-99)
  • R1: preset envelope rate 1 (0-99)
  • R2: preset envelope rate 2 (0-99)
  • R3: preset envelope rate 3 (0-99)
  • R4: preset envelope rate 4 (0-99)
  • L1: preset envelope level 1 (0-99)
  • L2: preset envelope level 2 (0-99)
  • L3: preset envelope level 3 (0-99)
  • L4: preset envelope level 4 (0-99)
  • keyVel: preset key velocity sensitivity (0-99)
  • rateScale: preset envelope rate scale
  • type: preset operator type
  • gain: general gain
  • gate: trigger signal

dx7_algo

DX7 algorithms. Implements the 32 DX7 algorithms (a quick Google search should give your more details on this). Each algorithm uses 6 operators

Usage:

dx7_algo(algN,egR1,egR2,egR3,egR4,egL1,egL2,egL3,egL4,outLevel,keyVelSens,ampModSens,opMode,opFreq,opDetune,opRateScale,feedback,lfoDelay,lfoDepth,lfoSpeed,freq,gain,gate) : _

Where:

  • algN: algorithm number (0-31, should be an int...)
  • egR1: preset envelope rates 1 (a list of 6 values between 0-99)
  • egR2: preset envelope rates 2 (a list of 6 values between 0-99)
  • egR3: preset envelope rates 3 (a list of 6 values between 0-99)
  • egR4: preset envelope rates 4 (a list of 6 values between 0-99)
  • egL1: preset envelope levels 1 (a list of 6 values between 0-99)
  • egL2: preset envelope levels 2 (a list of 6 values between 0-99)
  • egL3: preset envelope levels 3 (a list of 6 values between 0-99)
  • egL4: preset envelope levels 4 (a list of 6 values between 0-99)
  • outLev: preset output levels (a list of 6 values between 0-99)
  • keyVel: preset key velocity sensitivities (a list of 6 values between 0-99)
  • ampModSens: preset amplitude sensitivities (a list of 6 values between 0-99)
  • opMode: preset operator mode (a list of 6 values between 0-1)
  • opFreq: preset operator frequencies (a list of 6 values between 0-99)
  • opDetune: preset operator detuning (a list of 6 values between 0-99)
  • opRateScale: preset operator rate scale (a list of 6 values between 0-99)
  • feedback: preset operator feedback (a list of 6 values between 0-99)
  • lfoDelay: preset LFO delay (a list of 6 values between 0-99)
  • lfoDepth: preset LFO depth (a list of 6 values between 0-99)
  • lfoSpeed: preset LFO speed (a list of 6 values between 0-99)
  • freq: fundamental frequency
  • gain: general gain
  • gate: trigger signal

dx7_ui

Generic DX7 function where all parameters are controllable using UI elements. The master-with-mute branch must be used for this function to work... This function is MIDI-compatible.

Usage

dx7_ui : _

envelopes.lib

This library contains a collection of envelope generators. Its official prefix is en.

Functions Reference

smoothEnvelope

An envelope with an exponential attack and release. smoothEnvelope is a standard Faust function.

Usage

smoothEnvelope(ar,t) : _
  • ar: attack and release duration (s)
  • t: trigger signal (0-1)

ar

AR (Attack, Release) envelope generator (useful to create percussion envelopes). ar is a standard Faust function.

Usage

ar(a,r,t) : _

Where:

  • a: attack (sec)
  • r: release (sec)
  • t: trigger signal (0 or 1)

arfe

ARFE (Attack and Release-to-Final-value Exponentially) envelope generator. Approximately equal to smoothEnvelope(Attack/6.91) when Attack == Release. arfe is a standard Faust function.

Usage

arfe(a,r,f,g) : _

Where:

  • a, r: attack (sec), release (sec)
  • f: final value to approach upon release (such as 0)
  • g: gate signal ( >0 for attack, release begins when g returns to 0)

are

ARE (Attack, Release) envelope generator with Exponential segments. Approximately equal to smoothEnvelope(Attack/6.91) when Attack == Release. are is a standard Faust function.

Usage

are(a,r,g) : _

Where:

  • a, r: attack (sec), release (sec)
  • g: gate signal ( >0 for attack, release begins when g returns to 0)

asr

ASR (Attack, Sustain, Release) envelope generator. asr is a standard Faust function.

Usage

asr(a,s,r,g) : _

Where:

  • a, s, r: attack (sec), sustain (percentage of g), release (sec)
  • g: trigger signal ( >0 for attack, then release is when g back to 0)

adsr

ADSR (Attack, Decay, Sustain, Release) envelope generator. adsr is a standard Faust function.

Usage

adsr(a,d,s,r,g) : _

Where:

  • a, d, s, r: attack (sec), decay (sec), sustain level (percentage of max), release (sec)
  • g: gate signal ( >0 for attack, then release is when g back to 0)

dx7envelope

DX7 operator envelope generator with 4 independent rates and levels. It is essentially a 4 points BPF.

Usage

dx7_envelope(R1,R2,R3,R4,L1,L2,L3,L4,t) : _

Where:

  • RN: rates in seconds
  • LN: levels (0-1)
  • t: trigger signal

adsre

ADSRE (Attack, Decay, Sustain, Release) envelope generator with Exponential segments. adsre is a standard Faust function.

Usage

adsre(a,d,s,r,g) : _

Where:

  • a, d, s, r: attack (sec), decay (sec), sustain level (percentage of max), release (sec)
  • g: gate signal ( >0 for attack, then release is when g back to 0)

filters.lib

Faust Filters library; Its official prefix is fi.

The Filters library is organized into 18 sections:

  • Basic Filters
  • Comb Filters
  • Direct-Form Digital Filter Sections
  • Direct-Form Second-Order Biquad Sections
  • Ladder/Lattice Digital Filters
  • Useful Special Cases
  • Ladder/Lattice Allpass Filters
  • Digital Filter Sections Specified as Analog Filter Sections
  • Simple Resonator Filters
  • Butterworth Lowpass/Highpass Filters
  • Special Filter-Bank Delay-Equalizing Allpass Filters
  • Elliptic (Cauer) Lowpass Filters
  • Elliptic Highpass Filters
  • Butterworth Bandpass/Bandstop Filters
  • Elliptic Bandpass Filters
  • Parametric Equalizers (Shelf, Peaking)
  • Mth-Octave Filter-Banks
  • Arbritary-Crossover Filter-Banks and Spectrum Analyzers

For more information, see ../documentation/library.pdf

Basic Filters

zero

One zero filter. Difference equation: y(n) = x(n) - z * x(n-1).

Usage

_ : zero(z) : _

Where:

  • z: location of zero along real axis in z-plane

Reference

https://ccrma.stanford.edu/~jos/filters/One_Zero.html


pole

One pole filter. Could also be called a "leaky integrator". Difference equation: y(n) = x(n) + p * y(n-1).

Usage

_ : pole(z) : _

Where:

  • p: pole location = feedback coefficient

Reference

https://ccrma.stanford.edu/~jos/filters/One_Pole.html


integrator

Same as pole(1) [implemented separately for block-diagram clarity].


dcblockerat

DC 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).

Usage

_ : dcblockerat(fb) : _

Where:

  • fb: "break frequency" in Hz, i.e., -3 dB gain frequency.

Reference

https://ccrma.stanford.edu/~jos/pasp/Bilinear_Transformation.html


dcblocker

DC 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.

Usage

_ : dcblocker : _

Comb Filters

ff_comb

Feed-Forward Comb Filter. Note that ff_comb requires integer delays (uses delay internally). ff_comb is a standard Faust function.

Usage

_ : ff_comb(maxdel,intdel,b0,bM) : _

Where:

  • maxdel: maximum delay (a power of 2)
  • intdel: current (integer) comb-filter delay between 0 and maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • b0: gain applied to delay-line input
  • bM: gain applied to delay-line output and then summed with input

Reference

https://ccrma.stanford.edu/~jos/pasp/Feedforward_Comb_Filters.html


ff_fcomb

Feed-Forward Comb Filter. Note that ff_fcomb takes floating-point delays (uses fdelay internally). ff_fcomb is a standard Faust function.

Usage

_ : ff_fcomb(maxdel,del,b0,bM) : _

Where:

  • maxdel: maximum delay (a power of 2)
  • intdel: current (integer) comb-filter delay between 0 and maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • b0: gain applied to delay-line input
  • bM: gain applied to delay-line output and then summed with input

Reference

https://ccrma.stanford.edu/~jos/pasp/Feedforward_Comb_Filters.html


ffcombfilter

Typical special case of ff_comb() where: b0 = 1.


fb_comb

Feed-Back Comb Filter (integer delay).

Usage

_ : fb_comb(maxdel,intdel,b0,aN) : _

Where:

  • maxdel: maximum delay (a power of 2)
  • intdel: current (integer) comb-filter delay between 0 and maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • b0: gain applied to delay-line input and forwarded to output
  • aN: minus the gain applied to delay-line output before summing with the input and feeding to the delay line

Reference

https://ccrma.stanford.edu/~jos/pasp/Feedback_Comb_Filters.html


fb_fcomb

Feed-Back Comb Filter (floating point delay).

Usage

_ : fb_fcomb(maxdel,del,b0,aN) : _

Where:

  • maxdel: maximum delay (a power of 2)
  • intdel: current (integer) comb-filter delay between 0 and maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • b0: gain applied to delay-line input and forwarded to output
  • aN: minus the gain applied to delay-line output before summing with the input and feeding to the delay line

Reference

https://ccrma.stanford.edu/~jos/pasp/Feedback_Comb_Filters.html


rev1

Special case of fb_comb (rev1(maxdel,N,g)). The "rev1 section" dates back to the 1960s in computer-music reverberation. See the jcrev and brassrev in reverbs.lib for usage examples.


fbcombfilter and ffbcombfilter

Other special cases of Feed-Back Comb Filter.

Usage

_ : 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 maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • g: feedback gain

Reference

https://ccrma.stanford.edu/~jos/pasp/Feedback_Comb_Filters.html


allpass_comb

Schroeder 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.

Usage

_ : allpass_comb (maxdel,intdel,aN) : _

Where:

  • maxdel: maximum delay (a power of 2)
  • intdel: current (integer) comb-filter delay between 0 and maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • aN: minus the feedback gain

References


allpass_fcomb

Schroeder 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.

Usage

_ : 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 maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • aN: minus the feedback gain

References


rev2

Special case of allpass_comb (rev2(maxlen,len,g)). The "rev2 section" dates back to the 1960s in computer-music reverberation. See the jcrev and brassrev in reverbs.lib for usage examples.


allpass_fcomb5 and allpass_fcomb1a

Same 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);`).

Direct-Form Digital Filter Sections

iir

Nth-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.

Usage

  _ : iir(bcoeffs,acoeffs) : _

Where:

  • order: filter order (int) = max(#poles,#zeros)
  • bcoeffs: (b0,b1,...,b_order) = TF numerator coefficients
  • acoeffs: (a1,...,a_order) = TF denominator coeffs (a0=1)

Reference

https://ccrma.stanford.edu/~jos/filters/Four_Direct_Forms.html


fir

FIR filter (convolution of FIR filter coefficients with a signal)

Usage

_ : fir(bv) : _

fir is standard Faust function.

Where:

  • bv = b0,b1,...,bn is a parallel bank of coefficient signals.

Note

bv is processed using pattern-matching at compile time, so it must have this normal form (parallel signals).

Example

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 convN

Convolution of input signal with given coefficients.

Usage

_ : conv((k1,k2,k3,...,kN)) : _; // Argument = one signal bank
_ : convN(N,(k1,k2,k3,...)) : _; // Useful when N < count((k1,...))

tf1, tf2 and tf3

tfN = N'th-order direct-form digital filter.

Usage

_ : tf1(b0,b1,a1) : _
_ : tf2(b0,b1,b2,a1,a2) : _
_ : tf3(b0,b1,b2,b3,a1,a2,a3) : _

Where:

  • a: the poles
  • b: the zeros

Reference

https://ccrma.stanford.edu/~jos/fp/Direct_Form_I.html


notchw

Simple notch filter based on a biquad (tf2). notchw is a standard Faust function.

Usage:

_ : notchw(width,freq) : _

Where:

  • width: "notch width" in Hz (approximate)
  • freq: "notch frequency" in Hz

Reference

https://ccrma.stanford.edu/~jos/pasp/Phasing_2nd_Order_Allpass_Filters.html


Direct-Form Second-Order Biquad Sections

Direct-Form Second-Order Biquad Sections

Reference

https://ccrma.stanford.edu/~jos/filters/Four_Direct_Forms.html

tf21, tf22, tf22t and tf21t

tfN = N'th-order direct-form digital filter where:

  • tf21 is tf2, direct-form 1
  • tf22 is tf2, direct-form 2
  • tf22t is tf2, direct-form 2 transposed
  • tf21t is tf2, direct-form 1 transposed

Usage

_ : 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 poles
  • b: the zeros

Reference

https://ccrma.stanford.edu/~jos/fp/Direct_Form_I.html


Ladder/Lattice Digital Filters

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.

av2sv

Compute reflection coefficients sv from transfer-function denominator av.

Usage

sv = av2sv(av)

Where:

  • av: parallel signal bank a1,...,aN
  • sv: parallel signal bank s1,...,sN

where ro = ith reflection coefficient, and ai = coefficient of z^(-i) in the filter transfer-function denominator A(z).

Reference

https://ccrma.stanford.edu/~jos/filters/Step_Down_Procedure.html (where reflection coefficients are denoted by k rather than s).


bvav2nuv

Compute lattice tap coefficients from transfer-function coefficients.

