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646c4a929abfbbdf34e1e19d55ad66fdbb66e503448998de33b7c7801f1d9a50 | RuolunWeng/ruolunweng.github.io | STunedBar3.dsp | declare name "Three Rack Tuned Bars";
declare author "ER";//From "Tuned Bar" by Romain Michon ([email protected]);
/* =========== DESCRIPTION =============
- Three rack tuned bars
- Head = Silence/Resonance
- Tilt = High frequencies
- Front = High + Medium frequencies
- Bottom = High + Medium + Low frequencies
*/
import("stdfaust.lib");
instrument = library("instruments.lib");
//==================== INSTRUMENT =======================
process = vgroup("tunedBars",hgroup("[1]",par(i, 3, onerack(i,i,i))):>_);
onerack(d,n,e) = hgroup("bar %n", par(i, 5, tunedBar(d,i,e)));
tunedBar(d,n,e) =
((select-1)*-1) <:
//nModes resonances with nModes feedbacks for bow table look-up
par(i,nModes,(resonance(i,freqqy(n,e),gate(d,n))~_)):> + :
//Signal Scaling and stereo
*(4);
//==================== GUI SPECIFICATION ================
gain = 0.8;
gate(d,n) = position(d,n) : upfront;
position(d,n) = abs(hand(d) - n) < 0.5;
upfront(x) = x>x';
hand(0) = vslider("Instrument Hand 0 [acc:1 0 -10 0 14]", 0, 0, 5, 1):int:ba.automat(120, 15, 0.0);
hand(1) = vslider("Instrument Hand 1 [acc:1 0 -10 0 14]", 2, 0, 5, 1):int:ba.automat(240, 15, 0.0);
hand(2) = vslider("Instrument Hand 2 [acc:1 0 -10 0 10]", 4, 0, 5, 1):int:ba.automat(480, 15, 0.0);
select = 1;
//----------------------- Frequency Table --------------------
freq(0) = 184.99;
freq(1) = 207.65;
freq(2) = 233.08;
freq(3) = 277.18;
freq(4) = 311.12;
freq(d) = freq(d-5)*2;
freqqy(d,e) = freq(d+e*5);
//==================== MODAL PARAMETERS ================
preset = 2;
nMode(2) = 4;
modes(2,0) = 1;
basegains(2,0) = pow(0.999,1);
excitation(2,0,g) = 1*gain*g/nMode(2);
modes(2,1) = 4.0198391420;
basegains(2,1) = pow(0.999,2);
excitation(2,1,g) = 1*gain*g/nMode(2);
modes(2,2) = 10.7184986595;
basegains(2,2) = pow(0.999,3);
excitation(2,2,g) = 1*gain*g/nMode(2);
modes(2,3) = 18.0697050938;
basegains(2,3) = pow(0.999,4);
excitation(2,3,g) = 1*gain*g/nMode(2);
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//the number of modes depends on the preset being used
nModes = nMode(preset);
delayLengthBase(f) = ma.SR/f;
//delay lengths in number of samples
delayLength(x,f) = delayLengthBase(f)/modes(preset,x);
//delay lines
delayLine(x,f) = de.delay(4096,delayLength(x,f));
//Filter bank: fi.bandpass filters (declared in instrument.lib)
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x,f) = instrument.bandPass(f*modes(preset,x),radius);
//----------------------- Algorithm implementation ----------------------------
//One resonance
resonance(x,f,g) = + : + (excitation(preset,x,g)*select) : delayLine(x,f) : *(basegains(preset,x)) : bandPassFilter(x,f);
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/STunedBar3.dsp | faust | From "Tuned Bar" by Romain Michon ([email protected]);
=========== DESCRIPTION =============
- Three rack tuned bars
- Head = Silence/Resonance
- Tilt = High frequencies
- Front = High + Medium frequencies
- Bottom = High + Medium + Low frequencies
==================== INSTRUMENT =======================
nModes resonances with nModes feedbacks for bow table look-up
Signal Scaling and stereo
==================== GUI SPECIFICATION ================
----------------------- Frequency Table --------------------
==================== MODAL PARAMETERS ================
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
the number of modes depends on the preset being used
delay lengths in number of samples
delay lines
Filter bank: fi.bandpass filters (declared in instrument.lib)
----------------------- Algorithm implementation ----------------------------
One resonance | declare name "Three Rack Tuned Bars";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("tunedBars",hgroup("[1]",par(i, 3, onerack(i,i,i))):>_);
onerack(d,n,e) = hgroup("bar %n", par(i, 5, tunedBar(d,i,e)));
tunedBar(d,n,e) =
((select-1)*-1) <:
par(i,nModes,(resonance(i,freqqy(n,e),gate(d,n))~_)):> + :
*(4);
gain = 0.8;
gate(d,n) = position(d,n) : upfront;
position(d,n) = abs(hand(d) - n) < 0.5;
upfront(x) = x>x';
hand(0) = vslider("Instrument Hand 0 [acc:1 0 -10 0 14]", 0, 0, 5, 1):int:ba.automat(120, 15, 0.0);
hand(1) = vslider("Instrument Hand 1 [acc:1 0 -10 0 14]", 2, 0, 5, 1):int:ba.automat(240, 15, 0.0);
hand(2) = vslider("Instrument Hand 2 [acc:1 0 -10 0 10]", 4, 0, 5, 1):int:ba.automat(480, 15, 0.0);
select = 1;
freq(0) = 184.99;
freq(1) = 207.65;
freq(2) = 233.08;
freq(3) = 277.18;
freq(4) = 311.12;
freq(d) = freq(d-5)*2;
freqqy(d,e) = freq(d+e*5);
preset = 2;
nMode(2) = 4;
modes(2,0) = 1;
basegains(2,0) = pow(0.999,1);
excitation(2,0,g) = 1*gain*g/nMode(2);
modes(2,1) = 4.0198391420;
basegains(2,1) = pow(0.999,2);
excitation(2,1,g) = 1*gain*g/nMode(2);
modes(2,2) = 10.7184986595;
basegains(2,2) = pow(0.999,3);
excitation(2,2,g) = 1*gain*g/nMode(2);
modes(2,3) = 18.0697050938;
basegains(2,3) = pow(0.999,4);
excitation(2,3,g) = 1*gain*g/nMode(2);
nModes = nMode(preset);
delayLengthBase(f) = ma.SR/f;
delayLength(x,f) = delayLengthBase(f)/modes(preset,x);
delayLine(x,f) = de.delay(4096,delayLength(x,f));
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x,f) = instrument.bandPass(f*modes(preset,x),radius);
resonance(x,f,g) = + : + (excitation(preset,x,g)*select) : delayLine(x,f) : *(basegains(preset,x)) : bandPassFilter(x,f);
|
44c1e73f703349ecd1447528aeafa1a6e3f3dc96d87c2330ceaf66941c4b85db | RuolunWeng/ruolunweng.github.io | SCMajTunedBars.dsp | declare name "C Major Tuned Bars";
declare author "ER";//From "Tuned Bar" by Romain Michon ([email protected]);
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- C Major tuned bars
- Left = Low frequencies + slow rhythm/Silence
- Right = High frequencies + fast rhythm
*/
//==================== INSTRUMENT =======================
process = par(i, N, tunedBar(i)):>_;
tunedBar(n) =
((select-1)*-1) <:
//nModes resonances with nModes feedbacks for bow table look-up
par(i,nModes,(resonance(i,freq(n),gate(n))~_)) :> + :
//Signal Scaling and stereo
*(4);
//==================== GUI SPECIFICATION ================
N = 24;
gain = 0.8;
gate(n) = position(n) : upfront;
hand = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 12, 0, N, 1):si.smooth(0.999):min(N):max(0):int:ba.automat(B, 15, 0.0);
B = hslider("[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 60): si.smooth(0.99) : min(720) : max(180) : int;
position(n) = abs(hand - n) < 0.5;
upfront(x) = x>x';
select = 1;
nMode(2) = 4;
//-------------------- Frequency Table ------------------
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 174.61;
freq(4) = 195.99;
freq(5) = 220.00;
freq(6) = 246.94;
freq(d) = freq(d-7)*2;
//==================== MODAL PARAMETERS ================
preset = 2;
modes(2,0) = 1;
basegains(2,0) = pow(0.999,1);
excitation(2,0,g) = 1*gain*g/nMode(2);
modes(2,1) = 4.0198391420;
basegains(2,1) = pow(0.999,2);
excitation(2,1,g) = 1*gain*g/nMode(2);
modes(2,2) = 10.7184986595;
basegains(2,2) = pow(0.999,3);
excitation(2,2,g) = 1*gain*g/nMode(2);
modes(2,3) = 18.0697050938;
basegains(2,3) = pow(0.999,4);
excitation(2,3,g) = 1*gain*g/nMode(2);
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//the number of modes depends on the preset being used
nModes = nMode(preset);
delayLengthBase(f) = ma.SR/f;
//delay lengths in number of samples
delayLength(x,f) = delayLengthBase(f)/modes(preset,x);
//delay lines
delayLine(x,f) = de.delay(4096,delayLength(x,f));
//Filter bank: fi.bandpass filters (declared in instrument.lib)
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x,f) = instrument.bandPass(f*modes(preset,x),radius);
//----------------------- Algorithm implementation ----------------------------
//One resonance
resonance(x,f,g) = + : + (excitation(preset,x,g)*select) : delayLine(x,f) : *(basegains(preset,x)) : bandPassFilter(x,f);
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SCMajTunedBars.dsp | faust | From "Tuned Bar" by Romain Michon ([email protected]);
=============== DESCRIPTION ================= :
- C Major tuned bars
- Left = Low frequencies + slow rhythm/Silence
- Right = High frequencies + fast rhythm
==================== INSTRUMENT =======================
nModes resonances with nModes feedbacks for bow table look-up
Signal Scaling and stereo
==================== GUI SPECIFICATION ================
-------------------- Frequency Table ------------------
==================== MODAL PARAMETERS ================
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
the number of modes depends on the preset being used
delay lengths in number of samples
delay lines
Filter bank: fi.bandpass filters (declared in instrument.lib)
----------------------- Algorithm implementation ----------------------------
One resonance | declare name "C Major Tuned Bars";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = par(i, N, tunedBar(i)):>_;
tunedBar(n) =
((select-1)*-1) <:
par(i,nModes,(resonance(i,freq(n),gate(n))~_)) :> + :
*(4);
N = 24;
gain = 0.8;
gate(n) = position(n) : upfront;
hand = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 12, 0, N, 1):si.smooth(0.999):min(N):max(0):int:ba.automat(B, 15, 0.0);
B = hslider("[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 60): si.smooth(0.99) : min(720) : max(180) : int;
position(n) = abs(hand - n) < 0.5;
upfront(x) = x>x';
select = 1;
nMode(2) = 4;
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 174.61;
freq(4) = 195.99;
freq(5) = 220.00;
freq(6) = 246.94;
freq(d) = freq(d-7)*2;
preset = 2;
modes(2,0) = 1;
basegains(2,0) = pow(0.999,1);
excitation(2,0,g) = 1*gain*g/nMode(2);
modes(2,1) = 4.0198391420;
basegains(2,1) = pow(0.999,2);
excitation(2,1,g) = 1*gain*g/nMode(2);
modes(2,2) = 10.7184986595;
basegains(2,2) = pow(0.999,3);
excitation(2,2,g) = 1*gain*g/nMode(2);
modes(2,3) = 18.0697050938;
basegains(2,3) = pow(0.999,4);
excitation(2,3,g) = 1*gain*g/nMode(2);
nModes = nMode(preset);
delayLengthBase(f) = ma.SR/f;
delayLength(x,f) = delayLengthBase(f)/modes(preset,x);
delayLine(x,f) = de.delay(4096,delayLength(x,f));
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x,f) = instrument.bandPass(f*modes(preset,x),radius);
resonance(x,f,g) = + : + (excitation(preset,x,g)*select) : delayLine(x,f) : *(basegains(preset,x)) : bandPassFilter(x,f);
|
6905e6c0215e45affb3e09a30164c70f591d3392ffd1d68587862f4c96a55153 | RuolunWeng/ruolunweng.github.io | SChromaticTunedBars.dsp | declare name "Chromatic Tuned Bars";
declare author "ER";//From "Tuned Bar" by Romain Michon ([email protected]);
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- Chromatic tuned bars
- Left = Low frequencies + slow rhythm/Silence
- Right = High frequencies + fast rhythm
*/
//==================== INSTRUMENT =======================
process = par(i, N, tunedBar(i)):> fi.lowpass(1,5000);
tunedBar(n) =
((select-1)*-1) <:
//nModes resonances with nModes feedbacks for bow table look-up
par(i,nModes,(resonance(i,freq(n),gate(n))~_)):> + :
//Signal Scaling and stereo
*(4);
//==================== GUI SPECIFICATION ================
N = 24;
gain = 0.8;
gate(n) = position(n) : upfront;
hand = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 12, 0, N, 1):si.smooth(0.999):min(N):max(0):int:ba.automat(B, 15, 0.0);
B = hslider("[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 60): si.smooth(0.99) : min(720) : max(180) : int;
position(n) = abs(hand - n) < 0.5;
upfront(x) = x>x';
select = 1;
//-------------------- Frequency Table ------------------
freq(0) = 130.81;
freq(1) = 138.59;
freq(2) = 146.83;
freq(3) = 155.56;
freq(4) = 164.81;
freq(5) = 174.61;
freq(6) = 184.99;
freq(7) = 195.99;
freq(8) = 207.65;
freq(9) = 220.00;
freq(10) = 233.08;
freq(11) = 246.94;
//freq(12) = 261.62;
freq(d) = freq(d-12)*2;
//==================== MODAL PARAMETERS ================
preset = 2;
nMode(2) = 4;
modes(2,0) = 1;
basegains(2,0) = pow(0.999,1);
excitation(2,0,g) = 1*gain*g/nMode(2);
modes(2,1) = 4.0198391420;
basegains(2,1) = pow(0.999,2);
excitation(2,1,g) = 1*gain*g/nMode(2);
modes(2,2) = 10.7184986595;
basegains(2,2) = pow(0.999,3);
excitation(2,2,g) = 1*gain*g/nMode(2);
modes(2,3) = 18.0697050938;
basegains(2,3) = pow(0.999,4);
excitation(2,3,g) = 1*gain*g/nMode(2);
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//the number of modes depends on the preset being used
nModes = nMode(preset);
delayLengthBase(f) = ma.SR/f;
//delay lengths in number of samples
delayLength(x,f) = delayLengthBase(f)/modes(preset,x);
//delay lines
delayLine(x,f) = de.delay(4096,delayLength(x,f));
//Filter bank: fi.bandpass filters (declared in instrument.lib)
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x,f) = instrument.bandPass(f*modes(preset,x),radius);
//----------------------- Algorithm implementation ----------------------------
//One resonance
resonance(x,f,g) = + : + (excitation(preset,x,g)*select) : delayLine(x,f) : *(basegains(preset,x)) : bandPassFilter(x,f);
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SChromaticTunedBars.dsp | faust | From "Tuned Bar" by Romain Michon ([email protected]);
=============== DESCRIPTION ================= :
- Chromatic tuned bars
- Left = Low frequencies + slow rhythm/Silence
- Right = High frequencies + fast rhythm
==================== INSTRUMENT =======================
nModes resonances with nModes feedbacks for bow table look-up
Signal Scaling and stereo
==================== GUI SPECIFICATION ================
-------------------- Frequency Table ------------------
freq(12) = 261.62;
==================== MODAL PARAMETERS ================
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
the number of modes depends on the preset being used
delay lengths in number of samples
delay lines
Filter bank: fi.bandpass filters (declared in instrument.lib)
----------------------- Algorithm implementation ----------------------------
One resonance | declare name "Chromatic Tuned Bars";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = par(i, N, tunedBar(i)):> fi.lowpass(1,5000);
tunedBar(n) =
((select-1)*-1) <:
par(i,nModes,(resonance(i,freq(n),gate(n))~_)):> + :
*(4);
N = 24;
gain = 0.8;
gate(n) = position(n) : upfront;
hand = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 12, 0, N, 1):si.smooth(0.999):min(N):max(0):int:ba.automat(B, 15, 0.0);
B = hslider("[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 60): si.smooth(0.99) : min(720) : max(180) : int;
position(n) = abs(hand - n) < 0.5;
upfront(x) = x>x';
select = 1;
freq(0) = 130.81;
freq(1) = 138.59;
freq(2) = 146.83;
freq(3) = 155.56;
freq(4) = 164.81;
freq(5) = 174.61;
freq(6) = 184.99;
freq(7) = 195.99;
freq(8) = 207.65;
freq(9) = 220.00;
freq(10) = 233.08;
freq(11) = 246.94;
freq(d) = freq(d-12)*2;
preset = 2;
nMode(2) = 4;
modes(2,0) = 1;
basegains(2,0) = pow(0.999,1);
excitation(2,0,g) = 1*gain*g/nMode(2);
modes(2,1) = 4.0198391420;
basegains(2,1) = pow(0.999,2);
excitation(2,1,g) = 1*gain*g/nMode(2);
modes(2,2) = 10.7184986595;
basegains(2,2) = pow(0.999,3);
excitation(2,2,g) = 1*gain*g/nMode(2);
modes(2,3) = 18.0697050938;
basegains(2,3) = pow(0.999,4);
excitation(2,3,g) = 1*gain*g/nMode(2);
nModes = nMode(preset);
delayLengthBase(f) = ma.SR/f;
delayLength(x,f) = delayLengthBase(f)/modes(preset,x);
delayLine(x,f) = de.delay(4096,delayLength(x,f));
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x,f) = instrument.bandPass(f*modes(preset,x),radius);
resonance(x,f,g) = + : + (excitation(preset,x,g)*select) : delayLine(x,f) : *(basegains(preset,x)) : bandPassFilter(x,f);
|
e4ddabc6cbab58e12149def6d65e05f0bb20b7fba0f4064d9b85815337d72e58 | RuolunWeng/ruolunweng.github.io | SPentatonicFlute.dsp | declare name "Pentatonic Flute";
declare description "Nonlinear WaveGuide Flute";
declare author "ER";// Adapted from "Flute" by Romain Michon ([email protected]);
/* =============== DESCRIPTION ================= :
- Pentatonic flute
- Rocking = playing all notes from low to high frequencies
- Left = Silence/Slow rhythm
- Right = Fast rhythm
- Front = long notes
- Back = short notes
*/
import("stdfaust.lib");
instrument = library("instruments.lib");
//==================== INSTRUMENT =======================
flute(n) = (_ <: (flow(trigger(n)) + *(feedBack1) : embouchureDelay(freq(n)): poly) + *(feedBack2) : reflexionFilter)~(boreDelay(freq(n))) : *(env2(trigger(n)))*gain:_;
process = par(i, N, flute(i)):>_;
//==================== GUI SPECIFICATION ================
vibratoFreq = 2.5;
env1Attack = 0.06;
env1Release = 1;
//-------------------- Non-Variable Parameters -----------
N = 14;
gain = 1;
pressure = 0.9;
breathAmp = 0.01;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.1;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
//----------------------- Frequency Table --------------------
freq(0) = 184.99;
freq(1) = 207.65;
freq(2) = 233.08;
freq(3) = 277.18;
freq(4) = 311.12;
freq(d) = freq(d-5)*2;
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//Loops feedbacks gains
feedBack1 = 0.4;
feedBack2 = 0.4;
//Delay Lines
embouchureDelayLength(f) = (ma.SR/f)/2-2;
boreDelayLength(f) = ma.SR/f-2;
embouchureDelay(f) = de.fdelay(4096,embouchureDelayLength(f));
boreDelay(f) = de.fdelay(4096,boreDelayLength(f));
//Polinomial
poly = _ <: _ - _*_*_;
//jet filter is a lowwpass filter (declared in filter.lib)
reflexionFilter = fi.lowpass(1,2000);
//----------------------- Algorithm implementation ----------------------------
//Pressure envelope
env1(t) = en.adsr(env1Attack,env1Decay,0.9,env1Release,(t | pressureEnvelope))*pressure*1.1;
//Global envelope
env2(t) = en.asr(env2Attack,1,env2Release,t)*0.5;
//Vibrato Envelope
vibratoEnvelope(t) = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*vibratoGain;
vibrato(t) = os.osc(vibratoFreq)*vibratoEnvelope(t);
breath(t) = no.noise*env1(t);
flow(t) = env1(t) + breath(t)*breathAmp + vibrato(t);
//------------------------- Enveloppe Trigger --------------------------------------------
trigger(n) = position(n): trig
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
noteDuration = hslider("[3]Note Duration[unit:s][style:knob][acc:2 1 -10 0 10]", 0.166, 0.1, 0.25, 0.01)*44100 : min(11025) : max(4410):int;
trig = upfront : release(noteDuration) : >(0.0);
};
position(n) = abs(hand - n) < 0.5;
hand = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 7, 0, N, 1):int: ba.automat(bps, 15, 0.0)// => gate
with {
bps = hslider("[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SPentatonicFlute.dsp | faust | Adapted from "Flute" by Romain Michon ([email protected]);
=============== DESCRIPTION ================= :
- Pentatonic flute
- Rocking = playing all notes from low to high frequencies
- Left = Silence/Slow rhythm
- Right = Fast rhythm
- Front = long notes
- Back = short notes
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
-------------------- Non-Variable Parameters -----------
----------------------- Frequency Table --------------------
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
Loops feedbacks gains
Delay Lines
Polinomial
jet filter is a lowwpass filter (declared in filter.lib)
----------------------- Algorithm implementation ----------------------------
Pressure envelope
Global envelope
Vibrato Envelope
------------------------- Enveloppe Trigger --------------------------------------------
=> gate | declare name "Pentatonic Flute";
declare description "Nonlinear WaveGuide Flute";
import("stdfaust.lib");
instrument = library("instruments.lib");
flute(n) = (_ <: (flow(trigger(n)) + *(feedBack1) : embouchureDelay(freq(n)): poly) + *(feedBack2) : reflexionFilter)~(boreDelay(freq(n))) : *(env2(trigger(n)))*gain:_;
process = par(i, N, flute(i)):>_;
vibratoFreq = 2.5;
env1Attack = 0.06;
env1Release = 1;
N = 14;
gain = 1;
pressure = 0.9;
breathAmp = 0.01;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.1;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
freq(0) = 184.99;
freq(1) = 207.65;
freq(2) = 233.08;
freq(3) = 277.18;
freq(4) = 311.12;
freq(d) = freq(d-5)*2;
feedBack1 = 0.4;
feedBack2 = 0.4;
embouchureDelayLength(f) = (ma.SR/f)/2-2;
boreDelayLength(f) = ma.SR/f-2;
embouchureDelay(f) = de.fdelay(4096,embouchureDelayLength(f));
boreDelay(f) = de.fdelay(4096,boreDelayLength(f));
poly = _ <: _ - _*_*_;
reflexionFilter = fi.lowpass(1,2000);
env1(t) = en.adsr(env1Attack,env1Decay,0.9,env1Release,(t | pressureEnvelope))*pressure*1.1;
env2(t) = en.asr(env2Attack,1,env2Release,t)*0.5;
vibratoEnvelope(t) = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*vibratoGain;
vibrato(t) = os.osc(vibratoFreq)*vibratoEnvelope(t);
breath(t) = no.noise*env1(t);
flow(t) = env1(t) + breath(t)*breathAmp + vibrato(t);
trigger(n) = position(n): trig
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
noteDuration = hslider("[3]Note Duration[unit:s][style:knob][acc:2 1 -10 0 10]", 0.166, 0.1, 0.25, 0.01)*44100 : min(11025) : max(4410):int;
trig = upfront : release(noteDuration) : >(0.0);
};
position(n) = abs(hand - n) < 0.5;
with {
bps = hslider("[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
|
7cd40b90b8d9ab0c71fde62971afd9cd25e0c41a1d12548aa9795cf68d8421ca | RuolunWeng/ruolunweng.github.io | SPentatonicBlowBottle.dsp | declare name "Pentatonic BlowBottle";
declare author "ER";//Adapted from Blow Bottle by Romain Michon ([email protected]);
/* =========== DESCRITPION =============
- Pentatonic Blow Bottles
- Left = Low frequencies/ Silence/ Slow rhythm
- Right = High frequencies/ Fast rhythm
- Front = Long notes
- Back = Short notes
*/
import("stdfaust.lib");
instrument = library("instruments.lib");
//==================== INSTRUMENT =======================
process = vgroup("Blowhistle Bottles", par(i, N, blow(i)) :>*(2));
blow(n)=
//differential pressure
(-(breathPressure(trigger(n))) <:
((+(1))*randPressure((trigger(n))) : +(breathPressure(trigger(n)))) - *(instrument.jetTable),_ : baPaF(n),_)~_: !,_:
//signal scaling
fi.dcblocker*envelopeG(trigger(n))*(0.5)
with {
baPaF(n) = bandPassFilter(freq(n));
};
//==================== GUI SPECIFICATION ================
N = 15;
position(n) = abs(hand - n) < 0.5;
hand = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 8, 0, N, 1) : si.smooth(0.999) : min(24) : max(0) :int: ba.automat(bps, 15, 0.0)
with{
bps = hslider("[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
envelopeAttack = 0.01;
vibratoFreq = 5;
vibratoGain = 0.1;
//--------------------- Non-variable Parameters -------------
gain = 0.5;
noiseGain = 0.5;
pressure = 1.2;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.01;
envelopeDecay = 0.01;
envelopeRelease = 0.05;
//----------------------- Frequency Table --------------------
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 195.99;
freq(4) = 220.00;
freq(d) = freq(d-5)*2;
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//botlle radius
bottleRadius = 0.999;
bandPassFilter(f) = instrument.bandPass(f,bottleRadius);
//----------------------- Algorithm implementation ----------------------------
//global envelope is of type attack - decay - sustain - release
envelopeG(t) = gain*en.adsr(gain*envelopeAttack,envelopeDecay,0.8,envelopeRelease,t);
//pressure envelope is also ADSR
envelope(t) = pressure*en.adsr(gain*0.02,0.01,0.8,gain*0.2,t);
//vibrato
vibrato(t) = os.osc(vibratoFreq)*vibratoGain*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*os.osc(vibratoFreq);
//breat pressure
breathPressure(t) = envelope(t) + vibrato(t);
//breath no.noise
randPressure(t) = noiseGain*no.noise*breathPressure(t) ;
//------------------------- Enveloppe Trigger --------------------------------------------
trigger(n) = position(n): trig
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
noteDuration = hslider("[3]Note Duration[unit:s][style:knob][acc:2 1 -10 0 10]", 0.166, 0.1, 0.2, 0.01)*44100 : min(8820) : max(4410):int;
trig = upfront : release(noteDuration) : >(0.0);
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SPentatonicBlowBottle.dsp | faust | Adapted from Blow Bottle by Romain Michon ([email protected]);
=========== DESCRITPION =============
- Pentatonic Blow Bottles
- Left = Low frequencies/ Silence/ Slow rhythm
- Right = High frequencies/ Fast rhythm
- Front = Long notes
- Back = Short notes
==================== INSTRUMENT =======================
differential pressure
signal scaling
==================== GUI SPECIFICATION ================
--------------------- Non-variable Parameters -------------
----------------------- Frequency Table --------------------
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
botlle radius
----------------------- Algorithm implementation ----------------------------
