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755ee317f9b8ce4583c5d47d70ab86508803d4f2fa523eb79018ece0f828e822 | s-e-a-m/faust-libraries | msheadphones.dsp | declare name "MID SIDE HEADPHONES";
declare version "001";
declare author "Giuseppe Silvi";
declare license "GNU-GPL-v3";
declare copyright "(c)SEAM 2019";
declare description "MID SIDE PANNER";
import("stdfaust.lib");
//import("../seam.lib");
deg2rad = *(ma.PI/180);
mspan(x) = m, s
with{
pot = vslider("[01] Azimuth [style:knob]", 0, -180, 180, 0.1) : deg2rad : si.smoo;
m = (0.5 * x) + (0.5 * (x * cos(pot)));
s = x *(sin(-pot));
};
process = os.osc(250)*0.0123 : mspan;
| https://raw.githubusercontent.com/s-e-a-m/faust-libraries/9120cccb9335f42407062eb4bf149188d8018b07/examples/msheadphones.dsp | faust | import("../seam.lib"); | declare name "MID SIDE HEADPHONES";
declare version "001";
declare author "Giuseppe Silvi";
declare license "GNU-GPL-v3";
declare copyright "(c)SEAM 2019";
declare description "MID SIDE PANNER";
import("stdfaust.lib");
deg2rad = *(ma.PI/180);
mspan(x) = m, s
with{
pot = vslider("[01] Azimuth [style:knob]", 0, -180, 180, 0.1) : deg2rad : si.smoo;
m = (0.5 * x) + (0.5 * (x * cos(pot)));
s = x *(sin(-pot));
};
process = os.osc(250)*0.0123 : mspan;
|
7fecd94a725658d736d5cef84d5745fcee6bd8d00b292990e0fdd68bf5db66d0 | s-e-a-m/faust-libraries | abmodule.dsp | declare name "MICHAEL GERZON AB-MODULE";
declare version "001";
declare author "Giuseppe Silvi";
declare license "GNU-GPL-v3";
declare copyright "(c)SEAM 2019";
declare description "MICHAEL GERZON AB-MODULE";
import("stdfaust.lib");
//import("../../seam.lib");
abmodule(LFU,RFD,RBU,LBD) = W,X,Y,Z
with{
W = (0.5 * (LFU + RFD + RBU + LBD));
X = (0.5 * (LFU + RFD - RBU - LBD));
Y = (0.5 * (LFU - RFD - RBU + LBD));
Z = (0.5 * (LFU - RFD + RBU - LBD));
};
process = abmodule;
| https://raw.githubusercontent.com/s-e-a-m/faust-libraries/9120cccb9335f42407062eb4bf149188d8018b07/examples/vst/abmodule.dsp | faust | import("../../seam.lib"); | declare name "MICHAEL GERZON AB-MODULE";
declare version "001";
declare author "Giuseppe Silvi";
declare license "GNU-GPL-v3";
declare copyright "(c)SEAM 2019";
declare description "MICHAEL GERZON AB-MODULE";
import("stdfaust.lib");
abmodule(LFU,RFD,RBU,LBD) = W,X,Y,Z
with{
W = (0.5 * (LFU + RFD + RBU + LBD));
X = (0.5 * (LFU + RFD - RBU - LBD));
Y = (0.5 * (LFU - RFD - RBU + LBD));
Z = (0.5 * (LFU - RFD + RBU - LBD));
};
process = abmodule;
|
a4f208cff65c9bc6f2cac1b912eac2865a491c6b3281f63412d3d8ecbc0af791 | friskgit/snares | disperse.dsp | // -*- compile-command: "cd .. && make jack src=src/disperse.dsp && cd -"; -*-&& cd -"; -*-
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
//---------------`Disperse audio randomly over x channels` --------------------------
//
// Each hit is output to a channel <= channels as controlled by the lfo
// in rndctrl. Due to the ma.fabs, there is a greater chance that signal
// is sent to lower outputs than higher
//
//
// 18 Juli 2019 Henrik Frisk [email protected]
//---------------------------------------------------
// GUI
posgroup(x) = vgroup("[0]position", x);
// Set the number of channels at compile time.
channels = 14;
integ(x) = x - ma.frac(x);
//imp = ba.pulse(hslider("tempo", 5000, 500, 10000, 1));
// Control the output channel
focus = posgroup(hslider("[1]disperse", 1, 0, 1, 0.0001));
position = posgroup(hslider("[0]position", 1, 0, channels, 1));
rate = ma.SR/1000.0;
rndctrl = (no.lfnoise(rate) * (channels + 1)) * focus : ma.fabs + position : int ;
outputctrl(imp) = rndctrl : ba.sAndH(imp);
// Wrap channels around the array.
ch_wrapped(imp) = ma.modulo(outputctrl(imp), channels);
// Main gate
process(imp, sig) = sig : ba.selectoutn(channels, ch_wrapped(imp)) ;
| https://raw.githubusercontent.com/friskgit/snares/bb43ea5e706a0ead6d65dd176a5c492b2f5d8f74/faust/snare/src/disperse.dsp | faust | -*- compile-command: "cd .. && make jack src=src/disperse.dsp && cd -"; -*-&& cd -"; -*-
---------------`Disperse audio randomly over x channels` --------------------------
Each hit is output to a channel <= channels as controlled by the lfo
in rndctrl. Due to the ma.fabs, there is a greater chance that signal
is sent to lower outputs than higher
18 Juli 2019 Henrik Frisk [email protected]
---------------------------------------------------
GUI
Set the number of channels at compile time.
imp = ba.pulse(hslider("tempo", 5000, 500, 10000, 1));
Control the output channel
Wrap channels around the array.
Main gate |
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
posgroup(x) = vgroup("[0]position", x);
channels = 14;
integ(x) = x - ma.frac(x);
focus = posgroup(hslider("[1]disperse", 1, 0, 1, 0.0001));
position = posgroup(hslider("[0]position", 1, 0, channels, 1));
rate = ma.SR/1000.0;
rndctrl = (no.lfnoise(rate) * (channels + 1)) * focus : ma.fabs + position : int ;
outputctrl(imp) = rndctrl : ba.sAndH(imp);
ch_wrapped(imp) = ma.modulo(outputctrl(imp), channels);
process(imp, sig) = sig : ba.selectoutn(channels, ch_wrapped(imp)) ;
|
915ac99d2b4762f217fd43b6681b68154c3145c3353215107b5c4c2500611616 | jameslnrd/mi_introduction_workshop_2020 | harmonicOscillator2.dsp | declare name "Harmonic Oscillator v2";
declare author "James Leonard";
declare date "April 2020";
/* ========= DESCRITPION =============
The simplest mass-interaction construct: a harmonic oscillator, built by assembling three elements:
a mass, a fixed point and a spring-damper interaction.
The resulting model is identical to an mi.osc element (which integrates them together).
- inputs: force impulse
- outputs: oscillator' position.
- controls: none.
*/
import("stdfaust.lib");
in1 = button("Frc Input 1"): ba.impulsify * 0.25; //write a specific force input signal operation here
OutGain = 1;
model = (
mi.mass(1., 0, 0., 0.),
mi.ground(0.),
par(i, nbFrcIn,_):
RoutingMassToLink ,
par(i, nbFrcIn,_):
mi.springDamper(0.1, 0.0003, 0., 0.),
par(i, nbOut+nbFrcIn, _):
RoutingLinkToMass
)~par(i, nbMass, _):
par(i, nbMass, !), par(i, nbOut , _)
with{
RoutingMassToLink(m0, m1) = /* routed positions */ m1, m0, /* outputs */ m0;
RoutingLinkToMass(l0_f1, l0_f2, p_out1, f_in1) = /* routed forces */ f_in1 + l0_f2, l0_f1, /* pass-through */ p_out1;
nbMass = 2;
nbFrcIn = 1;
nbOut = 1;
};
process = in1 : model:*(OutGain);
/*
========= MIMS SCRIPT USED FOR MODEL GENERATION =============
# MIMS script file
# Script author: James Leonard
# Assembled harmonic oscillator with M = 1, K = 0.1, Z = 0.0003
@m mass 1. 0. 0.
@g ground 0.
@sp springDamper @g @m 0.1 0.0003
# Add force input to the model
@in1 frcInput @m
# Add position output from the oscillator
@out1 posOutput @m
# end of MIMS script
*/ | https://raw.githubusercontent.com/jameslnrd/mi_introduction_workshop_2020/2f487dbc5b8e7cd83cbd962254e737bdb82948f6/00_BasicOscillator/harmonicOscillator2.dsp | faust | ========= DESCRITPION =============
The simplest mass-interaction construct: a harmonic oscillator, built by assembling three elements:
a mass, a fixed point and a spring-damper interaction.
The resulting model is identical to an mi.osc element (which integrates them together).
- inputs: force impulse
- outputs: oscillator' position.
- controls: none.
write a specific force input signal operation here
routed positions
outputs
routed forces
pass-through
========= MIMS SCRIPT USED FOR MODEL GENERATION =============
# MIMS script file
# Script author: James Leonard
# Assembled harmonic oscillator with M = 1, K = 0.1, Z = 0.0003
@m mass 1. 0. 0.
@g ground 0.
@sp springDamper @g @m 0.1 0.0003
# Add force input to the model
@in1 frcInput @m
# Add position output from the oscillator
@out1 posOutput @m
# end of MIMS script
| declare name "Harmonic Oscillator v2";
declare author "James Leonard";
declare date "April 2020";
import("stdfaust.lib");
OutGain = 1;
model = (
mi.mass(1., 0, 0., 0.),
mi.ground(0.),
par(i, nbFrcIn,_):
RoutingMassToLink ,
par(i, nbFrcIn,_):
mi.springDamper(0.1, 0.0003, 0., 0.),
par(i, nbOut+nbFrcIn, _):
RoutingLinkToMass
)~par(i, nbMass, _):
par(i, nbMass, !), par(i, nbOut , _)
with{
nbMass = 2;
nbFrcIn = 1;
nbOut = 1;
};
process = in1 : model:*(OutGain);
|
ee8d4e53149a7fc63a4fee5b0a9ca1b299f34b35c7a9090717e9db7add4701be | jameslnrd/mi_introduction_workshop_2020 | bowedOsc.dsp | declare name "Bowed Oscillator";
declare author "James Leonard";
declare date "April 2020";
/* ========= DESCRITPION =============
Friction-based interaction with a simple oscillator => cool squeaky sounds.
- inputs: position control of the "bowing" mass
- outputs: oscillator position.
- controls: none.
Note: the "type" parameter changes the way the friction interaction is calculated
(set to 0 for piecewise linear function or 1 for friction à-la-Bilbao).
*/
import("stdfaust.lib");
in1 = hslider("Bow Position", 0, 0, 100, 0.001):si.smoo:si.smoo:si.smoo; //Need very smooth position data here !
OutGain = 20;
type = 0;
model = (
mi.oscil(1., 0.1, 0.0003, 0, 0., 0.),
mi.posInput(1.):
RoutingMassToLink :
mi.nlBow(1.2, 0.001, type, 0., 1.),
par(i, nbOut, _):
RoutingLinkToMass
)~par(i, nbMass, _):
par(i, nbMass, !), par(i, nbOut , _)
with{
RoutingMassToLink(m0, m1) = /* routed positions */ m0, m1, /* outputs */ m0;
RoutingLinkToMass(l0_f1, l0_f2, p_out1) = /* routed forces */ l0_f1, l0_f2, /* pass-through */ p_out1;
nbMass = 2;
nbOut = 1;
};
process = in1 : model:*(OutGain);
/*
========= MIMS SCRIPT USED FOR MODEL GENERATION =============
# MIMS script file
# Script author: James Leonard
# parameter to switch how the
# bowing interaction is calculated
@type param 0
# Integrated harmonic oscillator
@o osc 1. 0.1 0.0003 0. 0.
# Position input, controlled by audio signal
@in1 posInput 1.
@b nlBow @o @in1 1.2 0.001 type
# Add position output from the oscillator
@out1 posOutput @o
# end of MIMS script
*/ | https://raw.githubusercontent.com/jameslnrd/mi_introduction_workshop_2020/2f487dbc5b8e7cd83cbd962254e737bdb82948f6/06_BowedOscillator/bowedOsc.dsp | faust | ========= DESCRITPION =============
Friction-based interaction with a simple oscillator => cool squeaky sounds.
- inputs: position control of the "bowing" mass
- outputs: oscillator position.
- controls: none.
Note: the "type" parameter changes the way the friction interaction is calculated
(set to 0 for piecewise linear function or 1 for friction à-la-Bilbao).
Need very smooth position data here !
routed positions
outputs
routed forces
pass-through
========= MIMS SCRIPT USED FOR MODEL GENERATION =============
# MIMS script file
# Script author: James Leonard
# parameter to switch how the
# bowing interaction is calculated
@type param 0
# Integrated harmonic oscillator
@o osc 1. 0.1 0.0003 0. 0.
# Position input, controlled by audio signal
@in1 posInput 1.
@b nlBow @o @in1 1.2 0.001 type
# Add position output from the oscillator
@out1 posOutput @o
# end of MIMS script
| declare name "Bowed Oscillator";
declare author "James Leonard";
declare date "April 2020";
import("stdfaust.lib");
OutGain = 20;
type = 0;
model = (
mi.oscil(1., 0.1, 0.0003, 0, 0., 0.),
mi.posInput(1.):
RoutingMassToLink :
mi.nlBow(1.2, 0.001, type, 0., 1.),
par(i, nbOut, _):
RoutingLinkToMass
)~par(i, nbMass, _):
par(i, nbMass, !), par(i, nbOut , _)
with{
nbMass = 2;
nbOut = 1;
};
process = in1 : model:*(OutGain);
|
d0bd42c06f782ab376b5de0509cbbac2e2ec7f71a4a6eaf8fcc2e77560f3b230 | grame-cncm/smartfaust | sfTrashShift.dsp | declare name "sfTrashShift";
declare version "1.1";
declare author "Christophe Lebreton";
declare license "BSD";
declare copyright "SmartFaust - GRAME(c)2013-2018";
import("stdfaust.lib");
//-------------------- MAIN -------------------------------
process = pitchshifter_drywet;
//--------------------------------------------------------------------------------------------------
// from FAUST example and adapted by Christophe Lebreton
// very simple real time pitch shifter
transpose (w, x, s, sig) = de.fdelay1s(d,sig)* ma.fmin(d/x,1) + de.fdelay1s(d+w,sig)*(1- ma.fmin(d/x,1))
with {
i = 1 - pow(2, s/12);
d = i : (+ : +(w) : fmod(_,w)) ~ _;
};
pitchshifter = transpose(w,x,s)
with {
//w = hslider("window [units (ms)]", 75, 10, 1000, 1)*SR*0.001;
w = (75)* ma.SR*(0.001);
//x = hslider("xfade [units (ms)]", 10, 1, 500, 1)*SR*0.001 : smooth (0.99);
x = w * 0.5;
s = (hslider("v:sfTrashShift parameter(s)/shift [units (cents)] [acc:0 1 -10 0 10][color: 255 0 0 ][hidden:1]", 0, -3600, 3600, 0.1))*0.01 : si.smooth (0.998);
};
dry_wet(x,y) = (1-c)*x + c*y
with {
c = hslider("v:sfTrashShift parameter(s)/dry_wet [acc:1 1 -10 0 10][color: 255 255 0 ][hidden:1] ",100,0,100,0.01):*(0.01):fi.lowpass(1,1):max(0):min(1);
};
pitchshifter_drywet = _<: _ , pitchshifter: dry_wet:*(volume):*(gain):*(out)
with {
volume = vslider ("h:sfTrashShift/Volume",1,0,2,0.001): si.smooth(0.998):max(0):min(2);
gain = hslider ("v:sfTrashShift parameter(s)/gain[acc:2 1 -10 0 10][color:255 255 0][hidden:1]",0.2,0,1,0.001): fi.lowpass(1,1):max(0):min(1);
out = checkbox ("h:sfTrashShift/ON/OFF"): si.smooth(0.998);
};
| https://raw.githubusercontent.com/grame-cncm/smartfaust/0a9c93ea7eda9899e1401402901848f221366c99/src/sfTrashShift/sfTrashShift.dsp | faust | -------------------- MAIN -------------------------------
--------------------------------------------------------------------------------------------------
from FAUST example and adapted by Christophe Lebreton
very simple real time pitch shifter
w = hslider("window [units (ms)]", 75, 10, 1000, 1)*SR*0.001;
x = hslider("xfade [units (ms)]", 10, 1, 500, 1)*SR*0.001 : smooth (0.99); | declare name "sfTrashShift";
declare version "1.1";
declare author "Christophe Lebreton";
declare license "BSD";
declare copyright "SmartFaust - GRAME(c)2013-2018";
import("stdfaust.lib");
process = pitchshifter_drywet;
transpose (w, x, s, sig) = de.fdelay1s(d,sig)* ma.fmin(d/x,1) + de.fdelay1s(d+w,sig)*(1- ma.fmin(d/x,1))
with {
i = 1 - pow(2, s/12);
d = i : (+ : +(w) : fmod(_,w)) ~ _;
};
pitchshifter = transpose(w,x,s)
with {
w = (75)* ma.SR*(0.001);
x = w * 0.5;
s = (hslider("v:sfTrashShift parameter(s)/shift [units (cents)] [acc:0 1 -10 0 10][color: 255 0 0 ][hidden:1]", 0, -3600, 3600, 0.1))*0.01 : si.smooth (0.998);
};
dry_wet(x,y) = (1-c)*x + c*y
with {
c = hslider("v:sfTrashShift parameter(s)/dry_wet [acc:1 1 -10 0 10][color: 255 255 0 ][hidden:1] ",100,0,100,0.01):*(0.01):fi.lowpass(1,1):max(0):min(1);
};
pitchshifter_drywet = _<: _ , pitchshifter: dry_wet:*(volume):*(gain):*(out)
with {
volume = vslider ("h:sfTrashShift/Volume",1,0,2,0.001): si.smooth(0.998):max(0):min(2);
gain = hslider ("v:sfTrashShift parameter(s)/gain[acc:2 1 -10 0 10][color:255 255 0][hidden:1]",0.2,0,1,0.001): fi.lowpass(1,1):max(0):min(1);
out = checkbox ("h:sfTrashShift/ON/OFF"): si.smooth(0.998);
};
|
869adf6f42d10499485b3d2a9bc91193f048f771aaa4a5b8d71e68efa87007f6 | s-e-a-m/faust-libraries | cardiosc_plot.dsp | declare name "MID SIDE PANNER - LEFT RIGHT LOUDSPEAKER";
declare version "001";
declare author "Giuseppe Silvi";
declare license "GNU-GPL-v3";
declare copyright "(c)SEAM 2019";
declare description "MID SIDE PANNER - LEFT RIGHT LOUDSPEAKER";
import("stdfaust.lib");
import("../../seam.lib");
pisweep = (os.lf_trianglepos(1)*360)-180;
rad = 45 : deg2rad;
//process = os.osc(50), 0 : mspan;
process = os.osc(50) <: *(sin(rad)), *(cos(rad));
| https://raw.githubusercontent.com/s-e-a-m/faust-libraries/9120cccb9335f42407062eb4bf149188d8018b07/plots/dsp/cardiosc_plot.dsp | faust | process = os.osc(50), 0 : mspan; | declare name "MID SIDE PANNER - LEFT RIGHT LOUDSPEAKER";
declare version "001";
declare author "Giuseppe Silvi";
declare license "GNU-GPL-v3";
declare copyright "(c)SEAM 2019";
declare description "MID SIDE PANNER - LEFT RIGHT LOUDSPEAKER";
import("stdfaust.lib");
import("../../seam.lib");
pisweep = (os.lf_trianglepos(1)*360)-180;
rad = 45 : deg2rad;
process = os.osc(50) <: *(sin(rad)), *(cos(rad));
|
521b151eca25ee9c21639e1b8242c431bdf2b039af03496e0798eea895c12059 | afalaize/faust | vumeter.dsp | declare name "vumeter";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
//-------------------------------------------------
// Simple vumeter
//-------------------------------------------------
import("stdfaust.lib");
vmeter(x) = attach(x, envelop(x) : vbargraph("[2][unit:dB]", -70, +5));
hmeter(x) = attach(x, envelop(x) : hbargraph("[2][unit:dB]", -70, +5));
envelop = abs : max ~ -(1.0/ma.SR) : max(ba.db2linear(-70)) : ba.linear2db;
process = hmeter,hmeter;
| https://raw.githubusercontent.com/afalaize/faust/8f9f5fe3aa167eaeecc15a99d4da984ac2797be3/examples/analysis/vumeter.dsp | faust | -------------------------------------------------
Simple vumeter
------------------------------------------------- | declare name "vumeter";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
import("stdfaust.lib");
vmeter(x) = attach(x, envelop(x) : vbargraph("[2][unit:dB]", -70, +5));
hmeter(x) = attach(x, envelop(x) : hbargraph("[2][unit:dB]", -70, +5));
envelop = abs : max ~ -(1.0/ma.SR) : max(ba.db2linear(-70)) : ba.linear2db;
process = hmeter,hmeter;
|
15cb7f1e45d806d9ae21bbb1705192edb4d4b0600a1fe7e816a6571787c8dc7a | afalaize/faust | osci.dsp | declare name "osci";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2009";
//-----------------------------------------------
// Sinusoidal Oscillator
// (with linear interpolation)
//-----------------------------------------------
import("stdfaust.lib");
vol = hslider("volume [unit:dB]", 0, -96, 0, 0.1) : ba.db2linear : si.smoo ;
freq = hslider("freq [unit:Hz]", 1000, 20, 24000, 1);
process = vgroup("Oscillator", os.osci(freq) * vol);
| https://raw.githubusercontent.com/afalaize/faust/8f9f5fe3aa167eaeecc15a99d4da984ac2797be3/examples/generator/osci.dsp | faust | -----------------------------------------------
Sinusoidal Oscillator
(with linear interpolation)
----------------------------------------------- | declare name "osci";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2009";
import("stdfaust.lib");
vol = hslider("volume [unit:dB]", 0, -96, 0, 0.1) : ba.db2linear : si.smoo ;
freq = hslider("freq [unit:Hz]", 1000, 20, 24000, 1);
process = vgroup("Oscillator", os.osci(freq) * vol);
|
40d7dcf24b805069728dd0d049b863f0f5f9ba06f59372d8898688ab579a2475 | SMERM/BN-Tedesco | spanner_1x8.dsp | declare filename "spanner_1x8.dsp"; declare name "spanner_1x8"; declare name "spanner_1x8";
declare version "1.0";
declare author "THC-SCALAS";
declare license "BSD";
declare copyright "Cecilia-labs";
//==========================================================
//
// spanner_1x8
// traspose panner_1x8
//
//==========================================================
import("stdfaust.lib");
nch = 8; //NUMERO DI CANALI IN USCITA
angstep = 360.0/nch; //DISTANZA ANGOLARE INTERSPEAKER IN GRADI
ang = nentry("Angle[style:knob]",360,0,360,0.1) : si.smooth(ba.tau2pole(0.02)); //DIREZIONE ANGOLARE DELLA SORGENTE IN GRADI
sep = hslider("Separation", 12, 0, 100, 0.1) : si.smooth(ba.tau2pole(0.02)); //SEPARAZIONE INTERSPEAKER DELLA SORGENTE
angr = ang*ma.PI/180; //ANGOLO IN RADIANTI
angstepr= angstep*ma.PI/180; //DISTANZA ANGOLARE INTERSPEAKER IN RADIANTI
//*********** CALCOLA I SEGNALE PER TUTTELE 8 USCITE ***********************//
process(sig) = sig*(pow(10, ((0.5*cos(angstepr*1-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*2-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*3-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*4-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*5-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*6-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*7-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*8-angr)-0.5)*sep)*0.2));
| https://raw.githubusercontent.com/SMERM/BN-Tedesco/2a77e1707f7e64c512dd40d58d29c0db8092463d/COME-04/20200507/spanner_1x8.dsp | faust | ==========================================================
spanner_1x8
traspose panner_1x8
==========================================================
NUMERO DI CANALI IN USCITA
DISTANZA ANGOLARE INTERSPEAKER IN GRADI
DIREZIONE ANGOLARE DELLA SORGENTE IN GRADI
SEPARAZIONE INTERSPEAKER DELLA SORGENTE
ANGOLO IN RADIANTI
DISTANZA ANGOLARE INTERSPEAKER IN RADIANTI
*********** CALCOLA I SEGNALE PER TUTTELE 8 USCITE ***********************//
| declare filename "spanner_1x8.dsp"; declare name "spanner_1x8"; declare name "spanner_1x8";
declare version "1.0";
declare author "THC-SCALAS";
declare license "BSD";
declare copyright "Cecilia-labs";
import("stdfaust.lib");
process(sig) = sig*(pow(10, ((0.5*cos(angstepr*1-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*2-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*3-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*4-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*5-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*6-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*7-angr)-0.5)*sep)*0.2)),
sig*(pow(10, ((0.5*cos(angstepr*8-angr)-0.5)*sep)*0.2));
|
604c9deae2f2134db6278577ad9c8795460dd8e9ea64893a607ffd3a14922aeb | inria-emeraude/syfala | lmsN.dsp | declare name "Least Mean Square Algorithm";
declare version "1.0";
declare author "Pierre Lecomte";
declare author "Loic Alexandre";
declare license "CC-BY-NC-SA-4.0";
import("stdfaust.lib");
N = 500;
coeffs = si.bus(N);
h = fi.fir((1,2,3,4,5)); // target filter
y = _:h; // Output signal from target system
h_hat(N) = (si.bus(N),(_<:(si.bus(N)))):ro.interleave(N,2):sum(i, N, (_,@(i):*)); // Adapted filter
y_hat(N) = ((si.bus(N)<:si.bus(2*N)),_):(si.bus(N),h_hat(N)); // Output signal from adapted system
buffer = _<:par(i,N,@(i)); // To obtain x_n, the reference signal at time n
signal = no.noise; // Reference signal
x = (signal:_<:(_,_,_));
mu = -0.0001; // Convergence coefficient (smaller for slower convergence)
// No input
// Output 0 = error signal (y - y_hat)
process = ((coeffs,x):(y_hat(N),y,_):(coeffs,(-<:(_,_*mu)),buffer):(coeffs,_,(_<:si.bus(N)),coeffs):(coeffs,_,ro.interleave(N,2)):(coeffs,_,par(i,N,*)):(coeffs,ro.cross1n(N)):(ro.interleave(N,2),_):(par(i,N,+),_))~si.bus(N):(par(i,N,!),_);
| https://raw.githubusercontent.com/inria-emeraude/syfala/95ed6765d73520362f6a1ad35e4a3b2a5e16fbc9/tools/multiN/dsp/lmsN.dsp | faust | target filter
Output signal from target system
Adapted filter
Output signal from adapted system
To obtain x_n, the reference signal at time n
Reference signal
Convergence coefficient (smaller for slower convergence)
No input
Output 0 = error signal (y - y_hat) | declare name "Least Mean Square Algorithm";
declare version "1.0";
declare author "Pierre Lecomte";
declare author "Loic Alexandre";
declare license "CC-BY-NC-SA-4.0";
import("stdfaust.lib");
N = 500;
coeffs = si.bus(N);
x = (signal:_<:(_,_,_));
process = ((coeffs,x):(y_hat(N),y,_):(coeffs,(-<:(_,_*mu)),buffer):(coeffs,_,(_<:si.bus(N)),coeffs):(coeffs,_,ro.interleave(N,2)):(coeffs,_,par(i,N,*)):(coeffs,ro.cross1n(N)):(ro.interleave(N,2),_):(par(i,N,+),_))~si.bus(N):(par(i,N,!),_);
|
f9b856f7070103ce40ba126f3bcda34c08d07782063513978711c0a967632c8c | inria-emeraude/syfala | lms_standalone.dsp | declare name "Least Mean Square Algorithm";
declare version "1.0";
declare author "Pierre Lecomte";
declare author "Loic Alexandre";
declare license "CC-BY-NC-SA-4.0";
import("stdfaust.lib");
N = 30; // Number of coefficients
coeffs = si.bus(N);
h = fi.fir((1,2,3,4,5)); // target filter
y = _:h; // Output signal from target system
h_hat(N) = (si.bus(N),(_<:(si.bus(N)))):ro.interleave(N,2):sum(i, N, (_,@(i):*)); // Adapted filter
y_hat(N) = ((si.bus(N)<:si.bus(2*N)),_):(si.bus(N),h_hat(N)); // Output signal from adapted system
buffer = _<:par(i,N,@(i)); // To obtain x_n, the reference signal at time n
signal = no.noise; // Reference signal
x = (signal:_<:(_,_,_));
mu = -0.0001; // Convergence coefficient (smaller for slower convergence)
// No input
// Output 0 = error signal (y - y_hat)
process = ((coeffs,x):(y_hat(N),y,_):(coeffs,(-<:(_,_*mu)),buffer):(coeffs,_,(_<:si.bus(N)),coeffs):(coeffs,_,ro.interleave(N,2)):(coeffs,_,par(i,N,*)):(coeffs,ro.cross1n(N)):(ro.interleave(N,2),_):(par(i,N,+),_))~si.bus(N):(par(i,N,!),_);
| https://raw.githubusercontent.com/inria-emeraude/syfala/95ed6765d73520362f6a1ad35e4a3b2a5e16fbc9/examples/lms_standalone.dsp | faust | Number of coefficients
target filter
Output signal from target system
Adapted filter
Output signal from adapted system
To obtain x_n, the reference signal at time n
Reference signal
Convergence coefficient (smaller for slower convergence)
No input
Output 0 = error signal (y - y_hat) | declare name "Least Mean Square Algorithm";
declare version "1.0";
declare author "Pierre Lecomte";
declare author "Loic Alexandre";
declare license "CC-BY-NC-SA-4.0";
import("stdfaust.lib");
coeffs = si.bus(N);
x = (signal:_<:(_,_,_));
process = ((coeffs,x):(y_hat(N),y,_):(coeffs,(-<:(_,_*mu)),buffer):(coeffs,_,(_<:si.bus(N)),coeffs):(coeffs,_,ro.interleave(N,2)):(coeffs,_,par(i,N,*)):(coeffs,ro.cross1n(N)):(ro.interleave(N,2),_):(par(i,N,+),_))~si.bus(N):(par(i,N,!),_);
|
5aba287c9f6e8ee48d46141310c85695da4256b04e4880d70af67333fd620a25 | rottingsounds/bitDSP-faust | DarioGen3.dsp | declare name "DarioGen3";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
// bit = library("bitDSP.lib");
bit_gen = library("bitDSP_gen.lib");
// SuperCollider
// export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
// faust2supercollider -I ../../lib -noprefix DarioGen3.dsp
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
// Final output
process = bit_gen.gen1(c1, c2) : si.bus(2);
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/d70843492b65bb2cb9cf97c2240905fefacc7383/synths/_sc/DarioGen3.dsp | faust | bit = library("bitDSP.lib");
SuperCollider
export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
faust2supercollider -I ../../lib -noprefix DarioGen3.dsp
Final output | declare name "DarioGen3";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit_gen = library("bitDSP_gen.lib");
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
process = bit_gen.gen1(c1, c2) : si.bus(2);
|
55e9120ad5d2d5ff2ccdf36be2fd08c6358af88971b928a05f68709ea2cc2b23 | rottingsounds/bitDSP-faust | DarioGen2.dsp | declare name "DarioGen2";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
// bit = library("bitDSP.lib");
bit_gen = library("bitDSP_gen.lib");
// SuperCollider
// export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
// faust2supercollider -I ../../lib -noprefix DarioGen2.dsp
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
// Final output
process = bit_gen.gen2(c1, c2) : si.bus(2);
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/d70843492b65bb2cb9cf97c2240905fefacc7383/synths/_sc/DarioGen2.dsp | faust | bit = library("bitDSP.lib");
SuperCollider
export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
faust2supercollider -I ../../lib -noprefix DarioGen2.dsp
Final output | declare name "DarioGen2";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit_gen = library("bitDSP_gen.lib");
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
process = bit_gen.gen2(c1, c2) : si.bus(2);
|
236896dceac16a85b502b2997e1b60a2fc8866fd018beaf2d7abb5b6db8e2f02 | rottingsounds/bitDSP-faust | DarioGen1.dsp | declare name "DarioGen1";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
// bit = library("bitDSP.lib");
bit_gen = library("bitDSP_gen.lib");
// SuperCollider
// export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
// faust2supercollider -I ../../lib -noprefix DarioGen1.dsp
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
// Final output
process = bit_gen.gen1(c1, c2) : si.bus(2);
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/d70843492b65bb2cb9cf97c2240905fefacc7383/synths/_sc/DarioGen1.dsp | faust | bit = library("bitDSP.lib");
SuperCollider
export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
faust2supercollider -I ../../lib -noprefix DarioGen1.dsp
Final output | declare name "DarioGen1";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit_gen = library("bitDSP_gen.lib");
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
process = bit_gen.gen1(c1, c2) : si.bus(2);
|
457a0959dd4613a24811dbac841c615658e67031b05c86a71d272ceef3d09774 | rottingsounds/bitDSP-faust | DarioGen4.dsp | declare name "DarioGen4";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
// bit = library("bitDSP.lib");
bit_gen = library("bitDSP_gen.lib");
// SuperCollider
// export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
// faust2supercollider -I ../../lib -noprefix DarioGen4.dsp
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
// Final output
process = bit_gen.gen4(c1, c2) : si.bus(2);
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/d70843492b65bb2cb9cf97c2240905fefacc7383/synths/_sc/DarioGen4.dsp | faust | bit = library("bitDSP.lib");
SuperCollider
export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
faust2supercollider -I ../../lib -noprefix DarioGen4.dsp
Final output | declare name "DarioGen4";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit_gen = library("bitDSP_gen.lib");
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
process = bit_gen.gen4(c1, c2) : si.bus(2);
|
d3a4cb12e796230e0ba92ebb4991e9537144d1fb7487fe28b3290fa428dc9148 | rottingsounds/bitDSP-faust | Higks.dsp | declare name "Higks";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
// bit = library("bitDSP.lib");
bit_gen = library("bitDSP_gen.lib");
// SuperCollider
// export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
// faust2supercollider -I ../../lib -noprefix Higks.dsp
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
// Final output
process = bit_gen.higks(c1, c2) : si.bus(2);
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/d70843492b65bb2cb9cf97c2240905fefacc7383/synths/_sc/Higks.dsp | faust | bit = library("bitDSP.lib");
SuperCollider
export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
faust2supercollider -I ../../lib -noprefix Higks.dsp
Final output | declare name "Higks";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit_gen = library("bitDSP_gen.lib");
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
process = bit_gen.higks(c1, c2) : si.bus(2);
|
4e9aa99e354f6aac204c509783e4783bcf9a7aeb004da79c23adc508deedcc6c | rottingsounds/bitDSP-faust | Bfb.dsp | declare name "Bfb";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
// bit = library("bitDSP.lib");
bit_gen = library("bitDSP_gen.lib");
// SuperCollider
// export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
// faust2supercollider -I ../../lib -noprefix Bfb.dsp
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
// Final output
process = bit_gen.bfb(c1, c2) : si.bus(2);
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/d70843492b65bb2cb9cf97c2240905fefacc7383/synths/_sc/Bfb.dsp | faust | bit = library("bitDSP.lib");
SuperCollider
export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
faust2supercollider -I ../../lib -noprefix Bfb.dsp
Final output | declare name "Bfb";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit_gen = library("bitDSP_gen.lib");
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
process = bit_gen.bfb(c1, c2) : si.bus(2);
|
7d0546ab51bfea4b8966a59611df3e45578c17622e369c58481e68829d57d19c | rottingsounds/bitDSP-faust | Trck.dsp | declare name "Trck";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
// bit = library("bitDSP.lib");
bit_gen = library("bitDSP_gen.lib");
// SuperCollider
// export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
// faust2supercollider -I ../../lib -noprefix Trck.dsp
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
// Final output
process = bit_gen.trck(c1, c2) : si.bus(2);
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/d70843492b65bb2cb9cf97c2240905fefacc7383/synths/_sc/Trck.dsp | faust | bit = library("bitDSP.lib");
SuperCollider
export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
faust2supercollider -I ../../lib -noprefix Trck.dsp
Final output | declare name "Trck";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit_gen = library("bitDSP_gen.lib");
c1 = hslider("c1",0,0,1,0.001);
c2 = hslider("c2",0.5,0,1,0.001);
process = bit_gen.trck(c1, c2) : si.bus(2);
|
492636850a58f92019579481e92865ef2f396732ef00b901e241c54f24f890c5 | Frando/studiox-switcher | switcher.dsp | declare name "studiox-switcher";
declare version "1.0";
declare author "Franz Heinzmann";
declare license "BSD";
declare options "[osc:on]";
import("stdfaust.lib");
merge2 = _,_: ba.parallelMean(2);
// helpers to build a VU meter
envelop = abs : max ~ -(1.0/ma.SR) : max(ba.db2linear(-70)) : ba.linear2db;
vumeterM(x) = envelop(x) : vbargraph("level[2][unit:dB][style:dB]", -60, +5);
vumeterS(a,b) = a,b <: _,_,_,_ :
(a, b, attach(0,vumeterM((a+b)/2)), 0) :>
_,_;
vumeter = _,_ : vumeterS(_,_);
vumeterI(i) = _,_ : vgroup("level/%i", vumeter) : _,_;
silenceDetect(
analysisWin,
dBSilenceTh,
timeSilenceTh,
xInput
) =
ba.linear2db(
an.rms_envelope_t19(
analysisWin,
xInput
)
)
< dBSilenceTh <: fi.pole > (timeSilenceTh * ma.SR);
stereoSilenceFallback(
analysisWin,
dBSilenceTh,
timeSilenceTh,
mainActive,
xMainL, xMainR, xBackupL, xBackupR
) =
ba.select2stereo(cond, xMainL, xMainR, xBackupL, xBackupR)
with {
cond =
ba.if(
mainActive,
silenceDetect(
analysisWin,
dBSilenceTh,
timeSilenceTh,
merge2(xMainL, xMainR)
),
1.0
);
};
applySilenceFallback(xBackupL, xBackupR, xMainL, xMainR) =
stereoSilenceFallback(
.01,
vslider("threshold[style:knob][unit:dB]", -60, -70, 0, 0.1),
vslider("timeout[style:knob]", 1.0, 0.1, 60.0, 0.1),
1.0,
xMainL, xMainR,
xBackupL, xBackupR
);
switcherN(N, xBackupL, xBackupR) =
par(n, N, _,_) : hgroup("active", selector(N))
with {
selector(1) = ba.select2stereo(
checkbox("1"),
xBackupL,
xBackupR,
_,_
);
selector(n) = ba.select2stereo(
checkbox("%n"),
selector(n-1),
_,_
);
};
fallbackSwitcherN(N, xBackupL, xBackupR) =
switcherN(N, xBackupL, xBackupR) : _,_ : applySilenceFallback(xBackupL, xBackupR);
inputMeters(N) = hgroup("input", par(n, N, vgroup("%n", vumeter)));
N = 3;
process = par(n, N + 1, _,_) : inputMeters(N + 1) : fallbackSwitcherN(N) : vumeter : _,_;
| https://raw.githubusercontent.com/Frando/studiox-switcher/84ce1d192c86c11aaf89edb5e097b29555f632f4/dsp/switcher.dsp | faust | helpers to build a VU meter | declare name "studiox-switcher";
declare version "1.0";
declare author "Franz Heinzmann";
declare license "BSD";
declare options "[osc:on]";
import("stdfaust.lib");
merge2 = _,_: ba.parallelMean(2);
envelop = abs : max ~ -(1.0/ma.SR) : max(ba.db2linear(-70)) : ba.linear2db;
vumeterM(x) = envelop(x) : vbargraph("level[2][unit:dB][style:dB]", -60, +5);
vumeterS(a,b) = a,b <: _,_,_,_ :
(a, b, attach(0,vumeterM((a+b)/2)), 0) :>
_,_;
vumeter = _,_ : vumeterS(_,_);
vumeterI(i) = _,_ : vgroup("level/%i", vumeter) : _,_;
silenceDetect(
analysisWin,
dBSilenceTh,
timeSilenceTh,
xInput
) =
ba.linear2db(
an.rms_envelope_t19(
analysisWin,
xInput
)
)
< dBSilenceTh <: fi.pole > (timeSilenceTh * ma.SR);
stereoSilenceFallback(
analysisWin,
dBSilenceTh,
timeSilenceTh,
mainActive,
xMainL, xMainR, xBackupL, xBackupR
) =
ba.select2stereo(cond, xMainL, xMainR, xBackupL, xBackupR)
with {
cond =
ba.if(
mainActive,
silenceDetect(
analysisWin,
dBSilenceTh,
timeSilenceTh,
merge2(xMainL, xMainR)
),
1.0
);
};
applySilenceFallback(xBackupL, xBackupR, xMainL, xMainR) =
stereoSilenceFallback(
.01,
vslider("threshold[style:knob][unit:dB]", -60, -70, 0, 0.1),
vslider("timeout[style:knob]", 1.0, 0.1, 60.0, 0.1),
1.0,
xMainL, xMainR,
xBackupL, xBackupR
);
switcherN(N, xBackupL, xBackupR) =
par(n, N, _,_) : hgroup("active", selector(N))
with {
selector(1) = ba.select2stereo(
checkbox("1"),
xBackupL,
xBackupR,
_,_
);
selector(n) = ba.select2stereo(
checkbox("%n"),
selector(n-1),
_,_
);
};
fallbackSwitcherN(N, xBackupL, xBackupR) =
switcherN(N, xBackupL, xBackupR) : _,_ : applySilenceFallback(xBackupL, xBackupR);
inputMeters(N) = hgroup("input", par(n, N, vgroup("%n", vumeter)));
N = 3;
process = par(n, N + 1, _,_) : inputMeters(N + 1) : fallbackSwitcherN(N) : vumeter : _,_;
|
30b48aef23af25448a628b3c24e5775f7dbc0e6badaad3794d88f05e09bd3ecb | magnetophon/MBdistortion | MBdistortion.dsp | declare name "MBdistortion";
declare author "Bart Brouns ([email protected]";
declare copyright "Bart Brouns";
declare version "1.1.1";
declare license "GPLv2";
import("stdfaust.lib");
MS = (checkbox("[0] mid/side
[tooltip: When this is checked, process mid-side, otherwise process stereo]"));
freq_group(x) = (hgroup("[1] crossover frequencies", x));
fc1 = freq_group(vslider("[1] L/LM [unit:Hz] [style:knob]
[tooltip: Crossover frequency (Hz) separating low and low middle frequencies]",
120 , 20, 1000, 1)):si.smooth(0.999);
fc2 = freq_group(vslider("[2] LM/HM [unit:Hz] [style:knob]
[tooltip: Crossover frequency (Hz) separating low middle and high middle frequencies]",
1000, 100, 3000, 1)):si.smooth(0.999);
fc3 = freq_group(vslider("[3] HM/H [unit:Hz] [style:knob]
[tooltip: Crossover frequency (Hz) separating high middle and high frequencies]",
6000, 500, 10000, 1)):si.smooth(0.999);
bands_group(x) = (hgroup("[2] frequency bands", x));
low_group(x) = bands_group(hgroup("[1] low", x));
lowmid_group(x) = bands_group(hgroup("[2] low mid", x));
himid_group(x) = bands_group(hgroup("[3] high mid", x));
high_group(x) = bands_group(hgroup("[4] high", x));
drive1A = low_group(vslider("[1] Drive A
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive1B = low_group(vslider("[2] Drive B
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset1A = low_group(vslider("[3] Offset A
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset1B = low_group(vslider("[4] Offset B
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive2A = lowmid_group(vslider("[1] Drive A
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive2B = lowmid_group(vslider("[2] Drive B
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset2A = lowmid_group(vslider("[3] Offset A
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset2B = lowmid_group(vslider("[4] Offset B
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive3A = himid_group(vslider("[1] Drive A
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive3B = himid_group(vslider("[2] Drive B
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset3A = himid_group(vslider("[3] Offset A
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset3B = himid_group(vslider("[4] Offset B
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive4A = high_group(vslider("[1] Drive A
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive4B = high_group(vslider("[2] Drive B
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset4A = high_group(vslider("[3] Offset A
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset4B = high_group(vslider("[4] Offset B
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
outgain_group(x) = (vgroup("[3] output gain", x));
drygain = outgain_group(hslider("[1] Dry gain
[unit:dB][tooltip: ]",
-144, -144, 0, 0.01)):ba.db2linear:si.smooth(0.999);
wetgain = outgain_group(hslider("[2] Wet gain
[unit:dB][tooltip: ]",
0, -144, 0, 0.01)):ba.db2linear:si.smooth(0.999);
MBdist( drive1,offset1, drive2,offset2, drive3,offset3, drive4,offset4)=
fi.filterbank(3,(fc1,fc2,fc3)):
(
ef.cubicnl(drive4,offset4),
ef.cubicnl(drive3,offset3),
ef.cubicnl(drive2,offset2),
ef.cubicnl(drive1,offset1)
) :>fi.dcblocker;
stereo2MS(MS, x,y) = (x+(MS*y)), ((MS*x) + ((MS*-2)+1)*y);
MS2stereo(MS, m,s) = ((m+(MS*s))/(MS+1)), (((MS*m) + ((MS*-2)+1)*s)/(MS+1));
MSMBdist(x,y) =
(
stereo2MS(MS, x,y) :
(MBdist( drive1A,offset1A, drive2A,offset2A, drive3A,offset3A, drive4A,offset4A)*wetgain,
MBdist( drive1B,offset1B, drive2B,offset2B, drive3B,offset3B, drive4B,offset4B)*wetgain) :
MS2stereo(MS)
)
,
((x:fi.filterbank(3,(fc1,fc2,fc3)):>_)*drygain,(y:fi.filterbank(3,(fc1,fc2,fc3)):>_)*drygain)
:>(_,_);
process(x,y) = MSMBdist(x,y);
| https://raw.githubusercontent.com/magnetophon/MBdistortion/612e511d7f6f09877f8bb804faacfb90a01afe9c/MBdistortion.dsp | faust | declare name "MBdistortion";
declare author "Bart Brouns ([email protected]";
declare copyright "Bart Brouns";
declare version "1.1.1";
declare license "GPLv2";
import("stdfaust.lib");
MS = (checkbox("[0] mid/side
[tooltip: When this is checked, process mid-side, otherwise process stereo]"));
freq_group(x) = (hgroup("[1] crossover frequencies", x));
fc1 = freq_group(vslider("[1] L/LM [unit:Hz] [style:knob]
[tooltip: Crossover frequency (Hz) separating low and low middle frequencies]",
120 , 20, 1000, 1)):si.smooth(0.999);
fc2 = freq_group(vslider("[2] LM/HM [unit:Hz] [style:knob]
[tooltip: Crossover frequency (Hz) separating low middle and high middle frequencies]",
1000, 100, 3000, 1)):si.smooth(0.999);
fc3 = freq_group(vslider("[3] HM/H [unit:Hz] [style:knob]
[tooltip: Crossover frequency (Hz) separating high middle and high frequencies]",
6000, 500, 10000, 1)):si.smooth(0.999);
bands_group(x) = (hgroup("[2] frequency bands", x));
low_group(x) = bands_group(hgroup("[1] low", x));
lowmid_group(x) = bands_group(hgroup("[2] low mid", x));
himid_group(x) = bands_group(hgroup("[3] high mid", x));
high_group(x) = bands_group(hgroup("[4] high", x));
drive1A = low_group(vslider("[1] Drive A
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive1B = low_group(vslider("[2] Drive B
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset1A = low_group(vslider("[3] Offset A
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset1B = low_group(vslider("[4] Offset B
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive2A = lowmid_group(vslider("[1] Drive A
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive2B = lowmid_group(vslider("[2] Drive B
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset2A = lowmid_group(vslider("[3] Offset A
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset2B = lowmid_group(vslider("[4] Offset B
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive3A = himid_group(vslider("[1] Drive A
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive3B = himid_group(vslider("[2] Drive B
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset3A = himid_group(vslider("[3] Offset A
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset3B = himid_group(vslider("[4] Offset B
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive4A = high_group(vslider("[1] Drive A
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
drive4B = high_group(vslider("[2] Drive B
[tooltip: Amount of distortion]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset4A = high_group(vslider("[3] Offset A
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
offset4B = high_group(vslider("[4] Offset B
[tooltip: Brings in even harmonics]",
0, 0, 1, 0.01)):si.smooth(0.999);
outgain_group(x) = (vgroup("[3] output gain", x));
drygain = outgain_group(hslider("[1] Dry gain
[unit:dB][tooltip: ]",
-144, -144, 0, 0.01)):ba.db2linear:si.smooth(0.999);
wetgain = outgain_group(hslider("[2] Wet gain
[unit:dB][tooltip: ]",
0, -144, 0, 0.01)):ba.db2linear:si.smooth(0.999);
MBdist( drive1,offset1, drive2,offset2, drive3,offset3, drive4,offset4)=
fi.filterbank(3,(fc1,fc2,fc3)):
(
ef.cubicnl(drive4,offset4),
ef.cubicnl(drive3,offset3),
ef.cubicnl(drive2,offset2),
ef.cubicnl(drive1,offset1)
) :>fi.dcblocker;
stereo2MS(MS, x,y) = (x+(MS*y)), ((MS*x) + ((MS*-2)+1)*y);
MS2stereo(MS, m,s) = ((m+(MS*s))/(MS+1)), (((MS*m) + ((MS*-2)+1)*s)/(MS+1));
MSMBdist(x,y) =
(
stereo2MS(MS, x,y) :
(MBdist( drive1A,offset1A, drive2A,offset2A, drive3A,offset3A, drive4A,offset4A)*wetgain,
MBdist( drive1B,offset1B, drive2B,offset2B, drive3B,offset3B, drive4B,offset4B)*wetgain) :
MS2stereo(MS)
)
,
((x:fi.filterbank(3,(fc1,fc2,fc3)):>_)*drygain,(y:fi.filterbank(3,(fc1,fc2,fc3)):>_)*drygain)
:>(_,_);
process(x,y) = MSMBdist(x,y);
|
|
fc6cc0319086b0544f131cdea95968d74ca13ac2a955f8a05687a1603326df7f | s-e-a-m/faust-libraries | m2bfmt.dsp | declare name "MICHAEL GERZON MONO TO BFORMAT ENCODER";
declare version "001";
declare author "Giuseppe Silvi";
declare license "GNU-GPL-v3";
declare copyright "(c)SEAM 2019";
declare description "MICHAEL GERZON MONO TO BFORMAT ENCODER";
import("stdfaust.lib");
import("../../seam.lib");
// LS and RS are dead channels to create VST routing consistency
m2bfmt(L,R,LS,RS) = W,X,Y,Z
with{
encoder(x) = hgroup("BFMT ENCODER", x);
azi = encoder(vslider("[01] Azimuth [style:knob]", 0, 0, 360, 0.1) : deg2rad : si.smoo);
elv = encoder(vslider("[01] Elevation [style:knob]", 0, 0, 360, 0.1) : deg2rad : si.smoo);
W = L * 0.707;
X = L * cos(azi) * cos(elv);
Y = L * sin(azi) * cos(elv);
Z = L * sin(elv);
};
process = m2bfmt;
| https://raw.githubusercontent.com/s-e-a-m/faust-libraries/9120cccb9335f42407062eb4bf149188d8018b07/examples/vst/m2bfmt.dsp | faust | LS and RS are dead channels to create VST routing consistency | declare name "MICHAEL GERZON MONO TO BFORMAT ENCODER";
declare version "001";
declare author "Giuseppe Silvi";
declare license "GNU-GPL-v3";
declare copyright "(c)SEAM 2019";
declare description "MICHAEL GERZON MONO TO BFORMAT ENCODER";
import("stdfaust.lib");
import("../../seam.lib");
m2bfmt(L,R,LS,RS) = W,X,Y,Z
with{
encoder(x) = hgroup("BFMT ENCODER", x);
azi = encoder(vslider("[01] Azimuth [style:knob]", 0, 0, 360, 0.1) : deg2rad : si.smoo);
elv = encoder(vslider("[01] Elevation [style:knob]", 0, 0, 360, 0.1) : deg2rad : si.smoo);
W = L * 0.707;
X = L * cos(azi) * cos(elv);
Y = L * sin(azi) * cos(elv);
Z = L * sin(elv);
};
process = m2bfmt;
|
6dc8f93ac874a79b85bcfae675fe9b3e87820ba89a3e4446e0fe64b18ef83f3e | madskjeldgaard/komet | plat.dsp | declare name "Plat";
declare author "Mads Kjeldgaard";
declare copyright "Mads Kjeldgaard";
declare version "1.00";
declare license "GPL";
import("stdfaust.lib");
import("lib/mkdelay.dsp");
// Static
order = 4;
numDelays = 8;
maxdelay = 0.1 * ma.SR;
// Controls
delay = vslider("delaytime",0.1,0.001,2.0,0.01) : *(ma.SR) : si.smoo;
fb = vslider("fb",0.1,0.001,2.0,0.01);
lpf = vslider("cutoff",3500,20.0,20000.0,1);
delayOffset=vslider("delayoffset",0.5,0.0,1.0,0.00001) : si.smoo;
modFreq=vslider("modFreq",0.05,0.0,1.0,0.00001) : si.smoo;
modDepth=vslider("modDepth",0.05,0.0,1.0,0.00001) : si.smoo;
// apFb = vslider("apFb",0.25,0.0,1.0,0.00001) : si.smoo;
// Process
process = _
<: mkd.parallel_comb_lpf(numDelays, order, maxdelay, delay, delayOffset, fb, lpf)
:> fi.allpass_fcomb(maxdelay,apdelay(modFreq, modDepth),fb) with{
apdelay(modFreq, modDepth) = os.lf_triangle(modFreq) + 1.0 : /(2.0)
: *(maxdelay)
: *(modDepth);
};
| https://raw.githubusercontent.com/madskjeldgaard/komet/defd9b0b2f4055dcb12b75565631a30152fa779c/faust/plat.dsp | faust | Static
Controls
apFb = vslider("apFb",0.25,0.0,1.0,0.00001) : si.smoo;
Process | declare name "Plat";
declare author "Mads Kjeldgaard";
declare copyright "Mads Kjeldgaard";
declare version "1.00";
declare license "GPL";
import("stdfaust.lib");
import("lib/mkdelay.dsp");
order = 4;
numDelays = 8;
maxdelay = 0.1 * ma.SR;
delay = vslider("delaytime",0.1,0.001,2.0,0.01) : *(ma.SR) : si.smoo;
fb = vslider("fb",0.1,0.001,2.0,0.01);
lpf = vslider("cutoff",3500,20.0,20000.0,1);
delayOffset=vslider("delayoffset",0.5,0.0,1.0,0.00001) : si.smoo;
modFreq=vslider("modFreq",0.05,0.0,1.0,0.00001) : si.smoo;
modDepth=vslider("modDepth",0.05,0.0,1.0,0.00001) : si.smoo;
process = _
<: mkd.parallel_comb_lpf(numDelays, order, maxdelay, delay, delayOffset, fb, lpf)
:> fi.allpass_fcomb(maxdelay,apdelay(modFreq, modDepth),fb) with{
apdelay(modFreq, modDepth) = os.lf_triangle(modFreq) + 1.0 : /(2.0)
: *(maxdelay)
: *(modDepth);
};
|
7711054442f63c5055f39bc66e365dd85e4d129e0444d47cc65058e4e7523538 | jameslnrd/mi_introduction_workshop_2020 | 2massChain.dsp | declare name "2-mass Chain";
declare author "James Leonard";
declare date "April 2020";
/* ========= DESCRITPION =============
A logical step from a simple oscillator: a chain of two masses connected by spring-dampers,
fixed at one end to a fixed point !