Usage

nuv = bvav2nuv(bv,av)

Where:

  • av: parallel signal bank a1,...,aN
  • bv: parallel signal bank b0,b1,...,aN
  • nuv: parallel signal bank nu1,...,nuN

where 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_lat2

Two-multiply latice IIR filter of arbitrary order.

Usage

_ : iir_lat2(bv,av) : _

Where:

  • bv: zeros as a bank of parallel signals
  • av: poles as a bank of parallel signals

allpassnt

Two-multiply lattice allpass (nested order-1 direct-form-ii allpasses).

Usage

_ : allpassnt(n,sv) : _

Where:

  • n: the order of the filter
  • sv: the reflection coefficients (-1 1)

iir_kl

Kelly-Lochbaum ladder IIR filter of arbitrary order.

Usage

_ : iir_kl(bv,av) : _

Where:

  • bv: zeros as a bank of parallel signals
  • av: poles as a bank of parallel signals

allpassnklt

Kelly-Lochbaum ladder allpass.

Usage:

_ : allpassklt(n,sv) : _

Where:

  • n: the order of the filter
  • sv: the reflection coefficients (-1 1)

iir_lat1

One-multiply latice IIR filter of arbitrary order.

Usage

_ : iir_lat1(bv,av) : _

Where:

  • bv: zeros as a bank of parallel signals
  • av: poles as a bank of parallel signals

allpassn1mt

One-multiply lattice allpass with tap lines.

Usage

_ : allpassn1mt(n,sv) : _

Where:

  • n: the order of the filter
  • sv: the reflection coefficients (-1 1)

iir_nl

Normalized ladder filter of arbitrary order.

Usage

_ : iir_nl(bv,av) : _

Where:

  • bv: zeros as a bank of parallel signals
  • av: poles as a bank of parallel signals

References


allpassnnlt

Normalized ladder allpass filter of arbitrary order.

Usage:

_ : allpassnnlt(n,sv) : _

Where:

  • n: the order of the filter
  • sv: the reflection coefficients (-1,1)

References


Useful Special Cases

tf2np

Biquad based on a stable second-order Normalized Ladder Filter (more robust to modulation than tf2 and protected against instability).

Usage

_ : tf2np(b0,b1,b2,a1,a2) : _

Where:

  • a: the poles
  • b: the zeros

wgr

Second-order transformer-normalized digital waveguide resonator.

Usage

_ : wgr(f,r) : _

Where:

  • f: resonance frequency (Hz)
  • r: loss factor for exponential decay (set to 1 to make a numerically stable oscillator)

References


nlf2

Second order normalized digital waveguide resonator.

Usage

_ : nlf2(f,r) : _

Where:

  • f: resonance frequency (Hz)
  • r: loss factor for exponential decay (set to 1 to make a sinusoidal oscillator)

Reference

https://ccrma.stanford.edu/~jos/pasp/Power_Normalized_Waveguide_Filters.html


apnl

Passive Nonlinear Allpass based on Pierce switching springs idea. Switch between allpass coefficient a1 and a2 at signal zero crossings.

Usage

_ : apnl(a1,a2) : _

Where:

  • a1 and a2: allpass coefficients

Reference

  • "A Passive Nonlinear Digital Filter Design ..." by John R. Pierce and Scott A. Van Duyne, JASA, vol. 101, no. 2, pp. 1120-1126, 1997

Ladder/Lattice Allpass Filters

An 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*.

References

allpassn

Two-multiply lattice - each section is two multiply-adds.

Usage:

_ : allpassn(n,sv) : _

Where:

  • n: the order of the filter
  • sv: the reflection coefficients (-1 1)

References

  • J. O. Smith and R. Michon, "Nonlinear Allpass Ladder Filters in FAUST", in Proceedings of the 14th International Conference on Digital Audio Effects (DAFx-11), Paris, France, September 19-23, 2011.

allpassnn

Normalized form - four multiplies and two adds per section, but coefficients can be time varying and nonlinear without "parametric amplification" (modulation of signal energy).

Usage:

_ : allpassnn(n,tv) : _

Where:

  • n: the order of the filter
  • tv: the reflection coefficients (-PI PI)

allpasskl

Kelly-Lochbaum form - four multiplies and two adds per section, but all signals have an immediate physical interpretation as traveling pressure waves, etc.

Usage:

_ : allpassnkl(n,sv) : _

Where:

  • n: the order of the filter
  • sv: the reflection coefficients (-1 1)

allpass1m

One-multiply form - one multiply and three adds per section. Normally the most efficient in special-purpose hardware.

Usage:

_ : allpassn1m(n,sv) : _

Where:

  • n: the order of the filter
  • sv: the reflection coefficients (-1 1)

Digital Filter Sections Specified as Analog Filter Sections

tf2s and tf2snp

Second-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.

Usage

_ : 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).

Example

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);

Method

Bilinear transform scaled for exact mapping of w1.

Reference

https://ccrma.stanford.edu/~jos/pasp/Bilinear_Transformation.html


tf3slf

Analogous 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.

Usage

_ : tf3slf(b3,b2,b1,b0,a3,a2,a1,a0) : _

tf1s

First-order direct-form digital filter, specified by ANALOG transfer-function polynomials B(s)/A(s), and a frequency-scaling parameter.

Usage

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).

Example

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

Method

Bilinear transform scaled for exact mapping of w1.

Reference

https://ccrma.stanford.edu/~jos/pasp/Bilinear_Transformation.html


tf2sb

Bandpass 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.

Usage

_ : tf2sb(b2,b1,b0,a1,a0,w1,wc) : _

tf1sb

First-to-second-order lowpass-to-bandpass section mapping, analogous to tf2sb above.

Usage

_ : tf1sb(b1,b0,a0,w1,wc) : _

Simple Resonator Filters

resonlp

Simple resonant lowpass filter based on tf2s (virtual analog). resonlp is a standard Faust function.

Usage

_ : resonlp(fc,Q,gain) : _
_ : resonhp(fc,Q,gain) : _
_ : resonbp(fc,Q,gain) : _

Where:

  • fc: center frequency (Hz)
  • Q: q
  • gain: gain (0-1)

resonhp

Simple resonant highpass filters based on tf2s (virtual analog). resonhp is a standard Faust function.

Usage

_ : resonlp(fc,Q,gain) : _
_ : resonhp(fc,Q,gain) : _
_ : resonbp(fc,Q,gain) : _

Where:

  • fc: center frequency (Hz)
  • Q: q
  • gain: gain (0-1)

resonbp

Simple resonant bandpass filters based on tf2s (virtual analog). resonbp is a standard Faust function.

Usage

_ : resonlp(fc,Q,gain) : _
_ : resonhp(fc,Q,gain) : _
_ : resonbp(fc,Q,gain) : _

Where:

  • fc: center frequency (Hz)
  • Q: q
  • gain: gain (0-1)

Butterworth Lowpass/Highpass Filters

lowpass

Nth-order Butterworth lowpass filter. lowpass is a standard Faust function.

Usage

_ : lowpass(N,fc) : _

Where:

  • N: filter order (number of poles) [nonnegative constant integer]
  • fc: desired cut-off frequency (-3dB frequency) in Hz

References


highpass

Nth-order Butterworth highpass filters. highpass is a standard Faust function.

Usage

_ : highpass(N,fc) : _

Where:

  • N: filter order (number of poles) [nonnegative constant integer]
  • fc: desired cut-off frequency (-3dB frequency) in Hz

References


lowpass0_highpass1

Special Filter-Bank Delay-Equalizing Allpass Filters

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_highpass

Elliptic (Cauer) Lowpass Filters

Elliptic (Cauer) Lowpass Filters

References

  • <http://en.wikipedia.org/wiki/Elliptic_filter
  • functions ncauer and ellip in Octave

lowpass3e

Third-order Elliptic (Cauer) lowpass filter.

Usage

_ : lowpass3e(fc) : _

Where:

  • fc: -3dB frequency in Hz

Design

For 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

lowpass6e

Sixth-order Elliptic/Cauer lowpass filter.

Usage

_ : lowpass6e(fc) : _

Where:

  • fc: -3dB frequency in Hz

Design

For 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

Elliptic Highpass Filters

highpass3e

Third-order Elliptic (Cauer) highpass filter. Inversion of lowpass3e wrt unit circle in s plane (s <- 1/s)

Usage

_ : highpass3e(fc) : _

Where:

  • fc: -3dB frequency in Hz

highpass6e

Sixth-order Elliptic/Cauer highpass filter. Inversion of lowpass3e wrt unit circle in s plane (s <- 1/s)

Usage

_ : highpass6e(fc) : _

Where:

  • fc: -3dB frequency in Hz

Butterworth Bandpass/Bandstop Filters

bandpass

Order 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.

Usage

_ : bandpass(Nh,fl,fu) : _

Where:

  • Nh: HALF the desired bandpass order (which is therefore even)
  • fl: lower -3dB frequency in Hz
  • fu: upper -3dB frequency in Hz Thus, the passband width is fu-fl, and its center frequency is (fl+fu)/2.

Reference

http://cnx.org/content/m16913/latest/


bandstop

Order 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.

Usage

_ : bandstop(Nh,fl,fu) : _

Where:

  • Nh: HALF the desired bandstop order (which is therefore even)
  • fl: lower -3dB frequency in Hz
  • fu: upper -3dB frequency in Hz Thus, the passband (stopband) width is fu-fl, and its center frequency is (fl+fu)/2.

Reference

http://cnx.org/content/m16913/latest/


Elliptic Bandpass Filters

bandpass6e

Order 12 elliptic bandpass filter analogous to bandpass(6).


bandpass12e

Order 24 elliptic bandpass filter analogous to bandpass(6).


Parametric Equalizers (Shelf, Peaking)

Parametric Equalizers (Shelf, Peaking)

References

low_shelf

First-order "low shelf" filter (gain boost|cut between dc and some frequency) low_shelf is a standard Faust function.

Usage

_ : 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).

Shelf Shape

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):

  • L0 > 0:
  • L(lf) = L0, f between 0 and fx = 1st corner frequency;
  • L(lf) = L0 - N * (lf - lfx), f between fx and f2 = 2nd corner frequency;
  • L(lf) = 0, lf > lf2.
  • lf2 = lfx + L0/N = dB-frequency at which level gets back to 0 dB.
  • L0 < 0:
  • L(lf) = L0, f between 0 and f1 = 1st corner frequency;
  • L(lf) = - N * (lfx - lf), f between f1 and lfx = 2nd corner frequency;
  • L(lf) = 0, lf > lfx.
  • lf1 = lfx + L0/N = dB-frequency at which level goes up from L0.

See lowshelf_other_freq.


high_shelf

First-order "high shelf" filter (gain boost|cut above some frequency). high_shelf is a standard Faust function.

Usage

_ : 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_eq

Second order "peaking equalizer" section (gain boost or cut near some frequency) Also called a "parametric equalizer" section. peak_eq is a standard Faust function.

Usage

_ : peak_eq(Lfx,fx,B) : _;

Where:

  • Lfx: level (dB) at fx (boost Lfx>0 or cut Lfx<0)
  • fx: peak frequency (Hz)
  • B: bandwidth (B) of peak in Hz

peak_eq_cq

Constant-Q second order peaking equalizer section.

Usage

_ : peak_eq_cq(Lfx,fx,Q) : _;

Where:

  • Lfx: level (dB) at fx
  • fx: boost or cut frequency (Hz)
  • Q: "Quality factor" = fx/B where B = bandwidth of peak in Hz

peak_eq_rm

Regalia-Mitra second order peaking equalizer section

Usage

_ : peak_eq_rm(Lfx,fx,tanPiBT) : _;

Where:

  • Lfx: level (dB) at fx
  • fx: 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 B

Reference

P.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_tilt

Spectral 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.

Usage

_ : spectral_tilt(N,f0,bw,alpha) : _

Where:

  • N: desired integer filter order (fixed at compile time)
  • f0: lower frequency limit for desired roll-off band
  • bw: bandwidth of desired roll-off band
  • alpha: slope of roll-off desired in nepers per neper (ln mag / ln radian freq)

Examples

See spectral_tilt_demo.

Reference

J.O. Smith and H.F. Smith, "Closed Form Fractional Integration and Differentiation via Real Exponentially Spaced Pole-Zero Pairs", arXiv.org publication arXiv:1606.06154 [cs.CE], June 7, 2016, http://arxiv.org/abs/1606.06154


levelfilter

Dynamic level lowpass filter. levelfilter is a standard Faust function.

Usage

_ : levelfilter(L,freq) : _

Where:

  • L: desired level (in dB) at Nyquist limit (SR/2), e.g., -60
  • freq: corner frequency (-3dB point) usually set to fundamental freq
  • N: Number of filters in series where L = L/N

Reference

https://ccrma.stanford.edu/realsimple/faust_strings/Dynamic_Level_Lowpass_Filter.html


levelfilterN

Dynamic level lowpass filter.

Usage

_ : levelfilterN(N,freq,L) : _

Where:

  • L: desired level (in dB) at Nyquist limit (SR/2), e.g., -60
  • freq: corner frequency (-3dB point) usually set to fundamental freq
  • N: Number of filters in series where L = L/N

Reference

https://ccrma.stanford.edu/realsimple/faust_strings/Dynamic_Level_Lowpass_Filter.html


Mth-Octave Filter-Banks

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.

Increasing Channel Isolation

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.

References

  • "Tree-structured complementary filter banks using all-pass sections", Regalia et al., IEEE Trans. Circuits & Systems, CAS-34:1470-1484, Dec. 1987
  • "Multirate Systems and Filter Banks", P. Vaidyanathan, Prentice-Hall, 1993
  • Elementary filter theory: https://ccrma.stanford.edu/~jos/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.