global envelope is of type attack - decay - sustain - release
pressure envelope is also ADSR
vibrato
breat pressure
breath no.noise
------------------------- Enveloppe Trigger -------------------------------------------- | declare name "Pentatonic BlowBottle";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("Blowhistle Bottles", par(i, N, blow(i)) :>*(2));
blow(n)=
(-(breathPressure(trigger(n))) <:
((+(1))*randPressure((trigger(n))) : +(breathPressure(trigger(n)))) - *(instrument.jetTable),_ : baPaF(n),_)~_: !,_:
fi.dcblocker*envelopeG(trigger(n))*(0.5)
with {
baPaF(n) = bandPassFilter(freq(n));
};
N = 15;
position(n) = abs(hand - n) < 0.5;
hand = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 8, 0, N, 1) : si.smooth(0.999) : min(24) : max(0) :int: ba.automat(bps, 15, 0.0)
with{
bps = hslider("[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
envelopeAttack = 0.01;
vibratoFreq = 5;
vibratoGain = 0.1;
gain = 0.5;
noiseGain = 0.5;
pressure = 1.2;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.01;
envelopeDecay = 0.01;
envelopeRelease = 0.05;
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 195.99;
freq(4) = 220.00;
freq(d) = freq(d-5)*2;
bottleRadius = 0.999;
bandPassFilter(f) = instrument.bandPass(f,bottleRadius);
envelopeG(t) = gain*en.adsr(gain*envelopeAttack,envelopeDecay,0.8,envelopeRelease,t);
envelope(t) = pressure*en.adsr(gain*0.02,0.01,0.8,gain*0.2,t);
vibrato(t) = os.osc(vibratoFreq)*vibratoGain*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*os.osc(vibratoFreq);
breathPressure(t) = envelope(t) + vibrato(t);
randPressure(t) = noiseGain*no.noise*breathPressure(t) ;
trigger(n) = position(n): trig
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
noteDuration = hslider("[3]Note Duration[unit:s][style:knob][acc:2 1 -10 0 10]", 0.166, 0.1, 0.2, 0.01)*44100 : min(8820) : max(4410):int;
trig = upfront : release(noteDuration) : >(0.0);
};
|
0699acbdd4f7c0ab842a146fe509689c158a7d084001aea8dc0cdee1d12a2e53 | RuolunWeng/ruolunweng.github.io | Modulations.dsp | declare name "Modulations";
declare author "ER";
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =========== DESCRIPTION ==============
- Non Linear Filter Modulators applied to a sinewave
- Head = Silence/Revereration/Higher Frequencies
- Bottom = Modulation n°3 = FM/ Lower Frequencies
- Rocking = Modulating Frequency (low to high)
- Front = Modulation n°2
- Left = Modulation n°0 & n°1
- Upward = swing from head/bottom/head (a bit like tennis racket) = interesting
*/
//======================== INSTRUMENT =============================
process = vgroup("Modulations",oscil <: seq(i, 3, NLFM(i)), NLFM3 :> fi.lowpass(1,2000) *(0.6) *(vol) <: instrReverbMod:*(vool),*(vool));
NLFM(n) = _ : instrument.nonLinearModulator((nonlinearity:si.smooth(0.999)),env(n),freq,typeMod(n),freqMod,nlfOrder) : _;
NLFM3 = _ : instrument.nonLinearModulator((nonlinearity:si.smooth(0.999)),env(3),freq,typeMod(3),freqMod,nlfOrder) : _;
oscil = os.osci(freq);
//======================== GUI SPECIFICATIONS =====================
freq = hslider("h:Instrument/ Frequency [unit:Hz][acc:1 1 -10 0 15]", 330, 100, 1200, 0.1):si.smooth(0.999);
freqMod = hslider("h:Instrument/Modulating Frequency[style:knob][unit:Hz][acc:0 0 -10 0 10]", 1200, 900, 1700, 0.1):si.smooth(0.999);
vol = (hslider("h:Instrument/ Oscillator Volume[style:knob][acc:1 0 -10 0 10]", 0.5, 0, 1, 0.01)^2):si.smooth(0.999);
vool = hslider("h:Instrument/ General Volume[style:knob][acc:1 1 -10 0 10]", 1, 0.75, 4, 0.01):si.smooth(0.999):min(4):max(0.75);
gate(0) = hslider("v:Modulations/Play Modulation 0 (ASR Envelope)[tooltip:noteOn = 1, noteOff = 0][acc:0 0 -30 0 10]", 0,0,1,1);
gate(1) = hslider("v:Modulations/Play Modulation 1 (ASR Envelope)[tooltip:noteOn = 1, noteOff = 0][acc:0 0 -30 0 5]", 0,0,1,1);
gate(2) = hslider("v:Modulations/Play Modulation 2 (ASR Envelope)[tooltip:noteOn = 1, noteOff = 0][acc:2 1 -30 0 10]", 0,0,1,1);
gate(3) = hslider("v:Modulations/Play Modulation 3 (ASR Envelope)[tooltip:noteOn = 1, noteOff = 0][acc:1 0 -10 0 10]", 0,0,1,1);
//------------------------ NLFM PARAMETERS ------------------------
nlfOrder = 6;
nonlinearity = 0.8;
typeMod(n) = n;
env(n) = ASR(n);
ASR(n) = en.asr(a,s,r,t(n));
a = 3;
s = 1;
r = 2;
t(n) = gate(n);
//----------------------- INSTRREVERB -------------------------------
instrReverbMod = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("v:Reverb/Reverberation Volume(InstrReverb)[acc:1 1 -10 0 10]",0.25,0.05,1,0.01) : si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("v:Reverb/Reverberation Room Size(InstrReverb)[acc:1 1 -10 0 10]", 0.5,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/Modulations.dsp | faust | =========== DESCRIPTION ==============
- Non Linear Filter Modulators applied to a sinewave
- Head = Silence/Revereration/Higher Frequencies
- Bottom = Modulation n°3 = FM/ Lower Frequencies
- Rocking = Modulating Frequency (low to high)
- Front = Modulation n°2
- Left = Modulation n°0 & n°1
- Upward = swing from head/bottom/head (a bit like tennis racket) = interesting
======================== INSTRUMENT =============================
======================== GUI SPECIFICATIONS =====================
------------------------ NLFM PARAMETERS ------------------------
----------------------- INSTRREVERB ------------------------------- | declare name "Modulations";
declare author "ER";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("Modulations",oscil <: seq(i, 3, NLFM(i)), NLFM3 :> fi.lowpass(1,2000) *(0.6) *(vol) <: instrReverbMod:*(vool),*(vool));
NLFM(n) = _ : instrument.nonLinearModulator((nonlinearity:si.smooth(0.999)),env(n),freq,typeMod(n),freqMod,nlfOrder) : _;
NLFM3 = _ : instrument.nonLinearModulator((nonlinearity:si.smooth(0.999)),env(3),freq,typeMod(3),freqMod,nlfOrder) : _;
oscil = os.osci(freq);
freq = hslider("h:Instrument/ Frequency [unit:Hz][acc:1 1 -10 0 15]", 330, 100, 1200, 0.1):si.smooth(0.999);
freqMod = hslider("h:Instrument/Modulating Frequency[style:knob][unit:Hz][acc:0 0 -10 0 10]", 1200, 900, 1700, 0.1):si.smooth(0.999);
vol = (hslider("h:Instrument/ Oscillator Volume[style:knob][acc:1 0 -10 0 10]", 0.5, 0, 1, 0.01)^2):si.smooth(0.999);
vool = hslider("h:Instrument/ General Volume[style:knob][acc:1 1 -10 0 10]", 1, 0.75, 4, 0.01):si.smooth(0.999):min(4):max(0.75);
gate(0) = hslider("v:Modulations/Play Modulation 0 (ASR Envelope)[tooltip:noteOn = 1, noteOff = 0][acc:0 0 -30 0 10]", 0,0,1,1);
gate(1) = hslider("v:Modulations/Play Modulation 1 (ASR Envelope)[tooltip:noteOn = 1, noteOff = 0][acc:0 0 -30 0 5]", 0,0,1,1);
gate(2) = hslider("v:Modulations/Play Modulation 2 (ASR Envelope)[tooltip:noteOn = 1, noteOff = 0][acc:2 1 -30 0 10]", 0,0,1,1);
gate(3) = hslider("v:Modulations/Play Modulation 3 (ASR Envelope)[tooltip:noteOn = 1, noteOff = 0][acc:1 0 -10 0 10]", 0,0,1,1);
nlfOrder = 6;
nonlinearity = 0.8;
typeMod(n) = n;
env(n) = ASR(n);
ASR(n) = en.asr(a,s,r,t(n));
a = 3;
s = 1;
r = 2;
t(n) = gate(n);
instrReverbMod = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("v:Reverb/Reverberation Volume(InstrReverb)[acc:1 1 -10 0 10]",0.25,0.05,1,0.01) : si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("v:Reverb/Reverberation Room Size(InstrReverb)[acc:1 1 -10 0 10]", 0.5,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
4263ce312c201946d4d65122df5a46cfc14868be3e558aa6eff2280681c43339 | RuolunWeng/ruolunweng.github.io | SPentatonicDryHarp.dsp | declare name "PentatonicDryHarp";
declare author "ER";//Adapted from Harpe by Yann Orlarey;
//Modification Grame July 2015
/* =============== DESCRIPTION ================= :
- Pentatonic dry harp
- Left = Lower frequencies/Silence when still
- Front = Resonance (longer notes)
- Back = No Resonance (dry notes)
- Right = Higher frequencies/Fast rhythm
- Rocking = plucking all strings one by one
*/
//-----------------------------------------------
// Harpe : simple string instrument
// (based on Karplus-Strong)
//
//-----------------------------------------------
import("stdfaust.lib");
instrument = library("instruments.lib");
KEY = 60; // basic midi key
NCY = 15; // note cycle length
CCY = 15; // control cycle length
BPS = 360; // general tempo (ba.beat per sec)
//-------------------------------Harpe----------------------------------
// Harpe is a simple string instrument. Move the "hand" to play the
// various strings
//-----------------------------------------------------------------------
process = hgroup("harp", h : harpe(C,N,K) :> *(l),*(l))
with {
N = 21; // number of strings
K = 48; // Midi key of first string
h = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 11, 0, N, 1) : int: ba.automat(bps, 15, 0.0)
with {
bps = hslider("h:[2]Parameters/[1]Speed[style:knob][acc:0 1 -12 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
l = 0.9;
C = 0.5;
};
//----------------------------------Harpe--------------------------------
// USAGE: hand : harpe(C,10,60) : _,_;
// C is the filter coefficient 0..1
// Build a N (10) strings harpe using a pentatonic scale
// based on midi key b (60)
// Each string is triggered by a specific
// position of the "hand"
//-----------------------------------------------------------------------
harpe(C,N,b) = _ <: par(i, N, position(i+1)
: string(C,Penta(b).degree2Hz(i), att, lvl)
: pan((i+0.5)/N) )
:> _,_
with {
att = hslider("h:[2]Parameters/[2]Resonance[style:knob][acc:2 1 -12 0 10]", 5, 0.1, 10, 0.01):min(10):max(0.1);
lvl = 1;
pan(p) = _ <: *(sqrt(1-p)), *(sqrt(p));
position(a,x) = abs(x - a) < 0.5;
};
//----------------------------------Penta-------------------------------
// Pentatonic scale with degree to midi and degree to Hz conversion
// USAGE: Penta(60).degree2midi(3) ==> 67 midikey
// Penta(60).degree2Hz(4) ==> 440 Hz
//-----------------------------------------------------------------------
Penta(key) = environment {
A4Hz = 440;
degree2midi(0) = key+0;
degree2midi(1) = key+2;
degree2midi(2) = key+4;
degree2midi(3) = key+7;
degree2midi(4) = key+9;
degree2midi(d) = degree2midi(d-5)+12;
degree2Hz(d) = A4Hz*semiton(degree2midi(d)-69) with { semiton(n) = 2.0^(n/12.0); };
};
//----------------------------------String-------------------------------
// A karplus-strong string.
//
// USAGE: string(440Hz, 4s, 1.0, button("play"))
// or button("play") : string(440Hz, 4s, 1.0)
//-----------------------------------------------------------------------
string(coef, freq, t60, level, trig) = no.noise*level
: *(trig : trigger(freq2samples(freq)))
: resonator(freq2samples(freq), att)
with {
resonator(d,a) = (+ : @(d-1)) ~ (average : *(a));
average(x) = (x*(1+coef)+x'*(1-coef))/2;
trigger(n) = upfront : + ~ decay(n) : >(0.0);
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
freq2samples(f) = 44100.0/f;
att = pow(0.001,1.0/(freq*t60)); // attenuation coefficient
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SPentatonicDryHarp.dsp | faust | Adapted from Harpe by Yann Orlarey;
Modification Grame July 2015
=============== DESCRIPTION ================= :
- Pentatonic dry harp
- Left = Lower frequencies/Silence when still
- Front = Resonance (longer notes)
- Back = No Resonance (dry notes)
- Right = Higher frequencies/Fast rhythm
- Rocking = plucking all strings one by one
-----------------------------------------------
Harpe : simple string instrument
(based on Karplus-Strong)
-----------------------------------------------
basic midi key
note cycle length
control cycle length
general tempo (ba.beat per sec)
-------------------------------Harpe----------------------------------
Harpe is a simple string instrument. Move the "hand" to play the
various strings
-----------------------------------------------------------------------
number of strings
Midi key of first string
----------------------------------Harpe--------------------------------
USAGE: hand : harpe(C,10,60) : _,_;
C is the filter coefficient 0..1
Build a N (10) strings harpe using a pentatonic scale
based on midi key b (60)
Each string is triggered by a specific
position of the "hand"
-----------------------------------------------------------------------
----------------------------------Penta-------------------------------
Pentatonic scale with degree to midi and degree to Hz conversion
USAGE: Penta(60).degree2midi(3) ==> 67 midikey
Penta(60).degree2Hz(4) ==> 440 Hz
-----------------------------------------------------------------------
----------------------------------String-------------------------------
A karplus-strong string.
USAGE: string(440Hz, 4s, 1.0, button("play"))
or button("play") : string(440Hz, 4s, 1.0)
-----------------------------------------------------------------------
attenuation coefficient | declare name "PentatonicDryHarp";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = hgroup("harp", h : harpe(C,N,K) :> *(l),*(l))
with {
h = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 11, 0, N, 1) : int: ba.automat(bps, 15, 0.0)
with {
bps = hslider("h:[2]Parameters/[1]Speed[style:knob][acc:0 1 -12 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
l = 0.9;
C = 0.5;
};
harpe(C,N,b) = _ <: par(i, N, position(i+1)
: string(C,Penta(b).degree2Hz(i), att, lvl)
: pan((i+0.5)/N) )
:> _,_
with {
att = hslider("h:[2]Parameters/[2]Resonance[style:knob][acc:2 1 -12 0 10]", 5, 0.1, 10, 0.01):min(10):max(0.1);
lvl = 1;
pan(p) = _ <: *(sqrt(1-p)), *(sqrt(p));
position(a,x) = abs(x - a) < 0.5;
};
Penta(key) = environment {
A4Hz = 440;
degree2midi(0) = key+0;
degree2midi(1) = key+2;
degree2midi(2) = key+4;
degree2midi(3) = key+7;
degree2midi(4) = key+9;
degree2midi(d) = degree2midi(d-5)+12;
degree2Hz(d) = A4Hz*semiton(degree2midi(d)-69) with { semiton(n) = 2.0^(n/12.0); };
};
string(coef, freq, t60, level, trig) = no.noise*level
: *(trig : trigger(freq2samples(freq)))
: resonator(freq2samples(freq), att)
with {
resonator(d,a) = (+ : @(d-1)) ~ (average : *(a));
average(x) = (x*(1+coef)+x'*(1-coef))/2;
trigger(n) = upfront : + ~ decay(n) : >(0.0);
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
freq2samples(f) = 44100.0/f;
};
|
4fa3eb339e3d2c60ab0de41ace8695efc4ace6bff2d67d86145be289ed8d4e05 | RuolunWeng/ruolunweng.github.io | SCMajFlute.dsp | declare name "C Major Flute";
declare author "ER";// Adapted from "Nonlinear WaveGuide Flute" by Romain Michon ([email protected])";
import("stdfaust.lib");
instrument=library("instruments.lib");
/* =============== DESCRIPTION ================= :
- C Major flute
- Rocking = playing all notes from low to high frequencies
- Left = Silence/Slow rhythm
- Right = Fast rhythm
- Front = long notes
- Back = short notes
*/
//==================== INSTRUMENT =======================
flute(n) = (_ <: (flow(trigger(n)) + *(feedBack1) : embouchureDelay(freq(n)): poly) + *(feedBack2) : reflexionFilter)~(boreDelay(freq(n))) : *(env2(trigger(n)))*gain:_;
process = vgroup("C Maj Flute", par(i, N, flute(i)):>_);
//==================== GUI SPECIFICATION ================
vibratoFreq = 2.5;
env1Attack = 0.06;
env1Release = 1;
//-------------------- Non-Variable Parameters -----------
N = 17;
gain = 1;
pressure = 0.9;
breathAmp = 0.01;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.1;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
//----------------------- Frequency Table --------------------
freq(0) = 261.62;
freq(1) = 293.66;
freq(2) = 329.62;
freq(3) = 349.22;
freq(4) = 391.99;
freq(5) = 440.00;
freq(6) = 493.88;
freq(d) = freq(d-7)*2;
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//Loops feedbacks gains
feedBack1 = 0.4;
feedBack2 = 0.4;
//Delay Lines
embouchureDelayLength(f) = (ma.SR/f)/2-2;
boreDelayLength(f) = ma.SR/f-2;
embouchureDelay(f) = de.fdelay(4096,embouchureDelayLength(f));
boreDelay(f) = de.fdelay(4096,boreDelayLength(f));
//Polinomial
poly = _ <: _ - _*_*_;
//jet filter is a lowwpass filter (declared in filter.lib)
reflexionFilter = fi.lowpass(1,2000);
//----------------------- Algorithm implementation ----------------------------
//Pressure envelope
env1(t) = en.adsr(env1Attack,env1Decay,0.9,env1Release,(t | pressureEnvelope))*pressure*1.1;
//Global envelope
env2(t) = en.asr(env2Attack,1,env2Release,t)*0.5;
//Vibrato Envelope
vibratoEnvelope(t) = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*vibratoGain;
vibrato(t) = os.osc(vibratoFreq)*vibratoEnvelope(t);
breath(t) = no.noise*env1(t);
flow(t) = env1(t) + breath(t)*breathAmp + vibrato(t);
//------------------------- Enveloppe Trigger --------------------------------------------
trigger(n) = position(n): trig
with{
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
noteDuration = hslider("h:[1]/[3]Note Duration[unit:s][style:knob][acc:2 1 -10 0 10]", 0.166, 0.1, 0.25, 0.01)*44100 : min(11025) : max(4410):int;
trig = upfront : release(noteDuration) : >(0.0);
};
position(n) = abs(hand - n) < 0.5;
hand = hslider("h:[1]/[1]Instrument Hand[acc:0 1 -12 0 10]", 9, 0, N, 1): ba.automat(bps, 15, 0.0)// => gate
with{
bps = hslider("h:[1]/[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SCMajFlute.dsp | faust | Adapted from "Nonlinear WaveGuide Flute" by Romain Michon ([email protected])";
=============== DESCRIPTION ================= :
- C Major flute
- Rocking = playing all notes from low to high frequencies
- Left = Silence/Slow rhythm
- Right = Fast rhythm
- Front = long notes
- Back = short notes
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
-------------------- Non-Variable Parameters -----------
----------------------- Frequency Table --------------------
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
Loops feedbacks gains
Delay Lines
Polinomial
jet filter is a lowwpass filter (declared in filter.lib)
----------------------- Algorithm implementation ----------------------------
Pressure envelope
Global envelope
Vibrato Envelope
------------------------- Enveloppe Trigger --------------------------------------------
=> gate | declare name "C Major Flute";
import("stdfaust.lib");
instrument=library("instruments.lib");
flute(n) = (_ <: (flow(trigger(n)) + *(feedBack1) : embouchureDelay(freq(n)): poly) + *(feedBack2) : reflexionFilter)~(boreDelay(freq(n))) : *(env2(trigger(n)))*gain:_;
process = vgroup("C Maj Flute", par(i, N, flute(i)):>_);
vibratoFreq = 2.5;
env1Attack = 0.06;
env1Release = 1;
N = 17;
gain = 1;
pressure = 0.9;
breathAmp = 0.01;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.1;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
freq(0) = 261.62;
freq(1) = 293.66;
freq(2) = 329.62;
freq(3) = 349.22;
freq(4) = 391.99;
freq(5) = 440.00;
freq(6) = 493.88;
freq(d) = freq(d-7)*2;
feedBack1 = 0.4;
feedBack2 = 0.4;
embouchureDelayLength(f) = (ma.SR/f)/2-2;
boreDelayLength(f) = ma.SR/f-2;
embouchureDelay(f) = de.fdelay(4096,embouchureDelayLength(f));
boreDelay(f) = de.fdelay(4096,boreDelayLength(f));
poly = _ <: _ - _*_*_;
reflexionFilter = fi.lowpass(1,2000);
env1(t) = en.adsr(env1Attack,env1Decay,0.9,env1Release,(t | pressureEnvelope))*pressure*1.1;
env2(t) = en.asr(env2Attack,1,env2Release,t)*0.5;
vibratoEnvelope(t) = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*vibratoGain;
vibrato(t) = os.osc(vibratoFreq)*vibratoEnvelope(t);
breath(t) = no.noise*env1(t);
flow(t) = env1(t) + breath(t)*breathAmp + vibrato(t);
trigger(n) = position(n): trig
with{
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
noteDuration = hslider("h:[1]/[3]Note Duration[unit:s][style:knob][acc:2 1 -10 0 10]", 0.166, 0.1, 0.25, 0.01)*44100 : min(11025) : max(4410):int;
trig = upfront : release(noteDuration) : >(0.0);
};
position(n) = abs(hand - n) < 0.5;
with{
bps = hslider("h:[1]/[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
|
3c7bb27724dbe54efe71322fbf78006aae29dcd0b5d3e844d80deb58f193bb01 | RuolunWeng/ruolunweng.github.io | STinkle.dsp | declare name "Tinkle";
declare description "Banded Waveguide Modeld Tibetan Bowl";
declare author "Romain Michon";
declare copyright "Romain Michon ([email protected])";
declare version "1.0";
declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
declare description "This instrument uses banded waveguide. For more information, see Essl, G. and Cook, P. Banded Waveguides: Towards Physical Modelling of Bar Percussion Instruments, Proceedings of the 1999 International Computer Music Conference.";
/* ============ DESCRIPTION ==============
- Tinkling bowls
- Rocking = Ringing all bowls from low to high frequencies
*/
import("stdfaust.lib");
instrument = library("instruments.lib");
//==================== INSTRUMENT =======================
process = hgroup("Tinkle",(((select-1)*-1) <:
//nModes resonances with nModes feedbacks for bow table look-up
par(i,nModes,(resonance(i)~_))):>+:fi.lowpass(1,5000)*(gain));
//==================== GUI SPECIFICATION ================
freq = hslider("[1]Frequency[unit:Hz][acc:0 1 -10 0 10]", 440,180,780,1);
gain = 0.7;
gate = 0;
select = hslider("[2]Play[style:knob][tooltip:0=Bow; 1=Strike][acc:1 0 -10 0 10]", 1,0,1,1);
integrationConstant = 0.01;
baseGain = 0.5;
//==================== MODAL PARAMETERS ================
preset = 0;
nMode(0) = 12;
modes(0,0) = 0.996108344;
basegains(0,0) = 0.999925960128219;
excitation(0,0) = 11.900357 / 10;
modes(0,1) = 1.0038916562;
basegains(0,1) = 0.999925960128219;
excitation(0,1) = 11.900357 / 10;
modes(0,2) = 2.979178;
basegains(0,2) = 0.999982774366897;
excitation(0,2) = 10.914886 / 10;
modes(0,3) = 2.99329767;
basegains(0,3) = 0.999982774366897;
excitation(0,3) = 10.914886 / 10;
modes(0,4) = 5.704452;
basegains(0,4) = 1.0;
excitation(0,4) = 42.995041 / 10;
modes(0,5) = 5.704452;
basegains(0,5) = 1.0;
excitation(0,5) = 42.995041 / 10;
modes(0,6) = 8.9982;
basegains(0,6) = 1.0;
excitation(0,6) = 40.063034 / 10;
modes(0,7) = 9.01549726;
basegains(0,7) = 1.0;
excitation(0,7) = 40.063034 / 10;
modes(0,8) = 12.83303;
basegains(0,8) = 0.999965497558225;
excitation(0,8) = 7.063034 / 10;
modes(0,9) = 12.807382;
basegains(0,9) = 0.999965497558225;
excitation(0,9) = 7.063034 / 10;
modes(0,10) = 17.2808219;
basegains(0,10) = 0.9999999999999999999965497558225;
excitation(0,10) = 57.063034 / 10;
modes(0,11) = 21.97602739726;
basegains(0,11) = 0.999999999999999965497558225;
excitation(0,11) = 57.063034 / 10;
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//the number of modes depends on the preset being used
nModes = nMode(preset);
delayLengthBase = ma.SR/freq;
//delay lengths in number of samples
delayLength(x) = delayLengthBase/modes(preset,x);
//delay lines
delayLine(x) = de.delay(4096,delayLength(x));
//Filter bank: fi.bandpass filters (declared in instrument.lib)
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x) = instrument.bandPass(freq*modes(preset,x),radius);
//----------------------- Algorithm implementation ----------------------------
//One resonance
resonance(x) = + : + (excitation(preset,x)*select) : delayLine(x) : *(basegains(preset,x)) : bandPassFilter(x);
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/STinkle.dsp | faust | Synthesis Tool Kit 4.3 (MIT style license);
============ DESCRIPTION ==============
- Tinkling bowls
- Rocking = Ringing all bowls from low to high frequencies
==================== INSTRUMENT =======================
nModes resonances with nModes feedbacks for bow table look-up
==================== GUI SPECIFICATION ================
==================== MODAL PARAMETERS ================
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
the number of modes depends on the preset being used
delay lengths in number of samples
delay lines
Filter bank: fi.bandpass filters (declared in instrument.lib)
----------------------- Algorithm implementation ----------------------------
One resonance | declare name "Tinkle";
declare description "Banded Waveguide Modeld Tibetan Bowl";
declare author "Romain Michon";
declare copyright "Romain Michon ([email protected])";
declare version "1.0";
declare description "This instrument uses banded waveguide. For more information, see Essl, G. and Cook, P. Banded Waveguides: Towards Physical Modelling of Bar Percussion Instruments, Proceedings of the 1999 International Computer Music Conference.";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = hgroup("Tinkle",(((select-1)*-1) <:
par(i,nModes,(resonance(i)~_))):>+:fi.lowpass(1,5000)*(gain));
freq = hslider("[1]Frequency[unit:Hz][acc:0 1 -10 0 10]", 440,180,780,1);
gain = 0.7;
gate = 0;
select = hslider("[2]Play[style:knob][tooltip:0=Bow; 1=Strike][acc:1 0 -10 0 10]", 1,0,1,1);
integrationConstant = 0.01;
baseGain = 0.5;
preset = 0;
nMode(0) = 12;
modes(0,0) = 0.996108344;
basegains(0,0) = 0.999925960128219;
excitation(0,0) = 11.900357 / 10;
modes(0,1) = 1.0038916562;
basegains(0,1) = 0.999925960128219;