- inputs: force impulse on the last mass of the chain.
- outputs: position of the last mass of the chain.
- controls: none
*/
import("stdfaust.lib");
in1 = button("Frc Input 1"): ba.impulsify;
OutGain = 0.1;
K1 = 0.1;
Z1 = 0.0003;
K2 = 0.1;
Z2 = 0.0003;
model = (
mi.ground(0.),
mi.mass(1., 0, 0., 0.),
mi.mass(1., 0, 0., 0.),
par(i, nbFrcIn,_):
RoutingMassToLink ,
par(i, nbFrcIn,_):
mi.springDamper(K1, Z1, 0., 0.),
mi.springDamper(K2, Z2, 0., 0.),
par(i, nbOut+nbFrcIn, _):
RoutingLinkToMass
)~par(i, nbMass, _):
par(i, nbMass, !), par(i, nbOut , _)
with{
RoutingMassToLink(m0, m1, m2) = /* routed positions */ m0, m1, m1, m2, /* outputs */ m2;
RoutingLinkToMass(l0_f1, l0_f2, l1_f1, l1_f2, p_out1, f_in1) = /* routed forces */ l0_f1, l0_f2 + l1_f1, f_in1 + l1_f2, /* pass-through */ p_out1;
nbMass = 3;
nbFrcIn = 1;
nbOut = 1;
};
process = in1 : model:*(OutGain);
/*
========= MIMS SCRIPT USED FOR MODEL GENERATION =============
# MIMS script file
# Script author: James Leonard
@K1 param 0.1
@Z1 param 0.0003
@K2 param 0.1
@Z2 param 0.0003
@g ground 0.
@m1 mass 1. 0. 0.
@m2 mass 1. 0. 0.
@s1 springDamper @g @m1 K1 Z1
@s2 springDamper @m1 @m2 K2 Z2
# Add force input to the model
@in1 frcInput @m2
# Add position output from the oscillator
@out1 posOutput @m2
# end of MIMS script
*/ | https://raw.githubusercontent.com/jameslnrd/mi_introduction_workshop_2020/2f487dbc5b8e7cd83cbd962254e737bdb82948f6/08_TwoMassChain/2massChain.dsp | faust | ========= DESCRITPION =============
A logical step from a simple oscillator: a chain of two masses connected by spring-dampers,
fixed at one end to a fixed point !
- inputs: force impulse on the last mass of the chain.
- outputs: position of the last mass of the chain.
- controls: none
routed positions
outputs
routed forces
pass-through
========= MIMS SCRIPT USED FOR MODEL GENERATION =============
# MIMS script file
# Script author: James Leonard
@K1 param 0.1
@Z1 param 0.0003
@K2 param 0.1
@Z2 param 0.0003
@g ground 0.
@m1 mass 1. 0. 0.
@m2 mass 1. 0. 0.
@s1 springDamper @g @m1 K1 Z1
@s2 springDamper @m1 @m2 K2 Z2
# Add force input to the model
@in1 frcInput @m2
# Add position output from the oscillator
@out1 posOutput @m2
# end of MIMS script
| declare name "2-mass Chain";
declare author "James Leonard";
declare date "April 2020";
import("stdfaust.lib");
in1 = button("Frc Input 1"): ba.impulsify;
OutGain = 0.1;
K1 = 0.1;
Z1 = 0.0003;
K2 = 0.1;
Z2 = 0.0003;
model = (
mi.ground(0.),
mi.mass(1., 0, 0., 0.),
mi.mass(1., 0, 0., 0.),
par(i, nbFrcIn,_):
RoutingMassToLink ,
par(i, nbFrcIn,_):
mi.springDamper(K1, Z1, 0., 0.),
mi.springDamper(K2, Z2, 0., 0.),
par(i, nbOut+nbFrcIn, _):
RoutingLinkToMass
)~par(i, nbMass, _):
par(i, nbMass, !), par(i, nbOut , _)
with{
nbMass = 3;
nbFrcIn = 1;
nbOut = 1;
};
process = in1 : model:*(OutGain);
|
b2ad1e81b612236daa5d6acb5e09b644c1e43994df5bc0fb0efe9651692a2c23 | rottingsounds/bitDSP-faust | DSM2bipolar.dsp | declare name "DSM2bipolar";
declare author "Till Bovermann, Dario Sanfilippo";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
// plot
// CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib DSM2bipolar.dsp
// ./DSM2bipolar -n 10
// compile
// CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib DSM2bipolar.dsp
// ./DSM2bipolar
// SuperCollider
// export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
// faust2supercollider -I ../faust/bitDSP-faust/lib -noprefix DSM2bipolar.dsp
// Final output
// Bipolar multi-bit signal to bipolar one-bit signal
process = bit.dsm2;
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/21cf36105c55b6e18969a867a319530a0ef1ea63/examples/_sc/DSM2bipolar.dsp | faust | plot
CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib DSM2bipolar.dsp
./DSM2bipolar -n 10
compile
CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib DSM2bipolar.dsp
./DSM2bipolar
SuperCollider
export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
faust2supercollider -I ../faust/bitDSP-faust/lib -noprefix DSM2bipolar.dsp
Final output
Bipolar multi-bit signal to bipolar one-bit signal | declare name "DSM2bipolar";
declare author "Till Bovermann, Dario Sanfilippo";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
process = bit.dsm2;
|
a917ca78818353ea977a6da71c44edd52d952bb7bc63d9aa2f6f35b3617766f0 | rottingsounds/bitDSP-faust | DSMAdd.dsp | declare name "DSMAdd";
declare author "Till Bovermann, Dario Sanfilippo";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
// plot
// CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib DSMAdd.dsp
// ./DSM2bipolar -n 10
// compile
// CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib DSMAdd.dsp
// ./DSM2bipolar
// SuperCollider
// export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
// faust2supercollider -I ../faust/bitDSP-faust/lib -noprefix DSMAdd.dsp
// Final output
// Bipolar multi-bit signal to bipolar one-bit signal
process = bit.bitstream_adder;
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/21cf36105c55b6e18969a867a319530a0ef1ea63/examples/_sc/DSMAdd.dsp | faust | plot
CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib DSMAdd.dsp
./DSM2bipolar -n 10
compile
CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib DSMAdd.dsp
./DSM2bipolar
SuperCollider
export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
faust2supercollider -I ../faust/bitDSP-faust/lib -noprefix DSMAdd.dsp
Final output
Bipolar multi-bit signal to bipolar one-bit signal | declare name "DSMAdd";
declare author "Till Bovermann, Dario Sanfilippo";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
process = bit.bitstream_adder;
|
1ee56568c9a4ea0137c28f425f25f22eadcff3cecc71c2831db4b2bdab4d0b99 | rottingsounds/bitDSP-faust | DSM2unipolar.dsp | declare name "DSM2unipolar";
declare author "Till Bovermann, Dario Sanfilippo";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
// plot
// CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib DSM2unipolar.dsp
// ./DSM2unipolar -n 10
// compile
// CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib DSM2unipolar.dsp
// ./DSM2unipolar
// SuperCollider
// export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
// faust2supercollider -I ../faust/bitDSP-faust/lib -noprefix DSM2unipolar.dsp
// Final output
// Bipolar multi-bit signal to bipolar one-bit signal
process = bit.dsm2 > 0;
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/21cf36105c55b6e18969a867a319530a0ef1ea63/examples/_sc/DSM2unipolar.dsp | faust | plot
CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib DSM2unipolar.dsp
./DSM2unipolar -n 10
compile
CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib DSM2unipolar.dsp
./DSM2unipolar
SuperCollider
export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
faust2supercollider -I ../faust/bitDSP-faust/lib -noprefix DSM2unipolar.dsp
Final output
Bipolar multi-bit signal to bipolar one-bit signal | declare name "DSM2unipolar";
declare author "Till Bovermann, Dario Sanfilippo";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
process = bit.dsm2 > 0;
|
c672b1927ea600b0593cdc19af5fb20829e1ecbf91d7a9d6718d4e464caacf18 | afalaize/faust | karplus.dsp | declare name "karplus";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
//-----------------------------------------------
// karplus-strong
//-----------------------------------------------
import("stdfaust.lib");
// Excitator
//-----------
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
trigger(n) = upfront : release(n) : >(0.0);
size = hslider("excitation [unit:f]", 128, 2, 512, 1);
// resonator
//------------
dur = hslider("duration [unit:f]", 128, 2, 512, 1);
att = hslider("attenuation", 0.1, 0, 1, 0.01);
average(x) = (x+x')/2;
resonator(d, a) = (+ : de.delay(4096, d-1.5)) ~ (average : *(1.0-a)) ;
process = no.noise * hslider("level", 0.5, 0, 1, 0.01)
: vgroup("excitator", *(button("play"): trigger(size)))
: vgroup("resonator", resonator(dur, att));
| https://raw.githubusercontent.com/afalaize/faust/8f9f5fe3aa167eaeecc15a99d4da984ac2797be3/examples/physicalModeling/old/karplus.dsp | faust | -----------------------------------------------
karplus-strong
-----------------------------------------------
Excitator
-----------
resonator
------------ | declare name "karplus";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
import("stdfaust.lib");
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
trigger(n) = upfront : release(n) : >(0.0);
size = hslider("excitation [unit:f]", 128, 2, 512, 1);
dur = hslider("duration [unit:f]", 128, 2, 512, 1);
att = hslider("attenuation", 0.1, 0, 1, 0.01);
average(x) = (x+x')/2;
resonator(d, a) = (+ : de.delay(4096, d-1.5)) ~ (average : *(1.0-a)) ;
process = no.noise * hslider("level", 0.5, 0, 1, 0.01)
: vgroup("excitator", *(button("play"): trigger(size)))
: vgroup("resonator", resonator(dur, att));
|
f865256dfacc5ce8abe6f41e359e7130c6f6c76fe31bad121e5f68a0a68fbdd6 | friskgit/snares | i_snare_phase.dsp | // -*- compile-command: "cd .. && make jack src=i_snare_phase.dsp && cd -"; -*-&& cd -"; -*-
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
import("math.lib") ; // for PI definition
import("music.lib") ; // for osci definition
//---------------`Four drum instances phased equally` --------------------------
//
// Generating an impulse and feeding it to a generic_snarefs. Each impulse is delayed by 25%
// and sent to a separate instance of generic_snarefs. This allows for faster impulses 4X the
// speed of the pulse which is in samples.
//
// disperse.dsp doe not pass on the impules as generic_snarefs does.
//
// 18 Juli 2019 Henrik Frisk [email protected]
//---------------------------------------------------
p = hslider("pulse", 1, 1, 10000, 1);// : si.smooth(0.999);
per = ma.SR / p : int : *(4);
//This is to avoid lagging when modulating the pulse
hit(t) = (diff(ba.period(t)) < 0) + impulse
with {
diff(x) = x <: _ - _';
impulse = 1 - 1';
};
delA = per : *(0.25);
delB = per : *(0.5);
delC = per : *(0.75);
imp_delA = hit(per) : de.sdelay(192000, 64, delA);
imp_delB = hit(per) : de.sdelay(192000, 64, delB);
imp_delC = hit(per) : de.sdelay(192000, 64, delC);
process = ((hit(per) : component("generic_snarefs.dsp")[accent = 2;]),
(imp_delA : component("generic_snarefs.dsp")),
(imp_delB : component("generic_snarefs.dsp")),
(imp_delC : component("generic_snarefs.dsp"))) :> _;
| https://raw.githubusercontent.com/friskgit/snares/bb43ea5e706a0ead6d65dd176a5c492b2f5d8f74/faust/snare/src/i_snare_phase.dsp | faust | -*- compile-command: "cd .. && make jack src=i_snare_phase.dsp && cd -"; -*-&& cd -"; -*-
for PI definition
for osci definition
---------------`Four drum instances phased equally` --------------------------
Generating an impulse and feeding it to a generic_snarefs. Each impulse is delayed by 25%
and sent to a separate instance of generic_snarefs. This allows for faster impulses 4X the
speed of the pulse which is in samples.
disperse.dsp doe not pass on the impules as generic_snarefs does.
18 Juli 2019 Henrik Frisk [email protected]
---------------------------------------------------
: si.smooth(0.999);
This is to avoid lagging when modulating the pulse |
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
per = ma.SR / p : int : *(4);
hit(t) = (diff(ba.period(t)) < 0) + impulse
with {
diff(x) = x <: _ - _';
impulse = 1 - 1';
};
delA = per : *(0.25);
delB = per : *(0.5);
delC = per : *(0.75);
imp_delA = hit(per) : de.sdelay(192000, 64, delA);
imp_delB = hit(per) : de.sdelay(192000, 64, delB);
imp_delC = hit(per) : de.sdelay(192000, 64, delC);
process = ((hit(per) : component("generic_snarefs.dsp")[accent = 2;]),
(imp_delA : component("generic_snarefs.dsp")),
(imp_delB : component("generic_snarefs.dsp")),
(imp_delC : component("generic_snarefs.dsp"))) :> _;
|
85949597c388c8ee1806b196f52f54760ebdbcc0b328e8b6f5c8ff700af0bd3b | friskgit/snares | snare.dsp | // -*- compile-command: "cd .. && make sc && cd -"; -*-
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
//---------------`Snare drum synth` --------------------------
// A take at a snare drum synth
//
// A single hit snare drum synth
//
// Where:
// * midi note 67-89
// * stiffness 0-0.55 (mapped to note as in note 67 -> 0)§
// * midi velocity is mapped to pressure
// A useful parameter setting is:
//
// 30 Juni 2018 Henrik Frisk [email protected]
//---------------------------------------------------
channels = 2;
//imp = ba.pulse(hslider("tempo", 1000, 500, 10000, 1));
// env = en.ar(0.000001, 0.1, button("play"));
env = en.ar(attack, rel, imp) * amp
with {
attack = hslider("attack", 0.00000001, 0, 0.1, 0.000000001) : si.smooth(0.1);
rel = hslider("rel", 0.1, 0.0000001, 0.5, 0.0000001) : si.smooth(0.2);
imp = button("gate");
amp = hslider("vol", 0.5, 0, 1, 0.0001);
};
// Control the output channel
focus = hslider("focus", 1, 0, 1, 0.0001);
position = hslider("position", 1, 0, channels, 1);
rate = ma.SR/1000.0;
rndctrl = (no.lfnoise(rate) * (channels + 1)) * focus : ma.fabs + position : int ;
outputctrl = rndctrl : ba.sAndH(imp);
n = no.multinoise(8) : par(i, 8, _ * env * 0.1);
filt = fi.resonbp(frq, q, gain)
with {
frq = hslider("freq", 200, 50, 5000, 0.1);
q = hslider("q", 1, 0.01, 10, 0.01);
gain = hslider("gain", 0, 0, 2, 0.00001);
};
ch_wrapped = ma.modulo(outputctrl, channels);
process = n : par(i, 8, filt);
// :> ba.selectoutn(channels, ch_wrapped);
//process = n : par(i, 8, filt) :> _,_;
| https://raw.githubusercontent.com/friskgit/snares/bb43ea5e706a0ead6d65dd176a5c492b2f5d8f74/max/snare%7E.mxo/snare.dsp | faust | -*- compile-command: "cd .. && make sc && cd -"; -*-
---------------`Snare drum synth` --------------------------
A take at a snare drum synth
A single hit snare drum synth
Where:
* midi note 67-89
* stiffness 0-0.55 (mapped to note as in note 67 -> 0)§
* midi velocity is mapped to pressure
A useful parameter setting is:
30 Juni 2018 Henrik Frisk [email protected]
---------------------------------------------------
imp = ba.pulse(hslider("tempo", 1000, 500, 10000, 1));
env = en.ar(0.000001, 0.1, button("play"));
Control the output channel
:> ba.selectoutn(channels, ch_wrapped);
process = n : par(i, 8, filt) :> _,_; |
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
channels = 2;
env = en.ar(attack, rel, imp) * amp
with {
attack = hslider("attack", 0.00000001, 0, 0.1, 0.000000001) : si.smooth(0.1);
rel = hslider("rel", 0.1, 0.0000001, 0.5, 0.0000001) : si.smooth(0.2);
imp = button("gate");
amp = hslider("vol", 0.5, 0, 1, 0.0001);
};
focus = hslider("focus", 1, 0, 1, 0.0001);
position = hslider("position", 1, 0, channels, 1);
rate = ma.SR/1000.0;
rndctrl = (no.lfnoise(rate) * (channels + 1)) * focus : ma.fabs + position : int ;
outputctrl = rndctrl : ba.sAndH(imp);
n = no.multinoise(8) : par(i, 8, _ * env * 0.1);
filt = fi.resonbp(frq, q, gain)
with {
frq = hslider("freq", 200, 50, 5000, 0.1);
q = hslider("q", 1, 0.01, 10, 0.01);
gain = hslider("gain", 0, 0, 2, 0.00001);
};
ch_wrapped = ma.modulo(outputctrl, channels);
process = n : par(i, 8, filt);
|
c1b07341ec500f0c4cfeedb9211871ffc5c7115df583b736404368e8fdc9c664 | chmaha/Enover | enover.dsp | declare name "Enover";
declare description "A feedback-delay-network reverb";
declare author "Julius O. Smith III, Christopher Arndt, chmaha";
declare copyright "Copyright (C) 2003-2019 by Julius O. Smith III <[email protected]>";
declare license "GPLv3";
declare version "0.1.0";
import("stdfaust.lib");
zita_rev1 = _,_ <: re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ : out_eq,_,_ :
dry_wet : out_level
with{
fsmax = 48000.0; // highest sampling rate that will be used
fdn_group(x) = hgroup(
"[0] Zita_Rev1 [tooltip: ~ ZITA REV1 FEEDBACK DELAY NETWORK (FDN) & SCHROEDER
ALLPASS-COMB REVERBERATOR (8x8). See Faust's reverbs.lib for documentation and
references]", x);
in_group(x) = fdn_group(hgroup("[1] Input", x));
rdel = in_group(vslider("[1] Initial Delay [unit:ms] [style:knob] [tooltip: Delay in ms
before reverberation begins]",40,20,100,1));
freq_group(x) = fdn_group(hgroup("[2] Decay Times in Bands (see tooltips)", x));
f1 = freq_group(vslider("[1] LowFreq X [unit:Hz] [style:knob] [scale:log] [tooltip:
Crossover frequency (Hz) separating low and middle frequencies]", 250, 50, 1000, 1));
t60dc = freq_group(vslider("[2] Bass Mult [unit:x] [style:knob] [scale:log]
[style:knob] [tooltip: Bass Mult = Low Decay is equal to Mid Decay x Low Mult]",
1.5, 0.5, 2, 0.1)) * t60m;
t60m = freq_group(vslider("[3] Mid Decay [unit:s] [style:knob] [scale:log] [tooltip:
T60 = time (in seconds) to decay 60dB in middle band]",2, 1, 8, 0.1));
f2 = freq_group(vslider("[4] HF Damping [unit:Hz] [style:knob] [scale:log]
[tooltip: Frequency (Hz) at which the high-frequency T60 is half the middle-band's T60]",
3000, 1500, 0.49*fsmax, 1));
out_eq = pareq_stereo(eq1f,eq1l,eq1q) : pareq_stereo(eq2f,eq2l,eq2q);
// Zolzer style peaking eq (not used in zita-rev1) (filters.lib):
// pareq_stereo(eqf,eql,Q) = peak_eq(eql,eqf,eqf/Q), peak_eq(eql,eqf,eqf/Q);
// Regalia-Mitra peaking eq with "Q" hard-wired near sqrt(g)/2 (filters.lib):
pareq_stereo(eqf,eql,Q) = fi.peak_eq_rm(eql,eqf,tpbt), fi.peak_eq_rm(eql,eqf,tpbt)
with {
tpbt = wcT/sqrt(max(0,g)); // tan(PI*B/SR), B bw in Hz (Q^2 ~ g/4)
wcT = 2*ma.PI*eqf/ma.SR; // peak frequency in rad/sample
g = ba.db2linear(eql); // peak gain
};
eq1_group(x) = fdn_group(hgroup("[3] RM Peaking Equalizer 1", x));
eq1f = 315;
eq1l = 0;
eq1q = 3;
eq2_group(x) = fdn_group(hgroup("[4] RM Peaking Equalizer 2", x));
eq2f = 1500;
eq2l = 0;
eq2q = 3;
out_group(x) = fdn_group(hgroup("[5] Output", x));
dry_wet(x,y) = *(wet) + dry*x, *(wet) + dry*y with {
wet = 0.5*(drywet+1.0);
dry = 1.0-wet;
};
drywet = out_group(vslider("[1] Wet/Dry Mix [style:knob] [tooltip: -1 = dry, 1 = wet]",
0, -1.0, 1.0, 0.01)) : si.smoo;
out_level = *(gain),*(gain);
gain = out_group(vslider("[2] Level [unit:dB] [style:knob] [tooltip: Output scale
factor]", 0, -20, 20, 0.1)) : ba.db2linear : si.smoo;
};
process = _,_ : zita_rev1 : _,_;
| https://raw.githubusercontent.com/chmaha/Enover/fc8f665d2d4b627e4196437d068fe0050deadf87/faust/enover.dsp | faust | highest sampling rate that will be used
Zolzer style peaking eq (not used in zita-rev1) (filters.lib):
pareq_stereo(eqf,eql,Q) = peak_eq(eql,eqf,eqf/Q), peak_eq(eql,eqf,eqf/Q);
Regalia-Mitra peaking eq with "Q" hard-wired near sqrt(g)/2 (filters.lib):
tan(PI*B/SR), B bw in Hz (Q^2 ~ g/4)
peak frequency in rad/sample
peak gain | declare name "Enover";
declare description "A feedback-delay-network reverb";
declare author "Julius O. Smith III, Christopher Arndt, chmaha";
declare copyright "Copyright (C) 2003-2019 by Julius O. Smith III <[email protected]>";
declare license "GPLv3";
declare version "0.1.0";
import("stdfaust.lib");
zita_rev1 = _,_ <: re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ : out_eq,_,_ :
dry_wet : out_level
with{
fdn_group(x) = hgroup(
"[0] Zita_Rev1 [tooltip: ~ ZITA REV1 FEEDBACK DELAY NETWORK (FDN) & SCHROEDER
ALLPASS-COMB REVERBERATOR (8x8). See Faust's reverbs.lib for documentation and
references]", x);
in_group(x) = fdn_group(hgroup("[1] Input", x));
rdel = in_group(vslider("[1] Initial Delay [unit:ms] [style:knob] [tooltip: Delay in ms
before reverberation begins]",40,20,100,1));
freq_group(x) = fdn_group(hgroup("[2] Decay Times in Bands (see tooltips)", x));
f1 = freq_group(vslider("[1] LowFreq X [unit:Hz] [style:knob] [scale:log] [tooltip:
Crossover frequency (Hz) separating low and middle frequencies]", 250, 50, 1000, 1));
t60dc = freq_group(vslider("[2] Bass Mult [unit:x] [style:knob] [scale:log]
[style:knob] [tooltip: Bass Mult = Low Decay is equal to Mid Decay x Low Mult]",
1.5, 0.5, 2, 0.1)) * t60m;
t60m = freq_group(vslider("[3] Mid Decay [unit:s] [style:knob] [scale:log] [tooltip:
T60 = time (in seconds) to decay 60dB in middle band]",2, 1, 8, 0.1));
f2 = freq_group(vslider("[4] HF Damping [unit:Hz] [style:knob] [scale:log]
[tooltip: Frequency (Hz) at which the high-frequency T60 is half the middle-band's T60]",
3000, 1500, 0.49*fsmax, 1));
out_eq = pareq_stereo(eq1f,eq1l,eq1q) : pareq_stereo(eq2f,eq2l,eq2q);
pareq_stereo(eqf,eql,Q) = fi.peak_eq_rm(eql,eqf,tpbt), fi.peak_eq_rm(eql,eqf,tpbt)
with {
};
eq1_group(x) = fdn_group(hgroup("[3] RM Peaking Equalizer 1", x));
eq1f = 315;
eq1l = 0;
eq1q = 3;
eq2_group(x) = fdn_group(hgroup("[4] RM Peaking Equalizer 2", x));
eq2f = 1500;
eq2l = 0;
eq2q = 3;
out_group(x) = fdn_group(hgroup("[5] Output", x));
dry_wet(x,y) = *(wet) + dry*x, *(wet) + dry*y with {
wet = 0.5*(drywet+1.0);
dry = 1.0-wet;
};
drywet = out_group(vslider("[1] Wet/Dry Mix [style:knob] [tooltip: -1 = dry, 1 = wet]",
0, -1.0, 1.0, 0.01)) : si.smoo;
out_level = *(gain),*(gain);
gain = out_group(vslider("[2] Level [unit:dB] [style:knob] [tooltip: Output scale
factor]", 0, -20, 20, 0.1)) : ba.db2linear : si.smoo;
};
process = _,_ : zita_rev1 : _,_;
|
9de4496beaa05d32e41227c23d2b6dcdf5a60f807118f73be82029d545419587 | rottingsounds/bitDSP-faust | boolOsc0.dsp | declare name "boolOsc0";
declare description "bool_osc_0 - example";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
// SuperCollider
// CXXFLAGS="-I ../../../../include" faust2supercollider -I ../../lib -noprefix boolOsc0.dsp
// plot
// CXXFLAGS="-I ../include" faust2csvplot -I ../lib boolOsc0.dsp
// ./boolOsc0 -n 10
// compile
// CXXFLAGS="-I ../../../include" faust2caqt -I ../lib boolOsc0.dsp
// ./boolOsc0
dt1 = int(hslider("dt1",0,0,1,0) * ma.SR);
dt2 = int(hslider("dt2",0,0,1,0) * ma.SR);
dt3 = int(hslider("dt3",0,0,1,0) * ma.SR);
dt4 = int(hslider("dt4",0,0,1,0) * ma.SR);
// mono out
process = bit.bool_osc0(dt1, dt2, dt3, dt4); | https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/21cf36105c55b6e18969a867a319530a0ef1ea63/examples/_sc/boolOsc0.dsp | faust | SuperCollider
CXXFLAGS="-I ../../../../include" faust2supercollider -I ../../lib -noprefix boolOsc0.dsp
plot
CXXFLAGS="-I ../include" faust2csvplot -I ../lib boolOsc0.dsp
./boolOsc0 -n 10
compile
CXXFLAGS="-I ../../../include" faust2caqt -I ../lib boolOsc0.dsp
./boolOsc0
mono out | declare name "boolOsc0";
declare description "bool_osc_0 - example";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
dt1 = int(hslider("dt1",0,0,1,0) * ma.SR);
dt2 = int(hslider("dt2",0,0,1,0) * ma.SR);
dt3 = int(hslider("dt3",0,0,1,0) * ma.SR);
dt4 = int(hslider("dt4",0,0,1,0) * ma.SR);
process = bit.bool_osc0(dt1, dt2, dt3, dt4); |
6f83840bb923386cdaebd2bad7ddc6d6d0790f63d05bf33ae3115396cab6ea8a | rottingsounds/BoolOscFB1 | BoolOscFB1.dsp | declare name "BoolOscFB1";
declare description "bool_osc FB alternative 1";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
// bit = library("bitDSP.lib");
// SuperCollider
// CXXFLAGS="-I ../../../../include" faust2supercollider -I ../../lib -noprefix BoolOscFB1.dsp
// plot
// CXXFLAGS="-I ../include" faust2csvplot -I ../lib boolOsc_fb.dsp
// ./boolOsc_fb -n 10
// compile
// CXXFLAGS="-I ../../../include" faust2caqt -I ../lib boolOsc_fb.dsp
// ./boolOsc_fb
// bool_osc1_mod(del1, del2) = node1 letrec {
// 'node1 = not(node1 xor node2 & node1) @ min(maxDel,(del1 % maxDel));
// 'node2 = not(node2 xor node1 xor node2) @ min(maxDel,(del2 % maxDel));
// };
// bool_osc2_mod(del1, del2) = node1 letrec {
// 'node1 = not(node1 & node2) @ min(maxDel,(del1 % maxDel));
// 'node2 = not(node1 & node2) @ min(maxDel,(del2 % maxDel));
// };
bool_osc1_mod(d1, d2) = node1 letrec {
'node1 = delay(d1, not(node1 xor node2 & node1));
'node2 = delay(d2, not(node2 xor node1 xor node2));
};
bool_osc2_mod(d1, d2) = node1 letrec {
'node1 = delay(d1, not(node1 & node2));
'node2 = delay(d2, not(node1 & node2));
};
delay(d, x) = de.delay(maxDel, (d % maxDel), x);
not(x) = rint(1 - x);
maxDel = ma.SR / 4;
oscfb1(leakcoef, bias1, bias2, mod1, mod2) = loop ~ _ with {
loop(fb) = bool_osc1_mod(
(fb : map(bias1, mod2)),
(fb : map(bias2, mod1))
) : leakdc(leakcoef);
};
oscfb2(leakcoef, bias1, bias2, mod1, mod2) = loop ~ _ with {
loop(fb) = bool_osc2_mod(
(fb : map(bias1, mod1)),
(fb : map(bias2, mod2))
) : leakdc(leakcoef);
};
map(bias, scale, in) = max(0, (bias * biasfac) + (scale * in * modfac)) with {
biasfac = 15000;
modfac = 15000;
};
// sc-like leakdc
leakdc(coef, x) = y letrec {
'y = x - x' + coef * y;
};
// MS processor
// ms(x, y, width) = (x + y) * 0.5, (x-y) * 0.5 * width;
ms(1, midIn, sideIn) = (midIn + sideIn) * 0.5, (midIn-sideIn) * 0.5;
ms(width, midIn, sideIn) = (mid + side) * 0.5, (mid-side) * 0.5 with {
mid = midIn;
side = sideIn * width;
};
rotate2(r, x, y) = xout, yout with {
xout = cos(r) * x + sin(r) * y;
yout = cos(r) * y - sin(r) * x;
};
// stereo out
process = (
oscfb1(leakcoef, bias1, bias2, mod1, mod2),
oscfb2(leakcoef, bias1, bias2, mod1, mod2)
)
: leakDC
: rotate2(rot)
: ms(1) : ms(width)
: vca(distort)
: tanh
: vca(amp)
: fi.lowpass(lporder, lpfreq), fi.lowpass(lporder, lpfreq)
// <: si.bus(2)
with {
mod1 = hslider("[01]mod1[scale:exp]", 0.0001, 0.0001, 1, 0.00001) : si.smoo;
mod2 = hslider("[02]mod2[scale:exp]", 0.0001, 0.0001, 1, 0.00001) : si.smoo;
bias1 = hslider("[03]bias1[scale:exp]", 0.0001, 0.0001, 1, 0.00001) : si.smoo;
bias2 = hslider("[04]bias2[scale:exp]", 0.0001, 0.0001, 1, 0.00001) : si.smoo;
leak = hslider("[05]leak", 0.01, 0, 1, 0.0001) : si.smoo;
leakcoef = 1 - (leak * 0.001);
distort = hslider("[06]distort[scale:exp]", 1, 1, 10, 0.001) : si.smoo;
amp = hslider("[07]amp", 0.5, 0, 1, 0.001) : si.smoo;
width = hslider("[08]width", 1, 0, 1, 0.001) : si.smoo;
rot = hslider("[09]rot", 0.25, 0, 1, 0.001) : si.smoo * ma.PI;
lpfreq = hslider("[10]lpfreq[scale:exp]", 10000, 10, 20000, 1) : si.smoo;
lporder = 4;
tanh = ma.tanh(_), ma.tanh(_);
leakDC = leakdc(0.999), leakdc(0.999);
vca(amp) = _ * amp, _ * amp;
};
| https://raw.githubusercontent.com/rottingsounds/BoolOscFB1/b6e58251575b296268d16c2f0c48256137f4f35c/faust/BoolOscFB1.dsp | faust | bit = library("bitDSP.lib");
SuperCollider
CXXFLAGS="-I ../../../../include" faust2supercollider -I ../../lib -noprefix BoolOscFB1.dsp
plot
CXXFLAGS="-I ../include" faust2csvplot -I ../lib boolOsc_fb.dsp
./boolOsc_fb -n 10
compile
CXXFLAGS="-I ../../../include" faust2caqt -I ../lib boolOsc_fb.dsp
./boolOsc_fb
bool_osc1_mod(del1, del2) = node1 letrec {
'node1 = not(node1 xor node2 & node1) @ min(maxDel,(del1 % maxDel));
'node2 = not(node2 xor node1 xor node2) @ min(maxDel,(del2 % maxDel));
};
bool_osc2_mod(del1, del2) = node1 letrec {
'node1 = not(node1 & node2) @ min(maxDel,(del1 % maxDel));
'node2 = not(node1 & node2) @ min(maxDel,(del2 % maxDel));
};
sc-like leakdc
MS processor
ms(x, y, width) = (x + y) * 0.5, (x-y) * 0.5 * width;
stereo out
<: si.bus(2)
| declare name "BoolOscFB1";
declare description "bool_osc FB alternative 1";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bool_osc1_mod(d1, d2) = node1 letrec {
'node1 = delay(d1, not(node1 xor node2 & node1));
'node2 = delay(d2, not(node2 xor node1 xor node2));
};
bool_osc2_mod(d1, d2) = node1 letrec {
'node1 = delay(d1, not(node1 & node2));
'node2 = delay(d2, not(node1 & node2));
};
delay(d, x) = de.delay(maxDel, (d % maxDel), x);
not(x) = rint(1 - x);
maxDel = ma.SR / 4;
oscfb1(leakcoef, bias1, bias2, mod1, mod2) = loop ~ _ with {
loop(fb) = bool_osc1_mod(
(fb : map(bias1, mod2)),
(fb : map(bias2, mod1))
) : leakdc(leakcoef);
};
oscfb2(leakcoef, bias1, bias2, mod1, mod2) = loop ~ _ with {
loop(fb) = bool_osc2_mod(
(fb : map(bias1, mod1)),
(fb : map(bias2, mod2))
) : leakdc(leakcoef);
};
map(bias, scale, in) = max(0, (bias * biasfac) + (scale * in * modfac)) with {
biasfac = 15000;
modfac = 15000;
};
leakdc(coef, x) = y letrec {
'y = x - x' + coef * y;
};
ms(1, midIn, sideIn) = (midIn + sideIn) * 0.5, (midIn-sideIn) * 0.5;
ms(width, midIn, sideIn) = (mid + side) * 0.5, (mid-side) * 0.5 with {
mid = midIn;
side = sideIn * width;
};
rotate2(r, x, y) = xout, yout with {
xout = cos(r) * x + sin(r) * y;
yout = cos(r) * y - sin(r) * x;
};
process = (
oscfb1(leakcoef, bias1, bias2, mod1, mod2),
oscfb2(leakcoef, bias1, bias2, mod1, mod2)
)
: leakDC
: rotate2(rot)
: ms(1) : ms(width)
: vca(distort)
: tanh
: vca(amp)
: fi.lowpass(lporder, lpfreq), fi.lowpass(lporder, lpfreq)
with {
mod1 = hslider("[01]mod1[scale:exp]", 0.0001, 0.0001, 1, 0.00001) : si.smoo;
mod2 = hslider("[02]mod2[scale:exp]", 0.0001, 0.0001, 1, 0.00001) : si.smoo;
bias1 = hslider("[03]bias1[scale:exp]", 0.0001, 0.0001, 1, 0.00001) : si.smoo;
bias2 = hslider("[04]bias2[scale:exp]", 0.0001, 0.0001, 1, 0.00001) : si.smoo;
leak = hslider("[05]leak", 0.01, 0, 1, 0.0001) : si.smoo;
leakcoef = 1 - (leak * 0.001);
distort = hslider("[06]distort[scale:exp]", 1, 1, 10, 0.001) : si.smoo;
amp = hslider("[07]amp", 0.5, 0, 1, 0.001) : si.smoo;
width = hslider("[08]width", 1, 0, 1, 0.001) : si.smoo;
rot = hslider("[09]rot", 0.25, 0, 1, 0.001) : si.smoo * ma.PI;
lpfreq = hslider("[10]lpfreq[scale:exp]", 10000, 10, 20000, 1) : si.smoo;
lporder = 4;
tanh = ma.tanh(_), ma.tanh(_);
leakDC = leakdc(0.999), leakdc(0.999);
vca(amp) = _ * amp, _ * amp;
};
|
c666d59a4e2cbebe49201f297fbc4f4dbaa8162cf5ca885423c60a3acaa76431 | LucaSpanedda/Sound_reading_and_writing_techniques_in_Faust | 1.01_Wavetable_Realtime_Recorder.dsp | // ---------------------------------------------------------------------------------
declare name "Realtime Recorder";
declare version "1.0";
declare author "Luca Spanedda";
/*
Oscillator from wave-table recording
*/
// import Standard Faust library
// https://github.com/grame-cncm/faustlibraries/
import("stdfaust.lib");
// GUI
freqgui = hslider("[1] Frequency",1,0,2,0.001) : si.smoo;
ampgui = hslider("[2] Amp",1,0,2,0.001) : si.smoo;
buttongui = button("[0] Rec");
// WAVETABLE RECORDER
looptable(recstart,freq) = _*buttongui : rwtable(dimension,0.0,indexwrite,_,indexread)
with{
offset = 2 : int; // Offset for write and read. For point the write index at 0 when stopped.
record = recstart : int; // record the memory with the int value of 1
dimension = 1024+offset : int; // dimension in samples the memory table
decimal(x)= x-int(x); // rescale the int (for the phasor)
phasor = ((ma.SR/dimension)/ma.SR)*freq : (+ : decimal)~ _; // phasor from 0 to 1
indexwrite = ((phasor*recstart)*(dimension-offset)):_+offset*recstart :int; // write the input
indexread = ((phasor)*(dimension-offset)):_+offset :int; // read the written signal+offset
};
wavetable = _ : looptable(buttongui,freqgui)*ampgui; // routing of the GUI
process = wavetable <: _,_;
// ---------------------------------------------------------------------------------
| https://raw.githubusercontent.com/LucaSpanedda/Sound_reading_and_writing_techniques_in_Faust/bb01eff05a51424c16420a00b383441d8973d85e/0_work-in-progress/1.01_Wavetable_Realtime_Recorder.dsp | faust | ---------------------------------------------------------------------------------
Oscillator from wave-table recording
import Standard Faust library
https://github.com/grame-cncm/faustlibraries/
GUI
WAVETABLE RECORDER
Offset for write and read. For point the write index at 0 when stopped.
