Faust Standard Libraries

January 4, 2017

Contents
Faust Libraries 15
Using the Faust Libraries . . . . . . . . . . . . . . . . . . . . . . . . . 15
Contributing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
New Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
New Libraries . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
General Organization . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
Coding Conventions . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Documentation . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Library Import . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
Demo Functions . . . . . . . . . . . . . . . . . . . . . . . . . . 20
Standard Functions . . . . . . . . . . . . . . . . . . . . . . . . 21
The question of licensing/authoring/copyrigth . . . . . . . . . . . . . . 21

Standard Functions 21
Analysis Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
Basic Elements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
Conversion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
Effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
Envelope Generators . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
Oscillators/Sound Generators . . . . . . . . . . . . . . . . . . . . . . . 24
Synths . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25

1
[Link] 25
Amplitude Tracking . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
amp_follower . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
amp_follower_ud . . . . . . . . . . . . . . . . . . . . . . . . . . 26
amp_follower_ar . . . . . . . . . . . . . . . . . . . . . . . . . . 27
Spectrum-Analyzers . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
mth_octave_analyzer . . . . . . . . . . . . . . . . . . . . . . . . 28
Mth-Octave Spectral Level . . . . . . . . . . . . . . . . . . . . . . . . 28
mth_octave_spectral_level6e . . . . . . . . . . . . . . . . . . 29
[third|half]_octave_[analyzer|filterbank] . . . . . . . . . 29
Arbritary-Crossover Filter-Banks and Spectrum Analyzers . . . . . . . 29
analyzer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30

[Link] 30
Conversion Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
samp2sec . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
sec2samp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
db2linear . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
linear2db . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
lin2LogGain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
log2LinGain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
tau2pole . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
pole2tau . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
midikey2hz . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
pianokey2hz . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
hz2pianokey . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
Counters and Time/Tempo Tools . . . . . . . . . . . . . . . . . . . . . 35
countdown . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
countup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
sweep . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
time . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36

2
 tempo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
period . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
pulse . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
pulsen . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
beat . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
pulse_countup . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
pulse_countdown . . . . . . . . . . . . . . . . . . . . . . . . . . 38
pulse_countup_loop . . . . . . . . . . . . . . . . . . . . . . . . 38
pulse_countdown_loop . . . . . . . . . . . . . . . . . . . . . . . 39
Array Processing/Pattern Matching . . . . . . . . . . . . . . . . . . . 39
count . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
take . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
subseq . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
Selectors (Conditions) . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
if . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
selector . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
selectn . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
select2stereo . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
Other . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
latch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
sAndH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
peakhold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
peakholder . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
impulsify . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
automat . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44
bpf . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44
bypass1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
bypass2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
toggle . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
on_and_off . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
selectoutn . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46

3
[Link] 47
Functions Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
compressor_mono . . . . . . . . . . . . . . . . . . . . . . . . . . 47
compressor_stereo . . . . . . . . . . . . . . . . . . . . . . . . . 48
limiter_1176_R4_mono . . . . . . . . . . . . . . . . . . . . . . . 48
limiter_1176_R4_stereo . . . . . . . . . . . . . . . . . . . . . . 49

[Link] 50
Basic Delay Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
fdelay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
sdelay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
Lagrange Interpolation . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
fdelaylti and fdelayltv . . . . . . . . . . . . . . . . . . . . . . 51
fdelay[n] . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52
Thiran Allpass Interpolation . . . . . . . . . . . . . . . . . . . . . . . 52
fdelay[n]a . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52

[Link] 53
Analyzers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
mth_octave_spectral_level_demo . . . . . . . . . . . . . . . . 53
Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54
parametric_eq_demo . . . . . . . . . . . . . . . . . . . . . . . . 54
spectral_tilt_demo . . . . . . . . . . . . . . . . . . . . . . . . 54
mth_octave_filterbank_demo and filterbank_demo . . . . . . 54
Effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
cubicnl_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
gate_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
compressor_demo . . . . . . . . . . . . . . . . . . . . . . . . . . 55
exciter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
moog_vcf_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
wah4_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56

4
 crybaby_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
vocoder_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
flanger_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57
phaser2_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57
freeverb_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57
stereo_reverb_tester . . . . . . . . . . . . . . . . . . . . . . . 57
fdnrev0_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
zita_rev_fdn_demo . . . . . . . . . . . . . . . . . . . . . . . . . 58
zita_rev1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
Generators . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
sawtooth_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
virtual_analog_oscillator_demo . . . . . . . . . . . . . . . . 59
oscrs_demo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59

[Link] 60
Functions Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
smoothEnvelope . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
ar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
asr . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
adsr . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61

[Link] 62
Basic Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
zero . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
pole . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
integrator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
dcblockerat . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
dcblocker . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
Comb Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
ff_comb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
ff_fcomb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
ffcombfilter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65

5
 fb_comb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
fb_fcomb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66
rev1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
fbcombfilter and ffbcombfilter . . . . . . . . . . . . . . . . . 67
allpass_comb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
allpass_fcomb . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68
rev2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
allpass_fcomb5 and allpass_fcomb1a . . . . . . . . . . . . . . 69
Direct-Form Digital Filter Sections . . . . . . . . . . . . . . . . . . . . 69
iir . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
fir . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
conv and convN . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
tf1, tf2 and tf3 . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
notchw . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
Direct-Form Second-Order Biquad Sections . . . . . . . . . . . . . . . 71
tf21, tf22, tf22t and tf21t . . . . . . . . . . . . . . . . . . . . 72
Ladder/Lattice Digital Filters . . . . . . . . . . . . . . . . . . . . . . . 72
av2sv . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72
bvav2nuv . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
iir_lat2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
allpassnt . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74
iir_kl . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74
allpassnklt . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
iir_lat1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
allpassn1mt . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
iir_nl . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
allpassnnlt . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
Useful Special Cases . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
tf2np . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
wgr . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
nlf2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78