excitation(0,1) = 11.900357 / 10;
modes(0,2) = 2.979178;
basegains(0,2) = 0.999982774366897;
excitation(0,2) = 10.914886 / 10;
modes(0,3) = 2.99329767;
basegains(0,3) = 0.999982774366897;
excitation(0,3) = 10.914886 / 10;
modes(0,4) = 5.704452;
basegains(0,4) = 1.0;
excitation(0,4) = 42.995041 / 10;
modes(0,5) = 5.704452;
basegains(0,5) = 1.0;
excitation(0,5) = 42.995041 / 10;
modes(0,6) = 8.9982;
basegains(0,6) = 1.0;
excitation(0,6) = 40.063034 / 10;
modes(0,7) = 9.01549726;
basegains(0,7) = 1.0;
excitation(0,7) = 40.063034 / 10;
modes(0,8) = 12.83303;
basegains(0,8) = 0.999965497558225;
excitation(0,8) = 7.063034 / 10;
modes(0,9) = 12.807382;
basegains(0,9) = 0.999965497558225;
excitation(0,9) = 7.063034 / 10;
modes(0,10) = 17.2808219;
basegains(0,10) = 0.9999999999999999999965497558225;
excitation(0,10) = 57.063034 / 10;
modes(0,11) = 21.97602739726;
basegains(0,11) = 0.999999999999999965497558225;
excitation(0,11) = 57.063034 / 10;
nModes = nMode(preset);
delayLengthBase = ma.SR/freq;
delayLength(x) = delayLengthBase/modes(preset,x);
delayLine(x) = de.delay(4096,delayLength(x));
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x) = instrument.bandPass(freq*modes(preset,x),radius);
resonance(x) = + : + (excitation(preset,x)*select) : delayLine(x) : *(basegains(preset,x)) : bandPassFilter(x);
|
f59d1b5b3f9d328e4a2cdbbd23ce10c0c3984753f0089ceb4420fce6077bebb3 | RuolunWeng/ruolunweng.github.io | SCMajBlowBottle.dsp | declare name "C Maj BlowBottle";
declare author "ER";//Adapted from Blow Bottle by Romain Michon ([email protected]);
/* =========== DESCRITPION =============
- C Major Blow Bottles
- Left = Low frequencies/ Silence/ Slow rhythm
- Right = High frequencies/ Fast rhythm
- Front = Long notes
- Back = Short notes
*/
import("stdfaust.lib");
instrument = library("instruments.lib");
//==================== INSTRUMENT =======================
process = vgroup("Blowhistle Bottles", par(i, N, blow(i)) :>*(2));
blow(n)=
//differential pressure
(-(breathPressure(trigger(n))) <:
((+(1))*randPressure((trigger(n))) : +(breathPressure(trigger(n)))) - *(instrument.jetTable),_ : baPaF(n),_)~_: !,_:
//signal scaling
fi.dcblocker*envelopeG(trigger(n))*(0.5)
with{
baPaF(n) = bandPassFilter(freq(n));
};
//==================== GUI SPECIFICATION ================
N = 16;
position(n) = abs(hand - n) < 0.5;
hand = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 12, 0, N, 1) : si.smooth(0.999) : min(24) : max(0) :int: ba.automat(bps, 15, 0.0)
with{
bps = hslider("[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
envelopeAttack = 0.01;
vibratoFreq = 5;
vibratoGain = 0.1;
//--------------------- Non-variable Parameters -------------
gain = 0.5;
noiseGain = 0.5;
pressure = 1.2;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.01;
envelopeDecay = 0.01;
envelopeRelease = 0.05;
//----------------------- Frequency Table --------------------
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 174.61;
freq(4) = 195.99;
freq(5) = 220.00;
freq(6) = 246.94;
freq(d) = freq(d-7)*2;
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//botlle radius
bottleRadius = 0.999;
bandPassFilter(f) = instrument.bandPass(f,bottleRadius);
//----------------------- Algorithm implementation ----------------------------
//global envelope is of type attack - decay - sustain - release
envelopeG(t) = gain*en.adsr(gain*envelopeAttack,envelopeDecay,0.8,envelopeRelease,t);
//pressure envelope is also ADSR
envelope(t) = pressure*en.adsr(gain*0.02,0.01,0.8,gain*0.2,t);
//vibrato
vibrato(t) = os.osc(vibratoFreq)*vibratoGain*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*os.osc(vibratoFreq);
//breat pressure
breathPressure(t) = envelope(t) + vibrato(t);
//breath no.noise
randPressure(t) = noiseGain*no.noise*breathPressure(t) ;
//------------------------- Enveloppe Trigger --------------------------------------------
trigger(n) = position(n): trig
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
noteDuration = hslider("[3]Note Duration[unit:s][style:knob][acc:2 1 -10 0 10]", 0.166, 0.1, 0.2, 0.01)*44100 : min(8820) : max(4410):int;
trig = upfront : release(noteDuration) : >(0.0);
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SCMajBlowBottle.dsp | faust | Adapted from Blow Bottle by Romain Michon ([email protected]);
=========== DESCRITPION =============
- C Major Blow Bottles
- Left = Low frequencies/ Silence/ Slow rhythm
- Right = High frequencies/ Fast rhythm
- Front = Long notes
- Back = Short notes
==================== INSTRUMENT =======================
differential pressure
signal scaling
==================== GUI SPECIFICATION ================
--------------------- Non-variable Parameters -------------
----------------------- Frequency Table --------------------
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
botlle radius
----------------------- Algorithm implementation ----------------------------
global envelope is of type attack - decay - sustain - release
pressure envelope is also ADSR
vibrato
breat pressure
breath no.noise
------------------------- Enveloppe Trigger -------------------------------------------- | declare name "C Maj BlowBottle";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("Blowhistle Bottles", par(i, N, blow(i)) :>*(2));
blow(n)=
(-(breathPressure(trigger(n))) <:
((+(1))*randPressure((trigger(n))) : +(breathPressure(trigger(n)))) - *(instrument.jetTable),_ : baPaF(n),_)~_: !,_:
fi.dcblocker*envelopeG(trigger(n))*(0.5)
with{
baPaF(n) = bandPassFilter(freq(n));
};
N = 16;
position(n) = abs(hand - n) < 0.5;
hand = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 12, 0, N, 1) : si.smooth(0.999) : min(24) : max(0) :int: ba.automat(bps, 15, 0.0)
with{
bps = hslider("[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
envelopeAttack = 0.01;
vibratoFreq = 5;
vibratoGain = 0.1;
gain = 0.5;
noiseGain = 0.5;
pressure = 1.2;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.01;
envelopeDecay = 0.01;
envelopeRelease = 0.05;
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 174.61;
freq(4) = 195.99;
freq(5) = 220.00;
freq(6) = 246.94;
freq(d) = freq(d-7)*2;
bottleRadius = 0.999;
bandPassFilter(f) = instrument.bandPass(f,bottleRadius);
envelopeG(t) = gain*en.adsr(gain*envelopeAttack,envelopeDecay,0.8,envelopeRelease,t);
envelope(t) = pressure*en.adsr(gain*0.02,0.01,0.8,gain*0.2,t);
vibrato(t) = os.osc(vibratoFreq)*vibratoGain*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*os.osc(vibratoFreq);
breathPressure(t) = envelope(t) + vibrato(t);
randPressure(t) = noiseGain*no.noise*breathPressure(t) ;
trigger(n) = position(n): trig
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
noteDuration = hslider("[3]Note Duration[unit:s][style:knob][acc:2 1 -10 0 10]", 0.166, 0.1, 0.2, 0.01)*44100 : min(8820) : max(4410):int;
trig = upfront : release(noteDuration) : >(0.0);
};
|
f069ba9b8b45689cee759d2e1ae7af0141aa3099c73f9a4a485ee521adaf1842 | RuolunWeng/ruolunweng.github.io | Kisana.dsp | declare name "Kisana";
declare author "Yann Orlarey";
//Modifications GRAME July 2015
/* ========= DESCRITPION =============
- Kisana : 3-loops string instrument (based on Karplus-Strong)
- Head = Silence
- Tilt = High frequencies
- Front = High + Medium frequencies
- Bottom = High + Medium + Low frequencies
- Left = Minimum brightness
- Right = Maximum birghtness
- Front = Long notes
- Back = Short notes
*/
import("stdfaust.lib");
KEY = 60; // basic midi key
NCY = 15; // note cycle length
CCY = 15; // control cycle length
BPS = 360; // general tempo (ba.beat per sec)
process = kisana;
//-------------------------------kisana----------------------------------
// USAGE: kisana : _,_;
// 3-loops string instrument
//-----------------------------------------------------------------------
kisana = vgroup("Kisana", harpe(C,11,48), harpe(C,11,60), (harpe(C,11,72) : *(1.5), *(1.5))
:>*(l))
with {
l = -20 : ba.db2linear;//hslider("[1]Volume",-20, -60, 0, 0.01) : ba.db2linear;
C = hslider("[2]Brightness[acc:0 1 -10 0 10]", 0.2, 0, 1, 0.01) : ba.automat(BPS, CCY, 0.0);
};
//----------------------------------Harpe--------------------------------
// USAGE: harpe(C,10,60) : _,_;
// C is the filter coefficient 0..1
// Build a N (10) strings harpe using a pentatonic scale
// based on midi key b (60)
// Each string is triggered by a specific
// position of the "hand"
//-----------------------------------------------------------------------
harpe(C,N,b) = hand(b) <: par(i, N, position(i+1)
: string(C,Penta(b).degree2Hz(i), att, lvl)
: pan((i+0.5)/N) )
:> _,_
with {
att = hslider("[3]Resonance[acc:2 1 -10 0 12]", 4, 0.1, 10, 0.01);
hand(48) = vslider("h:[1]Instrument Hands/1 (Note %b)[unit:pk][acc:1 0 -10 0 14]", 0, 0, N, 1) : int : ba.automat(120, CCY, 0.0);
hand(60) = vslider("h:[1]Instrument Hands/2 (Note %b)[unit:pk][acc:1 0 -10 0 14]", 2, 0, N, 1) : int : ba.automat(240, CCY, 0.0);
hand(72) = vslider("h:[1]Instrument Hands/3 (Note %b)[unit:pk][acc:1 0 -10 0 10]", 4, 0, N, 1) : int : ba.automat(480, CCY, 0.0);
//lvl = vslider("h:loop/level", 0, 0, 6, 1) : int : ba.automat(BPS, CCY, 0.0) : -(6) : ba.db2linear;
lvl = 1;
pan(p) = _ <: *(sqrt(1-p)), *(sqrt(p));
position(a,x) = abs(x - a) < 0.5;
};
//----------------------------------Penta-------------------------------
// Pentatonic scale with degree to midi and degree to Hz conversion
// USAGE: Penta(60).degree2midi(3) ==> 67 midikey
// Penta(60).degree2Hz(4) ==> 440 Hz
//-----------------------------------------------------------------------
Penta(key) = environment {
A4Hz = 440;
degree2midi(0) = key+0;
degree2midi(1) = key+2;
degree2midi(2) = key+4;
degree2midi(3) = key+7;
degree2midi(4) = key+9;
degree2midi(d) = degree2midi(d-5)+12;
degree2Hz(d) = A4Hz*semiton(degree2midi(d)-69) with { semiton(n) = 2.0^(n/12.0); };
};
//----------------------------------String-------------------------------
// A karplus-strong string.
//
// USAGE: string(440Hz, 4s, 1.0, button("play"))
// or button("play") : string(440Hz, 4s, 1.0)
//-----------------------------------------------------------------------
string(coef, freq, t60, level, trig) = no.noise*level
: *(trig : trigger(freq2samples(freq)))
: resonator(freq2samples(freq), att)
with {
resonator(d,a) = (+ : @(d-1)) ~ (average : *(a));
average(x) = (x*(1+coef)+x'*(1-coef))/2;
trigger(n) = upfront : + ~ decay(n) : >(0.0);
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
freq2samples(f) = 44100.0/f;
att = pow(0.001,1.0/(freq*t60)); // attenuation coefficient
random = +(12345)~*(1103515245);
noise = random/2147483647.0;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/Kisana.dsp | faust | Modifications GRAME July 2015
========= DESCRITPION =============
- Kisana : 3-loops string instrument (based on Karplus-Strong)
- Head = Silence
- Tilt = High frequencies
- Front = High + Medium frequencies
- Bottom = High + Medium + Low frequencies
- Left = Minimum brightness
- Right = Maximum birghtness
- Front = Long notes
- Back = Short notes
basic midi key
note cycle length
control cycle length
general tempo (ba.beat per sec)
-------------------------------kisana----------------------------------
USAGE: kisana : _,_;
3-loops string instrument
-----------------------------------------------------------------------
hslider("[1]Volume",-20, -60, 0, 0.01) : ba.db2linear;
----------------------------------Harpe--------------------------------
USAGE: harpe(C,10,60) : _,_;
C is the filter coefficient 0..1
Build a N (10) strings harpe using a pentatonic scale
based on midi key b (60)
Each string is triggered by a specific
position of the "hand"
-----------------------------------------------------------------------
lvl = vslider("h:loop/level", 0, 0, 6, 1) : int : ba.automat(BPS, CCY, 0.0) : -(6) : ba.db2linear;
----------------------------------Penta-------------------------------
Pentatonic scale with degree to midi and degree to Hz conversion
USAGE: Penta(60).degree2midi(3) ==> 67 midikey
Penta(60).degree2Hz(4) ==> 440 Hz
-----------------------------------------------------------------------
----------------------------------String-------------------------------
A karplus-strong string.
USAGE: string(440Hz, 4s, 1.0, button("play"))
or button("play") : string(440Hz, 4s, 1.0)
-----------------------------------------------------------------------
attenuation coefficient | declare name "Kisana";
declare author "Yann Orlarey";
import("stdfaust.lib");
process = kisana;
kisana = vgroup("Kisana", harpe(C,11,48), harpe(C,11,60), (harpe(C,11,72) : *(1.5), *(1.5))
:>*(l))
with {
C = hslider("[2]Brightness[acc:0 1 -10 0 10]", 0.2, 0, 1, 0.01) : ba.automat(BPS, CCY, 0.0);
};
harpe(C,N,b) = hand(b) <: par(i, N, position(i+1)
: string(C,Penta(b).degree2Hz(i), att, lvl)
: pan((i+0.5)/N) )
:> _,_
with {
att = hslider("[3]Resonance[acc:2 1 -10 0 12]", 4, 0.1, 10, 0.01);
hand(48) = vslider("h:[1]Instrument Hands/1 (Note %b)[unit:pk][acc:1 0 -10 0 14]", 0, 0, N, 1) : int : ba.automat(120, CCY, 0.0);
hand(60) = vslider("h:[1]Instrument Hands/2 (Note %b)[unit:pk][acc:1 0 -10 0 14]", 2, 0, N, 1) : int : ba.automat(240, CCY, 0.0);
hand(72) = vslider("h:[1]Instrument Hands/3 (Note %b)[unit:pk][acc:1 0 -10 0 10]", 4, 0, N, 1) : int : ba.automat(480, CCY, 0.0);
lvl = 1;
pan(p) = _ <: *(sqrt(1-p)), *(sqrt(p));
position(a,x) = abs(x - a) < 0.5;
};
Penta(key) = environment {
A4Hz = 440;
degree2midi(0) = key+0;
degree2midi(1) = key+2;
degree2midi(2) = key+4;
degree2midi(3) = key+7;
degree2midi(4) = key+9;
degree2midi(d) = degree2midi(d-5)+12;
degree2Hz(d) = A4Hz*semiton(degree2midi(d)-69) with { semiton(n) = 2.0^(n/12.0); };
};
string(coef, freq, t60, level, trig) = no.noise*level
: *(trig : trigger(freq2samples(freq)))
: resonator(freq2samples(freq), att)
with {
resonator(d,a) = (+ : @(d-1)) ~ (average : *(a));
average(x) = (x*(1+coef)+x'*(1-coef))/2;
trigger(n) = upfront : + ~ decay(n) : >(0.0);
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
freq2samples(f) = 44100.0/f;
random = +(12345)~*(1103515245);
noise = random/2147483647.0;
};
|
54ead1d05a7a380b8c70f3b0fffbf5e04e0eb9653b12c9803c36f945e1328eb1 | RuolunWeng/ruolunweng.github.io | SBrassMulti.dsp | declare name "Multiple Brass";
declare description "WaveGuide Brass instrument from STK";
declare author "ER"; //Adapted from Brass by Romain Michon ([email protected]);
import("stdfaust.lib");
instrument=library("instruments.lib");
/* ========= DESCRITPION ===========
- Triple Brass
- Left = Silence
- Other positions = interpolating brass voices
*/
//==================== INSTRUMENT =======================
process = vgroup("Brass Instrument", par(i, 3, brass(i)) :>_);
brass(n) = (borePressure <: deltaPressure(pressure(n)),_ :
(lipFilter(freq(n)) <: *(mouthPressure(pressure(n))),(1-_)),_ : _, * :> + :
fi.dcblocker) ~ (boreDelay(freq(n)))
*(gain(n)): fi.lowpass((n+1),((n+1)*1500));
//==================== GUI SPECIFICATION ================
//gate = checkbox(" Play");
gate = hslider(" ON/OFF", 0, 0, 1, 1);
freq(0) = hslider("h:Instrument/v:Frequencies/Frequency 1 [unit:Hz][acc:1 1 -10 0 10]",370,280,380, 0.01):si.smooth(0.999);
freq(1) = hslider("h:Instrument/v:Frequencies/Frequency 2 [unit:Hz][acc:0 1 -10 0 10]",440,380,550,0.01):si.smooth(0.999);
freq(2) = hslider("h:Instrument/v:Frequencies/Frequency 3 [unit:Hz][acc:2 0 -10 0 12]",587.32,550,700,0.01):si.smooth(0.999);
gain(0) = hslider("h:Instrument/v:Gain/Volume 1 [style:knob][acc:1 0 -10 0 12][tooltip:Gain (value between 0 and 1)]",0.5,0,1,0.01);
gain(1) = hslider("h:Instrument/v:Gain/Volume 2 [style:knob][acc:0 0 -10 0 12][tooltip:Gain (value between 0 and 1)]",0.5,0,1,0.01);
gain(2) = hslider("h:Instrument/v:Gain/Volume 3 [style:knob][acc:2 1 -10 0 10][tooltip:Gain (value between 0 and 1)]",0.5,0,0.5,0.01);
pressure(0) = 0.37;
pressure(1) = 0.68;
pressure(2) = 1.0;
lipTension = 0.780;
slideLength = 0.041;
vibratoFreq = 6;
vibratoGain = 0.05;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.1;
envelopeAttack = 0.01;
envelopeDecay = 0.001;
envelopeRelease = 2;
//==================== SIGNAL PROCESSING ================
//---------- Synthesis parameters computing and functions declaration ----------
//lips are simulated by a biquad filter whose output is squared and hard-clipped, bandPassH and saturationPos are declared in instrument.lib
lipFilterFrequency(f) = f*pow(4,(2*lipTension)-1);
lipFilter(f) = *(0.03) : instrument.bandPassH(lipFilterFrequency(f),0.997) <: * : instrument.saturationPos;
//delay times in number of samples
slideTarget(f) = ((ma.SR/f)*2 + 3)*(0.5 + slideLength);
boreDelay(f) = de.fdelay(4096,slideTarget(f));
//----------------------- Algorithm implementation ----------------------------
//vibrato
vibrato = vibratoGain*os.osc(vibratoFreq)*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate);
//envelope (Attack / Decay / Sustain / Release), breath pressure and vibrato
breathPressure(p) = p*en.adsr(envelopeAttack,envelopeDecay,1,envelopeRelease,gate) + vibrato;
mouthPressure(p) = 0.3*breathPressure(p);
//scale the delay feedback
borePressure = *(0.85);
//differencial presure
deltaPressure(p) = mouthPressure(p) - _;
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SBrassMulti.dsp | faust | Adapted from Brass by Romain Michon ([email protected]);
========= DESCRITPION ===========
- Triple Brass
- Left = Silence
- Other positions = interpolating brass voices
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
gate = checkbox(" Play");
==================== SIGNAL PROCESSING ================
---------- Synthesis parameters computing and functions declaration ----------
lips are simulated by a biquad filter whose output is squared and hard-clipped, bandPassH and saturationPos are declared in instrument.lib
delay times in number of samples
----------------------- Algorithm implementation ----------------------------
vibrato
envelope (Attack / Decay / Sustain / Release), breath pressure and vibrato
scale the delay feedback
differencial presure | declare name "Multiple Brass";
declare description "WaveGuide Brass instrument from STK";
import("stdfaust.lib");
instrument=library("instruments.lib");
process = vgroup("Brass Instrument", par(i, 3, brass(i)) :>_);
brass(n) = (borePressure <: deltaPressure(pressure(n)),_ :
(lipFilter(freq(n)) <: *(mouthPressure(pressure(n))),(1-_)),_ : _, * :> + :
fi.dcblocker) ~ (boreDelay(freq(n)))
*(gain(n)): fi.lowpass((n+1),((n+1)*1500));
gate = hslider(" ON/OFF", 0, 0, 1, 1);
freq(0) = hslider("h:Instrument/v:Frequencies/Frequency 1 [unit:Hz][acc:1 1 -10 0 10]",370,280,380, 0.01):si.smooth(0.999);
freq(1) = hslider("h:Instrument/v:Frequencies/Frequency 2 [unit:Hz][acc:0 1 -10 0 10]",440,380,550,0.01):si.smooth(0.999);
freq(2) = hslider("h:Instrument/v:Frequencies/Frequency 3 [unit:Hz][acc:2 0 -10 0 12]",587.32,550,700,0.01):si.smooth(0.999);
gain(0) = hslider("h:Instrument/v:Gain/Volume 1 [style:knob][acc:1 0 -10 0 12][tooltip:Gain (value between 0 and 1)]",0.5,0,1,0.01);
gain(1) = hslider("h:Instrument/v:Gain/Volume 2 [style:knob][acc:0 0 -10 0 12][tooltip:Gain (value between 0 and 1)]",0.5,0,1,0.01);
gain(2) = hslider("h:Instrument/v:Gain/Volume 3 [style:knob][acc:2 1 -10 0 10][tooltip:Gain (value between 0 and 1)]",0.5,0,0.5,0.01);
pressure(0) = 0.37;
pressure(1) = 0.68;
pressure(2) = 1.0;
lipTension = 0.780;
slideLength = 0.041;
vibratoFreq = 6;
vibratoGain = 0.05;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.1;
envelopeAttack = 0.01;
envelopeDecay = 0.001;
envelopeRelease = 2;
lipFilterFrequency(f) = f*pow(4,(2*lipTension)-1);
lipFilter(f) = *(0.03) : instrument.bandPassH(lipFilterFrequency(f),0.997) <: * : instrument.saturationPos;
slideTarget(f) = ((ma.SR/f)*2 + 3)*(0.5 + slideLength);
boreDelay(f) = de.fdelay(4096,slideTarget(f));
vibrato = vibratoGain*os.osc(vibratoFreq)*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate);
breathPressure(p) = p*en.adsr(envelopeAttack,envelopeDecay,1,envelopeRelease,gate) + vibrato;
mouthPressure(p) = 0.3*breathPressure(p);
borePressure = *(0.85);
deltaPressure(p) = mouthPressure(p) - _;
|
106bf3774e84d93677d29f0e7d7478b7929a8e888810ebd95450262fee550c39 | RuolunWeng/ruolunweng.github.io | AtonalSoftHarp.dsp | declare name "Atonal Soft Harp";
declare author "ER"; //Adapted from NLFeks by Julius Smith and Romain Michon;
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ======================== :
- Soft Atonal Harp
- Head = High frequencies + Reverberation
- Bottom = Low frequencies / Silence
- Swing + Right = Plucking all the strings one by one
- Left = Slow rhythm / Silence
- Right = Fast rhythm
- Front = Short and dry notes
- Back = Long and bright notes
- Back + horizontal shaking = vibrato
*/
//==================== INSTRUMENT =======================
process = par(i, N, NFLeks(i)):>_<: instrReverbHarp;
NFLeks(n) = filtered_excitation(n+1,P(freq(n)),freq(n)) : stringloop(freq(n));
//==================== GUI SPECIFICATION ================
N = 20;
hand = hslider("h:[1]/Instrument Hand[acc:1 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 0 -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 no.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 de.delay-line allocation)
ppdel(f) = beta*P(f); // pick position de.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));
//================================ REVERB ==============================
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[2]Reverberation/Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -30 0 12]", 0.1,0.05,1,0.01) : si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[2]Reverberation/ Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -30 0 12]", 0.2,0.05,1.7,0.01) : min(1.7):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/AtonalSoftHarp.dsp | faust | Adapted from NLFeks by Julius Smith and Romain Michon;
=============== DESCRIPTION ======================== :
- Soft Atonal Harp
- Head = High frequencies + Reverberation
- Bottom = Low frequencies / Silence
- Swing + Right = Plucking all the strings one by one
- Left = Slow rhythm / Silence
- Right = Fast rhythm
- Front = Short and dry notes
- Back = Long and bright notes
- Back + horizontal shaking = vibrato
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
=> gate
String decay time in seconds:
-60db decay time (sec)
---------------------------------- FREQUENCY TABLE ---------------------------
==================== SIGNAL PROCESSING ================
----------------------- noiseburst -------------------------
White no.noise burst (adapted from Faust's karplus.dsp example)
Requires music.lib (for no.noise)
fundamental period in samples
maximum P (for de.delay-line allocation)
pick position de.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
================================ REVERB ============================== | declare name "Atonal Soft Harp";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = par(i, N, NFLeks(i)):>_<: instrReverbHarp;
NFLeks(n) = filtered_excitation(n+1,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));
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[2]Reverberation/Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -30 0 12]", 0.1,0.05,1,0.01) : si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[2]Reverberation/ Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -30 0 12]", 0.2,0.05,1.7,0.01) : min(1.7):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
b792313d8ad33eca0e952330494464f83c708f42206655914474d4a510e56948 | RuolunWeng/ruolunweng.github.io | PentatonicSoftHarp.dsp | declare name "Pentatonic Soft Harp";
declare author "ER";//Adapted from "Nonlinear EKS" by Julius Smith and Romain Michon;
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION =================
- Reverberated pentatonic soft harp
- Left = Lower frequencies/Silence when still
- Front = Resonance
- Back = No resonance
- Right = Higher frequencies/Fast rhythm
- Head = Reverberation
- Rocking = plucking all strings one by one
*/
//==================== INSTRUMENT =======================
process = par(i, N, NFLeks(i)):>_<: instrReverbHarp;
NFLeks(n) = filtered_excitation(n,P(freq(n)),freq(n)) : stringloop(freq(n));
//==================== GUI SPECIFICATION ================
// standard MIDI voice parameters:
// NOTE: The labels MUST be "freq", "gain", and "gate" for faust2pd
N = 20;
hand = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 10, 0, N, 1) : ba.automat(bps, 15, 0.0)// => gate
with{
bps = hslider("[2]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 = 0.81;
beta = 0.5;
t60 = hslider("h:[3]Reverb/[1]Resonance[unit:s][acc:2 1 -10 0 10]", 5, 0.5, 10, 0.01); // -60db decay time (sec)
B = 0;