record the memory with the int value of 1
dimension in samples the memory table
rescale the int (for the phasor)
phasor from 0 to 1
write the input
read the written signal+offset
routing of the GUI
--------------------------------------------------------------------------------- | declare name "Realtime Recorder";
declare version "1.0";
declare author "Luca Spanedda";
import("stdfaust.lib");
freqgui = hslider("[1] Frequency",1,0,2,0.001) : si.smoo;
ampgui = hslider("[2] Amp",1,0,2,0.001) : si.smoo;
buttongui = button("[0] Rec");
looptable(recstart,freq) = _*buttongui : rwtable(dimension,0.0,indexwrite,_,indexread)
with{
};
process = wavetable <: _,_;
|
c740113e345c994aa96285975cbd9788530fd8f2a9594799a897d55555bce826 | inria-emeraude/syfala | lms_live.dsp | declare name "Least Mean Square Algorithm";
declare version "1.0";
declare author "Pierre Lecomte";
declare author "Loic Alexandre";
declare license "CC-BY-NC-SA-4.0";
import("stdfaust.lib");
N = 5; // Number of coefficients (should be >= 200 to identify a system with narrow frequency band)
coeffs = si.bus(N);
y = _; // y = x*[room impulse response]
h_hat(N) = (si.bus(N),(_<:(si.bus(N)))):ro.interleave(N,2):sum(i, N, (_,@(i):*)); // Adapted filter
y_hat(N) = ((si.bus(N)<:si.bus(2*N)),_):(si.bus(N),h_hat(N)); // Output from the adapted system
buffer = _<:par(i,N,@(i)); // To obtain x_n, the reference signal at time n
filter_freq = fi.lowpass(4,800); // Lowpass filter for frequency band reduction
signal = no.noise; // Excitation signal
x = (signal:_<:(_,_,_)); // Reference signal
in = _,x:ro.crossn1(3); // Inputs including the external microphone signal and the reference signal
mu = -0.0001; // Convergence coefficient (smaller for slower convergence)
// Input = microphone signal whichcorresponds to the target system ouput y
// Output 0 = x (white noise signal)
// Output 1 = error signal (y - y_hat)
process = ((coeffs,in):(y_hat(N),y,_,_):(coeffs,(-<:(_,_*mu)),buffer,_):(coeffs,_,(_<:si.bus(N)),coeffs,_):(coeffs,_,ro.interleave(N,2),_):(coeffs,_,par(i,N,*),_):(coeffs,ro.cross1n(N),_):(ro.interleave(N,2),_,_):(par(i,N,+),_,_))~si.bus(N):(par(i,N,!),ro.cross(2));
| https://raw.githubusercontent.com/inria-emeraude/syfala/95ed6765d73520362f6a1ad35e4a3b2a5e16fbc9/examples/lms_live.dsp | faust | Number of coefficients (should be >= 200 to identify a system with narrow frequency band)
y = x*[room impulse response]
Adapted filter
Output from the adapted system
To obtain x_n, the reference signal at time n
Lowpass filter for frequency band reduction
Excitation signal
Reference signal
Inputs including the external microphone signal and the reference signal
Convergence coefficient (smaller for slower convergence)
Input = microphone signal whichcorresponds to the target system ouput y
Output 0 = x (white noise signal)
Output 1 = error signal (y - y_hat) | declare name "Least Mean Square Algorithm";
declare version "1.0";
declare author "Pierre Lecomte";
declare author "Loic Alexandre";
declare license "CC-BY-NC-SA-4.0";
import("stdfaust.lib");
coeffs = si.bus(N);
process = ((coeffs,in):(y_hat(N),y,_,_):(coeffs,(-<:(_,_*mu)),buffer,_):(coeffs,_,(_<:si.bus(N)),coeffs,_):(coeffs,_,ro.interleave(N,2),_):(coeffs,_,par(i,N,*),_):(coeffs,ro.cross1n(N),_):(ro.interleave(N,2),_,_):(par(i,N,+),_,_))~si.bus(N):(par(i,N,!),ro.cross(2));
|
2b9e7139d114887475436b633e726dca09e08fbcc5cfccb3622088001e0d8ea6 | s-e-a-m/faust-libraries | lr2bfmt.dsp | declare name "MICHAEL GERZON STEREO TO BFORMAT ENCODER";
declare version "001";
declare author "Giuseppe Silvi";
declare license "GNU-GPL-v3";
declare copyright "(c)SEAM 2019";
declare description "MICHAEL GERZON STEREO TO BFORMAT ENCODER";
import("stdfaust.lib");
import("../../seam.lib");
// LS and RS are dead channels to create VST routing consistency
// lr2bfmt(L,R,LR,RS) = W,X,Y,Z
// with{
// azi = 45.0 : deg2rad;
// elv = 00.0 : deg2rad;
//
// WL = L * 0.707;
// XL = L * cos(azi) * cos(elv);
// YL = L * sin(azi) * cos(elv);
// ZL = L * sin(elv);
//
// WR = R * 0.707;
// XR = R * cos(-azi) * cos(elv);
// YR = R * sin(-azi) * cos(elv);
// ZR = R * sin(elv);
//
// W = 0.707 * (WL + WR);
// X = 0.707 * (XL + XR);
// Y = 0.707 * (YL + YR);
// Z = 0.707 * (ZL + ZR);
// };
process = lrms2bfmt;
| https://raw.githubusercontent.com/s-e-a-m/faust-libraries/9120cccb9335f42407062eb4bf149188d8018b07/examples/vst/lr2bfmt.dsp | faust | LS and RS are dead channels to create VST routing consistency
lr2bfmt(L,R,LR,RS) = W,X,Y,Z
with{
azi = 45.0 : deg2rad;
elv = 00.0 : deg2rad;
WL = L * 0.707;
XL = L * cos(azi) * cos(elv);
YL = L * sin(azi) * cos(elv);
ZL = L * sin(elv);
WR = R * 0.707;
XR = R * cos(-azi) * cos(elv);
YR = R * sin(-azi) * cos(elv);
ZR = R * sin(elv);
W = 0.707 * (WL + WR);
X = 0.707 * (XL + XR);
Y = 0.707 * (YL + YR);
Z = 0.707 * (ZL + ZR);
}; | declare name "MICHAEL GERZON STEREO TO BFORMAT ENCODER";
declare version "001";
declare author "Giuseppe Silvi";
declare license "GNU-GPL-v3";
declare copyright "(c)SEAM 2019";
declare description "MICHAEL GERZON STEREO TO BFORMAT ENCODER";
import("stdfaust.lib");
import("../../seam.lib");
process = lrms2bfmt;
|
a7d0f33e6af3129c2bdc757b0f79228b48db5ce4576bb27bc65967287a109a9c | sonosaurus/sonobus | compressjlc.dsp | //----------------------------`(dm.)compressor_demo`-------------------------
// Compressor demo application.
//
// #### Usage
//
// ```
// _,_ : compressor_demo : _,_;
// ```
//------------------------------------------------------------
declare name "compressor";
declare version "0.0";
declare author "JOS, revised by RM";
declare description "Compressor demo application";
import("stdfaust.lib");
compressorjlc_demo = ba.bypass2(cbp,compressor_stereo_demo)
with{
main_group(x) = vgroup("compressor [tooltip: Reference:
http://en.wikipedia.org/wiki/Dynamic_range_compression]", x);
comp_group(x) = main_group(hgroup("[0] comp", x));
env_group(x) = main_group(hgroup("[1] env", x));
gain_group(x) = main_group(hgroup("[2] gain", x));
cbp = main_group(checkbox("[0] Bypass [tooltip: When this is checked, the compressor
has no effect]"));
//gainview = co.compression_gain_mono(ratio,threshold,attack,release) :
gainview = co.compression_gain_mono(ratio,threshold,attack,release) : ba.linear2db :
gain_group(hbargraph("[1] outgain [unit:db] [tooltip: Current gain of
the compressor in linear gain]",0,2.0)) ;
displaygain = _,_ <: _,_,(abs,abs:+) : _,_,gainview : _,attach;
compressor_stereo_demo =
displaygain(co.compressor_stereo(ratio,threshold,attack,release)) :
*(makeupgain) , *(makeupgain);
//ctl_group(x) = knob_group(hgroup("[3] Compression Control", x));
ratio = comp_group(hslider("[1] ratio [style:knob]
[tooltip: A compression Ratio of N means that for each N dB increase in input
signal level above Threshold, the output level goes up 1 dB]",
5, 1, 20, 0.1));
threshold = comp_group(hslider("[0] threshold [unit:dB] [style:knob]
[tooltip: When the signal level exceeds the Threshold (in dB), its level
is compressed according to the Ratio]",
-30, -100, 10, 0.1));
// env_group(x) = knob_group(hgroup("[4] Compression Response", x));
attack = env_group(hslider("[0] attack [unit:sec] [style:knob] [scale:log]
[tooltip: Time constant in ms (1/e smoothing time) for the compression gain
to approach (exponentially) a new lower target level (the compression
`kicking in')]", 50, 1, 1000, 0.1)) : max(1/ma.SR);
release = env_group(hslider("[1] release [unit:sec] [style: knob] [scale:log]
[tooltip: Time constant in ms (1/e smoothing time) for the compression gain
to approach (exponentially) a new higher target level (the compression
'releasing')]", 500, 1, 1000, 0.1)) : max(1/ma.SR);
makeupgain = gain_group(hslider("[0] makeup gain [unit:dB]
[tooltip: The compressed-signal output level is increased by this amount
(in dB) to make up for the level lost due to compression]",
40, -96, 96, 0.1)) : ba.db2linear : si.smoo;
};
process = compressorjlc_demo; | https://raw.githubusercontent.com/sonosaurus/sonobus/52cb09f046b94ba2d5d9d47e85c8d9789d86c96e/scripts/compressjlc.dsp | faust | ----------------------------`(dm.)compressor_demo`-------------------------
Compressor demo application.
#### Usage
```
_,_ : compressor_demo : _,_;
```
------------------------------------------------------------
en.wikipedia.org/wiki/Dynamic_range_compression]", x);
gainview = co.compression_gain_mono(ratio,threshold,attack,release) :
ctl_group(x) = knob_group(hgroup("[3] Compression Control", x));
env_group(x) = knob_group(hgroup("[4] Compression Response", x)); |
declare name "compressor";
declare version "0.0";
declare author "JOS, revised by RM";
declare description "Compressor demo application";
import("stdfaust.lib");
compressorjlc_demo = ba.bypass2(cbp,compressor_stereo_demo)
with{
main_group(x) = vgroup("compressor [tooltip: Reference:
comp_group(x) = main_group(hgroup("[0] comp", x));
env_group(x) = main_group(hgroup("[1] env", x));
gain_group(x) = main_group(hgroup("[2] gain", x));
cbp = main_group(checkbox("[0] Bypass [tooltip: When this is checked, the compressor
has no effect]"));
gainview = co.compression_gain_mono(ratio,threshold,attack,release) : ba.linear2db :
gain_group(hbargraph("[1] outgain [unit:db] [tooltip: Current gain of
the compressor in linear gain]",0,2.0)) ;
displaygain = _,_ <: _,_,(abs,abs:+) : _,_,gainview : _,attach;
compressor_stereo_demo =
displaygain(co.compressor_stereo(ratio,threshold,attack,release)) :
*(makeupgain) , *(makeupgain);
ratio = comp_group(hslider("[1] ratio [style:knob]
[tooltip: A compression Ratio of N means that for each N dB increase in input
signal level above Threshold, the output level goes up 1 dB]",
5, 1, 20, 0.1));
threshold = comp_group(hslider("[0] threshold [unit:dB] [style:knob]
[tooltip: When the signal level exceeds the Threshold (in dB), its level
is compressed according to the Ratio]",
-30, -100, 10, 0.1));
attack = env_group(hslider("[0] attack [unit:sec] [style:knob] [scale:log]
[tooltip: Time constant in ms (1/e smoothing time) for the compression gain
to approach (exponentially) a new lower target level (the compression
`kicking in')]", 50, 1, 1000, 0.1)) : max(1/ma.SR);
release = env_group(hslider("[1] release [unit:sec] [style: knob] [scale:log]
[tooltip: Time constant in ms (1/e smoothing time) for the compression gain
to approach (exponentially) a new higher target level (the compression
'releasing')]", 500, 1, 1000, 0.1)) : max(1/ma.SR);
makeupgain = gain_group(hslider("[0] makeup gain [unit:dB]
[tooltip: The compressed-signal output level is increased by this amount
(in dB) to make up for the level lost due to compression]",
40, -96, 96, 0.1)) : ba.db2linear : si.smoo;
};
process = compressorjlc_demo; |
ad0c9caa8b6ac986ac12ae675e4fbc8a664b2fcab3572d26c6464e306da665ca | jpecquais/faustLab | jlpCompressor.dsp | import("lib/jlpLibs.lib");
import("stdfaust.lib");
declare name "Compressor";
declare author "Jean-Loup Pecquais";
declare version "1.00";
declare license "GPL3";
// TODO : TO BE IMPLEMENTED IN A PROPER LIB
delay(time) = @(max(0,floor(0.5+ma.SR*time)));
//TODO : FORCE FEEDFORWARD WHEN LAH IS ON
comp_BigBrother_nCh(strength,thresh,att,rel,hld,rms,knee,lad,link,FBFF,meter,N) =
si.bus(N) <: si.bus(N*2):
(
(
(
(ro.interleave(N,2):par(i, N*2, abs) :par(i, N, it.interpolate_linear(FBFF)) : jlpDyn.comp_genericGainComputer_nCh(strength*(1+(((FBFF*-1)+1)*1)),thresh,att,rel,hld,rms,knee,link,N))
,si.bus(N)
)
:(ro.interleave(N,2):par(i,N,meter*@(max(0,floor(0.5+ma.SR*lad)))))
)~si.bus(N)
);
comp_HybridComp_nCh(strength,thresh,att,rel,hld,rms,knee,lad,link,FBFF,meter,N) =
si.bus(N) <: si.bus(N*2) :
(
(
(( (jlpDyn.comp_genericGainComputer_nCh(strength,thresh,att,rel,hld,0.005,knee,0,N),jlpDyn.comp_genericGainComputer_nCh(strength,thresh,att,rel,hld,rms,knee,link,N)) : ro.interleave(N,2) : par(i, N, it.interpolate_linear(FBFF))),si.bus(N))
: (ro.interleave(N,2) : par(i,N,meter*delay(lad)))
)~si.bus(N)
);
comp_HybridComp_nCh2(strength,thresh,att,rel,hld,rms,knee,lad,link,FBFF,meter,N) =
si.bus(N) <: si.bus(N*2) : ((si.bus(N) <: si.bus(N*2)),(jlpDyn.crestFactorComputer_nCh(rms,link,N))):(si.bus(N*2),si.block(N)) :
(
(
(( (jlpDyn.comp_genericGainComputer_nCh(strength,thresh,att,rel,hld,0.005,knee,0,N),jlpDyn.comp_genericGainComputer_nCh(strength,thresh,att,rel,hld,rms,knee,link,N)) : ro.interleave(N,2) : par(i, N, it.interpolate_linear(FBFF))),si.bus(N))
: (ro.interleave(N,2) : par(i,N,meter*delay(lad)))
)~si.bus(N)
);
process = comp_HybridComp_nCh(strength,thresh,att,rel,hld,rms,knee,lad,0,1,meter,1)
//process = jlpDyn.crestFactorComputer_nCh(1,0,0,2)
with{
strength = hslider("Strenght", 0, 0, 1, 0.01);
thresh = hslider("Threshold", 0, -96, 0, 0.1);
att = hslider("Attack", 0, 0, 120, 0.1)/1000;
rel = hslider("Release", 0, 0, 3000, 0.1)/1000;
hld = hslider("Hold", 0, 0, 3000, 0.1)/1000;
rms = hslider("RMS Size", 0, 0, 3000, 0.1)/1000;
knee = hslider("Knee", 0, 0, 24, 0.1);
lad = ba.if(checkbox("Perfect attack"),att, 0);//hslider("Look Ahead", 0, 0, 120, 0.1);
meter = _<:(_, (ba.linear2db:max(maxGR):(hbargraph("GainReduction[1][unit:dB][tooltip: gain reduction in dB]", maxGR, 0)))):attach;
maxGR = 0;
}; | https://raw.githubusercontent.com/jpecquais/faustLab/822700f7f05a3a033f22bc3eaebd3640904ab703/dsp/Dynamic/jlpCompressor.dsp | faust | TODO : TO BE IMPLEMENTED IN A PROPER LIB
TODO : FORCE FEEDFORWARD WHEN LAH IS ON
process = jlpDyn.crestFactorComputer_nCh(1,0,0,2)
hslider("Look Ahead", 0, 0, 120, 0.1); | import("lib/jlpLibs.lib");
import("stdfaust.lib");
declare name "Compressor";
declare author "Jean-Loup Pecquais";
declare version "1.00";
declare license "GPL3";
delay(time) = @(max(0,floor(0.5+ma.SR*time)));
comp_BigBrother_nCh(strength,thresh,att,rel,hld,rms,knee,lad,link,FBFF,meter,N) =
si.bus(N) <: si.bus(N*2):
(
(
(
(ro.interleave(N,2):par(i, N*2, abs) :par(i, N, it.interpolate_linear(FBFF)) : jlpDyn.comp_genericGainComputer_nCh(strength*(1+(((FBFF*-1)+1)*1)),thresh,att,rel,hld,rms,knee,link,N))
,si.bus(N)
)
:(ro.interleave(N,2):par(i,N,meter*@(max(0,floor(0.5+ma.SR*lad)))))
)~si.bus(N)
);
comp_HybridComp_nCh(strength,thresh,att,rel,hld,rms,knee,lad,link,FBFF,meter,N) =
si.bus(N) <: si.bus(N*2) :
(
(
(( (jlpDyn.comp_genericGainComputer_nCh(strength,thresh,att,rel,hld,0.005,knee,0,N),jlpDyn.comp_genericGainComputer_nCh(strength,thresh,att,rel,hld,rms,knee,link,N)) : ro.interleave(N,2) : par(i, N, it.interpolate_linear(FBFF))),si.bus(N))
: (ro.interleave(N,2) : par(i,N,meter*delay(lad)))
)~si.bus(N)
);
comp_HybridComp_nCh2(strength,thresh,att,rel,hld,rms,knee,lad,link,FBFF,meter,N) =
si.bus(N) <: si.bus(N*2) : ((si.bus(N) <: si.bus(N*2)),(jlpDyn.crestFactorComputer_nCh(rms,link,N))):(si.bus(N*2),si.block(N)) :
(
(
(( (jlpDyn.comp_genericGainComputer_nCh(strength,thresh,att,rel,hld,0.005,knee,0,N),jlpDyn.comp_genericGainComputer_nCh(strength,thresh,att,rel,hld,rms,knee,link,N)) : ro.interleave(N,2) : par(i, N, it.interpolate_linear(FBFF))),si.bus(N))
: (ro.interleave(N,2) : par(i,N,meter*delay(lad)))
)~si.bus(N)
);
process = comp_HybridComp_nCh(strength,thresh,att,rel,hld,rms,knee,lad,0,1,meter,1)
with{
strength = hslider("Strenght", 0, 0, 1, 0.01);
thresh = hslider("Threshold", 0, -96, 0, 0.1);
att = hslider("Attack", 0, 0, 120, 0.1)/1000;
rel = hslider("Release", 0, 0, 3000, 0.1)/1000;
hld = hslider("Hold", 0, 0, 3000, 0.1)/1000;
rms = hslider("RMS Size", 0, 0, 3000, 0.1)/1000;
knee = hslider("Knee", 0, 0, 24, 0.1);
meter = _<:(_, (ba.linear2db:max(maxGR):(hbargraph("GainReduction[1][unit:dB][tooltip: gain reduction in dB]", maxGR, 0)))):attach;
maxGR = 0;
}; |
fde506e545978ab239094cb5bbf07eee0a87667a75d8c41e87aca800429c1571 | olegkapitonov/Kapitonov-Plugins-Pack | kpp_single2humbucker.dsp | /*
* Copyright (C) 2018-2020 Oleg Kapitonov
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* --------------------------------------------------------------------------
*/
/*
* This pugin is very primitive imitator of humbucker sound (Les Paul)
* for single-coiled guitars (Stratocaster).
* Good for playing hard rock and metal with Stratocaster.
* Probably bad for playing blues :))
*
*/
declare name "kpp_single2humbucker";
declare author "Oleg Kapitonov";
declare license "GPLv3";
declare version "1.2";
import("stdfaust.lib");
delay_samples = ma.SR / 2880 / 2;
process = output with {
effect_knob = vslider("Humbuckerize", 1, 0, 1, 0.001);
filter_knob = vslider("Bass Cut", 20, 20, 720, 0.001);
effect = fi.highpass(1,20)
<: _, de.delay(50, delay_samples) :
+ : fi.lowpass(2, 5500) : fi.peak_eq(6.0, 550, 750);
filter = fi.highpass(1,filter_knob);
output = _,_ :> _ <: (*(effect_knob) : effect), (*(1.0 - effect_knob)) : + :
filter : *(ba.db2linear(-10.0)) <: _,_ ;
};
| https://raw.githubusercontent.com/olegkapitonov/Kapitonov-Plugins-Pack/ed4541172d53ecf04bad43cd583365f278ccf176/LV2/kpp_single2humbucker/kpp_single2humbucker.dsp | faust |
* Copyright (C) 2018-2020 Oleg Kapitonov
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* --------------------------------------------------------------------------
* This pugin is very primitive imitator of humbucker sound (Les Paul)
* for single-coiled guitars (Stratocaster).
* Good for playing hard rock and metal with Stratocaster.
* Probably bad for playing blues :))
*
|
declare name "kpp_single2humbucker";
declare author "Oleg Kapitonov";
declare license "GPLv3";
declare version "1.2";
import("stdfaust.lib");
delay_samples = ma.SR / 2880 / 2;
process = output with {
effect_knob = vslider("Humbuckerize", 1, 0, 1, 0.001);
filter_knob = vslider("Bass Cut", 20, 20, 720, 0.001);
effect = fi.highpass(1,20)
<: _, de.delay(50, delay_samples) :
+ : fi.lowpass(2, 5500) : fi.peak_eq(6.0, 550, 750);
filter = fi.highpass(1,filter_knob);
output = _,_ :> _ <: (*(effect_knob) : effect), (*(1.0 - effect_knob)) : + :
filter : *(ba.db2linear(-10.0)) <: _,_ ;
};
|
fe603ee64f6dd5b3a0321e101fe391a7c9c25654dd6db586dfc1e00eb36cd9aa | olegkapitonov/Kapitonov-Plugins-Pack | kpp_single2humbucker.dsp | /*
* Copyright (C) 2018-2020 Oleg Kapitonov
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* --------------------------------------------------------------------------
*/
/*
* This pugin is very primitive imitator of humbucker sound (Les Paul)
* for single-coiled guitars (Stratocaster).
* Good for playing hard rock and metal with Stratocaster.
* Probably bad for playing blues :))
*
*/
declare name "kpp_single2humbucker";
declare author "Oleg Kapitonov";
declare license "GPLv3";
declare version "1.1";
import("stdfaust.lib");
delay_samples = ma.SR / 2880 / 2;
process = output with {
effect_knob = vslider("Humbuckerize", 0, 0, 1, 0.001);
filter_knob = vslider("Bass Cut", 20, 20, 720, 0.001);
effect = fi.highpass(1,20)
<: _, de.delay(50, delay_samples) :
+ : fi.lowpass(2, 5500) : fi.peak_eq(6.0, 550, 750);
filter = fi.highpass(1,filter_knob);
output = _ <: (*(effect_knob) : effect), (*(1.0 - effect_knob)) : + :
filter : *(ba.db2linear(-10.0)) : _ ;
};
| https://raw.githubusercontent.com/olegkapitonov/Kapitonov-Plugins-Pack/ed4541172d53ecf04bad43cd583365f278ccf176/LADSPA/kpp_single2humbucker/kpp_single2humbucker.dsp | faust |
* Copyright (C) 2018-2020 Oleg Kapitonov
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* --------------------------------------------------------------------------
* This pugin is very primitive imitator of humbucker sound (Les Paul)
* for single-coiled guitars (Stratocaster).
* Good for playing hard rock and metal with Stratocaster.
* Probably bad for playing blues :))
*
|
declare name "kpp_single2humbucker";
declare author "Oleg Kapitonov";
declare license "GPLv3";
declare version "1.1";
import("stdfaust.lib");
delay_samples = ma.SR / 2880 / 2;
process = output with {
effect_knob = vslider("Humbuckerize", 0, 0, 1, 0.001);
filter_knob = vslider("Bass Cut", 20, 20, 720, 0.001);
effect = fi.highpass(1,20)
<: _, de.delay(50, delay_samples) :
+ : fi.lowpass(2, 5500) : fi.peak_eq(6.0, 550, 750);
filter = fi.highpass(1,filter_knob);
output = _ <: (*(effect_knob) : effect), (*(1.0 - effect_knob)) : + :
filter : *(ba.db2linear(-10.0)) : _ ;
};
|
5832966bcfbb29a79a1bdff9a5bb9216027ab0fdac1c70997aff95c7da04870b | sebastien-clara/panoplie | tube12AX7.dsp | declare name "Tube Amp Emulation 12AX7";
declare author "Guitarix";
import("stdfaust.lib");
process = component("tubes.lib").T1_12AX7 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T2_12AX7 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T3_12AX7 : *(gain)
with {
preamp = vslider("Pregain",-6,-20,20,0.1) : ba.db2linear : si.smooth(0.999);
gain = vslider("Gain", -6, -20.0, 20.0, 0.1) : ba.db2linear : si.smooth(0.999);
}; | https://raw.githubusercontent.com/sebastien-clara/panoplie/bfb061ab2874a404826a2d62a5359dcd92264f30/audioFX/ampli/tube12AX7%7E.mxo/tube12AX7.dsp | faust | declare name "Tube Amp Emulation 12AX7";
declare author "Guitarix";
import("stdfaust.lib");
process = component("tubes.lib").T1_12AX7 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T2_12AX7 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T3_12AX7 : *(gain)
with {
preamp = vslider("Pregain",-6,-20,20,0.1) : ba.db2linear : si.smooth(0.999);
gain = vslider("Gain", -6, -20.0, 20.0, 0.1) : ba.db2linear : si.smooth(0.999);
}; |
|
97a6569ff7018bbe7d695010da4f649624f54556c900a185f32d04c6fe39b37b | sebastien-clara/panoplie | tube6C16.dsp | declare name "Tube Amp Emulation 6C16";
declare author "Guitarix";
import("stdfaust.lib");
process = component("tubes.lib").T1_6C16 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T2_6C16 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T3_6C16 : *(gain)
with {
preamp = vslider("Pregain",-6,-20,20,0.1) : ba.db2linear : si.smooth(0.999);
gain = vslider("Gain", -6, -20.0, 20.0, 0.1) : ba.db2linear : si.smooth(0.999);
}; | https://raw.githubusercontent.com/sebastien-clara/panoplie/bfb061ab2874a404826a2d62a5359dcd92264f30/audioFX/ampli/tube6C16%7E.mxo/tube6C16.dsp | faust | declare name "Tube Amp Emulation 6C16";
declare author "Guitarix";
import("stdfaust.lib");
process = component("tubes.lib").T1_6C16 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T2_6C16 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T3_6C16 : *(gain)
with {
preamp = vslider("Pregain",-6,-20,20,0.1) : ba.db2linear : si.smooth(0.999);
gain = vslider("Gain", -6, -20.0, 20.0, 0.1) : ba.db2linear : si.smooth(0.999);
}; |
|
28aadfb46a0235c289f97ddf2a8fac01cb397010e5174a08cee9d1e5079023bf | sebastien-clara/panoplie | tube6DJ8.dsp | declare name "Tube Amp Emulation 6DJ8";
declare author "Guitarix";
import("stdfaust.lib");
process = component("tubes.lib").T1_6DJ8 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T2_6DJ8 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T3_6DJ8 : *(gain)
with {
preamp = vslider("Pregain",-6,-20,20,0.1) : ba.db2linear : si.smooth(0.999);
gain = vslider("Gain", -6, -20.0, 20.0, 0.1) : ba.db2linear : si.smooth(0.999);
}; | https://raw.githubusercontent.com/sebastien-clara/panoplie/bfb061ab2874a404826a2d62a5359dcd92264f30/audioFX/ampli/tube6DJ8%7E.mxo/tube6DJ8.dsp | faust | declare name "Tube Amp Emulation 6DJ8";
declare author "Guitarix";
import("stdfaust.lib");
process = component("tubes.lib").T1_6DJ8 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T2_6DJ8 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T3_6DJ8 : *(gain)
with {
preamp = vslider("Pregain",-6,-20,20,0.1) : ba.db2linear : si.smooth(0.999);
gain = vslider("Gain", -6, -20.0, 20.0, 0.1) : ba.db2linear : si.smooth(0.999);
}; |
|
aa328a55da06c3cbd31b84cb6faad643f179823cc2cf65dc3bcdc8de487b1ea9 | sebastien-clara/panoplie | tube12AT7.dsp | declare name "Tube Amp Emulation 12AT7";
declare author "Guitarix";
import("stdfaust.lib");
process = component("tubes.lib").T1_12AT7 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T2_12AT7 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T3_12AT7 : *(gain)
with {
preamp = vslider("Pregain",-6,-20,20,0.1) : ba.db2linear : si.smooth(0.999);
gain = vslider("Gain", -6, -20.0, 20.0, 0.1) : ba.db2linear : si.smooth(0.999);
}; | https://raw.githubusercontent.com/sebastien-clara/panoplie/bfb061ab2874a404826a2d62a5359dcd92264f30/audioFX/ampli/tube12AT7%7E.mxo/tube12AT7.dsp | faust | declare name "Tube Amp Emulation 12AT7";
declare author "Guitarix";
import("stdfaust.lib");
process = component("tubes.lib").T1_12AT7 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T2_12AT7 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T3_12AT7 : *(gain)
with {
preamp = vslider("Pregain",-6,-20,20,0.1) : ba.db2linear : si.smooth(0.999);
gain = vslider("Gain", -6, -20.0, 20.0, 0.1) : ba.db2linear : si.smooth(0.999);
}; |
|
9657d3a413e9c9d2b03cf534ec3216e9068529e5ad91d2e150c4bf005ff0f2a6 | sebastien-clara/panoplie | tube12AU7.dsp | declare name "Tube Amp Emulation 12AU7";
declare author "Guitarix";
import("stdfaust.lib");
process = component("tubes.lib").T1_12AU7 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T2_12AU7 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T3_12AU7 : *(gain)
with {
preamp = vslider("Pregain",-6,-20,20,0.1) : ba.db2linear : si.smooth(0.999);
gain = vslider("Gain", -6, -20.0, 20.0, 0.1) : ba.db2linear : si.smooth(0.999);
}; | https://raw.githubusercontent.com/sebastien-clara/panoplie/bfb061ab2874a404826a2d62a5359dcd92264f30/audioFX/ampli/tube12AU7%7E.mxo/tube12AU7.dsp | faust | declare name "Tube Amp Emulation 12AU7";
declare author "Guitarix";
import("stdfaust.lib");
process = component("tubes.lib").T1_12AU7 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T2_12AU7 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T3_12AU7 : *(gain)
with {
preamp = vslider("Pregain",-6,-20,20,0.1) : ba.db2linear : si.smooth(0.999);
gain = vslider("Gain", -6, -20.0, 20.0, 0.1) : ba.db2linear : si.smooth(0.999);
}; |
|
80d098c8c4b62876e07d41d2116adcf6080c796f819f63bdeeb156445e58c90a | sebastien-clara/panoplie | tube6V6.dsp | declare name "Tube Amp Emulation 6V6";
declare author "Guitarix";
import("stdfaust.lib");
process = component("tubes.lib").T1_6V6 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T2_6V6 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T3_6V6 : *(gain)
with {
preamp = vslider("Pregain",-6,-20,20,0.1) : ba.db2linear : si.smooth(0.999);
gain = vslider("Gain", -6, -20.0, 20.0, 0.1) : ba.db2linear : si.smooth(0.999);
}; | https://raw.githubusercontent.com/sebastien-clara/panoplie/bfb061ab2874a404826a2d62a5359dcd92264f30/audioFX/ampli/tube6V6%7E.mxo/tube6V6.dsp | faust | declare name "Tube Amp Emulation 6V6";
declare author "Guitarix";
import("stdfaust.lib");
process = component("tubes.lib").T1_6V6 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T2_6V6 : *(preamp):
fi.lowpass(1,6531.0) : component("tubes.lib").T3_6V6 : *(gain)
with {
preamp = vslider("Pregain",-6,-20,20,0.1) : ba.db2linear : si.smooth(0.999);
gain = vslider("Gain", -6, -20.0, 20.0, 0.1) : ba.db2linear : si.smooth(0.999);
}; |
|
ce551794d4c9ae72253d48e5b7106551c09e2307f4d361cc8fdda0a7e8639f4f | rottingsounds/bitDSP-faust | bitDAC.dsp | declare name "bitDAC";
declare description "bitDAC - example";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit32 = library("bitDSP_int32.lib");
// plot
// CXXFLAGS="-I ../include" faust2csvplot -I ../lib bitDAC.dsp
// ./bitDAC -n 10
// compile
// CXXFLAGS="-I ../../../include" faust2caqt -I ../lib bitDAC.dsp
// ./bitDAC
dac_bits = int(hslider("dac bits",1,1,32,1));
dac_offset = min(int(hslider("dac offset",0,0,31,1)), 32-dac_bits);
// counting samples from start of execution, starting at 0
sample_count = int((1:+~_)) - 1;
// select bits of the samplecount and interpret that chunk as a PCM value
process = bit32.bitDAC(dac_offset, dac_bits, sample_count) <: _,_; | https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/21cf36105c55b6e18969a867a319530a0ef1ea63/examples/bitDAC.dsp | faust | plot
CXXFLAGS="-I ../include" faust2csvplot -I ../lib bitDAC.dsp
./bitDAC -n 10
compile
CXXFLAGS="-I ../../../include" faust2caqt -I ../lib bitDAC.dsp
./bitDAC
counting samples from start of execution, starting at 0
select bits of the samplecount and interpret that chunk as a PCM value | declare name "bitDAC";
declare description "bitDAC - example";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit32 = library("bitDSP_int32.lib");
dac_bits = int(hslider("dac bits",1,1,32,1));
dac_offset = min(int(hslider("dac offset",0,0,31,1)), 32-dac_bits);
sample_count = int((1:+~_)) - 1;
process = bit32.bitDAC(dac_offset, dac_bits, sample_count) <: _,_; |
eb0cbc77b590a52c27a1289eecf9d3f33b04a9aba3a901e796f43aa2250abca6 | pingdynasty/OwlPatches | Qompander.dsp | declare name "qompander";
declare version "1.2";
declare author "Bart Brouns";
declare license "GNU 3.0";
declare copyright "(c) Bart Brouns 2014";
declare credits "ported from qompander in pd by Katja Vetter";
declare see "http://www.katjaas.nl/compander/compander.html";
declare additional "filter coefficients by Olli Niemitalo";
//-----------------------------------------------
// imports
//-----------------------------------------------
import("stdfaust.lib");
//-----------------------------------------------
// the GUI
//-----------------------------------------------
qompanderGroup(x) = (vgroup("[0] qompander [tooltip: Reference: http://www.katjaas.nl/compander/compander.html]", x));
factor = qompanderGroup(hslider("[0] Factor[unit:: 1][style:knob][OWL:PARAMETER_A]", 3, 0.8, 8, 0.01):si.smooth(0.999));
threshold = qompanderGroup(hslider("[1] Threshold [unit: dB][style:knob][OWL:PARAMETER_B]", -40, -96, -20, 0.01):si.smooth(0.999));
attack = qompanderGroup(hslider("[2] Attack[unit: ms][style:knob][OWL:PARAMETER_C]", 1, 1, 20, 0.01):si.smooth(0.999));
release = qompanderGroup(hslider("[3] Release[unit: ms][style:knob][OWL:PARAMETER_D]", 20, 20, 1000, 0.01):si.smooth(0.999));
//-----------------------------------------------
// the DSP
//-----------------------------------------------
magnitude = (threshold):ba.db2linear;
exponent = log(magnitude)/log(sin(factor*magnitude*ma.PI/2));
// to go from puredata biquad coefficients to max/msp and faust notation: the first two parameters are negated and put last
olli1(x) = x: fi.tf2(0.161758, 0, -1, 0, -0.161758):fi.tf2(0.733029, 0, -1, 0, -0.733029):fi.tf2(0.94535 , 0, -1, 0, -0.94535 ):fi.tf2(0.990598, 0, -1, 0, -0.990598);
olli2(x) = x:mem: fi.tf2(0.479401, 0, -1, 0, -0.479401):fi.tf2(0.876218, 0, -1, 0, -0.876218):fi.tf2(0.976599, 0, -1, 0, -0.976599):fi.tf2(0.9975 , 0, -1, 0, -0.9975 );
pyth(x) = sqrt((olli1(x)*olli1(x))+(olli2(x)*olli2(x))):max(0.00001):min(100); //compute instantaneous amplitudes
attackDecay(x) = pyth(x) :an.amp_follower_ud(attack/1000,release/1000);
mapping(x) = attackDecay(x) : ((sin((min(1/factor)*(factor/4)) * (2*ma.PI)): max(0.0000001):min(1),exponent) : pow );
qompander(x) = (mapping(x) / attackDecay(x))<: (_,olli1(x):*),(_,olli2(x):*):+:_*(sqrt(0.5));
process(x) = qompander(x);
| https://raw.githubusercontent.com/pingdynasty/OwlPatches/2be8a65bb257b53ee7ee0b9d4b5a1ad249e16dab/Faust/Qompander.dsp | faust | -----------------------------------------------
imports
-----------------------------------------------
-----------------------------------------------
the GUI
-----------------------------------------------
-----------------------------------------------
the DSP
-----------------------------------------------
to go from puredata biquad coefficients to max/msp and faust notation: the first two parameters are negated and put last
compute instantaneous amplitudes | declare name "qompander";
declare version "1.2";
declare author "Bart Brouns";
declare license "GNU 3.0";
declare copyright "(c) Bart Brouns 2014";
declare credits "ported from qompander in pd by Katja Vetter";
declare see "http://www.katjaas.nl/compander/compander.html";
declare additional "filter coefficients by Olli Niemitalo";
import("stdfaust.lib");
qompanderGroup(x) = (vgroup("[0] qompander [tooltip: Reference: http://www.katjaas.nl/compander/compander.html]", x));
factor = qompanderGroup(hslider("[0] Factor[unit:: 1][style:knob][OWL:PARAMETER_A]", 3, 0.8, 8, 0.01):si.smooth(0.999));
threshold = qompanderGroup(hslider("[1] Threshold [unit: dB][style:knob][OWL:PARAMETER_B]", -40, -96, -20, 0.01):si.smooth(0.999));
attack = qompanderGroup(hslider("[2] Attack[unit: ms][style:knob][OWL:PARAMETER_C]", 1, 1, 20, 0.01):si.smooth(0.999));
release = qompanderGroup(hslider("[3] Release[unit: ms][style:knob][OWL:PARAMETER_D]", 20, 20, 1000, 0.01):si.smooth(0.999));
magnitude = (threshold):ba.db2linear;
exponent = log(magnitude)/log(sin(factor*magnitude*ma.PI/2));
olli1(x) = x: fi.tf2(0.161758, 0, -1, 0, -0.161758):fi.tf2(0.733029, 0, -1, 0, -0.733029):fi.tf2(0.94535 , 0, -1, 0, -0.94535 ):fi.tf2(0.990598, 0, -1, 0, -0.990598);
olli2(x) = x:mem: fi.tf2(0.479401, 0, -1, 0, -0.479401):fi.tf2(0.876218, 0, -1, 0, -0.876218):fi.tf2(0.976599, 0, -1, 0, -0.976599):fi.tf2(0.9975 , 0, -1, 0, -0.9975 );
attackDecay(x) = pyth(x) :an.amp_follower_ud(attack/1000,release/1000);
mapping(x) = attackDecay(x) : ((sin((min(1/factor)*(factor/4)) * (2*ma.PI)): max(0.0000001):min(1),exponent) : pow );
qompander(x) = (mapping(x) / attackDecay(x))<: (_,olli1(x):*),(_,olli2(x):*):+:_*(sqrt(0.5));
process(x) = qompander(x);
|
562cea5e761efa826f2f3cfdc47a399776cc2adc3fd7a3c21beeb7daa902bc6c | magnetophon/faustExperiments | slidingOp.dsp | declare name "LazyLeveler";
declare version "0.1";
declare author "Bart Brouns";
declare license "GPLv3";
import("stdfaust.lib");
// N = hslider("maxN", 0, 0, maxN, 1);
// maxN = pow(2,4);
N = maxN;
maxN = 14;
process =
// pow(2,32);
// maxn;
slidingMax(N,maxN);
// ba.slidingMax(N,maxN);
slidingMin(n,maxn) = slidingReduce(min,n,maxn,ma.INFINITY);
slidingMax(n,maxn) = slidingReduce(max,n,maxn,0-ma.INFINITY);
slidingReduce(op,N,0,disabledVal) = _;
slidingReduce(op,N,maxN,disabledVal) =
sequentialOperatorParOut(maxNrBits(maxN+1)-1,op)
:
par(i,maxNrBits(maxN+1)
, _@sumOfPrevBlockSizes(i)
: useVal(i)
) : combine(maxNrBits(maxN+1))
with {
sequentialOperatorParOut(N,op) =
seq(i, N, operator(i));
operator(i) =
myBus(i)
,
(_<:
_ , op(_,_@(pow2(i) ))
) ;
// todo:
myBus(0) = 0:! ;
myBus(i) = si.bus(i);
// The sum of all the sizes of the previous blocks
sumOfPrevBlockSizes(0) = 0;
sumOfPrevBlockSizes(i) = (ba.subseq((allBlockSizes),0,i):>_);
allBlockSizes = par(i, maxNrBits(maxN), (pow2(i)) * isUsed(i));
maxNrBits(n) = int2nrOfBits(n);
// Apply <op> to <N> parallel input signals
combine(2) = op;
combine(N) = op(combine(N-1),_);
// Decide wether or not to use a certain value, based on N
useVal(i) =
select2(isUsed(i), disabledVal, _)
;
isUsed(i) = ba.take(i+1, (int2bin(N+1,maxN*2+1)));
pow2(i) = 1<<i;
// same as:
// pow2(i) = int(pow(2,i));
// but in the block diagram, it will be displayed as a number, instead of a formula
// convert N into a list of ones and zeros
int2bin(N,maxN) = par(j, maxNrBits(maxN), int(floor((N)/(pow2(j))))%2);
// calculate how many ones and zeros are needed to represent maxN
int2nrOfBits(N) = int(floor(log(N)/log(2))+1);
};
| https://raw.githubusercontent.com/magnetophon/faustExperiments/0cb3f5027fc4350b3b448e51e2b8515107fc5e64/slidingOp.dsp | faust | N = hslider("maxN", 0, 0, maxN, 1);
maxN = pow(2,4);
pow(2,32);
maxn;
ba.slidingMax(N,maxN);
todo:
The sum of all the sizes of the previous blocks
Apply <op> to <N> parallel input signals
Decide wether or not to use a certain value, based on N
same as:
pow2(i) = int(pow(2,i));
but in the block diagram, it will be displayed as a number, instead of a formula
convert N into a list of ones and zeros
calculate how many ones and zeros are needed to represent maxN | declare name "LazyLeveler";
declare version "0.1";
declare author "Bart Brouns";
declare license "GPLv3";
import("stdfaust.lib");
N = maxN;
maxN = 14;
process =
slidingMax(N,maxN);
slidingMin(n,maxn) = slidingReduce(min,n,maxn,ma.INFINITY);
slidingMax(n,maxn) = slidingReduce(max,n,maxn,0-ma.INFINITY);
slidingReduce(op,N,0,disabledVal) = _;
slidingReduce(op,N,maxN,disabledVal) =
sequentialOperatorParOut(maxNrBits(maxN+1)-1,op)
:
par(i,maxNrBits(maxN+1)
, _@sumOfPrevBlockSizes(i)
: useVal(i)
) : combine(maxNrBits(maxN+1))
with {
sequentialOperatorParOut(N,op) =
seq(i, N, operator(i));
operator(i) =
myBus(i)
,
(_<:
_ , op(_,_@(pow2(i) ))
) ;
myBus(0) = 0:! ;
myBus(i) = si.bus(i);
sumOfPrevBlockSizes(0) = 0;
sumOfPrevBlockSizes(i) = (ba.subseq((allBlockSizes),0,i):>_);
allBlockSizes = par(i, maxNrBits(maxN), (pow2(i)) * isUsed(i));
maxNrBits(n) = int2nrOfBits(n);
combine(2) = op;
combine(N) = op(combine(N-1),_);
useVal(i) =
select2(isUsed(i), disabledVal, _)
;
isUsed(i) = ba.take(i+1, (int2bin(N+1,maxN*2+1)));
pow2(i) = 1<<i;
int2bin(N,maxN) = par(j, maxNrBits(maxN), int(floor((N)/(pow2(j))))%2);
int2nrOfBits(N) = int(floor(log(N)/log(2))+1);
};
|
a15784adb5fa65f95d345569e58a710e5f8505718c1e1420f457119fe37d71f3 | friskgit/snares | filter_bank.dsp | // -*- compile-command: "cd .. && make jack src=src/filter_bank.dsp && cd -"; -*-
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
//---------------`Filterbank for snaredrum` --------------------------
//
// A filterbank for use with snare drum synths and channel disperser. It
// takes a trigger and a signal as input and outputs as many channels as
// there are filterbands.
//
// Parameter 'bands' is set at compile time.