6
 apnl . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78
Ladder/Lattice Allpass Filters . . . . . . . . . . . . . . . . . . . . . . 79
allpassn . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
allpassnn . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
allpasskl . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80
allpass1m . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80
Digital Filter Sections Specified as Analog Filter Sections . . . . . . . 81
tf2s and tf2snp . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
tf3slf . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
tf1s . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 82
tf2sb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
tf1sb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
Simple Resonator Filters . . . . . . . . . . . . . . . . . . . . . . . . . . 83
resonlp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
resonhp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84
resonbp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84
Butterworth Lowpass/Highpass Filters . . . . . . . . . . . . . . . . . . 85
lowpass . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
highpass . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
lowpass0_highpass1 . . . . . . . . . . . . . . . . . . . . . . . . 86
Special Filter-Bank Delay-Equalizing Allpass Filters . . . . . . . . . . 86
lowpass_plus|minus_highpass . . . . . . . . . . . . . . . . . . . 86
Elliptic (Cauer) Lowpass Filters . . . . . . . . . . . . . . . . . . . . . 86
lowpass3e . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86
lowpass6e . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
Elliptic Highpass Filters . . . . . . . . . . . . . . . . . . . . . . . . . . 87
highpass3e . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
highpass6e . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
Butterworth Bandpass/Bandstop Filters . . . . . . . . . . . . . . . . . 88
bandpass . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
bandstop . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89

7
 Elliptic Bandpass Filters . . . . . . . . . . . . . . . . . . . . . . . . . . 89
bandpass6e . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
bandpass12e . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
Parametric Equalizers (Shelf, Peaking) . . . . . . . . . . . . . . . . . . 89
low_shelf . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
high_shelf . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
peak_eq . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
peak_eq_cq . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
peak_eq_rm . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
spectral_tilt . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
levelfilter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
levelfilterN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
Mth-Octave Filter-Banks . . . . . . . . . . . . . . . . . . . . . . . . . 94
mth_octave_filterbank[n] . . . . . . . . . . . . . . . . . . . . 95
Arbritary-Crossover Filter-Banks and Spectrum Analyzers . . . . . . . 96
filterbank . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
filterbanki . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96

[Link] 97
encoder . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97
decoder . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
decoderStereo . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
Optimization Functions . . . . . . . . . . . . . . . . . . . . . . . . . . 99
optimBasic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
optimMaxRe . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
optimInPhase . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
Usage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
wider . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
map . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
rotate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101

8
[Link] 101
Functions Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
SR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
BS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102
PI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102
FTZ . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102
neg . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
sub(x,y) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
inv . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
cbrt . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
hypot . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
ldexp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104
scalb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104
log1p . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104
logb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104
ilogb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
log2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
expm1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
acosh . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
asinh . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
atanh . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
sinh . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
cosh . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
tanh . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
erf . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
erfc . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
gamma . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
lgamma . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
J0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
J1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
Jn . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108

9
 Y0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109
Y1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109
Yn . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109
fabs, fmax, fmin . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
np2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
frac . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
isnan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
chebychev . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
chebychevpoly . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
diffn . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112

[Link] 112
Dynamic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113
cubicnl . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113
gate_mono . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113
gate_stereo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114
Filtering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114
speakerbp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114
piano_dispersion_filter . . . . . . . . . . . . . . . . . . . . . 115
stereo_width . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115
Time Based . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116
echo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116
Pitch Shifting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117
transpose . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117
Meshes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117
mesh_square . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117

[Link] 118
Functions Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
noise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
multirandom . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
multinoise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119

10
 noises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
pink_noise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
pink_noise_vm . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
lfnoise, lfnoise0 and lfnoiseN . . . . . . . . . . . . . . . . . 121

[Link] 121
Wave-Table-Based Oscillators . . . . . . . . . . . . . . . . . . . . . . . 121
sinwaveform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
coswaveform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
phasor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
oscsin . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
osccos . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
oscp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
osci . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
LFOs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
lf_imptrain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
lf_pulsetrainpos . . . . . . . . . . . . . . . . . . . . . . . . . . 124
lf_squarewavepos . . . . . . . . . . . . . . . . . . . . . . . . . . 125
lf_squarewave . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
lf_trianglepos . . . . . . . . . . . . . . . . . . . . . . . . . . . 126
lf_triangle . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 126
Low Frequency Sawtooths . . . . . . . . . . . . . . . . . . . . . . . . . 126
lf_rawsaw . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127
lf_sawpos . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127
lf_saw . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127
lf_sawpos_phase . . . . . . . . . . . . . . . . . . . . . . . . . . 128
Bandlimited Sawtooth . . . . . . . . . . . . . . . . . . . . . . . . . . . 128
sawN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
sawNp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
saw2dpw . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
saw3 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130

11
 sawtooth . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130
saw2f2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130
saw2f4 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130
Bandlimited Pulse, Square, and Impulse Trains . . . . . . . . . . . . . 130
pulsetrainN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
pulsetrain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
squareN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
square . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
impulse . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
imptrainN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
imptrain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 133
triangleN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 133
triangle . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 133
Filter-Based Oscillators . . . . . . . . . . . . . . . . . . . . . . . . . . 133
oscb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 134
oscrq . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 134
oscrs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
oscrc . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
osc . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 136
oscs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 136
Waveguide-Resonator-Based Osccilators . . . . . . . . . . . . . . . . . 136
oscw . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 136
oscws . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
oscwq . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
oscw . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138