L = -10 : ba.db2linear;
//---------------------------------- FREQUENCY TABLE ---------------------------
freq(0) = 184.99;
freq(1) = 207.65;
freq(2) = 233.08;
freq(3) = 277.18;
freq(4) = 311.12;
freq(d) = freq(d-5)*2;
//==================== SIGNAL PROCESSING ================
//----------------------- noiseburst -------------------------
// White no.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);
};
//nlfOrder = 6;
P(f) = ma.SR/f ; // fundamental period in samples
Pmax = 4096; // maximum P (for de.delay-line allocation)
ppdel(f) = beta*P(f); // pick position de.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));
//================================= REVERB ==============================
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[3]Reverb/[1]Reverberation Volume(InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[3]Reverb/[2]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1.7,0.01):min(1.3):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/PentatonicSoftHarp.dsp | faust | Adapted from "Nonlinear EKS" by Julius Smith and Romain Michon;
=============== DESCRIPTION =================
- Reverberated pentatonic soft harp
- Left = Lower frequencies/Silence when still
- Front = Resonance
- Back = No resonance
- Right = Higher frequencies/Fast rhythm
- Head = Reverberation
- Rocking = plucking all strings one by one
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
standard MIDI voice parameters:
NOTE: The labels MUST be "freq", "gain", and "gate" for faust2pd
=> gate
-60db decay time (sec)
---------------------------------- FREQUENCY TABLE ---------------------------
==================== SIGNAL PROCESSING ================
----------------------- noiseburst -------------------------
White no.noise burst (adapted from Faust's karplus.dsp example)
Requires music.lib (for no.noise)
nlfOrder = 6;
fundamental period in samples
maximum P (for de.delay-line allocation)
pick position de.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
================================= REVERB ============================== | declare name "Pentatonic Soft Harp";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = par(i, N, NFLeks(i)):>_<: instrReverbHarp;
NFLeks(n) = filtered_excitation(n,P(freq(n)),freq(n)) : stringloop(freq(n));
N = 20;
with{
bps = hslider("[2]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 = 0.81;
beta = 0.5;
B = 0;
L = -10 : ba.db2linear;
freq(0) = 184.99;
freq(1) = 207.65;
freq(2) = 233.08;
freq(3) = 277.18;
freq(4) = 311.12;
freq(d) = freq(d-5)*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));
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[3]Reverb/[1]Reverberation Volume(InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[3]Reverb/[2]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1.7,0.01):min(1.3):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
0b441e02bc7634ea175b0010ffca7c78508a19f1849d400e7c018766eddb9298 | RuolunWeng/ruolunweng.github.io | Brass.dsp | declare name "Brass";
declare description "WaveGuide Brass instrument from STK";
declare author "Romain Michon ([email protected])";
declare copyright "Romain Michon";
declare version "1.0";
declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
declare description "A simple brass instrument waveguide model, a la Cook (TBone, HosePlayer).";
declare reference "https://ccrma.stanford.edu/~jos/pasp/Brasses.html";
//Modification GRAME July 2015
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- Brass instrument
- Head = Reverb/Silence
- Upward = Higher frequency
- Downward = Lower frequency
*/
//==================== INSTRUMENT =======================
process = vgroup("Brass Instrument", Brass <: InstrReverBrass :>_);
Brass = (borePressure <: deltaPressure,_ :
(lipFilter <: *(mouthPressure),(1-_)),_ : _, * :> + :
fi.dcblocker) ~ (boreDelay) :
*(gain)*(2);
//==================== GUI SPECIFICATION ================
freq = hslider("h:[1]Instrument/Frequency[1][unit:Hz] [tooltip:Tone frequency][acc:1 1 -10 0 10]", 300,170,700,1):si.smooth(0.999);
gain = 0.8;
gate = checkbox("h:[1]Instrument/ ON/OFF (ASR Envelope)");
lipTension = 0.780;
pressure = 1;
slideLength = 0.041;
vibratoFreq = hslider("v:[3]Parameters/h:/Vibrato Frequency (Vibrato Envelope)[unit:Hz][style:knob][unit:Hz][acc:0 1 -10 0 10]", 5,1,10,0.01);
vibratoGain = 0.05;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.1;
envelopeDecay = 0.001;
envelopeAttack = 0.005;
envelopeRelease = 0.07;
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//lips are simulated by a biquad filter whose output is squared and hard-clipped, instrument.bandPassH and instrument.saturationPos are declared in instrument.lib
lipFilterFrequency = freq*pow(4,(2*lipTension)-1);
lipFilter = *(0.03) : instrument.bandPassH(lipFilterFrequency,0.997) <: * : instrument.saturationPos;
//de.delay times in number of samples
slideTarget = ((ma.SR/freq)*2 + 3)*(0.5 + slideLength);
boreDelay = de.fdelay(4096,slideTarget);
//----------------------- Algorithm implementation ----------------------------
//vibrato
vibrato = vibratoGain*os.osc(vibratoFreq)*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate);
//envelope (Attack / Decay / Sustain / Release), breath pressure and vibrato
breathPressure = pressure*en.adsr(envelopeAttack,envelopeDecay,1,envelopeRelease,gate) + vibrato;
mouthPressure = 0.3*breathPressure;
//scale the de.delay feedback
borePressure = *(0.85);
//differencial presure
deltaPressure = mouthPressure - _;
//-------------------------------- InstrReverb ---------------------------------
InstrReverBrass = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
roomSize = hslider("v:[4]Reverb/Reverberation Room Size (InstrReverb)[acc:1 1 -15 0 12]", 0.2,0.05,1.7,0.01) : min(1.7) : max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/Brass.dsp | faust | Synthesis Tool Kit 4.3 (MIT style license);
Modification GRAME July 2015
=============== DESCRIPTION ================= :
- Brass instrument
- Head = Reverb/Silence
- Upward = Higher frequency
- Downward = Lower frequency
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
lips are simulated by a biquad filter whose output is squared and hard-clipped, instrument.bandPassH and instrument.saturationPos are declared in instrument.lib
de.delay times in number of samples
----------------------- Algorithm implementation ----------------------------
vibrato
envelope (Attack / Decay / Sustain / Release), breath pressure and vibrato
scale the de.delay feedback
differencial presure
-------------------------------- InstrReverb --------------------------------- | declare name "Brass";
declare description "WaveGuide Brass instrument from STK";
declare author "Romain Michon ([email protected])";
declare copyright "Romain Michon";
declare version "1.0";
declare description "A simple brass instrument waveguide model, a la Cook (TBone, HosePlayer).";
declare reference "https://ccrma.stanford.edu/~jos/pasp/Brasses.html";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("Brass Instrument", Brass <: InstrReverBrass :>_);
Brass = (borePressure <: deltaPressure,_ :
(lipFilter <: *(mouthPressure),(1-_)),_ : _, * :> + :
fi.dcblocker) ~ (boreDelay) :
*(gain)*(2);
freq = hslider("h:[1]Instrument/Frequency[1][unit:Hz] [tooltip:Tone frequency][acc:1 1 -10 0 10]", 300,170,700,1):si.smooth(0.999);
gain = 0.8;
gate = checkbox("h:[1]Instrument/ ON/OFF (ASR Envelope)");
lipTension = 0.780;
pressure = 1;
slideLength = 0.041;
vibratoFreq = hslider("v:[3]Parameters/h:/Vibrato Frequency (Vibrato Envelope)[unit:Hz][style:knob][unit:Hz][acc:0 1 -10 0 10]", 5,1,10,0.01);
vibratoGain = 0.05;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.1;
envelopeDecay = 0.001;
envelopeAttack = 0.005;
envelopeRelease = 0.07;
lipFilterFrequency = freq*pow(4,(2*lipTension)-1);
lipFilter = *(0.03) : instrument.bandPassH(lipFilterFrequency,0.997) <: * : instrument.saturationPos;
slideTarget = ((ma.SR/freq)*2 + 3)*(0.5 + slideLength);
boreDelay = de.fdelay(4096,slideTarget);
vibrato = vibratoGain*os.osc(vibratoFreq)*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate);
breathPressure = pressure*en.adsr(envelopeAttack,envelopeDecay,1,envelopeRelease,gate) + vibrato;
mouthPressure = 0.3*breathPressure;
borePressure = *(0.85);
deltaPressure = mouthPressure - _;
InstrReverBrass = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
roomSize = hslider("v:[4]Reverb/Reverberation Room Size (InstrReverb)[acc:1 1 -15 0 12]", 0.2,0.05,1.7,0.01) : min(1.7) : max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
fed7c24adfaa6f55ac2097fe4d68ebbea8fd1da7fa9ba73e7c027f5b4070eeaf | RuolunWeng/ruolunweng.github.io | SBlowhistleBottle.dsp | declare name "Blowhistle Bottle";
declare author "ER"; //From "Blow bottle" by Romain Michon;
declare version "1.0";
declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
declare description "This object implements a helmholtz resonator (biquad filter) with a polynomial jet excitation (a la Cook).";
/* =============== DESCRIPTION ================= :
- Blow bottles with whistling echo.
- Left : silence/dying echo.
- Front : single blow bottle.
- Back : maximum whistling echo
- Bottom : bottle + whistling echo
- Rocking : changes tone of blow bottle.
*/
import("stdfaust.lib");
instrument = library("instruments.lib");
//==================== INSTRUMENT =======================
process = vgroup("Blowhistle Bottles", par(i, N, blow(i)) :>_);
blow(n)= par(i, 2,
//differential pressure
(-(breathPressure(trigger(n))) <:
((+(1))*randPressure((trigger(n))) : +(breathPressure(trigger(n)))) - *(instrument.jetTable),_ : baPaF(i,n),_)~_: !,_:
//signal scaling
fi.dcblocker*envelopeG(trigger(n))*(0.5)<:+(voice(i,n))*resonGain(i)):>_
with{
baPaF(0,n) = bandPassFilter(freq(n));
baPaF(1,n) = bandPassFilter(freq(n)*8);
voice(0,n) = 0*n;
voice(1,n) = 1*(fi.resonbp(freq(n)*8,Q,gain):echo);
resonGain(0) = 1;
resonGain(1) =(hslider("v:[1]Instrument/Whistle Volume[acc:2 0 -10 0 10]", 0.07, 0, 0.2, 0.001))^2:si.smooth(0.999);
echo = _:+~(@(delayEcho):*(feedback));
delayEcho = 44100;
feedback = hslider("h:[2]Echo/Echo Intensity [style:knob][acc:2 0 -10 0 10]", 0.48, 0.2, 0.98, 0.01):si.smooth(0.999):min(0.98):max(0.2);
};
//==================== GUI SPECIFICATION ================
N = 10;
Q = 30;
position(n) = abs(hand - n) < 0.5;
hand = hslider("v:[1]Instrument/Instrument Hand[acc:0 1 -10 0 10]", 5, 0, N, 1):int:ba.automat(360, 15, 0.0);
envelopeAttack = 0.01;
vibratoFreq = 5;
vibratoGain = 0.1;
//--------------------- Non-variable Parameters -------------
gain = 0.5;
noiseGain = 0.5;
pressure = 1.2;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.01;
envelopeDecay = 0.01;
envelopeRelease = 0.5;
//----------------------- Frequency Table --------------------
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 195.99;
freq(4) = 220.00;
freq(d) = freq(d-5)*2;
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//botlle radius
bottleRadius = 0.999;
bandPassFilter(f) = instrument.bandPass(f,bottleRadius);
//----------------------- Algorithm implementation ----------------------------
//global envelope is of type attack - decay - sustain - release
envelopeG(t) = gain*en.adsr(gain*envelopeAttack,envelopeDecay,0.8,envelopeRelease,t);
//pressure envelope is also ADSR
envelope(t) = pressure*en.adsr(gain*0.02,0.01,0.8,gain*0.2,t);
//vibrato
vibrato(t) = os.osc(vibratoFreq)*vibratoGain*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*os.osc(vibratoFreq);
//breat pressure
breathPressure(t) = envelope(t) + vibrato(t);
//breath no.noise
randPressure(t) = noiseGain*no.noise*breathPressure(t) ;
//------------------------- Enveloppe Trigger --------------------------------------------
trigger(n) = position(n): trig
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
trig = upfront : release(8820) : >(0.0);
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SBlowhistleBottle.dsp | faust | From "Blow bottle" by Romain Michon;
Synthesis Tool Kit 4.3 (MIT style license);
=============== DESCRIPTION ================= :
- Blow bottles with whistling echo.
- Left : silence/dying echo.
- Front : single blow bottle.
- Back : maximum whistling echo
- Bottom : bottle + whistling echo
- Rocking : changes tone of blow bottle.
==================== INSTRUMENT =======================
differential pressure
signal scaling
==================== GUI SPECIFICATION ================
--------------------- Non-variable Parameters -------------
----------------------- Frequency Table --------------------
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
botlle radius
----------------------- Algorithm implementation ----------------------------
global envelope is of type attack - decay - sustain - release
pressure envelope is also ADSR
vibrato
breat pressure
breath no.noise
------------------------- Enveloppe Trigger -------------------------------------------- | declare name "Blowhistle Bottle";
declare version "1.0";
declare description "This object implements a helmholtz resonator (biquad filter) with a polynomial jet excitation (a la Cook).";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("Blowhistle Bottles", par(i, N, blow(i)) :>_);
blow(n)= par(i, 2,
(-(breathPressure(trigger(n))) <:
((+(1))*randPressure((trigger(n))) : +(breathPressure(trigger(n)))) - *(instrument.jetTable),_ : baPaF(i,n),_)~_: !,_:
fi.dcblocker*envelopeG(trigger(n))*(0.5)<:+(voice(i,n))*resonGain(i)):>_
with{
baPaF(0,n) = bandPassFilter(freq(n));
baPaF(1,n) = bandPassFilter(freq(n)*8);
voice(0,n) = 0*n;
voice(1,n) = 1*(fi.resonbp(freq(n)*8,Q,gain):echo);
resonGain(0) = 1;
resonGain(1) =(hslider("v:[1]Instrument/Whistle Volume[acc:2 0 -10 0 10]", 0.07, 0, 0.2, 0.001))^2:si.smooth(0.999);
echo = _:+~(@(delayEcho):*(feedback));
delayEcho = 44100;
feedback = hslider("h:[2]Echo/Echo Intensity [style:knob][acc:2 0 -10 0 10]", 0.48, 0.2, 0.98, 0.01):si.smooth(0.999):min(0.98):max(0.2);
};
N = 10;
Q = 30;
position(n) = abs(hand - n) < 0.5;
hand = hslider("v:[1]Instrument/Instrument Hand[acc:0 1 -10 0 10]", 5, 0, N, 1):int:ba.automat(360, 15, 0.0);
envelopeAttack = 0.01;
vibratoFreq = 5;
vibratoGain = 0.1;
gain = 0.5;
noiseGain = 0.5;
pressure = 1.2;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.01;
envelopeDecay = 0.01;
envelopeRelease = 0.5;
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 195.99;
freq(4) = 220.00;
freq(d) = freq(d-5)*2;
bottleRadius = 0.999;
bandPassFilter(f) = instrument.bandPass(f,bottleRadius);
envelopeG(t) = gain*en.adsr(gain*envelopeAttack,envelopeDecay,0.8,envelopeRelease,t);
envelope(t) = pressure*en.adsr(gain*0.02,0.01,0.8,gain*0.2,t);
vibrato(t) = os.osc(vibratoFreq)*vibratoGain*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*os.osc(vibratoFreq);
breathPressure(t) = envelope(t) + vibrato(t);
randPressure(t) = noiseGain*no.noise*breathPressure(t) ;
trigger(n) = position(n): trig
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
trig = upfront : release(8820) : >(0.0);
};
|
66d47fb2e29d871081ac07abc88b25a1f83cbb44ed4277af3fb822eb73d406dd | RuolunWeng/ruolunweng.github.io | STunedBar6.dsp | declare name "Tuned Bar";
declare description "Nonlinear Banded Waveguide Models";
declare name "Six Rack Tuned Bars";
declare author "ER";//From "Tuned Bar" by Romain Michon ([email protected]);
/* =========== DESCRIPTION =============
- Six rack tuned bars
- Head = Silence/Resonance
- Tilt = High frequencies
- Front = High + Medium frequencies
- Bottom = High + Medium + Low frequencies
*/
import("stdfaust.lib");
instrument = library("instruments.lib");
//==================== INSTRUMENT =======================
process = vgroup("tunedBars",hgroup("[1]",par(i, 6, onerack(i,i,i))):>_);
onerack(h,n,e) = hgroup("Bar %n", par(i, 5, tunedBar(h,i,e)));
tunedBar(h,n,e) =
((select-1)*-1) <:
//nModes resonances with nModes feedbacks for bow table look-up
par(i,nModes,(resonance(i,freqqy(n,e),gate(h,n))~_)) :> + :
//Signal Scaling and stereo
*(4);
//==================== GUI SPECIFICATION ================
gain = 0.8;
gate(h,n) = position(h,n) : upfront;
hand(0) = vslider("Instrument Hand[acc:1 0 -10 0 18]", 0, 0, 5, 1):int:ba.automat(120, 15, 0.0);
hand(1) = vslider("Instrument Hand[acc:1 0 -10 0 18]", 0, 0, 5, 1):int:ba.automat(120, 15, 0.0);
hand(2) = vslider("Instrument Hand[acc:1 0 -10 0 14]", 2, 0, 5, 1):int:ba.automat(240, 15, 0.0);
hand(3) = vslider("Instrument Hand[acc:1 0 -10 0 14]", 2, 0, 5, 1):int:ba.automat(240, 15, 0.0);
hand(4) = vslider("Instrument Hand[acc:1 0 -10 0 10]", 4, 0, 5, 1):int:ba.automat(480, 15, 0.0);
hand(5) = vslider("Instrument Hand[acc:1 0 -10 0 10]", 4, 0, 5, 1):int:ba.automat(480, 15, 0.0);
position(h,n) = abs(hand(h) - n) < 0.5;
upfront(x) = x>x';
select = 1;
integrationConstant = 0;
baseGain = 1;
//----------------------- Frequency Table --------------------
freq(0) = 92.49;
freq(1) = 103.82;
freq(2) = 116.54;
freq(3) = 138.59;
freq(4) = 155.56;
freq(d) = freq(d-5)*2;
freqqy(d,e) = freq(d+e*5);
//==================== MODAL PARAMETERS ================
preset = 2;
nMode(2) = 4;
modes(2,0) = 1;
basegains(2,0) = pow(0.999,1);
excitation(2,0,g) = 1*gain*g/nMode(2);
modes(2,1) = 4.0198391420;
basegains(2,1) = pow(0.999,2);
excitation(2,1,g) = 1*gain*g/nMode(2);
modes(2,2) = 10.7184986595;
basegains(2,2) = pow(0.999,3);
excitation(2,2,g) = 1*gain*g/nMode(2);
modes(2,3) = 18.0697050938;
basegains(2,3) = pow(0.999,4);
excitation(2,3,g) = 1*gain*g/nMode(2);
//==================== SIGNAL PROCESSING ================
//----------------------- Nonlinear filter ----------------------------
//nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
//nonlinear filter order
nlfOrder = 6;
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//the number of modes depends on the preset being used
nModes = nMode(preset);
delayLengthBase(f) = ma.SR/f;
//delay lengths in number of samples
delayLength(x,f) = delayLengthBase(f)/modes(preset,x);
//delay lines
delayLine(x,f) = de.delay(4096,delayLength(x,f));
//Filter bank: fi.bandpass filters (declared in instrument.lib)
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x,f) = instrument.bandPass(f*modes(preset,x),radius);
//----------------------- Algorithm implementation ----------------------------
//One resonance
resonance(x,f,g) = + : + (excitation(preset,x,g)*select) : delayLine(x,f) : *(basegains(preset,x)) : bandPassFilter(x,f);
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/STunedBar6.dsp | faust | From "Tuned Bar" by Romain Michon ([email protected]);
=========== DESCRIPTION =============
- Six rack tuned bars
- Head = Silence/Resonance
- Tilt = High frequencies
- Front = High + Medium frequencies
- Bottom = High + Medium + Low frequencies
==================== INSTRUMENT =======================
nModes resonances with nModes feedbacks for bow table look-up
Signal Scaling and stereo
==================== GUI SPECIFICATION ================
----------------------- Frequency Table --------------------
==================== MODAL PARAMETERS ================
==================== SIGNAL PROCESSING ================
----------------------- Nonlinear filter ----------------------------
nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
nonlinear filter order
----------------------- Synthesis parameters computing and functions declaration ----------------------------
the number of modes depends on the preset being used
delay lengths in number of samples
delay lines
Filter bank: fi.bandpass filters (declared in instrument.lib)
----------------------- Algorithm implementation ----------------------------
One resonance | declare name "Tuned Bar";
declare description "Nonlinear Banded Waveguide Models";
declare name "Six Rack Tuned Bars";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("tunedBars",hgroup("[1]",par(i, 6, onerack(i,i,i))):>_);
onerack(h,n,e) = hgroup("Bar %n", par(i, 5, tunedBar(h,i,e)));
tunedBar(h,n,e) =
((select-1)*-1) <:
par(i,nModes,(resonance(i,freqqy(n,e),gate(h,n))~_)) :> + :
*(4);
gain = 0.8;
gate(h,n) = position(h,n) : upfront;
hand(0) = vslider("Instrument Hand[acc:1 0 -10 0 18]", 0, 0, 5, 1):int:ba.automat(120, 15, 0.0);
hand(1) = vslider("Instrument Hand[acc:1 0 -10 0 18]", 0, 0, 5, 1):int:ba.automat(120, 15, 0.0);
hand(2) = vslider("Instrument Hand[acc:1 0 -10 0 14]", 2, 0, 5, 1):int:ba.automat(240, 15, 0.0);
hand(3) = vslider("Instrument Hand[acc:1 0 -10 0 14]", 2, 0, 5, 1):int:ba.automat(240, 15, 0.0);
hand(4) = vslider("Instrument Hand[acc:1 0 -10 0 10]", 4, 0, 5, 1):int:ba.automat(480, 15, 0.0);
hand(5) = vslider("Instrument Hand[acc:1 0 -10 0 10]", 4, 0, 5, 1):int:ba.automat(480, 15, 0.0);
position(h,n) = abs(hand(h) - n) < 0.5;
upfront(x) = x>x';
select = 1;
integrationConstant = 0;
baseGain = 1;
freq(0) = 92.49;
freq(1) = 103.82;
freq(2) = 116.54;
freq(3) = 138.59;
freq(4) = 155.56;
freq(d) = freq(d-5)*2;
freqqy(d,e) = freq(d+e*5);
preset = 2;
nMode(2) = 4;
modes(2,0) = 1;
basegains(2,0) = pow(0.999,1);
excitation(2,0,g) = 1*gain*g/nMode(2);
modes(2,1) = 4.0198391420;
basegains(2,1) = pow(0.999,2);
excitation(2,1,g) = 1*gain*g/nMode(2);
modes(2,2) = 10.7184986595;
basegains(2,2) = pow(0.999,3);
excitation(2,2,g) = 1*gain*g/nMode(2);
modes(2,3) = 18.0697050938;
basegains(2,3) = pow(0.999,4);
excitation(2,3,g) = 1*gain*g/nMode(2);
nlfOrder = 6;
nModes = nMode(preset);
delayLengthBase(f) = ma.SR/f;
delayLength(x,f) = delayLengthBase(f)/modes(preset,x);
delayLine(x,f) = de.delay(4096,delayLength(x,f));
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x,f) = instrument.bandPass(f*modes(preset,x),radius);
resonance(x,f,g) = + : + (excitation(preset,x,g)*select) : delayLine(x,f) : *(basegains(preset,x)) : bandPassFilter(x,f);
|
7576ff9aad06a197efb47bf379d565565bc1ac5cbf40de51455f793f3a2655bd | RuolunWeng/ruolunweng.github.io | PentatonicDryHarp.dsp | declare name "PentatonicDryHarp";
declare author "ER";//Adapted from Harpe by Yann Orlarey;
//Modification Grame July 2015
/* =============== DESCRIPTION ================= :
- Reverberated pentatonic dry harp
- Left = Lower frequencies/Silence when still
- Front = Resonance (longer notes)
- Back = No Resonance (dry notes)
- Right = Higher frequencies/Fast rhythm
- Head = Reverberation
- Rocking = plucking all strings one by one
*/
//-----------------------------------------------
// Harpe : simple string instrument
// (based on Karplus-Strong)
//
//-----------------------------------------------
import("stdfaust.lib");
instrument = library("instruments.lib");
KEY = 60; // basic midi key
NCY = 15; // note cycle length
CCY = 15; // control cycle length
BPS = 360; // general tempo (ba.beat per sec)
//-------------------------------Harpe----------------------------------
// Harpe is a simple string instrument. Move the "hand" to play the
// various strings
//-----------------------------------------------------------------------
process = vgroup("Harp", h : harpe(C,N,K) :> instrReverbHarp : *(l),*(l))
with {
N = 21; // number of strings
K = 48; // Midi key of first string
h = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 11, 0, N, 1) : int: ba.automat(bps, 15, 0.0)
with{
bps = hslider("h:[2]Parameters/[1]Speed[style:knob][acc:0 1 -12 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
l = 0.9;
C = 0.5;
};
//----------------------------------Harpe--------------------------------
// USAGE: hand : harpe(C,10,60) : _,_;
// C is the filter coefficient 0..1
// Build a N (10) strings harpe using a pentatonic scale
// based on midi key b (60)
// Each string is triggered by a specific
// position of the "hand"
//-----------------------------------------------------------------------
harpe(C,N,b) = _ <: par(i, N, position(i+1)
: string(C,Penta(b).degree2Hz(i), att, lvl)
: pan((i+0.5)/N) )
:> _,_
with {
att = hslider("h:[2]Parameters/[2]Resonance[style:knob][acc:2 1 -12 0 10]", 5, 0.1, 10, 0.01):min(10):max(0.1);
lvl = 1;
pan(p) = _ <: *(sqrt(1-p)), *(sqrt(p));
position(a,x) = abs(x - a) < 0.5;
};
//----------------------------------Penta-------------------------------
// Pentatonic scale with degree to midi and degree to Hz conversion
// USAGE: Penta(60).degree2midi(3) ==> 67 midikey
// Penta(60).degree2Hz(4) ==> 440 Hz
//-----------------------------------------------------------------------
Penta(key) = environment {
A4Hz = 440;
degree2midi(0) = key+0;
degree2midi(1) = key+2;
degree2midi(2) = key+4;
degree2midi(3) = key+7;
degree2midi(4) = key+9;
degree2midi(d) = degree2midi(d-5)+12;
degree2Hz(d) = A4Hz*semiton(degree2midi(d)-69) with { semiton(n) = 2.0^(n/12.0); };
};
//----------------------------------String-------------------------------
// A karplus-strong string.