Usage

_ : 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,_); // 3rd-order Butterworth
_ : mth_octave_filterbank5(M,ftop,N) : par(i,N,_); // 5th-order Butterworth
mth_octave_filterbank_default = mth_octave_filterbank5;

Where:

  • O: order of filter used to split each frequency band into two
  • M: number of band-slices per octave
  • ftop: highest band-split crossover frequency (e.g., 20 kHz)
  • N: total number of bands (including dc and Nyquist)

Arbritary-Crossover Filter-Banks and Spectrum Analyzers

These are similar to the Mth-octave analyzers above, except that the band-split frequencies are passed explicitly as arguments.

filterbank

Filter bank. filterbank is a standard Faust function.

Usage

_ : 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)) : _,_,_

filterbanki

Inverted-dc filter bank.

Usage

_ : 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)) : _,_,_

hoa.lib

Faust library for high order ambisonic. Its official prefix is ho.

encoder

Ambisonic encoder. Encodes a signal in the circular harmonics domain depending on an order of decomposition and an angle.

Usage

encoder(n, x, a) : _

Where:

  • n: the order
  • x: the signal
  • a: the angle

decoder

Decodes an ambisonics sound field for a circular array of loudspeakers.

Usage

_ : decoder(n, p) : _

Where:

  • n: the order
  • p: the number of speakers

Note

Number of loudspeakers must be greater or equal to 2n+1. It's preferable to use 2n+2 loudspeakers.


decoderStereo

Decodes 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 order

Optimization Functions

Functions to weight the circular harmonics signals depending to the ambisonics optimization. It can be basic for no optimization, maxRe or inPhase.

optimBasic

The basic optimization has no effect and should be used for a perfect circle of loudspeakers with one listener at the perfect center loudspeakers array.

Usage

_ : optimBasic(n) : _ 

Where:

  • n: the order

optimMaxRe

The maxRe optimization optimize energy vector. It should be used for an auditory confined in the center of the loudspeakers array.

Usage

_ : optimMaxRe(n) : _ 

Where:

  • n: the order

optimInPhase

The inPhase Optimization optimize energy vector and put all loudspeakers signals n phase. It should be used for an auditory.

Usage

``
optimInPhase(n) : _ ``

here:

n: the order


wider

Can 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.

Usage

_ : wider(n,w) : _ 

Where:

  • n: the order
  • w: the width value between 0 - 1

map

It simulate the distance of the source by applying a gain on the signal and a wider processing on the soundfield.

Usage

map(n, x, r, a)

Where:

  • n: the order
  • x: the signal
  • r: the radius
  • a: the angle in radian

rotate

Rotates the sound field.

Usage

_ : rotate(n, a) : _ 

Where:

  • n: the order
  • a: the angle in radian

maths.lib

Mathematic library for Faust. Its official prefix is ma.

Functions Reference

SR

Current sampling rate (between 1000Hz and 192000Hz). Constant during program execution.

Usage

SR : _

BS

Current block-size. Can change during the execution.

Usage

BS : _

PI

Constant PI in double precisio.n

Usage

PI : _

FTZ

Flush 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.

Usage

_ : ftz : _

See : http://docs.oracle.com/cd/E19957-01/806-3568/ncg_math.html


neg

Invert the sign (-x) of a signal.

Usage

_ : neg : _

sub(x,y)

Subtract x and y.


inv

Compute the inverse (1/x) of the input signal.

Usage

_ : inv : _

cbrt

Computes the cube root of of the input signal.

Usage

_ : cbrt : _

hypot

Computes the euclidian distance of the two input signals sqrt(xx+yy) without undue overflow or underflow.

Usage

_,_ : hypot : _

ldexp

Takes two input signals: x and n, and multiplies x by 2 to the power n.

Usage

_,_ : ldexp : _

scalb

Takes two input signals: x and n, and multiplies x by 2 to the power n.

Usage

_,_ : scalb : _

log1p

Computes log(1 + x) without undue loss of accuracy when x is nearly zero.

Usage

_ : log1p : _

logb

Return exponent of the input signal as a floating-point number.

Usage

_ : logb : _

ilogb

Return exponent of the input signal as an integer number.

Usage

_ : ilogb : _

log2

Returns the base 2 logarithm of x.

Usage

_ : log2 : _

expm1

Return exponent of the input signal minus 1 with better precision.

Usage

_ : expm1 : _

acosh

Computes the principle value of the inverse hyperbolic cosine of the input signal.

Usage

_ : acosh : _

asinh

Computes the inverse hyperbolic sine of the input signal.

Usage

_ : asinh : _

atanh

Computes the inverse hyperbolic tangent of the input signal.

Usage

_ : atanh : _

sinh

Computes the hyperbolic sine of the input signal.

Usage

_ : sinh : _

cosh

Computes the hyperbolic cosine of the input signal.

Usage

_ : cosh : _

tanh

Computes the hyperbolic tangent of the input signal.

Usage

_ : tanh : _

erf

Computes the error function of the input signal.

Usage

_ : erf : _

erfc

Computes the complementary error function of the input signal.

Usage

_ : erfc : _

gamma

Computes the gamma function of the input signal.

Usage

_ : gamma : _

lgamma

Calculates the natural logorithm of the absolute value of the gamma function of the input signal.

Usage

_ : lgamma : _

J0

Computes the Bessel function of the first kind of order 0 of the input signal.

Usage

_ : J0 : _

J1

Computes the Bessel function of the first kind of order 1 of the input signal.

Usage

_ : J1 : _

Jn

Computes the Bessel function of the first kind of order n (first input signal) of the second input signal.

Usage

_,_ : Jn : _

Y0

Computes the linearly independent Bessel function of the second kind of order 0 of the input signal.

Usage

_ : Y0 : _

Y1

Computes the linearly independent Bessel function of the second kind of order 1 of the input signal.

Usage

_ : Y0 : _

Yn

Computes the linearly independent Bessel function of the second kind of order n (first input signal) of the second input signal.

Usage

_,_ : Yn : _

fabs, fmax, fmin

Just for compatibility...

fabs = abs
fmax = max
fmin = min

np2

Gives the next power of 2 of x.

Usage

np2(n) : _

Where:

  • n: an integer

frac

Gives the fractional part of n.

Usage

frac(n) : _

Where:

  • n: a decimal number

modulo

Modulus operation.

Usage

modulo(x,N) : _

Where:

  • x: the numerator
  • N: the denominator

isnan

Return non-zero if and only if x is a NaN.

Usage

isnan(x)
_ : isnan : _

Where:

  • x: signal to analyse

chebychev

Chebychev transformation of order n.

Usage

_ : chebychev(n) : _

Where:

  • n: the order of the polynomial

Semantics

T[0](x) = 1,
T[1](x) = x,
T[n](x) = 2x*T[n-1](x) - T[n-2](x)

Reference

http://en.wikipedia.org/wiki/Chebyshev_polynomial


chebychevpoly

Linear combination of the first Chebyshev polynomials.

Usage

_ : chebychevpoly((c0,c1,...,cn)) : _

Where:

  • cn: the different Chebychevs polynomials such that: chebychevpoly((c0,c1,...,cn)) = Sum of chebychev(i)*ci

Reference

http://www.csounds.com/manual/html/chebyshevpoly.html


diffn

Negated first-order difference.

Usage

_ : diffn : _

signum

The signum function signum(x) is defined as -1 for x<0, 0 for x==0, and 1 for x>0;

Usage

_ : signum : _

misceffects.lib

This library contains a collection of audio effects. Its official prefix is ef.

Dynamic

cubicnl

Cubic nonlinearity distortion. cubicnl is a standard Faust library.

Usage:

_ : cubicnl(drive,offset) : _
_ : cubicnl_nodc(drive,offset) : _

Where:

  • drive: distortion amount, between 0 and 1
  • offset: constant added before nonlinearity to give even harmonics. Note: offset can introduce a nonzero mean - feed cubicnl output to dcblocker to remove this.

References:


gate_mono

Mono signal gate. gate_mono is a standard Faust function.

Usage

_ : 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)

References


gate_stereo

Stereo signal gates. gate_stereo is a standard Faust function.

Usage

 _,_ : 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)

References


Filtering

speakerbp

Dirt-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

Usage

speakerbp(f1,f2)
_ : speakerbp(130,5000) : _

piano_dispersion_filter

Piano dispersion allpass filter in closed form.

Usage

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 Hz

Outputs

  • MINUS the estimated delay at f0 of allpass chain in samples, provided in negative form to facilitate subtraction from delay-line length.
  • Output signal from allpass chain

stereo_width

Stereo Width effect using the Blumlein Shuffler technique. stereo_width is a standard Faust function.

Usage

_,_ : stereo_width(w) : _,_

Where:

  • w: stereo width between 0 and 1

At 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".

Reference

  • "Applications of Blumlein Shuffling to Stereo Microphone Techniques" Michael A. Gerzon, JAES vol. 42, no. 6, June 1994

Time Based

echo

A simple echo effect.

echo is a standard Faust function

Usage

_ : echo(maxDuration,duration,feedback) : _

Where:

  • maxDuration: the max echo duration in seconds
  • duration: the echo duration in seconds
  • feedback: the feedback coefficient

Pitch Shifting

transpose

A simple pitch shifter based on 2 delay lines. transpose is a standard Faust function.

Usage

_ : transpose(w, x, s) : _

Where:

  • w: the window length (samples)
  • x: crossfade duration duration (samples)
  • s: shift (semitones)

Meshes

mesh_square

Square Rectangular Digital Waveguide Mesh.

Usage

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,...)

Reference

https://ccrma.stanford.edu/~jos/pasp/Digital_Waveguide_Mesh.html

Signal Order In and Out

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);.

Example

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.


noises.lib

Faust Noise Generator Library. Its official prefix is no.

Functions Reference

noise

White noise generator (outputs random number between -1 and 1). Noise is a standard Faust function.

Usage

noise : _

multirandom

Generates multiple decorrelated random numbers in parallel.

Usage

multirandom(n) : si.bus(n)

Where:

  • n: the number of decorrelated random numbers in parallel

multinoise

Generates multiple decorrelated noises in parallel.

Usage

multinoise(n) : si.bus(n)

Where:

  • n: the number of decorrelated random numbers in parallel

noises

TODO.


pink_noise

Pink noise (1/f noise) generator (third-order approximation) pink_noise is a standard Faust function.

Usage

pink_noise : _;

Reference:

https://ccrma.stanford.edu/~jos/sasp/Example_Synthesis_1_F_Noise.html


pink_noise_vm

Multi pink noise generator.

Usage

pink_noise_vm(N) : _;

Where:

  • N: number of latched white-noise processes to sum, not to exceed sizeof(int) in C++ (typically 32).

References


lfnoise, lfnoise0 and lfnoiseN

Low-frequency noise generators (Butterworth-filtered downsampled white noise)

Usage

lfnoise0(rate) : _;   // new random number every int(SR/rate) samples or so
lfnoiseN(N,rate) : _; // same as "lfnoise0(rate) : lowpass(N,rate)" [see filters.lib]
lfnoise(rate) : _;    // same as "lfnoise0(rate) : seq(i,5,lowpass(N,rate))" (no overshoot)

Example

(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

sparse_noise_vm

sparse noise generator.

Usage

sparse_noise(f0) : _;

Where:

  • f0: average frequency of noise impulses per second

Random impulses in the amplitude range -1 to 1 are generated at an average rate of f0 impulses per second.

Reference

  • See velvet_noise

velvet_noise_vm

velvet noise generator.

Usage

velvet_noise(amp,f0) : _;

Where:

  • amp: amplitude of noise impulses (positive and negative)
  • f0: average frequency of noise impulses per second

Reference

  • Matti Karjalainen and Hanna Jarvelainen, "Reverberation Modeling Using Velvet Noise", in Proc. 30th Int. Conf. Intelligent Audio Environments (AES07), March 2007.

gnoise

approximate zero-mean, unit-variance Gaussian white noise generator

Usage

gnoise(N) : _;

Where:

  • N: number of uniform random numbers added to approximate Gaussian white noise

Reference

  • See Central Limit Theorem

oscillators.lib

This library contains a collection of sound generators. Its official prefix is os.

Wave-Table-Based Oscillators

sinwaveform

Sine waveform ready to use with a rdtable.

Usage

sinwaveform(tablesize) : _

Where:

  • tablesize: the table size

coswaveform

Cosine waveform ready to use with a rdtable.

Usage

coswaveform(tablesize) : _

Where:

  • tablesize: the table size

phasor

A simple phasor to be used with a rdtable. phasor is a standard Faust function.

Usage

phasor(tablesize,freq) : _

Where:

  • tablesize: the table size
  • freq: the frequency of the wave (Hz)

hs_phasor

Hardsyncing phasor to be used with an rdtable.

Usage

hs_phasor(ts,freq,c) :  _

Where:

  • ts: the tablesize for the related sine wavetable
  • freq: the fundamental frequency of the phasor
  • c: a clock signal, c>0 resets phase to 0

oscsin

Sine wave oscillator. oscsin is a standard Faust function.

Usage

oscsin(freq) : _

Where:

  • freq: the frequency of the wave (Hz)

hs_oscsin

Sin lookup table with hardsyncing phase.

Usage

hs_oscsin(freq,c) : _

Where:

  • freq: the fundamental frequency of the phasor
  • c: a clock signal, c>0 resets phase to 0

osccos

Cosine wave oscillator.

Usage

osccos(freq) : _

Where:

  • freq: the frequency of the wave (Hz)

oscp

A sine wave generator with controllable phase.

Usage

oscp(freq,p) : _

Where:

  • freq: the frequency of the wave (Hz)
  • p: the phase in radian

osci

Interpolated phase sine wave oscillator.