[Link] 138
Functions Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
flanger_mono . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
flanger_stereo . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
phaser2_mono . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 140
phaser2_stereo . . . . . . . . . . . . . . . . . . . . . . . . . . . 140

12
[Link] 141
chain(A:B:. . . ) . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
Requires . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
input(x) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
output() . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
terminations(a,b,c) . . . . . . . . . . . . . . . . . . . . . . . . . . 143
Requires . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 143
fullTerminations(a,b,c) . . . . . . . . . . . . . . . . . . . . . . . . 143
Requires . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 143
leftTermination(a,b) . . . . . . . . . . . . . . . . . . . . . . . . . 143
Requires . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 144
rightTermination(b,c) . . . . . . . . . . . . . . . . . . . . . . . . 144
Requires . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 144
waveguide(nMax,n) . . . . . . . . . . . . . . . . . . . . . . . . . . 144
idealString(length,reflexion,xPosition,x) . . . . . . . . . . . . . . 145

[Link] 145
Functions Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
jcrev . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
satrev . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 146
mono_freeverb . . . . . . . . . . . . . . . . . . . . . . . . . . . . 146
stereo_freeverb . . . . . . . . . . . . . . . . . . . . . . . . . . 147
fdnrev0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 147
zita_rev_fdn . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 148
zita_rev1_stereo . . . . . . . . . . . . . . . . . . . . . . . . . . 148
zita_rev1_ambi . . . . . . . . . . . . . . . . . . . . . . . . . . . 149

[Link] 149
Functions Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
cross . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
crossnn . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 150
crossn1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 150

13
 interleave . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 150
butterfly . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 151
hadamard . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 151
recursivize . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152

[Link] 152
Functions Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152
bus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152
block . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 153
interpolate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 153
smooth . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 154
smoo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 154
polySmooth . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 154
bsmooth . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
lag_ud . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
dot . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155

[Link] 156
panner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156
spat . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156
stereoize . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157

[Link] 157
popFilterPerc . . . . . . . . . . . . . . . . . . . . . . . . . . . . 158
dubDub . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 158
sawTrombone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 158
combString . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 159
additiveDrum . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 159
fm . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 160

14
[Link] 160
Functions Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . 161
moog_vcf . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 161
moog_vcf_2b[n] . . . . . . . . . . . . . . . . . . . . . . . . . . . 161
wah4 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 162
autowah . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 162
crybaby . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
vocoder . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163

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:
[Link] 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
[Link] in your Faust code:

import("[Link]");

This will give you access to all the Faust libraries through a series of environments:

sf: [Link]
an: [Link]
ba: [Link]
co: [Link]

15
 de: [Link]
dm: [Link]
en: [Link]
fi: [Link]
ho: [Link]
ma: [Link]
ef: [Link]
os: [Link]
no: [Link]
pf: [Link]
pm: [Link]
re: [Link]
ro: [Link]
si: [Link]
sp: [Link]
sy: [Link]
ve: [Link]

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

import("[Link]");
process = [Link](440);

In this case, were calling the osc function from [Link].

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

import("[Link]");
process = [Link](440);

Alternatively, environments can be created by hand:

os = library("[Link]");
process = [Link](440);

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

import("[Link]");
process = osc(440);

16
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. [Link]) 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 [Link] with its

own environment (2 letters - see [Link]).
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):

17
//############### libraryName ##################
// Description
//
// * Section Name 1
// * Section Name 2
// * ...
//
// It should be used using the `[...]` environment:
//
// ```
// [...] = library("libraryName");
// process = [...].functionCall;
// ```
//
// Another option is to import `[Link]` which already contains the `[...]`
// environment:
//
// ```
// import("[Link]");
// 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:

[Link]
[Link]
[Link]
[Link]
[Link]
[Link]
[Link]
[Link]
[Link]
[Link]
[Link]

18
 [Link]
[Link]
[Link]
[Link]
[Link]
[Link]
[Link]
[Link]
[Link] (not documented but example in /examples/misc)
[Link] (not documented but example in /examples/misc)
[Link]

Other deprecated libraries such as [Link], etc. are present but are not
documented to not confuse new users.
The doumentation of each library can be found in /documentation/[Link]
or in /documentation/[Link].
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 [Link] 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 declara-
tions (equivalent to ## in markdown) and //--- for function declarations
(equivalent to #### in markdown - see [Link] for an example).
Sections in function documentation should be declared as #### markdown
title.
Each function documentation provides a Usage section (see [Link]).

19
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
[Link] library was created and is imported by all the libraries:

an = library("[Link]");
ba = library("[Link]");
co = library("[Link]");
de = library("[Link]");
dm = library("[Link]");
en = library("[Link]");
fi = library("[Link]");
ho = library("[Link]");
ma = library("[Link]");
ef = library("[Link]");
os = library("[Link]");
no = library("[Link]");
pf = library("[Link]");
pm = library("[Link]");
re = library("[Link]");
ro = library("[Link]");
sp = library("[Link]");
si = library("[Link]");
sy = library("[Link]");
ve = library("[Link]");

For example, if we wanted to use the smooth function which is now declared in
[Link], we would do the following:

import("[Link]");

process = [Link](0.999);

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

Demo Functions

Demo functions are placed in [Link] 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.