//
// USAGE: string(440Hz, 4s, 1.0, button("play"))
// or button("play") : string(440Hz, 4s, 1.0)
//-----------------------------------------------------------------------
string(coef, freq, t60, level, trig) = no.noise*level
: *(trig : trigger(freq2samples(freq)))
: resonator(freq2samples(freq), att)
with {
resonator(d,a) = (+ : @(d-1)) ~ (average : *(a));
average(x) = (x*(1+coef)+x'*(1-coef))/2;
trigger(n) = upfront : + ~ decay(n) : >(0.0);
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
freq2samples(f) = 44100.0/f;
att = pow(0.001,1.0/(freq*t60)); // attenuation coefficient
};
//================================= REVERB ==============================
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[3]Reverb/[1]Reverberation Volume(InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[3]Reverb/[2]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1.3,0.01):min(1.3):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/PentatonicDryHarp.dsp | faust | Adapted from Harpe by Yann Orlarey;
Modification Grame July 2015
=============== DESCRIPTION ================= :
- Reverberated pentatonic dry harp
- Left = Lower frequencies/Silence when still
- Front = Resonance (longer notes)
- Back = No Resonance (dry notes)
- Right = Higher frequencies/Fast rhythm
- Head = Reverberation
- Rocking = plucking all strings one by one
-----------------------------------------------
Harpe : simple string instrument
(based on Karplus-Strong)
-----------------------------------------------
basic midi key
note cycle length
control cycle length
general tempo (ba.beat per sec)
-------------------------------Harpe----------------------------------
Harpe is a simple string instrument. Move the "hand" to play the
various strings
-----------------------------------------------------------------------
number of strings
Midi key of first string
----------------------------------Harpe--------------------------------
USAGE: hand : harpe(C,10,60) : _,_;
C is the filter coefficient 0..1
Build a N (10) strings harpe using a pentatonic scale
based on midi key b (60)
Each string is triggered by a specific
position of the "hand"
-----------------------------------------------------------------------
----------------------------------Penta-------------------------------
Pentatonic scale with degree to midi and degree to Hz conversion
USAGE: Penta(60).degree2midi(3) ==> 67 midikey
Penta(60).degree2Hz(4) ==> 440 Hz
-----------------------------------------------------------------------
----------------------------------String-------------------------------
A karplus-strong string.
USAGE: string(440Hz, 4s, 1.0, button("play"))
or button("play") : string(440Hz, 4s, 1.0)
-----------------------------------------------------------------------
attenuation coefficient
================================= REVERB ============================== | declare name "PentatonicDryHarp";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("Harp", h : harpe(C,N,K) :> instrReverbHarp : *(l),*(l))
with {
h = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 11, 0, N, 1) : int: ba.automat(bps, 15, 0.0)
with{
bps = hslider("h:[2]Parameters/[1]Speed[style:knob][acc:0 1 -12 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
l = 0.9;
C = 0.5;
};
harpe(C,N,b) = _ <: par(i, N, position(i+1)
: string(C,Penta(b).degree2Hz(i), att, lvl)
: pan((i+0.5)/N) )
:> _,_
with {
att = hslider("h:[2]Parameters/[2]Resonance[style:knob][acc:2 1 -12 0 10]", 5, 0.1, 10, 0.01):min(10):max(0.1);
lvl = 1;
pan(p) = _ <: *(sqrt(1-p)), *(sqrt(p));
position(a,x) = abs(x - a) < 0.5;
};
Penta(key) = environment {
A4Hz = 440;
degree2midi(0) = key+0;
degree2midi(1) = key+2;
degree2midi(2) = key+4;
degree2midi(3) = key+7;
degree2midi(4) = key+9;
degree2midi(d) = degree2midi(d-5)+12;
degree2Hz(d) = A4Hz*semiton(degree2midi(d)-69) with { semiton(n) = 2.0^(n/12.0); };
};
string(coef, freq, t60, level, trig) = no.noise*level
: *(trig : trigger(freq2samples(freq)))
: resonator(freq2samples(freq), att)
with {
resonator(d,a) = (+ : @(d-1)) ~ (average : *(a));
average(x) = (x*(1+coef)+x'*(1-coef))/2;
trigger(n) = upfront : + ~ decay(n) : >(0.0);
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
freq2samples(f) = 44100.0/f;
};
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[3]Reverb/[1]Reverberation Volume(InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[3]Reverb/[2]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1.3,0.01):min(1.3):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
cd40a914d5824b3622adbf6ed3f3f815cc55c07a623227bef88abd6ffe3ce053 | RuolunWeng/ruolunweng.github.io | CMajSoftHarp.dsp | declare name "C Major Soft Harp";
declare author "ER";//Adapted from Nonlinear EKS by Julius Smith and Romain Michon;
declare reference "http://ccrma.stanford.edu/~jos/pasp/vegf.html";
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- Reverberated C Major soft harp
- Left = Lower frequencies/Silence when still
- Front = Resonance
- Back = No resonance
- Right = Higher frequencies/Fast rhythm
- Head = Reverberation
- Rocking = plucking all strings one by one
*/
//==================== INSTRUMENT =======================
process = vgroup("Soft Harp - C Major",par(i, N, NFLeks(i)):>_<: instrReverbHarp);
NFLeks(n) = filtered_excitation(n,P(freq(n)),freq(n)) : stringloop(freq(n));
//==================== GUI SPECIFICATION ================
N = 24;
hand = hslider("h:[1]/Instrument Hand[acc:0 1 -10 0 10]", 12, 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 = 0.81;
beta = 0.5;
t60 = hslider("h:[2]Reverb/ Resonance (InstrReverb)[unit:s][acc:0 0 -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) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 174.61;
freq(4) = 195.99;
freq(5) = 220.00;
freq(6) = 246.94;
freq(d) = freq(d-7)*(2);
//==================== SIGNAL PROCESSING ================
//----------------------- noiseburst -------------------------
// White no.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 de.delay-line allocation)
ppdel(f) = beta*P(f); // pick position de.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));
//================================= REVERB ==============================
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[2]Reverb/ Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -30 0 17]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[2]Reverb/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -30 0 16]", 0.72,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/CMajSoftHarp.dsp | faust | Adapted from Nonlinear EKS by Julius Smith and Romain Michon;
=============== DESCRIPTION ================= :
- Reverberated C Major soft harp
- Left = Lower frequencies/Silence when still
- Front = Resonance
- Back = No resonance
- Right = Higher frequencies/Fast rhythm
- Head = Reverberation
- Rocking = plucking all strings one by one
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
=> gate
-60db decay time (sec)
---------------------------------- FREQUENCY TABLE ---------------------------
==================== SIGNAL PROCESSING ================
----------------------- noiseburst -------------------------
White no.noise burst (adapted from Faust's karplus.dsp example)
Requires music.lib (for no.noise)
fundamental period in samples
maximum P (for de.delay-line allocation)
pick position de.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
================================= REVERB ============================== | declare name "C Major Soft Harp";
declare reference "http://ccrma.stanford.edu/~jos/pasp/vegf.html";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("Soft Harp - C Major",par(i, N, NFLeks(i)):>_<: instrReverbHarp);
NFLeks(n) = filtered_excitation(n,P(freq(n)),freq(n)) : stringloop(freq(n));
N = 24;
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 = 0.81;
beta = 0.5;
B = 0;
L = -10 : ba.db2linear;
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 174.61;
freq(4) = 195.99;
freq(5) = 220.00;
freq(6) = 246.94;
freq(d) = freq(d-7)*(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));
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[2]Reverb/ Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -30 0 17]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[2]Reverb/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -30 0 16]", 0.72,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
e2000a420ebf5da41ac3639f768eea4b399e80c0f970dab91b4033cd223c6dfc | RuolunWeng/ruolunweng.github.io | SCMajDryHarp.dsp | declare name "CMajDryHarp";
declare author "ER";//Adapted from Harpe by Yann Orlarey;
//Modification Grame July 2015
import("stdfaust.lib");
/* =============== DESCRIPTION ================= :
- C Major dry harp
- Left = Lower frequencies/Silence when still
- Front = Resonance (longer notes)
- Back = No Resonance (dry notes)
- Right = Higher frequencies/Fast rhythm
- Rocking = plucking all strings one by one
*/
//-----------------------------------------------
// Harpe : simple string instrument
// (based on Karplus-Strong)
//
//-----------------------------------------------
KEY = 60; // basic midi key
NCY = 15; // note cycle length
CCY = 15; // control cycle length
BPS = 360; // general tempo (ba.beat per sec)
//-------------------------------Harpe----------------------------------
// Harpe is a simple string instrument. Move the "hand" to play the
// various strings
//-----------------------------------------------------------------------
process = vgroup("harpe", h : harpe(C,N,K) :> *(l),*(l))
with {
N = 48; // number of strings
K = 36; // Midi key of first string
h = hslider("[1]Instrument Hand[1] [acc:0 1 -10 0 10]", 24, 0, N, 1) : int: ba.automat(bps, 15, 0.0)
with{
bps = hslider("h:[2]Parameters/[1]Speed[style:knob][acc:0 1 -12 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
//l = vslider("h:parameters/volume [style:knob][unit: dB]", -20, -60, 0, 0.01) : ba.db2linear;
l = -10 : ba.db2linear;
C = 0.5;
};
//----------------------------------Harpe--------------------------------
// USAGE: hand : harpe(C,10,60) : _,_;
// C is the filter coefficient 0..1
// Build a N (10) strings harpe using a pentatonic scale
// based on midi key b (60)
// Each string is triggered by a specific
// position of the "hand"
//-----------------------------------------------------------------------
harpe(C,N,b) = _ <: par(i, N, position(i+1)
:string(C,Major(b).degree2Hz(i), att, lvl)
//envReader(twig,enveloppe):*
: pan((i+0.5)/N) )
:> _,_
with {
att = hslider("h:[2]Parameters/[2]Resonance[style:knob][acc:2 1 -12 0 10]", 5, 0.1, 10, 0.01):min(10):max(0.1);
lvl = 1;
pan(p) = _ <: *(sqrt(1-p)), *(sqrt(p));
position(a,x) = abs(x - a) < 0.5;
};
//----------------------------------Penta-------------------------------
// Pentatonic scale with degree to midi and degree to Hz conversion
// USAGE: Penta(60).degree2midi(3) ==> 67 midikey
// Penta(60).degree2Hz(4) ==> 440 Hz
//-----------------------------------------------------------------------
//---------------------------------- Major -------------------------------
// Major scale.
// From Pentatonic scale with degree to midi and degree to Hz conversion
// USAGE: Penta(60).degree2midi(3) ==> 67 midikey
// Penta(60).degree2Hz(4) ==> 440 Hz
//-----------------------------------------------------------------------
Major(key) = environment {
A4Hz = 440;
degree2midi(0) = key+0;
degree2midi(1) = key+2;
degree2midi(2) = key+4;
degree2midi(3) = key+5;
degree2midi(4) = key+7;
degree2midi(5) = key+9;
degree2midi(6) = key+11;
degree2midi(7) = key+12;
degree2midi(d) = degree2midi(d-8)+12;
degree2Hz(d) = A4Hz*semiton(degree2midi(d)-69) with { semiton(n) = 2.0^(n/12.0); };
};
//----------------------------------String-------------------------------
// A karplus-strong string.
//
// USAGE: string(440Hz, 4s, 1.0, button("play"))
// or button("play") : string(440Hz, 4s, 1.0)
//-----------------------------------------------------------------------
string(coef, freq, t60, level, trig) = no.noise*level
: *(trig : trigger(freq2samples(freq)))
: resonator(freq2samples(freq), att)
with {
resonator(d,a) = (+ : @(d-1)) ~ (average : *(a));
average(x) = (x*(1+coef)+x'*(1-coef))/2;
trigger(n) = upfront : + ~ decay(n) : >(0.0);
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
freq2samples(f) = 44100.0/f;
att = pow(0.001,1.0/(freq*t60)); // attenuation coefficient
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SCMajDryHarp.dsp | faust | Adapted from Harpe by Yann Orlarey;
Modification Grame July 2015
=============== DESCRIPTION ================= :
- C Major dry harp
- Left = Lower frequencies/Silence when still
- Front = Resonance (longer notes)
- Back = No Resonance (dry notes)
- Right = Higher frequencies/Fast rhythm
- Rocking = plucking all strings one by one
-----------------------------------------------
Harpe : simple string instrument
(based on Karplus-Strong)
-----------------------------------------------
basic midi key
note cycle length
control cycle length
general tempo (ba.beat per sec)
-------------------------------Harpe----------------------------------
Harpe is a simple string instrument. Move the "hand" to play the
various strings
-----------------------------------------------------------------------
number of strings
Midi key of first string
l = vslider("h:parameters/volume [style:knob][unit: dB]", -20, -60, 0, 0.01) : ba.db2linear;
----------------------------------Harpe--------------------------------
USAGE: hand : harpe(C,10,60) : _,_;
C is the filter coefficient 0..1
Build a N (10) strings harpe using a pentatonic scale
based on midi key b (60)
Each string is triggered by a specific
position of the "hand"
-----------------------------------------------------------------------
envReader(twig,enveloppe):*
----------------------------------Penta-------------------------------
Pentatonic scale with degree to midi and degree to Hz conversion
USAGE: Penta(60).degree2midi(3) ==> 67 midikey
Penta(60).degree2Hz(4) ==> 440 Hz
-----------------------------------------------------------------------
---------------------------------- Major -------------------------------
Major scale.
From Pentatonic scale with degree to midi and degree to Hz conversion
USAGE: Penta(60).degree2midi(3) ==> 67 midikey
Penta(60).degree2Hz(4) ==> 440 Hz
-----------------------------------------------------------------------
----------------------------------String-------------------------------
A karplus-strong string.
USAGE: string(440Hz, 4s, 1.0, button("play"))
or button("play") : string(440Hz, 4s, 1.0)
-----------------------------------------------------------------------
attenuation coefficient | declare name "CMajDryHarp";
import("stdfaust.lib");
process = vgroup("harpe", h : harpe(C,N,K) :> *(l),*(l))
with {
h = hslider("[1]Instrument Hand[1] [acc:0 1 -10 0 10]", 24, 0, N, 1) : int: ba.automat(bps, 15, 0.0)
with{
bps = hslider("h:[2]Parameters/[1]Speed[style:knob][acc:0 1 -12 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
l = -10 : ba.db2linear;
C = 0.5;
};
harpe(C,N,b) = _ <: par(i, N, position(i+1)
:string(C,Major(b).degree2Hz(i), att, lvl)
: pan((i+0.5)/N) )
:> _,_
with {
att = hslider("h:[2]Parameters/[2]Resonance[style:knob][acc:2 1 -12 0 10]", 5, 0.1, 10, 0.01):min(10):max(0.1);
lvl = 1;
pan(p) = _ <: *(sqrt(1-p)), *(sqrt(p));
position(a,x) = abs(x - a) < 0.5;
};
Major(key) = environment {
A4Hz = 440;
degree2midi(0) = key+0;
degree2midi(1) = key+2;
degree2midi(2) = key+4;
degree2midi(3) = key+5;
degree2midi(4) = key+7;
degree2midi(5) = key+9;
degree2midi(6) = key+11;
degree2midi(7) = key+12;
degree2midi(d) = degree2midi(d-8)+12;
degree2Hz(d) = A4Hz*semiton(degree2midi(d)-69) with { semiton(n) = 2.0^(n/12.0); };
};
string(coef, freq, t60, level, trig) = no.noise*level
: *(trig : trigger(freq2samples(freq)))
: resonator(freq2samples(freq), att)
with {
resonator(d,a) = (+ : @(d-1)) ~ (average : *(a));
average(x) = (x*(1+coef)+x'*(1-coef))/2;
trigger(n) = upfront : + ~ decay(n) : >(0.0);
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
freq2samples(f) = 44100.0/f;
};
|
ff6b8c9bc10d2bd58e0d7474c936bcaf22a22d7f538bafa4d43872784486dca6 | RuolunWeng/ruolunweng.github.io | Meow.dsp | declare name "Meow";
declare description "WaveGuide Brass instrument from STK";
declare author "ER"; //From Brass by Romain Michon ([email protected]);
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- Triple brass mimicking mewing cats
- Left = silence.
- Rocking from top Left to Front/Right : one cat mewing
- Rotation = two cats in turn
- Back/Front = Tutti
*/
//==================== INSTRUMENT =======================
process = vgroup("MEOW", par(i, 3, brass(i)) :> crybb);
brass(n) = (borePressure <: deltaPressure(pressure(n)),_ :
(lipFilter(freq(n)) <: *(mouthPressure(pressure(n))),(1-_)),_ : _, * :> + :
fi.dcblocker) ~ (boreDelay(freq(n)))
*(gain(n)): fi.lowpass((n+1),((n+1)*1500));
//==================== GUI SPECIFICATION ================
//gate = checkbox(" Play[1]");
gate = hslider(" ON/OFF", 0, 0, 1, 1):int;
freq(0) = hslider("h:Instrument/v:Frequencies/Frequency 1 [unit:Hz][acc:1 0 -10 0 10][tooltip:Tone frequency]",370,280,380, 0.01):si.smooth(0.999);
freq(1) = hslider("h:Instrument/v:Frequencies/Frequency 2 [unit:Hz][acc:0 0 -10 0 10][tooltip:Tone frequency]",440,380,550,0.01):si.smooth(0.999);
freq(2) = hslider("h:Instrument/v:Frequencies/Frequency 3 [unit:Hz][acc:2 1 -10 0 12][tooltip:Tone frequency]",587.32,550,700,0.01):si.smooth(0.999);
gain(0) = hslider("h:Instrument/v:Gain/Volume 1 [style:knob][acc:1 1 -12 0 12][tooltip:Gain (value between 0 and 1)]",0.5,0,1,0.01);
gain(1) = hslider("h:Instrument/v:Gain/Volume 2 [style:knob][acc:0 1 -12 0 12][tooltip:Gain (value between 0 and 1)]",0.5,0,1,0.01);
gain(2) = hslider("h:Instrument/v:Gain/Volume 3 [style:knob][acc:2 0 -12 0 10][tooltip:Gain (value between 0 and 1)]",0.5,0,0.5,0.01);
pressure(0) = 0.37;
pressure(1) = 0.68;
pressure(2) = 1.0;
lipTension = 0.780;
slideLength = 0.041;
vibratoFreq = 6;
vibratoGain = 0.05;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.1;
envelopeAttack = 0.01;
envelopeDecay = 0.001;
envelopeRelease = 2;
//==================== SIGNAL PROCESSING ================
crybb = ve.crybaby(wah) with {
wah = hslider("Wah Wah[acc:0 0 -15 10 0]", 0.5,0,1,0.01) : ba.automat(360, 15, 0.0);
};
//--------- Synthesis parameters computing and functions declaration -----------
//lips are simulated by a biquad filter whose output is squared and hard-clipped, instrument.bandPassH and instrument.saturationPos are declared in instrument.lib
lipFilterFrequency(f) = f*pow(4,(2*lipTension)-1);
lipFilter(f) = *(0.03) : instrument.bandPassH(lipFilterFrequency(f),0.997) <: * : instrument.saturationPos;
//de.delay times in number of samples
slideTarget(f) = ((ma.SR/f)*2 + 3)*(0.5 + slideLength);
boreDelay(f) = de.fdelay(4096,slideTarget(f));
//----------------------- Algorithm implementation ----------------------------
//vibrato
vibrato = vibratoGain*os.osc(vibratoFreq)*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate);
//envelope (Attack / Decay / Sustain / Release), breath pressure and vibrato
breathPressure(p) = p*en.adsr(envelopeAttack,envelopeDecay,1,envelopeRelease,gate) + vibrato;
mouthPressure(p) = 0.3*breathPressure(p);
//scale the de.delay feedback
borePressure = *(0.85);
//differencial presure
deltaPressure(p) = mouthPressure(p) - _;
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/Meow.dsp | faust | From Brass by Romain Michon ([email protected]);
=============== DESCRIPTION ================= :
- Triple brass mimicking mewing cats
- Left = silence.