Usage

osci(freq) : _

Where:

  • freq: the frequency of the wave (Hz)

LFOs

Low-Frequency Oscillators (LFOs) have prefix lf_ (no aliasing suppression, which is not audible at LF).

lf_imptrain

Unit-amplitude low-frequency impulse train. lf_imptrain is a standard Faust function.

Usage

lf_imptrain(freq) : _

Where:

  • freq: frequency in Hz

lf_pulsetrainpos

Unit-amplitude nonnegative LF pulse train, duty cycle between 0 and 1

Usage

lf_pulsetrainpos(freq,duty) : _

Where:

  • freq: frequency in Hz
  • duty: duty cycle between 0 and 1

lf_pulsetrain

Unit-amplitude zero-mean LF pulse train, duty cycle between 0 and 1

Usage

lf_pulsetrain(freq,duty) : _

Where:

  • freq: frequency in Hz
  • duty: duty cycle between 0 and 1

lf_squarewavepos

Positive LF square wave in [0,1]

Usage

lf_squarewavepos(freq) : _

Where:

  • freq: frequency in Hz

lf_squarewave

Zero-mean unit-amplitude LF square wave. lf_squarewave is a standard Faust function.

Usage

lf_squarewave(freq) : _

Where:

  • freq: frequency in Hz

lf_trianglepos

Positive unit-amplitude LF positive triangle wave

Usage

lf_trianglepos(freq) : _

Where:

  • freq: frequency in Hz

Low Frequency Sawtooths

Sawtooth 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_rawsaw

Simple sawtooth waveform oscillator between 0 and period in samples.

Usage

lf_rawsaw(periodsamps)

Where:

  • periodsamps: number of periods per samples

lf_sawpos_phase

Simple sawtooth waveform oscillator between 0 and 1 with phase control.

Usage

lf_sawpos_phase(freq,phase)

Where:

  • freq: frequency
  • phase: phase

Bandlimited Sawtooth

//------------------sawN-------------------------------- Bandlimited Sawtooth

sawN(N,freq), sawNp, saw2dpw(freq), saw2(freq), saw3(freq), saw4(freq), saw5(freq), saw6(freq), sawtooth(freq), saw2f2(freq) saw2f4(freq)

Method 1 (saw2)

Polynomial Transition Regions (PTR) (for aliasing suppression)

Reference

Method 2 (sawN)

Differentiated Polynomial Waves (DPW) (for aliasing suppression)

Reference

"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.

Other Cases

Correction-filtered versions of saw2: saw2f2, saw2f4 The correction filter compensates "droop" near half the sampling rate. See reference for sawN.

Usage

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 order
  • freq: frequency in Hz
  • phase: phase

sawNp

TODO: MarkDown doc in comments


saw2dpw

TODO: MarkDown doc in comments


saw3

TODO: MarkDown doc in comments


sawtooth

Alias-free sawtooth wave. 2nd order interpolation (based on saw2). sawtooth is a standard Faust function.

Usage

sawtooth(freq) : _

Where:

  • freq: frequency

saw2f2

TODO: MarkDown doc in comments


saw2f4

TODO: MarkDown doc in comments


Bandlimited Pulse, Square, and Impulse Trains

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.

Usage

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 order
  • freq: frequency in Hz

pulsetrainN

TODO: MarkDown doc in comments


pulsetrain

Bandlimited pulse train oscillator. Based on pulsetrainN(2). pulsetrain is a standard Faust function.

Usage

pulsetrain(freq, duty) : _

Where:

  • freq: frequency
  • duty: duty cycle between 0 and 1

squareN

TODO: MarkDown doc in comments


square

Bandlimited square wave oscillator. Based on squareN(2). square is a standard Faust function.

Usage

square(freq) : _

Where:

  • freq: frequency

impulse

One-time impulse generated when the Faust process is started. impulse is a standard Faust function.

Usage

impulse : _

imptrainN

TODO: MarkDown doc in comments


imptrain

Bandlimited impulse train generator. Based on imptrainN(2). imptrain is a standard Faust function.

Usage

imptrain(freq) : _

Where:

  • freq: frequency

triangleN

TODO: MarkDown doc in comments


triangle

Bandlimited triangle wave oscillator. Based on triangleN(2). triangle is a standard Faust function.

Usage

triangle(freq) : _

Where:

  • freq: frequency

Filter-Based Oscillators

Filter-Based Oscillators

Usage

osc[b|r|rs|rc|s|w](f), where f = frequency in Hz.

References

oscb

Sinusoidal oscillator based on the biquad.

Usage

oscb(freq) : _

Where:

  • freq: frequency

oscrq

Sinusoidal (sine and cosine) oscillator based on 2D vector rotation, = undamped "coupled-form" resonator = lossless 2nd-order normalized ladder filter.

Usage

oscrq(freq) : _,_

Where:

  • freq: frequency

Reference


oscrs

Sinusoidal (sine) oscillator based on 2D vector rotation, = undamped "coupled-form" resonator = lossless 2nd-order normalized ladder filter.

Usage

oscrs(freq) : _

Where:

  • freq: frequency

Reference


oscrc

Sinusoidal (cosine) oscillator based on 2D vector rotation, = undamped "coupled-form" resonator = lossless 2nd-order normalized ladder filter.

Usage

oscrc(freq) : _

Where:

  • freq: frequency

Reference


osc

Default sine wave oscillator (same as oscrs). osc is a standard Faust function.

Usage

osc(freq) : _

Where:

  • freq: the frequency of the wave (Hz)

oscs

Sinusoidal oscillator based on the state variable filter = undamped "modified-coupled-form" resonator = "magic circle" algorithm used in graphics


Waveguide-Resonator-Based Oscillators

Sinusoidal oscillator based on the waveguide resonator wgr.

oscw

Sinusoidal oscillator based on the waveguide resonator wgr. Unit-amplitude cosine oscillator.

Usage

oscwc(freq) : _

Where:

  • freq: frequency

Reference


oscws

Sinusoidal oscillator based on the waveguide resonator wgr. Unit-amplitude sine oscillator

Usage

oscws(freq) : _

Where:

  • freq: frequency

Reference


oscwq

Sinusoidal oscillator based on the waveguide resonator wgr. Unit-amplitude cosine and sine (quadrature) oscillator.

Usage

oscwq(freq) : _

Where:

  • freq: frequency

Reference


oscw

Sinusoidal oscillator based on the waveguide resonator wgr. Unit-amplitude cosine oscillator (default)

Usage

oscw(freq) : _

Where:

  • freq: frequency

Reference


lf_sawpos

Simple sawtooth waveform oscillator between 0 and 1.

Usage

lf_sawpos(freq)

Where:

  • freq: frequency

lf_saw

Simple sawtooth waveform. lf_saw is a standard Faust function.

Usage

lf_saw(freq)

Where:

  • freq: frequency

lf_triangle

Positive unit-amplitude LF triangle wave lf_triangle is a standard Faust function.

Usage

lf_triangle(freq) : _

Where:

  • freq: frequency in Hz

phaflangers.lib

A library of phasor and flanger effects. Its official prefix is pf.

Functions Reference

flanger_mono

Mono flanging effect.

Usage:

_ : flanger_mono(dmax,curdel,depth,fb,invert) : _;

Where:

  • dmax: maximum delay-line length (power of 2) - 10 ms typical
  • curdel: 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 sum

Reference

https://ccrma.stanford.edu/~jos/pasp/Flanging.html


flanger_stereo

Stereo flanging effect. flanger_stereo is a standard Faust function.

Usage:

_,_ : flanger_stereo(dmax,curdel1,curdel2,depth,fb,invert) : _,_;

Where:

  • dmax: maximum delay-line length (power of 2) - 10 ms typical
  • curdel: 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 sum

Reference

https://ccrma.stanford.edu/~jos/pasp/Flanging.html


phaser2_mono

Mono phasing effect.

Phaser

_ : 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 Hz
  • frqmin: approximate minimum frequency of first spectral notch in Hz
  • fratio: ratio of adjacent notch frequencies
  • frqmax: approximate maximum frequency of first spectral notch in Hz
  • speed: 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 sum

Reference:


phaser2_stereo

Stereo phasing effect. phaser2_stereo is a standard Faust function.

Phaser

_ : 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 Hz
  • frqmin: approximate minimum frequency of first spectral notch in Hz
  • fratio: ratio of adjacent notch frequencies
  • frqmax: approximate maximum frequency of first spectral notch in Hz
  • speed: 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 sum

Reference:


physmodels.lib

Faust physical modeling library; Its official prefix is pm.

This library provides an environment to facilitate physical modeling of musical instruments. It contains dozens of functions implementing low and high level elements going from a simple waveguide to fully operational models with built-in UI, etc.

It is organized as follows:

  • Global Variables: Useful pre-defined variables for physical modeling (e.g., speed of sound, etc.).
  • Conversion Tools: Conversion functions specific to physical modeling (e.g., length to frequency, etc.).
  • Bidirectional Utilities: Functions to create bidirectional block diagrams for physical modeling.
  • Basic Elements: waveguides, specific types of filters, etc.
  • String Instruments: various types of strings (e.g., steel, nylon, etc.), bridges, guitars, etc.
  • Bowed String Instruments: parts and models specific to bowed string instruments (e.g., bows, bridges, violins, etc.).
  • Wind Instrument: parts and models specific to wind string instruments (e.g., reeds, mouthpieces, flutes, clarinets, etc.).
  • Exciters: pluck generators, "blowers", etc.
  • Modal Percussions: percussion instruments based on modal models.
  • Vocal Synthesis: functions for various vocal synthesis techniques (e.g., fof, source/filter, etc.) and vocal synthesizers.
  • Misc Functions: any other functions that don't fit in the previous category (e.g., nonlinear filters, etc.)

This library is part of the Faust Physical Modeling ToolKit. More information on how to use this library can be found on this page: https://ccrma.stanford.edu/~rmichon/pmFaust. Tutorials on how to make physical models of musical instruments using Faust can be found here as well.

Global Variables

Useful pre-defined variables for physical modeling.

speedOfSound

Speed of sound in meters per second (340m/s).


maxLength

The default maximum length (3) in meters of strings and tubes used in this library. This variable should be overriden to allow longer strings or tubes.


Conversion Tools

Useful conversion tools for physical modeling.

f2l

Frequency to length in meters.

Usage

f2l(freq) : distanceInMeters

Where:

  • freq: the frequency

l2f

Length in meters to frequency.

Usage

l2f(length) : freq

Where:

  • length: length/distance in meters

l2s

Length in meters to number of samples.

Usage

l2s(l) : numberOfSamples

Where:

  • l: length in meters

Bidirectional Utilities

Set of fundamental functions to create bi-directional block diagrams in Faust. These elements are used as the basis of this library to connect high level elements (e.g., mouthpieces, strings, bridge, instrument body, etc.). Each block has 3 inputs and 3 outputs. The first input/output carry left going waves, the second input/output carry right going waves, and the third input/output is used to carry any potential output signal to the end of the algorithm.

basicBlock

Empty bidirectional block to be used with chain: 3 signals ins and 3 signals out.

Usage

chain(basicBlock : basicBlock : etc.)

chain

Creates a chain of bidirectional blocks. Blocks must have 3 inputs and outputs. The first input/output carry left going waves, the second input/output carry right going waves, and the third input/output is used to carry any potential output signal to the end of the algorithm. The implied one sample delay created by the ~ operator is generalized to the left and right going waves. Thus, n blocks in chain() will add an n samples delay to both left and right going waves.

Usage

leftGoingWaves,rightGoingWaves,mixedOutput : chain( A : B ) : leftGoingWaves,rightGoingWaves,mixedOutput
with{
     A = _,_,_;
     B = _,_,_;
};

inLeftWave

Adds a signal to left going waves anywhere in a chain of blocks.

Usage

model(x) = chain(A : inLeftWave(x) : B)

Where A and B are bidirectional blocks and x is the signal added to left going waves in that chain.


inRightWave

Adds a signal to right going waves anywhere in a chain of blocks.

Usage

model(x) = chain(A : inRightWave(x) : B)

Where A and B are bidirectional blocks and x is the signal added to right going waves in that chain.


in

Adds a signal to left and right going waves anywhere in a chain of blocks.

Usage

model(x) = chain(A : in(x) : B)

Where A and B are bidirectional blocks and x is the signal added to left and right going waves in that chain.


outLeftWave

Sends the signal of left going waves to the output channel of the chain.

Usage

chain(A : outLeftWave : B)

Where A and B are bidirectional blocks.


outRightWave

Sends the signal of right going waves to the output channel of the chain.

Usage

chain(A : outRightWave : B)

Where A and B are bidirectional blocks.


out

Sends the signal of right and left going waves to the output channel of the chain.

Usage

chain(A : out : B)

Where A and B are bidirectional blocks.


terminations

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. Of courses, this function can be nested within a chain.

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
};

lTermination

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 and can be nested within another chain.

Usage

lTerminations(a,b)
with{
     a = *(-1); // left termination
     b = chain(D : E : F); // bidirectional chain of blocks (D, E, F, etc.)
};

rTermination

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 and can be nested within another chain.

Usage

rTerminations(b,c)
with{
     b = chain(D : E : F); // bidirectional chain of blocks (D, E, F, etc.)
     c = *(-1); // right termination
};

closeIns

Closes the inputs of a bidirectional chain in all directions.

Usage

closeIns : chain(...) :  _,_,_

closeOuts

Closes the outputs of a bidirectional chain in all directions except for the main signal output (3d output).

Usage

_,_,_ : chain(...) : _

endChain

Closes the inputs and outputs of a bidirectional chain in all directions except for the main signal output (3d output).