20
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/[Link] 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 [Link]
and only one of them is considered to be standard, etc.

The question of licensing/authoring/copyrigth

Now that Faust libraries are not author specific, each function will be able to
have its own licence/author declaration. This means that some libraries wont
have a global licence/author/copyright declaration like it used to be the case.

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 [Link].

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 [Link] Pulses at a specific tempo

Block [Link] Terminate n signals

21
Function Type Function Name Description

Break Point Function [Link] Beak Point Function (BPF)

Bus [Link] Bus of n signals
Bypass (Mono) ba.bypass1 Mono bypass
Bypass (Stereo) ba.bypass2 Stereo bypass
Count Elements [Link] Count elements in a list
Count Down [Link] Samples count down
Count Up [Link] Samples count up
Delay (Integer) [Link] Integer delay
Delay (Float) [Link] Fractional delay
Impulsify [Link] Turns a signal into an impulse
Sample and Hold [Link] Sample and hold
Signal Crossing [Link] Cross n signals
Smoother (Default) [Link] Exponential smoothing
Smoother [Link] Exponential smoothing with controllable pole
Take Element [Link] Take en element from a list
Time [Link] 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
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

22
Function Type Function Name Description

Auto Wah [Link] Auto-Wah effect

Compressor co.compressor_mono Dynamic range compressor
Distortion [Link] Cubic nonlinearity distortion
Crybaby [Link] Crybaby wah pedal
Echo [Link] 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) [Link] Simple Schroeder reverberator
Reverb (Zita) re.zita_rev1_stereo High quality FDN reverberator
Panner [Link] Linear stereo panner
Pitch Shift [Link] Simple pitch shifter
Panner [Link] N outputs spatializer
Speaker Simulator [Link] Simple speaker simulator
Stereo Width ef.stereo_width Stereo width effect
Vocoder [Link] Simple vocoder
Wah ve.wah4 Wah effect

Envelope Generators

Function Type Function Name Description

ADSR [Link] Attack/Decay/Sustain/Release envelope generator

AR [Link] Attack/Release envelope generator
ASR [Link] Attack/Sustain/Release envelope generator
Exponential [Link] Exponential envelope generator

Filters

23
Function Type Function Name Description

Bandpass (Butterworth) [Link] Generic butterworth bandpass

Bandpass (Resonant) [Link] Virtual analog resonant bandpass
Bandstop (Butterworth) [Link] 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 [Link] Default dc blocker
Filterbank [Link] Generic filter bank
FIR (Arbitrary Order) [Link] Nth-order FIR filter
High Shelf fi.high_shelf High shelf
Highpass (Butterworth) [Link] Nth-order Butterworth highpass
Highpass (Resonant) [Link] Virtual analog resonant highpass
IIR (Arbitrary Order) [Link] Nth-order IIR filter
Level Filter [Link] Dynamic level lowpass
Low Shelf fi.low_shelf Low shelf
Lowpass (Butterworth) [Link] Nth-order Butterworth lowpass
Lowpass (Resonant) [Link] Virtual analog resonant lowpass
Notch Filter [Link] Simple notch filter
Peak Equalizer fi.peak_eq Peaking equalizer section

Oscillators/Sound Generators

Function Type Function Name Description

Impulse [Link] Generate an impulse on start-up

Impulse Train [Link] Band-limited impulse train
Phasor [Link] Simple phasor
Pink Noise no.pink_noise Pink noise generator
Pulse Train [Link] Band-limited pulse train
Pulse Train (Low Frequency) os.lf_imptrain Low-frequency pulse train
Sawtooth [Link] Band-limited sawtooth wave

24
Function Type Function Name Description

Sawtooth (Low Frequency) os.lf_saw Low-frequency sawtooth wave

Sine (Filter-Based) [Link] Sine oscillator (filter-based)
Sine (Table-Based) [Link] Sine oscillator (table-based)
Square [Link] Band-limited square wave
Square (Low Frequency) os.lf_squarewave Low-frequency square wave
Triangle [Link] Band-limited triangle wave
Triangle (Low Frequency) os.lf_triangle Low-frequency triangle wave
White Noise [Link] White noise generator

Synths

Function Type Function Name Description

Additive Drum [Link] Additive synthesis drum

Bandpassed Sawtooth [Link] Sawtooth through resonant bandpass
Comb String [Link] String model based on a comb filter
FM [Link] Frequency modulation synthesizer
Lowpassed Sawtooth [Link] Trombone based on a filtered sawtooth
Popping Filter [Link] Popping filter percussion instrument

[Link]
This library contains a collection of tools to analyze signals.
It should be used using the an environment:

an = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the an envi-

ronment:

import("[Link]");
process = [Link];

25
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 Engineers Handbook, Bernie Hutchins, Ithaca NY, 1975 Elec-

tronotes 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.

26
Reference

Digital Dynamic Range Compressor Design A Tutorial and Analysis,

by Dimitrios Giannoulis, Michael Massberg, and Joshua D. Reiss http://
[Link]/~josh/documents/GiannoulisMassbergReiss-dynamicrangecompression-JAES2012.
pdf

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) : _;

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
[Link]. 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).

27
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, Re-

galia 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: [Link]

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.

28
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.

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.

29
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)) : _,_,_

[Link]
A library of basic elements for Faust organized in 5 sections:

Conversion Tools
Counters and Time/Tempo Tools
Array Processing/Pattern Matching
Selectors (Conditions)
Other Tools (Misc)

It should be used using the ba environment:

ba = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the ba envi-

ronment:

import("[Link]");
process = [Link];

30
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.