- Rocking from top Left to Front/Right : one cat mewing
- Rotation = two cats in turn
- Back/Front = Tutti
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
gate = checkbox(" Play[1]");
==================== SIGNAL PROCESSING ================
--------- Synthesis parameters computing and functions declaration -----------
lips are simulated by a biquad filter whose output is squared and hard-clipped, instrument.bandPassH and instrument.saturationPos are declared in instrument.lib
de.delay times in number of samples
----------------------- Algorithm implementation ----------------------------
vibrato
envelope (Attack / Decay / Sustain / Release), breath pressure and vibrato
scale the de.delay feedback
differencial presure | declare name "Meow";
declare description "WaveGuide Brass instrument from STK";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("MEOW", par(i, 3, brass(i)) :> crybb);
brass(n) = (borePressure <: deltaPressure(pressure(n)),_ :
(lipFilter(freq(n)) <: *(mouthPressure(pressure(n))),(1-_)),_ : _, * :> + :
fi.dcblocker) ~ (boreDelay(freq(n)))
*(gain(n)): fi.lowpass((n+1),((n+1)*1500));
gate = hslider(" ON/OFF", 0, 0, 1, 1):int;
freq(0) = hslider("h:Instrument/v:Frequencies/Frequency 1 [unit:Hz][acc:1 0 -10 0 10][tooltip:Tone frequency]",370,280,380, 0.01):si.smooth(0.999);
freq(1) = hslider("h:Instrument/v:Frequencies/Frequency 2 [unit:Hz][acc:0 0 -10 0 10][tooltip:Tone frequency]",440,380,550,0.01):si.smooth(0.999);
freq(2) = hslider("h:Instrument/v:Frequencies/Frequency 3 [unit:Hz][acc:2 1 -10 0 12][tooltip:Tone frequency]",587.32,550,700,0.01):si.smooth(0.999);
gain(0) = hslider("h:Instrument/v:Gain/Volume 1 [style:knob][acc:1 1 -12 0 12][tooltip:Gain (value between 0 and 1)]",0.5,0,1,0.01);
gain(1) = hslider("h:Instrument/v:Gain/Volume 2 [style:knob][acc:0 1 -12 0 12][tooltip:Gain (value between 0 and 1)]",0.5,0,1,0.01);
gain(2) = hslider("h:Instrument/v:Gain/Volume 3 [style:knob][acc:2 0 -12 0 10][tooltip:Gain (value between 0 and 1)]",0.5,0,0.5,0.01);
pressure(0) = 0.37;
pressure(1) = 0.68;
pressure(2) = 1.0;
lipTension = 0.780;
slideLength = 0.041;
vibratoFreq = 6;
vibratoGain = 0.05;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.1;
envelopeAttack = 0.01;
envelopeDecay = 0.001;
envelopeRelease = 2;
crybb = ve.crybaby(wah) with {
wah = hslider("Wah Wah[acc:0 0 -15 10 0]", 0.5,0,1,0.01) : ba.automat(360, 15, 0.0);
};
lipFilterFrequency(f) = f*pow(4,(2*lipTension)-1);
lipFilter(f) = *(0.03) : instrument.bandPassH(lipFilterFrequency(f),0.997) <: * : instrument.saturationPos;
slideTarget(f) = ((ma.SR/f)*2 + 3)*(0.5 + slideLength);
boreDelay(f) = de.fdelay(4096,slideTarget(f));
vibrato = vibratoGain*os.osc(vibratoFreq)*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate);
breathPressure(p) = p*en.adsr(envelopeAttack,envelopeDecay,1,envelopeRelease,gate) + vibrato;
mouthPressure(p) = 0.3*breathPressure(p);
borePressure = *(0.85);
deltaPressure(p) = mouthPressure(p) - _;
|
938917b089803ede9918934114b1f998e087b2815a55ce9494bc9a46f5f51cb8 | RuolunWeng/ruolunweng.github.io | PentatonicFlute.dsp | declare name "Pentatonic Flute";
declare description "Nonlinear WaveGuide Flute";
declare author "ER";// Adapted from "Flute" by Romain Michon ([email protected]);
/* =============== DESCRIPTION ================= :
- Pentatonic flute
- Rocking = playing all notes from low to high frequencies
- Left = Silence/Slow rhythm
- Right = Fast rhythm
- Head = Reverberation
- Front = long notes
- Back = short notes
*/
import("stdfaust.lib");
instrument = library("instruments.lib");
//==================== INSTRUMENT =======================
flute(n) = (_ <: (flow(trigger(n)) + *(feedBack1) : embouchureDelay(freq(n)): poly) + *(feedBack2) : reflexionFilter)~(boreDelay(freq(n))) : *(env2(trigger(n)))*gain:_;
process = par(i, N, flute(i)):>_<: instrReverbFlute;
//==================== GUI SPECIFICATION ================
vibratoFreq = 2.5;
env1Attack = 0.06;
env1Release = 1;
//-------------------- Non-Variable Parameters -----------
N = 14;
gain = 1;
pressure = 0.9;
breathAmp = 0.01;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.1;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
//----------------------- Frequency Table --------------------
freq(0) = 184.99;
freq(1) = 207.65;
freq(2) = 233.08;
freq(3) = 277.18;
freq(4) = 311.12;
freq(d) = freq(d-5)*2;
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//Loops feedbacks gains
feedBack1 = 0.4;
feedBack2 = 0.4;
//Delay Lines
embouchureDelayLength(f) = (ma.SR/f)/2-2;
boreDelayLength(f) = ma.SR/f-2;
embouchureDelay(f) = de.fdelay(4096,embouchureDelayLength(f));
boreDelay(f) = de.fdelay(4096,boreDelayLength(f));
//Polinomial
poly = _ <: _ - _*_*_;
//jet filter is a lowwpass filter (declared in filter.lib)
reflexionFilter = fi.lowpass(1,2000);
//----------------------- Algorithm implementation ----------------------------
//Pressure envelope
env1(t) = en.adsr(env1Attack,env1Decay,0.9,env1Release,(t | pressureEnvelope))*pressure*1.1;
//Global envelope
env2(t) = en.asr(env2Attack,1,env2Release,t)*0.5;
//Vibrato Envelope
vibratoEnvelope(t) = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*vibratoGain;
vibrato(t) = os.osc(vibratoFreq)*vibratoEnvelope(t);
breath(t) = no.noise*env1(t);
flow(t) = env1(t) + breath(t)*breathAmp + vibrato(t);
//------------------------- Enveloppe Trigger --------------------------------------------
trigger(n) = position(n): trig
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
noteDuration = hslider("[3]Note Duration[unit:s][style:knob][acc:2 1 -10 0 10]", 0.166, 0.1, 0.25, 0.01)*44100 : min(11025) : max(4410):int;
trig = upfront : release(noteDuration) : >(0.0);
};
position(n) = abs(hand - n) < 0.5;
hand = hslider("[1]Instrument Hand[acc:0 1 -10 0 10]", 7, 0, N, 1):int: ba.automat(bps, 15, 0.0)// => gate
with{
bps = hslider("[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
//----------------------- INSTRREVERB ----------------------------
// GUI for re.zita_rev1_stereo from effect.lib
//
// USAGE:
// _,_ : instrRerveb
instrReverbFlute = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[4]Reverb/[1]Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[4]Reverb/[2]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.5,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/PentatonicFlute.dsp | faust | Adapted from "Flute" by Romain Michon ([email protected]);
=============== DESCRIPTION ================= :
- Pentatonic flute
- Rocking = playing all notes from low to high frequencies
- Left = Silence/Slow rhythm
- Right = Fast rhythm
- Head = Reverberation
- Front = long notes
- Back = short notes
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
-------------------- Non-Variable Parameters -----------
----------------------- Frequency Table --------------------
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
Loops feedbacks gains
Delay Lines
Polinomial
jet filter is a lowwpass filter (declared in filter.lib)
----------------------- Algorithm implementation ----------------------------
Pressure envelope
Global envelope
Vibrato Envelope
------------------------- Enveloppe Trigger --------------------------------------------
=> gate
----------------------- INSTRREVERB ----------------------------
GUI for re.zita_rev1_stereo from effect.lib
USAGE:
_,_ : instrRerveb | declare name "Pentatonic Flute";
declare description "Nonlinear WaveGuide Flute";
import("stdfaust.lib");
instrument = library("instruments.lib");
flute(n) = (_ <: (flow(trigger(n)) + *(feedBack1) : embouchureDelay(freq(n)): poly) + *(feedBack2) : reflexionFilter)~(boreDelay(freq(n))) : *(env2(trigger(n)))*gain:_;
process = par(i, N, flute(i)):>_<: instrReverbFlute;
vibratoFreq = 2.5;
env1Attack = 0.06;
env1Release = 1;
N = 14;
gain = 1;
pressure = 0.9;
breathAmp = 0.01;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.1;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
freq(0) = 184.99;
freq(1) = 207.65;
freq(2) = 233.08;
freq(3) = 277.18;
freq(4) = 311.12;
freq(d) = freq(d-5)*2;
feedBack1 = 0.4;
feedBack2 = 0.4;
embouchureDelayLength(f) = (ma.SR/f)/2-2;
boreDelayLength(f) = ma.SR/f-2;
embouchureDelay(f) = de.fdelay(4096,embouchureDelayLength(f));
boreDelay(f) = de.fdelay(4096,boreDelayLength(f));
poly = _ <: _ - _*_*_;
reflexionFilter = fi.lowpass(1,2000);
env1(t) = en.adsr(env1Attack,env1Decay,0.9,env1Release,(t | pressureEnvelope))*pressure*1.1;
env2(t) = en.asr(env2Attack,1,env2Release,t)*0.5;
vibratoEnvelope(t) = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*vibratoGain;
vibrato(t) = os.osc(vibratoFreq)*vibratoEnvelope(t);
breath(t) = no.noise*env1(t);
flow(t) = env1(t) + breath(t)*breathAmp + vibrato(t);
trigger(n) = position(n): trig
with {
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
noteDuration = hslider("[3]Note Duration[unit:s][style:knob][acc:2 1 -10 0 10]", 0.166, 0.1, 0.25, 0.01)*44100 : min(11025) : max(4410):int;
trig = upfront : release(noteDuration) : >(0.0);
};
position(n) = abs(hand - n) < 0.5;
with{
bps = hslider("[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
instrReverbFlute = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[4]Reverb/[1]Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[4]Reverb/[2]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.5,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
d30fe32cf8e4cf92e9ffca1e860104243cd08883949470a498ead7647c07d210 | RuolunWeng/ruolunweng.github.io | CMajFlute.dsp | declare name "C Major Flute";
declare author "ER";// Adapted from "Nonlinear WaveGuide Flute" by Romain Michon ([email protected])";
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- C Major flute
- Rocking = playing all notes from low to high frequencies
- Left = Silence/Slow rhythm
- Right = Fast rhythm
- Head = Reverberation
- Front = long notes
- Back = short notes
*/
//==================== INSTRUMENT =======================
flute(n) = (_ <: (flow(trigger(n)) + *(feedBack1) : embouchureDelay(freq(n)): poly) + *(feedBack2) : reflexionFilter)~(boreDelay(freq(n))) : *(env2(trigger(n)))*gain:_;
process = vgroup("C Maj Flute", par(i, N, flute(i)):>_<: instrReverbFlute);
//==================== GUI SPECIFICATION ================
vibratoFreq = 2.5;
env1Attack = 0.06;
env1Release = 1;
//-------------------- Non-Variable Parameters -----------
N = 17;
gain = 1;
pressure = 0.9;
breathAmp = 0.01;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.1;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
//----------------------- Frequency Table --------------------
freq(0) = 261.62;
freq(1) = 293.66;
freq(2) = 329.62;
freq(3) = 349.22;
freq(4) = 391.99;
freq(5) = 440.00;
freq(6) = 493.88;
freq(d) = freq(d-7)*2;
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//Loops feedbacks gains
feedBack1 = 0.4;
feedBack2 = 0.4;
//Delay Lines
embouchureDelayLength(f) = (ma.SR/f)/2-2;
boreDelayLength(f) = ma.SR/f-2;
embouchureDelay(f) = de.fdelay(4096,embouchureDelayLength(f));
boreDelay(f) = de.fdelay(4096,boreDelayLength(f));
//Polinomial
poly = _ <: _ - _*_*_;
//jet filter is a lowwpass filter (declared in filter.lib)
reflexionFilter = fi.lowpass(1,2000);
//----------------------- Algorithm implementation ----------------------------
//Pressure envelope
env1(t) = en.adsr(env1Attack,env1Decay,0.9,env1Release,(t | pressureEnvelope))*pressure*1.1;
//Global envelope
env2(t) = en.asr(env2Attack,1,env2Release,t)*0.5;
//Vibrato Envelope
vibratoEnvelope(t) = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*vibratoGain;
vibrato(t) = os.osc(vibratoFreq)*vibratoEnvelope(t);
breath(t) = no.noise*env1(t);
flow(t) = env1(t) + breath(t)*breathAmp + vibrato(t);
//------------------------- Enveloppe Trigger --------------------------------------------
trigger(n) = position(n): trig
with{
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
noteDuration = hslider("h:[1]/[3]Note Duration[unit:s][style:knob][acc:2 1 -10 0 10]", 0.166, 0.1, 0.25, 0.01)*44100 : min(11025) : max(4410):int;
trig = upfront : release(noteDuration) : >(0.0);
};
position(n) = abs(hand - n) < 0.5;
hand = hslider("h:[1]/[1]Instrument Hand[acc:0 1 -12 0 10]", 9, 0, N, 1): ba.automat(bps, 15, 0.0)// => gate
with {
bps = hslider("h:[1]/[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
//----------------------- INSTRREVERB ----------------------------
// GUI for re.zita_rev1_stereo from effect.lib
//
// USAGE:
// _,_ : instrRerveb
instrReverbFlute = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[2]Reverb/[1]Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[2]Reverb/[2]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.5,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/CMajFlute.dsp | faust | Adapted from "Nonlinear WaveGuide Flute" by Romain Michon ([email protected])";
=============== DESCRIPTION ================= :
- C Major flute
- Rocking = playing all notes from low to high frequencies
- Left = Silence/Slow rhythm
- Right = Fast rhythm
- Head = Reverberation
- Front = long notes
- Back = short notes
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
-------------------- Non-Variable Parameters -----------
----------------------- Frequency Table --------------------
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
Loops feedbacks gains
Delay Lines
Polinomial
jet filter is a lowwpass filter (declared in filter.lib)
----------------------- Algorithm implementation ----------------------------
Pressure envelope
Global envelope
Vibrato Envelope
------------------------- Enveloppe Trigger --------------------------------------------
=> gate
----------------------- INSTRREVERB ----------------------------
GUI for re.zita_rev1_stereo from effect.lib
USAGE:
_,_ : instrRerveb | declare name "C Major Flute";
import("stdfaust.lib");
instrument = library("instruments.lib");
flute(n) = (_ <: (flow(trigger(n)) + *(feedBack1) : embouchureDelay(freq(n)): poly) + *(feedBack2) : reflexionFilter)~(boreDelay(freq(n))) : *(env2(trigger(n)))*gain:_;
process = vgroup("C Maj Flute", par(i, N, flute(i)):>_<: instrReverbFlute);
vibratoFreq = 2.5;
env1Attack = 0.06;
env1Release = 1;
N = 17;
gain = 1;
pressure = 0.9;
breathAmp = 0.01;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.1;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
freq(0) = 261.62;
freq(1) = 293.66;
freq(2) = 329.62;
freq(3) = 349.22;
freq(4) = 391.99;
freq(5) = 440.00;
freq(6) = 493.88;
freq(d) = freq(d-7)*2;
feedBack1 = 0.4;
feedBack2 = 0.4;
embouchureDelayLength(f) = (ma.SR/f)/2-2;
boreDelayLength(f) = ma.SR/f-2;
embouchureDelay(f) = de.fdelay(4096,embouchureDelayLength(f));
boreDelay(f) = de.fdelay(4096,boreDelayLength(f));
poly = _ <: _ - _*_*_;
reflexionFilter = fi.lowpass(1,2000);
env1(t) = en.adsr(env1Attack,env1Decay,0.9,env1Release,(t | pressureEnvelope))*pressure*1.1;
env2(t) = en.asr(env2Attack,1,env2Release,t)*0.5;
vibratoEnvelope(t) = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*vibratoGain;
vibrato(t) = os.osc(vibratoFreq)*vibratoEnvelope(t);
breath(t) = no.noise*env1(t);
flow(t) = env1(t) + breath(t)*breathAmp + vibrato(t);
trigger(n) = position(n): trig
with{
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
noteDuration = hslider("h:[1]/[3]Note Duration[unit:s][style:knob][acc:2 1 -10 0 10]", 0.166, 0.1, 0.25, 0.01)*44100 : min(11025) : max(4410):int;
trig = upfront : release(noteDuration) : >(0.0);
};
position(n) = abs(hand - n) < 0.5;
with {
bps = hslider("h:[1]/[2]Speed[style:knob][acc:0 1 -10 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
instrReverbFlute = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[2]Reverb/[1]Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[2]Reverb/[2]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.5,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
dc9431b93dfaa663d2add99dd2b016d1a627ea5ee7bcf579443d9b454f985dbe | RuolunWeng/ruolunweng.github.io | ChromaticSoftHarp.dsp | declare name "Chromatic Soft Harp";
declare author "ER";//Adapted from Nonlinear EKS by Julius Smith and Romain Michon;
declare reference "http://ccrma.stanford.edu/~jos/pasp/vegf.html";
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION =================
- Reverberated soft chromatic harp
- Left = Lower frequencies/Silence when still
- Front = Resonance
- Back = No resonance
- Right = Higher frequencies/Fast rhythm
- Head = Reverberation
- Rocking = plucking all strings one by one
*/
//==================== INSTRUMENT =======================
process = par(i, N, NFLeks(i)):>_<: instrReverbHarp : *(vol),*(vol);
NFLeks(n) = filtered_excitation(n+1,P(freq(n)),freq(n)) : stringloop(freq(n));
//==================== GUI SPECIFICATION ================
// standard MIDI voice parameters:
// NOTE: The labels MUST be "freq", "gain", and "gate" for faust2pd
N = 24;
hand = hslider("h:[1]/Instrument Hand[acc:0 1 -10 0 10]", 12, 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;
vol = 2;
pickangle = 0.9;
beta = 0.5;
// String decay time in seconds:
t60 = hslider("h:[2]Reverb/[1]Resonance (InstrReverb)[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) = 130.81;
freq(1) = 138.59;
freq(2) = 146.83;
freq(3) = 155.56;
freq(4) = 164.81;
freq(5) = 174.61;
freq(6) = 184.99;
freq(7) = 195.99;
freq(8) = 207.65;
freq(9) = 220.00;
freq(10) = 233.08;
freq(11) = 246.94;
freq(d) = freq(d-12)*(2);
//==================== SIGNAL PROCESSING ================
//----------------------- noiseburst -------------------------
// White no.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);
};
//nlfOrder = 6;
P(f) = ma.SR/f ; // fundamental period in samples
Pmax = 4096; // maximum P (for de.delay-line allocation)
ppdel(f) = beta*P(f); // pick position de.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));
//================================= REVERB ==============================
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[2]Reverb/[2]Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -30 0 17]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[2]Reverb/[3]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -30 0 16]", 0.72,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/ChromaticSoftHarp.dsp | faust | Adapted from Nonlinear EKS by Julius Smith and Romain Michon;
=============== DESCRIPTION =================
- Reverberated soft chromatic harp
- Left = Lower frequencies/Silence when still
- Front = Resonance
- Back = No resonance
- Right = Higher frequencies/Fast rhythm
- Head = Reverberation
- Rocking = plucking all strings one by one
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
standard MIDI voice parameters:
NOTE: The labels MUST be "freq", "gain", and "gate" for faust2pd
=> gate
String decay time in seconds:
-60db decay time (sec)
---------------------------------- FREQUENCY TABLE ---------------------------
==================== SIGNAL PROCESSING ================
----------------------- noiseburst -------------------------
White no.noise burst (adapted from Faust's karplus.dsp example)
Requires music.lib (for no.noise)
nlfOrder = 6;
fundamental period in samples
maximum P (for de.delay-line allocation)
pick position de.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
================================= REVERB ============================== | declare name "Chromatic Soft Harp";
declare reference "http://ccrma.stanford.edu/~jos/pasp/vegf.html";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = par(i, N, NFLeks(i)):>_<: instrReverbHarp : *(vol),*(vol);
NFLeks(n) = filtered_excitation(n+1,P(freq(n)),freq(n)) : stringloop(freq(n));
N = 24;
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;
vol = 2;
pickangle = 0.9;
beta = 0.5;
B = 0;
L = -10 : ba.db2linear;
freq(0) = 130.81;
freq(1) = 138.59;
freq(2) = 146.83;
freq(3) = 155.56;
freq(4) = 164.81;
freq(5) = 174.61;
freq(6) = 184.99;
freq(7) = 195.99;
freq(8) = 207.65;
freq(9) = 220.00;
freq(10) = 233.08;
freq(11) = 246.94;
freq(d) = freq(d-12)*(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));
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[2]Reverb/[2]Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -30 0 17]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[2]Reverb/[3]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -30 0 16]", 0.72,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
d5e406ea04e10deb4455bbb03c55f619839061abb64bcdd1adc4d8f1a267690c | RuolunWeng/ruolunweng.github.io | SFlute.dsp | declare name "Flute";
declare description "Nonlinear WaveGuide Flute";
declare author "Romain Michon ([email protected])";
declare copyright "Romain Michon";
declare version "1.0";
declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
declare description "A simple flute based on Smith algorythm: https://ccrma.stanford.edu/~jos/pasp/Flutes_Recorders_Pipe_Organs.html";
//Modifications GRAME July 2015
/* =========== DESCRITPION ===========
- Flute
- Turn ON flute (0=OFF, 1=ON)
- Head = High frequencies/ Silence
- Bottom = Low frequencies
- Left = No vibrato
- Right = Fast vibrato
- Front = Full sound
- Back = Breathy sound
*/
import("stdfaust.lib");
instrument = library("instruments.lib");
//==================== INSTRUMENT =======================
flute = (_ <: (flow + *(feedBack1) : embouchureDelay: poly) + *(feedBack2) : reflexionFilter)~(boreDelay) : NLFM : *(env2)*gain:_;
process = flute;
//==================== GUI SPECIFICATION ================
freq = hslider("[1]Frequency[acc:1 1 -10 0 10]", 440,247,1200,1):si.smooth(0.999);
pressure = hslider("[2]Pressure[style:knob][acc:1 0 -10 0 10]", 0.96, 0.2, 0.99, 0.01):si.smooth(0.999):min(0.99):max(0.2);
breathAmp = hslider("[3]Breath Noise[style:knob][acc:2 0 -10 0 10]", 0.02, 0.01, 0.2, 0.01):si.smooth(0.999):min(0.2):max(0.01);
gate = hslider("[0]ON/OFF (ASR Envelope)",0,0,1,1);
vibratoFreq = hslider("[4]Vibrato Freq (Vibrato Envelope)[style:knob][unit:Hz][acc:0 1 -10 0 10]", 4,0.5,8,0.1);
env1Attack = 0.1;//hslider("h:Parameters/Press_Env_Attack[unit:s][style:knob][acc:1 0 -10 0 10][tooltip:Pressure envelope attack duration]",0.05,0.05,0.2,0.01);
//-------------------- Non-Variable Parameters -----------
gain = 1;
typeModulation = 0;
nonLinearity = 0;
frequencyMod = 220;
nonLinAttack = 0.1;
vibratoGain = 0.05;
vibratoBegin = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
env1Release = 0.5;
//==================== SIGNAL PROCESSING ================
//----------------------- Nonlinear filter ----------------------------
//nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
//nonlinear filter order
nlfOrder = 6;
//attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
envelopeMod = en.asr(nonLinAttack,1,0.1,gate);
//nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
//for using it with waveguide instruments
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//Loops feedbacks gains
feedBack1 = 0.4;
feedBack2 = 0.4;
//Delay Lines
embouchureDelayLength = (ma.SR/freq)/2-2;
boreDelayLength = ma.SR/freq-2;
embouchureDelay = de.fdelay(4096,embouchureDelayLength);
boreDelay = de.fdelay(4096,boreDelayLength);
//Polinomial
poly = _ <: _ - _*_*_;
//jet filter is a lowwpass filter (declared in filter.lib)
reflexionFilter = fi.lowpass(1,2000);
//----------------------- Algorithm implementation ----------------------------
//Pressure envelope
env1 = en.adsr(env1Attack,env1Decay,0.9,env1Release,(gate | pressureEnvelope))*pressure*1.1;
//Global envelope
env2 = en.asr(env2Attack,1,env2Release,gate)*0.5;
//Vibrato Envelope
vibratoEnvelope = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*vibratoGain;
vibrato = os.osc(vibratoFreq)*vibratoEnvelope;
breath = no.noise*env1;
flow = env1 + breath*breathAmp + vibrato;
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SFlute.dsp | faust | Synthesis Tool Kit 4.3 (MIT style license);
Modifications GRAME July 2015
=========== DESCRITPION ===========
- Flute
- Turn ON flute (0=OFF, 1=ON)
- Head = High frequencies/ Silence
- Bottom = Low frequencies
- Left = No vibrato
- Right = Fast vibrato
- Front = Full sound
- Back = Breathy sound
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
hslider("h:Parameters/Press_Env_Attack[unit:s][style:knob][acc:1 0 -10 0 10][tooltip:Pressure envelope attack duration]",0.05,0.05,0.2,0.01);
-------------------- Non-Variable Parameters -----------
==================== SIGNAL PROCESSING ================
----------------------- Nonlinear filter ----------------------------
nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
nonlinear filter order
attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
for using it with waveguide instruments
----------------------- Synthesis parameters computing and functions declaration ----------------------------
Loops feedbacks gains
Delay Lines
Polinomial
jet filter is a lowwpass filter (declared in filter.lib)
----------------------- Algorithm implementation ----------------------------
Pressure envelope
Global envelope
Vibrato Envelope | declare name "Flute";
declare description "Nonlinear WaveGuide Flute";
declare author "Romain Michon ([email protected])";
declare copyright "Romain Michon";
declare version "1.0";
declare description "A simple flute based on Smith algorythm: https://ccrma.stanford.edu/~jos/pasp/Flutes_Recorders_Pipe_Organs.html";
import("stdfaust.lib");
instrument = library("instruments.lib");
flute = (_ <: (flow + *(feedBack1) : embouchureDelay: poly) + *(feedBack2) : reflexionFilter)~(boreDelay) : NLFM : *(env2)*gain:_;
process = flute;
freq = hslider("[1]Frequency[acc:1 1 -10 0 10]", 440,247,1200,1):si.smooth(0.999);
pressure = hslider("[2]Pressure[style:knob][acc:1 0 -10 0 10]", 0.96, 0.2, 0.99, 0.01):si.smooth(0.999):min(0.99):max(0.2);
breathAmp = hslider("[3]Breath Noise[style:knob][acc:2 0 -10 0 10]", 0.02, 0.01, 0.2, 0.01):si.smooth(0.999):min(0.2):max(0.01);
gate = hslider("[0]ON/OFF (ASR Envelope)",0,0,1,1);
vibratoFreq = hslider("[4]Vibrato Freq (Vibrato Envelope)[style:knob][unit:Hz][acc:0 1 -10 0 10]", 4,0.5,8,0.1);
gain = 1;
typeModulation = 0;
nonLinearity = 0;
frequencyMod = 220;
nonLinAttack = 0.1;
vibratoGain = 0.05;
vibratoBegin = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
env1Release = 0.5;
nlfOrder = 6;
envelopeMod = en.asr(nonLinAttack,1,0.1,gate);
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
feedBack1 = 0.4;
feedBack2 = 0.4;
embouchureDelayLength = (ma.SR/freq)/2-2;
boreDelayLength = ma.SR/freq-2;
embouchureDelay = de.fdelay(4096,embouchureDelayLength);
boreDelay = de.fdelay(4096,boreDelayLength);
poly = _ <: _ - _*_*_;
reflexionFilter = fi.lowpass(1,2000);
env1 = en.adsr(env1Attack,env1Decay,0.9,env1Release,(gate | pressureEnvelope))*pressure*1.1;
env2 = en.asr(env2Attack,1,env2Release,gate)*0.5;
vibratoEnvelope = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*vibratoGain;
vibrato = os.osc(vibratoFreq)*vibratoEnvelope;
breath = no.noise*env1;
flow = env1 + breath*breathAmp + vibrato;
|
b3cb562dddea4f176164b1a77c9bccc5e157fb1876aea9af320ac43621df9aa0 | RuolunWeng/ruolunweng.github.io | SRandomFlute.dsp | declare name "Random Flute";
declare author "ER";//Adapted from "Nonlinear WaveGuide Flute" by Romain Michon ([email protected]);
import("stdfaust.lib");
instrument = library("instruments.lib");
/* ============== DESCRIPTION ================
- Random frequency flute
- Left = Slow rhythm/long notes/silence
- Right = Fast rhythm/short note
*/
//==================== INSTRUMENT =======================
flute = (_ <: (flow + *(feedBack1) : embouchureDelay: poly) + *(feedBack2) : reflexionFilter)~(boreDelay) : NLFM : *(env2)*gain:_;
process = flute;
//==================== GUI SPECIFICATION ================
pressure = 1;
breathAmp = hslider("h:[3]Parameters/Breath Noise[style:knob][acc:0 1 -10 0 10]", 0.02, 0.01, 0.05, 0.0001):si.smooth(0.999):min(0.05):max(0.01);
gate = pulsaflute.gate;
vibratoFreq = 5;
env1Attack = 0.05;
//--------------------------- Random Frequency ---------------------------
freq = gate : randfreq : si.smooth(0.99) : fi.lowpass (1, 3000);
randfreq(g) = no.noise : sampleAndhold(sahgate(g))*(1500)+(100)
with {
sampleAndhold(t) = select2(t) ~_;
sahgate(g) = g : upfront : counter -(3) <=(0);
upfront(x) = abs(x-x')>0.5;
counter(g) = (+(1):*(1-g))~_;
};
//----------------------- Pulsar --------------------------------------
pulsaflute = environment {
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
fade = (0.5);
speed = hslider ("h:[1]Pulse/[1]Speed (Granulator)[unit:Hz][style:knob][acc:0 1 -10 0 10]", 3,1,6,0.0001):fi.lowpass(1,1);
proba = hslider ("h:[1]Pulse/[2]Probability (Granulator)[unit:%][style:knob][acc:0 1 -10 0 10]", 88,60,100,1) *(0.01):fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
//-------------------- Non-Variable Parameters -----------
N = 27;
gain = 1;
typeModulation = 0;
nonLinearity = 0;
frequencyMod = 220;
nonLinAttack = 0.1;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
env1Release = 0.5;
//==================== SIGNAL PROCESSING ================
//----------------------- Nonlinear filter ----------------------------
//nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
//nonlinear filter order
nlfOrder = 6;
//attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
envelopeMod = en.asr(nonLinAttack,1,0.1,gate);
//nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
//for using it with waveguide instruments
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//Loops feedbacks gains
feedBack1 = 0.4;
feedBack2 = 0.4;
//Delay Lines
embouchureDelayLength = (ma.SR/freq)/2-2;
boreDelayLength = ma.SR/freq-2;
embouchureDelay = de.fdelay(4096,embouchureDelayLength);
boreDelay = de.fdelay(4096,boreDelayLength);
//Polinomial
poly = _ <: _ - _*_*_;
//jet filter is a lowwpass filter (declared in filter.lib)
reflexionFilter = fi.lowpass(1,2000);
//----------------------- Algorithm implementation ----------------------------
//Pressure envelope
env1 = en.adsr(env1Attack,env1Decay,0.9,env1Release,(gate | pressureEnvelope))*pressure*1.1;
//Global envelope
env2 = en.asr(env2Attack,1,env2Release,gate)*0.5;
//Vibrato Envelope
vibratoEnvelope = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*vibratoGain;
vibrato = os.osc(vibratoFreq)*vibratoEnvelope;
breath = no.noise*env1;
flow = env1 + breath*breathAmp + vibrato;
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SRandomFlute.dsp | faust | Adapted from "Nonlinear WaveGuide Flute" by Romain Michon ([email protected]);
============== DESCRIPTION ================
- Random frequency flute
- Left = Slow rhythm/long notes/silence
- Right = Fast rhythm/short note
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
--------------------------- Random Frequency ---------------------------
----------------------- Pulsar --------------------------------------
-------------------- Non-Variable Parameters -----------
==================== SIGNAL PROCESSING ================
----------------------- Nonlinear filter ----------------------------
nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
nonlinear filter order
attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
for using it with waveguide instruments
----------------------- Synthesis parameters computing and functions declaration ----------------------------
Loops feedbacks gains
Delay Lines
Polinomial
jet filter is a lowwpass filter (declared in filter.lib)
----------------------- Algorithm implementation ----------------------------
Pressure envelope
Global envelope
Vibrato Envelope | declare name "Random Flute";
import("stdfaust.lib");
instrument = library("instruments.lib");
flute = (_ <: (flow + *(feedBack1) : embouchureDelay: poly) + *(feedBack2) : reflexionFilter)~(boreDelay) : NLFM : *(env2)*gain:_;
process = flute;
pressure = 1;
breathAmp = hslider("h:[3]Parameters/Breath Noise[style:knob][acc:0 1 -10 0 10]", 0.02, 0.01, 0.05, 0.0001):si.smooth(0.999):min(0.05):max(0.01);
gate = pulsaflute.gate;
vibratoFreq = 5;
env1Attack = 0.05;
freq = gate : randfreq : si.smooth(0.99) : fi.lowpass (1, 3000);
randfreq(g) = no.noise : sampleAndhold(sahgate(g))*(1500)+(100)
with {
sampleAndhold(t) = select2(t) ~_;
sahgate(g) = g : upfront : counter -(3) <=(0);
upfront(x) = abs(x-x')>0.5;
counter(g) = (+(1):*(1-g))~_;
};
pulsaflute = environment {
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
fade = (0.5);
speed = hslider ("h:[1]Pulse/[1]Speed (Granulator)[unit:Hz][style:knob][acc:0 1 -10 0 10]", 3,1,6,0.0001):fi.lowpass(1,1);
proba = hslider ("h:[1]Pulse/[2]Probability (Granulator)[unit:%][style:knob][acc:0 1 -10 0 10]", 88,60,100,1) *(0.01):fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
N = 27;
gain = 1;
typeModulation = 0;
nonLinearity = 0;
frequencyMod = 220;
nonLinAttack = 0.1;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
env1Release = 0.5;
nlfOrder = 6;
envelopeMod = en.asr(nonLinAttack,1,0.1,gate);
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
feedBack1 = 0.4;
feedBack2 = 0.4;
embouchureDelayLength = (ma.SR/freq)/2-2;
boreDelayLength = ma.SR/freq-2;
embouchureDelay = de.fdelay(4096,embouchureDelayLength);
boreDelay = de.fdelay(4096,boreDelayLength);
poly = _ <: _ - _*_*_;
reflexionFilter = fi.lowpass(1,2000);
env1 = en.adsr(env1Attack,env1Decay,0.9,env1Release,(gate | pressureEnvelope))*pressure*1.1;
env2 = en.asr(env2Attack,1,env2Release,gate)*0.5;
vibratoEnvelope = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*vibratoGain;
vibrato = os.osc(vibratoFreq)*vibratoEnvelope;
breath = no.noise*env1;
flow = env1 + breath*breathAmp + vibrato;
|
85057ad282435db7046d26ef04c57ae950948691dda623d75b8feff96d685fc3 | RuolunWeng/ruolunweng.github.io | BouncyHarp.dsp | declare name "Bouncy Harp";
declare author "ER"; //From Nonlinear EKS by Julius Smith and Romain Michon;
import("stdfaust.lib");
/* =============== DESCRIPTION ================= :
Do not hesitate to make swift and abrupt gestures.