Usage

endChain(chain(...)) : _

Basic Elements

Basic elements for physical modeling (e.g., waveguides, specific filters, etc.).

waveguideN

A series of waveguide functions based on various types of delays (see fdelay[n]).

List of functions

  • waveguideUd: unit delay waveguide
  • waveguideFd: fractional delay waveguide
  • waveguideFd2: second order fractional delay waveguide
  • waveguideFd4: fourth order fractional delay waveguide

Usage

chain(A : waveguideUd(nMax,n) : B)

Where:

  • nMax: the maximum length of the delays in the waveguide
  • n: the length of the delay lines in samples.

waveguide

Standard pm.lib waveguide (based on waveguideFd4).

Usage

chain(A : waveguide(nMax,n) : B)

Where:

  • nMax: the maximum length of the delays in the waveguide
  • n: the length of the delay lines in samples.

bridgeFilter

Generic two zeros bridge FIR filter (as implemented in the STK) that can be used to implement the reflectance violin, guitar, etc. bridges.

Usage

_ : bridge(brightness,absorption) : _

Where:

  • brightness: controls the damping of high frequencies (0-1)
  • absorption: controls the absorption of the brige and thus the t60 of the string plugged to it (0-1) (1 = 20 seconds)

modeFilter

Resonant bandpass filter that can be used to implement a single resonance (mode).

Usage

_ : modeFilter(freq,t60,gain) : _

Where:

  • freq: mode frequency
  • t60: mode resonance duration (in seconds)
  • gain: mode gain (0-1)

String Instruments

Low and high level string instruments parts. Most of the elements in this section can be used in a bidirectional chain.

stringSegment

A string segment without terminations (just a simple waveguide).

Usage

chain(A : stringSegment(maxLength,length) : B)

Where:

  • maxLength: the maximum length of the string in meters (should be static)
  • length: the length of the string in meters

openString

A bidirectional block implementing a basic "generic" string with a selectable excitation position. Lowpass filters are built-in and allow to simulate the effect of dispersion on the sound and thus to change the "stiffness" of the string.

Usage

chain(... : openString(length,stiffness,pluckPosition,excitation) : ...)

Where:

  • length: the length of the string in meters
  • stiffness: the stiffness of the string (0-1) (1 for max stiffness)
  • pluckPosition: excitation position (0-1) (1 is bottom)
  • excitation: the excitation signal

nylonString

A bidirectional block implementing a basic nylon string with selectable excitation position. This element is based on openString and has a fix stiffness corresponding to that of a nylon string.

Usage

chain(... : nylonString(length,pluckPosition,excitation) : ...)

Where:

  • length: the length of the string in meters
  • pluckPosition: excitation position (0-1) (1 is bottom)
  • excitation: the excitation signal

steelString

A bidirectional block implementing a basic steel string with selectable excitation position. This element is based on openString and has a fix stiffness corresponding to that of a steel string.

Usage

chain(... : steelString(length,pluckPosition,excitation) : ...)

Where:

  • length: the length of the string in meters
  • pluckPosition: excitation position (0-1) (1 is bottom)
  • excitation: the excitation signal

openStringPick

A bidirectional block implementing a "generic" string with selectable excitation position. It also has a built-in pickup whose position is the same as the excitation position. Thus, moving the excitation position will also move the pickup.

Usage

chain(... : openStringPick(length,stiffness,pluckPosition,excitation) : ...)

Where:

  • length: the length of the string in meters
  • stiffness: the stiffness of the string (0-1) (1 for max stiffness)
  • pluckPosition: excitation position (0-1) (1 is bottom)
  • excitation: the excitation signal

openStringPickUp

A bidirectional block implementing a "generic" string with selectable excitation position and stiffness. It also has a built-in pickup whose position can be independenly selected. The only constraint is that the pickup has to be placed after the excitation position.

Usage

chain(... : openStringPickUp(length,stiffness,pluckPosition,excitation) : ...)

Where:

  • length: the length of the string in meters
  • stiffness: the stiffness of the string (0-1) (1 for max stiffness)
  • pluckPosition: pluck position between the top of the string and the pickup (0-1) (1 for same as pickup position)
  • pickupPosition: position of the pickup on the string (0-1) (1 is bottom)
  • excitation: the excitation signal

openStringPickDown

A bidirectional block implementing a "generic" string with selectable excitation position and stiffness. It also has a built-in pickup whose position can be independenly selected. The only constraint is that the pickup has to be placed before the excitation position.

Usage

chain(... : openStringPickDown(length,stiffness,pluckPosition,excitation) : ...)

Where:

  • length: the length of the string in meters
  • stiffness: the stiffness of the string (0-1) (1 for max stiffness)
  • pluckPosition: pluck position on the string (0-1) (1 is bottom)
  • pickupPosition: position of the pickup between the top of the string and the excitation position (0-1) (1 is excitation position)
  • excitation: the excitation signal

ksReflexionFilter

The "typical" one-zero Karplus-strong feedforward reflexion filter. This filter will be typically used in a termination (see below).

Usage

terminations(_,chain(...),ksReflexionFilter)

rStringRigidTermination

Bidirectional block implementing a right rigid string termination (no damping, just phase inversion).

Usage

chain(rStringRigidTermination : stringSegment : ...)

lStringRigidTermination

Bidirectional block implementing a left rigid string termination (no damping, just phase inversion).

Usage

chain(... : stringSegment : lStringRigidTermination)

elecGuitarBridge

Bidirectional block implementing a simple electric guitar bridge. This block is based on bridgeFilter. The bridge doesn't implement transmittance since it is not meant to be connected to a body (unlike acoustic guitar). It also partially sets the resonance duration of the string with the nuts used on the other side.

Usage

chain(... : stringSegment : elecGuitarBridge)

elecGuitarNuts

Bidirectional block implementing a simple electric guitar nuts. This block is based on bridgeFilter and does essentially the same thing as elecGuitarBridge, but on the other side of the chain. It also partially sets the resonance duration of the string with the bridge used on the other side.

Usage

chain(elecGuitarNuts : stringSegment : ...)

guitarBridge

Bidirectional block implementing a simple acoustic guitar bridge. This bridge damps more hight frequencies than elecGuitarBridge and implements a transmittance filter. It also partially sets the resonance duration of the string with the nuts used on the other side.

Usage

chain(... : stringSegment : guitarBridge)

guitarNuts

Bidirectional block implementing a simple acoustic guitar nuts. This nuts damps more hight frequencies than elecGuitarNuts and implements a transmittance filter. It also partially sets the resonance duration of the string with the bridge used on the other side.

Usage

chain(guitarNuts : stringSegment : ...)

idealString

An "ideal" string with rigid terminations and where the plucking position and the pick-up position are the same. Since terminations are rigid, this string will ring forever.

Usage

1-1' : idealString(length,reflexion,xPosition,excitation)

With: * length: the length of the string in meters * pluckPosition: the plucking position (0.001-0.999) * excitation: the input signal for the excitation


ks

A Karplus-Strong string (in that case, the string is implemented as a one dimension waveguide).

Usage

ks(length,damping,excitation) : _

Where:

  • length: the length of the string in meters
  • damping: string damping (0-1)
  • excitation: excitation signal

ks_ui_MIDI

Ready-to-use, MIDI-enabled Karplus-Strong string with buil-in UI.

Usage

ks_ui_MIDI : _

elecGuitarModel

A simple electric guitar model (without audio effects, of course) with selectable pluck position. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function. Pitch is changed by changing the length of the string and not through a finger model.

Usage

elecGuitarModel(length,pluckPosition,mute,excitation) : _

Where:

  • length: the length of the string in meters
  • pluckPosition: pluck position (0-1) (1 is on the bridge)
  • mute: mute coefficient (1 for no mute and 0 for instant mute)
  • excitation: excitation signal

elecGuitar

A simple electric guitar model with steel strings (based on elecGuitarModel) implementing an excitation model. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function.

Usage

elecGuitar(length,pluckPosition,trigger) : _

Where:

  • length: the length of the string in meters
  • pluckPosition: pluck position (0-1) (1 is on the bridge)
  • mute: mute coefficient (1 for no mute and 0 for instant mute)
  • gain: gain of the pluck (0-1)
  • trigger: trigger signal (1 for on, 0 for off)

elecGuitar_ui_MIDI

Ready-to-use MIDI-enabled electric guitar physical model with built-in UI.

Usage

elecGuitar_ui_MIDI : _

guitarBody

WARNING: not implemented yet! Bidirectional block implementing a simple acoustic guitar body.

Usage

chain(... : guitarBody)

guitarModel

A simple acoustic guitar model with steel strings and selectable excitation position. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function. Pitch is changed by changing the length of the string and not through a finger model. WARNING: this function doesn't currently implement a body (just strings and bridge)

Usage

guitarModel(length,pluckPosition,excitation) : _

Where:

  • length: the length of the string in meters
  • pluckPosition: pluck position (0-1) (1 is on the bridge)
  • excitation: excitation signal

guitar

A simple acoustic guitar model with steel strings (based on guitarModel) implementing an excitation model. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function.

Usage

guitar(length,pluckPosition,trigger) : _

Where:

  • length: the length of the string in meters
  • pluckPosition: pluck position (0-1) (1 is on the bridge)
  • gain: gain of the excitation
  • trigger: trigger signal (1 for on, 0 for off)

guitar_ui_MIDI

Ready-to-use MIDI-enabled steel strings acoustic guitar physical model with built-in UI.

Usage

guitar_ui_MIDI : _

nylonGuitarModel

A simple acoustic guitar model with nylon strings and selectable excitation position. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function. Pitch is changed by changing the length of the string and not through a finger model. WARNING: this function doesn't currently implement a body (just strings and bridge)

Usage

nylonGuitarModel(length,pluckPosition,excitation) : _

Where:

  • length: the length of the string in meters
  • pluckPosition: pluck position (0-1) (1 is on the bridge)
  • excitation: excitation signal

nylonGuitar

A simple acoustic guitar model with steel strings (based on nylonGuitarModel) implementing an excitation model. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function.

Usage

nylonGuitar(length,pluckPosition,trigger) : _

Where:

  • length: the length of the string in meters
  • pluckPosition: pluck position (0-1) (1 is on the bridge)
  • gain: gain of the excitation (0-1)
  • trigger: trigger signal (1 for on, 0 for off)

nylonGuitar_ui_MIDI

Ready-to-use MIDI-enabled nylon strings acoustic guitar physical model with built-in UI.

Usage

nylonGuitar_ui_MIDI : _

Bowed String Instruments

Low and high level basic string instruments parts. Most of the elements in this section can be used in a bidirectional chain.

bowTable

Extremely basic bow table that can be used to implement a wide range of bow types for many different bowed string instruments (violin, cello, etc.)

Usage

excitation : bowTable(offeset,slope) : _

Where:

  • excitation: an excitation signal
  • offset: table offset
  • slope: table slope

violinBowTable

Violin bow table based on bowTable.

Usage

bowVelocity : violinBowTable(bowPressure) : _

Where:

  • bowVelocity: velocity of the bow/excitation signal (0-1)
  • bowPressure: bow pressure on the string (0-1)

bowInteraction

Bidirectional block implementing the interaction of a bow in a chain.

Usage

chain(... : stringSegment : bowInteraction(bowTable) : stringSegment : ...)

Where:

  • bowTable: the bow table

violinBow

Bidirectional block implementing a violin bow and its interaction with a string.

Usage

chain(... : stringSegment : violinBow(bowPressure,bowVelocity) : stringSegment : ...)

Where:

  • bowVelocity: velocity of the bow / excitation signal (0-1)
  • bowPressure: bow pressure on the string (0-1)

violinBowedString

Violin bowed string bidirectional block with controllable bow position. Terminations are not implemented in this model.

Usage

chain(nuts : violinBowedString(stringLength,bowPressure,bowVelocity,bowPosition) : bridge)

Where:

  • stringLength: the length of the string in meters
  • bowVelocity: velocity of the bow / excitation signal (0-1)
  • bowPressure: bow pressure on the string (0-1)
  • bowPosition: the position of the bow on the string (0-1)

violinNuts

Bidirectional block implementing simple violin nuts. This function is based on bridgeFilter.

Usage

chain(violinNuts : stringSegment : ...)

violinBridge

Bidirectional block implementing a simple violin bridge. This function is based on bridgeFilter.

Usage

chain(... : stringSegment : violinBridge

violinBody

Bidirectional block implementing a simple violin body (just a simple resonant lowpass filter).

Usage

chain(... : stringSegment : violinBridge : violinBody)

violinModel

Ready-to-use simple violin physical model. This model implements a single string. Additional strings should be created by making a polyphonic applications out of this function. Pitch is changed by changing the length of the string (and not through a finger model).

Usage

violinModel(stringLength,bowPressure,bowVelocity,bridgeReflexion,
bridgeAbsorption,bowPosition) : _

Where:

  • stringLength: the length of the string in meters
  • bowVelocity: velocity of the bow / excitation signal (0-1)
  • bowPressure: bow pressure on the string (0-1))
  • bowPosition: the position of the bow on the string (0-1)

violin_ui

Ready-to-use violin physical model with built-in UI.

Usage

violinModel_ui : _

violin_ui_MIDI

Ready-to-use MIDI-enabled violin physical model with built-in UI.

Usage

violin_ui_MIDI : _

Wind Instruments

Low and high level basic wind instruments parts. Most of the elements in this section can be used in a bidirectional chain.

openTube

A tube segment without terminations (same as stringSegment).