31
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).

32
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.

33
Usage

midikey2hz(mk) : _

Where:

mk: the MIDI key number

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

34
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 starting point of the countup

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.

35
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

36
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

beat

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

Usage

beat(t) : _

Where:

t: tempo in BPM

37
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.

38
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)

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

39
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 doesnt have proper lists. Lists are simulated with parallel com-
positions and there is no empty list

Selectors (Conditions)

if

if-then-else implemented with a select2.

40
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)

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)

41
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.

42
Usage

_ : sAndH(t) : _

Where:

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

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.

43
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 = [Link](x0,y0) : [Link](x1,y1);

A more involved break-point function can contains any number of intermediate

points:

f = [Link](x0,y0) : [Link](x1,y1) : [Link](x2,y2) : [Link](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 = [Link](x0,y0) : ... : [Link](xi,yi) : ... : [Link](xn,yn);

implements a break-point function f such that :

44
 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.

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

45
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)

46
Example

process = 1 : selectoutn(3, sel) : par(i,3,bar) ;

sel = hslider("volume",0,0,2,1) : int;
bar = vbargraph("[Link]", 0, 1);

[Link]
A library of compressor effects.
It should be used using the co environment:

co = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the co envi-

ronment:

import("[Link]");
process = [Link];

Functions Reference

compressor_mono

Mono dynamic range compressors. compressor_mono is a standard Faust func-

tion

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

47
References

[Link]
[Link]
[Link]
Albert Graefs faust2pd/examples/synth/compressor_.dsp
More features: [Link]

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

[Link]
[Link]
[Link]
Albert Graefs faust2pd/examples/synth/compressor_.dsp
More features: [Link]

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

48
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: [Link]

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: [Link]

49
[Link]
This library contains a collection of delay functions.
It should be used using the de environment:

de = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the de envi-

ronment:

import("[Link]");
process = [Link];

Basic Delay Functions

delay

Simple d samples delay where n is the maximum delay length as a number of

samples (it needs to be a power of 2). Unlike the @ delay operator, this function
allows to preallocate memory which means that d can be changed dynamically
at run time as long as it remains smaller than n. delay is a standard Faust
function.

Usage

_ : delay(n,d) : _

Where:

n: the max delay length as a power of 2

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 (it needs to be a power of 2 -
see delay()). fdelay is a standard Faust function.

50
Usage

_ : fdelay(n,d) : _

Where:

n: the max delay length as a power of 2

d: the delay length as a number of samples (float)

sdelay

s(mooth)delay: a mono delay that doesnt click and doesnt transpose when the
delay time is changed.

Usage

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

Where :

N: maximal delay in samples (must be a constant power of 2, for example

65536)
it: interpolation time (in samples) for example 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.

51
References

[Link]
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 avail-
able where n is the order of the interpolation.

Thiran Allpass Interpolation

Thiran Allpass Interpolation

Reference [Link]
[Link]

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.

52
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 inter-
polation under time-varying conditions. (You may hear clicks when changing
the delay rapidly.)
First-order allpass interpolation, delay d in [0.5,1.5]

[Link]
This library contains a set of demo functions based on examples located in the
/examples folder.
It should be used using the dm environment:

dm = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the dm envi-

ronment:

import("[Link]");
process = [Link];

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

53
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

54
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 : _,_;

exciter

Psychoacoustic harmonic exciter, with GUI.

Usage

_ : exciter : _

55
References

[Link]
[Link]
Harmonic_Exciter

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 : _ ;

vocoder_demo

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

56
Usage

_ : vocoder_demo : _;

flanger_demo

Flanger effect application.

Usage

_,_ : flanger_demo : _,_;

phaser2_demo

Phaser effect demo application.

Usage

_,_ : phaser2_demo : _,_;

freeverb_demo

Freeverb demo application.

Usage

_,_ : freeverb_demo : _,_;

stereo_reverb_tester

Handy test inputs for reverberator demos below.

57
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

[Link](8) : zita_rev_fdn_demo : [Link](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 param-
eters are to be varied in real time, use smooth(0.999) or the like in the same
way.

58
Usage

_,_ : zita_rev1 : _,_

Reference [Link]
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 : _

59
[Link]
This library contains a collection of envelope generators.
It should be used using the en environment:

en = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the en envi-

ronment:

import("[Link]");
process = [Link];

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:

60
 a: attack (sec)
r: release (sec)
t: trigger signal (0 or 1)

asr

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

function.

Usage

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

Where:

a, s, r: attack (sec), sustain (percentage of t), release (sec)

t: trigger signal ( >0 for attack, then release is when t back to 0)

adsr

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

Faust function.

Usage

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

Where:

a, d, s, r: attack (sec), decay (sec), sustain (percentage of t), release (sec)

t: trigger signal ( >0 for attack, then release is when t back to 0)

61
[Link]
A library of filters and of more advanced filter-based sound processor organized
in 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

It should be used using the fi environment:

fi = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the fi envi-

ronment:

import("[Link]");
process = [Link];

Basic Filters

zero

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

62
Usage

_ : zero(z) : _

Where:

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

Reference [Link]

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 [Link]

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 + 2PI fb) by the low-frequency-
matching bilinear transform method (i.e., the standard frequency-scaling constant
2*SR).

63
Usage

_ : dcblockerat(fb) : _

Where:

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

Reference [Link]
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

64
Reference [Link]
[Link]

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 [Link]
[Link]

ffcombfilter

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

fb_comb

Feed-Back Comb Filter (integer delay).