//
// 18 Juli 2019 Henrik Frisk [email protected]
//---------------------------------------------------
bands = 16;
// Control the output channel
del_group(x) = vgroup("delay", x);
del = del_group(hslider("delay", 0, 0, 1024, 1));
fb = fi.mth_octave_filterbank(order,M,ftop,bands) : par(i, bands, (*(ba.db2linear(fader(bands-i))))) :
par(i, bands, de.sdelay(8192, 512, *(del, i)))
with {
M = 2;
order = 1;
ftop = 10000;
bp1 = ba.bypass1;
slider_group(x) = mofb_group(hgroup("[1]", x));
mofb_group(x) = vgroup("constant-q filter bank (Butterworth dyadic tree) [tooltip: See Faust's filters.lib for more info", x);
fader(i) = slider_group(vslider("Band%2i [unit:dB] [tooltip: Bandpass filter gain in dB]", -10, -70, 10, 0.1)) : si.smoo;
bp = bypass_group(checkbox("[0] Bypassc[tooltip: When this is checked, the filter-bank has no effect]"));
};
ch_wrapped(x) = ma.modulo((offset+x), bands);
offset = hslider("offset", 0, 0, bands, 1);
process(trig, sig) = sig : fb : par(i, bands, ba.selectoutn(bands, ch_wrapped(i))) :> par(i, bands, _);
| https://raw.githubusercontent.com/friskgit/snares/bb43ea5e706a0ead6d65dd176a5c492b2f5d8f74/faust/snare/src/filter_bank.dsp | faust | -*- compile-command: "cd .. && make jack src=src/filter_bank.dsp && cd -"; -*-
---------------`Filterbank for snaredrum` --------------------------
A filterbank for use with snare drum synths and channel disperser. It
takes a trigger and a signal as input and outputs as many channels as
there are filterbands.
Parameter 'bands' is set at compile time.
18 Juli 2019 Henrik Frisk [email protected]
---------------------------------------------------
Control the output channel |
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
bands = 16;
del_group(x) = vgroup("delay", x);
del = del_group(hslider("delay", 0, 0, 1024, 1));
fb = fi.mth_octave_filterbank(order,M,ftop,bands) : par(i, bands, (*(ba.db2linear(fader(bands-i))))) :
par(i, bands, de.sdelay(8192, 512, *(del, i)))
with {
M = 2;
order = 1;
ftop = 10000;
bp1 = ba.bypass1;
slider_group(x) = mofb_group(hgroup("[1]", x));
mofb_group(x) = vgroup("constant-q filter bank (Butterworth dyadic tree) [tooltip: See Faust's filters.lib for more info", x);
fader(i) = slider_group(vslider("Band%2i [unit:dB] [tooltip: Bandpass filter gain in dB]", -10, -70, 10, 0.1)) : si.smoo;
bp = bypass_group(checkbox("[0] Bypassc[tooltip: When this is checked, the filter-bank has no effect]"));
};
ch_wrapped(x) = ma.modulo((offset+x), bands);
offset = hslider("offset", 0, 0, bands, 1);
process(trig, sig) = sig : fb : par(i, bands, ba.selectoutn(bands, ch_wrapped(i))) :> par(i, bands, _);
|
5328da9f9f7d7745808e015b19e8b04d4d9e758f87885d0388de01ecfe58d5ee | friskgit/snares | snares.dsp | // -*- compile-command: "cd .. && make jack src=src/snares.dsp && cd -"; -*-
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
//---------------`Snare drum synth` --------------------------
// A take at a snare drum synth
//
// Continuously playing, optionally multichannel dispersing (see focus parameter).
//
// Where:
// * midi note 67-89
// * stiffness 0-0.55 (mapped to note as in note 67 -> 0)§
// * midi velocity is mapped to pressure
// A useful parameter setting is:
//
// 30 Juni 2018 Henrik Frisk [email protected]
//---------------------------------------------------
channels = 14;
//imp = ba.pulse(hslider("tempo", 10000, 500, 10000, 1));
imp = ba.beat(hslider("tempo", 100, 1, 2000, 1));
master_env = 1; //en.smoothEnvelope(0.1, button("play"));
env = en.ar(attack, rel, imp) * amp
with {
attack = hslider("attack", 0.00000001, 0, 0.1, 0.000000001) : si.smooth(0.1);
rel = hslider("rel", 0.1, 0.0000001, 0.5, 0.0000001) : si.smooth(0.2);
amp = hslider("vol", 0.5, 0, 1, 0.0001);
};
// Control the output channel
focus = hslider("focus", 1, 0, 1, 0.0001);
position = hslider("position", 1, 0, channels, 1);
rate = ma.SR/1000.0;
rndctrl = (no.lfnoise(rate) * (channels + 1)) * focus : ma.fabs + position : int ;
outputctrl = rndctrl : ba.sAndH(imp);
// Filter
n = no.multinoise(8) : par(i, 8, _ * env * 0.1);
filt = fi.resonbp(frq, q, gain)
with {
frq = hslider("freq", 200, 50, 5000, 0.1);
q = hslider("q", 1, 0.01, 10, 0.01);
gain = hslider("gain", 0, 0, 2, 0.00001);
};
// Wrap channels
ch_wrapped = ma.modulo(outputctrl, channels);
//process = n : par(i, 8, filt);
process = n : par(i, 8, filt) :> _,_ :> *(_, master_env) :> ba.selectoutn(channels, ch_wrapped);
| https://raw.githubusercontent.com/friskgit/snares/bb43ea5e706a0ead6d65dd176a5c492b2f5d8f74/faust/snare/src/extras/snares.dsp | faust | -*- compile-command: "cd .. && make jack src=src/snares.dsp && cd -"; -*-
---------------`Snare drum synth` --------------------------
A take at a snare drum synth
Continuously playing, optionally multichannel dispersing (see focus parameter).
Where:
* midi note 67-89
* stiffness 0-0.55 (mapped to note as in note 67 -> 0)§
* midi velocity is mapped to pressure
A useful parameter setting is:
30 Juni 2018 Henrik Frisk [email protected]
---------------------------------------------------
imp = ba.pulse(hslider("tempo", 10000, 500, 10000, 1));
en.smoothEnvelope(0.1, button("play"));
Control the output channel
Filter
Wrap channels
process = n : par(i, 8, filt); |
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
channels = 14;
imp = ba.beat(hslider("tempo", 100, 1, 2000, 1));
env = en.ar(attack, rel, imp) * amp
with {
attack = hslider("attack", 0.00000001, 0, 0.1, 0.000000001) : si.smooth(0.1);
rel = hslider("rel", 0.1, 0.0000001, 0.5, 0.0000001) : si.smooth(0.2);
amp = hslider("vol", 0.5, 0, 1, 0.0001);
};
focus = hslider("focus", 1, 0, 1, 0.0001);
position = hslider("position", 1, 0, channels, 1);
rate = ma.SR/1000.0;
rndctrl = (no.lfnoise(rate) * (channels + 1)) * focus : ma.fabs + position : int ;
outputctrl = rndctrl : ba.sAndH(imp);
n = no.multinoise(8) : par(i, 8, _ * env * 0.1);
filt = fi.resonbp(frq, q, gain)
with {
frq = hslider("freq", 200, 50, 5000, 0.1);
q = hslider("q", 1, 0.01, 10, 0.01);
gain = hslider("gain", 0, 0, 2, 0.00001);
};
ch_wrapped = ma.modulo(outputctrl, channels);
process = n : par(i, 8, filt) :> _,_ :> *(_, master_env) :> ba.selectoutn(channels, ch_wrapped);
|
a5feefe76a60641267dc096bf0796540aecf3647d408f5ed49474f4eac74c312 | dariosanfilippo/modified_hindmarsh-rose | modified_hindmarshrose.dsp | // =============================================================================
// Modified Hindmarsh-Rose complex generator
// =============================================================================
//
// Complex sound generator based on modified Hindmarsh-Rose equations.
// The model is structurally-stable through hyperbolic tangent function
// saturators and allows for parameters in unstable ranges to explore
// different dynamics. Furthermore, this model includes DC-blockers in the
// feedback paths to counterbalance a tendency towards fixed-point attractors
// – thus enhancing complex behaviours – and obtain signals suitable for audio.
// Besides the original parameters in the model, this system includes a
// saturating threshold determining the positive and negative bounds in the
// equations, while the output peaks are within the [-1.0; 1.0] range.
//
// The system can be triggered by an impulse or by a constant of arbitrary
// values for deterministic and reproducable behaviours. Alternatively,
// the oscillator can be fed with external inputs to be used as a nonlinear
// distortion unit.
//
// =============================================================================
import("stdfaust.lib");
declare name "Modified Hindmarsh-Rose complex generator";
declare author "Dario Sanfilippo";
declare copyright "Copyright (C) 2021 Dario Sanfilippo
<[email protected]>";
declare version "1.1";
declare license "GPL v3.0 license";
// Hindmarsh–Rose model
// dx/dt=y+F(x)-z+I
// dy/dt=G(x)-y
// dz/dt=r(s(x-x_r)-z)
// F(x)=-ax^3+bx^2
// G(x)c-dx^2
hindmarshrose(l, a, b, I, c, d, r, s, x_R, dt, x_0, y_0, z_0) =
x_level(out * (x / l)) ,
y_level(out * (y / l)) ,
z_level(out * (z / l))
letrec {
'x = fi.highpass(1, 10, tanh(l, (x_0 + x + (y + phi(x) - z + I) * dt)));
'y = fi.highpass(1, 10, tanh(l, (y_0 + y + (psi(x) - y) * dt)));
'z = fi.highpass(1, 10, tanh(l, (z_0 + z + (r * (s * (x - x_R) - z))
* dt)));
}
with {
phi(x) = b * x ^ 2.0 - a * x ^ 3.0;
psi(x) = c - d * x ^ 2.0;
};
// smoothing function for click-free parameter variations using
// a one-pole low-pass with a 20-Hz cut-off frequency.
smooth(x) = fi.pole(pole, x * (1.0 - pole))
with {
pole = exp(-2.0 * ma.PI * 20.0 / ma.SR);
};
// tanh() saturator with adjustable saturating threshold
tanh(l, x) = l * ma.tanh(x / l);
// GUI parameters
x_level(x) = attach(x , abs(x) : ba.linear2db :
levels_group(hbargraph("[5]x[style:dB]", -60, 0)));
y_level(x) = attach(x , abs(x) : ba.linear2db :
levels_group(hbargraph("[6]y[style:dB]", -60, 0)));
z_level(x) = attach(x , abs(x) : ba.linear2db :
levels_group(hbargraph("[7]z[style:dB]", -60, 0)));
x_param_group(x) = vgroup("[0]x-parameters", x);
y_param_group(x) = vgroup("[1]y-parameters", x);
z_param_group(x) = vgroup("[2]z-parameters", x);
global_group(x) = vgroup("[3]Global", x);
levels_group(x) = hgroup("[4]Levels (dB)", x);
a = x_param_group(hslider("[1]a[scale:exp]", 1, 0, 20, .000001) : smooth);
b = x_param_group(hslider("[2]b[scale:exp]", 3, 0, 20, .000001) : smooth);
I = x_param_group(hslider("[3]I", 2, -20, 20, .000001) : smooth);
c = y_param_group(hslider("[4]c", 1, -20, 20, .000001) : smooth);
d = y_param_group(hslider("[5]d[scale:exp]", 5, 0, 20, .000001) : smooth);
r = z_param_group(hslider("[6]r[scale:exp]", 1, 0, 2, .000001) : smooth);
s = z_param_group(hslider("[7]s[scale:exp]", 4, 0, 20, .000001) : smooth);
x_r = z_param_group(hslider("[8]x_R", 1.6, -20, 20, .000001) : smooth);
dt = global_group(
hslider("[4]dt (integration step)[scale:exp]", 0.1, 0.000001, 1, .000001) :
smooth);
input(x) = global_group(nentry("[3]Input value", 1, 0, 10, .000001) <:
_ * impulse + _ * checkbox("[1]Constant inputs") +
x * checkbox("[0]External inputs"));
impulse = button("[2]Impulse inputs") : ba.impulsify;
limit = global_group(
hslider("[5]Saturation limit[scale:exp]", 4, 1, 64, .000001) : smooth);
out = global_group(hslider("[6]Output scaling[scale:exp]", 0, 0, 1, .000001) :
smooth);
process(x1, x2, x3) =
hindmarshrose(limit, a, b, I, c, d, r, s, x_r, dt, input(x1),
input(x2), input(x3));
| https://raw.githubusercontent.com/dariosanfilippo/modified_hindmarsh-rose/b19fa8bdd70aa59576980cfa2e01b145a0f939d2/modified_hindmarshrose.dsp | faust | =============================================================================
Modified Hindmarsh-Rose complex generator
=============================================================================
Complex sound generator based on modified Hindmarsh-Rose equations.
The model is structurally-stable through hyperbolic tangent function
saturators and allows for parameters in unstable ranges to explore
different dynamics. Furthermore, this model includes DC-blockers in the
feedback paths to counterbalance a tendency towards fixed-point attractors
– thus enhancing complex behaviours – and obtain signals suitable for audio.
Besides the original parameters in the model, this system includes a
saturating threshold determining the positive and negative bounds in the
equations, while the output peaks are within the [-1.0; 1.0] range.
The system can be triggered by an impulse or by a constant of arbitrary
values for deterministic and reproducable behaviours. Alternatively,
the oscillator can be fed with external inputs to be used as a nonlinear
distortion unit.
=============================================================================
Hindmarsh–Rose model
dx/dt=y+F(x)-z+I
dy/dt=G(x)-y
dz/dt=r(s(x-x_r)-z)
F(x)=-ax^3+bx^2
G(x)c-dx^2
smoothing function for click-free parameter variations using
a one-pole low-pass with a 20-Hz cut-off frequency.
tanh() saturator with adjustable saturating threshold
GUI parameters |
import("stdfaust.lib");
declare name "Modified Hindmarsh-Rose complex generator";
declare author "Dario Sanfilippo";
declare copyright "Copyright (C) 2021 Dario Sanfilippo
<[email protected]>";
declare version "1.1";
declare license "GPL v3.0 license";
hindmarshrose(l, a, b, I, c, d, r, s, x_R, dt, x_0, y_0, z_0) =
x_level(out * (x / l)) ,
y_level(out * (y / l)) ,
z_level(out * (z / l))
letrec {
'x = fi.highpass(1, 10, tanh(l, (x_0 + x + (y + phi(x) - z + I) * dt)));
'y = fi.highpass(1, 10, tanh(l, (y_0 + y + (psi(x) - y) * dt)));
'z = fi.highpass(1, 10, tanh(l, (z_0 + z + (r * (s * (x - x_R) - z))
* dt)));
}
with {
phi(x) = b * x ^ 2.0 - a * x ^ 3.0;
psi(x) = c - d * x ^ 2.0;
};
smooth(x) = fi.pole(pole, x * (1.0 - pole))
with {
pole = exp(-2.0 * ma.PI * 20.0 / ma.SR);
};
tanh(l, x) = l * ma.tanh(x / l);
x_level(x) = attach(x , abs(x) : ba.linear2db :
levels_group(hbargraph("[5]x[style:dB]", -60, 0)));
y_level(x) = attach(x , abs(x) : ba.linear2db :
levels_group(hbargraph("[6]y[style:dB]", -60, 0)));
z_level(x) = attach(x , abs(x) : ba.linear2db :
levels_group(hbargraph("[7]z[style:dB]", -60, 0)));
x_param_group(x) = vgroup("[0]x-parameters", x);
y_param_group(x) = vgroup("[1]y-parameters", x);
z_param_group(x) = vgroup("[2]z-parameters", x);
global_group(x) = vgroup("[3]Global", x);
levels_group(x) = hgroup("[4]Levels (dB)", x);
a = x_param_group(hslider("[1]a[scale:exp]", 1, 0, 20, .000001) : smooth);
b = x_param_group(hslider("[2]b[scale:exp]", 3, 0, 20, .000001) : smooth);
I = x_param_group(hslider("[3]I", 2, -20, 20, .000001) : smooth);
c = y_param_group(hslider("[4]c", 1, -20, 20, .000001) : smooth);
d = y_param_group(hslider("[5]d[scale:exp]", 5, 0, 20, .000001) : smooth);
r = z_param_group(hslider("[6]r[scale:exp]", 1, 0, 2, .000001) : smooth);
s = z_param_group(hslider("[7]s[scale:exp]", 4, 0, 20, .000001) : smooth);
x_r = z_param_group(hslider("[8]x_R", 1.6, -20, 20, .000001) : smooth);
dt = global_group(
hslider("[4]dt (integration step)[scale:exp]", 0.1, 0.000001, 1, .000001) :
smooth);
input(x) = global_group(nentry("[3]Input value", 1, 0, 10, .000001) <:
_ * impulse + _ * checkbox("[1]Constant inputs") +
x * checkbox("[0]External inputs"));
impulse = button("[2]Impulse inputs") : ba.impulsify;
limit = global_group(
hslider("[5]Saturation limit[scale:exp]", 4, 1, 64, .000001) : smooth);
out = global_group(hslider("[6]Output scaling[scale:exp]", 0, 0, 1, .000001) :
smooth);
process(x1, x2, x3) =
hindmarshrose(limit, a, b, I, c, d, r, s, x_r, dt, input(x1),
input(x2), input(x3));
|
a61ddf8811b9b6780f352440ced31d41e5be03b74a18aca5fefb72c90730c67f | gabrielsanchez/faust-guitarix-sc37 | sloop.dsp | declare id "sloop";
declare version "1.0";
declare author "brummer";
declare license "BSD";
import("stdfaust.lib");
B = checkbox("Capture"); // Capture sound while pressed
C = checkbox("Play");
I = int(B); // convert button signal from float to integer
R = (I-I') <= 0; // Reset capture when button is pressed
D = (+(I):*(R))~_; // Compute capture duration while button is pressed: 0..NNNN0..MMM
capture = ( *(B) : (+ : de.de.fdelay(2097152, D-1)) ~ *(1.0-B)) *(C);
si.smooth(c) = *(1-c) : +~*(c);
level = hslider("gain", 0, -96, 4, 0.1) : ba.db2linear : si.smooth(0.999);
process = vgroup( "SampleLooper", _ <:_, (capture : *(level)):>_ ) ;
| https://raw.githubusercontent.com/gabrielsanchez/faust-guitarix-sc37/c0608695e24870abb56f7f0d1355cbf2f563ed30/Faust/sloop.dsp | faust | Capture sound while pressed
convert button signal from float to integer
Reset capture when button is pressed
Compute capture duration while button is pressed: 0..NNNN0..MMM | declare id "sloop";
declare version "1.0";
declare author "brummer";
declare license "BSD";
import("stdfaust.lib");
C = checkbox("Play");
capture = ( *(B) : (+ : de.de.fdelay(2097152, D-1)) ~ *(1.0-B)) *(C);
si.smooth(c) = *(1-c) : +~*(c);
level = hslider("gain", 0, -96, 4, 0.1) : ba.db2linear : si.smooth(0.999);
process = vgroup( "SampleLooper", _ <:_, (capture : *(level)):>_ ) ;
|
f3a0a7073fb6f2e8a7af29753c6335de8e741e452515c433e6d8614463a02813 | sekisushai/ambitools | hoa_mic_encoder_lebedev50.dsp | declare name "MemsBedev HOA Encoder";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2014";
// Description: This tool does the Discrete Spherical Fourier Transform (DSFT) [1] of signal from a rigid spherical microphone with 50-node Lebedev grid geometry [1].
// CAUTION: This tool does only the DSFT, to obtain the Ambisonics components you should filter the signals with corresponding radial filters [1].
// Inputs: 50
// Outputs: (M+1)^2
// References:
// [1] P. Lecomte, P.-A. Gauthier, C. Langrenne, A. Garcia, and A. Berry, “On the use of a Lebedev grid for Ambisonics,” in Audio Engineering Society Convention 139, 2015.
import("stdfaust.lib");
import("lebedev.lib");
import("ymn.lib");
import("gui.lib");
// Maximum order M=5 for 50-node Lebedev grid [1].
M = 5;
ins = 50;
outs = (M+1)^2;
// VU-Meters activation (choose between vumeteron or off)
insvumeter = insvumeteroff;
outsvumeter = outsvumeteroff;
insvumeteron = par(i,ins,id(i,0));
insvumeteroff = par(i,ins,_);
outsvumeteron = par(i,M+1,metermute(i));
outsvumeteroff = par(i,outs,_);
vol = hslider("[1]Gain[unit:dB][style:knob]", 0, -10, 50, 0.1) : ba.db2linear : si.smooth(0.999);
// Vector of weighted spherical harmonics : spherical harmonics times the speaker weight for weighed quadrature rules [1].
row(i) = par(j,ins,yacn(i,azimuth(j),elevation(j))*(weight5(j)));
process = hgroup("[0]Inputs",insvumeter)<:par(i,outs,buswg(row(i)):>_):hgroup("[1]Outputs",par(i,outs,*(vol)):outsvumeter); | https://raw.githubusercontent.com/sekisushai/ambitools/2d21b7cc7cfe9bc35d91d51ec05bf9250372f0ce/Faust/src/hoa_mic_encoder_lebedev50.dsp | faust | Description: This tool does the Discrete Spherical Fourier Transform (DSFT) [1] of signal from a rigid spherical microphone with 50-node Lebedev grid geometry [1].
CAUTION: This tool does only the DSFT, to obtain the Ambisonics components you should filter the signals with corresponding radial filters [1].
Inputs: 50
Outputs: (M+1)^2
References:
[1] P. Lecomte, P.-A. Gauthier, C. Langrenne, A. Garcia, and A. Berry, “On the use of a Lebedev grid for Ambisonics,” in Audio Engineering Society Convention 139, 2015.
Maximum order M=5 for 50-node Lebedev grid [1].
VU-Meters activation (choose between vumeteron or off)
Vector of weighted spherical harmonics : spherical harmonics times the speaker weight for weighed quadrature rules [1]. | declare name "MemsBedev HOA Encoder";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2014";
import("stdfaust.lib");
import("lebedev.lib");
import("ymn.lib");
import("gui.lib");
M = 5;
ins = 50;
outs = (M+1)^2;
insvumeter = insvumeteroff;
outsvumeter = outsvumeteroff;
insvumeteron = par(i,ins,id(i,0));
insvumeteroff = par(i,ins,_);
outsvumeteron = par(i,M+1,metermute(i));
outsvumeteroff = par(i,outs,_);
vol = hslider("[1]Gain[unit:dB][style:knob]", 0, -10, 50, 0.1) : ba.db2linear : si.smooth(0.999);
row(i) = par(j,ins,yacn(i,azimuth(j),elevation(j))*(weight5(j)));
process = hgroup("[0]Inputs",insvumeter)<:par(i,outs,buswg(row(i)):>_):hgroup("[1]Outputs",par(i,outs,*(vol)):outsvumeter); |
3a9f356497df9453a8cd301f66188f9958085844188e43603e159a6b80b050f1 | scottericpetersen/OMI-Faust-Workshop | pulsee_v2.dsp | declare name " Pulsee ";
declare author " Scott E. Petersen " ;
declare copyright " (c) SEP 2023 ";
declare version " 0.02a ";
declare license " BSD ";
// Import the standard library so we can use preexisting objects
import("stdfaust.lib");
/* -- OSCILLATOR SECTION -- comment/uncomment to choose your source. For this workshop, use os.lf_imptrain and keep lfn line uncommented */
// src = no.noise*0.1;
// src = os.pulsetrain(2 + lfn, 0.5)*0.2;
// src = os.square(1 + lfn)*0.25;
lfn = no.lfnoiseN(3, 48000/100) : abs * drift;
src = os.lf_imptrain(pfreq + lfn)*amp;
/* -- GUI SECTION -- */
// knob for overall amplitude scaler
amp = oscGroup(hslider("[1]amp[style:knob]",0.1,0.0,0.3,0.001)) : si.smoo;
// knob for impulse frequency
pfreq = oscGroup(hslider("[2]freq[style:knob]",1.0,0.0,30.0,0.1)) : si.smoo;
// knob for random number amount
drift = oscGroup(hslider("[3]drift[style:knob]",0.0,0.0,10.0,0.1)) : si.smoo;
// button for filter order
order = filtGroup(hslider("[1]order[style:menu{'2nd':0;'4th':1}]",0,0,1,1));
// knob for filter cutoff
cutoff = filtGroup(hslider("cut[style:knob]",500,10,2800,10)) : si.smoo;
// knob for delay
delay = combGroup(hslider("delay[style:knob]",mdel/2, 0, mdel, 0.1)) : si.smoo;
// knob for feedback
fdbk = combGroup(hslider("fdbk[style:knob]",0.2,0.0,1.0,0.01)) : si.smoo;
// knob for feedback
mix = verbGroup(hslider("wet amnt[style:knob]",0.01,0.0,0.1,0.001)) : si.smoo;
// Groups for GUI elements listed above
oscGroup(x) = pulseeGUI(hgroup("[A]Oscillator",x));
filtGroup(y) = pulseeGUI(hgroup("[B]Filter",y));
combGroup(z) = pulseeGUI(hgroup("[C]Comb", z));
verbGroup(v) = pulseeGUI(hgroup("[D]Reverb", v));
pulseeGUI(a) = hgroup("PULSEE", a);
/* -- FILTER SECTION -- */
// low pass
filter2 = src : fi.lowpass(2, cutoff);
filter4 = src : fi.lowpass(4, cutoff);
//comb
mdel = 8092;
comb = fi.fb_comb(mdel, delay, delay, fdbk);
// panning
pan = sp.panner(0.5);
// reverb
verb = re.stereo_freeverb(0.91521,0.9459,0.495, 7);
// signal with no reverb, panned
dry = select2(order, filter2, filter4) : comb : pan : _*(1-mix), _*(1-mix); // inverse of mix for wet/dry
// dry signal processed through reverb
wet = dry : _*mix, _*mix : verb ;
/* -- OUTPUT -- */
process = wet,dry :> _,_; // here we do not "sum" like wet + dry, but parallel process the two, resulting in 4 chans out which have to be reduced to two using the merge operator.
| https://raw.githubusercontent.com/scottericpetersen/OMI-Faust-Workshop/28d952b574cd8f08c82416b6fdbfbc8fa9f9de74/examples/omi/pulsee_v2.dsp | faust | Import the standard library so we can use preexisting objects
-- OSCILLATOR SECTION -- comment/uncomment to choose your source. For this workshop, use os.lf_imptrain and keep lfn line uncommented
src = no.noise*0.1;
src = os.pulsetrain(2 + lfn, 0.5)*0.2;
src = os.square(1 + lfn)*0.25;
-- GUI SECTION --
knob for overall amplitude scaler
knob for impulse frequency
knob for random number amount
button for filter order
knob for filter cutoff
knob for delay
knob for feedback
knob for feedback
Groups for GUI elements listed above
-- FILTER SECTION --
low pass
comb
panning
reverb
signal with no reverb, panned
inverse of mix for wet/dry
dry signal processed through reverb
-- OUTPUT --
here we do not "sum" like wet + dry, but parallel process the two, resulting in 4 chans out which have to be reduced to two using the merge operator. | declare name " Pulsee ";
declare author " Scott E. Petersen " ;
declare copyright " (c) SEP 2023 ";
declare version " 0.02a ";
declare license " BSD ";
import("stdfaust.lib");
lfn = no.lfnoiseN(3, 48000/100) : abs * drift;
src = os.lf_imptrain(pfreq + lfn)*amp;
amp = oscGroup(hslider("[1]amp[style:knob]",0.1,0.0,0.3,0.001)) : si.smoo;
pfreq = oscGroup(hslider("[2]freq[style:knob]",1.0,0.0,30.0,0.1)) : si.smoo;
drift = oscGroup(hslider("[3]drift[style:knob]",0.0,0.0,10.0,0.1)) : si.smoo;
order = filtGroup(hslider("[1]order[style:menu{'2nd':0;'4th':1}]",0,0,1,1));
cutoff = filtGroup(hslider("cut[style:knob]",500,10,2800,10)) : si.smoo;
delay = combGroup(hslider("delay[style:knob]",mdel/2, 0, mdel, 0.1)) : si.smoo;
fdbk = combGroup(hslider("fdbk[style:knob]",0.2,0.0,1.0,0.01)) : si.smoo;
mix = verbGroup(hslider("wet amnt[style:knob]",0.01,0.0,0.1,0.001)) : si.smoo;
oscGroup(x) = pulseeGUI(hgroup("[A]Oscillator",x));
filtGroup(y) = pulseeGUI(hgroup("[B]Filter",y));
combGroup(z) = pulseeGUI(hgroup("[C]Comb", z));
verbGroup(v) = pulseeGUI(hgroup("[D]Reverb", v));
pulseeGUI(a) = hgroup("PULSEE", a);
filter2 = src : fi.lowpass(2, cutoff);
filter4 = src : fi.lowpass(4, cutoff);
mdel = 8092;
comb = fi.fb_comb(mdel, delay, delay, fdbk);
pan = sp.panner(0.5);
verb = re.stereo_freeverb(0.91521,0.9459,0.495, 7);
wet = dry : _*mix, _*mix : verb ;
|
630ca4cc6ff0e0bb72dfc52e1586af49494217ab52db66eac1ef360140887655 | Msc-program/Jacklink | compressordsp.dsp | declare name "compressor";
declare version "0.0";
declare author "Julius Smith";
declare license "MIT Style STK-4.2";
declare description "Compressor demo application, adapted from the Faust Library's dm.compressor_demo in demos.lib";
declare documentation "https://faustlibraries.grame.fr/libs/compressors/#cocompressor_mono";
import("stdfaust.lib");
//----------------------------`(dm.)compressor_mono_demo`-------------------------
// Mono Compressor
//
// #### Usage
//
// ```
// _ : compressor_mono_demo : _;
// ```
//------------------------------------------------------------
compressor_demo = ba.bypass1(cbp,compressor_mono_demo)
with {
comp_group(x) = vgroup("COMPRESSOR [tooltip: References:
https://faustlibraries.grame.fr/libs/compressors/
http://en.wikipedia.org/wiki/Dynamic_range_compression]", x);
meter_group(x) = comp_group(hgroup("[0]", x));
knob_group(x) = comp_group(hgroup("[1]", x));
cbp = meter_group(checkbox("[0] Bypass [tooltip: When this is checked, the compressor
has no effect]"));
gainview = co.compression_gain_mono(ratio,threshold,attack,release) : ba.linear2db :
meter_group(hbargraph("[1] Compressor Gain [unit:dB] [tooltip: Compressor gain in dB]",-50,+10));
displaygain = _ <: _,abs : _,gainview : attach;
compressor_stereo_demo =
displaygain(co.compressor_stereo(ratio,threshold,attack,release)) :
*(makeupgain), *(makeupgain);
compressor_mono_demo =
displaygain(co.compressor_mono(ratio,threshold,attack,release)) :
*(makeupgain);
ctl_group(x) = knob_group(hgroup("[3] Compression Control", x));
ratio = ctl_group(hslider("[0] Ratio [style:knob]
[tooltip: A compression Ratio of N means that for each N dB increase in input
signal level above Threshold, the output level goes up 1 dB]",
2, 1, 20, 0.1));
threshold = ctl_group(hslider("[1] Threshold [unit:dB] [style:knob]
[tooltip: When the signal level exceeds the Threshold (in dB), its level
is compressed according to the Ratio]",
-24, -100, 10, 0.1));
env_group(x) = knob_group(hgroup("[4] Compression Response", x));
attack = env_group(hslider("[1] Attack [unit:ms] [style:knob] [scale:log]
[tooltip: Time constant in ms (1/e smoothing time) for the compression gain
to approach (exponentially) a new lower target level (the compression
`kicking in')]", 15, 1, 1000, 0.1)) : *(0.001) : max(1/ma.SR);
release = env_group(hslider("[2] Release [unit:ms] [style: knob] [scale:log]
[tooltip: Time constant in ms (1/e smoothing time) for the compression gain
to approach (exponentially) a new higher target level (the compression
'releasing')]", 40, 1, 1000, 0.1)) : *(0.001) : max(1/ma.SR);
makeupgain = comp_group(hslider("[5] MakeUpGain [unit:dB]
[tooltip: The compressed-signal output level is increased by this amount
(in dB) to make up for the level lost due to compression]",
2, -96, 96, 0.1)) : ba.db2linear;
};
process = _ : compressor_demo : _;
| https://raw.githubusercontent.com/Msc-program/Jacklink/70b8634173e66d89884bb77b70b7b3ed01f71f79/faust-src/compressordsp.dsp | faust | ----------------------------`(dm.)compressor_mono_demo`-------------------------
Mono Compressor
#### Usage
```
_ : compressor_mono_demo : _;
```
------------------------------------------------------------
faustlibraries.grame.fr/libs/compressors/
en.wikipedia.org/wiki/Dynamic_range_compression]", x); | declare name "compressor";
declare version "0.0";
declare author "Julius Smith";
declare license "MIT Style STK-4.2";
declare description "Compressor demo application, adapted from the Faust Library's dm.compressor_demo in demos.lib";
declare documentation "https://faustlibraries.grame.fr/libs/compressors/#cocompressor_mono";
import("stdfaust.lib");
compressor_demo = ba.bypass1(cbp,compressor_mono_demo)
with {
comp_group(x) = vgroup("COMPRESSOR [tooltip: References:
meter_group(x) = comp_group(hgroup("[0]", x));
knob_group(x) = comp_group(hgroup("[1]", x));
cbp = meter_group(checkbox("[0] Bypass [tooltip: When this is checked, the compressor
has no effect]"));
gainview = co.compression_gain_mono(ratio,threshold,attack,release) : ba.linear2db :
meter_group(hbargraph("[1] Compressor Gain [unit:dB] [tooltip: Compressor gain in dB]",-50,+10));
displaygain = _ <: _,abs : _,gainview : attach;
compressor_stereo_demo =
displaygain(co.compressor_stereo(ratio,threshold,attack,release)) :
*(makeupgain), *(makeupgain);
compressor_mono_demo =
displaygain(co.compressor_mono(ratio,threshold,attack,release)) :
*(makeupgain);
ctl_group(x) = knob_group(hgroup("[3] Compression Control", x));
ratio = ctl_group(hslider("[0] Ratio [style:knob]
[tooltip: A compression Ratio of N means that for each N dB increase in input
signal level above Threshold, the output level goes up 1 dB]",
2, 1, 20, 0.1));
threshold = ctl_group(hslider("[1] Threshold [unit:dB] [style:knob]
[tooltip: When the signal level exceeds the Threshold (in dB), its level
is compressed according to the Ratio]",
-24, -100, 10, 0.1));
env_group(x) = knob_group(hgroup("[4] Compression Response", x));
attack = env_group(hslider("[1] Attack [unit:ms] [style:knob] [scale:log]
[tooltip: Time constant in ms (1/e smoothing time) for the compression gain
to approach (exponentially) a new lower target level (the compression
`kicking in')]", 15, 1, 1000, 0.1)) : *(0.001) : max(1/ma.SR);
release = env_group(hslider("[2] Release [unit:ms] [style: knob] [scale:log]
[tooltip: Time constant in ms (1/e smoothing time) for the compression gain
to approach (exponentially) a new higher target level (the compression
'releasing')]", 40, 1, 1000, 0.1)) : *(0.001) : max(1/ma.SR);
makeupgain = comp_group(hslider("[5] MakeUpGain [unit:dB]
[tooltip: The compressed-signal output level is increased by this amount
(in dB) to make up for the level lost due to compression]",
2, -96, 96, 0.1)) : ba.db2linear;
};
process = _ : compressor_demo : _;
|
56ebe845c833711f01291b01c8c8ceb3cada5e7a1916a90771b864cb0e1d7b6a | afalaize/faust | bandFilter.dsp | // WARNING: This a "legacy example based on a deprecated library". Check filters.lib
// for more accurate examples of filter functions
declare name "bandFilter";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
import("stdfaust.lib");
//---------------------second order filter--------------------------
// filter(Q,F,G)
// Q : quality factor [1..100]
// F : frequency (Hz)
// G : gain [0..1]
//------------------------------------------------------------------
filter(Q,F,G) = fi.TF2( (1 + K/Q + K*K) / D,
2 * (K*K - 1) / D,
(1 - K/Q + K*K) / D,
2 * (K*K - 1) / D,
(1 - V*K/Q + K*K) / D
)
with {
V = ba.db2linear(G);
K = tan(ma.PI*F/ma.SR);
D = 1 + V*K/Q + K*K;
};
//--------------- Band Filter with user interface ------------------
// bandfilter(F)
// F : default frequency (Hz)
//
//------------------------------------------------------------------
bandfilter(F) = filter( nentry("Q factor [style:knob]",50,0.1,100,0.1),
nentry("freq [unit:Hz][style:knob]", F, 20, 20000, 1),
0 - vslider("gain [unit:dB]", 0, -50, 50, 0.1)
);
//------------------------- Process --------------------------------
process = vgroup("Bandfilter", bandfilter(1000));
| https://raw.githubusercontent.com/afalaize/faust/8f9f5fe3aa167eaeecc15a99d4da984ac2797be3/examples/filtering/bandFilter.dsp | faust | WARNING: This a "legacy example based on a deprecated library". Check filters.lib
for more accurate examples of filter functions
---------------------second order filter--------------------------
filter(Q,F,G)
Q : quality factor [1..100]
F : frequency (Hz)
G : gain [0..1]
------------------------------------------------------------------
--------------- Band Filter with user interface ------------------
bandfilter(F)
F : default frequency (Hz)
------------------------------------------------------------------
------------------------- Process -------------------------------- |
declare name "bandFilter";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
import("stdfaust.lib");
filter(Q,F,G) = fi.TF2( (1 + K/Q + K*K) / D,
2 * (K*K - 1) / D,
(1 - K/Q + K*K) / D,
2 * (K*K - 1) / D,
(1 - V*K/Q + K*K) / D
)
with {
V = ba.db2linear(G);
K = tan(ma.PI*F/ma.SR);
D = 1 + V*K/Q + K*K;
};
bandfilter(F) = filter( nentry("Q factor [style:knob]",50,0.1,100,0.1),
nentry("freq [unit:Hz][style:knob]", F, 20, 20000, 1),
0 - vslider("gain [unit:dB]", 0, -50, 50, 0.1)
);
process = vgroup("Bandfilter", bandfilter(1000));
|
3b1511349ca72fe851c435384f68cfce7062b43b06823e864ca8d826988cbaa3 | rottingsounds/bitDSP-faust | DSMSine.dsp | declare name "DSMSine";
declare author "Till Bovermann, Dario Sanfilippo";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
// plot
// CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib dsm2-example.dsp
// ./DSMSine -n 10
// compile
// CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib dsm2-example.dsp
// open ./DSMSine
// SuperCollider
// export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
// faust2supercollider -I ../faust/bitDSP-faust/lib -noprefix DSMSine.dsp
freq = hslider("freq",100,0,20000,0);
// High-precision sinewave
sine(f) = sin(os.phasor(2 * ma.PI, f));
// Bipolar multi-bit signal to bipolar one-bit signal
// Standard test with a 1 kHz tone
onebitstream = bit.dsm2(sine(freq));
// Final output
process = onebitstream;
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/21cf36105c55b6e18969a867a319530a0ef1ea63/examples/_sc/DSMSine.dsp | faust | plot
CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib dsm2-example.dsp
./DSMSine -n 10
compile
CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib dsm2-example.dsp
open ./DSMSine
SuperCollider
export SUPERCOLLIDER_HEADERS=/localvol/sound/src/supercollider/include/
faust2supercollider -I ../faust/bitDSP-faust/lib -noprefix DSMSine.dsp
High-precision sinewave
Bipolar multi-bit signal to bipolar one-bit signal
Standard test with a 1 kHz tone
Final output | declare name "DSMSine";
declare author "Till Bovermann, Dario Sanfilippo";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
freq = hslider("freq",100,0,20000,0);
sine(f) = sin(os.phasor(2 * ma.PI, f));
onebitstream = bit.dsm2(sine(freq));
process = onebitstream;
|
16c79a8e4eda52d08a5cd68ba11f6ceae98b347cdfbf9ae0d24f4793f819ecee | PierreKy-org/plugins_server_webaudiomodules | granulator.dsp | // FROM FAUST DEMO
// Designed to use the Analog Input for parameter controls.
//
///////////////////////////////////////////////////////////////////////////////////////////////////
//
// ANALOG IN:
// ANALOG 0 : Grain Size
// ANALOG 1 : Speed
// ANALOG 2 : Probability
// (others analog inputs are not used)
//
///////////////////////////////////////////////////////////////////////////////////////////////////
process = vgroup("Granulator", environment {
declare name "Granulator";
declare author "Adapted from sfIter by Christophe Lebreton";
/* =========== DESCRIPTION =============
- The granulator takes very small parts of a sound, called GRAINS, and plays them at a varying speed
- Front = Medium size grains
- Back = short grains
- Left Slow rhythm
- Right = Fast rhythm
- Bottom = Regular occurrences
- Head = Irregular occurrences
*/
import("stdfaust.lib");
process = hgroup("Granulator", *(excitation : ampf));
excitation = noiseburst(gate,P) * (gain);
ampf = an.amp_follower_ud(duree_env,duree_env);
//----------------------- NOISEBURST -------------------------
noiseburst(gate,P) = no.noise : *(gate : trigger(P))
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0)/n;
release(n) = + ~ decay(n);
trigger(n) = upfront : release(n) : > (0.0);
};
//-------------------------------------------------------------
P = freq; // fundamental period in samples
freq = hslider("[1]GrainSize[BELA: ANALOG_0]", 200,5,2205,1);
// the frequency gives the white noise band width
Pmax = 4096; // maximum P (for de.delay-line allocation)
// PHASOR_BIN //////////////////////////////
phasor_bin(init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
gate = phasor_bin(1) :-(0.001):pulsar;
gain = 1;
// PULSAR //////////////////////////////
// Pulsar allows to create a more or less random 'pulse'(proba).
pulsar = _<:((_<(ratio_env)):@(100))*(proba>(_,abs(no.noise):ba.latch));
speed = hslider ("[2]Speed[BELA: ANALOG_1]", 10,1,20,0.0001):fi.lowpass(1,1);
ratio_env = 0.5;
fade = (0.5); // min > 0 to avoid division by 0
proba = hslider ("[3]Probability[BELA: ANALOG_2]", 70,50,100,1) * (0.01):fi.lowpass(1,1);
duree_env = 1/(speed: / (ratio_env*(0.25)*fade));
}.process);
| https://raw.githubusercontent.com/PierreKy-org/plugins_server_webaudiomodules/a8162fbb0a9341ad67d3cbb78714e7a2f7c56b0b/plugins/granulator/granulator.dsp | faust | FROM FAUST DEMO
Designed to use the Analog Input for parameter controls.
/////////////////////////////////////////////////////////////////////////////////////////////////
ANALOG IN:
ANALOG 0 : Grain Size
ANALOG 1 : Speed
ANALOG 2 : Probability
(others analog inputs are not used)
/////////////////////////////////////////////////////////////////////////////////////////////////
=========== DESCRIPTION =============
- The granulator takes very small parts of a sound, called GRAINS, and plays them at a varying speed
- Front = Medium size grains
- Back = short grains
- Left Slow rhythm
- Right = Fast rhythm
- Bottom = Regular occurrences
- Head = Irregular occurrences
----------------------- NOISEBURST -------------------------
-------------------------------------------------------------
fundamental period in samples
the frequency gives the white noise band width
maximum P (for de.delay-line allocation)
PHASOR_BIN //////////////////////////////
PULSAR //////////////////////////////
Pulsar allows to create a more or less random 'pulse'(proba).
min > 0 to avoid division by 0
|
process = vgroup("Granulator", environment {
declare name "Granulator";
declare author "Adapted from sfIter by Christophe Lebreton";
import("stdfaust.lib");
process = hgroup("Granulator", *(excitation : ampf));
excitation = noiseburst(gate,P) * (gain);
ampf = an.amp_follower_ud(duree_env,duree_env);
noiseburst(gate,P) = no.noise : *(gate : trigger(P))
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0)/n;
release(n) = + ~ decay(n);
trigger(n) = upfront : release(n) : > (0.0);
};
freq = hslider("[1]GrainSize[BELA: ANALOG_0]", 200,5,2205,1);
phasor_bin(init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
gate = phasor_bin(1) :-(0.001):pulsar;
gain = 1;
pulsar = _<:((_<(ratio_env)):@(100))*(proba>(_,abs(no.noise):ba.latch));
speed = hslider ("[2]Speed[BELA: ANALOG_1]", 10,1,20,0.0001):fi.lowpass(1,1);
ratio_env = 0.5;
proba = hslider ("[3]Probability[BELA: ANALOG_2]", 70,50,100,1) * (0.01):fi.lowpass(1,1);
duree_env = 1/(speed: / (ratio_env*(0.25)*fade));
}.process);
|
8905bc904a74557d73b02f8816041bf8e8864bb53a3288811b12d0c7429d90e8 | jameslnrd/mi_introduction_workshop_2020 | hammerOsc.dsp | declare name "Hammered Oscillator";
declare author "James Leonard";
declare date "April 2020";
/* ========= DESCRITPION =============
An oscillator struck by a hammer (a second "loose" oscillator placed above it)
- inputs: force impulse onto the hammer mass
- outputs: oscillator and hammer positions.
- controls: none.
Note: The "clicking" sound is because we are listening to the hammer mass position on the
second output channel. Listening to this mass allows to plot the collision, but if you
were just interested in the oscillator's sound you should only listen to that.
*/
import("stdfaust.lib");
in1 = button("Frc Input 1"): ba.impulsify * -0.1;
OutGain = 1;
model = (
mi.oscil(1., 0.1, 0.0003, 0, 0., 0.),
mi.mass(0.3, 0, 1., 1.),
mi.ground(1.),
par(i, nbFrcIn,_):
RoutingMassToLink ,
par(i, nbFrcIn,_):
mi.springDamper(0.0001, 0.05, 1., 1.),
mi.collision(0.1, 0.001, 0, 0., 1.),
par(i, nbOut+nbFrcIn, _):
RoutingLinkToMass
)~par(i, nbMass, _):
par(i, nbMass, !), par(i, nbOut , _)
with{
RoutingMassToLink(m0, m1, m2) = /* routed positions */ m2, m1, m0, m1, /* outputs */ m0, m1;
RoutingLinkToMass(l0_f1, l0_f2, l1_f1, l1_f2, p_out1, p_out2, f_in1) = /* routed forces */ l1_f1, f_in1 + l0_f2 + l1_f2, l0_f1, /* pass-through */ p_out1, p_out2;
nbMass = 3;
nbFrcIn = 1;
nbOut = 2;
};
process = in1 : model:*(OutGain), *(OutGain);
/*
========= MIMS SCRIPT USED FOR MODEL GENERATION =============
# MIMS script file
# Script author: James Leonard
# Integrated harmonic oscillator
@o osc 1. 0.1 0.0003 0. 0.
# A slow moving oscillator placed above the other
# serving as a hammer
@ham mass 0.3 1. 0.
@g ground 1.