Usage

chain(A : openTube(maxLength,length) : B)

Where:

  • maxLength: the maximum length of the tube in meters (should be static)
  • length: the length of the tube in meters

reedTable

Extremely basic reed table that can be used to implement a wide range of single reed types for many different instruments (saxophone, clarinet, etc.).

Usage

excitation : reedTable(offeset,slope) : _

Where:

  • excitation: an excitation signal
  • offset: table offset
  • slope: table slope

fluteJetTable

Extremely basic flute jet table.

Usage

excitation : fluteJetTable : _

Where:

  • excitation: an excitation signal

brassLipsTable

Simple brass lips/mouthpiece table. Since this implementation is very basic and that the lips and tube of the instrument are coupled to each other, the length of that tube must be provided here.

Usage

excitation : brassLipsTable(tubeLength,lipsTension) : _

Where:

  • excitation: an excitation signal (can be DC)
  • tubeLength: length in meters of the tube connected to the mouthpiece
  • lipsTension: tension of the lips (0-1) (default: 0.5)

clarinetReed

Clarinet reed based on reedTable with controllable stiffness.

Usage

excitation : clarinetReed(stiffness) : _

Where:

  • excitation: an excitation signal
  • stiffness: reed stiffness (0-1)

clarinetMouthPiece

Bidirectional block implementing a clarinet mouthpiece as well as the various interactions happening with traveling waves. This element is ready to be plugged to a tube...

Usage

chain(clarinetMouthPiece(reedStiffness,pressure) : tube : etc.)

Where:

  • pressure: the pressure of the air flow (DC) created by the virtual performer (0-1). This can also be any kind of signal that will directly injected in the mouthpiece (e.g., breath noise, etc.)
  • reedStiffness: reed stiffness (0-1)

brassLips

Bidirectional block implementing a brass mouthpiece as well as the various interactions happening with traveling waves. This element is ready to be plugged to a tube...

Usage

chain(brassLips(tubeLength,lipsTension,pressure) : tube : etc.)

Where:

  • tubeLength: length in meters of the tube connected to the mouthpiece
  • lipsTension: tension of the lips (0-1) (default: 0.5)
  • pressure: the pressure of the air flow (DC) created by the virtual performer (0-1). This can also be any kind of signal that will directly injected in the mouthpiece (e.g., breath noise, etc.)

fluteEmbouchure

Bidirectional block implementing a flute embouchure as well as the various interactions happening with traveling waves. This element is ready to be plugged between tubes segments...

Usage

chain(... : tube : fluteEmbouchure(pressure) : tube : etc.)

Where:

  • pressure: the pressure of the air flow (DC) created by the virtual performer (0-1). This can also be any kind of signal that will directly injected in the mouthpiece (e.g., breath noise, etc.)

wBell

Generic wind instrument bell bidirectional block that should be placed at the end of a chain.

Usage

chain(... : wBell(opening))

Where:

  • opening: the "opening" of bell (0-1)

fluteHead

Simple flute head implementing waves reflexion.

Usage

chain(fluteHead : tube : ...)

fluteFoot

Simple flute foot implementing waves reflexion and dispersion.

Usage

chain(... : tube : fluteFoot)

clarinetModel

A simple clarinet physical model without tone holes (pitch is changed by changing the length of the tube of the instrument).

Usage

clarinetModel(length,pressure,reedStiffness,bellOpening) : _

Where:

  • tubeLength: the length of the tube in meters
  • pressure: the pressure of the air flow created by the virtual performer (0-1). This can also be any kind of signal that will directly injected in the mouthpiece (e.g., breath noise, etc.)
  • reedStiffness: reed stiffness (0-1)
  • bellOpening: the opening of bell (0-1)

clarinetModel_ui

Same as clarinetModel but with a built-in UI. This function doesn't implement a virtual "blower", thus pressure remains an argument here.

Usage

clarinetModel_ui(pressure) : _

Where:

  • pressure: the pressure of the air flow created by the virtual performer (0-1). This can also be any kind of signal that will be directly injected in the mouthpiece (e.g., breath noise, etc.)

clarinet_ui

Ready-to-use clarinet physical model with built-in UI based on clarinetModel.

Usage

clarinet_ui : _

clarinet_ui_MIDI

Ready-to-use MIDI compliant clarinet physical model with built-in UI.

Usage

clarinet_ui_MIDI : _

brassModel

A simple generic brass instrument physical model without pistons (pitch is changed by changing the length of the tube of the instrument). This model is kind of hard to control and might not sound very good if bad parameters are given to it...

Usage

brassModel(tubeLength,lipsTension,mute,pressure) : _

Where:

  • tubeLength: the length of the tube in meters
  • lipsTension: tension of the lips (0-1) (default: 0.5)
  • mute: mute opening at the end of the instrument (0-1) (default: 0.5)
  • pressure: the pressure of the air flow created by the virtual performer (0-1). This can also be any kind of signal that will directly injected in the mouthpiece (e.g., breath noise, etc.)

brassModel_ui

Same as brassModel but with a built-in UI. This function doesn't implement a virtual "blower", thus pressure remains an argument here.

Usage

brassModel_ui(pressure) : _

Where:

  • pressure: the pressure of the air flow created by the virtual performer (0-1). This can also be any kind of signal that will be directly injected in the mouthpiece (e.g., breath noise, etc.)

brass_ui

Ready-to-use brass instrument physical model with built-in UI based on brassModel.

Usage

brass_ui : _

brass_ui_MIDI

Ready-to-use MIDI-controllable brass instrument physical model with built-in UI.

Usage

brass_ui_MIDI : _

fluteModel

A simple generic brass instrument physical model without tone holes (pitch is changed by changing the length of the tube of the instrument).

Usage

fluteModel(tubeLength,lipsTension,mute,pressure) : _

Where:

  • tubeLength: the length of the tube in meters
  • mouthPosition: position of the mouth on the embouchure (0-1) (default: 0.5)
  • pressure: the pressure of the air flow created by the virtual performer (0-1). This can also be any kind of signal that will directly injected in the mouthpiece (e.g., breath noise, etc.)

fluteModel_ui

Same as fluteModel but with a built-in UI. This function doesn't implement a virtual "blower", thus pressure remains an argument here.

Usage

fluteModel_ui(pressure) : _

Where:

  • pressure: the pressure of the air flow created by the virtual performer (0-1). This can also be any kind of signal that will be directly injected in the mouthpiece (e.g., breath noise, etc.)

flute_ui

Ready-to-use flute physical model with built-in UI based on fluteModel.

Usage

flute_ui : _

flute_ui_MIDI

Ready-to-use MIDI-controllable flute physical model with built-in UI.

Usage

brass_ui_MIDI : _

Exciters

Various kind of excitation signal generators.

impulseExcitation

Creates an impulse excitation of one sample.

Usage

gate = button('gate');
impulseExcitation(gate) : chain;

Where:

  • gate: a gate button

strikeModel

Creates a filtered noise excitation.

Usage

gate = button('gate');
strikeModel(LPcutoff,HPcutoff,sharpness,gain,gate) : chain;

Where:

  • HPcutoff: highpass cutoff frequency
  • LPcutoff: lowpass cutoff frequency
  • sharpness: sharpness of the attack and release (0-1)
  • gain: gain of the excitation
  • gate: a gate button/trigger signal (0/1)

strike

Strikes generator with controllable excitation position.

Usage

gate = button('gate');
strike(exPos,sharpness,gain,gate) : chain;

Where:

  • exPos: excitation position wiht 0: for max low freqs and 1: for max high freqs. So, on membrane for example, 0 would be the middle and 1 the edge
  • sharpness: sharpness of the attack and release (0-1)
  • gain: gain of the excitation
  • gate: a gate button/trigger signal (0/1)

pluckString

Creates a plucking excitation signal.

Usage

trigger = button('gate');
pluckString(stringLength,cutoff,maxFreq,sharpness,trigger)

Where:

  • stringLength: length of the string to pluck
  • cutoff: cutoff ratio (1 for default)
  • maxFreq: max frequency ratio (1 for default)
  • sharpness: sharpness of the attack and release (1 for default)
  • gain: gain of the excitation (0-1)
  • trigger: trigger signal (1 for on, 0 for off)

blower

A virtual blower creating a DC signal with some breath noise in it.

Usage

blower(pressure,breathGain,breathCutoff) : _

Where:

  • pressure: pressure (0-1)
  • breathGain: breath noise gain (0-1) (recommended: 0.005)
  • breathCutoff: breath cuttoff frequency (Hz) (recommended: 2000)

blower_ui

Same as blower but with a built-in UI.

Usage

blower : somethingToBeBlown

High and low level functions for modal synthesis of percussion instruments.

djembeModel

Dirt-simple djembe modal physical model. Mode parameters are empirically calculated and don't correspond to any measurements or 3D model. They kind of sound good though :).

Usage

excitation : djembeModel(freq)

Where:

  • excitation: excitation signal
  • freq: fundamental frequency of the bar

djembe

Dirt-simple djembe modal physical model. Mode parameters are empirically calculated and don't correspond to any measurements or 3D model. They kind of sound good though :).

This model also implements a virtual "exciter".

Usage

djembe(freq,strikePosition,strikeSharpness,gain,trigger

Where:

  • freq: fundamental frequency of the model
  • strikePosition: strike position (0 for the middle of the membrane and 1 for the edge)
  • strikeSharpness: sharpness of the strike (0-5, default: 0.5)
  • gain: gain of the strike
  • trigger: trigger signal (0: off, 1: on)

djembe_ui_MIDI

Simple MIDI controllable djembe physical model with built-in UI.

Usage

djembe_ui_MIDI : _

marimbaBarModel

Generic marimba tone bar modal model.

This model was generated using mesh2faust from a 3D CAD model of a marimba tone bar (libraries/modalmodels/marimbaBar). The corresponding CAD model is that of a C2 tone bar (original fundamental frequency: ~65Hz). While marimbaBarModel allows to translate the harmonic content of the generated sound by providing a frequency (freq), mode transposition has limits and the model will sound less and less like a marimba tone bar as it diverges from C2. To make an accurate model of a marimba, we'd want to have an independent model for each bar...

This model contains 5 excitation positions going linearly from the center bottom to the center top of the bar. Obviously, a model with more excitation position could be regenerated using mesh2faust.

Usage

excitation : marimbaBarModel(freq,exPos,t60,t60DecayRatio,t60DecaySlope)

Where:

  • excitation: excitation signal
  • freq: fundamental frequency of the bar
  • exPos: excitation position (0-4)
  • t60: T60 in seconds (recommended value: 0.1)
  • t60DecayRatio: T60 decay ratio (recommended value: 1)
  • t60DecaySlope: T60 decay slope (recommended value: 5)

marimbaResTube

Simple marimba resonance tube.

Usage

marimbaResTube(tubeLength,excitation)

Where:

  • tubeLength: the length of the tube in meters
  • excitation: the excitation signal (audio in)

marimbaModel

Simple marimba physical model implementing a single tone bar connected to tube. This model is scalable and can be adapted to any size of bar/tube (see marimbaBarModel to know more about the limitations of this type of system).

Usage

excitation : marimbaModel(freq,exPos) : _

Where:

  • freq: the frequency of the bar/tube couple
  • exPos: excitation position (0-4)

marimba

Simple marimba physical model implementing a single tone bar connected to tube. This model is scalable and can be adapted to any size of bar/tube (see marimbaBarModel to know more about the limitations of this type of system).

This function also implement a virtual exciter to drive the model.

Usage

excitation : marimba(freq,strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _

Where:

  • excitation: the excitation signal
  • freq: the frequency of the bar/tube couple
  • strikePosition: strike position (0-4)
  • strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)
  • strikeSharpness: shaarpness of the strike (recommened: ~0.25)
  • gain: gain of the strike (0-1)
  • trigger signal (0: off, 1: on)

marimba_ui_MIDI

Simple MIDI controllable marimba physical model with built-in UI implementing a single tone bar connected to tube. This model is scalable and can be adapted to any size of bar/tube (see marimbaBarModel to know more about the limitations of this type of system).

Usage

marimba_ui_MIDI : _

churchBellModel

Generic church bell modal model generated by mesh2faust from libraries/modalmodels/churchBell.

Modeled after T. Rossing and R. Perrin, Vibrations of Bells, Applied Acoustics 2, 1987.

Model height is 301 mm.

This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.

Usage

excitation : churchBellModel(nModes,exPos,t60,t60DecayRatio,t60DecaySlope)

Where:

  • excitation: the excitation signal
  • nModes: number of synthesized modes (max: 50)
  • exPos: excitation position (0-6)
  • t60: T60 in seconds (recommended value: 0.1)
  • t60DecayRatio: T60 decay ratio (recommended value: 1)
  • t60DecaySlope: T60 decay slope (recommended value: 5)

churchBell

Generic church bell modal model.

Modeled after T. Rossing and R. Perrin, Vibrations of Bells, Applied Acoustics 2, 1987.

Model height is 301 mm.

This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.

This function also implement a virtual exciter to drive the model.

Usage

excitation : churchBell(strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _

Where:

  • excitation: the excitation signal
  • strikePosition: strike position (0-6)
  • strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)
  • strikeSharpness: shaarpness of the strike (recommened: ~0.25)
  • gain: gain of the strike (0-1)
  • trigger signal (0: off, 1: on)

churchBell_ui

Church bell physical model based on churchBell with built-in UI.

Usage

churchBell_ui : _

englishBellModel

English church bell modal model generated by mesh2faust from libraries/modalmodels/englishBell.

Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell' Tone, Archives of Foundry Engineering, 2016.

Model height is 1 m.

This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.