65
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 [Link]
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 [Link]
html

66
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
[Link] 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 [Link]
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 imple-

mentation here is direct-form-2 requiring only one delay line.

67
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

[Link]
[Link]
[Link]

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 imple-

mentation 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

68
References

[Link]
[Link]
[Link]

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 [Link] 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)

69
Reference [Link]
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,...))

70
tf1, tf2 and tf3

tfN = Nth-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 [Link]

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 [Link]
Allpass_Filters.html

Direct-Form Second-Order Biquad Sections

Direct-Form Second-Order Biquad Sections

71
Reference [Link]
html

tf21, tf22, tf22t and tf21t

tfN = Nth-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 [Link]

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.

72
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 [Link]
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 ith 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 or arbitrary order.

73
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 reflexion coefficients (-1 1)

iir_kl

Kelly-Lochbaum ladder IIR filter or arbitrary order.

Usage

_ : iir_kl(bv,av) : _

Where:

bv: zeros as a bank of parallel signals

av: poles as a bank of parallel signals

74
allpassnklt

Kelly-Lochbaum ladder allpass.

Usage:

_ : allpassklt(n,sv) : _

Where:

n: the order of the filter

sv: the reflexion coefficients (-1 1)

iir_lat1

One-multiply latice IIR filter or 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 reflexion coefficients (-1 1)

75
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

J. D. Markel and A. H. Gray, Linear Prediction of Speech, New York:

Springer Verlag, 1976.
[Link]
[Link]

allpassnnlt

Normalized ladder allpass filter of arbitrary order.

Usage:

_ : allpassnnlt(n,sv) : _

Where:

n: the order of the filter

sv: the reflexion coefficients (-1,1)

References

J. D. Markel and A. H. Gray, Linear Prediction of Speech, New York:

Springer Verlag, 1976.
[Link]
[Link]

76
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

[Link]
[Link]
[Link]
html

77
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 [Link]
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

78
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

[Link]
html
[Link]
Linear Prediction of Speech, Markel and Gray, Springer Verlag, 1976

allpassn

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

Usage:

_ : allpassn(n,sv) : _

Where:

n: the order of the filter

sv: the reflexion 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).

79
Usage:

_ : allpassnn(n,tv) : _

Where:

n: the order of the filter

tv: the reflexion 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 reflexion 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 reflexion coefficients (-1 1)

80
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 [Link]
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.

81
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

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 [Link]
html

82
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) : _

83
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)

84
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

[Link]
html
butter function in Octave ("[z,p,g] = butter(N,1,s);")

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

[Link]
html
butter function in Octave ("[z,p,g] = butter(N,1,s);")

85
lowpass0_highpass1

TODO

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

TODO

Elliptic (Cauer) Lowpass Filters

Elliptic (Cauer) Lowpass Filters

References

<[Link]
functions ncauer and ellip in Octave

lowpass3e

Third-order Elliptic (Cauer) lowpass filter.

Usage

_ : lowpass3e(fc) : _

Where:

fc: -3dB frequency in Hz

86
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

87
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
+ wc2/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 [Link]

88
bandstop

Order 2*Nh Butterworth bandstop filter made using the transformation s <- s
+ wc2/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 [Link]

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)

89
References

[Link]
[Link]
Digital Audio Signal Processing, Udo Zolzer, Wiley, 1999, p. 124
[Link]
[Link]
[Link] in the Faust distribution
[Link] in the faust2pd distribution

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 approxima-
tion 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.

90
 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:

91
 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.)

92
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 Link to appear here when write up is done

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

93
Reference [Link]
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 [Link]
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 [Link]. 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)))

94
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, Re-

galia 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: [Link]

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,_); // 3d-order Butterworth

_ : mth_octave_filterbank5(M,ftop,N) : par(i,N,_); // 5th-roder Butterworth
mth_octave_filterbank_default = mth_octave_analyzer6e;

Where:

O: order of filter used to split each frequency band into two

95
 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.

96
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)) : _,_,_

[Link]
Faust library for high order ambisonic.
It should be used using the ho environment:

ho = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the ho envi-

ronment:

import("[Link]");
process = [Link];

encoder

Ambisonic encoder. Encodes a signal in the circular harmonics domain depending

on an order of decomposition and an angle.

97
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. Its 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

98
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.

99
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

100
rotate

Rotates the sound field.

Usage

_ : rotate(n, a) : _

Where:

n: the order
a: the angle in radian

[Link]
Mathematic library for Faust. Some functions are implemenented as Faust
foreign functions of math.h functions that are not part of Fausts primitives.
Defines also various constants and several utilities.
It should be used using the fi environment:

ma = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the ma envi-

ronment:

import("[Link]");
process = [Link];

Functions Reference

SR

Current sampling rate (between 1Hz and 192000Hz). Constant during program
execution.

101
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 : [Link]

102
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.

103
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.

104
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.

105
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.

106
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.

107
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.

108
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 : _

109
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

isnan

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

110
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 [Link]

chebychevpoly

Linear combination of the first Chebyshev polynomials.

111
Usage

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

Where:

cn: the different Chebychevs polynomials such that: cheby-

chevpoly((c0,c1,. . . ,cn)) = Sum of chebychev(i)*ci

Reference [Link]

diffn

Negated first-roder difference.