@sp springDamper @g @ham 0.0001 0.05
# Add force input to the hammer
@in1 frcInput @ham
@c contact @o @ham 0.1 0.001
# Add position output from the oscillator
@out1 posOutput @o
@out2 posOutput @ham
# end of MIMS script
*/ | https://raw.githubusercontent.com/jameslnrd/mi_introduction_workshop_2020/2f487dbc5b8e7cd83cbd962254e737bdb82948f6/03_HammerTime/hammerOsc.dsp | faust | ========= DESCRITPION =============
An oscillator struck by a hammer (a second "loose" oscillator placed above it)
- inputs: force impulse onto the hammer mass
- outputs: oscillator and hammer positions.
- controls: none.
Note: The "clicking" sound is because we are listening to the hammer mass position on the
second output channel. Listening to this mass allows to plot the collision, but if you
were just interested in the oscillator's sound you should only listen to that.
routed positions
outputs
routed forces
pass-through
========= MIMS SCRIPT USED FOR MODEL GENERATION =============
# MIMS script file
# Script author: James Leonard
# Integrated harmonic oscillator
@o osc 1. 0.1 0.0003 0. 0.
# A slow moving oscillator placed above the other
# serving as a hammer
@ham mass 0.3 1. 0.
@g ground 1.
@sp springDamper @g @ham 0.0001 0.05
# Add force input to the hammer
@in1 frcInput @ham
@c contact @o @ham 0.1 0.001
# Add position output from the oscillator
@out1 posOutput @o
@out2 posOutput @ham
# end of MIMS script
| declare name "Hammered Oscillator";
declare author "James Leonard";
declare date "April 2020";
import("stdfaust.lib");
in1 = button("Frc Input 1"): ba.impulsify * -0.1;
OutGain = 1;
model = (
mi.oscil(1., 0.1, 0.0003, 0, 0., 0.),
mi.mass(0.3, 0, 1., 1.),
mi.ground(1.),
par(i, nbFrcIn,_):
RoutingMassToLink ,
par(i, nbFrcIn,_):
mi.springDamper(0.0001, 0.05, 1., 1.),
mi.collision(0.1, 0.001, 0, 0., 1.),
par(i, nbOut+nbFrcIn, _):
RoutingLinkToMass
)~par(i, nbMass, _):
par(i, nbMass, !), par(i, nbOut , _)
with{
nbMass = 3;
nbFrcIn = 1;
nbOut = 2;
};
process = in1 : model:*(OutGain), *(OutGain);
|
0978644648aa9dedcb11df2c29460aadee4adf8027f460af5f1a221dbbe2b52a | osam-cologne/faustsimplesynth | faustsimplesynth.dsp | // faustsimplesynth - a simple sawtooth wave synthesizer implemented in Faust
// Copyright (C) 2018 Daniel Appelt
// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation, either version 3 of the License, or
// (at your option) any later version.
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
// You should have received a copy of the GNU General Public License
// along with this program. If not, see <http://www.gnu.org/licenses/>.
declare name "faustsimplesynth";
declare version "0.1.0";
declare author "Daniel Appelt";
declare license "GPL3";
declare copyright "Copyright (C) 2018 Daniel Appelt";
// Declaring nvoices takes care of creating a synth plugin / app
declare nvoices "1";
import("stdfaust.lib");
// MIDI note input (by naming convention)
midigate = button("gate [hidden]");
midifreq = hslider("freq [unit:Hz][hidden]", 440, 20, 20000, 1);
midigain = hslider("gain [hidden]", 0.5, 0, 1, 0.001);
// ADSR amp envelope
a = hslider("h:Env/[0]Attack [unit: s][scale:log][style:knob]", 0.1, 0.001, 2, 0.001);
d = hslider("h:Env/[1]Decay [unit: s][scale:log][style:knob]", 0.3, 0.001, 2, 0.001);
s = hslider("h:Env/[2]Sustain [style:knob]", 0.8, 0, 1, 0.01);
r = hslider("h:Env/[2]Release [unit: s][scale:log][style:knob]", 2.0, 0.001, 4, 0.001);
// Volume always sets the maximum loudness
amp_vol = hslider("h:Amplifier/[0]Volume [style:knob]", 0.5, 0, 1, 0.01);
// Velocity determines how much of the volume will be affected by MIDI note velocity.
// 0 = no effect, 1 = max velocity leads to max volume, -1 = min velocity leads to max volume
amp_vel = hslider("h:Amplifier/[1]Velocity [style:knob]", 0, -1, 1, 0.01);
amp_mod = (midigain * amp_vel + (amp_vel < 0) + 1 - abs(amp_vel)) * en.adsr(a, d, s, r, midigate);
process = midifreq : os.sawtooth * amp_mod * amp_vol;
| https://raw.githubusercontent.com/osam-cologne/faustsimplesynth/a283944a816dd41fd4986a7dede5a14399c04200/src/faustsimplesynth.dsp | faust | faustsimplesynth - a simple sawtooth wave synthesizer implemented in Faust
Copyright (C) 2018 Daniel Appelt
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program. If not, see <http://www.gnu.org/licenses/>.
Declaring nvoices takes care of creating a synth plugin / app
MIDI note input (by naming convention)
ADSR amp envelope
Volume always sets the maximum loudness
Velocity determines how much of the volume will be affected by MIDI note velocity.
0 = no effect, 1 = max velocity leads to max volume, -1 = min velocity leads to max volume |
declare name "faustsimplesynth";
declare version "0.1.0";
declare author "Daniel Appelt";
declare license "GPL3";
declare copyright "Copyright (C) 2018 Daniel Appelt";
declare nvoices "1";
import("stdfaust.lib");
midigate = button("gate [hidden]");
midifreq = hslider("freq [unit:Hz][hidden]", 440, 20, 20000, 1);
midigain = hslider("gain [hidden]", 0.5, 0, 1, 0.001);
a = hslider("h:Env/[0]Attack [unit: s][scale:log][style:knob]", 0.1, 0.001, 2, 0.001);
d = hslider("h:Env/[1]Decay [unit: s][scale:log][style:knob]", 0.3, 0.001, 2, 0.001);
s = hslider("h:Env/[2]Sustain [style:knob]", 0.8, 0, 1, 0.01);
r = hslider("h:Env/[2]Release [unit: s][scale:log][style:knob]", 2.0, 0.001, 4, 0.001);
amp_vol = hslider("h:Amplifier/[0]Volume [style:knob]", 0.5, 0, 1, 0.01);
amp_vel = hslider("h:Amplifier/[1]Velocity [style:knob]", 0, -1, 1, 0.01);
amp_mod = (midigain * amp_vel + (amp_vel < 0) + 1 - abs(amp_vel)) * en.adsr(a, d, s, r, midigate);
process = midifreq : os.sawtooth * amp_mod * amp_vol;
|
36b6b1dd77bd663d212dbeafb31db7681720615685bb55f3838babfa5e5881a7 | clearly-broken-software/Uprising | test.dsp | import("stdfaust.lib");
declare author "Bart Brouns";
declare license "GPLv3";
xVal = hslider("xVal", 0, 0, maxSamples, 1)/defaultSR*ma.SR;
yVal = hslider("yVal", 0, 0, 1, 0.001);
slope = hslider("slope", 0, -1, 1, 0.001);
goBtn = button("resetBtn");//:ba.impulsify;
maxSeconds = 5;
maxSamples = maxSeconds*defaultSR;
defaultSR = 48000;
process =
envelope;
// ramp from 0 to 1 in n samples.
// when reset == 1, go back to 0.
ramp(n,reset) = select2(reset,_+(1/n):min(1),0)~_;
xVal_target(reset) = rwtable(3, 0.0,nextValIndex(reset),xVal,currentValIndex(reset));
yVal_target(reset) = rwtable(3, 0.0,nextValIndex(reset),yVal,currentValIndex(reset));
currentValIndex(reset) = reset :ba.toggle;
nextValIndex(reset) = currentValIndex(reset)*-1+1;
xfade(a,b,x) = a*(1-x)+b*x;
envelope = FB~_ with {
FB(prev) = ((xfade(ba.sAndH(reset),yVal_target(reset),position)+(yVal*startPulse))~(_))
,(reset':ba.impulsify:hbargraph("[1]endpoint", 0, 1))
with {
reset = ( (prev==yVal_target(_)) :xor(goBtn) )~(_<:(_,_));
position = ramp(xVal_target(reset),reset):hbargraph("[0]position", 0, 1);
// the commented out stuff is WIP for changing the curve of the ramp
// position = ramp(xVal_target(reset),reset):hbargraph("pos", 0, 1):pow(power):max(0):min(1);
};
startPulse = 1-1';
// power = select2(slope<0,slope*px+1,1+slope*mx):hbargraph("power", 0, 10);
// px = hslider("px", 1, 1, 9, 1);
// mx = hslider("mx", 0.5, 0, 1, 0.001);
}; | https://raw.githubusercontent.com/clearly-broken-software/Uprising/89f5b49d90cd47611da7e7dc2009061768716b4c/plugins/uprising/dsp/faust/test.dsp | faust | :ba.impulsify;
ramp from 0 to 1 in n samples.
when reset == 1, go back to 0.
the commented out stuff is WIP for changing the curve of the ramp
position = ramp(xVal_target(reset),reset):hbargraph("pos", 0, 1):pow(power):max(0):min(1);
power = select2(slope<0,slope*px+1,1+slope*mx):hbargraph("power", 0, 10);
px = hslider("px", 1, 1, 9, 1);
mx = hslider("mx", 0.5, 0, 1, 0.001); | import("stdfaust.lib");
declare author "Bart Brouns";
declare license "GPLv3";
xVal = hslider("xVal", 0, 0, maxSamples, 1)/defaultSR*ma.SR;
yVal = hslider("yVal", 0, 0, 1, 0.001);
slope = hslider("slope", 0, -1, 1, 0.001);
maxSeconds = 5;
maxSamples = maxSeconds*defaultSR;
defaultSR = 48000;
process =
envelope;
ramp(n,reset) = select2(reset,_+(1/n):min(1),0)~_;
xVal_target(reset) = rwtable(3, 0.0,nextValIndex(reset),xVal,currentValIndex(reset));
yVal_target(reset) = rwtable(3, 0.0,nextValIndex(reset),yVal,currentValIndex(reset));
currentValIndex(reset) = reset :ba.toggle;
nextValIndex(reset) = currentValIndex(reset)*-1+1;
xfade(a,b,x) = a*(1-x)+b*x;
envelope = FB~_ with {
FB(prev) = ((xfade(ba.sAndH(reset),yVal_target(reset),position)+(yVal*startPulse))~(_))
,(reset':ba.impulsify:hbargraph("[1]endpoint", 0, 1))
with {
reset = ( (prev==yVal_target(_)) :xor(goBtn) )~(_<:(_,_));
position = ramp(xVal_target(reset),reset):hbargraph("[0]position", 0, 1);
};
startPulse = 1-1';
}; |
8377f08e3c150ae8f9d741e065315617a020231da2661c1e31773149ec264821 | sonejostudios/DeLooper | DeLooper.dsp | declare name "DeLooper";
declare version "1.0";
declare author "Vincent Rateau";
declare license "GPL v3";
declare reference "www.sonejo.net";
declare description "Sample-accurate Looper/Delay with free mode and midi-clock sync mode";
import("stdfaust.lib");
// variables at compilation time
loopsec = 32 ; //max loop time (sec)
process = masterloop ;
masterloop = _ <: _*(1-switcher), _*switcher : ((loop*loopvol*mute), _) :> _ ;
switcher = (rec*looptime) == 0 : si.smooth(0.99) ; // bypass signal if looptime == 0
loop = + ~ de.fdelay(8388608, looptime)*erase*loopfback ; // fdelay better because sdelay makes a dobble signal bug in free mode
//Compute the loop time in free mode and sync mode and switch between them
////////////////////////////////////
looptime = select2(sync, counttime, barssync) : _/divide : looplength : int
with {
counttime = ba.countup(ma.SR*loopsec, (1-setloop) ) : ba.sAndH(setloop != new) ; // : looplength2 ;
// sync mode with more-or-less function
barssync = counttime <: barsync2, _ : moreless : _ - midiclock2beat : _ + setloop*midiclock2beat ;
barsync2 = _ / midiclock2beat : int : _*midiclock2beat : _ + midiclock2beat ;
moreless(x,y) = select2(setrange(x,y), x, x + midiclock2beat ) ;
setrange(x,y) = (y > (x - midiclock2beat/2))*(y < (x + midiclock2beat/2)) ;
};
// GUI
/////////////////////
setloop = checkbox("set loop length[midi:ctrl 59]") ;
rec = checkbox("rec[midi:ctrl 60]") ;
new = button("set new loop[midi:ctrl 58]") : ba.impulsify ;
looplength = vbargraph("loop length",0, 44100*loopsec) ; // (== looplength2 in free mode, but snyced in sync mode)
//looplength2 = vbargraph("real loop length",0, 44100*loopsec) ; // the real number of the sample counter
sync = checkbox("sync") ;
erase = 1-button("erase") : si.smooth(0.99);
loopfback = hslider("feedback",1,0,1,0.01) : si.smooth(0.999);
loopvol = hslider("loop vol[midi:ctrl 118]",1,0,1,0.01) : si.smooth(0.999);
mute = 1-checkbox("mute") : si.smooth(0.999);
divide = nentry("divide loop length by",1,1,100,1) : si.smooth(ba.tau2pole(2));
//MIDICLOCK to BEAT (AMOUNT OF SAMPLES IN 1 BEAT) to BPM and SAMPLES
//////////////////////////////////
//send midi clock signal, count sample amount, latch stable signal between beat recognition (after 16000 samples), convert to bpm,
// convert bpm to sample-accurate loop length (in sample)
midiclock2beat = vgroup("MIDI Clock",((clocker, play)) : attach : midi2count <: (_@16000==_@16001), _ : ba.latch : s2bpm : int : bpm2s : result3)
with{
//clockersim = ba.pulse(1837.5 /2) ; // replace it with clocker for internal clock for testing
clocker = checkbox("Clock Signal[midi:clock]") ; // create a square signal (1/0), changing state at each received clock
play = checkbox("Start/Stop Signal[midi:start] [midi:stop]") ; // just to show start stop signal
// takes clocker signal(24 changes/beat), count samples between every changes, latch the highest number.
// re-latch to fix the highest number, * by 24 to get on beat (in samples)
midi2count = _ <: _ != _@1 : ba.countup(8388608,_) : result1 <: _==0,_@1 : ba.latch <: _>_@1, _ : ba.latch : _*24 : result2;
result1 = _ ; //: vbargraph("samplecount midi", 0, 8388608);
result2 = _ ; //: vbargraph("sample amount midi2", 0, 8388608);
result3 = _ ; //: vbargraph("one-beat length (samples) from bpm", 0, 8388608);
// convert bpm to sample amount for loop length
bpm2s = 60/_ : _* ma.SR ;
// round down sampleholder and convert it to bpm
s2bpm = _/10 : int : _*10 : ma.SR/_ : _*60 : int : bpm ;
bpm = vbargraph("bpm", 0, 240.0) ;
};
| https://raw.githubusercontent.com/sonejostudios/DeLooper/0ff9ac414d6c7daa3ef494d43524b9e1c9e3f7f5/DeLooper.dsp | faust | variables at compilation time
max loop time (sec)
bypass signal if looptime == 0
fdelay better because sdelay makes a dobble signal bug in free mode
Compute the loop time in free mode and sync mode and switch between them
//////////////////////////////////
: looplength2 ;
sync mode with more-or-less function
GUI
///////////////////
(== looplength2 in free mode, but snyced in sync mode)
looplength2 = vbargraph("real loop length",0, 44100*loopsec) ; // the real number of the sample counter
MIDICLOCK to BEAT (AMOUNT OF SAMPLES IN 1 BEAT) to BPM and SAMPLES
////////////////////////////////
send midi clock signal, count sample amount, latch stable signal between beat recognition (after 16000 samples), convert to bpm,
convert bpm to sample-accurate loop length (in sample)
clockersim = ba.pulse(1837.5 /2) ; // replace it with clocker for internal clock for testing
create a square signal (1/0), changing state at each received clock
just to show start stop signal
takes clocker signal(24 changes/beat), count samples between every changes, latch the highest number.
re-latch to fix the highest number, * by 24 to get on beat (in samples)
: vbargraph("samplecount midi", 0, 8388608);
: vbargraph("sample amount midi2", 0, 8388608);
: vbargraph("one-beat length (samples) from bpm", 0, 8388608);
convert bpm to sample amount for loop length
round down sampleholder and convert it to bpm | declare name "DeLooper";
declare version "1.0";
declare author "Vincent Rateau";
declare license "GPL v3";
declare reference "www.sonejo.net";
declare description "Sample-accurate Looper/Delay with free mode and midi-clock sync mode";
import("stdfaust.lib");
process = masterloop ;
masterloop = _ <: _*(1-switcher), _*switcher : ((loop*loopvol*mute), _) :> _ ;
looptime = select2(sync, counttime, barssync) : _/divide : looplength : int
with {
barssync = counttime <: barsync2, _ : moreless : _ - midiclock2beat : _ + setloop*midiclock2beat ;
barsync2 = _ / midiclock2beat : int : _*midiclock2beat : _ + midiclock2beat ;
moreless(x,y) = select2(setrange(x,y), x, x + midiclock2beat ) ;
setrange(x,y) = (y > (x - midiclock2beat/2))*(y < (x + midiclock2beat/2)) ;
};
setloop = checkbox("set loop length[midi:ctrl 59]") ;
rec = checkbox("rec[midi:ctrl 60]") ;
new = button("set new loop[midi:ctrl 58]") : ba.impulsify ;
sync = checkbox("sync") ;
erase = 1-button("erase") : si.smooth(0.99);
loopfback = hslider("feedback",1,0,1,0.01) : si.smooth(0.999);
loopvol = hslider("loop vol[midi:ctrl 118]",1,0,1,0.01) : si.smooth(0.999);
mute = 1-checkbox("mute") : si.smooth(0.999);
divide = nentry("divide loop length by",1,1,100,1) : si.smooth(ba.tau2pole(2));
midiclock2beat = vgroup("MIDI Clock",((clocker, play)) : attach : midi2count <: (_@16000==_@16001), _ : ba.latch : s2bpm : int : bpm2s : result3)
with{
midi2count = _ <: _ != _@1 : ba.countup(8388608,_) : result1 <: _==0,_@1 : ba.latch <: _>_@1, _ : ba.latch : _*24 : result2;
bpm2s = 60/_ : _* ma.SR ;
s2bpm = _/10 : int : _*10 : ma.SR/_ : _*60 : int : bpm ;
bpm = vbargraph("bpm", 0, 240.0) ;
};
|
51bd5286935b6bfb490f5088cca10b48317ff6aefe952728cab067dbb2a2f112 | afalaize/faust | DNN.dsp | // Forward Deep Neural Net (DNN), any number of layers of any size each
declare name "DNN";
declare author "JOS";
declare license "STK-4.3";
import("stdfaust.lib");
layerSizes = (8,5,8); // autoencoder with 8 in & out, 5-state hidden layer
w(m,n,k) = m*100+n*10+k; // placeholder weights: m=layer, n=fromNode, k=destNode
M = ba.count(layerSizes);
N(l) = ba.take(l+1,layerSizes); // Nodes per layer
process = seq(m, M-1, layer(m))
// look at weights:
// process = par(m,M,par(n,N(m),par(k,N(m),w(m,n,k))))
with {
layer(m) = weights(m) :> nonlinearities(m);
nonlinearities(m) = bus(N(m)*N(m+1)) :> par(n,N(m+1),nl(n));
weights(m) = bus(N(m)) <: par(n,N(m),(bus(N(m+1))<:wts(m,n)));
wts(m,n) = bus(N(m+1)) : par(k,N(m+1),*(w(m,n,k)));
nl(n,x) = x * (x>0); // ReLU
bus(N) = par(k,N,_);
};
| https://raw.githubusercontent.com/afalaize/faust/8f9f5fe3aa167eaeecc15a99d4da984ac2797be3/examples/filtering/DNN.dsp | faust | Forward Deep Neural Net (DNN), any number of layers of any size each
autoencoder with 8 in & out, 5-state hidden layer
placeholder weights: m=layer, n=fromNode, k=destNode
Nodes per layer
look at weights:
process = par(m,M,par(n,N(m),par(k,N(m),w(m,n,k))))
ReLU |
declare name "DNN";
declare author "JOS";
declare license "STK-4.3";
import("stdfaust.lib");
M = ba.count(layerSizes);
process = seq(m, M-1, layer(m))
with {
layer(m) = weights(m) :> nonlinearities(m);
nonlinearities(m) = bus(N(m)*N(m+1)) :> par(n,N(m+1),nl(n));
weights(m) = bus(N(m)) <: par(n,N(m),(bus(N(m+1))<:wts(m,n)));
wts(m,n) = bus(N(m+1)) : par(k,N(m+1),*(w(m,n,k)));
bus(N) = par(k,N,_);
};
|
27814aa5531e4dbc4ee7eadca0578d3290aa4bb09e9a3dcc9086e538b236f38f | CICM-research-composition/livepatching | help.dsp | process = vgroup("SAtonalSoftHarp",environment{declare name "Atonal Soft Harp";
declare author "ER"; //Adapted from NLFeks by Julius Smith and Romain Michon;
/* =============== DESCRIPTION ======================== :
- Soft Atonal Harp
- Swing = Plucking all the strings one by one
- Left = Slow rhythm /Low frequencies/ Silence
- Right = Fast rhythm/ High frequencies
- Back = Short and dry notes
- Front = Long and bright notes
*/
import("stdfaust.lib");
//==================== INSTRUMENT =======================
process = par(i, N, NFLeks(i)):>_;
NFLeks(n) = filtered_excitation(n,P(freq(n)),freq(n)) : stringloop(freq(n));
//==================== GUI SPECIFICATION ================
N = 20;
hand = hslider("h:[1]/Hand[acc:0 1 -10 0 10]", 10, 0, N, 1) : ba.automat(bps, 15, 0.0)// => gate
with{
bps = hslider("h:[1]/Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
gain = 1;
pickangle = 0.9;
beta = 0.5;
// String decay time in seconds:
t60 = hslider("h:[2]Reverberation/ Resonance[unit:s][acc:2 1 -10 0 10]", 5, 0.5, 10, 0.01):min(10):max(0.5); // -60db decay time (sec)
B = 0;
L = -10 : ba.db2linear;
//---------------------------------- FREQUENCY TABLE ---------------------------
freq(0) = 200;
freq(1) = 215;
freq(2) = 230;
freq(3) = 245;
freq(4) = 260;
freq(5) = 275;
freq(d) = freq(d-6)*(2);
//==================== SIGNAL PROCESSING ================
//----------------------- noiseburst -------------------------
// White noise burst (adapted from Faust's karplus.dsp example)
// Requires music.lib (for no.noise)
noiseburst(d,e) = no.noise : *(trigger(d,e))
with{
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0)/n;
release(n) = + ~ decay(n);
position(d) = abs(hand - d) < 0.5;
trigger(d,n) = position(d) : upfront : release(n) : > (0.0);
};
P(f) = ma.SR/f ; // fundamental period in samples
Pmax = 4096; // maximum P (for delay-line allocation)
ppdel(f) = beta*P(f); // pick position delay
pickposfilter(f) = fi.ffcombfilter(Pmax,ppdel(f),-1); // defined in filter.lib
excitation(d,e) = noiseburst(d,e) : *(gain); // defined in signal.lib
rho(f) = pow(0.001,1.0/(f*t60)); // multiplies loop-gain
// Original EKS damping filter:
b1 = 0.5*B; b0 = 1.0-b1; // S and 1-S
dampingfilter1(f,x) = rho(f) * ((b0 * x) + (b1 * x'));
// Linear phase FIR3 damping filter:
h0 = (1.0 + B)/2; h1 = (1.0 - B)/4;
dampingfilter2(f,x) = rho(f) * (h0 * x' + h1*(x+x''));
loopfilter(f) = dampingfilter2(f); // or dampingfilter1
filtered_excitation(d,e,f) = excitation(d,e) : si.smooth(pickangle)
: pickposfilter(f) : fi.levelfilter(L,f); // see filter.lib
stringloop(f) = (+ : de.fdelay4(Pmax, P(f)-2)) ~ (loopfilter(f));
}.process);
| https://raw.githubusercontent.com/CICM-research-composition/livepatching/da47667b3f5236b4b81cb05593f35d408caaca9e/source/help.dsp | faust | Adapted from NLFeks by Julius Smith and Romain Michon;
=============== DESCRIPTION ======================== :
- Soft Atonal Harp
- Swing = Plucking all the strings one by one
- Left = Slow rhythm /Low frequencies/ Silence
- Right = Fast rhythm/ High frequencies
- Back = Short and dry notes
- Front = Long and bright notes
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
=> gate
String decay time in seconds:
-60db decay time (sec)
---------------------------------- FREQUENCY TABLE ---------------------------
==================== SIGNAL PROCESSING ================
----------------------- noiseburst -------------------------
White noise burst (adapted from Faust's karplus.dsp example)
Requires music.lib (for no.noise)
fundamental period in samples
maximum P (for delay-line allocation)
pick position delay
defined in filter.lib
defined in signal.lib
multiplies loop-gain
Original EKS damping filter:
S and 1-S
Linear phase FIR3 damping filter:
or dampingfilter1
see filter.lib | process = vgroup("SAtonalSoftHarp",environment{declare name "Atonal Soft Harp";
import("stdfaust.lib");
process = par(i, N, NFLeks(i)):>_;
NFLeks(n) = filtered_excitation(n,P(freq(n)),freq(n)) : stringloop(freq(n));
N = 20;
with{
bps = hslider("h:[1]/Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
gain = 1;
pickangle = 0.9;
beta = 0.5;
B = 0;
L = -10 : ba.db2linear;
freq(0) = 200;
freq(1) = 215;
freq(2) = 230;
freq(3) = 245;
freq(4) = 260;
freq(5) = 275;
freq(d) = freq(d-6)*(2);
noiseburst(d,e) = no.noise : *(trigger(d,e))
with{
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0)/n;
release(n) = + ~ decay(n);
position(d) = abs(hand - d) < 0.5;
trigger(d,n) = position(d) : upfront : release(n) : > (0.0);
};
dampingfilter1(f,x) = rho(f) * ((b0 * x) + (b1 * x'));
h0 = (1.0 + B)/2; h1 = (1.0 - B)/4;
dampingfilter2(f,x) = rho(f) * (h0 * x' + h1*(x+x''));
filtered_excitation(d,e,f) = excitation(d,e) : si.smooth(pickangle)
stringloop(f) = (+ : de.fdelay4(Pmax, P(f)-2)) ~ (loopfilter(f));
}.process);
|
63fcd9dc923f0deee8318cba9745e5f89733158d01372a4ab1e2bfebbb845fa3 | friskgit/snares | snare_multi.dsp | // -*- compile-command: "cd .. && make sc && cd -"; -*-
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
//---------------`Snare drum synth` --------------------------
// A take at a snare drum synth
//
// Each hit is output to a channel <= channels as controlled by the lfo
// in rndctrl. Due to the ma.fabs, there is a greater chance that signal
// is sent to lower outputs than higher
//
// Where:
// * midi note 67-89
// * stiffness 0-0.55 (mapped to note as in note 67 -> 0)§
// * midi velocity 75-127
// * midi velocity is mapped to pressure
// A useful parameter setting is:
//
//
// 30 Juni 2018 Henrik Frisk [email protected]
//---------------------------------------------------
channels = 14;
integ(x) = x - ma.frac(x);
imp = ba.pulse(hslider("tempo", 1000, 500, 10000, 1));
// env = en.ar(0.000001, 0.1, button("play"));
env = en.ar(attack, rel, imp) * amp
with {
attack = hslider("attack", 0.00000001, 0, 0.1, 0.000000001) : si.smooth(0.1);
rel = hslider("rel", 0.1, 0.0000001, 0.5, 0.0000001) : si.smooth(0.2);
// imp = button("gate");
amp = hslider("vol", 0.5, 0, 1, 0.0001);
};
// Control the output channel
focus = hslider("focus", 1, 0, 1, 0.0001);
position = hslider("position", 1, 0, channels, 1);
rate = ma.SR/1000.0;
rndctrl = (no.lfnoise(rate) * (channels + 1)) * focus : ma.fabs + position : int ;
outputctrl = rndctrl : ba.sAndH(imp);
n = no.multinoise(8) : par(i, 8, _ * env * 0.1);
filt = fi.resonbp(frq, q, gain)
with {
frq = hslider("freq", 200, 50, 5000, 0.1);
q = hslider("q", 1, 0.01, 10, 0.01);
gain = hslider("gain", 0, 0, 2, 0.00001);
};
ch_wrapped = ma.modulo(outputctrl, channels);
process = n : par(i, 8, filt) :> ba.selectoutn(channels, ch_wrapped);
//process = n : par(i, 8, filt) :> _,_;
| https://raw.githubusercontent.com/friskgit/snares/bb43ea5e706a0ead6d65dd176a5c492b2f5d8f74/faust/snare/src/snare_multi.dsp | faust | -*- compile-command: "cd .. && make sc && cd -"; -*-
---------------`Snare drum synth` --------------------------
A take at a snare drum synth
Each hit is output to a channel <= channels as controlled by the lfo
in rndctrl. Due to the ma.fabs, there is a greater chance that signal
is sent to lower outputs than higher
Where:
* midi note 67-89
* stiffness 0-0.55 (mapped to note as in note 67 -> 0)§
* midi velocity 75-127
* midi velocity is mapped to pressure
A useful parameter setting is:
30 Juni 2018 Henrik Frisk [email protected]
---------------------------------------------------
env = en.ar(0.000001, 0.1, button("play"));
imp = button("gate");
Control the output channel
process = n : par(i, 8, filt) :> _,_; |
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
channels = 14;
integ(x) = x - ma.frac(x);
imp = ba.pulse(hslider("tempo", 1000, 500, 10000, 1));
env = en.ar(attack, rel, imp) * amp
with {
attack = hslider("attack", 0.00000001, 0, 0.1, 0.000000001) : si.smooth(0.1);
rel = hslider("rel", 0.1, 0.0000001, 0.5, 0.0000001) : si.smooth(0.2);
amp = hslider("vol", 0.5, 0, 1, 0.0001);
};
focus = hslider("focus", 1, 0, 1, 0.0001);
position = hslider("position", 1, 0, channels, 1);
rate = ma.SR/1000.0;
rndctrl = (no.lfnoise(rate) * (channels + 1)) * focus : ma.fabs + position : int ;
outputctrl = rndctrl : ba.sAndH(imp);
n = no.multinoise(8) : par(i, 8, _ * env * 0.1);
filt = fi.resonbp(frq, q, gain)
with {
frq = hslider("freq", 200, 50, 5000, 0.1);
q = hslider("q", 1, 0.01, 10, 0.01);
gain = hslider("gain", 0, 0, 2, 0.00001);
};
ch_wrapped = ma.modulo(outputctrl, channels);
process = n : par(i, 8, filt) :> ba.selectoutn(channels, ch_wrapped);
|
3075f484cc33c701fce57660447e394e40776ef34ff3efd01b6de8477aee6964 | friskgit/snares | i_snare_phase_disp.dsp | // -*- compile-command: "cd .. && make sc src=i_snare_phase_disp.dsp && cd -"; -*-&& cd -"; -*-
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
import("math.lib") ; // for PI definition
import("music.lib") ; // for osci definition
//---------------`Snare drum dispersing over X channels` --------------------------
//
// Generating an impulse and feeding it to a generic_snarefs. Each impulse is delayed by 25%
// and sent to a separate instance of generic_snarefs. This allows for faster impulses 4X the
// speed of the pulse which is in samples.
//
// disperse.dsp doe not pass on the impules as generic_snarefs does.
//
// 18 Juli 2019 Henrik Frisk [email protected]
//---------------------------------------------------
p = hslider("pulse[scale:exp]", 1, 1, 2000, 0.1) : si.smooth(0.5);
//per = hslider("pulse", 192000, 10, 192000, 1) : *(4);
per = ma.SR / p : int : *(4);
//p = hslider("pulse", 1, 1, 1000, 0.01);
imp = ba.pulse(per);
impsp = ba.pulse(per / 4);
// imp = os.imptrain(p);
delA = per : *(0.25);
delB = per : *(0.5);
delC = per : *(0.75);
imp_delA = imp : de.sdelay(192000, 64, delA);
imp_delB = imp : de.sdelay(192000, 64, delB);
imp_delC = imp : de.sdelay(192000, 64, delC);
//imp = os.impulse;
// divisor = 0.25;
// snares(d) = imp : de.sdelay(192000, 64, (per : *(divisor * d))) : component("generic_snarefs.dsp");
// process = par(i, 4, snares(i+1));
// n = (p / 20000) - 1 : ma.fabs : ma.log1p;
process = impsp, ((imp : component("generic_snarefs.dsp")),
(imp_delA : component("generic_snarefs.dsp")),
(imp_delB : component("generic_snarefs.dsp")),
(imp_delC : component("generic_snarefs.dsp")) :> _ ) : component("disperse.dsp")[channels = 29;];
| https://raw.githubusercontent.com/friskgit/snares/bb43ea5e706a0ead6d65dd176a5c492b2f5d8f74/faust/snare/src/i_snare_phase_disp.dsp | faust | -*- compile-command: "cd .. && make sc src=i_snare_phase_disp.dsp && cd -"; -*-&& cd -"; -*-
for PI definition
for osci definition
---------------`Snare drum dispersing over X channels` --------------------------
Generating an impulse and feeding it to a generic_snarefs. Each impulse is delayed by 25%
and sent to a separate instance of generic_snarefs. This allows for faster impulses 4X the
speed of the pulse which is in samples.
disperse.dsp doe not pass on the impules as generic_snarefs does.
18 Juli 2019 Henrik Frisk [email protected]
---------------------------------------------------
per = hslider("pulse", 192000, 10, 192000, 1) : *(4);
p = hslider("pulse", 1, 1, 1000, 0.01);
imp = os.imptrain(p);
imp = os.impulse;
divisor = 0.25;
snares(d) = imp : de.sdelay(192000, 64, (per : *(divisor * d))) : component("generic_snarefs.dsp");
process = par(i, 4, snares(i+1));
n = (p / 20000) - 1 : ma.fabs : ma.log1p; |
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
p = hslider("pulse[scale:exp]", 1, 1, 2000, 0.1) : si.smooth(0.5);
per = ma.SR / p : int : *(4);
imp = ba.pulse(per);
impsp = ba.pulse(per / 4);
delA = per : *(0.25);
delB = per : *(0.5);
delC = per : *(0.75);
imp_delA = imp : de.sdelay(192000, 64, delA);
imp_delB = imp : de.sdelay(192000, 64, delB);
imp_delC = imp : de.sdelay(192000, 64, delC);
process = impsp, ((imp : component("generic_snarefs.dsp")),
(imp_delA : component("generic_snarefs.dsp")),
(imp_delB : component("generic_snarefs.dsp")),
(imp_delC : component("generic_snarefs.dsp")) :> _ ) : component("disperse.dsp")[channels = 29;];
|
fc45fb557c5528aa7c201bec3f2c45de3ccac7234f7ff08249625869ac344d48 | afalaize/faust | karplus32.dsp | declare name "karplus32";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
//-----------------------------------------------
// karplus-strong
// with 32 resonators in parallel
//-----------------------------------------------
import("stdfaust.lib");
// Excitator
//-----------
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0)/n;
release(n) = + ~ decay(n);
trigger(n) = upfront : release(n) : >(0.0) : +(leak);
leak = 1.0/65536.0;
size = hslider("excitation (samples)", 128, 2, 512, 1);
// Resonator
//-----------
dur = hslider("duration (samples)", 128, 2, 512, 1);
att = hslider("attenuation", 0.1, 0, 1, 0.01);
average(x) = (x+x')/2;
resonator(d, a) = (+ : de.delay(4096, d-1.5)) ~ (average : *(1.0-a)) ;
// Polyphony
//-----------
detune = hslider("detune", 32, 0, 512, 1);
polyphony = hslider("polyphony", 1, 0, 32, 1);
output = hslider("output volume", 0.5, 0, 1, 0.1);
process = vgroup("karplus32",
vgroup("noise generator", no.noise * hslider("level", 0.5, 0, 1, 0.1))
: vgroup("excitator", *(button("play"): trigger(size)))
<: vgroup("resonator x32", par(i,32, resonator(dur+i*detune, att) * (polyphony > i)))
:> *(output),*(output)
);
| https://raw.githubusercontent.com/afalaize/faust/8f9f5fe3aa167eaeecc15a99d4da984ac2797be3/examples/physicalModeling/old/karplus32.dsp | faust | -----------------------------------------------
karplus-strong
with 32 resonators in parallel
-----------------------------------------------
Excitator
-----------
Resonator
-----------
Polyphony
----------- | declare name "karplus32";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
import("stdfaust.lib");
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0)/n;
release(n) = + ~ decay(n);
trigger(n) = upfront : release(n) : >(0.0) : +(leak);
leak = 1.0/65536.0;
size = hslider("excitation (samples)", 128, 2, 512, 1);
dur = hslider("duration (samples)", 128, 2, 512, 1);
att = hslider("attenuation", 0.1, 0, 1, 0.01);
average(x) = (x+x')/2;
resonator(d, a) = (+ : de.delay(4096, d-1.5)) ~ (average : *(1.0-a)) ;
detune = hslider("detune", 32, 0, 512, 1);
polyphony = hslider("polyphony", 1, 0, 32, 1);
output = hslider("output volume", 0.5, 0, 1, 0.1);
process = vgroup("karplus32",
vgroup("noise generator", no.noise * hslider("level", 0.5, 0, 1, 0.1))
: vgroup("excitator", *(button("play"): trigger(size)))
<: vgroup("resonator x32", par(i,32, resonator(dur+i*detune, att) * (polyphony > i)))
:> *(output),*(output)
);
|
5797d29298865cb574def425fecc17a7a7880d2bb85c3707d1c4adb72a9b4f7d | afalaize/faust | tester.dsp | declare name "tester";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
//-----------------------------------------------
// Tester : tests louspeakers
// Send a test signal( sine, noise, pink) to one
// of 8 loudspeakers
//-----------------------------------------------
import("stdfaust.lib");
// TODO: this should be rewritten with the pink noise function of noises.lib
pink = f : (+ ~ g) with {
f(x) = 0.04957526213389*x - 0.06305581334498*x' + 0.01483220320740*x'';
g(x) = 1.80116083982126*x - 0.80257737639225*x';
};
// User interface
//----------------
vol = hslider("[2] volume [unit:dB]", -96, -96, 0, 1): ba.db2linear : si.smoo;
freq = hslider("[1] freq [unit:Hz]", 1000, 10, 20000, 1);
dest = hslider("[3] destination", 0, 0, 8, 1);
testsignal = os.osci(freq)*checkbox("sine wave")
+ no.noise * checkbox("white noise")
+ pink(no.noise) * ba.db2linear(20) * checkbox("pink noise");
process = vgroup( "Audio Tester",
testsignal*vol
<: par(i, 8, *(dest==i))
);
| https://raw.githubusercontent.com/afalaize/faust/8f9f5fe3aa167eaeecc15a99d4da984ac2797be3/examples/misc/tester.dsp | faust | -----------------------------------------------
Tester : tests louspeakers
Send a test signal( sine, noise, pink) to one
of 8 loudspeakers
-----------------------------------------------
TODO: this should be rewritten with the pink noise function of noises.lib
User interface
---------------- | declare name "tester";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
import("stdfaust.lib");
pink = f : (+ ~ g) with {
f(x) = 0.04957526213389*x - 0.06305581334498*x' + 0.01483220320740*x'';
g(x) = 1.80116083982126*x - 0.80257737639225*x';
};
vol = hslider("[2] volume [unit:dB]", -96, -96, 0, 1): ba.db2linear : si.smoo;
freq = hslider("[1] freq [unit:Hz]", 1000, 10, 20000, 1);
dest = hslider("[3] destination", 0, 0, 8, 1);
testsignal = os.osci(freq)*checkbox("sine wave")
+ no.noise * checkbox("white noise")
+ pink(no.noise) * ba.db2linear(20) * checkbox("pink noise");
process = vgroup( "Audio Tester",
testsignal*vol
<: par(i, 8, *(dest==i))
);
|
cd2c4a4b7cb0e9c8ba9317175b2d4b1651357ad173f84ff64028d494cb3a0ad2 | sekisushai/ambitools | hoa_mirroring.dsp | declare name "HOA scene mirroring";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2015";
// Description: This tool mirros an HOA scene: Axes left-right, front-back, up-down can be reversed by changing a the sign of particular spherical harmonics [1,p.46-47]
// References:
// [1] Kronlachner, M. (2014). Spatial Transformations for the Alteration of Ambisonic Recordings. Graz University Of Technology, Austria.
// Inputs: (M+1)^2
// Outputs: (M+1)^2
import("stdfaust.lib");
ud=checkbox("up-down");
lr=checkbox("left-right");
fb=checkbox("front-back");
M = 10;
//up-down switching: all spherical harmonics with order and degree as m+n odd
updown = par(m,M+1,par(i,2*m+1,term
with{ term = _<:ba.if((ud==1) & (i%2==1),_*-1,_); // i+m^2 = m^2+m+n
})
);
//left-right switching : all spherical harmonics of degree n<0
leftright = par(m,M+1,par(i,2*m+1,term
with{ term = _<:ba.if((lr==1) & ((i-m)<0),_*-1,_); // i+m^2 = m^2+m+n
})
);
//front-back switching : all spherical harmonics of degree (n<0 & n even) or (n>0 & n odd)
frontback = par(m,M+1,par(i,2*m+1,term
with{ term = _<:
ba.if(
((fb==1)
&
(
(
((i-m)<0)
&
((i-m)%2==0)
)
|
(
((i-m)>0)
&
((i-m)%2==1)
)
)
)
,_*-1,_); // i+m^2 = m^2+m+n
})
);
process=vgroup("HOA scene mirroring",updown:leftright:frontback);
//EXAMPLE AT ORDER 5
//up-down switching : all spherical harmonics with order and degree as m+n odd
// updown=(
// _,
// _,u,_,
// _,u,_,u,_,
// _,u,_,u,_,u,_,
// _,u,_,u,_,u,_,u,_,
// _,u,_,u,_,u,_,u,_,u,_
// )
// with { u = (_<:ba.if(ud==1,_*-1,_)); };
//left-right switching : all spherical harmonics of degree n<0
// leftright=(
// _,
// l,_,_,
// l,l,_,_,_,
// l,l,l,_,_,_,_,
// l,l,l,l,_,_,_,_,_,
// l,l,l,l,l,_,_,_,_,_,_
// )
// with { l = (_<:ba.if(lr==1,_*-1,_)); };
//front-back switching : all spherical harmonics of degree (n<0 & n even) and (n>0 & n odd)
// frontback=(
// _,
// _,_,f,
// f,_,_,f,_,
// _,f,_,_,f,_,f,
// f,_,f,_,_,f,_,f,_,
// _,f,_,f,_,_,f,_,f,_,f
// )
// with { f = (_<:ba.if(fb==1,_*-1,_)); }; | https://raw.githubusercontent.com/sekisushai/ambitools/2d21b7cc7cfe9bc35d91d51ec05bf9250372f0ce/Faust/src/hoa_mirroring.dsp | faust | Description: This tool mirros an HOA scene: Axes left-right, front-back, up-down can be reversed by changing a the sign of particular spherical harmonics [1,p.46-47]
References:
[1] Kronlachner, M. (2014). Spatial Transformations for the Alteration of Ambisonic Recordings. Graz University Of Technology, Austria.
Inputs: (M+1)^2
Outputs: (M+1)^2
up-down switching: all spherical harmonics with order and degree as m+n odd
i+m^2 = m^2+m+n
left-right switching : all spherical harmonics of degree n<0
i+m^2 = m^2+m+n
front-back switching : all spherical harmonics of degree (n<0 & n even) or (n>0 & n odd)
i+m^2 = m^2+m+n
EXAMPLE AT ORDER 5
up-down switching : all spherical harmonics with order and degree as m+n odd
updown=(
_,
_,u,_,
_,u,_,u,_,
_,u,_,u,_,u,_,
_,u,_,u,_,u,_,u,_,
_,u,_,u,_,u,_,u,_,u,_
)
with { u = (_<:ba.if(ud==1,_*-1,_)); };
left-right switching : all spherical harmonics of degree n<0
leftright=(
_,
l,_,_,
l,l,_,_,_,
l,l,l,_,_,_,_,
l,l,l,l,_,_,_,_,_,
l,l,l,l,l,_,_,_,_,_,_
)
with { l = (_<:ba.if(lr==1,_*-1,_)); };
front-back switching : all spherical harmonics of degree (n<0 & n even) and (n>0 & n odd)
frontback=(
_,
_,_,f,
f,_,_,f,_,
_,f,_,_,f,_,f,
f,_,f,_,_,f,_,f,_,
_,f,_,f,_,_,f,_,f,_,f
)
with { f = (_<:ba.if(fb==1,_*-1,_)); }; | declare name "HOA scene mirroring";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2015";
import("stdfaust.lib");
ud=checkbox("up-down");
lr=checkbox("left-right");
fb=checkbox("front-back");
M = 10;
updown = par(m,M+1,par(i,2*m+1,term
})
);
leftright = par(m,M+1,par(i,2*m+1,term
})
);
frontback = par(m,M+1,par(i,2*m+1,term
with{ term = _<:
ba.if(
((fb==1)
&
(
(
((i-m)<0)
&
((i-m)%2==0)
)
|
(
((i-m)>0)
&
((i-m)%2==1)
)
)
)
})
);
process=vgroup("HOA scene mirroring",updown:leftright:frontback);
|
20670b6d9a323c7c096899e859420c369ec031a9ae0936c78c97ca7c410d550a | sekisushai/ambitools | hoa_mic_encoder_eigenmike32.dsp | declare name "eigenmike32 HOA Encoder";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2015";
// Description: This tool does the Discrete Spherical Fourier Transform (DSFT) [1] of signal from a rigid spherical microphone with Pentakis dodecahedron geometry (mh acoustics eigenmike32 [1]).
// CAUTION: This tool does only the DSFT, to obtain the Ambisonics components you should filter the signals with corresponding radial filters [2].
// Inputs: 32
// Outputs: (M+1)^2
// References:
// [1] G. Elko, R. A. Kubli, and J. Meyer, “Audio system based on at least second-order eigenbeams,” 2009.
// [2] S. Moreau, J. Daniel, and S. Bertet, “3d sound field recording with higher order ambisonics-objective measurements and validation of spherical microphone,” in Audio Engineering Society Convention 120, 2006, pp. 1–24.
// [3] P. Lecomte, P.-A. Gauthier, C. Langrenne, A. Garcia, and A. Berry, “On the use of a Lebedev grid for Ambisonics,” in Audio Engineering Society Convention 139, 2015.
import("stdfaust.lib");
import("eigenmike32.lib");
import("ymn.lib");
import("gui.lib");
// Maximum order M=4 for eigenmike32 [2].