- Head : Silence/reverb.
- Swing : To pluck the strings of the harp.
- Fishing rod with abrupt stop in Head position : bouncing string effect.
- Frying Pan and Tennis Racket : to pluck a single bouncing string.
- LOOPING MODE :
==> Bottom position/Rotation around Bottom = record loop
==> Head = listen to loop
==> Swift mouvements around head = siren/scratched record effect
*/
//==================== INSTRUMENT =======================
process = par(i, N, NFLeks(i)):>_<: select2(byPass,capture,_) <: instrReverbHarp;
NFLeks(n) = filtered_excitation(n,P(octave(n)),octave(n)) : stringloop(octave(n));
capture = _<:capt,_ : select2(B)
with{
B = hand > (0.5); // Capture sound while hand plays
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
capt = *(B) : (+ : de.delay(1048576, D-1)) ~ *(1.0-B) ;
};
//==================== GUI SPECIFICATION ================
N = 15;
hand = hslider("[1]Instrument Hand (Loop mode: hand>0 = recording, 0 = playback)[acc:1 0 -8 0 11]", 0, 0, N, 1);// => gate
gain = 1;
byPass = checkbox("[7]Loop Mode ON/OFF (max 20s)") : reverse;//In loop capture mode : hand>0 = recording, 0 = stop recording/playback (Y axis upward)
reverse = select2(_, 1, 0);
pickangle = 0.9 * hslider("[3]Dry/Soft Strings[acc:2 1 -10 0 10]", 0.45,0,0.9,0.1);
beta = hslider("[4]Picking Position [acc:2 1 -10 0 10]", 0.13, 0.02, 0.5, 0.01);
t60 = hslider("[5]Resonance (InstrReverb)[acc:1 1 -10 0 10]", 5, 0.5, 10, 0.01); // -60db decay time (sec)
B = 0.5;
L = -10 : ba.db2linear;
//---------------------------------- FREQUENCY TABLE ---------------------------
freq(0) = 115;
freq(1) = 130;
freq(2) = 145;
freq(3) = 160;
freq(4) = 175;
freq(d) = freq(d-5)*(2);
octave(d) = freq(d) * hslider("[2]Hight[acc:0 0 -10 0 10]", 3, 1, 6, 0.1) : si.smooth(0.999);
//==================== SIGNAL PROCESSING ================
//----------------------- noiseburst -------------------------
// White no.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 de.delay-line allocation)
ppdel(f) = beta*P(f); // pick position de.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));
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("v:[8]Reverb/ Reverberation Volume (InstrReverb)[acc:1 1 -10 20 0 0.5] ",0.5,0.1,1,0.01) : si.smooth(0.999);
roomSize = hslider("v:[8]Reverb/ÒReverberation Room Size (InstrReverb)[acc:1 1 -10 0 25]", 0.72,0.01,2,0.01);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/BouncyHarp.dsp | faust | From Nonlinear EKS by Julius Smith and Romain Michon;
=============== DESCRIPTION ================= :
Do not hesitate to make swift and abrupt gestures.
- Head : Silence/reverb.
- Swing : To pluck the strings of the harp.
- Fishing rod with abrupt stop in Head position : bouncing string effect.
- Frying Pan and Tennis Racket : to pluck a single bouncing string.
- LOOPING MODE :
==> Bottom position/Rotation around Bottom = record loop
==> Head = listen to loop
==> Swift mouvements around head = siren/scratched record effect
==================== INSTRUMENT =======================
Capture sound while hand plays
convert button signal from float to integer
Reset capture when button is pressed
Compute capture duration while button is pressed: 0..NNNN0..MMM
==================== GUI SPECIFICATION ================
=> gate
In loop capture mode : hand>0 = recording, 0 = stop recording/playback (Y axis upward)
-60db decay time (sec)
---------------------------------- FREQUENCY TABLE ---------------------------
==================== SIGNAL PROCESSING ================
----------------------- noiseburst -------------------------
White no.noise burst (adapted from Faust's karplus.dsp example)
Requires music.lib (for no.noise)
fundamental period in samples
maximum P (for de.delay-line allocation)
pick position de.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 | declare name "Bouncy Harp";
import("stdfaust.lib");
process = par(i, N, NFLeks(i)):>_<: select2(byPass,capture,_) <: instrReverbHarp;
NFLeks(n) = filtered_excitation(n,P(octave(n)),octave(n)) : stringloop(octave(n));
capture = _<:capt,_ : select2(B)
with{
capt = *(B) : (+ : de.delay(1048576, D-1)) ~ *(1.0-B) ;
};
N = 15;
gain = 1;
reverse = select2(_, 1, 0);
pickangle = 0.9 * hslider("[3]Dry/Soft Strings[acc:2 1 -10 0 10]", 0.45,0,0.9,0.1);
beta = hslider("[4]Picking Position [acc:2 1 -10 0 10]", 0.13, 0.02, 0.5, 0.01);
B = 0.5;
L = -10 : ba.db2linear;
freq(0) = 115;
freq(1) = 130;
freq(2) = 145;
freq(3) = 160;
freq(4) = 175;
freq(d) = freq(d-5)*(2);
octave(d) = freq(d) * hslider("[2]Hight[acc:0 0 -10 0 10]", 3, 1, 6, 0.1) : si.smooth(0.999);
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));
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("v:[8]Reverb/ Reverberation Volume (InstrReverb)[acc:1 1 -10 20 0 0.5] ",0.5,0.1,1,0.01) : si.smooth(0.999);
roomSize = hslider("v:[8]Reverb/ÒReverberation Room Size (InstrReverb)[acc:1 1 -10 0 25]", 0.72,0.01,2,0.01);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
f8836c6be0ed718f79305e4258fe18cdf874ea7232d467ba787c0c649357bc86 | RuolunWeng/ruolunweng.github.io | StalactiteHarp.dsp | declare name "Stalactite Harp";
declare author "ER"; //From Non-linear EKS by Julius Smith and Romain Michon;
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- Stalactite harp mimicking the sound of drops of water in a cave
- Head = Reverberation
- Left = Rare drops
- Right = Frequent and rapidly falling drops
- Back = Harp/Silence
*/
//==================== INSTRUMENT =======================
process = vgroup("Stalactite Harp",par(i, N, NFLeks(i)):>_<: instrReverbHarp);
NFLeks(n) = filtered_excitation(n+1,P(freq(n)),freq(n)) : stringloop(freq(n)) : fi.lowpass(1,8000);
//==================== GUI SPECIFICATION ================
N = 14;
hand = hslider("[1]Instrument Hand[acc:2 1 -10 0 10]", 8, 0, N, 1) : ba.automat(360, 15, 0.0);// => gate
gain = 1;
pickangle = 0.81;
beta = 0.5;
t60 = hslider("h:[3]Reverb/[1]Resonance (InstrReverb)[style:knob][acc:0 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) = 1108.73;
freq(1) = 1244.50;
freq(2) = 1479.97;
freq(3) = 1661.21;
freq(4) = 1864.65;
freq(d) = freq(d-5)*2;
//==================== SIGNAL PROCESSING ================
//----------------------- noiseburst -------------------------
// White no.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) = select2(position(d),0,pulsaxo.gate) : upfront : release(n) : > (0.0);
};
pulsaxo = environment{
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
fade = (0.5); // min > 0 pour eviter division par 0
speed = hslider ("h:[2]Pulse/[1]Speed (Granulator)[unit:Hz][style:knob][acc:0 1 -15 0 8]", 2,0.1,10,0.0001):fi.lowpass(1,1);
proba = hslider ("h:[2]Pulse/[2]Probability (Granulator)[unit:%][style:knob][acc:2 0 -15 0 10]", 95,20,100,1) * (0.01) : fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
P(f) = ma.SR/f ; // fundamental period in samples
Pmax = 4096; // maximum P (for de.delay-line allocation)
ppdel(f) = beta*P(f); // pick position de.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));// : NLFM(f));
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[3]Reverb/[2]Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -30 0 13]", 0.2,0.05,1,0.01) : si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[3]Reverb/[3]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -30 0 13]", 0.72,0.05,1.7,0.01):min(1.7):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/StalactiteHarp.dsp | faust | From Non-linear EKS by Julius Smith and Romain Michon;
=============== DESCRIPTION ================= :
- Stalactite harp mimicking the sound of drops of water in a cave
- Head = Reverberation
- Left = Rare drops
- Right = Frequent and rapidly falling drops
- Back = Harp/Silence
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
=> gate
-60db decay time (sec)
---------------------------------- FREQUENCY TABLE ---------------------------
==================== SIGNAL PROCESSING ================
----------------------- noiseburst -------------------------
White no.noise burst (adapted from Faust's karplus.dsp example)
Requires music.lib (for no.noise)
min > 0 pour eviter division par 0
fundamental period in samples
maximum P (for de.delay-line allocation)
pick position de.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
: NLFM(f)); | declare name "Stalactite Harp";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("Stalactite Harp",par(i, N, NFLeks(i)):>_<: instrReverbHarp);
NFLeks(n) = filtered_excitation(n+1,P(freq(n)),freq(n)) : stringloop(freq(n)) : fi.lowpass(1,8000);
N = 14;
gain = 1;
pickangle = 0.81;
beta = 0.5;
B = 0;
L = -10 : ba.db2linear;
freq(0) = 1108.73;
freq(1) = 1244.50;
freq(2) = 1479.97;
freq(3) = 1661.21;
freq(4) = 1864.65;
freq(d) = freq(d-5)*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) = select2(position(d),0,pulsaxo.gate) : upfront : release(n) : > (0.0);
};
pulsaxo = environment{
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
speed = hslider ("h:[2]Pulse/[1]Speed (Granulator)[unit:Hz][style:knob][acc:0 1 -15 0 8]", 2,0.1,10,0.0001):fi.lowpass(1,1);
proba = hslider ("h:[2]Pulse/[2]Probability (Granulator)[unit:%][style:knob][acc:2 0 -15 0 10]", 95,20,100,1) * (0.01) : fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
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)
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[3]Reverb/[2]Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -30 0 13]", 0.2,0.05,1,0.01) : si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[3]Reverb/[3]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -30 0 13]", 0.72,0.05,1.7,0.01):min(1.7):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
bad9f084d37bb26314c079e3bffd91acbcdeeb59cd0b0950f0817c4b603cf8fc | RuolunWeng/ruolunweng.github.io | SClarinet.dsp | declare name "Clarinet";
declare description "Nonlinear WaveGuide Clarinet";
declare author "Romain Michon";
declare copyright "Romain Michon ([email protected])";
declare version "1.0";
declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
declare description "A simple clarinet physical model, as discussed by Smith (1986), McIntyre, Schumacher, Woodhouse (1983), and others.";
declare reference "https://ccrma.stanford.edu/~jos/pasp/Woodwinds.html";
//Modification Grame July 2015
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- Clarinet responding to vigorous gestures
- Turn ON clarinet (0=OFF, 1=ON)
- Head = High frequencies/Silence when hold still
- Tilt = very soft sound
- Bottom = Low frequencies
- Right = Breathy clarinet
- Fishing rod (vigorous mouvements) :
==> Downward = to reach lower frequencies
==> Upward = To 'through' the sound in the air = vanishes, comes back when Tilt
- Rocking = from full sound to breathy sound
- Shaking in right position = no.noise impulses
*/
//==================== INSTRUMENT =======================
process = vgroup("CLARINET",
//Commuted Loss Filtering
(_,(breathPressure <: _,_) : (filter*-0.95 - _ <:
//Non-Linear Scattering
*(reedTable)) + _) ~
//Delay with Feedback
(delayLine):// : NLFM) :
//scaling and stereo
*(gain)*1.5);
//==================== GUI SPECIFICATION ================
freq = hslider("h:[2]Instrument/Frequency[unit:Hz][tooltip:Tone frequency][acc:1 1 -14 0 10]", 440,110,1300,0.01):si.smooth(0.999);
gain = 1;
gate = hslider("[1]ON/OFF",0,0,1,1);
reedStiffness = hslider("h:[3]Parameters/Instrument Stiffness[style:knob][acc:0 1 -12 0 12]", 0.25,0.01,1,0.01);
noiseGain = hslider("h:[3]Parameters/Breath Noise[style:knob][acc:0 1 -10 0 12]", 0.02,0,0.12,0.01);
pressure = hslider("h:[3]Parameters/ Pressure[style:knob][acc:1 0 -10 0 10]", 0.8,0.25,1,0.01);
vibratoFreq = 5;
vibratoGain = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.01;
envelopeAttack = 0.1;
envelopeDecay = 0.05;
envelopeRelease = 0.1;
//==================== SIGNAL PROCESSING ======================
//----------------------- Synthesis PARAMETERS computing and functions declaration ----------------------------
//reed table PARAMETERS
reedTableOffset = 0.7;
reedTableSlope = -0.44 + (0.26*reedStiffness);
//the reed function is declared in INSTRUMENT.lib
reedTable = instrument.reed(reedTableOffset,reedTableSlope);
//delay line with a length adapted in function of the order of nonlinear filter
delayLength = ma.SR/freq*0.5 - 1.5;// - (nlfOrder*nonLinearity)*(typeModulation < 2);
delayLine = de.fdelay(4096,delayLength);
//one zero filter used as a allpass: pole is set to -1
filter = instrument.oneZero0(0.5,0.5);
//stereoizer is declared in INSTRUMENT.lib and implement a stereo spacialisation in function of
//the frequency period in number of samples
//stereo = stereoizerCla(ma.SR/freq);
//----------------------- Algorithm implementation ----------------------------
//Breath pressure + vibrato + breath no.noise + envelope (Attack / Decay / Sustain / Release)
envelope = en.adsr(envelopeAttack,envelopeDecay,1,envelopeRelease,gate)*pressure*0.9;
vibrato = os.osc(vibratoFreq)*vibratoGain*
instrument.envVibrato(0.1*2*vibratoAttack,0.9*2*vibratoAttack,100,vibratoRelease,gate);
breath = envelope + envelope*no.noise*noiseGain;
breathPressure = breath + breath*vibrato;
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SClarinet.dsp | faust | Synthesis Tool Kit 4.3 (MIT style license);
Modification Grame July 2015
=============== DESCRIPTION ================= :
- Clarinet responding to vigorous gestures
- Turn ON clarinet (0=OFF, 1=ON)
- Head = High frequencies/Silence when hold still
- Tilt = very soft sound
- Bottom = Low frequencies
- Right = Breathy clarinet
- Fishing rod (vigorous mouvements) :
==> Downward = to reach lower frequencies
==> Upward = To 'through' the sound in the air = vanishes, comes back when Tilt
- Rocking = from full sound to breathy sound
- Shaking in right position = no.noise impulses
==================== INSTRUMENT =======================
Commuted Loss Filtering
Non-Linear Scattering
Delay with Feedback
: NLFM) :
scaling and stereo
==================== GUI SPECIFICATION ================
==================== SIGNAL PROCESSING ======================
----------------------- Synthesis PARAMETERS computing and functions declaration ----------------------------
reed table PARAMETERS
the reed function is declared in INSTRUMENT.lib
delay line with a length adapted in function of the order of nonlinear filter
- (nlfOrder*nonLinearity)*(typeModulation < 2);
one zero filter used as a allpass: pole is set to -1
stereoizer is declared in INSTRUMENT.lib and implement a stereo spacialisation in function of
the frequency period in number of samples
stereo = stereoizerCla(ma.SR/freq);
----------------------- Algorithm implementation ----------------------------
Breath pressure + vibrato + breath no.noise + envelope (Attack / Decay / Sustain / Release) | declare name "Clarinet";
declare description "Nonlinear WaveGuide Clarinet";
declare author "Romain Michon";
declare copyright "Romain Michon ([email protected])";
declare version "1.0";
declare description "A simple clarinet physical model, as discussed by Smith (1986), McIntyre, Schumacher, Woodhouse (1983), and others.";
declare reference "https://ccrma.stanford.edu/~jos/pasp/Woodwinds.html";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("CLARINET",
(_,(breathPressure <: _,_) : (filter*-0.95 - _ <:
*(reedTable)) + _) ~
*(gain)*1.5);
freq = hslider("h:[2]Instrument/Frequency[unit:Hz][tooltip:Tone frequency][acc:1 1 -14 0 10]", 440,110,1300,0.01):si.smooth(0.999);
gain = 1;
gate = hslider("[1]ON/OFF",0,0,1,1);
reedStiffness = hslider("h:[3]Parameters/Instrument Stiffness[style:knob][acc:0 1 -12 0 12]", 0.25,0.01,1,0.01);
noiseGain = hslider("h:[3]Parameters/Breath Noise[style:knob][acc:0 1 -10 0 12]", 0.02,0,0.12,0.01);
pressure = hslider("h:[3]Parameters/ Pressure[style:knob][acc:1 0 -10 0 10]", 0.8,0.25,1,0.01);
vibratoFreq = 5;
vibratoGain = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.01;
envelopeAttack = 0.1;
envelopeDecay = 0.05;
envelopeRelease = 0.1;
reedTableOffset = 0.7;
reedTableSlope = -0.44 + (0.26*reedStiffness);
reedTable = instrument.reed(reedTableOffset,reedTableSlope);
delayLine = de.fdelay(4096,delayLength);
filter = instrument.oneZero0(0.5,0.5);
envelope = en.adsr(envelopeAttack,envelopeDecay,1,envelopeRelease,gate)*pressure*0.9;
vibrato = os.osc(vibratoFreq)*vibratoGain*
instrument.envVibrato(0.1*2*vibratoAttack,0.9*2*vibratoAttack,100,vibratoRelease,gate);
breath = envelope + envelope*no.noise*noiseGain;
breathPressure = breath + breath*vibrato;
|
eaa5aff005c6a3078842a1c7693c203d6871d5f558569c54e635f84d48957ad4 | RuolunWeng/ruolunweng.github.io | STibetanBowl.dsp | declare name "Tibetan Bowl";
declare description "Banded Waveguide Modeld Tibetan Bowl";
declare author "Romain Michon";
declare copyright "Romain Michon ([email protected])";
declare version "1.0";
declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
declare description "This instrument uses banded waveguide. For more information, see Essl, G. and Cook, P. Banded Waveguides: Towards Physical Modelling of Bar Percussion Instruments, Proceedings of the 1999 International Computer Music Conference.";
//Modifications GRAME July 2015
/* ============ DESCRIPTION =============
- Tibetan Bowl
- Set the frequency manually
- Fishing rod/Front shaking = Ringing the bowl
- Right = maximum modulation
- Rocking = modulating the sound
*/
import("stdfaust.lib");
instrument = library("instruments.lib");
//==================== INSTRUMENT =======================
process = (((select-1)*-1) <:
//nModes resonances with nModes feedbacks for bow table look-up
par(i,nModes,(resonance(i)~_))):>+:
NLFM :> fi.lowpass(1, 5000);
//==================== GUI SPECIFICATION ================
freq = hslider("[1]Frequency[unit:Hz][tooltip:Tone frequency]",440,180,780,1);
gain = 0.5;
gate = 0;
select = hslider("[0]Play[tooltip:0=Bow; 1=Strike] [acc:2 1 -10 0 10]", 0,0,1,1);
baseGain = 0.5;
typeModulation = 3;
nonLinearity = hslider("[2]Modulation[acc:0 1 -10 0 10][tooltip:Nonlinearity factor (value between 0 and 1)]",0.02,0,0.1,0.001):si.smooth(0.999);
frequencyMod = hslider("[3]Modulation Frequency[unit:Hz][acc:0 0 -10 0 10]", 220,150,500,0.1):si.smooth(0.999);
nonLinAttack = 0.1;
//==================== MODAL PARAMETERS ================
preset = 0;
nMode(0) = 12;
modes(0,0) = 0.996108344;
basegains(0,0) = 0.999925960128219;
excitation(0,0) = 11.900357 / 10;
modes(0,1) = 1.0038916562;
basegains(0,1) = 0.999925960128219;
excitation(0,1) = 11.900357 / 10;
modes(0,2) = 2.979178;
basegains(0,2) = 0.999982774366897;
excitation(0,2) = 10.914886 / 10;
modes(0,3) = 2.99329767;
basegains(0,3) = 0.999982774366897;
excitation(0,3) = 10.914886 / 10;
modes(0,4) = 5.704452;
basegains(0,4) = 1.0;
excitation(0,4) = 42.995041 / 10;
modes(0,5) = 5.704452;
basegains(0,5) = 1.0;
excitation(0,5) = 42.995041 / 10;
modes(0,6) = 8.9982;
basegains(0,6) = 1.0;
excitation(0,6) = 40.063034 / 10;
modes(0,7) = 9.01549726;
basegains(0,7) = 1.0;
excitation(0,7) = 40.063034 / 10;
modes(0,8) = 12.83303;
basegains(0,8) = 0.999965497558225;
excitation(0,8) = 7.063034 / 10;
modes(0,9) = 12.807382;
basegains(0,9) = 0.999965497558225;
excitation(0,9) = 7.063034 / 10;
modes(0,10) = 17.2808219;
basegains(0,10) = 0.9999999999999999999965497558225;
excitation(0,10) = 57.063034 / 10;
modes(0,11) = 21.97602739726;
basegains(0,11) = 0.999999999999999965497558225;
excitation(0,11) = 57.063034 / 10;
//==================== SIGNAL PROCESSING ================
//----------------------- Nonlinear filter ----------------------------
//nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
//nonlinear filter order
nlfOrder = 6;
//nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
//for using it with waveguide instruments
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),1,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//the number of modes depends on the preset being used
nModes = nMode(preset);
delayLengthBase = ma.SR/freq;
//delay lengths in number of samples
delayLength(x) = delayLengthBase/modes(preset,x);
//delay lines
delayLine(x) = de.delay(4096,delayLength(x));
//Filter bank: fi.bandpass filters (declared in instrument.lib)
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x) = instrument.bandPass(freq*modes(preset,x),radius);
//One resonance
resonance(x) = + : + (excitation(preset,x)*select) : delayLine(x) : *(basegains(preset,x)) : bandPassFilter(x);
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/STibetanBowl.dsp | faust | Synthesis Tool Kit 4.3 (MIT style license);
Modifications GRAME July 2015
============ DESCRIPTION =============
- Tibetan Bowl
- Set the frequency manually
- Fishing rod/Front shaking = Ringing the bowl
- Right = maximum modulation
- Rocking = modulating the sound
==================== INSTRUMENT =======================
nModes resonances with nModes feedbacks for bow table look-up
==================== GUI SPECIFICATION ================
==================== MODAL PARAMETERS ================
==================== SIGNAL PROCESSING ================
----------------------- Nonlinear filter ----------------------------
nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
nonlinear filter order
nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
for using it with waveguide instruments
----------------------- Synthesis parameters computing and functions declaration ----------------------------
the number of modes depends on the preset being used
delay lengths in number of samples
delay lines
Filter bank: fi.bandpass filters (declared in instrument.lib)
One resonance | declare name "Tibetan Bowl";
declare description "Banded Waveguide Modeld Tibetan Bowl";
declare author "Romain Michon";
declare copyright "Romain Michon ([email protected])";
declare version "1.0";
declare description "This instrument uses banded waveguide. For more information, see Essl, G. and Cook, P. Banded Waveguides: Towards Physical Modelling of Bar Percussion Instruments, Proceedings of the 1999 International Computer Music Conference.";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = (((select-1)*-1) <:
par(i,nModes,(resonance(i)~_))):>+:
NLFM :> fi.lowpass(1, 5000);
freq = hslider("[1]Frequency[unit:Hz][tooltip:Tone frequency]",440,180,780,1);
gain = 0.5;
gate = 0;
select = hslider("[0]Play[tooltip:0=Bow; 1=Strike] [acc:2 1 -10 0 10]", 0,0,1,1);
baseGain = 0.5;
typeModulation = 3;
nonLinearity = hslider("[2]Modulation[acc:0 1 -10 0 10][tooltip:Nonlinearity factor (value between 0 and 1)]",0.02,0,0.1,0.001):si.smooth(0.999);
frequencyMod = hslider("[3]Modulation Frequency[unit:Hz][acc:0 0 -10 0 10]", 220,150,500,0.1):si.smooth(0.999);
nonLinAttack = 0.1;
preset = 0;
nMode(0) = 12;
modes(0,0) = 0.996108344;
basegains(0,0) = 0.999925960128219;
excitation(0,0) = 11.900357 / 10;
modes(0,1) = 1.0038916562;
basegains(0,1) = 0.999925960128219;
excitation(0,1) = 11.900357 / 10;
modes(0,2) = 2.979178;
basegains(0,2) = 0.999982774366897;
excitation(0,2) = 10.914886 / 10;
modes(0,3) = 2.99329767;
basegains(0,3) = 0.999982774366897;
excitation(0,3) = 10.914886 / 10;
modes(0,4) = 5.704452;
basegains(0,4) = 1.0;
excitation(0,4) = 42.995041 / 10;
modes(0,5) = 5.704452;
basegains(0,5) = 1.0;
excitation(0,5) = 42.995041 / 10;
modes(0,6) = 8.9982;
basegains(0,6) = 1.0;
excitation(0,6) = 40.063034 / 10;
modes(0,7) = 9.01549726;
basegains(0,7) = 1.0;
excitation(0,7) = 40.063034 / 10;
modes(0,8) = 12.83303;
basegains(0,8) = 0.999965497558225;
excitation(0,8) = 7.063034 / 10;
modes(0,9) = 12.807382;
basegains(0,9) = 0.999965497558225;
excitation(0,9) = 7.063034 / 10;
modes(0,10) = 17.2808219;
basegains(0,10) = 0.9999999999999999999965497558225;
excitation(0,10) = 57.063034 / 10;
modes(0,11) = 21.97602739726;
basegains(0,11) = 0.999999999999999965497558225;
excitation(0,11) = 57.063034 / 10;
nlfOrder = 6;
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),1,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
nModes = nMode(preset);
delayLengthBase = ma.SR/freq;
delayLength(x) = delayLengthBase/modes(preset,x);