Usage

_ : diffn : _

[Link]
This library contains a collection of audio effects.
It should be used using the ef environment:

ef = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the ef envi-

ronment:

import("[Link]");
process = [Link];

112
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:

[Link]
[Link]

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)

113
References

[Link]
[Link]
[Link]

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

[Link]
[Link]
[Link]

Filtering

speakerbp

Dirt-simple speaker simulator (overall bandpass eq with observed roll-offs above

and below the passband).

114
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.

115
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

116
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 [Link]
[Link]

117
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.

[Link]
A library of noise generators.
It should be used using the no environment:

no = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the no envi-

ronment:

118
import("[Link]");
process = [Link];

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) : _

Where:

n: the number of decorrelated random numbers in parallel

multinoise

Generates multiple decorrelated noises in parallel.

Usage

multinoise(n) : _

Where:

n: the number of decorrelated random numbers in parallel

119
noises

TODO.

pink_noise

Pink noise (1/f noise) generator (third-order approximation) pink_noise is a

standard Faust function.

Usage

pink_noise : _;

Reference: [Link]
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

[Link]
[Link]

120
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 [Link]]
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 [Link])
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

[Link]
This library contains a collection of sound generators.
It should be used using the os environment:

os = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the os envi-

ronment:

import("[Link]");
process = [Link];

Wave-Table-Based Oscillators

sinwaveform

Sine waveform ready to use with a rdtable.

121
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)

122
oscsin

Sine wave oscillator. oscsin is a standard Faust function.

Usage

oscsin(freq) : _

Where:

freq: the frequency of the wave (Hz)

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

123
osci

Interpolated phase sine wave oscillator.

Usage

osci(freq) : _

Where:

freq: the frequency of the wave (Hz)

LFOs

Low-frequency oscillators have prefix lf_ (no aliasing suppression, signal-means

not necessarily zero).

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

124
Usage

lf_pulsetrainpos(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

125
lf_trianglepos

Positive unit-amplitude LF positive triangle wave

Usage

lf_trianglepos(freq) : _

Where:

freq: frequency in Hz

lf_triangle

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

function.

Usage

lf_triangle(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 sam-
pling 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).

126
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

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

127
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

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 suppres-

sion)

Reference

Kleimola, J.; Valimaki, V., Reducing Aliasing from Synthetic Audio

Signals Using Polynomial Transition Regions, in Signal Processing Letters,
IEEE , vol.19, no.2, pp.67-70, Feb. 2012
[Link]
pdf?sequence=9
[Link]

Method 2 (sawN) Differentiated Polynomial Waves (DPW) (for aliasing sup-

pression)

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.

128
Other Cases Correction-filtered versions of saw2: saw2f2, saw2f4 The cor-
rection 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

sawN

TODO: implemented but not documented. For now, you can look at the source
code.

sawNp

TODO: implemented but not documented. For now, you can look at the source
code.

saw2dpw

TODO: implemented but not documented. For now, you can look at the source
code.

129
saw3

TODO: implemented but not documented. For now, you can look at the source
code.

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: implemented but not documented. For now, you can look at the source
code.

saw2f4

TODO: implemented but not documented. For now, you can look at the source
code.

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.

130
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: implemented but not documented. For now, you can look at the source
code.

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

131
squareN

TODO: implemented but not documented. For now, you can look at the source
code.

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: implemented but not documented. For now, you can look at the source
code.

132
imptrain

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

standard Faust function.

Usage

imptrain(freq) : _

Where:

freq: frequency

triangleN

TODO: implemented but not documented. For now, you can look at the source
code.

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

133
Usage

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

References

[Link]
[Link]

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

[Link]
[Link]

134
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

[Link]
[Link]

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

[Link]
[Link]

135
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 Osccilators

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

136
Reference

[Link]
html

oscws

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

oscillator

Usage

oscws(freq) : _

Where:

freq: frequency

Reference

[Link]
html

oscwq

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

cosine and sine (quadrature) oscillator.

Usage

oscwq(freq) : _

Where:

freq: frequency

137
Reference

[Link]
html

oscw

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

cosine oscillator (default)

Usage

oscw(freq) : _

Where:

freq: frequency

Reference

[Link]
html

[Link]
A library of compressor effects.
It should be used using the pf environment:

pf = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the pf envi-

ronment:

import("[Link]");
process = [Link];

138
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 [Link]

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 [Link]

139
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:

[Link]
[Link]
An Allpass Approach to Digital Phasing and Flanging, Julius O. Smith
III, Proc. Int. Computer Music Conf. (ICMC-84), pp. 103-109, Paris,
1984.
CCRMA Tech. Report STAN-M-21: [Link]
stanms/stanm21/

phaser2_stereo

Stereo phasing effect. phaser2_stereo is a standard Faust function.

140
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:

[Link]
[Link]
An Allpass Approach to Digital Phasing and Flanging, Julius O. Smith
III, Proc. Int. Computer Music Conf. (ICMC-84), pp. 103-109, Paris,
1984.
CCRMA Tech. Report STAN-M-21: [Link]
stanms/stanm21/

[Link]
Faust physical modeling library.
It should be used using the fi environment:

pm = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the pm envi-

ronment:

import("[Link]");
process = [Link];

141
chain(A:B:. . . )

Creates a chain of bidirectional blocks. Blocks must have 3 inputs and out-
puts. The first input/output correspond to the left going signal, the second
input/output correspond to the right going signal and the third input/output
is the mix of the main signal output. The implied one sample delay created by
the ~ operator is generalized to the left and right going waves. Thus, n blocks
in chain() will add an n samples delay to both the left and right going waves.
### Usage

rightGoingWaves,leftGoingWaves,mixedOutput : chain(A:B) : rightGoingWaves,leftGoingWaves,mix

with{
A = _,_,_;
B = _,_,_;
};