M = 4;
ins = 32;
outs = (M+1)^2;
vol = hslider("[1]Gain[unit:dB][style:knob]", 0, -10, 50, 0.1) : ba.db2linear : si.smooth(0.999);
// Vector of weighted spherical harmonics : spherical harmonics times the speaker weight for weighed quadrature rules [3].
row(i) = par(j,ins,yacn(i,azimuth(j),elevation(j))*weight(j));
process = hgroup("[0]Inputs",par(i,ins,id(i,0)))<:par(i,outs,buswg(row(i)):>_):hgroup("[1]Outputs",par(i,outs,*(vol)):par(i,M+1,metermute(i))); | https://raw.githubusercontent.com/sekisushai/ambitools/2d21b7cc7cfe9bc35d91d51ec05bf9250372f0ce/Faust/src/hoa_mic_encoder_eigenmike32.dsp | faust | Description: This tool does the Discrete Spherical Fourier Transform (DSFT) [1] of signal from a rigid spherical microphone with Pentakis dodecahedron geometry (mh acoustics eigenmike32 [1]).
CAUTION: This tool does only the DSFT, to obtain the Ambisonics components you should filter the signals with corresponding radial filters [2].
Inputs: 32
Outputs: (M+1)^2
References:
[1] G. Elko, R. A. Kubli, and J. Meyer, “Audio system based on at least second-order eigenbeams,” 2009.
[2] S. Moreau, J. Daniel, and S. Bertet, “3d sound field recording with higher order ambisonics-objective measurements and validation of spherical microphone,” in Audio Engineering Society Convention 120, 2006, pp. 1–24.
[3] P. Lecomte, P.-A. Gauthier, C. Langrenne, A. Garcia, and A. Berry, “On the use of a Lebedev grid for Ambisonics,” in Audio Engineering Society Convention 139, 2015.
Maximum order M=4 for eigenmike32 [2].
Vector of weighted spherical harmonics : spherical harmonics times the speaker weight for weighed quadrature rules [3]. | declare name "eigenmike32 HOA Encoder";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2015";
import("stdfaust.lib");
import("eigenmike32.lib");
import("ymn.lib");
import("gui.lib");
M = 4;
ins = 32;
outs = (M+1)^2;
vol = hslider("[1]Gain[unit:dB][style:knob]", 0, -10, 50, 0.1) : ba.db2linear : si.smooth(0.999);
row(i) = par(j,ins,yacn(i,azimuth(j),elevation(j))*weight(j));
process = hgroup("[0]Inputs",par(i,ins,id(i,0)))<:par(i,outs,buswg(row(i)):>_):hgroup("[1]Outputs",par(i,outs,*(vol)):par(i,M+1,metermute(i))); |
9dff396c75de403d77397f5ab50145529afed191877d3b39170ebacdddf9b9ea | sekisushai/ambitools | hoa_converter_acn_n3d_to_fuma.dsp | declare name "HOA Converter : ACN N3D to FuMa";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2016";
import("stdfaust.lib");
import("gui.lib");
//Description : this tool converts HOA signals defined with a convention 1 to HOA signals defined with convention 2. Proposed conventions are ACN N3D, ACN SN3D, FuMa. For ACN to FuMa, the ordering change is as in [1]
//[1] https://en.wikipedia.org/wiki/Ambisonic_data_exchange_formats
// Input ACN: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
// Output FuMa: 0 3 1 2 6 7 5 8 4 12 13 11 14 10 15 9 : W XYZ RSTUV KLMNOPQ
// Input FuMa: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 : W XYZ RSTUV KLMNOPQ
// Output ACN: 0 2 3 1 6 8 4 5 7 15 13 11 9 10 12 14
// Maximum required order (M = 3 for FuMa).
M = 3;
// Number of inputs
ins = (M+1)^2;
outs = ins;
// ACN_N3D Input
conversion(1,3) = par(i,M+1,ACNFuMa(i)); // ACN_N3D to FuMa
ACNFuMa(0) = _*(1/sqrt(2));
ACNFuMa(1) = ro.cross(3):(_,ro.cross(2)):
(_*(1/sqrt(3)),_*(1/sqrt(3)),_*(1/sqrt(3)));
ACNFuMa(2) = (ro.cross(3),_,_):(_,ro.cross(3),_):(_,_,ro.cross(2),_):(_,_,_,ro.cross(2)):
(_*(1/sqrt(5)),_*(2/sqrt(15)),_*(2/sqrt(15)),_*(2/sqrt(15)),_*(2/sqrt(15)));
ACNFuMa(3) = (ro.cross(4),_,_,_):(_,ro.cross(4),_,_):(_,_,ro.cross(3),_,_):(_,_,_,ro.cross(3),_):(_,_,_,_,ro.cross(2),_):(_,_,_,_,_,ro.cross(2)):
(_*(1/sqrt(7)),_*sqrt(45/224),_*sqrt(45/224),_*(3/sqrt(35)),_*(3/sqrt(35)),_*sqrt(8/35),_*sqrt(8/35));
ACNFuMa(m) = par(i,2*m+1,!:0);
process = si.bus(ins):hgroup("[1]ACN N3D",par(i,M+1,meterm(i))):conversion(1,3):hgroup("[2]FuMa",par(i,M+1,meterm(i))); | https://raw.githubusercontent.com/sekisushai/ambitools/2d21b7cc7cfe9bc35d91d51ec05bf9250372f0ce/Faust/src/hoa_converter_acn_n3d_to_fuma.dsp | faust | Description : this tool converts HOA signals defined with a convention 1 to HOA signals defined with convention 2. Proposed conventions are ACN N3D, ACN SN3D, FuMa. For ACN to FuMa, the ordering change is as in [1]
[1] https://en.wikipedia.org/wiki/Ambisonic_data_exchange_formats
Input ACN: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Output FuMa: 0 3 1 2 6 7 5 8 4 12 13 11 14 10 15 9 : W XYZ RSTUV KLMNOPQ
Input FuMa: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 : W XYZ RSTUV KLMNOPQ
Output ACN: 0 2 3 1 6 8 4 5 7 15 13 11 9 10 12 14
Maximum required order (M = 3 for FuMa).
Number of inputs
ACN_N3D Input
ACN_N3D to FuMa | declare name "HOA Converter : ACN N3D to FuMa";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2016";
import("stdfaust.lib");
import("gui.lib");
M = 3;
ins = (M+1)^2;
outs = ins;
ACNFuMa(0) = _*(1/sqrt(2));
ACNFuMa(1) = ro.cross(3):(_,ro.cross(2)):
(_*(1/sqrt(3)),_*(1/sqrt(3)),_*(1/sqrt(3)));
ACNFuMa(2) = (ro.cross(3),_,_):(_,ro.cross(3),_):(_,_,ro.cross(2),_):(_,_,_,ro.cross(2)):
(_*(1/sqrt(5)),_*(2/sqrt(15)),_*(2/sqrt(15)),_*(2/sqrt(15)),_*(2/sqrt(15)));
ACNFuMa(3) = (ro.cross(4),_,_,_):(_,ro.cross(4),_,_):(_,_,ro.cross(3),_,_):(_,_,_,ro.cross(3),_):(_,_,_,_,ro.cross(2),_):(_,_,_,_,_,ro.cross(2)):
(_*(1/sqrt(7)),_*sqrt(45/224),_*sqrt(45/224),_*(3/sqrt(35)),_*(3/sqrt(35)),_*sqrt(8/35),_*sqrt(8/35));
ACNFuMa(m) = par(i,2*m+1,!:0);
process = si.bus(ins):hgroup("[1]ACN N3D",par(i,M+1,meterm(i))):conversion(1,3):hgroup("[2]FuMa",par(i,M+1,meterm(i))); |
d9bd8ffe5f963d03aff26d6d3c03f601bd7d3ca5437a2a7253a11225d76ea474 | tonal-glyph/faustus | dbmeter.dsp | declare name "dbmeter";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
//-------------------------------------------------
// A dB Vumeter
//-------------------------------------------------
import("stdfaust.lib");
vmeter(x) = attach(x, envelop(x) : vbargraph("[unit:dB]", -70, 10));
hmeter(x) = attach(x, envelop(x) : hbargraph("[unit:dB]", -70, 10));
envelop = abs : max(ba.db2linear(-70)) : ba.linear2db : min(10) : max ~ -(80.0/ma.SR);
null(x) = attach(0,x);
process = hgroup("8 channels dB meter", par(i,8, vgroup("%i", vmeter : null)));
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/analysis/dbmeter.dsp | faust | -------------------------------------------------
A dB Vumeter
------------------------------------------------- | declare name "dbmeter";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
import("stdfaust.lib");
vmeter(x) = attach(x, envelop(x) : vbargraph("[unit:dB]", -70, 10));
hmeter(x) = attach(x, envelop(x) : hbargraph("[unit:dB]", -70, 10));
envelop = abs : max(ba.db2linear(-70)) : ba.linear2db : min(10) : max ~ -(80.0/ma.SR);
null(x) = attach(0,x);
process = hgroup("8 channels dB meter", par(i,8, vgroup("%i", vmeter : null)));
|
66a7ae4236b0b8f6fd8024b9bde45921b902b71253baad1cd8165866a9ea1c9f | LucaSpanedda/Musical_Plugins | Granulator2HEADS.dsp | declare name "Timestretching Overlap Add to One";
declare version "xxx";
declare author "Luca Spanedda";
declare license "GNU-GPL-v3";
declare copyright "(c)Luca Spanedda 2022";
declare description "Overlap Add to One Timestretcher";
// FAUST standard library
import("stdfaust.lib");
// Phasor Function
LetrecPhasor(f) = Xn
letrec{
'Xn = (Xn+(f/ma.SR))-int(Xn);
};
// Sample and Hold Circuit: Sig --> circuit(control signal for trigger)
SAH(trig,x) = loop
with{
loop = ((_,x) : Selector(trig))~_
with{
Selector(sel) = (_*(1-sel)+_*(sel));
};
};
floorint(x) = x-int(x);
// read section
// stretchFactor - 0 Normal / 1 Extreme stretch (Freeze)
stretch = LetrecPhasor(1-hslider("[7]Stretch Factor", 0, 0, 1, .001))
* hslider("[5]Read Section", 0, 0, 1, .001);
// Jitter Amount in the position for the reads
jitter = no.noise*hslider("[6]Read Jitter", 0, 0, 1, .001);
// position in the Buffer for the Reads
cntrlRead = hslider("[4]Read Position", 0, 0, 1, .001)+stretch+jitter : floorint;
// Timestretcher - sum of the 2 Head Reads
// Bufpos = 0 to 1 signals for the reads
timeStretcher(bufPos, x) = x <: head1 + head2 <: _,_, rIdxgraph, wIdxgraph
with{
offset = 2;
// tableMax = table Max Dimension - 10 Seconds
tableMax = 192000 * 10 + offset;
// L = buffer dimension in seconds
L = ma.SR * hslider("[2]Table Dimension[unit:Sec]", 1, 1, 10, 1);
// Write index - ramp 0 to L
wIdx = offset + ((+(1) : %(L)) ~ _) * checkbox("[3]Record") : int;
buffer(p, x) = it.frwtable(3, tableMax, .0, wIdx, x, p);
// Hanning window Equation
hann(x) = sin(ma.frac(x) * ma.PI) ^ 2.0;
// Grain in Milliseconds
grainms = 1000/hslider("[8]Grain Dimension[unit:ms.]", 80, 1, 1000, 1) : si.smoo;
// Position of the grain in the Buffer
timePhase = offset + (bufPos * L);
// two Heads for the read
// 0°
ph1 = LetrecPhasor(grainms);
// 180*
ph2 = ma.frac(.5 + ph1);
// Buffer positions = Position in the Buffer + Grain Read
pos1 = SAH(ph1 < ph1', timePhase) + ph1*(ma.SR/grainms);
pos2 = SAH(ph2 < ph2', timePhase) + ph2*(ma.SR/grainms);
// Windows + Buffer Reads
head1 = hann(ph1) * buffer(pos1);
head2 = hann(ph2) * buffer(pos2);
wIdxgraph = (wIdx/L) : hbargraph("[0]Write Head",0,1) : _*ma.EPSILON;
rIdxgraph = bufPos : hbargraph("[1]Read Head",0,1) : _*ma.EPSILON;
};
process = timeStretcher(cntrlRead); | https://raw.githubusercontent.com/LucaSpanedda/Musical_Plugins/d17556035378ac5c6a0a2033de7f0a93390e94e1/Granulator%202%20Heads/Granulator2HEADS.dsp | faust | FAUST standard library
Phasor Function
Sample and Hold Circuit: Sig --> circuit(control signal for trigger)
read section
stretchFactor - 0 Normal / 1 Extreme stretch (Freeze)
Jitter Amount in the position for the reads
position in the Buffer for the Reads
Timestretcher - sum of the 2 Head Reads
Bufpos = 0 to 1 signals for the reads
tableMax = table Max Dimension - 10 Seconds
L = buffer dimension in seconds
Write index - ramp 0 to L
Hanning window Equation
Grain in Milliseconds
Position of the grain in the Buffer
two Heads for the read
0°
180*
Buffer positions = Position in the Buffer + Grain Read
Windows + Buffer Reads
| declare name "Timestretching Overlap Add to One";
declare version "xxx";
declare author "Luca Spanedda";
declare license "GNU-GPL-v3";
declare copyright "(c)Luca Spanedda 2022";
declare description "Overlap Add to One Timestretcher";
import("stdfaust.lib");
LetrecPhasor(f) = Xn
letrec{
'Xn = (Xn+(f/ma.SR))-int(Xn);
};
SAH(trig,x) = loop
with{
loop = ((_,x) : Selector(trig))~_
with{
Selector(sel) = (_*(1-sel)+_*(sel));
};
};
floorint(x) = x-int(x);
stretch = LetrecPhasor(1-hslider("[7]Stretch Factor", 0, 0, 1, .001))
* hslider("[5]Read Section", 0, 0, 1, .001);
jitter = no.noise*hslider("[6]Read Jitter", 0, 0, 1, .001);
cntrlRead = hslider("[4]Read Position", 0, 0, 1, .001)+stretch+jitter : floorint;
timeStretcher(bufPos, x) = x <: head1 + head2 <: _,_, rIdxgraph, wIdxgraph
with{
offset = 2;
tableMax = 192000 * 10 + offset;
L = ma.SR * hslider("[2]Table Dimension[unit:Sec]", 1, 1, 10, 1);
wIdx = offset + ((+(1) : %(L)) ~ _) * checkbox("[3]Record") : int;
buffer(p, x) = it.frwtable(3, tableMax, .0, wIdx, x, p);
hann(x) = sin(ma.frac(x) * ma.PI) ^ 2.0;
grainms = 1000/hslider("[8]Grain Dimension[unit:ms.]", 80, 1, 1000, 1) : si.smoo;
timePhase = offset + (bufPos * L);
ph1 = LetrecPhasor(grainms);
ph2 = ma.frac(.5 + ph1);
pos1 = SAH(ph1 < ph1', timePhase) + ph1*(ma.SR/grainms);
pos2 = SAH(ph2 < ph2', timePhase) + ph2*(ma.SR/grainms);
head1 = hann(ph1) * buffer(pos1);
head2 = hann(ph2) * buffer(pos2);
wIdxgraph = (wIdx/L) : hbargraph("[0]Write Head",0,1) : _*ma.EPSILON;
rIdxgraph = bufPos : hbargraph("[1]Read Head",0,1) : _*ma.EPSILON;
};
process = timeStretcher(cntrlRead); |
efac903b3af6aa55762cd96c2e7d46d3940a020474f7dc5823d175f1d0d50d30 | rottingsounds/bitDSP-faust | bitRot.dsp | declare name "bitRot";
declare description "bitRot - example";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
// plot
// CXXFLAGS="-I ../include" faust2csvplot -I ../lib bitRot.dsp
// ./bitRot -n 10
// compile
// CXXFLAGS="-I ../../../include" faust2caqt -I ../lib bitRot.dsp
// ./bitRot
/////////////////////////// UI ////////////////////////////////
chance = hslider("chance", 0, 0, 1, 0.0001);
type = hslider("type" , 1, 1, 3, 1);
amp = hslider("amp" , 0, 0, 1, 0.001) : si.smoo;
/////////////////////////// Input /////////////////////////////
bitSin = bit.ddsm1(os.osc(50));
dirac = 1 - 1';
silence = fbPath(dirac) ~ _ with {
fbPath(init, fb) = -fb + init;
};
// noise needs to be unimodal
noise = (no.noise + 1) * 0.5;
// process = bitSin <: _, bitrot(noise, chance, type) : bit2mbit, bit2mbit with {
// bit2mbit(x) = fi.lowpass(4, 4000, x);
// };
process = bitSin : bit.bitrot(noise, chance, type) : outPCM(2, amp) with {
outPCM(N, amp, x) = fi.lowpass(2, 4000, x) * amp : leakdc(0.999) <: si.bus(N);
leakdc(coef, x) = y letrec {
'y = x - x' + coef * y;
};
};
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/21cf36105c55b6e18969a867a319530a0ef1ea63/examples/bitRot.dsp | faust | plot
CXXFLAGS="-I ../include" faust2csvplot -I ../lib bitRot.dsp
./bitRot -n 10
compile
CXXFLAGS="-I ../../../include" faust2caqt -I ../lib bitRot.dsp
./bitRot
///////////////////////// UI ////////////////////////////////
///////////////////////// Input /////////////////////////////
noise needs to be unimodal
process = bitSin <: _, bitrot(noise, chance, type) : bit2mbit, bit2mbit with {
bit2mbit(x) = fi.lowpass(4, 4000, x);
}; | declare name "bitRot";
declare description "bitRot - example";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
chance = hslider("chance", 0, 0, 1, 0.0001);
type = hslider("type" , 1, 1, 3, 1);
amp = hslider("amp" , 0, 0, 1, 0.001) : si.smoo;
bitSin = bit.ddsm1(os.osc(50));
dirac = 1 - 1';
silence = fbPath(dirac) ~ _ with {
fbPath(init, fb) = -fb + init;
};
noise = (no.noise + 1) * 0.5;
process = bitSin : bit.bitrot(noise, chance, type) : outPCM(2, amp) with {
outPCM(N, amp, x) = fi.lowpass(2, 4000, x) * amp : leakdc(0.999) <: si.bus(N);
leakdc(coef, x) = y letrec {
'y = x - x' + coef * y;
};
};
|
91c96e1008ba5b4d8eeadf7292d79d9a4e6a1941d5ef0163a110dca368d34a10 | grame-cncm/GameLAN | Sequenceur.dsp | declare name "Sequenceur";
declare author "Developpement Grame - CNCM par Elodie Rabibisoa et Romain Constant.";
import ("stdfaust.lib");
process = sequenceur * onOff <:_,_;
N = 4;
onOff = checkbox("[1]ON/OFF");
sequenceur = par(i, N, play(sample(i), file_index, i) * (sample_pick == i)) :>_ ;
seq_select = ba.beat(bps) : ba.pulse_countup_loop(15, 1) : hbargraph("Suivi",0,15);
bps = bpm.a25*(typeBpm == 7)+
bpm.b37*(typeBpm == 6) +
bpm.c50*(typeBpm == 5) +
bpm.d62*(typeBpm == 4) +
bpm.e75*(typeBpm == 3) +
bpm.f100*(typeBpm == 2) +
bpm.g125*(typeBpm == 1) +
bpm.h150*(typeBpm == 0);
typeBpm = vslider("[1]Tempo [style:radio{'150 BPM':0;'125 BPM':1;'100 BPM':2;'75 BPM':3;'62.5 BPM':4;'50 BPM':5;'37.5 BPM':6;'25 BPM':7}]", 0, 0, 7, 1):int;
bpm = environment { // bpm * 4, semiquaver (16th)
a25 = 25 * 4;
b37 = 37.5 * 4;
c50 = 50 * 4;
d62 = 62.5 * 4;
e75 = 75 * 4;
f100 = 100 * 4;
g125 = 125 * 4;
h150 = 150 * 4;
};
check(0) = checkbox("h:[04]/01") * 1;
check(1) = checkbox("h:[04]/02") * 2;
check(2) = checkbox("h:[05]/03") * 3;
check(3) = checkbox("h:[05]/04") * 4;
check(4) = checkbox("h:[06]/05") * 5;
check(5) = checkbox("h:[06]/06") * 6;
check(6) = checkbox("h:[07]/07") * 7;
check(7) = checkbox("h:[07]/08") * 8;
check(8) = checkbox("h:[08]/09") * 9;
check(9) = checkbox("h:[08]/10") * 10;
check(10) = checkbox("h:[09]/11") * 11;
check(11) = checkbox("h:[09]/12") * 12;
check(12) = checkbox("h:[10]/13") * 13;
check(13) = checkbox("h:[10]/14") * 14;
check(14) = checkbox("h:[11]/15") * 15;
check(15) = checkbox("h:[11]/16") * 16;
// ------------------------------------ Samples -------------------------------------
sample_pick = hslider("[0]Samples[style:radio{'Bip Square':0;'Hi-Hat':1;'Kick':2;'Snare':3}]", 0, 0, 3, 1);
sample(0) = soundfile("sample_2 [url:bipsquare_oneshot.flac]", 1);
sample(1) = soundfile("sample_4 [url:Hihat_oneshot_N.flac]", 1);
sample(2) = soundfile("sample_6 [url:Kick_oneshot_N.flac]", 1);
sample(3) = soundfile("sample_8 [url:Snare_oneshot_N.flac]", 1);
// ------------------------------------ Player --------------------------------------
file_index = 0;
trigger = par(i, 16, vgroup("[3]Steps",check(i)) == (seq_select + 1) : upfront) :>_;
upfront(x) = (x-x')>0.5;
counter(sampleSize) = trigger : decrease > (0.0)
with { //trig impulse to launch stream of 1
decay(y) = y - (y>0.0)/sampleDuration;
decrease = +~decay;
sampleDuration = hslider("Decay[acc:0 0 -8 0 8][hidden:1]", 1, 0.001, 1.5, 0.001) * (ma.SR): min(sampleSize) : max(44) : int;
};
index(sampleSize) = +(counter(sampleSize))~_ * (1 - (trigger : upfront)) : int; //increment loop with reinit to 0 through reversed impulse (trig : upfront)
play(s, part) = (part, reader(s)) : outs(s)
with {
length(s) = part,0 : s : _,si.block(outputs(s)-1);
srate(s) = part,0 : s : !,_,si.block(outputs(s)-2);
outs(s) = s : si.block(2), si.bus(outputs(s)-2);
reader(s,n) = index(length(s));
};
| https://raw.githubusercontent.com/grame-cncm/GameLAN/8d1dc26d709d721d27ec1156fbb66b03478f2529/sequenceur/Sequenceur.dsp | faust | bpm * 4, semiquaver (16th)
------------------------------------ Samples -------------------------------------
------------------------------------ Player --------------------------------------
trig impulse to launch stream of 1
increment loop with reinit to 0 through reversed impulse (trig : upfront) | declare name "Sequenceur";
declare author "Developpement Grame - CNCM par Elodie Rabibisoa et Romain Constant.";
import ("stdfaust.lib");
process = sequenceur * onOff <:_,_;
N = 4;
onOff = checkbox("[1]ON/OFF");
sequenceur = par(i, N, play(sample(i), file_index, i) * (sample_pick == i)) :>_ ;
seq_select = ba.beat(bps) : ba.pulse_countup_loop(15, 1) : hbargraph("Suivi",0,15);
bps = bpm.a25*(typeBpm == 7)+
bpm.b37*(typeBpm == 6) +
bpm.c50*(typeBpm == 5) +
bpm.d62*(typeBpm == 4) +
bpm.e75*(typeBpm == 3) +
bpm.f100*(typeBpm == 2) +
bpm.g125*(typeBpm == 1) +
bpm.h150*(typeBpm == 0);
typeBpm = vslider("[1]Tempo [style:radio{'150 BPM':0;'125 BPM':1;'100 BPM':2;'75 BPM':3;'62.5 BPM':4;'50 BPM':5;'37.5 BPM':6;'25 BPM':7}]", 0, 0, 7, 1):int;
a25 = 25 * 4;
b37 = 37.5 * 4;
c50 = 50 * 4;
d62 = 62.5 * 4;
e75 = 75 * 4;
f100 = 100 * 4;
g125 = 125 * 4;
h150 = 150 * 4;
};
check(0) = checkbox("h:[04]/01") * 1;
check(1) = checkbox("h:[04]/02") * 2;
check(2) = checkbox("h:[05]/03") * 3;
check(3) = checkbox("h:[05]/04") * 4;
check(4) = checkbox("h:[06]/05") * 5;
check(5) = checkbox("h:[06]/06") * 6;
check(6) = checkbox("h:[07]/07") * 7;
check(7) = checkbox("h:[07]/08") * 8;
check(8) = checkbox("h:[08]/09") * 9;
check(9) = checkbox("h:[08]/10") * 10;
check(10) = checkbox("h:[09]/11") * 11;
check(11) = checkbox("h:[09]/12") * 12;
check(12) = checkbox("h:[10]/13") * 13;
check(13) = checkbox("h:[10]/14") * 14;
check(14) = checkbox("h:[11]/15") * 15;
check(15) = checkbox("h:[11]/16") * 16;
sample_pick = hslider("[0]Samples[style:radio{'Bip Square':0;'Hi-Hat':1;'Kick':2;'Snare':3}]", 0, 0, 3, 1);
sample(0) = soundfile("sample_2 [url:bipsquare_oneshot.flac]", 1);
sample(1) = soundfile("sample_4 [url:Hihat_oneshot_N.flac]", 1);
sample(2) = soundfile("sample_6 [url:Kick_oneshot_N.flac]", 1);
sample(3) = soundfile("sample_8 [url:Snare_oneshot_N.flac]", 1);
file_index = 0;
trigger = par(i, 16, vgroup("[3]Steps",check(i)) == (seq_select + 1) : upfront) :>_;
upfront(x) = (x-x')>0.5;
counter(sampleSize) = trigger : decrease > (0.0)
decay(y) = y - (y>0.0)/sampleDuration;
decrease = +~decay;
sampleDuration = hslider("Decay[acc:0 0 -8 0 8][hidden:1]", 1, 0.001, 1.5, 0.001) * (ma.SR): min(sampleSize) : max(44) : int;
};
play(s, part) = (part, reader(s)) : outs(s)
with {
length(s) = part,0 : s : _,si.block(outputs(s)-1);
srate(s) = part,0 : s : !,_,si.block(outputs(s)-2);
outs(s) = s : si.block(2), si.bus(outputs(s)-2);
reader(s,n) = index(length(s));
};
|
a0d0bf6adf5b9d830938bb6cdd6241c56852e6e79aa46a6e4a5f431fa03c89b9 | grame-cncm/GameLAN | AttacKey.dsp | declare name "Attackey";
declare author "Developpement Grame - CNCM par Elodie Rabibisoa, Romain Constant et Stéphane Letz.";
import("stdfaust.lib");
declare nvoices "12";
// Specific syntax for faust2android, [style:keyboard] doesn't exist in iOS
process = vgroup("Attackey [style:keyboard]", instru);
freq = hslider("freq", 349.23, 261.63, 783.99, 0.001);
gain = hslider("gain",0.5,0,1,0.01);
gate = button("gate");
envelope = en.adsr(0.01,0.01,0.9,0.1,gate)*gain;
instru = play1(noteOn, instrument) * envelope * volume : attackey_reverb * 0.5 <:_,_;
instrument = hslider("Instruments[style:radio{'1':0;'2':1;'3':2;'4':3;'5':4}]", 0, 0, 4, 1);
volume = hslider("Volume [acc: 0 0 -8 0 0][hidden:1]", 1, 0, 1, 0.001):si.smoo;
noteOn = soundfile("Instrus [url:{'Piano_F.flac';'Ether_F.flac';'Bell_F.flac';'Saw_F.flac';'Vibraphone_F.flac'}]", 1);
// -------------------- Interpolation Players ------------------- //
speed = freq/(349.23*2); //reference pitch = F * 2 (midi keyboard plays one octave higher)
srate(s, part) = part,0 : s : !,_,si.block(outputs(s)-2):float;
// Reset when button is pressed (0 when trig is on, 1 when trig is off)
reset(trig) = (trig-trig') <= 0;
// Ramp with a given step, reset when trig is on
ramp(trig, step) = (+(step):*(reset(trig))) ~ _;
// Outputs
outs(s, level) = s : si.block(2), bus_level(outputs(s)-2) with { bus_level(n) = par(i,n,*(level)); };
// Plays a soundfile given a parametric 'reader'
player(s, part, reader, level) = (part, reader(s,part)) : outs(s,level);
// Plays a soundfile given a parametric 'reader' with linear interpolation
linear_player(s, part, reader, level) = (lplayer(id0), lplayer(id1))
: ro.interleave(sound_outs, 2)
: par(i, sound_outs, linear(c))
with {
lplayer(reader) = (part, reader) : outs(s, level);
reader1 = reader(s, part);
id0 = int(reader1);
id1 = id0 + 1;
c = reader1 - id0;
sound_outs = outputs(s)-2;
linear(c,v0,v1) = v0*(1-c)+v1*c;
};
// Plays a soundfile given a parametric 'reader' with cubic interpolation
cubic_player(s, part, reader, level)
= (lplayer(id0), lplayer(id1), lplayer(id2), lplayer(id3))
: ro.interleave(sound_outs, 4)
: par(i, sound_outs, cubic(c))
with {
lplayer(reader) = (part, reader) : outs(s, level);
reader1 = reader(s, part);
id0 = int(reader1);
id1 = id0 + 1;
id2 = id1 + 1;
id3 = id2 + 1;
c = reader1 - id0;
sound_outs = outputs(s)-2;
cubic(c,v0,v1,v2,v3) = v1 + 0.5 * c * (v2 - v0 + c * (2.0*v0 - 5.0*v1 + 4.0*v2 - v3 + c*(3.0*(v1 -v2) + v3 - v0)));
};
fullsample_reader(gate) = \(s,part).(ramp(gate, speed*srate(s,part)/ma.SR));
play1(s, part) = cubic_player(s, part, fullsample_reader(gate), 1);
// -------------------- Reverb ------------------- //
attackey_reverb = _<: instrReverb :>_;
instrReverb = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = 1;
roomSize = 1;
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/grame-cncm/GameLAN/8d1dc26d709d721d27ec1156fbb66b03478f2529/attacKey/AttacKey.dsp | faust | Specific syntax for faust2android, [style:keyboard] doesn't exist in iOS
-------------------- Interpolation Players ------------------- //
reference pitch = F * 2 (midi keyboard plays one octave higher)
Reset when button is pressed (0 when trig is on, 1 when trig is off)
Ramp with a given step, reset when trig is on
Outputs
Plays a soundfile given a parametric 'reader'
Plays a soundfile given a parametric 'reader' with linear interpolation
Plays a soundfile given a parametric 'reader' with cubic interpolation
-------------------- Reverb ------------------- //
| declare name "Attackey";
declare author "Developpement Grame - CNCM par Elodie Rabibisoa, Romain Constant et Stéphane Letz.";
import("stdfaust.lib");
declare nvoices "12";
process = vgroup("Attackey [style:keyboard]", instru);
freq = hslider("freq", 349.23, 261.63, 783.99, 0.001);
gain = hslider("gain",0.5,0,1,0.01);
gate = button("gate");
envelope = en.adsr(0.01,0.01,0.9,0.1,gate)*gain;
instru = play1(noteOn, instrument) * envelope * volume : attackey_reverb * 0.5 <:_,_;
instrument = hslider("Instruments[style:radio{'1':0;'2':1;'3':2;'4':3;'5':4}]", 0, 0, 4, 1);
volume = hslider("Volume [acc: 0 0 -8 0 0][hidden:1]", 1, 0, 1, 0.001):si.smoo;
noteOn = soundfile("Instrus [url:{'Piano_F.flac';'Ether_F.flac';'Bell_F.flac';'Saw_F.flac';'Vibraphone_F.flac'}]", 1);
srate(s, part) = part,0 : s : !,_,si.block(outputs(s)-2):float;
reset(trig) = (trig-trig') <= 0;
ramp(trig, step) = (+(step):*(reset(trig))) ~ _;
outs(s, level) = s : si.block(2), bus_level(outputs(s)-2) with { bus_level(n) = par(i,n,*(level)); };
player(s, part, reader, level) = (part, reader(s,part)) : outs(s,level);
linear_player(s, part, reader, level) = (lplayer(id0), lplayer(id1))
: ro.interleave(sound_outs, 2)
: par(i, sound_outs, linear(c))
with {
lplayer(reader) = (part, reader) : outs(s, level);
reader1 = reader(s, part);
id0 = int(reader1);
id1 = id0 + 1;
c = reader1 - id0;
sound_outs = outputs(s)-2;
linear(c,v0,v1) = v0*(1-c)+v1*c;
};
cubic_player(s, part, reader, level)
= (lplayer(id0), lplayer(id1), lplayer(id2), lplayer(id3))
: ro.interleave(sound_outs, 4)
: par(i, sound_outs, cubic(c))
with {
lplayer(reader) = (part, reader) : outs(s, level);
reader1 = reader(s, part);
id0 = int(reader1);
id1 = id0 + 1;
id2 = id1 + 1;
id3 = id2 + 1;
c = reader1 - id0;
sound_outs = outputs(s)-2;
cubic(c,v0,v1,v2,v3) = v1 + 0.5 * c * (v2 - v0 + c * (2.0*v0 - 5.0*v1 + 4.0*v2 - v3 + c*(3.0*(v1 -v2) + v3 - v0)));
};
fullsample_reader(gate) = \(s,part).(ramp(gate, speed*srate(s,part)/ma.SR));
play1(s, part) = cubic_player(s, part, fullsample_reader(gate), 1);
attackey_reverb = _<: instrReverb :>_;
instrReverb = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = 1;
roomSize = 1;
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
10379a1260eb4638f19eb9f00fecb72f3a36b189f341ee5812e8fbc2d8a218c6 | publicsamples/Xolotls-Weird-Delay | granul.dsp | process = vgroup("Granulator", environment {
declare name "Granulator";
declare author "Adapted from sfIter by Christophe Lebreton";
/* =========== DESCRIPTION =============
- The granulator takes very small parts of a sound, called GRAINS, and plays them at a varying speed
- Front = Medium size grains
- Back = short grains
- Left Slow rhythm
- Right = Fast rhythm
- Bottom = Regular occurrences
- Head = Irregular occurrences
*/
import("stdfaust.lib");
process = hgroup("Granulator", *(excitation : ampf)), hgroup("Granulator", *(excitation : ampf));
excitation = noiseburst(gate,P) * (gain);
ampf = an.amp_follower_ud(duree_env,duree_env);
//----------------------- NOISEBURST -------------------------
noiseburst(gate,P) = no.noise : *(gate : trigger(P))
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0)/n;
release(n) = + ~ decay(n);
trigger(n) = upfront : release(n) : > (0.0);
};
//-------------------------------------------------------------
P = freq; // fundamental period in samples
freq = hslider("[1]GrainSize[BELA: ANALOG_0]", 200,5,2205,1);
// the frequency gives the white noise band width
Pmax = 4096; // maximum P (for de.delay-line allocation)
// PHASOR_BIN //////////////////////////////
phasor_bin(init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
gate = phasor_bin(1) :-(0.001):pulsar;
gain = 1;
// PULSAR //////////////////////////////
// Pulsar allows to create a more or less random 'pulse'(proba).
pulsar = _<:((_<(ratio_env)):@(100))*(proba>(_,abs(no.noise):ba.latch));
speed = hslider ("[2]Speed[BELA: ANALOG_1]", 10,1,20,0.0001):fi.lowpass(1,1);
ratio_env = 0.5;
fade = (0.5); // min > 0 to avoid division by 0
proba = hslider ("[3]Probability[BELA: ANALOG_2]", 70,50,100,1) * (0.01):fi.lowpass(1,1);
duree_env = 1/(speed: / (ratio_env*(0.25)*fade));
}.process);
| https://raw.githubusercontent.com/publicsamples/Xolotls-Weird-Delay/dc304161575f8abca9ae4a878efde701126b18be/DspNetworks/CodeLibrary/faust/granul.dsp | faust | =========== DESCRIPTION =============
- The granulator takes very small parts of a sound, called GRAINS, and plays them at a varying speed
- Front = Medium size grains
- Back = short grains
- Left Slow rhythm
- Right = Fast rhythm
- Bottom = Regular occurrences
- Head = Irregular occurrences
----------------------- NOISEBURST -------------------------
-------------------------------------------------------------
fundamental period in samples
the frequency gives the white noise band width
maximum P (for de.delay-line allocation)
PHASOR_BIN //////////////////////////////
PULSAR //////////////////////////////
Pulsar allows to create a more or less random 'pulse'(proba).
min > 0 to avoid division by 0 | process = vgroup("Granulator", environment {
declare name "Granulator";
declare author "Adapted from sfIter by Christophe Lebreton";
import("stdfaust.lib");
process = hgroup("Granulator", *(excitation : ampf)), hgroup("Granulator", *(excitation : ampf));
excitation = noiseburst(gate,P) * (gain);
ampf = an.amp_follower_ud(duree_env,duree_env);
noiseburst(gate,P) = no.noise : *(gate : trigger(P))
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0)/n;
release(n) = + ~ decay(n);
trigger(n) = upfront : release(n) : > (0.0);
};
freq = hslider("[1]GrainSize[BELA: ANALOG_0]", 200,5,2205,1);
phasor_bin(init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
gate = phasor_bin(1) :-(0.001):pulsar;
gain = 1;
pulsar = _<:((_<(ratio_env)):@(100))*(proba>(_,abs(no.noise):ba.latch));
speed = hslider ("[2]Speed[BELA: ANALOG_1]", 10,1,20,0.0001):fi.lowpass(1,1);
ratio_env = 0.5;
proba = hslider ("[3]Probability[BELA: ANALOG_2]", 70,50,100,1) * (0.01):fi.lowpass(1,1);
duree_env = 1/(speed: / (ratio_env*(0.25)*fade));
}.process);
|
333cfe84799a6488297243aefdbd80810b7a7629192cfd7401c7ca6c54538e1a | RuolunWeng/ruolunweng.github.io | Loop.dsp | declare name "Loop";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
/* =========== DESCRITPION ===========
- Record and Loop up to 20s of sound
- Activate the Loop Mode : 1 = ON, 0 = OFF
- When Loop Mode is active :
==> Head = Not recording/ Looping the recorded sound
==> Bottom = Recording sound
==> Swift movements around Head = scratched record effect
*/
import("stdfaust.lib");
B = hslider("Start/Stop Recording (Max 20s)[acc:1 0 -10 0 12]", 1,0,1,1); // Capture sound while pressed
I = int(B); // convert button signal from float to integer
R = (I-I') <= 0; // Reset capture when button is pressed
D = (+(I):*(R))~_; // Compute capture duration while button is pressed: 0..NNNN0..MMM
capture = *(B) : (+ : de.delay(1048576, D-1)) ~ *(1.0-B) ;
level = hslider("Volume [unit:dB]", 0, -96, 4, 0.1) : ba.db2linear : si.smooth(0.999);
captONOFF = hslider("Loop Mode ON/OFF",0,0,1,1);
process = vgroup( "LOOP", _<:_,(_<:capture,_ : select2(B)): select2(captONOFF) *(level) ) ;
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/effects/Loop.dsp | faust | =========== DESCRITPION ===========
- Record and Loop up to 20s of sound
- Activate the Loop Mode : 1 = ON, 0 = OFF
- When Loop Mode is active :
==> Head = Not recording/ Looping the recorded sound
==> Bottom = Recording sound
==> Swift movements around Head = scratched record effect
Capture sound while pressed
convert button signal from float to integer
Reset capture when button is pressed
Compute capture duration while button is pressed: 0..NNNN0..MMM | declare name "Loop";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
import("stdfaust.lib");
capture = *(B) : (+ : de.delay(1048576, D-1)) ~ *(1.0-B) ;
level = hslider("Volume [unit:dB]", 0, -96, 4, 0.1) : ba.db2linear : si.smooth(0.999);
captONOFF = hslider("Loop Mode ON/OFF",0,0,1,1);
process = vgroup( "LOOP", _<:_,(_<:capture,_ : select2(B)): select2(captONOFF) *(level) ) ;
|
36ef2b4afe5eca544d94b8d06e14896b6949fa4f00b9a54d552f1d6b9580d5f9 | afalaize/faust | tester2.dsp | declare name "tester2";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2014";
//-----------------------------------------------
// Stereo Audio Tester : send a test signal (sine,
// noise, pink) on a stereo channel
//-----------------------------------------------
import("stdfaust.lib");
pink = f : (+ ~ g) with {
f(x) = 0.04957526213389*x - 0.06305581334498*x' + 0.01483220320740*x'';
g(x) = 1.80116083982126*x - 0.80257737639225*x';
};
// User interface
//----------------
transition(n) = \(old,new).(ba.if(old<new, min(old+1.0/n,new), max(old-1.0/n,new))) ~ _;
vol = hslider("[2] volume [unit:dB]", -96, -96, 0, 1): ba.db2linear : si.smoo;
freq = hslider("[1] freq [unit:Hz][scale:log]", 440, 40, 20000, 1);
wave = hslider("[3] signal [style:menu{'white noise':0;'pink noise':1;'sine':2}]", 0, 0, 2, 1) : int;
dest = hslider("[4] channel [style:radio{'none':0;'left':1;'right':2;'both':3}]", 0, 0, 3, 1) : int;
testsignal = no.noise, pink(no.noise), os.osci(freq): select3(wave);
process = vgroup("Stereo Audio Tester",
testsignal*vol
<: par(i, 2, *((dest & (i+1)) != 0 : transition(4410)))
);
| https://raw.githubusercontent.com/afalaize/faust/8f9f5fe3aa167eaeecc15a99d4da984ac2797be3/examples/misc/tester2.dsp | faust | -----------------------------------------------
Stereo Audio Tester : send a test signal (sine,
noise, pink) on a stereo channel
-----------------------------------------------
User interface
---------------- | declare name "tester2";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2014";
import("stdfaust.lib");
pink = f : (+ ~ g) with {
f(x) = 0.04957526213389*x - 0.06305581334498*x' + 0.01483220320740*x'';
g(x) = 1.80116083982126*x - 0.80257737639225*x';
};
transition(n) = \(old,new).(ba.if(old<new, min(old+1.0/n,new), max(old-1.0/n,new))) ~ _;
vol = hslider("[2] volume [unit:dB]", -96, -96, 0, 1): ba.db2linear : si.smoo;
freq = hslider("[1] freq [unit:Hz][scale:log]", 440, 40, 20000, 1);
wave = hslider("[3] signal [style:menu{'white noise':0;'pink noise':1;'sine':2}]", 0, 0, 2, 1) : int;
dest = hslider("[4] channel [style:radio{'none':0;'left':1;'right':2;'both':3}]", 0, 0, 3, 1) : int;
testsignal = no.noise, pink(no.noise), os.osci(freq): select3(wave);
process = vgroup("Stereo Audio Tester",
testsignal*vol
<: par(i, 2, *((dest & (i+1)) != 0 : transition(4410)))
);
|
bdfdfb0bef7e7c9338628302818b0532acb91b48886423172cb6b18b8d9272d6 | afalaize/faust | lfBoost.dsp | // WARNING: This a "legacy example based on a deprecated library". Check filters.lib
// for more accurate examples of filter functions
declare name "lfboost";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
//------------------------------------------------------------------
// DAFX, Digital Audio Effects (Wiley ed.)
// chapter 2 : filters
// section 2.3 : Equalizers
// page 53 : second order shelving filter design
//------------------------------------------------------------------
import("stdfaust.lib");
//----------------------low frequency boost filter -------------------------------
// lfboost(F,G)
// F : frequency (in Hz)
// G : gain (in dB)
//
//--------------------------------------------------------------------------------
lfboost(F,G) = fi.TF2((1 + sqrt(2*V)*K + V*K*K) / denom,
2 * (V*K*K - 1) / denom,
(1 - sqrt(2*V)*K + V*K*K) / denom,
2 * (K*K - 1) / denom,
(1 - sqrt(2)*K + K*K) / denom)
with {
V = ba.db2linear(G);
K = tan(ma.PI*F/ma.SR);
denom = 1 + sqrt(2)*K + K*K;
};
//====================low frequency boost process ===============================
process = vgroup("lowboost", lfboost(nentry("freq [unit:Hz][style:knob]", 100, 20, 150, 1),
vslider("gain [unit:dB]", 0, -20, 20, 0.1)));
| https://raw.githubusercontent.com/afalaize/faust/8f9f5fe3aa167eaeecc15a99d4da984ac2797be3/examples/filtering/lfBoost.dsp | faust | WARNING: This a "legacy example based on a deprecated library". Check filters.lib
for more accurate examples of filter functions
------------------------------------------------------------------
DAFX, Digital Audio Effects (Wiley ed.)
chapter 2 : filters
section 2.3 : Equalizers
page 53 : second order shelving filter design
------------------------------------------------------------------
----------------------low frequency boost filter -------------------------------
lfboost(F,G)
F : frequency (in Hz)
G : gain (in dB)
--------------------------------------------------------------------------------
====================low frequency boost process =============================== |
declare name "lfboost";
declare version "1.0";
declare author "Grame";
declare license "BSD";
declare copyright "(c)GRAME 2006";
import("stdfaust.lib");
lfboost(F,G) = fi.TF2((1 + sqrt(2*V)*K + V*K*K) / denom,
2 * (V*K*K - 1) / denom,
(1 - sqrt(2*V)*K + V*K*K) / denom,
2 * (K*K - 1) / denom,
(1 - sqrt(2)*K + K*K) / denom)
with {
V = ba.db2linear(G);
K = tan(ma.PI*F/ma.SR);
denom = 1 + sqrt(2)*K + K*K;
};
process = vgroup("lowboost", lfboost(nentry("freq [unit:Hz][style:knob]", 100, 20, 150, 1),
vslider("gain [unit:dB]", 0, -20, 20, 0.1)));
|
ed172e7478a9401e3b77f5e9cea6b0977cb901158d321ee4884ffb316f8039db | olegkapitonov/Kapitonov-Plugins-Pack | kpp_fuzz.dsp | /*
* Copyright (C) 2018-2020 Oleg Kapitonov
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* --------------------------------------------------------------------------
*/
/*
* This plugin is a vintage fuzz pedal emulator.
*
* Process chain:
*
* input->pre_filter->*drive_knob->fuzz->tone->*volume_knob->output
*
* pre-filter - lowpass, 1 order, 1720 Hz. Emulates effect
* of low impedance of vintage pedal.
*
* distortion - 2 cascades, asymmetric quadratic distortion and clipper.
* Emulates distortion of old bad class A transistor cascades.
* tone - highshelf 720 Hz.