Usage

excitation : englishBellModel(nModes,exPos,t60,t60DecayRatio,t60DecaySlope)

Where:

  • excitation: the excitation signal
  • nModes: number of synthesized modes (max: 50)
  • exPos: excitation position (0-6)
  • t60: T60 in seconds (recommended value: 0.1)
  • t60DecayRatio: T60 decay ratio (recommended value: 1)
  • t60DecaySlope: T60 decay slope (recommended value: 5)

englishBell

English church bell modal model.

Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell' Tone, Archives of Foundry Engineering, 2016.

Model height is 1 m.

This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.

This function also implement a virtual exciter to drive the model.

Usage

excitation : englishBell(strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _

Where:

  • excitation: the excitation signal
  • strikePosition: strike position (0-6)
  • strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)
  • strikeSharpness: shaarpness of the strike (recommened: ~0.25)
  • gain: gain of the strike (0-1)
  • trigger signal (0: off, 1: on)

englishBell_ui

English church bell physical model based on englishBell with built-in UI.

Usage

englishBell_ui : _

frenchBellModel

French church bell modal model generated by mesh2faust from libraries/modalmodels/frenchBell.

Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell' Tone, Archives of Foundry Engineering, 2016.

Model height is 1 m.

This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.

Usage

excitation : frenchBellModel(nModes,exPos,t60,t60DecayRatio,t60DecaySlope)

Where:

  • excitation: the excitation signal
  • nModes: number of synthesized modes (max: 50)
  • exPos: excitation position (0-6)
  • t60: T60 in seconds (recommended value: 0.1)
  • t60DecayRatio: T60 decay ratio (recommended value: 1)
  • t60DecaySlope: T60 decay slope (recommended value: 5)

frenchBell

French church bell modal model.

Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell' Tone, Archives of Foundry Engineering, 2016.

Model height is 1 m.

This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.

This function also implement a virtual exciter to drive the model.

Usage

excitation : frenchBell(strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _

Where:

  • excitation: the excitation signal
  • strikePosition: strike position (0-6)
  • strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)
  • strikeSharpness: shaarpness of the strike (recommened: ~0.25)
  • gain: gain of the strike (0-1)
  • trigger signal (0: off, 1: on)

frenchBell_ui

French church bell physical model based on frenchBell with built-in UI.

Usage

frenchBell_ui : _

germanBellModel

German church bell modal model generated by mesh2faust from libraries/modalmodels/germanBell.

Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell' Tone, Archives of Foundry Engineering, 2016.

Model height is 1 m.

This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.

Usage

excitation : germanBellModel(nModes,exPos,t60,t60DecayRatio,t60DecaySlope)

Where:

  • excitation: the excitation signal
  • nModes: number of synthesized modes (max: 50)
  • exPos: excitation position (0-6)
  • t60: T60 in seconds (recommended value: 0.1)
  • t60DecayRatio: T60 decay ratio (recommended value: 1)
  • t60DecaySlope: T60 decay slope (recommended value: 5)

germanBell

German church bell modal model.

Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell' Tone, Archives of Foundry Engineering, 2016.

Model height is 1 m.

This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.

This function also implement a virtual exciter to drive the model.

Usage

excitation : germanBell(strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _

Where:

  • excitation: the excitation signal
  • strikePosition: strike position (0-6)
  • strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)
  • strikeSharpness: shaarpness of the strike (recommened: ~0.25)
  • gain: gain of the strike (0-1)
  • trigger signal (0: off, 1: on)

germanBell_ui

German church bell physical model based on germanBell with built-in UI.

Usage

germanBell_ui : _

russianBellModel

Russian church bell modal model generated by mesh2faust from libraries/modalmodels/russianBell.

Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell' Tone, Archives of Foundry Engineering, 2016.

Model height is 2 m.

This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.

Usage

excitation : russianBellModel(nModes,exPos,t60,t60DecayRatio,t60DecaySlope)

Where:

  • excitation: the excitation signal
  • nModes: number of synthesized modes (max: 50)
  • exPos: excitation position (0-6)
  • t60: T60 in seconds (recommended value: 0.1)
  • t60DecayRatio: T60 decay ratio (recommended value: 1)
  • t60DecaySlope: T60 decay slope (recommended value: 5)

russianBell

Russian church bell modal model.

Modeled after D. Bartocha and . Baron, Influence of Tin Bronze Melting and Pouring Parameters on Its Properties and Bell' Tone, Archives of Foundry Engineering, 2016.

Model height is 2 m.

This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.

This function also implement a virtual exciter to drive the model.

Usage

excitation : russianBell(strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _

Where:

  • excitation: the excitation signal
  • strikePosition: strike position (0-6)
  • strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)
  • strikeSharpness: shaarpness of the strike (recommened: ~0.25)
  • gain: gain of the strike (0-1)
  • trigger signal (0: off, 1: on)

russianBell_ui

Russian church bell physical model based on russianBell with built-in UI.

Usage

russianBell_ui : _

standardBellModel

Standard church bell modal model generated by mesh2faust from libraries/modalmodels/standardBell.

Modeled after T. Rossing and R. Perrin, Vibrations of Bells, Applied Acoustics 2, 1987.

Model height is 1.8 m.

This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.

Usage

excitation : standardBellModel(nModes,exPos,t60,t60DecayRatio,t60DecaySlope)

Where:

  • excitation: the excitation signal
  • nModes: number of synthesized modes (max: 50)
  • exPos: excitation position (0-6)
  • t60: T60 in seconds (recommended value: 0.1)
  • t60DecayRatio: T60 decay ratio (recommended value: 1)
  • t60DecaySlope: T60 decay slope (recommended value: 5)

standardBell

Standard church bell modal model.

Modeled after T. Rossing and R. Perrin, Vibrations of Bells, Applied Acoustics 2, 1987.

Model height is 1.8 m.

This model contains 7 excitation positions going linearly from the bottom to the top of the bell. Obviously, a model with more excitation position could be regenerated using mesh2faust.

This function also implement a virtual exciter to drive the model.

Usage

excitation : standardBell(strikePosition,strikeCutoff,strikeSharpness,gain,trigger) : _

Where:

  • excitation: the excitation signal
  • strikePosition: strike position (0-6)
  • strikeCutoff: cuttoff frequency of the strike genarator (recommended: ~7000Hz)
  • strikeSharpness: shaarpness of the strike (recommened: ~0.25)
  • gain: gain of the strike (0-1)
  • trigger signal (0: off, 1: on)

standardBell_ui

Standard church bell physical model based on standardBell with built-in UI.

Usage

standardBell_ui : _

Vocal Synthesis

Vocal synthesizer functions (source/filter, fof, etc.).

formantValues

Formant data values.

The formant data used here come from the CSOUND manual http://www.csounds.com/manual/html/.

Usage

ba.take(j+1,formantValues.f(i)) : _
ba.take(j+1,formantValues.g(i)) : _
ba.take(j+1,formantValues.bw(i)) : _

Where:

  • i: formant number
  • j: (voiceType*nFormants)+vowel
  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)

voiceGender

Calculate the gender for the provided voiceType value. (0: male, 1: female)

Usage

voiceGender(voiceType) : _

Where:

  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)

skirtWidthMultiplier

Calculates value to multiply bandwidth to obtain skirtwidth for a Fof filter.

Usage

skirtWidthMultiplier(vowel,freq,gender) : _

Where:

  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)
  • freq: the fundamental frequency of the excitation signal
  • gender: gender of the voice used in the fof filter (0: male, 1: female)

autobendFreq

Autobends the center frequencies of formants 1 and 2 based on the fundamental frequency of the excitation signal and leaves all other formant frequencies unchanged. Ported from chant-lib. Reference: https://ccrma.stanford.edu/~rmichon/chantLib/

Usage

_ : autobendFreq(n,freq,voiceType) : _

Where:

  • n: formant index
  • freq: the fundamental frequency of the excitation signal
  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • input is the center frequency of the corresponding formant

vocalEffort

Changes the gains of the formants based on the fundamental frequency of the excitation signal. Higher formants are reinforced for higher fundamental frequencies. Ported from chant-lib. Reference: https://ccrma.stanford.edu/~rmichon/chantLib/

Usage

_ : vocalEffort(freq,gender) : _

Where:

  • freq: the fundamental frequency of the excitation signal
  • gender: the gender of the voice type (0: male, 1: female)
  • input is the linear amplitude of the formant

fof

Function to generate a single Formant-Wave-Function. Reference: https://ccrma.stanford.edu/~mjolsen/pdfs/smc2016_MOlsenFOF.pdf

Usage

_ : fof(fc,bw,a,g) : _

Where:

  • fc: formant center frequency,
  • bw: formant bandwidth (Hz),
  • sw: formant skirtwidth (Hz)
  • g: linear scale factor (g=1 gives 0dB amplitude response at fc)
  • input is an impulse signal to excite filter

fofSH

FOF with sample and hold used on bw and a parameter used in the filter-cycling FOF function fofCycle. Reference: https://ccrma.stanford.edu/~mjolsen/pdfs/smc2016_MOlsenFOF.pdf

Usage

_ : fofSH(fc,bw,a,g) : _

Where: all parameters same as for fof


fofCycle

FOF implementation where time-varying filter parameter noise is mitigated by using a cycle of n sample and hold FOF filters. Reference: https://ccrma.stanford.edu/~mjolsen/pdfs/smc2016_MOlsenFOF.pdf

Usage

_ : fofCycle(fc,bw,a,g,n) : _

Where:

  • n: the number of FOF filters to cycle through
  • all other parameters are same as for fof

fofSmooth

FOF implementation where time-varying filter parameter noise is mitigated by lowpass filtering the filter parameters bw and a with smooth.

Usage

_ : fofSmooth(fc,bw,sw,g,tau) : _

Where:

  • tau: the desired smoothing time constant in seconds
  • all other parameters are same as for fof

formantFilterFofCycle

Formant filter based on a single FOF filter. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. A cycle of n fof filters with sample-and-hold is used so that the fof filter parameters can be varied in realtime. This technique is more robust but more computationally expensive than formantFilterFofSmooth.Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.

Usage

_ : formantFilterFofCycle(voiceType,vowel,nFormants,i,freq) : _

Where:

  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)
  • nFormants: number of formant regions in frequency domain, typically 5
  • i: formant number (i.e. 0 - 4) used to index formant data value arrays
  • freq: fundamental frequency of excitation signal. Used to calculate rise time of envelope

formantFilterFofSmooth

Formant filter based on a single FOF filter. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Fof filter parameters are lowpass filtered to mitigate possible noise from varying them in realtime. Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.

Usage

_ : formantFilterFofSmooth(voiceType,vowel,nFormants,i,freq) : _

Where:

  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)
  • nFormants: number of formant regions in frequency domain, typically 5
  • i: formant number (i.e. 1 - 5) used to index formant data value arrays
  • freq: fundamental frequency of excitation signal. Used to calculate rise time of envelope

formantFilterBP

Formant filter based on a single resonant bandpass filter. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.

Usage

_ : formantFilterBP(voiceType,vowel,nFormants,i,freq) : _

Where:

  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)
  • nFormants: number of formant regions in frequency domain, typically 5
  • i: formant index used to index formant data value arrays
  • freq: fundamental frequency of excitation signal.

formantFilterbank

Formant filterbank which can use different types of filterbank functions and different excitation signals. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.

Usage

_ : formantFilterbank(voiceType,vowel,formantGen,freq) : _

Where:

  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)
  • formantGen: the specific formant filterbank function (i.e. FormantFilterbankBP, FormantFilterbankFof,...)
  • freq: fundamental frequency of excitation signal. Needed for FOF version to calculate rise time of envelope

formantFilterbankFofCycle

Formant filterbank based on a bank of fof filters. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.

Usage

_ : formantFilterbankFofCycle(voiceType,vowel,freq) : _

Where:

  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)
  • freq: the fundamental frequency of the excitation signal. Needed to calculate the skirtwidth of the FOF envelopes and for the autobendFreq and vocalEffort functions

formantFilterbankFofSmooth

Formant filterbank based on a bank of fof filters. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.

Usage

_ : formantFilterbankFofSmooth(voiceType,vowel,freq) : _

Where:

  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)
  • freq: the fundamental frequency of the excitation signal. Needed to calculate the skirtwidth of the FOF envelopes and for the autobendFreq and vocalEffort functions

formantFilterbankBP

Formant filterbank based on a bank of resonant bandpass filters. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the provided source to be realistic.

Usage

_ : formantFilterbankBP(voiceType,vowel) : _

Where:

  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u)
  • freq: the fundamental frequency of the excitation signal. Needed for the autobendFreq and vocalEffort functions

SFFormantModel

Simple formant/vocal synthesizer based on a source/filter model. The source and filterbank must be specified by the user. filterbank must take the same input parameters as formantFilterbank (BP/FofCycle /FofSmooth). Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the synthesized voice to be realistic.

Usage

SFFormantModel(voiceType,vowel,exType,freq,gain,source,filterbank,isFof) : _

Where:

  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u
  • exType: voice vs. fricative sound ratio (0-1 where 1 is 100% fricative)
  • freq: the fundamental frequency of the source signal
  • gain: linear gain multiplier to multiply the source by
  • isFof: whether model is FOF based (0: no, 1: yes)

SFFormantModelFofCycle

Simple formant/vocal synthesizer based on a source/filter model. The source is just a periodic impulse and the "filter" is a bank of FOF filters. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the synthesized voice to be realistic. This model does not work with noise in the source signal so exType has been removed and model does not depend on SFFormantModel function.

Usage

SFFormantModelFofCycle(voiceType,vowel,freq,gain) : _

Where:

  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u
  • freq: the fundamental frequency of the source signal
  • gain: linear gain multiplier to multiply the source by

SFFormantModelFofSmooth

Simple formant/vocal synthesizer based on a source/filter model. The source is just a periodic impulse and the "filter" is a bank of FOF filters. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the synthesized voice to be realistic.