Requires

[Link] (crossnn)

input(x)

Adds a waveguide input anywhere between 2 blocks in a chain of blocks (see

chain()). ### Usage

string(x) = chain(A:input(x):B)

Where x is the input signal to be added to the chain.

output()

Adds a waveguide output anywhere between 2 blocks in a chain of blocks and

sends it to the mix output channel (see chain()). ### Usage

chain(A:output:B)

142
terminations(a,b,c)

Creates terminations on both sides of a chain() without closing the inputs and
outputs of the bidirectional signals chain. As for chain(), this function adds a
1 sample delay to the bidirectional signal both ways. ### Usage

rightGoingWaves,leftGoingWaves,mixedOutput : terminations(a,b,c) : rightGoingWaves,leftGoing

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

Requires

[Link] (crossnn)

fullTerminations(a,b,c)

Same as terminations() but closes the inputs and outputs of the bidirectional
chain (only the mixed output remains). ### Usage

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

Requires

[Link] (crossnn)

leftTermination(a,b)

Creates a termination on the left side of a chain() without closing the inputs
and outputs of the bidirectional signals chain. This function adds a 1 sample
delay near the termination. ### Usage

143
rightGoingWaves,leftGoingWaves,mixedOutput : terminations(a,b) : rightGoingWaves,leftGoingWa
with{
a = *(-1); // left termination
b = chain(D:E:F); // bidirectional chain of blocks (D, E, F, etc.)
};

Requires

[Link] (crossnn)

rightTermination(b,c)

Creates a termination on the right side of a chain() without closing the inputs
and outputs of the bidirectional signals chain. This function adds a 1 sample
delay near the termination. ### Usage

rightGoingWaves,leftGoingWaves,mixedOutput : terminations(b,c) : rightGoingWaves,leftGoingWa

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

Requires

[Link] (crossnn)

waveguide(nMax,n)

A simple waveguide block based on a 4th order fractional delay. ### Usage

rightGoingWaves,leftGoingWaves,mixedOutput : waveguide(nMax,n) : rightGoingWaves,leftGoingWa

With: * nMax: the maximum length of the waveguide in samples * n the length
of the waveguide in samples. ### Requires [Link] (fdelay4)

144
idealString(length,reflexion,xPosition,x)

An ideal string with rigid terminations and where the plucking position and the
pick-up position are the same. ### Usage

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

With: * length: the length of the string in meters * reflexion: the coefficient
of reflexion (0-0.99999999) * pluckPosition: the plucking position (0.001-0.999)
* x: the input signal for the excitation ### Requires [Link] (crossnn)
[Link] (fdelay4)

[Link]
A library of reverb effects.
It should be used using the re environment:

re = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the re envi-

ronment:

import("[Link]");
process = [Link];

Functions Reference

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

145
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 [Link]

Usage

_ : satrev : _,_

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)

146
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)

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)

147
Reference [Link]

zita_rev_fdn

Internal 8x8 late-reverberation FDN used in the FOSS Linux reverb zita-rev1 by
Fons Adriaensen fons@[Link]. 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

[Link]
[Link]

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)

148
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)

[Link]
A library of basic elements to handle signal routing in Faust.
It should be used using the si environment:

ro = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the si envi-

ronment:

import("[Link]");
process = [Link];

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)

149
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+1 row), x(0+2row), . . . , x(1+0 row),
x(1+1row), x(1+2 row), . . .

150
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 = 2k.

Usage

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

Where:

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

Note: Implementation contributed by Remy Muller.

151
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

[Link]
A library of basic elements to handle signals in Faust.
It should be used using the si environment:

si = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the si envi-

ronment:

import("[Link]");
process = [Link];

Functions Reference

bus

n parallel cables. bus is a standard Faust function.

152
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)

153
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: [Link]
2_ADSR.html

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 doesnt 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.

154
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

bsmooth

Block smooth linear interpolation during a block of samples.

Usage

hslider(...) : bsmooth : _

lag_ud

Lag filter with separate times for up and down.

Usage

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

dot

Dot product for two vectors of size n.

155
Usage

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

Where:

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

[Link]
This library contains a collection of tools for sound spatialization.
It should be used using the sp environment:

sp = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the sp envi-

ronment:

import("[Link]");
process = [Link];

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.

156
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

[Link]
This library contains a collection of envelope generators.
It should be used using the sy environment:

sy = library("[Link]");
process = [Link];

Another option is to import [Link] which already contains the sy envi-

ronment:

import("[Link]");
process = [Link];

157
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.

158
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:

159
 freq: the resonance frequency of the drum
freqRatio: a list of ratio to choose the frequency of the mode in func-
tion 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)

[Link]
A library of virtual analog filter effects.
It should be used using the ve environment:

ve = library("[Link]");
process = [Link];

160
Another option is to import [Link] which already contains the ve envi-
ronment:

import("[Link]");
process = [Link];

Functions Reference
moog_vcf

Moog Voltage Controlled Filter (VCF) in analog form. Moog VCF im-
plemented 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

[Link]
[Link]

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](res4,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).

161
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 [Link]

autowah

Auto-wah effect. autowah is a standard Faust function.

Usage

_ : autowah(level) : _;

Where:

level: amount of effect desired (0 to 1).

162
crybaby

Digitized CryBaby wah pedal. crybaby is a standard Faust function.

Usage

_ : crybaby(wah) : _

Where:

wah: pedal angle from 0 to 1

Reference [Link]

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

163