*/
declare name "kpp_fuzz";
declare author "Oleg Kapitonov";
declare license "GPLv3";
declare version "1.1";
import("stdfaust.lib");
process = output with {
fuzz = vslider("fuzz",0,0,100,0.01);
tone = vslider("tone",0,-15,0,0.1);
volume = vslider("volume",0.5,0,1,0.001);
clamp = min(2.0) : max(-2.0);
pre_filter = fi.dcblocker : fi.lowpass(1, 1720.0);
distortion = *(ba.db2linear(fuzz/10)) : +(0.5) : max(0.0) : min(1.0)
: fi.highpass(1, 120) : +(0.5)
<: _,_ : * : -(0.4) : *(ba.db2linear(fuzz/5)) : max(-1.0) : min(1.0) : +(0.15);
filter = fi.high_shelf(tone + 7.5, 720.0);
stomp = pre_filter : *(10.0) : distortion : filter :
*(ba.db2linear(volume * 50.0 ) / 100.0) :
/(20.0);
output = _ : stomp : _;
};
| https://raw.githubusercontent.com/olegkapitonov/Kapitonov-Plugins-Pack/ed4541172d53ecf04bad43cd583365f278ccf176/LADSPA/kpp_fuzz/kpp_fuzz.dsp | faust |
* Copyright (C) 2018-2020 Oleg Kapitonov
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* --------------------------------------------------------------------------
* This plugin is a vintage fuzz pedal emulator.
*
* Process chain:
*
* input->pre_filter->*drive_knob->fuzz->tone->*volume_knob->output
*
* pre-filter - lowpass, 1 order, 1720 Hz. Emulates effect
* of low impedance of vintage pedal.
*
* distortion - 2 cascades, asymmetric quadratic distortion and clipper.
* Emulates distortion of old bad class A transistor cascades.
* tone - highshelf 720 Hz.
|
declare name "kpp_fuzz";
declare author "Oleg Kapitonov";
declare license "GPLv3";
declare version "1.1";
import("stdfaust.lib");
process = output with {
fuzz = vslider("fuzz",0,0,100,0.01);
tone = vslider("tone",0,-15,0,0.1);
volume = vslider("volume",0.5,0,1,0.001);
clamp = min(2.0) : max(-2.0);
pre_filter = fi.dcblocker : fi.lowpass(1, 1720.0);
distortion = *(ba.db2linear(fuzz/10)) : +(0.5) : max(0.0) : min(1.0)
: fi.highpass(1, 120) : +(0.5)
<: _,_ : * : -(0.4) : *(ba.db2linear(fuzz/5)) : max(-1.0) : min(1.0) : +(0.15);
filter = fi.high_shelf(tone + 7.5, 720.0);
stomp = pre_filter : *(10.0) : distortion : filter :
*(ba.db2linear(volume * 50.0 ) / 100.0) :
/(20.0);
output = _ : stomp : _;
};
|
b30888878d498c59dfdd37a4d351363645c60f27919498b016b025078198a15d | madskjeldgaard/mkfaustplugins | TwoMassChain.dsp | declare name "TwoMassChain";
declare author "James Leonard";
declare date "April 2020";
/* ========= DESCRITPION =============
A logical step from a simple oscillator: a chain of two masses connected by spring-dampers,
fixed at one end to a fixed point !
- inputs: force impulse on the last mass of the chain.
- outputs: position of the last mass of the chain.
- controls: none
*/
import("stdfaust.lib");
in1 = _;
OutGain = 0.1;
p1 = hslider("k1",0.1,0.00001,1.0,0.001);
p2 = hslider("k2",0.1,0.00001,1.0,0.001);
p3 = hslider("z1",0.003,0.00001,1.0,0.001);
p4 = hslider("z2",0.003,0.00001,1.0,0.001);
K1 = p1;
Z1 = p3;
K2 = p2;
Z2 = p4;
model = (
mi.ground(0.),
mi.mass(1., 0, 0., 0.),
mi.mass(1., 0, 0., 0.),
par(i, nbFrcIn,_):
RoutingMassToLink ,
par(i, nbFrcIn,_):
mi.springDamper(K1, Z1, 0., 0.),
mi.springDamper(K2, Z2, 0., 0.),
par(i, nbOut+nbFrcIn, _):
RoutingLinkToMass
)~par(i, nbMass, _):
par(i, nbMass, !), par(i, nbOut , _)
with{
RoutingMassToLink(m0, m1, m2) = /* routed positions */ m0, m1, m1, m2, /* outputs */ m2;
RoutingLinkToMass(l0_f1, l0_f2, l1_f1, l1_f2, p_out1, f_in1) = /* routed forces */ l0_f1, l0_f2 + l1_f1, f_in1 + l1_f2, /* pass-through */ p_out1;
nbMass = 3;
nbFrcIn = 1;
nbOut = 1;
};
process = in1 : model:*(OutGain);
/*
========= MIMS SCRIPT USED FOR MODEL GENERATION =============
# MIMS script file
# Script author: James Leonard
@K1 param 0.1
@Z1 param 0.0003
@K2 param 0.1
@Z2 param 0.0003
@g ground 0.
@m1 mass 1. 0. 0.
@m2 mass 1. 0. 0.
@s1 springDamper @g @m1 K1 Z1
@s2 springDamper @m1 @m2 K2 Z2
# Add force input to the model
@in1 frcInput @m2
# Add position output from the oscillator
@out1 posOutput @m2
# end of MIMS script
*/
| https://raw.githubusercontent.com/madskjeldgaard/mkfaustplugins/fd7cf250788174b5efa6ae3294997609830875d1/plugins/TwoMassChain/TwoMassChain.dsp | faust | ========= DESCRITPION =============
A logical step from a simple oscillator: a chain of two masses connected by spring-dampers,
fixed at one end to a fixed point !
- inputs: force impulse on the last mass of the chain.
- outputs: position of the last mass of the chain.
- controls: none
routed positions
outputs
routed forces
pass-through
========= MIMS SCRIPT USED FOR MODEL GENERATION =============
# MIMS script file
# Script author: James Leonard
@K1 param 0.1
@Z1 param 0.0003
@K2 param 0.1
@Z2 param 0.0003
@g ground 0.
@m1 mass 1. 0. 0.
@m2 mass 1. 0. 0.
@s1 springDamper @g @m1 K1 Z1
@s2 springDamper @m1 @m2 K2 Z2
# Add force input to the model
@in1 frcInput @m2
# Add position output from the oscillator
@out1 posOutput @m2
# end of MIMS script
| declare name "TwoMassChain";
declare author "James Leonard";
declare date "April 2020";
import("stdfaust.lib");
in1 = _;
OutGain = 0.1;
p1 = hslider("k1",0.1,0.00001,1.0,0.001);
p2 = hslider("k2",0.1,0.00001,1.0,0.001);
p3 = hslider("z1",0.003,0.00001,1.0,0.001);
p4 = hslider("z2",0.003,0.00001,1.0,0.001);
K1 = p1;
Z1 = p3;
K2 = p2;
Z2 = p4;
model = (
mi.ground(0.),
mi.mass(1., 0, 0., 0.),
mi.mass(1., 0, 0., 0.),
par(i, nbFrcIn,_):
RoutingMassToLink ,
par(i, nbFrcIn,_):
mi.springDamper(K1, Z1, 0., 0.),
mi.springDamper(K2, Z2, 0., 0.),
par(i, nbOut+nbFrcIn, _):
RoutingLinkToMass
)~par(i, nbMass, _):
par(i, nbMass, !), par(i, nbOut , _)
with{
nbMass = 3;
nbFrcIn = 1;
nbOut = 1;
};
process = in1 : model:*(OutGain);
|
a13e58df787b5ea6746f8fac84437f12d17662b27d19120765e897338abffb46 | jameslnrd/mi_introduction_workshop_2020 | nlOsc.dsp | declare name "Non Linear Oscillator";
declare author "James Leonard";
declare date "April 2020";
/* ========= DESCRITPION =============
A non-linear oscillator (containing a cubic stiffness term, making pitch amplitude dependant)
- inputs: force impulse
- outputs: oscillator position.
- controls: value of the non-linear stiffness component.
Note: we are using the nlSpringDamperClipped interaction that defines an upper bound for stiffness.
This can save from numerical divergence when large displacements cause very large reaction forces.
*/
import("stdfaust.lib");
in1 = button("Hammer Input Force"): ba.impulsify* -0.1;
OutGain = 1;
nlK = hslider("non-linear stiffness", 0.005, 0., 0.1, 0.0001);
model = (
mi.mass(1., 0, 0., 0.),
mi.ground(0.),
mi.mass(0.3, 0, 1., 1.),
mi.ground(1.),
par(i, nbFrcIn,_):
RoutingMassToLink ,
par(i, nbFrcIn,_):
mi.nlSpringDamperClipped(0.03, nlK, 0.8, 0.0002, 0., 0.),
mi.springDamper(0.0001, 0.05, 1., 1.),
mi.collision(0.1, 0.001, 0, 0., 1.),
par(i, nbOut+nbFrcIn, _):
RoutingLinkToMass
)~par(i, nbMass, _):
par(i, nbMass, !), par(i, nbOut , _)
with{
RoutingMassToLink(m0, m1, m2, m3) = /* routed positions */ m1, m0, m3, m2, m0, m2, /* outputs */ m0;
RoutingLinkToMass(l0_f1, l0_f2, l1_f1, l1_f2, l2_f1, l2_f2, p_out1, f_in1) = /* routed forces */ l0_f2 + l2_f1, l0_f1, f_in1 + l1_f2 + l2_f2, l1_f1, /* pass-through */ p_out1;
nbMass = 4;
nbFrcIn = 1;
nbOut = 1;
};
process = in1 : model:*(OutGain);
/*
========= MIMS SCRIPT USED FOR MODEL GENERATION =============
# MIMS script file
# Script author: James Leonard
@nlK param 0.005
# Build a non-linear harmonic oscillator
@m mass 1. 0. 0.
@g ground 0.
@nl nlSpringDamper @g @m 0.03 nlK 0.8 0.0002
# A slow moving oscillator placed above the other
# serving as a hammer
@ham mass 0.3 1. 0.
@g2 ground 1.
@sp springDamper @g2 @ham 0.0001 0.05
# Add force input to the hammer
@in1 frcInput @ham
@c contact @m @ham 0.1 0.001
# Add position output from the oscillator
@out1 posOutput @m
# @out2 posOutput @ham
# end of MIMS script
*/ | https://raw.githubusercontent.com/jameslnrd/mi_introduction_workshop_2020/2f487dbc5b8e7cd83cbd962254e737bdb82948f6/07_NonLinearOscillator/nlOsc.dsp | faust | ========= DESCRITPION =============
A non-linear oscillator (containing a cubic stiffness term, making pitch amplitude dependant)
- inputs: force impulse
- outputs: oscillator position.
- controls: value of the non-linear stiffness component.
Note: we are using the nlSpringDamperClipped interaction that defines an upper bound for stiffness.
This can save from numerical divergence when large displacements cause very large reaction forces.
routed positions
outputs
routed forces
pass-through
========= MIMS SCRIPT USED FOR MODEL GENERATION =============
# MIMS script file
# Script author: James Leonard
@nlK param 0.005
# Build a non-linear harmonic oscillator
@m mass 1. 0. 0.
@g ground 0.
@nl nlSpringDamper @g @m 0.03 nlK 0.8 0.0002
# A slow moving oscillator placed above the other
# serving as a hammer
@ham mass 0.3 1. 0.
@g2 ground 1.
@sp springDamper @g2 @ham 0.0001 0.05
# Add force input to the hammer
@in1 frcInput @ham
@c contact @m @ham 0.1 0.001
# Add position output from the oscillator
@out1 posOutput @m
# @out2 posOutput @ham
# end of MIMS script
| declare name "Non Linear Oscillator";
declare author "James Leonard";
declare date "April 2020";
import("stdfaust.lib");
in1 = button("Hammer Input Force"): ba.impulsify* -0.1;
OutGain = 1;
nlK = hslider("non-linear stiffness", 0.005, 0., 0.1, 0.0001);
model = (
mi.mass(1., 0, 0., 0.),
mi.ground(0.),
mi.mass(0.3, 0, 1., 1.),
mi.ground(1.),
par(i, nbFrcIn,_):
RoutingMassToLink ,
par(i, nbFrcIn,_):
mi.nlSpringDamperClipped(0.03, nlK, 0.8, 0.0002, 0., 0.),
mi.springDamper(0.0001, 0.05, 1., 1.),
mi.collision(0.1, 0.001, 0, 0., 1.),
par(i, nbOut+nbFrcIn, _):
RoutingLinkToMass
)~par(i, nbMass, _):
par(i, nbMass, !), par(i, nbOut , _)
with{
nbMass = 4;
nbFrcIn = 1;
nbOut = 1;
};
process = in1 : model:*(OutGain);
|
228ad962cc828a65929e1ec38c4accd8d47b83375f86198d63896a2b13020a0d | magnetophon/DigiDrie | korg35lpf_test.dsp | declare korg35LPF author "Eric Tarr";
declare korg35LPF license "MIT-style STK-4.3 license";
import("stdfaust.lib");
korg35LPF(freq,Q) = _ <: (s1,s2,s3,y) : !,!,!,_
letrec{
's1 = _-s1:_*(alpha*2):_+s1;
's2 = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2):_*alpha0:_-s3:_*alpha:_+s3:_*K:_-s2:_*(alpha*2):_+s2;
's3 = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2):_*alpha0:_-s3:_*(alpha*2):_+s3;
'y = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2) :_*alpha0:_-s3:_*alpha:_+s3;
}
with{
// freq = 2*(10^(3*normFreq+1));
K = 2.0*(Q - 0.707)/(10.0 - 0.707); // target.
wd = 2*ma.PI*freq;
T = 1/ma.SR;
wa = (2/T)*tan(wd*T/2);
g = wa*T/2;
G = g/(1.0 + g);
alpha = G; // target.
B3 = (K - K*G)/(1 + g); // target.
B2 = -1/(1 + g); // target.
alpha0 = 1/(1 - K*G + K*G*G); // target.
};
korg35LPF_approx(freq,Q) = _ <: (s1,s2,s3,y) : !,!,!,_
letrec{
's1 = _-s1:_*(alpha*2):_+s1;
's2 = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2):_*alpha0:_-s3:_*alpha:_+s3:_*K:_-s2:_*(alpha*2):_+s2;
's3 = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2):_*alpha0:_-s3:_*(alpha*2):_+s3;
'y = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2) :_*alpha0:_-s3:_*alpha:_+s3;
}
with{
// Only valid in [0, 0.498). 0.5 is nyquist frequency.
tan_halfpi_approx(x) = (
4.189308700355015e-05 +
4.290568649086532 * x +
-2.657498976290899 * x * x +
-1.5163927048819992 * x * x * x
) / (
1.3667229106607917 +
-0.8644224895636948 * x +
-4.828883069406347 * x * x +
2.181672945531366 * x * x * x
);
// freq = 2*(10^(3*normFreq+1));
K = 2.0*(Q - 0.707)/(10.0 - 0.707);
g = tan_halfpi_approx(freq / ma.SR);
G = g/(1.0 + g);
alpha = G;
B3 = (K - K*G)/(1 + g);
B2 = -1/(1 + g);
alpha0 = 1/(1 - K*G + K*G*G);
};
freq = hslider("normFreq", 124, -12, 124, 1e-5) : ba.pianokey2hz : min(20000);
Q = hslider("Q", 0, 0, 10, 1e-5);
process = _ <: korg35LPF(freq, Q), korg35LPF_approx(freq, Q);
| https://raw.githubusercontent.com/magnetophon/DigiDrie/a9f79d502e1f8d522e5f47e0c460ae99e80f9441/faust/benchmark/korg35lfp/korg35lpf_test.dsp | faust | freq = 2*(10^(3*normFreq+1));
target.
target.
target.
target.
target.
Only valid in [0, 0.498). 0.5 is nyquist frequency.
freq = 2*(10^(3*normFreq+1)); | declare korg35LPF author "Eric Tarr";
declare korg35LPF license "MIT-style STK-4.3 license";
import("stdfaust.lib");
korg35LPF(freq,Q) = _ <: (s1,s2,s3,y) : !,!,!,_
letrec{
's1 = _-s1:_*(alpha*2):_+s1;
's2 = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2):_*alpha0:_-s3:_*alpha:_+s3:_*K:_-s2:_*(alpha*2):_+s2;
's3 = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2):_*alpha0:_-s3:_*(alpha*2):_+s3;
'y = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2) :_*alpha0:_-s3:_*alpha:_+s3;
}
with{
wd = 2*ma.PI*freq;
T = 1/ma.SR;
wa = (2/T)*tan(wd*T/2);
g = wa*T/2;
G = g/(1.0 + g);
};
korg35LPF_approx(freq,Q) = _ <: (s1,s2,s3,y) : !,!,!,_
letrec{
's1 = _-s1:_*(alpha*2):_+s1;
's2 = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2):_*alpha0:_-s3:_*alpha:_+s3:_*K:_-s2:_*(alpha*2):_+s2;
's3 = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2):_*alpha0:_-s3:_*(alpha*2):_+s3;
'y = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2) :_*alpha0:_-s3:_*alpha:_+s3;
}
with{
tan_halfpi_approx(x) = (
4.189308700355015e-05 +
4.290568649086532 * x +
-2.657498976290899 * x * x +
-1.5163927048819992 * x * x * x
) / (
1.3667229106607917 +
-0.8644224895636948 * x +
-4.828883069406347 * x * x +
2.181672945531366 * x * x * x
);
K = 2.0*(Q - 0.707)/(10.0 - 0.707);
g = tan_halfpi_approx(freq / ma.SR);
G = g/(1.0 + g);
alpha = G;
B3 = (K - K*G)/(1 + g);
B2 = -1/(1 + g);
alpha0 = 1/(1 - K*G + K*G*G);
};
freq = hslider("normFreq", 124, -12, 124, 1e-5) : ba.pianokey2hz : min(20000);
Q = hslider("Q", 0, 0, 10, 1e-5);
process = _ <: korg35LPF(freq, Q), korg35LPF_approx(freq, Q);
|
b788d980a683521f0b3dc1287707e0d16055b7c6418bf3a12827e50504bb84e9 | johannphilippe/paw2022 | seq_poly_detect.dsp |
declare name "polyphonic_detection";
declare version "1.0";
declare author "Johann Philippe";
declare license "MIT";
declare copyright "(c) Johann Philippe 2022";
import("stdfaust.lib");
mpulse(smps_dur, trig) = pulsation
with {
count = ba.countdown(smps_dur, trig > 0 );
//count = -(1)~_, smps_dur : select2(trig);
pulsation = 0, 1 : select2(count > 0);
};
mpulse_dur(duration, trig) = mpulse(ba.sec2samp(duration), trig);
line(time, sig) = res
letrec {
'changed = (sig' != sig) | (time' != time);
'steps = ma.SR * time;
'cntup = ba.countup(steps ,changed);
'diff = ( sig - res);
'inc = diff / steps : ba.sAndH(changed);
'res = res, res + inc : select2(cntup < steps);
};
/*
poly_detector : A polyphonic pitch detector based on parallel bandpass filters
Input Arguments :
* thresh : threshold detection - when the RMS level of a band crosses the threshold, it will output the RMS of the band, else 0
* rms_avg : RMS average (duration in seconds)
* sig : input signal
Output :
* N_BANDS parallel signals. Value of each signal is the RMS of the band if this RMS level is above threshold, else 0. Bandpass filters frequencies are MIDI notes from 20 to (20 + N_BANDS)
*/
// Increase N_FILTER for more accuracy, reduce it to increase processing speed
N_FILTER = 3;
// From midi note 20 to 20 + 105 (125)
N_BANDS = 105;
poly_detector(thresh, rms_avg,atq, rel, sig) = 0 : seq(n, N_BANDS, chain(20 + n))
with {
// Precision of filters is 1/4 tone up and down of the center frequency
//filter(note) = fi.resonbp(ba.midikey2hz(note), 1000, 0.005) ; //fi.bandpass(1, ba.midikey2hz(note - 0.5), ba.midikey2hz(note + 0.5));
filter(note) = fi.bandpass(1, ba.midikey2hz(note - 0.5), ba.midikey2hz(note + 0.5));
// Sequential butterworth bandpass filter
band( note) = seq(n, N_FILTER, filter(note));
oscilo(note, amp) = (os.osc(ba.midikey2hz(note) ) + (os.osc(ba.midikey2hz(note) * 0.5) *0.5 ) ) : *(env) : *(itp_amp)
with {
trig = amp : mpulse_dur(atq);
env = trig : en.are(atq, rel);
itp_amp = amp : ba.sAndH(amp > 0) : line(0.1);
};
// Filters the input signal, and calls RMS detection
chain(note) = +(syn)
with {
syn = sig : band(note) : fi.dcblocker : detect : oscilo(note);
detect(band) = 0, brms : select2(brms > thresh)
with {
brms = band : an.rms_envelope_rect(rms_avg);
};
};
};
reverberate(mix, sig) = sig : re.mono_freeverb(0.8, 0.8, 0.4, 1) : _ * mix + sig * (1 - mix);
rms_avg = hslider("rms", 0.01, 0.0001, 1, 0.00001);
atq = hslider("attack", 0.1, 0.05, 1, 0.01);
rel = hslider("release", 0.1, 0.01, 3, 0.01);
threshold = hslider("thresh", 0.001, 0.0001, 1, 0.0001);
mix = hslider("drymix", 0, 0, 1, 0.01) : si.smoo;
process = _ <: poly_detector(threshold, rms_avg, atq, rel), _ : reverberate(0.4), _ * mix;
| https://raw.githubusercontent.com/johannphilippe/paw2022/d9b921a44e72bab11e457a13a1b43a4eabca53df/examples/seq_poly_detect.dsp | faust | count = -(1)~_, smps_dur : select2(trig);
poly_detector : A polyphonic pitch detector based on parallel bandpass filters
Input Arguments :
* thresh : threshold detection - when the RMS level of a band crosses the threshold, it will output the RMS of the band, else 0
* rms_avg : RMS average (duration in seconds)
* sig : input signal
Output :
* N_BANDS parallel signals. Value of each signal is the RMS of the band if this RMS level is above threshold, else 0. Bandpass filters frequencies are MIDI notes from 20 to (20 + N_BANDS)
Increase N_FILTER for more accuracy, reduce it to increase processing speed
From midi note 20 to 20 + 105 (125)
Precision of filters is 1/4 tone up and down of the center frequency
filter(note) = fi.resonbp(ba.midikey2hz(note), 1000, 0.005) ; //fi.bandpass(1, ba.midikey2hz(note - 0.5), ba.midikey2hz(note + 0.5));
Sequential butterworth bandpass filter
Filters the input signal, and calls RMS detection |
declare name "polyphonic_detection";
declare version "1.0";
declare author "Johann Philippe";
declare license "MIT";
declare copyright "(c) Johann Philippe 2022";
import("stdfaust.lib");
mpulse(smps_dur, trig) = pulsation
with {
count = ba.countdown(smps_dur, trig > 0 );
pulsation = 0, 1 : select2(count > 0);
};
mpulse_dur(duration, trig) = mpulse(ba.sec2samp(duration), trig);
line(time, sig) = res
letrec {
'changed = (sig' != sig) | (time' != time);
'steps = ma.SR * time;
'cntup = ba.countup(steps ,changed);
'diff = ( sig - res);
'inc = diff / steps : ba.sAndH(changed);
'res = res, res + inc : select2(cntup < steps);
};
N_FILTER = 3;
N_BANDS = 105;
poly_detector(thresh, rms_avg,atq, rel, sig) = 0 : seq(n, N_BANDS, chain(20 + n))
with {
filter(note) = fi.bandpass(1, ba.midikey2hz(note - 0.5), ba.midikey2hz(note + 0.5));
band( note) = seq(n, N_FILTER, filter(note));
oscilo(note, amp) = (os.osc(ba.midikey2hz(note) ) + (os.osc(ba.midikey2hz(note) * 0.5) *0.5 ) ) : *(env) : *(itp_amp)
with {
trig = amp : mpulse_dur(atq);
env = trig : en.are(atq, rel);
itp_amp = amp : ba.sAndH(amp > 0) : line(0.1);
};
chain(note) = +(syn)
with {
syn = sig : band(note) : fi.dcblocker : detect : oscilo(note);
detect(band) = 0, brms : select2(brms > thresh)
with {
brms = band : an.rms_envelope_rect(rms_avg);
};
};
};
reverberate(mix, sig) = sig : re.mono_freeverb(0.8, 0.8, 0.4, 1) : _ * mix + sig * (1 - mix);
rms_avg = hslider("rms", 0.01, 0.0001, 1, 0.00001);
atq = hslider("attack", 0.1, 0.05, 1, 0.01);
rel = hslider("release", 0.1, 0.01, 3, 0.01);
threshold = hslider("thresh", 0.001, 0.0001, 1, 0.0001);
mix = hslider("drymix", 0, 0, 1, 0.01) : si.smoo;
process = _ <: poly_detector(threshold, rms_avg, atq, rel), _ : reverberate(0.4), _ * mix;
|
0d7f50cf193eb0021c13d0137d6df1ca066277b1fb11758e4c0893a5329406b4 | afalaize/faust | FFT.dsp | // Radix 2 FFT, decimation in time, real and imag parts interleaved
declare name "FFT";
declare author "JOS";
declare license "STK-4.3";
import("stdfaust.lib");
N=32; // FFT size (power of 2)
// Number of frequency bins (including dc and SR/2) is N/2+1
No2 = N>>1;
signal = amp * cosine with {
cosine = select2(k==0,
select2(k==No2,
2.0*os.oscrc(f(k)), // 2x since negative-frequencies not displayed
1-1':+~*(-1) // Alternating sequence: 1, -1, 1, -1
),
1.0); // make sure phase is zero (freq jumps around)
f(k) = float(k) * ma.SR / float(N); // only test FFT bin frequencies
k = hslider("[2] FFT Bin Number",N/4,0,No2,0.001) : int <: _,dpy : attach;
dpy = hbargraph("[3] Measured FFT Bin Number",0,No2);
amp = hslider("[4] Amplitude",0.1,0,1,0.001);
};
process = signal : dm.fft_spectral_level_demo(N) <: _,_;
| https://raw.githubusercontent.com/afalaize/faust/8f9f5fe3aa167eaeecc15a99d4da984ac2797be3/examples/analysis/FFT.dsp | faust | Radix 2 FFT, decimation in time, real and imag parts interleaved
FFT size (power of 2)
Number of frequency bins (including dc and SR/2) is N/2+1
2x since negative-frequencies not displayed
Alternating sequence: 1, -1, 1, -1
make sure phase is zero (freq jumps around)
only test FFT bin frequencies |
declare name "FFT";
declare author "JOS";
declare license "STK-4.3";
import("stdfaust.lib");
No2 = N>>1;
signal = amp * cosine with {
cosine = select2(k==0,
select2(k==No2,
),
k = hslider("[2] FFT Bin Number",N/4,0,No2,0.001) : int <: _,dpy : attach;
dpy = hbargraph("[3] Measured FFT Bin Number",0,No2);
amp = hslider("[4] Amplitude",0.1,0,1,0.001);
};
process = signal : dm.fft_spectral_level_demo(N) <: _,_;
|
0992bafc69a7a5ddadd4c77862f632dd79a0d2e88703a2fe5a0717ed1d29e13d | rottingsounds/bitDSP-faust | bitstreamAdder.dsp | // This is a simple example that tests an adder for 1-bit streams.
// In this example, in particular, we will add together a stream
// of 1s and a stream of 0s. Summing two opposite values results
// in a zero-ed output, which consists of alternating 0s and 1s
// in the delta-sigma-modulated domain.
declare name "Bitstream adder";
declare author "Dario Sanfilippo";
declare reference "O'leary, P., & Maloberti, F. (1990). Bit stream adder
for oversampling coded data. Electronics Letters, 26(20), 1708-1709.";
import("stdfaust.lib");
bit = library("bitDSP.lib");
// plot
// CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib
// bitstream_adder-example.dsp
// ./bitstream_adder-example -n 50
// compile
// CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib
// bitstream_adder-example.dsp
// ./bitstream_adder-example
process = bit.bitstream_adder(0, 1);
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/21cf36105c55b6e18969a867a319530a0ef1ea63/examples/bitstreamAdder.dsp | faust | This is a simple example that tests an adder for 1-bit streams.
In this example, in particular, we will add together a stream
of 1s and a stream of 0s. Summing two opposite values results
in a zero-ed output, which consists of alternating 0s and 1s
in the delta-sigma-modulated domain.
plot
CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib
bitstream_adder-example.dsp
./bitstream_adder-example -n 50
compile
CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib
bitstream_adder-example.dsp
./bitstream_adder-example |
declare name "Bitstream adder";
declare author "Dario Sanfilippo";
declare reference "O'leary, P., & Maloberti, F. (1990). Bit stream adder
for oversampling coded data. Electronics Letters, 26(20), 1708-1709.";
import("stdfaust.lib");
bit = library("bitDSP.lib");
process = bit.bitstream_adder(0, 1);
|
6f13e9d4f4d8083cf7821cd2af21e87137c2b0c90ab3b50ee58d1408732b61be | olegkapitonov/KPP-VST3 | kpp_fuzz.dsp | /*
* Copyright (C) 2018-2020 Oleg Kapitonov
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* --------------------------------------------------------------------------
*/
/*
* This plugin is a vintage fuzz pedal emulator.
*
* Process chain:
*
* input->pre_filter->*drive_knob->fuzz->tone->*volume_knob->output
*
* pre-filter - lowpass, 1 order, 1720 Hz. Emulates effect
* of low impedance of vintage pedal.
*
* distortion - 2 cascades, asymmetric distortion and clipper.
* Emulates distortion of old bad class A transistor cascades.
* tone - highshelf 720 Hz.
*/
declare name "kpp_fuzz";
declare author "Oleg Kapitonov";
declare license "GPLv3";
declare version "1.2";
import("stdfaust.lib");
process = output with {
// Bypass button, 0 - pedal on, 1 -pedal off (bypass on)
bypass = checkbox("99_bypass");
fuzz = vslider("fuzz",50,0,100,0.01);
tone = vslider("tone",-7.5,-15,0,0.1);
volume = vslider("volume",0.5,0,1,0.001);
clamp = min(2.0) : max(-2.0);
pre_filter = fi.dcblocker : fi.lowpass(1, 2000.0);
biaser(Uin) = Uout letrec {
'Ulimited = Uin : max(-50.0 + Ubias) : -(Ubias);
'Ubias = min(Ubias + 100.0*Ulimited/ma.SR - 0.0*Ubias/ma.SR, 2000.0);
'Uout = Uin - Ubias;
};
distortion = *(100.0) : *(ba.db2linear(fuzz/5.0) - 1.0) : biaser :
*(ba.db2linear(fuzz/100.0*6.0)) :
max(-50.0) : min(100.0) : fi.dcblocker;
filter = fi.high_shelf(tone + 12.5, 720.0);
stomp = pre_filter : filter : distortion :
*(ba.db2linear(volume * 25.0 ) / 100.0) :
/(20.0);
output = _,_ : + : ba.bypass1(bypass, stomp) <: _,_;
};
| https://raw.githubusercontent.com/olegkapitonov/KPP-VST3/91af48938c94d5a72009e01ef139bc3de8cf8dcd/kpp_fuzz/include/kpp_fuzz.dsp | faust |
* Copyright (C) 2018-2020 Oleg Kapitonov
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* --------------------------------------------------------------------------
* This plugin is a vintage fuzz pedal emulator.
*
* Process chain:
*
* input->pre_filter->*drive_knob->fuzz->tone->*volume_knob->output
*
* pre-filter - lowpass, 1 order, 1720 Hz. Emulates effect
* of low impedance of vintage pedal.
*
* distortion - 2 cascades, asymmetric distortion and clipper.
* Emulates distortion of old bad class A transistor cascades.
* tone - highshelf 720 Hz.
Bypass button, 0 - pedal on, 1 -pedal off (bypass on) |
declare name "kpp_fuzz";
declare author "Oleg Kapitonov";
declare license "GPLv3";
declare version "1.2";
import("stdfaust.lib");
process = output with {
bypass = checkbox("99_bypass");
fuzz = vslider("fuzz",50,0,100,0.01);
tone = vslider("tone",-7.5,-15,0,0.1);
volume = vslider("volume",0.5,0,1,0.001);
clamp = min(2.0) : max(-2.0);
pre_filter = fi.dcblocker : fi.lowpass(1, 2000.0);
biaser(Uin) = Uout letrec {
'Ulimited = Uin : max(-50.0 + Ubias) : -(Ubias);
'Ubias = min(Ubias + 100.0*Ulimited/ma.SR - 0.0*Ubias/ma.SR, 2000.0);
'Uout = Uin - Ubias;
};
distortion = *(100.0) : *(ba.db2linear(fuzz/5.0) - 1.0) : biaser :
*(ba.db2linear(fuzz/100.0*6.0)) :
max(-50.0) : min(100.0) : fi.dcblocker;
filter = fi.high_shelf(tone + 12.5, 720.0);
stomp = pre_filter : filter : distortion :
*(ba.db2linear(volume * 25.0 ) / 100.0) :
/(20.0);
output = _,_ : + : ba.bypass1(bypass, stomp) <: _,_;
};
|
bcfd494d965d9550486fbce23ccd4c1a68d8d6c0d9f8452ec0c1215220efdd1e | SpotlightKid/faustfilters | oberheim.dsp | declare name "Oberheim";
declare description "Oberheim multi-mode, state-variable filter";
declare author "Christopher Arndt";
declare license "MIT-style STK-4.3 license";
import("stdfaust.lib");
//==================================Oberheim Filters======================================
// The following filter (4 types) is an implementation of the virtual analog
// model described in Section 7.2 of the Will Pirkle book, "Designing Software
// Synthesizer Plug-ins in C++. It is based on the block diagram in Figure 7.5.
//
// The Oberheim filter is a state-variable filter with soft-clipping distortion
// within the circuit.
//
// In many VA filters, distortion is accomplished using the "tanh" function.
// For this Faust implementation, that distortion function was replaced with
// the `(ef.)cubicnl` function.
//========================================================================================
//------------------`oberheim`-----------------
// Generic multi-outputs Oberheim filter (see description above).
//
// #### Usage
//
// ```
// _ : oberheim(normFreq,Q) : _,_,_,_
// ```
//
// Where:
//
// * `freq`: cutoff / center frequency (20.0 - 20000.0 Hzz)
// * `Q`: q (0.5 - 10.0)
//---------------------------------------------------------------------
declare oberheim author "Eric Tarr";
declare oberheim license "MIT-style STK-4.3 license";
oberheim(freq, Q) = _<:(s1,s2,ybsf,ybpf,yhpf,ylpf) : !,!,_,_,_,_
letrec{
's1 = _-s2:_-(s1*FBs1):_*alpha0:_*g<:_,(_+s1:ef.cubicnl(0.0,0)):>_;
's2 = _-s2:_-(s1*FBs1):_*alpha0:_*g:_+s1:ef.cubicnl(0.0,0):_*g*2:_+s2;
// Compute the BSF, BPF, HPF, LPF outputs
'ybsf = _-s2:_-(s1*FBs1):_*alpha0<:(_*g:_+s1:ef.cubicnl(0.0,0):_*g:_+s2),_:>_;
'ybpf = _-s2:_-(s1*FBs1):_*alpha0:_*g:_+s1:ef.cubicnl(0.0,0);
'yhpf = _-s2:_-(s1*FBs1):_*alpha0;
'ylpf = _-s2:_-(s1*FBs1):_*alpha0:_*g :_+s1:ef.cubicnl(0.0,0):_*g:_+s2;
}
with{
// freq = 2*(10^(3*normFreq+1));
wd = 2*ma.PI*freq;
T = 1/ma.SR;
wa = (2/T)*tan(wd*T/2);
g = wa*T/2;
G = g/(1.0 + g);
R = 1/(2*Q);
FBs1 = (2*R+g);
alpha0 = 1/(1 + 2*R*g + g*g);
};
q = hslider("[1]Q[symbol: q][abbrev: q][style:knob]", 1.0, 0.5, 10.0, 0.01);
cutoff = hslider("[0]Cutoff frequency[symbol: cutoff][abbrev: cutoff][unit: hz][scale: log][style: knob]", 20000.0, 20.0, 20000, 0.1):si.smoo;
process = oberheim(cutoff, q);
| https://raw.githubusercontent.com/SpotlightKid/faustfilters/8dfb35de7b83935806abe950187e056623b6c01a/faust/oberheim.dsp | faust | ==================================Oberheim Filters======================================
The following filter (4 types) is an implementation of the virtual analog
model described in Section 7.2 of the Will Pirkle book, "Designing Software
Synthesizer Plug-ins in C++. It is based on the block diagram in Figure 7.5.
The Oberheim filter is a state-variable filter with soft-clipping distortion
within the circuit.
In many VA filters, distortion is accomplished using the "tanh" function.
For this Faust implementation, that distortion function was replaced with
the `(ef.)cubicnl` function.
========================================================================================
------------------`oberheim`-----------------
Generic multi-outputs Oberheim filter (see description above).
#### Usage
```
_ : oberheim(normFreq,Q) : _,_,_,_
```
Where:
* `freq`: cutoff / center frequency (20.0 - 20000.0 Hzz)
* `Q`: q (0.5 - 10.0)
---------------------------------------------------------------------
Compute the BSF, BPF, HPF, LPF outputs
freq = 2*(10^(3*normFreq+1)); | declare name "Oberheim";
declare description "Oberheim multi-mode, state-variable filter";
declare author "Christopher Arndt";
declare license "MIT-style STK-4.3 license";
import("stdfaust.lib");
declare oberheim author "Eric Tarr";
declare oberheim license "MIT-style STK-4.3 license";
oberheim(freq, Q) = _<:(s1,s2,ybsf,ybpf,yhpf,ylpf) : !,!,_,_,_,_
letrec{
's1 = _-s2:_-(s1*FBs1):_*alpha0:_*g<:_,(_+s1:ef.cubicnl(0.0,0)):>_;
's2 = _-s2:_-(s1*FBs1):_*alpha0:_*g:_+s1:ef.cubicnl(0.0,0):_*g*2:_+s2;
'ybsf = _-s2:_-(s1*FBs1):_*alpha0<:(_*g:_+s1:ef.cubicnl(0.0,0):_*g:_+s2),_:>_;
'ybpf = _-s2:_-(s1*FBs1):_*alpha0:_*g:_+s1:ef.cubicnl(0.0,0);
'yhpf = _-s2:_-(s1*FBs1):_*alpha0;
'ylpf = _-s2:_-(s1*FBs1):_*alpha0:_*g :_+s1:ef.cubicnl(0.0,0):_*g:_+s2;
}
with{
wd = 2*ma.PI*freq;
T = 1/ma.SR;
wa = (2/T)*tan(wd*T/2);
g = wa*T/2;
G = g/(1.0 + g);
R = 1/(2*Q);
FBs1 = (2*R+g);
alpha0 = 1/(1 + 2*R*g + g*g);
};
q = hslider("[1]Q[symbol: q][abbrev: q][style:knob]", 1.0, 0.5, 10.0, 0.01);
cutoff = hslider("[0]Cutoff frequency[symbol: cutoff][abbrev: cutoff][unit: hz][scale: log][style: knob]", 20000.0, 20.0, 20000, 0.1):si.smoo;
process = oberheim(cutoff, q);
|
00a243daaac4fc94394d8d8e8e3d08a32bbd23e93bc8c9ee519aba3dbe7852da | mwicat/faustfx | fig8verb.dsp | /*
* Synthetic reverberator simulator
*
* Washy and synthetic reverb, sounds gorgeous when modulated.
* The core architecture is two channels of allpass filter series
* terminated by delays with their feedbacks crossed as described by Keith Barr at Spin Semiconductor
* (http://www.spinsemi.com/knowledge_base/effects.html#Reverberation)
*
* The algorithm is based on NI Reaktor Metaverb.
* Metaverb is a reverb effect from the ensemble "Cha Osc" by Stephan Schmitt.
*/
declare name "fig8verb";
declare author "Marek Wiewiorski";
declare version "0.4";
declare license "MIT";
import("stdfaust.lib");
size_scale(0) = 0.506392;
size_scale(1) = 0.803821;
size_scale(2) = 1;
size_scale(3) = 1.49834;
fig8verb(
max_dtime,
size,
diffusion,
feedback_gain,
lfo_freq,
lfo_amount,
hs_gain,
hs_freq,
ls_gain,
ls_freq) = (route_in : +,+ : core) ~ feedback with {
route_in = route(4, 4, (1, 3), (2, 2), (3, 1), (4, 4));
filter_hs = fi.highshelf(1, hs_gain, hs_freq);
filter_ls = fi.lowshelf(1, ls_gain, ls_freq);
filter = filter_hs : filter_ls;
lfo_phase(i) = ma.PI/4 * i;
lfo(i) = lfo_amount * os.oscp(lfo_freq, lfo_phase(i));
dtime(ch, i) = (size_scale(i) * size) * (1 + lfo(ch*i));
allpass(ch, i) = fi.allpass_fcomb(max_dtime, dtime(ch, i), diffusion);
diffusor(ch) = filter : seq(i, 3, allpass(ch, i));
delay(ch) = de.fdelay(max_dtime, dtime(ch, 3));
feedback = par(i, 2, delay(i) * feedback_gain);
core = par(i, 2, diffusor(i));
};
fig8verb_ui = (_,_) <: ((_, _), fig8verb(
max_dtime,
size,
diffusion,
feedback,
lfo_freq,
lfo_amount,
hs_gain,
hs_freq,
ls_gain,
ls_freq)) : route(4, 4, (1, 1), (2, 3), (3, 2), (4, 4)) : par(i, 2, si.interpolate(mix)) with {
max_dtime = 5000;
general_group(x) = hgroup("[0] General [tooltip:General controls]", x);
mix = general_group(vslider("[0] Mix [unit:%] [tooltip:Mix] [style:knob]", 100, 0, 100, 0.1)) * 0.01;
size = general_group(vslider("[1] Size [unit:%] [tooltip:Scale size in percents] [style:knob]", 60, 0, 100, 0.1)) * 0.01 * max_dtime : si.smooth(0.99);
feedback = general_group(vslider("[2] Feedback [unit:%] [tooltip:Feedback amount] [style:knob]", 60, 0, 100, 0.1)) * 0.01;
diffusion = general_group(vslider("[3] Diffusion [unit:%] [tooltip:Diffusion amount] [style:knob]", 66, 0, 100, 0.1)) * 0.01;
damping_group(x) = hgroup("[1] Damping [tooltip:Damping filters in the feedback loop]", x);
hs_gain = damping_group(hslider("[0] HighShelf [unit:dB] [tooltip:High shelf gain] [style:knob]", 0, -24, 24, 0.1));
hs_freq = damping_group(hslider("[1] HighFreq [unit:Hz] [style:knob] [tooltip:High shelf frequency] [scale:log]", 4000, 500, 10000, 10));
ls_gain = damping_group(hslider("[2] LowShelf [unit:dB] [tooltip:Low shelf gain] [style:knob]", 0, -24, 24, 0.1));
ls_freq = damping_group(hslider("[3] LowFreq [unit:Hz] [tooltip:Low Shelf Frequency] [style:knob] [scale:log]", 100, 100, 4000, 10));
modulation_group(x) = hgroup("[2] Modulation [tooltip:Control over modulation of delay times] [style:knob]", x);
lfo_freq = modulation_group(vslider("[0] Rate [unit:Hz] [tooltip:Modulation LFO frequency] [style:knob]", 0.7, 0, 10, 0.1));
lfo_amount = modulation_group(vslider("[1] Depth [unit:%] [tooltip:Modulation LFO amount] [style:knob]", 0.3, 0, 5, 0.6)) * 0.01;
};
process = fig8verb_ui;
| https://raw.githubusercontent.com/mwicat/faustfx/3fded3fd42b221eccbeec4ef79a28c0faa0bb6e9/docs/fig8verb/fig8verb.dsp | faust |
* Synthetic reverberator simulator
*
* Washy and synthetic reverb, sounds gorgeous when modulated.
* The core architecture is two channels of allpass filter series
* terminated by delays with their feedbacks crossed as described by Keith Barr at Spin Semiconductor
* (http://www.spinsemi.com/knowledge_base/effects.html#Reverberation)
*
* The algorithm is based on NI Reaktor Metaverb.