Usage

SFFormantModelFofSmooth(voiceType,vowel,freq,gain) : _

Where:

  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u
  • freq: the fundamental frequency of the source signal
  • gain: linear gain multiplier to multiply the source by

SFFormantModelBP

Simple formant/vocal synthesizer based on a source/filter model. The source is just a sawtooth wave and the "filter" is a bank of resonant bandpass filters. Formant parameters are linearly interpolated allowing to go smoothly from one vowel to another. Voice type can be selected but must correspond to the frequency range of the synthesized voice to be realistic.

The formant data used here come from the CSOUND manual http://www.csounds.com/manual/html/.

Usage

SFFormantModelBP(voiceType,vowel,exType,freq,gain) : _

Where:

  • voiceType: the voice type (0: alto, 1: bass, 2: countertenor, 3: soprano, 4: tenor)
  • vowel: the vowel (0: a, 1: e, 2: i, 3: o, 4: u
  • exType: voice vs. fricative sound ratio (0-1 where 1 is 100% fricative)
  • freq: the fundamental frequency of the source signal
  • gain: linear gain multiplier to multiply the source by

SFFormantModelFofCycle_ui

Ready-to-use source-filter vocal synthesizer with built-in user interface.

Usage

SFFormantModelFofCycle_ui : _

SFFormantModelFofSmooth_ui

Ready-to-use source-filter vocal synthesizer with built-in user interface.

Usage

SFFormantModelFofSmooth_ui : _

SFFormantModelBP_ui

Ready-to-use source-filter vocal synthesizer with built-in user interface.

Usage

SFFormantModelBP_ui : _

SFFormantModelFofCycle_ui_MIDI

Ready-to-use MIDI-controllable source-filter vocal synthesizer.

Usage

SFFormantModelFofCycle_ui_MIDI : _

SFFormantModelFofSmooth_ui_MIDI

Ready-to-use MIDI-controllable source-filter vocal synthesizer.

Usage

SFFormantModelFofSmooth_ui_MIDI : _

SFFormantModelBP_ui_MIDI

Ready-to-use MIDI-controllable source-filter vocal synthesizer.

Usage

SFFormantModelBP_ui_MIDI : _

Misc Functions

Various miscellaneous functions.

allpassNL

Bidirectional block adding nonlinearities in both directions in a chain. Nonlinearities are created by modulating the coefficients of a passive allpass filter by the signal it is processing.

Usage

chain(... : allpassNL(nonlinearity) : ...)

Where:

  • nonlinearity: amount of nonlinearity to be added (0-1)

reverbs.lib

A library of reverb effects. Its official prefix is re.

Schroeder Reverberators

jcrev

This 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

Usage

_ : jcrev : _,_,_,_

satrev

This 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

Usage

_ : satrev : _,_

Feedback Delay Network (FDN) Reverberators

fdnrev0

Pure Feedback Delay Network Reverberator (generalized for easy scaling). fdnrev0 is a standard Faust function.

Usage

<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 length
  • delays: N delay lines, N a power of 2, lengths perferably coprime
  • BBSO: odd positive integer = order of bandsplit desired at freqs
  • freqs: NB-1 crossover frequencies separating desired frequency bands
  • durs: NB decay times (t60) desired for the various bands
  • loopgainmax: scalar gain between 0 and 1 used to "squelch" the reverb
  • nonl: nonlinearity (0 to 0.999..., 0 being linear)

Reference

https://ccrma.stanford.edu/~jos/pasp/FDN_Reverberation.html


zita_rev_fdn

Internal 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.

Usage

bus(8) : zita_rev_fdn(f1,f2,t60dc,t60m,fsmax) : bus(8)

Where:

  • f1: crossover frequency (Hz) separating dc and midrange frequencies
  • f2: 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)

Reference


zita_rev1_stereo

Extend zita_rev_fdn to include zita_rev1 input/output mapping in stereo mode. zita_rev1_stereo is a standard Faust function.

Usage

_,_ : 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_ambi

Extend zita_rev_fdn to include zita_rev1 input/output mapping in "ambisonics mode", as provided in the Linux C++ version.

Usage

_,_ : 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)


Freeverb

mono_freeverb

A 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.

Usage

_ : 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)

License

While this version is licensed LGPL (with exception) along with other GRAME library functions, the file freeverb.dsp in the examples directory of older Faust distributions, such as faust-0.9.85, was released under the BSD license, which is less restrictive.


stereo_freeverb

A 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.

Usage

_,_ : 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)

routes.lib

A library of basic elements to handle signal routing in Faust. Its official prefix is ro.

Functions Reference

cross

Cross n signals: (x1,x2,..,xn) -> (xn,..,x2,x1). cross is a standard Faust function.

Usage

cross(n)
_,_,_ : cross(3) : _,_,_

Where:

  • n: number of signals (int, must be known at compile time)

Note

Special case: cross2:

cross2 = _,cross(2),_;

crossnn

Cross two bus(n)s.

Usage

_,_,... : crossmm(n) : _,_,...

Where:

  • n: the number of signals in the bus

crossn1

Cross bus(n) and bus(1).

Usage

_,_,... : crossn1(n) : _,_,...

Where:

  • n: the number of signals in the first bus

interleave

Interleave 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), ...

Usage

_,_,_,_,_,_ : interleave(row,column) : _,_,_,_,_,_

Where:

  • row: the number of row (int, known at compile time)
  • column: the number of column (int, known at compile time)

butterfly

Addition (first half) then substraction (second half) of interleaved signals.

Usage

_,_,_,_ : butterfly(n) : _,_,_,_

Where:

  • n: size of the butterfly (n is int, even and known at compile time)

hadamard

Hadamard matrix function of size n = 2^k.

Usage

_,_,_,_ : hadamard(n) : _,_,_,_

Where:

  • n: 2^k, size of the matrix (int, must be known at compile time)

Note:

Implementation contributed by Remy Muller.


recursivize

Create a recursion from two arbitrary processors p and q.

Usage

_,_ : recursivize(p,q) : _,_

Where:

  • p: the forward arbitrary processor
  • q: the feedback arbitrary processor

signals.lib

A library of basic elements to handle signals in Faust. Its official prefix is si.

Functions Reference

bus

n parallel cables. bus is a standard Faust function.

Usage

bus(n)
bus(4) : _,_,_,_

Where:

  • n: is an integer known at compile time that indicates the number of parallel cables.

block

Block - terminate n signals. block is a standard Faust function.

Usage

_,_,... : block(n) : _,...

Where:

  • n: the number of signals to be blocked

interpolate

Linear interpolation between two signals.

Usage

_,_ : interpolate(i) : _

Where:

  • i: interpolation control between 0 and 1 (0: first input; 1: second input)

smoo

Smoothing function based on smooth ideal to smooth UI signals (sliders, etc.) down. smoo is a standard Faust function.

Usage

hslider(...) : smoo;

polySmooth

A 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.

Usage

hslider(...) : polysmooth(g,s,d) : _

Where:

  • g: the gate/trigger signal used when making polyphonic synths
  • s: the smoothness (see smooth)
  • d: the number of samples to wait before the signal start being smoothed after g switched to 1

smoothAndH

A smoothing function based on smooth that holds its output signal when a trigger is sent to it. This feature is convenient when implementing polyphonic instruments to prevent some smoothed parameter to change when a note-off event is sent.

Usage

hslider(...) : smoothAndH(g,s) : _

Where:

  • g: the hold signal (0 for hold, 1 for bypass)
  • s: the smoothness (see smooth)

bsmooth

Block smooth linear interpolation during a block of samples.

Usage

hslider(...) : bsmooth : _

dot

Dot product for two vectors of size n.

Usage

_,_,_,_,_,_ : dot(n) : _

Where:

  • n: size of the vectors (int, must be known at compile time)

smooth

Exponential smoothing by a unity-dc-gain one-pole lowpass. smooth is a standard Faust function.

Usage:

_ : smooth(tau2pole(tau)) : _

Where:

  • tau: desired smoothing time constant in seconds, or
hslider(...) : 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.

Reference:

https://ccrma.stanford.edu/~jos/mdft/Convolution_Example_2_ADSR.html


cbus

n parallel cables for complex signals. cbus is a standard Faust function.

Usage

cbus(n)
cbus(4) : (r0,i0), (r1,i1), (r2,i2), (r3,i3)

Where:

  • n: is an integer known at compile time that indicates the number of parallel cables.
  • each complex number is represented by two real signals as (real,imag)

cmul

multiply two complex signals pointwise. cmul is a standard Faust function.

Usage

(r1,i1) : cmul(r2,i2) : (_,_);

Where:

  • Each complex number is represented by two real signals as (real,imag), so
  • (r1,i1) = real and imaginary parts of signal 1
  • (r2,i2) = real and imaginary parts of signal 2

lag_ud

Lag filter with separate times for up and down.

Usage

_ : lag_ud(up, dn, signal) : _;

spats.lib

This library contains a collection of tools for sound spatialization. Its official prefix is sp.

panner

A simple linear stereo panner. panner is a standard Faust function.

Usage

_ : panner(g) : _,_

Where:

  • g: the panning (0-1)

spat

GMEM SPAT: n-outputs spatializer. spat is a standard Faust function.

Usage

_ : spat(n,r,d) : _,_,...

Where:

  • n: number of outputs
  • r: rotation (between 0 et 1)
  • d: distance of the source (between 0 et 1)

stereoize

Transform an arbitrary processor p into a stereo processor with 2 inputs and 2 outputs.

Usage

_,_ : stereoize(p) : _,_

Where:

  • p: the arbitrary processor

synths.lib

This library contains a collection of envelope generators. Its official prefix is sy.

popFilterPerc

A simple percussion instrument based on a "popped" resonant bandpass filter. popFilterPerc is a standard Faust function.

Usage

popFilterDrum(freq,q,gate) : _;

Where:

  • freq: the resonance frequency of the instrument
  • q: the q of the res filter (typically, 5 is a good value)
  • gate: the trigger signal (0 or 1)

dubDub

A simple synth based on a sawtooth wave filtered by a resonant lowpass. dubDub is a standard Faust function.

Usage

dubDub(freq,ctFreq,q,gate) : _;

Where:

  • freq: frequency of the sawtooth
  • ctFreq: cutoff frequency of the filter
  • q: Q of the filter
  • gate: the trigger signal (0 or 1)

sawTrombone

A simple trombone based on a lowpassed sawtooth wave. sawTrombone is a standard Faust function.

Usage

sawTrombone(att,freq,gain,gate) : _

Where:

  • att: exponential attack duration in s (typically 0.01)
  • freq: the frequency
  • gain: the gain (0-1)
  • gate: the gate (0 or 1)

combString

Simplest string physical model ever based on a comb filter. combString is a standard Faust function.

Usage

combString(freq,res,gate) : _;

Where:

  • freq: the frequency of the string
  • res: string T60 (resonance time) in second
  • gate: trigger signal (0 or 1)

additiveDrum

A simple drum using additive synthesis. additiveDrum is a standard Faust function.

Usage

additiveDrum(freq,freqRatio,gain,harmDec,att,rel,gate) : _

Where:

  • freq: the resonance frequency of the drum
  • freqRatio: 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 second
  • rel: release duration in second
  • gate: trigger signal (0 or 1)

fm

An FM synthesizer with an arbitrary number of modulators connected as a sequence. fm is a standard Faust function.

Usage

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)

vaeffects.lib

A library of virtual analog filter effects. Its official prefix is ve.

Functions Reference

moog_vcf

Moog "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.

Usage

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)

References


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).

Usage

moog_vcf_2b(res,fr)
moog_vcf_2bn(res,fr)

Where:

  • fr: corner-resonance frequency in Hz
  • res: Normalized amount of corner-resonance between 0 and 1 (0 is min resonance, 1 is maximum)

wah4

Wah effect, 4th order. wah4 is a standard Faust function.

Usage

_ : wah4(fr) : _

Where:

  • fr: resonance frequency in Hz

Reference

https://ccrma.stanford.edu/~jos/pasp/vegf.html


autowah

Auto-wah effect. autowah is a standard Faust function.

Usage

_ : autowah(level) : _;

Where:

  • level: amount of effect desired (0 to 1).

crybaby

Digitized CryBaby wah pedal. crybaby is a standard Faust function.

Usage

_ : crybaby(wah) : _

Where:

  • wah: "pedal angle" from 0 to 1

Reference

https://ccrma.stanford.edu/~jos/pasp/vegf.html


vocoder

A very simple vocoder where the spectrum of the modulation signal is analyzed using a filter bank. vocoder is a standard Faust function.

Usage

_ : vocoder(nBands,att,rel,BWRatio,source,excitation) : _;

Where:

  • nBands: Number of vocoder bands
  • att: Attack time in seconds
  • rel: Release time in seconds
  • BWRatio: Coefficient to adjust the bandwidth of each band (0.1 - 2)
  • source: Modulation signal
  • excitation: Excitation/Carrier signal

Licenses

STK 4.3 License

Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files (the "Software"), to deal in the Software without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is furnished to do so, subject to the following conditions:

The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software.

Any person wishing to distribute modifications to the Software is asked to send the modifications to the original developer so that they can be incorporated into the canonical version. For software copyrighted by Julius O. Smith III, email your modifications to . This is, however, not a binding provision of this license.

THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.

LGPL License

This program is free software; you can redistribute it and/or modify it under the terms of the GNU Lesser General Public License as published by the Free Software Foundation; either version 2.1 of the License, or (at your option) any later version.

This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for more details.

You should have received a copy of the GNU Lesser General Public License along with the GNU C Library; if not, write to the Free Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.