* Metaverb is a reverb effect from the ensemble "Cha Osc" by Stephan Schmitt.
|
declare name "fig8verb";
declare author "Marek Wiewiorski";
declare version "0.4";
declare license "MIT";
import("stdfaust.lib");
size_scale(0) = 0.506392;
size_scale(1) = 0.803821;
size_scale(2) = 1;
size_scale(3) = 1.49834;
fig8verb(
max_dtime,
size,
diffusion,
feedback_gain,
lfo_freq,
lfo_amount,
hs_gain,
hs_freq,
ls_gain,
ls_freq) = (route_in : +,+ : core) ~ feedback with {
route_in = route(4, 4, (1, 3), (2, 2), (3, 1), (4, 4));
filter_hs = fi.highshelf(1, hs_gain, hs_freq);
filter_ls = fi.lowshelf(1, ls_gain, ls_freq);
filter = filter_hs : filter_ls;
lfo_phase(i) = ma.PI/4 * i;
lfo(i) = lfo_amount * os.oscp(lfo_freq, lfo_phase(i));
dtime(ch, i) = (size_scale(i) * size) * (1 + lfo(ch*i));
allpass(ch, i) = fi.allpass_fcomb(max_dtime, dtime(ch, i), diffusion);
diffusor(ch) = filter : seq(i, 3, allpass(ch, i));
delay(ch) = de.fdelay(max_dtime, dtime(ch, 3));
feedback = par(i, 2, delay(i) * feedback_gain);
core = par(i, 2, diffusor(i));
};
fig8verb_ui = (_,_) <: ((_, _), fig8verb(
max_dtime,
size,
diffusion,
feedback,
lfo_freq,
lfo_amount,
hs_gain,
hs_freq,
ls_gain,
ls_freq)) : route(4, 4, (1, 1), (2, 3), (3, 2), (4, 4)) : par(i, 2, si.interpolate(mix)) with {
max_dtime = 5000;
general_group(x) = hgroup("[0] General [tooltip:General controls]", x);
mix = general_group(vslider("[0] Mix [unit:%] [tooltip:Mix] [style:knob]", 100, 0, 100, 0.1)) * 0.01;
size = general_group(vslider("[1] Size [unit:%] [tooltip:Scale size in percents] [style:knob]", 60, 0, 100, 0.1)) * 0.01 * max_dtime : si.smooth(0.99);
feedback = general_group(vslider("[2] Feedback [unit:%] [tooltip:Feedback amount] [style:knob]", 60, 0, 100, 0.1)) * 0.01;
diffusion = general_group(vslider("[3] Diffusion [unit:%] [tooltip:Diffusion amount] [style:knob]", 66, 0, 100, 0.1)) * 0.01;
damping_group(x) = hgroup("[1] Damping [tooltip:Damping filters in the feedback loop]", x);
hs_gain = damping_group(hslider("[0] HighShelf [unit:dB] [tooltip:High shelf gain] [style:knob]", 0, -24, 24, 0.1));
hs_freq = damping_group(hslider("[1] HighFreq [unit:Hz] [style:knob] [tooltip:High shelf frequency] [scale:log]", 4000, 500, 10000, 10));
ls_gain = damping_group(hslider("[2] LowShelf [unit:dB] [tooltip:Low shelf gain] [style:knob]", 0, -24, 24, 0.1));
ls_freq = damping_group(hslider("[3] LowFreq [unit:Hz] [tooltip:Low Shelf Frequency] [style:knob] [scale:log]", 100, 100, 4000, 10));
modulation_group(x) = hgroup("[2] Modulation [tooltip:Control over modulation of delay times] [style:knob]", x);
lfo_freq = modulation_group(vslider("[0] Rate [unit:Hz] [tooltip:Modulation LFO frequency] [style:knob]", 0.7, 0, 10, 0.1));
lfo_amount = modulation_group(vslider("[1] Depth [unit:%] [tooltip:Modulation LFO amount] [style:knob]", 0.3, 0, 5, 0.6)) * 0.01;
};
process = fig8verb_ui;
|
83a829947c6fd75b30bc809219885a68ae9f76c85dea8c44ab79943d1b596781 | rottingsounds/bitDSP-faust | dsm2.dsp | declare name "DSM2";
declare description "Second-order delta-sigma modulator example";
Second-order delta-sigma modulator - example
declare author "Dario Sanfilippo";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
// plot
// CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib dsm2.dsp
// ./dsm2 -n 10
// compile
// CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib dsm2.dsp
// ./dsm2
// High-precision sinewave
sine(f) = sin(os.phasor(2 * ma.PI, f));
// Bipolar multi-bit signal to bipolar one-bit signal
// Standard test with a 1 kHz tone
onebitstream = bit.dsm2(sine(1000));
// Bipolar one-bit signal to bipolar multi-bit signal
// The process of low-passing corresponds to averaging
// The low-pass cut-off sets the target bandwidth
// The low-pass resolution of the coefficients sets the bitdepth
// The low-pass order determines the accuracy in the noise removal
multibitstream = fi.lowpass(4, 1000, onebitstream);
// Final output
process = multibitstream;
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/21cf36105c55b6e18969a867a319530a0ef1ea63/examples/dsm2.dsp | faust | plot
CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib dsm2.dsp
./dsm2 -n 10
compile
CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib dsm2.dsp
./dsm2
High-precision sinewave
Bipolar multi-bit signal to bipolar one-bit signal
Standard test with a 1 kHz tone
Bipolar one-bit signal to bipolar multi-bit signal
The process of low-passing corresponds to averaging
The low-pass cut-off sets the target bandwidth
The low-pass resolution of the coefficients sets the bitdepth
The low-pass order determines the accuracy in the noise removal
Final output | declare name "DSM2";
declare description "Second-order delta-sigma modulator example";
Second-order delta-sigma modulator - example
declare author "Dario Sanfilippo";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
sine(f) = sin(os.phasor(2 * ma.PI, f));
onebitstream = bit.dsm2(sine(1000));
multibitstream = fi.lowpass(4, 1000, onebitstream);
process = multibitstream;
|
278366ea7c5aec4d3aac5e18769fcd2cdf18b0d05dd2e6a26eeb445f5aadbe39 | rottingsounds/bitDSP-faust | dsm1.dsp | declare name "dsm1";
declare description "First-order delta-sigma modulator - example";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
// plot
// CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib dsm1-example.dsp
// ./dsm1-example -n 10
// compile
// CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib dsm1-example.dsp
// ./dsm1-example
// High-precision sinewave
sine(f) = sin(os.phasor(2 * ma.PI, f));
// Bipolar multi-bit signal to bipolar one-bit signal
// Standard test with a 1 kHz tone
onebitstream = bit.dsm1(sine(1000));
// Bipolar one-bit signal to bipolar multi-bit signal
// The process of low-passing corresponds to averaging
// The low-pass cut-off sets the target bandwidth
// The low-pass resolution of the coefficients sets the bitdepth
// The low-pass order determines the accuracy in the noise removal
multibitstream = fi.lowpass(4, 1000, onebitstream);
// Final output
process = multibitstream;
| https://raw.githubusercontent.com/rottingsounds/bitDSP-faust/21cf36105c55b6e18969a867a319530a0ef1ea63/examples/dsm1.dsp | faust | plot
CXXFLAGS="-I ../include" faust2csvplot -double -I ../lib dsm1-example.dsp
./dsm1-example -n 10
compile
CXXFLAGS="-I ../../../include" faust2caqt -double -I ../lib dsm1-example.dsp
./dsm1-example
High-precision sinewave
Bipolar multi-bit signal to bipolar one-bit signal
Standard test with a 1 kHz tone
Bipolar one-bit signal to bipolar multi-bit signal
The process of low-passing corresponds to averaging
The low-pass cut-off sets the target bandwidth
The low-pass resolution of the coefficients sets the bitdepth
The low-pass order determines the accuracy in the noise removal
Final output | declare name "dsm1";
declare description "First-order delta-sigma modulator - example";
declare author "Till Bovermann";
declare reference "http://rottingsounds.org";
import("stdfaust.lib");
bit = library("bitDSP.lib");
sine(f) = sin(os.phasor(2 * ma.PI, f));
onebitstream = bit.dsm1(sine(1000));
multibitstream = fi.lowpass(4, 1000, onebitstream);
process = multibitstream;
|
810bc31712f1527a9c3a184bcb1e223bc8ed227365e422881226ff90630bd6d5 | friskgit/snares | plain_snare.dsp | // -*- compile-command: "cd .. && make jack src=src/snare.dsp && cd -"; -*-
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
//---------------`Snare drum synth` --------------------------
// A take at a snare drum synth
//
// A single hit snare drum synth controllable with midi. Each hit is distribute over `channels` speakers.
// It has its own trigger
//
// Where:
// * midi note 67-89
// * stiffness 0-0.55 (mapped to note as in note 67 -> 0)§
// * midi velocity is mapped to pressure
//
// A useful parameter setting is:
//
// 30 Juni 2018 Henrik Frisk [email protected]
//---------------------------------------------------
// Set the number of channels at compile time.
channels = 16;
// Main impulse for generating one hit.
imp = button("gate");
// GUI
posgroup(x) = vgroup("position", x);
snaregroup(x) = vgroup("snare", x);
// Main envelope
env = en.ar(attack, rel, imp) * amp
with {
attack = snaregroup(hslider("attack", 0.00000001, 0, 0.1, 0.000000001) : si.smooth(0.1));
rel = snaregroup(hslider("rel", 0.1, 0.0000001, 0.5, 0.0000001) : si.smooth(0.2));
amp = snaregroup(hslider("vol", 0.5, 0, 1, 0.0001));
};
// Control the output channel
pimp = imp : ba.impulsify;
focus = posgroup(hslider("disperse", 1, 0, 1, 0.0001));
position = posgroup(hslider("displace", 1, 0, channels, 1));
rate = ma.SR/1000.0;
rndctrl = (no.lfnoise(rate) * (channels + 1)) * focus : ma.fabs + position : int ;
outputctrl = rndctrl : ba.sAndH(pimp);
// Wrap channels around the array.
ch_wrapped = ma.modulo(outputctrl, channels);
// Noise generation and filter
snare(n) = no.multinoise(8) : par(i, 8, _ * env * 0.1);
filt = fi.resonbp(frq, q, gain)
with {
frq = snaregroup(hslider("freq", 200, 50, 5000, 0.1));
q = snaregroup(hslider("q", 1, 0.01, 10, 0.01));
gain = snaregroup(hslider("gain", 0, 0, 2, 0.00001));
};
process = vgroup("snaredrum",
snare : par(i, 8, filt) :> _,_ :>
ba.selectoutn(channels, ch_wrapped)
);
| https://raw.githubusercontent.com/friskgit/snares/bb43ea5e706a0ead6d65dd176a5c492b2f5d8f74/faust/snare/src/extras/plain_snare.dsp | faust | -*- compile-command: "cd .. && make jack src=src/snare.dsp && cd -"; -*-
---------------`Snare drum synth` --------------------------
A take at a snare drum synth
A single hit snare drum synth controllable with midi. Each hit is distribute over `channels` speakers.
It has its own trigger
Where:
* midi note 67-89
* stiffness 0-0.55 (mapped to note as in note 67 -> 0)§
* midi velocity is mapped to pressure
A useful parameter setting is:
30 Juni 2018 Henrik Frisk [email protected]
---------------------------------------------------
Set the number of channels at compile time.
Main impulse for generating one hit.
GUI
Main envelope
Control the output channel
Wrap channels around the array.
Noise generation and filter |
declare version " 0.1 ";
declare author " Henrik Frisk " ;
declare author " henrikfr ";
declare license " BSD ";
declare copyright "(c) dinergy 2018 ";
import("stdfaust.lib");
channels = 16;
imp = button("gate");
posgroup(x) = vgroup("position", x);
snaregroup(x) = vgroup("snare", x);
env = en.ar(attack, rel, imp) * amp
with {
attack = snaregroup(hslider("attack", 0.00000001, 0, 0.1, 0.000000001) : si.smooth(0.1));
rel = snaregroup(hslider("rel", 0.1, 0.0000001, 0.5, 0.0000001) : si.smooth(0.2));
amp = snaregroup(hslider("vol", 0.5, 0, 1, 0.0001));
};
pimp = imp : ba.impulsify;
focus = posgroup(hslider("disperse", 1, 0, 1, 0.0001));
position = posgroup(hslider("displace", 1, 0, channels, 1));
rate = ma.SR/1000.0;
rndctrl = (no.lfnoise(rate) * (channels + 1)) * focus : ma.fabs + position : int ;
outputctrl = rndctrl : ba.sAndH(pimp);
ch_wrapped = ma.modulo(outputctrl, channels);
snare(n) = no.multinoise(8) : par(i, 8, _ * env * 0.1);
filt = fi.resonbp(frq, q, gain)
with {
frq = snaregroup(hslider("freq", 200, 50, 5000, 0.1));
q = snaregroup(hslider("q", 1, 0.01, 10, 0.01));
gain = snaregroup(hslider("gain", 0, 0, 2, 0.00001));
};
process = vgroup("snaredrum",
snare : par(i, 8, filt) :> _,_ :>
ba.selectoutn(channels, ch_wrapped)
);
|
1bcee603cd2dbd5bc39963e1ea475e1538a582efab9a151f7130523accd1f01e | RuolunWeng/ruolunweng.github.io | SNoiseS.dsp | declare name "Noises";
declare author "ER";
declare version "1.0";
import("stdfaust.lib");
/* ============ DESCRIPTION =============:
- White Noise and Pink Noise generator.
- Head = silence.
- Fishing rod = Volume variation.
- Right = silence.
- Face = pink no.noise.
- Left = white no.noise.
- Rocking to switch from one no.noise to another.
*/
// pink no.noise filter (-3dB per octave), see musicdsp.org
p = f : (+ ~ g) with {
f(x) = 0.04957526213389*x - 0.06305581334498*x' +
0.01483220320740*x'';
g(x) = 1.80116083982126*x - 0.80257737639225*x';
};
// white no.noise generator
rand = +(12345)~*(1103515245);
w = rand/2147483647.0;
White = w * hslider("White Noise Volume[acc:1 0 -10 0 10][style:knob]", 0.5, 0, 2, 0.01);
Pink = (w : p) * (2) * hslider("Pink Noise Volume[acc:1 0 -10 0 10][tooltip:0=Mute, 1=White Noise, 2=Pink Noise][style:knob]", 0.5, 0, 2, 0.01);
NoiseType = hslider("Noise Type[acc:0 0 -10 0 10]", 1,0,2,1);
Ntype(n) = abs(NoiseType - n) < 0.5;
Noise(0) = 0;
Noise(1) = Pink;
Noise(2) = White;
process = par(i, 3, Noise(i) * Ntype(i)) :>_;
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SNoiseS.dsp | faust | ============ DESCRIPTION =============:
- White Noise and Pink Noise generator.
- Head = silence.
- Fishing rod = Volume variation.
- Right = silence.
- Face = pink no.noise.
- Left = white no.noise.
- Rocking to switch from one no.noise to another.
pink no.noise filter (-3dB per octave), see musicdsp.org
white no.noise generator | declare name "Noises";
declare author "ER";
declare version "1.0";
import("stdfaust.lib");
p = f : (+ ~ g) with {
f(x) = 0.04957526213389*x - 0.06305581334498*x' +
0.01483220320740*x'';
g(x) = 1.80116083982126*x - 0.80257737639225*x';
};
rand = +(12345)~*(1103515245);
w = rand/2147483647.0;
White = w * hslider("White Noise Volume[acc:1 0 -10 0 10][style:knob]", 0.5, 0, 2, 0.01);
Pink = (w : p) * (2) * hslider("Pink Noise Volume[acc:1 0 -10 0 10][tooltip:0=Mute, 1=White Noise, 2=Pink Noise][style:knob]", 0.5, 0, 2, 0.01);
NoiseType = hslider("Noise Type[acc:0 0 -10 0 10]", 1,0,2,1);
Ntype(n) = abs(NoiseType - n) < 0.5;
Noise(0) = 0;
Noise(1) = Pink;
Noise(2) = White;
process = par(i, 3, Noise(i) * Ntype(i)) :>_;
|
1f2d2286106107416e2fa49fbb101ccc6a0ad929762a54871fa516bf5792f13c | sonejostudios/SuperCutSequencer | SuperCutSequencer.dsp | declare name "SuperCutSequencer";
declare version "1.2";
declare author "Vincent Rateau";
declare license "GPL v3";
declare reference "www.sonejo.net";
declare description "Cut 'On/Off' Sequencer 8 steps with smooth) synced to Midi-Clock Beats and Midi-Clock Start/Stop";
//import("signals.lib");
import("stdfaust.lib");
// Cut "On/Off" Sequencer 8 steps with smooth) synced to Midi-Clock Beats and Midi-Clock Start/Stop.
process = cutsequencer, cutsequencer;
//GLOBAL VARIABLES
///////////////////////////////////////
// midi ctrl out numbers to start iterations
//steppush and stepled are used to on/off the steps (and light back the state of each step )
//seqled shows the current played step
steppush = 57; // controls checkboxes step on/off
stepled = 57; // controls lightning of buttons if step is on or off
seqled = 41; // controls lighning of played step
//CUT SEQUENCER
///////////////////////////////////////
cutsequencer = _ <: (_ <: cutseq :> _), _ : drywet :> _
with{
// dry / wet knob
drywet = _* drywetgui , _* (1-drywetgui) ;
drywetgui = vslider("[08]Dry/Wet[style:knob][midi: ctrl 52]", 0, 0, 1, 0.001) : s ;
//create par 8 with on/off checkboxes, send them sif (counter conditions)
cutseq(a,b,c,d,e,f,g,h) = (par(i,8, hgroup("[4]Step On/Off", stepmute(i + steppush, i + stepled))) : sif) :
hgroup("[02]",result) : s*a, s*b, s*c, s*d, s*e, s*f, s*g, s*h ;
// checkboxes and bargraphs for each steps
stepmute(j, k) = checkbox("%j [midi: ctrl %j]") :_*stepcolor :
vbargraph("%k [style: led] [midi:ctrl %k]", 0, 127) : _/stepcolor
with{
// midi out led color
stepcolor = nentry("step color", 66, 0, 127, 1);
};
//smooth
s = si.smooth(0.999);
//s = smooth(tau2pole(interpTime));
//interpTime = hslider("Interpolation Time (s)[style:knob]",0.05,0,1,0.001);
// sequ is the sequencer created by the midi clock.
// sif = if sequ == 0, play track 0. etc. trig (1) is activated for now with lf pulsetrain
sif = par(i,8, _*(sequ==i));
//sif = _*(sequ == 0), _*(sequ == 1), _*(sequ == 2), _*(sequ == 3), _*(sequ == 4), _*(sequ == 5), _*(sequ == 6), _*(sequ == 7);
// choose a sequencer between midi clock and bpl slider
// bring midi clock sequence to conditions
//sequ = sequence : hbargraph("[2]Sequence", 0, 7) ;
sequ = seqchooser : hbargraph("[07]Sequence", 0, 7) ;
//gui sequence and lightning
//result = par(o,8, _);
result = par(o,8, leds(o + seqled));
leds(p) = vbargraph("[09]LED %p [style: led] [midi: ctrl %p]", 0, 1);
};
// SEQUENCE from BPM Slider and Sequencer Chooser
////////////////////////////////////////////
seqchooser = checkbox("[01]Sequencer Source : Midi-Clock / Bpm Slider"), sequence, sequence2bpm : select2 ;
sequence2bpm = os.lf_imptrain(bpmslider) : ba.pulse_countup_loop(7, 1) ;
bpmslider = hslider("[05]BPM Slider",120,20,240,0.01) : _/60 : _/scale ;
// scale the 8-step sequence
////////////////////////////////////////////
scale = vslider("[06]Sequencer Scaling[style:menu{'faster (1/4)':-2 ; 'fast (1/2)':-1 ; 'no scaling (1)':0 ; 'slow (2)':1 ; 'slower (4)':2}]",
0, -2, 2, 1) <: ((_==-2)*0.25) , ((_==-1)*0.5) , ((_==0)*1) , ((_==1)*2), ((_==2)*4) :> _;
// SEQUENCE from MIDI CLOCK
////////////////////////////////////////////
sequence = clocker : midiclock : clock2beat
with{
// clocker is a square signal (1/0), changing state at each received midi clock
clocker = checkbox("[03]MIDI clock[midi:clock]");
// squarewave for testing only (instead of midi clock "clocker")
//sqwv = lf_squarewavepos(frqslider) ; // : vbargraph("squarewave", 0, 1) ;
//frqslider = hslider("send sqwave / sec * 24", 1, 0, 10, 0.1)*24;
// count 24 pulse and reset
midiclock = sq2pulse : counter(24*scale) // : vbargraph("counter loop 24", 0, 30);
with{
// detect front, (create pulse from square wave)
sq2pulse(x) = (x-x') != 0.0 ;
};
// pulse once a beat and add 1 to sequence number (0 to 8)
clock2beat = _ == 0 <: _-_' : _ >0 : counter(8) ; //: vbargraph("counter loop 8", 0, 10);
// count and multiply by 1 as long as counter < n (last number in loop), otherwise multiply by 0 = reset seq to zero
counter(n) = + ~ cond(n)
with{
// condition inside the loop. play resets sequence to 0
cond(n) = _ <: _, _ : ( _ < n) * _ :> _ * play ;
// Start / Stop button controlled with MIDI start/stop messages inside the loop (if stop then reset to 0)
play = checkbox("[04]Sequence Start / Stop [midi:start] [midi:stop]");
};
};
| https://raw.githubusercontent.com/sonejostudios/SuperCutSequencer/f5f4cdca3542c01dabde1306add6dad7ca8f940c/SuperCutSequencer.dsp | faust | import("signals.lib");
Cut "On/Off" Sequencer 8 steps with smooth) synced to Midi-Clock Beats and Midi-Clock Start/Stop.
GLOBAL VARIABLES
/////////////////////////////////////
midi ctrl out numbers to start iterations
steppush and stepled are used to on/off the steps (and light back the state of each step )
seqled shows the current played step
controls checkboxes step on/off
controls lightning of buttons if step is on or off
controls lighning of played step
CUT SEQUENCER
/////////////////////////////////////
dry / wet knob
create par 8 with on/off checkboxes, send them sif (counter conditions)
checkboxes and bargraphs for each steps
midi out led color
smooth
s = smooth(tau2pole(interpTime));
interpTime = hslider("Interpolation Time (s)[style:knob]",0.05,0,1,0.001);
sequ is the sequencer created by the midi clock.
sif = if sequ == 0, play track 0. etc. trig (1) is activated for now with lf pulsetrain
sif = _*(sequ == 0), _*(sequ == 1), _*(sequ == 2), _*(sequ == 3), _*(sequ == 4), _*(sequ == 5), _*(sequ == 6), _*(sequ == 7);
choose a sequencer between midi clock and bpl slider
bring midi clock sequence to conditions
sequ = sequence : hbargraph("[2]Sequence", 0, 7) ;
gui sequence and lightning
result = par(o,8, _);
SEQUENCE from BPM Slider and Sequencer Chooser
//////////////////////////////////////////
scale the 8-step sequence
//////////////////////////////////////////
SEQUENCE from MIDI CLOCK
//////////////////////////////////////////
clocker is a square signal (1/0), changing state at each received midi clock
squarewave for testing only (instead of midi clock "clocker")
sqwv = lf_squarewavepos(frqslider) ; // : vbargraph("squarewave", 0, 1) ;
frqslider = hslider("send sqwave / sec * 24", 1, 0, 10, 0.1)*24;
count 24 pulse and reset
: vbargraph("counter loop 24", 0, 30);
detect front, (create pulse from square wave)
pulse once a beat and add 1 to sequence number (0 to 8)
: vbargraph("counter loop 8", 0, 10);
count and multiply by 1 as long as counter < n (last number in loop), otherwise multiply by 0 = reset seq to zero
condition inside the loop. play resets sequence to 0
Start / Stop button controlled with MIDI start/stop messages inside the loop (if stop then reset to 0) | declare name "SuperCutSequencer";
declare version "1.2";
declare author "Vincent Rateau";
declare license "GPL v3";
declare reference "www.sonejo.net";
declare description "Cut 'On/Off' Sequencer 8 steps with smooth) synced to Midi-Clock Beats and Midi-Clock Start/Stop";
import("stdfaust.lib");
process = cutsequencer, cutsequencer;
cutsequencer = _ <: (_ <: cutseq :> _), _ : drywet :> _
with{
drywet = _* drywetgui , _* (1-drywetgui) ;
drywetgui = vslider("[08]Dry/Wet[style:knob][midi: ctrl 52]", 0, 0, 1, 0.001) : s ;
cutseq(a,b,c,d,e,f,g,h) = (par(i,8, hgroup("[4]Step On/Off", stepmute(i + steppush, i + stepled))) : sif) :
hgroup("[02]",result) : s*a, s*b, s*c, s*d, s*e, s*f, s*g, s*h ;
stepmute(j, k) = checkbox("%j [midi: ctrl %j]") :_*stepcolor :
vbargraph("%k [style: led] [midi:ctrl %k]", 0, 127) : _/stepcolor
with{
stepcolor = nentry("step color", 66, 0, 127, 1);
};
s = si.smooth(0.999);
sif = par(i,8, _*(sequ==i));
sequ = seqchooser : hbargraph("[07]Sequence", 0, 7) ;
result = par(o,8, leds(o + seqled));
leds(p) = vbargraph("[09]LED %p [style: led] [midi: ctrl %p]", 0, 1);
};
seqchooser = checkbox("[01]Sequencer Source : Midi-Clock / Bpm Slider"), sequence, sequence2bpm : select2 ;
sequence2bpm = os.lf_imptrain(bpmslider) : ba.pulse_countup_loop(7, 1) ;
bpmslider = hslider("[05]BPM Slider",120,20,240,0.01) : _/60 : _/scale ;
scale = vslider("[06]Sequencer Scaling[style:menu{'faster (1/4)':-2 ; 'fast (1/2)':-1 ; 'no scaling (1)':0 ; 'slow (2)':1 ; 'slower (4)':2}]",
0, -2, 2, 1) <: ((_==-2)*0.25) , ((_==-1)*0.5) , ((_==0)*1) , ((_==1)*2), ((_==2)*4) :> _;
sequence = clocker : midiclock : clock2beat
with{
clocker = checkbox("[03]MIDI clock[midi:clock]");
with{
sq2pulse(x) = (x-x') != 0.0 ;
};
counter(n) = + ~ cond(n)
with{
cond(n) = _ <: _, _ : ( _ < n) * _ :> _ * play ;
play = checkbox("[04]Sequence Start / Stop [midi:start] [midi:stop]");
};
};
|
374c07be6c433463d548ebef5417cb74e909e79a128e2576c23e0b627c939cf2 | sekisushai/ambitools | hoa_converter_fuma_to_acn_n3d.dsp | declare name "HOA Converter : FuMa to ACN N3D";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2016";
import("stdfaust.lib");
import("gui.lib");
//Description : this tool converts HOA signals defined with a convention 1 to HOA signals defined with convention 2. Proposed conventions are ACN N3D, ACN SN3D, FuMa. For ACN to FuMa, the ordering change is as in [1]
//[1] https://en.wikipedia.org/wiki/Ambisonic_data_exchange_formats
// Input ACN: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
// Output FuMa: 0 3 1 2 6 7 5 8 4 12 13 11 14 10 15 9 : W XYZ RSTUV KLMNOPQ
// Input FuMa: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 : W XYZ RSTUV KLMNOPQ
// Output ACN: 0 2 3 1 6 8 4 5 7 15 13 11 9 10 12 14
// Maximum required order (M = 3 for FuMa).
M = 3;
// Number of inputs
ins = (M+1)^2;
outs = ins;
// FuMa Input
conversion(3,1) = par(i,M+1,FuMaACN(i)); // FuMa to ACN_N3D
FuMaACN(0) = _*sqrt(2);
FuMaACN(1) = (ro.cross(2),_):(_,ro.cross(2)):
(_*sqrt(3),_*sqrt(3),_*sqrt(3));
FuMaACN(2) = ro.cross(5):(_,ro.cross(2),_,_):(_,_,ro.cross(3)):
(_*(sqrt(15)/2),_*(sqrt(15)/2),_*sqrt(5),_*(sqrt(15)/2),_*(sqrt(15)/2));
FuMaACN(3) = ro.cross(7):(_,ro.cross(2),ro.cross(2),_,_):(_,_,ro.cross(2),_,_,_):(_,_,_,ro.cross(4)):
(_*sqrt(35/8),_*(sqrt(35)/3),_*sqrt(224/45),_*sqrt(7),_*sqrt(224/45),_*(sqrt(35)/3),_*sqrt(35/8));
FuMaACN(m) = par(i,2*m+1,!:0); // normally they shouldn't be FuMa components for M>3
process = si.bus(ins):hgroup("[1]FuMa",par(i,M+1,meterm(i))):conversion(3,1):hgroup("[2]ACN N3D",par(i,M+1,meterm(i))); | https://raw.githubusercontent.com/sekisushai/ambitools/2d21b7cc7cfe9bc35d91d51ec05bf9250372f0ce/Faust/src/hoa_converter_fuma_to_acn_n3d.dsp | faust | Description : this tool converts HOA signals defined with a convention 1 to HOA signals defined with convention 2. Proposed conventions are ACN N3D, ACN SN3D, FuMa. For ACN to FuMa, the ordering change is as in [1]
[1] https://en.wikipedia.org/wiki/Ambisonic_data_exchange_formats
Input ACN: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Output FuMa: 0 3 1 2 6 7 5 8 4 12 13 11 14 10 15 9 : W XYZ RSTUV KLMNOPQ
Input FuMa: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 : W XYZ RSTUV KLMNOPQ
Output ACN: 0 2 3 1 6 8 4 5 7 15 13 11 9 10 12 14
Maximum required order (M = 3 for FuMa).
Number of inputs
FuMa Input
FuMa to ACN_N3D
normally they shouldn't be FuMa components for M>3 | declare name "HOA Converter : FuMa to ACN N3D";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2016";
import("stdfaust.lib");
import("gui.lib");
M = 3;
ins = (M+1)^2;
outs = ins;
FuMaACN(0) = _*sqrt(2);
FuMaACN(1) = (ro.cross(2),_):(_,ro.cross(2)):
(_*sqrt(3),_*sqrt(3),_*sqrt(3));
FuMaACN(2) = ro.cross(5):(_,ro.cross(2),_,_):(_,_,ro.cross(3)):
(_*(sqrt(15)/2),_*(sqrt(15)/2),_*sqrt(5),_*(sqrt(15)/2),_*(sqrt(15)/2));
FuMaACN(3) = ro.cross(7):(_,ro.cross(2),ro.cross(2),_,_):(_,_,ro.cross(2),_,_,_):(_,_,_,ro.cross(4)):
(_*sqrt(35/8),_*(sqrt(35)/3),_*sqrt(224/45),_*sqrt(7),_*sqrt(224/45),_*(sqrt(35)/3),_*sqrt(35/8));
process = si.bus(ins):hgroup("[1]FuMa",par(i,M+1,meterm(i))):conversion(3,1):hgroup("[2]ACN N3D",par(i,M+1,meterm(i))); |
001de413be5bd03810701f5c5a9cd5ec19619b2971c489d7d17d408ccfec23af | sekisushai/ambitools | hoa_converter_acn_sn3d_to_fuma.dsp | declare name "HOA Converter : ACN SN3D to FuMa";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2016";
import("stdfaust.lib");
import("gui.lib");
//Description : this tool converts HOA signals defined with a convention 1 to HOA signals defined with convention 2. Proposed conventions are ACN N3D, ACN SN3D, FuMa. For ACN to FuMa, the ordering change is as in [1]
//[1] https://en.wikipedia.org/wiki/Ambisonic_data_exchange_formats
// Input ACN: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
// Output FuMa: 0 3 1 2 6 7 5 8 4 12 13 11 14 10 15 9 : W XYZ RSTUV KLMNOPQ
// Input FuMa: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 : W XYZ RSTUV KLMNOPQ
// Output ACN: 0 2 3 1 6 8 4 5 7 15 13 11 9 10 12 14
// Maximum required order (M = 3 for FuMa).
M = 3;
// Number of inputs
ins = (M+1)^2;
outs = ins;
// ACN_SN3D Input
conversion(2,3) = par(m,M+1,par(n,2*m+1,_*sqrt(2*m+1))):par(i,M+1,ACNFuMa(i)); // ACN_SN3D to FuMa : ACN_SN3D to ACN_N3D to FuMa
ACNFuMa(0) = _*(1/sqrt(2));
ACNFuMa(1) = ro.cross(3):(_,ro.cross(2)):
(_*(1/sqrt(3)),_*(1/sqrt(3)),_*(1/sqrt(3)));
ACNFuMa(2) = (ro.cross(3),_,_):(_,ro.cross(3),_):(_,_,ro.cross(2),_):(_,_,_,ro.cross(2)):
(_*(1/sqrt(5)),_*(2/sqrt(15)),_*(2/sqrt(15)),_*(2/sqrt(15)),_*(2/sqrt(15)));
ACNFuMa(3) = (ro.cross(4),_,_,_):(_,ro.cross(4),_,_):(_,_,ro.cross(3),_,_):(_,_,_,ro.cross(3),_):(_,_,_,_,ro.cross(2),_):(_,_,_,_,_,ro.cross(2)):
(_*(1/sqrt(7)),_*sqrt(45/224),_*sqrt(45/224),_*(3/sqrt(35)),_*(3/sqrt(35)),_*sqrt(8/35),_*sqrt(8/35));
ACNFuMa(m) = par(i,2*m+1,!:0);
process = si.bus(ins):hgroup("[1]ACN SN3D",par(i,M+1,meterm(i))):conversion(2,3):hgroup("[2]FuMa",par(i,M+1,meterm(i))); | https://raw.githubusercontent.com/sekisushai/ambitools/2d21b7cc7cfe9bc35d91d51ec05bf9250372f0ce/Faust/src/hoa_converter_acn_sn3d_to_fuma.dsp | faust | Description : this tool converts HOA signals defined with a convention 1 to HOA signals defined with convention 2. Proposed conventions are ACN N3D, ACN SN3D, FuMa. For ACN to FuMa, the ordering change is as in [1]
[1] https://en.wikipedia.org/wiki/Ambisonic_data_exchange_formats
Input ACN: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Output FuMa: 0 3 1 2 6 7 5 8 4 12 13 11 14 10 15 9 : W XYZ RSTUV KLMNOPQ
Input FuMa: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 : W XYZ RSTUV KLMNOPQ
Output ACN: 0 2 3 1 6 8 4 5 7 15 13 11 9 10 12 14
Maximum required order (M = 3 for FuMa).
Number of inputs
ACN_SN3D Input
ACN_SN3D to FuMa : ACN_SN3D to ACN_N3D to FuMa | declare name "HOA Converter : ACN SN3D to FuMa";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2016";
import("stdfaust.lib");
import("gui.lib");
M = 3;
ins = (M+1)^2;
outs = ins;
ACNFuMa(0) = _*(1/sqrt(2));
ACNFuMa(1) = ro.cross(3):(_,ro.cross(2)):
(_*(1/sqrt(3)),_*(1/sqrt(3)),_*(1/sqrt(3)));
ACNFuMa(2) = (ro.cross(3),_,_):(_,ro.cross(3),_):(_,_,ro.cross(2),_):(_,_,_,ro.cross(2)):
(_*(1/sqrt(5)),_*(2/sqrt(15)),_*(2/sqrt(15)),_*(2/sqrt(15)),_*(2/sqrt(15)));
ACNFuMa(3) = (ro.cross(4),_,_,_):(_,ro.cross(4),_,_):(_,_,ro.cross(3),_,_):(_,_,_,ro.cross(3),_):(_,_,_,_,ro.cross(2),_):(_,_,_,_,_,ro.cross(2)):
(_*(1/sqrt(7)),_*sqrt(45/224),_*sqrt(45/224),_*(3/sqrt(35)),_*(3/sqrt(35)),_*sqrt(8/35),_*sqrt(8/35));
ACNFuMa(m) = par(i,2*m+1,!:0);
process = si.bus(ins):hgroup("[1]ACN SN3D",par(i,M+1,meterm(i))):conversion(2,3):hgroup("[2]FuMa",par(i,M+1,meterm(i))); |
9eb0df1e7c25af00842582cb6d32dfcfeb4926690ee9e9ef3c58bcbc28a93da6 | sekisushai/ambitools | hoa_beamforming_hypercardioid_to_mono.dsp | declare name "HOA-Beamformer to Mono";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2015";
// Description: This tool helps to extract a monophonic signal from the HOA scene with a beampatern. The Ambisonics components inputs are combined to produce a monophonic output as if the capture was done with a directionnal microphone. The beampatern provided are regular hypercardioid up to order 5 [1].
// References:
// [1] Meyer, J., & Elko, G. (2002). A highly scalable spherical microphone array based on an orthonormal decomposition of the soundfield. 2002 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2, II–1781–II–1784.
// Inputs: (M+1)^2
// Outputs: 1
import("stdfaust.lib");
import("ymn.lib");
import("gui.lib");
// Maximum order of original HOA scene.
M = 3;
// Output gain
vol = vslider("[2]Output Gain", 0, -10, 10, 0.1) : ba.db2linear : si.smooth(0.999);
t = vslider("[3]Azimuth", 0, 0, 360, 0.1)*ma.PI/180;
d = vslider("[4]Elevation", 0, -90, 90, 0.1)*ma.PI/180;
order(s) = hslider("Order[style:knob]",0,0,M,0.0001)<:select2(s,int,_); // Order of the beampattern used for filtering, order=0 is a bypass.
crossfade(i,x) = par(j,i,_*(1-abs(x-j):max(0))):>_; // linear crossfade between order.
step = checkbox("Int/Float");
norm(m) = 1/sqrt(2*m+1);
// ORDER 1
hypercoeff(1,0) = 0.24993;
hypercoeff(1,1) = 0.433017;
// ORDER 2
hypercoeff(2,0) = 0.11112;
hypercoeff(2,1) = 0.19245;
hypercoeff(2,2) = 0.248448;
// ORDER 3
hypercoeff(3,0) = 0.0625128;
hypercoeff(3,1) = 0.108241;
hypercoeff(3,2) = 0.139751;
hypercoeff(3,3) = 0.165365;
hypercoeff(x1,x2) = 0;
g(beam,m) = hypercoeff(beam,m)*norm(m);
gvec(beam,M) = par(m,M+1,par(n,2*m+1,_*g(beam,m)));
process = hgroup("Inputs",par(i,M+1,metermute(i)):yvec((M+1)^2,t,d)<:par(m,M,gvec(m+1,M):>_*vol):crossfade(M,order(step)-1)):hgroup("Output",hmeter); | https://raw.githubusercontent.com/sekisushai/ambitools/2d21b7cc7cfe9bc35d91d51ec05bf9250372f0ce/Faust/src/hoa_beamforming_hypercardioid_to_mono.dsp | faust | Description: This tool helps to extract a monophonic signal from the HOA scene with a beampatern. The Ambisonics components inputs are combined to produce a monophonic output as if the capture was done with a directionnal microphone. The beampatern provided are regular hypercardioid up to order 5 [1].
References:
[1] Meyer, J., & Elko, G. (2002). A highly scalable spherical microphone array based on an orthonormal decomposition of the soundfield. 2002 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2, II–1781–II–1784.
Inputs: (M+1)^2
Outputs: 1
Maximum order of original HOA scene.
Output gain
Order of the beampattern used for filtering, order=0 is a bypass.
linear crossfade between order.
ORDER 1
ORDER 2
ORDER 3 | declare name "HOA-Beamformer to Mono";
declare version "1.0";
declare author "Pierre Lecomte";
declare license "GPL";
declare copyright "(c) Pierre Lecomte 2015";
import("stdfaust.lib");
import("ymn.lib");
import("gui.lib");
M = 3;
vol = vslider("[2]Output Gain", 0, -10, 10, 0.1) : ba.db2linear : si.smooth(0.999);
t = vslider("[3]Azimuth", 0, 0, 360, 0.1)*ma.PI/180;
d = vslider("[4]Elevation", 0, -90, 90, 0.1)*ma.PI/180;
step = checkbox("Int/Float");
norm(m) = 1/sqrt(2*m+1);
hypercoeff(1,0) = 0.24993;
hypercoeff(1,1) = 0.433017;
hypercoeff(2,0) = 0.11112;
hypercoeff(2,1) = 0.19245;
hypercoeff(2,2) = 0.248448;
hypercoeff(3,0) = 0.0625128;
hypercoeff(3,1) = 0.108241;
hypercoeff(3,2) = 0.139751;
hypercoeff(3,3) = 0.165365;
hypercoeff(x1,x2) = 0;
g(beam,m) = hypercoeff(beam,m)*norm(m);
gvec(beam,M) = par(m,M+1,par(n,2*m+1,_*g(beam,m)));
process = hgroup("Inputs",par(i,M+1,metermute(i)):yvec((M+1)^2,t,d)<:par(m,M,gvec(m+1,M):>_*vol):crossfade(M,order(step)-1)):hgroup("Output",hmeter); |
75221fef392a77baa88ab586da44b6c498044f697fa88ba6318ebeab392d957a | ljwall/er-301-units | Andromeda.dsp | import("stdfaust.lib");
declare andromeda author "Liam Wall";
declare andromeda license "MIT-style STK-4.3 license";
andromeda(decay, low_pass, high_pass) = _,_ : + : *(0.5) : diffusion_network : (+~chain) <: chain_l, chain_r with {
// allpass using delay with fixed size
allpass_f(t, a) = (+ <: @(t),*(a)) ~ *(-a) : mem,_ : +;
i_diff1 = 0.75;
i_diff2 = 0.625;
diffusion_network = allpass_f(235, i_diff1) : allpass_f(177, i_diff1) : allpass_f(627, i_diff2) : allpass_f(458, i_diff2);
line = de.fdelayltv(2, 28800);
taps = (0.047, 0.120, 0.134, 0.146, 0.158, 0.169, 0.180, 0.190, 0.200, 0.209, 0.217, 0.233, 0.240, 0.244, 0.225, 0.247);
mod = hslider("mod", 50, 0, 100, 0);
min_mod = 31, 25, 19, 11;
mid_mod = 130, 63, 43, 20;
max_mod = 313, 310, 251, 250;
epsilon = par(i, 4,
(mod <= 50)*(ba.take(i+1, min_mod) + (mod/50) * (ba.take(i+1, mid_mod) - ba.take(i+1, min_mod))) +
(mod > 50)*(ba.take(i+1, mid_mod) + (mod/50 - 1) * (ba.take(i+1, max_mod) - ba.take(i+1, mid_mod)))
);
e1 = ba.take(1, epsilon) / 1000000;
e2 = ba.take(2, epsilon) / 1000000;
e3 = ba.take(3, epsilon) / 1000000;
e4 = ba.take(4, epsilon) / 1000000;
mods = x,xq,-x,-xq , y,yq,-y,-yq, z,zq,-z,-zq, a,aq,-a,-aq letrec {
'xq = os.impulse + xq - e1*x;
'x = e1 * (xq - e1 *x) + x;
'yq = os.impulse + yq - e2*y;
'y = e2 * (yq - e2 *y) + y;
'zq = os.impulse + zq - e3*z;
'z = e3 * (zq - e3 *z) + z;
'aq = os.impulse + aq - e4*a;
'a = e4 * (aq - e4 *a) + a;
};
limiter(x) = 2 * x / sqrt(x*x +4);
depth = ba.sec2samp(0.004);
chain = _
<: par(i, ba.count(taps), line(ba.sec2samp(ba.take(i+1, taps)) + depth*ba.take(1 + (i % (ba.count(mods))), mods)))
:> /(ba.count(taps))
: fi.lowpass(1, low_pass)
: fi.highpass(1, high_pass)
: *(decay)
: limiter;
line_out = de.delay(24000);
taps_l = (0.060, 0.137, 0.175, 0.190);
taps_r = (0.077, 0.112, 0.160, 0.212);
chain_l = _ <: par(i, ba.count(taps_l), line_out(ba.take(i+1, taps_l) : ba.sec2samp)) :> /(ba.count(taps_l));
chain_r = _ <: par(i, ba.count(taps_r), line_out(ba.take(i+1, taps_r) : ba.sec2samp)) :> /(ba.count(taps_r));
};
declare er301_in1 "InL";
declare er301_in2 "InR";
declare er301_out1 "OutL";
declare er301_out2 "OutR";
process = _,_ <: _,_,andromeda(decay_ctrl, low_ctrl, high_ctrl): dry_wet_mix(dry_wet_ctr) with {
decay_ctrl = hslider("Decay", 0.8, 0, 5, 0.001) : si.smoo;
low_ctrl = hslider("HighCut", 20000, 100, 20000, 100) : min(20000) : max(100);
high_ctrl = hslider("LowCut", 20, 20, 20000, 100) : min(20000) : max(20);
dry_wet_ctr = hslider("DryWet", 0.25, 0, 1, 0.001) : si.smoo;
dry_wet_mix(mix, dry_l, dry_r, wet_l, wet_r) = (1-mix) * dry_l, (1-mix) * dry_r, mix * wet_l, mix * wet_r :> _,_;
};
| https://raw.githubusercontent.com/ljwall/er-301-units/4f59543a76e84c823d770b2101c1d6f9b3d1eb74/faustian/dsp/Andromeda.dsp | faust | allpass using delay with fixed size | import("stdfaust.lib");
declare andromeda author "Liam Wall";
declare andromeda license "MIT-style STK-4.3 license";
andromeda(decay, low_pass, high_pass) = _,_ : + : *(0.5) : diffusion_network : (+~chain) <: chain_l, chain_r with {
allpass_f(t, a) = (+ <: @(t),*(a)) ~ *(-a) : mem,_ : +;
i_diff1 = 0.75;
i_diff2 = 0.625;
diffusion_network = allpass_f(235, i_diff1) : allpass_f(177, i_diff1) : allpass_f(627, i_diff2) : allpass_f(458, i_diff2);
line = de.fdelayltv(2, 28800);
taps = (0.047, 0.120, 0.134, 0.146, 0.158, 0.169, 0.180, 0.190, 0.200, 0.209, 0.217, 0.233, 0.240, 0.244, 0.225, 0.247);
mod = hslider("mod", 50, 0, 100, 0);
min_mod = 31, 25, 19, 11;
mid_mod = 130, 63, 43, 20;
max_mod = 313, 310, 251, 250;
epsilon = par(i, 4,
(mod <= 50)*(ba.take(i+1, min_mod) + (mod/50) * (ba.take(i+1, mid_mod) - ba.take(i+1, min_mod))) +
(mod > 50)*(ba.take(i+1, mid_mod) + (mod/50 - 1) * (ba.take(i+1, max_mod) - ba.take(i+1, mid_mod)))
);
e1 = ba.take(1, epsilon) / 1000000;
e2 = ba.take(2, epsilon) / 1000000;
e3 = ba.take(3, epsilon) / 1000000;
e4 = ba.take(4, epsilon) / 1000000;
mods = x,xq,-x,-xq , y,yq,-y,-yq, z,zq,-z,-zq, a,aq,-a,-aq letrec {
'xq = os.impulse + xq - e1*x;
'x = e1 * (xq - e1 *x) + x;
'yq = os.impulse + yq - e2*y;
'y = e2 * (yq - e2 *y) + y;
'zq = os.impulse + zq - e3*z;
'z = e3 * (zq - e3 *z) + z;
'aq = os.impulse + aq - e4*a;
'a = e4 * (aq - e4 *a) + a;
};
limiter(x) = 2 * x / sqrt(x*x +4);
depth = ba.sec2samp(0.004);
chain = _
<: par(i, ba.count(taps), line(ba.sec2samp(ba.take(i+1, taps)) + depth*ba.take(1 + (i % (ba.count(mods))), mods)))
:> /(ba.count(taps))
: fi.lowpass(1, low_pass)
: fi.highpass(1, high_pass)
: *(decay)
: limiter;
line_out = de.delay(24000);
taps_l = (0.060, 0.137, 0.175, 0.190);
taps_r = (0.077, 0.112, 0.160, 0.212);
chain_l = _ <: par(i, ba.count(taps_l), line_out(ba.take(i+1, taps_l) : ba.sec2samp)) :> /(ba.count(taps_l));
chain_r = _ <: par(i, ba.count(taps_r), line_out(ba.take(i+1, taps_r) : ba.sec2samp)) :> /(ba.count(taps_r));
};
declare er301_in1 "InL";
declare er301_in2 "InR";
declare er301_out1 "OutL";
declare er301_out2 "OutR";
process = _,_ <: _,_,andromeda(decay_ctrl, low_ctrl, high_ctrl): dry_wet_mix(dry_wet_ctr) with {
decay_ctrl = hslider("Decay", 0.8, 0, 5, 0.001) : si.smoo;
low_ctrl = hslider("HighCut", 20000, 100, 20000, 100) : min(20000) : max(100);
high_ctrl = hslider("LowCut", 20, 20, 20000, 100) : min(20000) : max(20);
dry_wet_ctr = hslider("DryWet", 0.25, 0, 1, 0.001) : si.smoo;
dry_wet_mix(mix, dry_l, dry_r, wet_l, wet_r) = (1-mix) * dry_l, (1-mix) * dry_r, mix * wet_l, mix * wet_r :> _,_;
};
|
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