delayLine(x) = de.delay(4096,delayLength(x));
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x) = instrument.bandPass(freq*modes(preset,x),radius);
resonance(x) = + : + (excitation(preset,x)*select) : delayLine(x) : *(basegains(preset,x)) : bandPassFilter(x);
|
a0958f65433340898ec5f3d400d44753c35da32db6d4ba72431ed598bff3b0b1 | RuolunWeng/ruolunweng.github.io | TunedBars.dsp | declare name "Tuned Bars";
declare author "ER";//From "Tuned Bar" by Romain Michon ([email protected]);
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- Cascading tuned bars
- Head = Reverberation / Silence
- Bottom = Chime
- Left = Low frequencies + slow rhythm
- Right = High frequencies + fast rhythm
- Reversed Fishing Rod = start at bottom or left or right and go rapidly up to head then down again
- Left/Bottom/Left = Low pitched chime
- Right/Bottom/Right = High pitched chime
- Back/Front/Geiger counter = Chime
*/
//==================== INSTRUMENT =======================
process = par(i, N, tunedBar(i)):>_<: drywet(_,echo) <: instrReverbChime : *(2),*(2);
tunedBar(n) =
((select-1)*-1) <:
//nModes resonances with nModes feedbacks for bow table look-up
par(i,nModes,(resonance(i,octave(n),gate(n))~_)):> + :
//Signal Scaling and stereo
*(4);
//==================== GUI SPECIFICATION ================
N = 10;
gain = 1;
gate(n) = position(n) : upfront;
hand = hslider("[1]Instrument Hand[acc:1 0 -10 0 10]", 5, 0, N, 1):si.smooth(0.999):min(N):max(0):int:ba.automat(B, 15, 0.0);
B = hslider("[3]Speed[style:knob][acc:0 1 -10 0 10]", 360, 120, 720, 60): si.smooth(0.99) : min(720) : max(120) : int;
hight = hslider("[2]Hight[acc:0 1 -10 0 10]", 4, 0.5, 8, 0.1);//:si.smooth(0.999);
octave(d) = freq(d)*(hight);
position(n) = abs(hand - n) < 0.5;
upfront(x) = x>x';
select = 1;
//----------------------- Frequency Table --------------------
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 184.99;
freq(4) = 207.65;
freq(5) = 233.08;
freq(d) = freq(d-6)*2;
//==================== MODAL PARAMETERS ================
preset = 2;
nMode(2) = 4;
modes(2,0) = 1;
basegains(2,0) = pow(0.999,1);
excitation(2,0,g) = 1*gain*g/nMode(2);
modes(2,1) = 4.0198391420;
basegains(2,1) = pow(0.999,2);
excitation(2,1,g) = 1*gain*g/nMode(2);
modes(2,2) = 10.7184986595;
basegains(2,2) = pow(0.999,3);
excitation(2,2,g) = 1*gain*g/nMode(2);
modes(2,3) = 18.0697050938;
basegains(2,3) = pow(0.999,4);
excitation(2,3,g) = 1*gain*g/nMode(2);
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//the number of modes depends on the preset being used
nModes = nMode(preset);
//bow table parameters
tableOffset = 0;
tableSlope = 10 - (9*bowPressure);
delayLengthBase(f) = ma.SR/f;
//de.delay lengths in number of samples
delayLength(x,f) = delayLengthBase(f)/modes(preset,x);
//de.delay lines
delayLine(x,f) = de.delay(4096,delayLength(x,f));
//Filter bank: fi.bandpass filters (declared in instrument.lib)
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x,f) = instrument.bandPass(f*modes(preset,x),radius);
//----------------------- Algorithm implementation ----------------------------
//One resonance
resonance(x,f,g) = + : + (excitation(preset,x,g)*select) : delayLine(x,f) : *(basegains(preset,x)) : bandPassFilter(x,f);
echo = +~(@(22050)*(feedback));
//feedback = hslider("Echo Intensity (Feedback)[acc:1 1 -5 0 12]", 0.1, 0.05, 0.65, 0.01):si.smooth(0.999):min(0.05):max(0.65);
feedback = 0.8;
drywet(x,y) = (1-c)*x + c*y
with {
c = hslider("[4]Echo Intensity[style:knob][unit:%][acc:1 1 -8 0 10]", 20,0,99,0.01)*(0.01):si.smooth(0.999):min(0.99):max(0.001);
};
//instrReverb from instrument.lib
instrReverbChime = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
roomSize = hslider("h:[5]Reverb/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 12]", 0.2,0.1,1.7,0.01):min(1.7):max(0.1);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/TunedBars.dsp | faust | From "Tuned Bar" by Romain Michon ([email protected]);
=============== DESCRIPTION ================= :
- Cascading tuned bars
- Head = Reverberation / Silence
- Bottom = Chime
- Left = Low frequencies + slow rhythm
- Right = High frequencies + fast rhythm
- Reversed Fishing Rod = start at bottom or left or right and go rapidly up to head then down again
- Left/Bottom/Left = Low pitched chime
- Right/Bottom/Right = High pitched chime
- Back/Front/Geiger counter = Chime
==================== INSTRUMENT =======================
nModes resonances with nModes feedbacks for bow table look-up
Signal Scaling and stereo
==================== GUI SPECIFICATION ================
:si.smooth(0.999);
----------------------- Frequency Table --------------------
==================== MODAL PARAMETERS ================
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
the number of modes depends on the preset being used
bow table parameters
de.delay lengths in number of samples
de.delay lines
Filter bank: fi.bandpass filters (declared in instrument.lib)
----------------------- Algorithm implementation ----------------------------
One resonance
feedback = hslider("Echo Intensity (Feedback)[acc:1 1 -5 0 12]", 0.1, 0.05, 0.65, 0.01):si.smooth(0.999):min(0.05):max(0.65);
instrReverb from instrument.lib | declare name "Tuned Bars";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = par(i, N, tunedBar(i)):>_<: drywet(_,echo) <: instrReverbChime : *(2),*(2);
tunedBar(n) =
((select-1)*-1) <:
par(i,nModes,(resonance(i,octave(n),gate(n))~_)):> + :
*(4);
N = 10;
gain = 1;
gate(n) = position(n) : upfront;
hand = hslider("[1]Instrument Hand[acc:1 0 -10 0 10]", 5, 0, N, 1):si.smooth(0.999):min(N):max(0):int:ba.automat(B, 15, 0.0);
B = hslider("[3]Speed[style:knob][acc:0 1 -10 0 10]", 360, 120, 720, 60): si.smooth(0.99) : min(720) : max(120) : int;
octave(d) = freq(d)*(hight);
position(n) = abs(hand - n) < 0.5;
upfront(x) = x>x';
select = 1;
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 184.99;
freq(4) = 207.65;
freq(5) = 233.08;
freq(d) = freq(d-6)*2;
preset = 2;
nMode(2) = 4;
modes(2,0) = 1;
basegains(2,0) = pow(0.999,1);
excitation(2,0,g) = 1*gain*g/nMode(2);
modes(2,1) = 4.0198391420;
basegains(2,1) = pow(0.999,2);
excitation(2,1,g) = 1*gain*g/nMode(2);
modes(2,2) = 10.7184986595;
basegains(2,2) = pow(0.999,3);
excitation(2,2,g) = 1*gain*g/nMode(2);
modes(2,3) = 18.0697050938;
basegains(2,3) = pow(0.999,4);
excitation(2,3,g) = 1*gain*g/nMode(2);
nModes = nMode(preset);
tableOffset = 0;
tableSlope = 10 - (9*bowPressure);
delayLengthBase(f) = ma.SR/f;
delayLength(x,f) = delayLengthBase(f)/modes(preset,x);
delayLine(x,f) = de.delay(4096,delayLength(x,f));
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x,f) = instrument.bandPass(f*modes(preset,x),radius);
resonance(x,f,g) = + : + (excitation(preset,x,g)*select) : delayLine(x,f) : *(basegains(preset,x)) : bandPassFilter(x,f);
echo = +~(@(22050)*(feedback));
feedback = 0.8;
drywet(x,y) = (1-c)*x + c*y
with {
c = hslider("[4]Echo Intensity[style:knob][unit:%][acc:1 1 -8 0 10]", 20,0,99,0.01)*(0.01):si.smooth(0.999):min(0.99):max(0.001);
};
instrReverbChime = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
roomSize = hslider("h:[5]Reverb/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 12]", 0.2,0.1,1.7,0.01):min(1.7):max(0.1);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
92a9d58046103c011247b549e1db14da3a57dff5f2a9c8b1f4312652796cd310 | RuolunWeng/ruolunweng.github.io | BlowhistleBottle.dsp | declare name "Blowhistle Bottles";
declare author "ER"; //From "Blow bottle" by Romain Michon;
declare version "1.0";
declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
declare description "This object implements a helmholtz resonator (biquad filter) with a polynomial jet excitation (a la Cook).";
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- Blow bottles with whistling echo.
- Left : silence/dying echo.
- Head : reverberation
- Front : single blow bottle.
- Back : maximum whistling echo
- Bottom : bottle + whistling echo without reverberation
- Rocking : changes tone of blow bottle.
- Fishing rod : varying reverberation.
*/
//==================== INSTRUMENT =======================
process = vgroup("Blowhistle Bottles", par(i, N, blow(i)) :>_<: instrReverblow: (*(1),*(1)));
blow(n)= par(i, 2,
//differential pressure
(-(breathPressure(trigger(n))) <:
((+(1))*randPressure((trigger(n))) : +(breathPressure(trigger(n)))) - *(instrument.jetTable),_ : baPaF(i,n),_)~_: !,_:
//signal scaling
fi.dcblocker*envelopeG(trigger(n))*(0.5)<:+(voice(i,n))*resonGain(i)):>_
with{
baPaF(0,n) = bandPassFilter(freq(n));
baPaF(1,n) = bandPassFilter(freq(n)*8);
voice(0,n) = 0*n;
voice(1,n) = 1*(fi.resonbp(freq(n)*8,Q,gain):echo);
resonGain(0) = 1;
resonGain(1) =(hslider("v:[1]Instrument/Whistle Volume[acc:2 0 -10 0 10]", 0.07, 0, 0.2, 0.001))^2:si.smooth(0.999);
echo = _:+~(@(delayEcho):*(feedback));
delayEcho = 44100;
feedback = hslider("h:[2]Echo/Echo Intensity [style:knob][acc:2 0 -10 0 10]", 0.48, 0.2, 0.98, 0.01):si.smooth(0.999):min(0.98):max(0.2);
};
//==================== GUI SPECIFICATION ================
N = 10;
Q = 30;
position(n) = abs(hand - n) < 0.5;
hand = hslider("v:[1]Instrument/Instrument Hand[acc:0 1 -10 0 10]", 5, 0, N, 1):int:ba.automat(360, 15, 0.0);
envelopeAttack = 0.01;
vibratoFreq = 5;
vibratoGain = 0.1;
//--------------------- Non-variable Parameters -------------
gain = 0.5;
noiseGain = 0.5;
pressure = 1.2;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.01;
envelopeDecay = 0.01;
envelopeRelease = 0.5;
//----------------------- Frequency Table --------------------
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 195.99;
freq(4) = 220.00;
freq(d) = freq(d-5)*2;
//==================== SIGNAL PROCESSING ================
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//botlle radius
bottleRadius = 0.999;
bandPassFilter(f) = instrument.bandPass(f,bottleRadius);
//----------------------- Algorithm implementation ----------------------------
//global envelope is of type attack - decay - sustain - release
envelopeG(t) = gain*en.adsr(gain*envelopeAttack,envelopeDecay,0.8,envelopeRelease,t);
//pressure envelope is also ADSR
envelope(t) = pressure*en.adsr(gain*0.02,0.01,0.8,gain*0.2,t);
//vibrato
vibrato(t) = os.osc(vibratoFreq)*vibratoGain*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*os.osc(vibratoFreq);
//breat pressure
breathPressure(t) = envelope(t) + vibrato(t);
//breath no.noise
randPressure(t) = noiseGain*no.noise*breathPressure(t) ;
//------------------------- Enveloppe Trigger --------------------------------------------
trigger(n) = position(n): trig
with{
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
trig = upfront : release(8820) : >(0.0);
};
//------------------------ InstrReverb ----------------------------------------
//from instrument.lib
instrReverblow = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[3]Reverb/ Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.237,0.137,1,0.01) : si.smooth(0.999);
roomSize = hslider("h:[3]Reverb/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.72,0.4,2,0.01);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/BlowhistleBottle.dsp | faust | From "Blow bottle" by Romain Michon;
Synthesis Tool Kit 4.3 (MIT style license);
=============== DESCRIPTION ================= :
- Blow bottles with whistling echo.
- Left : silence/dying echo.
- Head : reverberation
- Front : single blow bottle.
- Back : maximum whistling echo
- Bottom : bottle + whistling echo without reverberation
- Rocking : changes tone of blow bottle.
- Fishing rod : varying reverberation.
==================== INSTRUMENT =======================
differential pressure
signal scaling
==================== GUI SPECIFICATION ================
--------------------- Non-variable Parameters -------------
----------------------- Frequency Table --------------------
==================== SIGNAL PROCESSING ================
----------------------- Synthesis parameters computing and functions declaration ----------------------------
botlle radius
----------------------- Algorithm implementation ----------------------------
global envelope is of type attack - decay - sustain - release
pressure envelope is also ADSR
vibrato
breat pressure
breath no.noise
------------------------- Enveloppe Trigger --------------------------------------------
------------------------ InstrReverb ----------------------------------------
from instrument.lib | declare name "Blowhistle Bottles";
declare version "1.0";
declare description "This object implements a helmholtz resonator (biquad filter) with a polynomial jet excitation (a la Cook).";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("Blowhistle Bottles", par(i, N, blow(i)) :>_<: instrReverblow: (*(1),*(1)));
blow(n)= par(i, 2,
(-(breathPressure(trigger(n))) <:
((+(1))*randPressure((trigger(n))) : +(breathPressure(trigger(n)))) - *(instrument.jetTable),_ : baPaF(i,n),_)~_: !,_:
fi.dcblocker*envelopeG(trigger(n))*(0.5)<:+(voice(i,n))*resonGain(i)):>_
with{
baPaF(0,n) = bandPassFilter(freq(n));
baPaF(1,n) = bandPassFilter(freq(n)*8);
voice(0,n) = 0*n;
voice(1,n) = 1*(fi.resonbp(freq(n)*8,Q,gain):echo);
resonGain(0) = 1;
resonGain(1) =(hslider("v:[1]Instrument/Whistle Volume[acc:2 0 -10 0 10]", 0.07, 0, 0.2, 0.001))^2:si.smooth(0.999);
echo = _:+~(@(delayEcho):*(feedback));
delayEcho = 44100;
feedback = hslider("h:[2]Echo/Echo Intensity [style:knob][acc:2 0 -10 0 10]", 0.48, 0.2, 0.98, 0.01):si.smooth(0.999):min(0.98):max(0.2);
};
N = 10;
Q = 30;
position(n) = abs(hand - n) < 0.5;
hand = hslider("v:[1]Instrument/Instrument Hand[acc:0 1 -10 0 10]", 5, 0, N, 1):int:ba.automat(360, 15, 0.0);
envelopeAttack = 0.01;
vibratoFreq = 5;
vibratoGain = 0.1;
gain = 0.5;
noiseGain = 0.5;
pressure = 1.2;
vibratoBegin = 0.05;
vibratoAttack = 0.5;
vibratoRelease = 0.01;
envelopeDecay = 0.01;
envelopeRelease = 0.5;
freq(0) = 130.81;
freq(1) = 146.83;
freq(2) = 164.81;
freq(3) = 195.99;
freq(4) = 220.00;
freq(d) = freq(d-5)*2;
bottleRadius = 0.999;
bandPassFilter(f) = instrument.bandPass(f,bottleRadius);
envelopeG(t) = gain*en.adsr(gain*envelopeAttack,envelopeDecay,0.8,envelopeRelease,t);
envelope(t) = pressure*en.adsr(gain*0.02,0.01,0.8,gain*0.2,t);
vibrato(t) = os.osc(vibratoFreq)*vibratoGain*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,t)*os.osc(vibratoFreq);
breathPressure(t) = envelope(t) + vibrato(t);
randPressure(t) = noiseGain*no.noise*breathPressure(t) ;
trigger(n) = position(n): trig
with{
upfront(x) = (x-x') > 0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
trig = upfront : release(8820) : >(0.0);
};
instrReverblow = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[3]Reverb/ Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.237,0.137,1,0.01) : si.smooth(0.999);
roomSize = hslider("h:[3]Reverb/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.72,0.4,2,0.01);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
96cbc3f8228c7189c0f660f71ecb23a07819ea580cc0db3f481577bbedb5cbbf | RuolunWeng/ruolunweng.github.io | CMajDryHarp.dsp |
declare name "CMajDryHarp";
declare author "ER";//Adapted from Harpe by Yann Orlarey;
//Modification Grame July 2015
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- Reverberated C Major dry harp
- Left = Lower frequencies/Silence when still
- Front = Resonance (longer notes)
- Back = No Resonance (dry notes)
- Right = Higher frequencies/Fast rhythm
- Head = Reverberation
- Rocking = plucking all strings one by one
*/
//-----------------------------------------------
// Harpe : simple string instrument
// (based on Karplus-Strong)
//
//-----------------------------------------------
KEY = 60; // basic midi key
NCY = 15; // note cycle length
CCY = 15; // control cycle length
BPS = 360; // general tempo (ba.beat per sec)
//-------------------------------Harpe----------------------------------
// Harpe is a simple string instrument. Move the "hand" to play the
// various strings
//-----------------------------------------------------------------------
process = vgroup("Harp", h : harpe(C,N,K) : instrReverbHarp : *(l),*(l))
with {
N = 48; // number of strings
K = 36; // Midi key of first string
h = hslider("[1]Instrument Hand[1] [acc:0 1 -10 0 10]", 24, 0, N, 1) : int: ba.automat(bps, 15, 0.0)
with{
bps = hslider("h:[2]Parameters/[1]Speed[style:knob][acc:0 1 -12 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
//l = vslider("h:parameters/volume [style:knob][unit: dB]", -20, -60, 0, 0.01) : ba.db2linear;
l = -10 : ba.db2linear;
C = 0.5;
};
//----------------------------------Harpe--------------------------------
// USAGE: hand : harpe(C,10,60) : _,_;
// C is the filter coefficient 0..1
// Build a N (10) strings harpe using a pentatonic scale
// based on midi key b (60)
// Each string is triggered by a specific
// position of the "hand"
//-----------------------------------------------------------------------
harpe(C,N,b) = _ <: par(i, N, position(i+1)
:string(C,Major(b).degree2Hz(i), att, lvl)
//envReader(twig,enveloppe):*
: pan((i+0.5)/N) )
:> _,_
with {
att = hslider("h:[2]Parameters/[2]Resonance[style:knob][acc:2 1 -12 0 10]", 5, 0.1, 10, 0.01):min(10):max(0.1);
lvl = 1;
pan(p) = _ <: *(sqrt(1-p)), *(sqrt(p));
position(a,x) = abs(x - a) < 0.5;
};
//----------------------------------Penta-------------------------------
// Pentatonic scale with degree to midi and degree to Hz conversion
// USAGE: Penta(60).degree2midi(3) ==> 67 midikey
// Penta(60).degree2Hz(4) ==> 440 Hz
//-----------------------------------------------------------------------
//---------------------------------- Major -------------------------------
// Major scale.
// From Pentatonic scale with degree to midi and degree to Hz conversion
// USAGE: Penta(60).degree2midi(3) ==> 67 midikey
// Penta(60).degree2Hz(4) ==> 440 Hz
//-----------------------------------------------------------------------
Major(key) = environment {
A4Hz = 440;
degree2midi(0) = key+0;
degree2midi(1) = key+2;
degree2midi(2) = key+4;
degree2midi(3) = key+5;
degree2midi(4) = key+7;
degree2midi(5) = key+9;
degree2midi(6) = key+11;
degree2midi(7) = key+12;
degree2midi(d) = degree2midi(d-8)+12;
degree2Hz(d) = A4Hz*semiton(degree2midi(d)-69) with { semiton(n) = 2.0^(n/12.0); };
};
//----------------------------------String-------------------------------
// A karplus-strong string.
//
// USAGE: string(440Hz, 4s, 1.0, button("play"))
// or button("play") : string(440Hz, 4s, 1.0)
//-----------------------------------------------------------------------
string(coef, freq, t60, level, trig) = no.noise*level
: *(trig : trigger(freq2samples(freq)))
: resonator(freq2samples(freq), att)
with {
resonator(d,a) = (+ : @(d-1)) ~ (average : *(a));
average(x) = (x*(1+coef)+x'*(1-coef))/2;
trigger(n) = upfront : + ~ decay(n) : >(0.0);
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
freq2samples(f) = 44100.0/f;
att = pow(0.001,1.0/(freq*t60)); // attenuation coefficient
};
//================================= REVERB ==============================
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[3]Reverb/ Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[3]Reverb/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1.3,0.01):min(1.3):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/CMajDryHarp.dsp | faust | Adapted from Harpe by Yann Orlarey;
Modification Grame July 2015
=============== DESCRIPTION ================= :
- Reverberated C Major dry harp
- Left = Lower frequencies/Silence when still
- Front = Resonance (longer notes)
- Back = No Resonance (dry notes)
- Right = Higher frequencies/Fast rhythm
- Head = Reverberation
- Rocking = plucking all strings one by one
-----------------------------------------------
Harpe : simple string instrument
(based on Karplus-Strong)
-----------------------------------------------
basic midi key
note cycle length
control cycle length
general tempo (ba.beat per sec)
-------------------------------Harpe----------------------------------
Harpe is a simple string instrument. Move the "hand" to play the
various strings
-----------------------------------------------------------------------
number of strings
Midi key of first string
l = vslider("h:parameters/volume [style:knob][unit: dB]", -20, -60, 0, 0.01) : ba.db2linear;
----------------------------------Harpe--------------------------------
USAGE: hand : harpe(C,10,60) : _,_;
C is the filter coefficient 0..1
Build a N (10) strings harpe using a pentatonic scale
based on midi key b (60)
Each string is triggered by a specific
position of the "hand"
-----------------------------------------------------------------------
envReader(twig,enveloppe):*
----------------------------------Penta-------------------------------
Pentatonic scale with degree to midi and degree to Hz conversion
USAGE: Penta(60).degree2midi(3) ==> 67 midikey
Penta(60).degree2Hz(4) ==> 440 Hz
-----------------------------------------------------------------------
---------------------------------- Major -------------------------------
Major scale.
From Pentatonic scale with degree to midi and degree to Hz conversion
USAGE: Penta(60).degree2midi(3) ==> 67 midikey
Penta(60).degree2Hz(4) ==> 440 Hz
-----------------------------------------------------------------------
----------------------------------String-------------------------------
A karplus-strong string.
USAGE: string(440Hz, 4s, 1.0, button("play"))
or button("play") : string(440Hz, 4s, 1.0)
-----------------------------------------------------------------------
attenuation coefficient
================================= REVERB ============================== |
declare name "CMajDryHarp";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("Harp", h : harpe(C,N,K) : instrReverbHarp : *(l),*(l))
with {
h = hslider("[1]Instrument Hand[1] [acc:0 1 -10 0 10]", 24, 0, N, 1) : int: ba.automat(bps, 15, 0.0)
with{
bps = hslider("h:[2]Parameters/[1]Speed[style:knob][acc:0 1 -12 0 10]", 480, 180, 720, 1):si.smooth(0.999) : min(720) : max(180) : int;
};
l = -10 : ba.db2linear;
C = 0.5;
};
harpe(C,N,b) = _ <: par(i, N, position(i+1)
:string(C,Major(b).degree2Hz(i), att, lvl)
: pan((i+0.5)/N) )
:> _,_
with {
att = hslider("h:[2]Parameters/[2]Resonance[style:knob][acc:2 1 -12 0 10]", 5, 0.1, 10, 0.01):min(10):max(0.1);
lvl = 1;
pan(p) = _ <: *(sqrt(1-p)), *(sqrt(p));
position(a,x) = abs(x - a) < 0.5;
};
Major(key) = environment {
A4Hz = 440;
degree2midi(0) = key+0;
degree2midi(1) = key+2;
degree2midi(2) = key+4;
degree2midi(3) = key+5;
degree2midi(4) = key+7;
degree2midi(5) = key+9;
degree2midi(6) = key+11;
degree2midi(7) = key+12;
degree2midi(d) = degree2midi(d-8)+12;
degree2Hz(d) = A4Hz*semiton(degree2midi(d)-69) with { semiton(n) = 2.0^(n/12.0); };
};
string(coef, freq, t60, level, trig) = no.noise*level
: *(trig : trigger(freq2samples(freq)))
: resonator(freq2samples(freq), att)
with {
resonator(d,a) = (+ : @(d-1)) ~ (average : *(a));
average(x) = (x*(1+coef)+x'*(1-coef))/2;
trigger(n) = upfront : + ~ decay(n) : >(0.0);
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
freq2samples(f) = 44100.0/f;
};
instrReverbHarp = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[3]Reverb/ Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1,0.01):si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[3]Reverb/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -10 0 10]", 0.2,0.05,1.3,0.01):min(1.3):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
933392d183dbe3c6272dc2cdc7a9c2b70b091e3dc35619f442f696a7cc145d77 | RuolunWeng/ruolunweng.github.io | SPulsaxophone.dsp | declare name "Solo Pulsaxophone";
declare author "ER"; //From Saxophone by Romain Michon;
/* =============== DESCRIPTION ================= :
- Pulsing saxophone
- Head = High frequencies
- Bottom = Low frequencies
- Right = Fast
- Left = Slow
- Head to Right, Head to Left = interesting transitions
*/
import("stdfaust.lib");
instrument = library("instruments.lib");
//==================== INSTRUMENT =======================
process = vgroup("PULSAXO",
(bodyFilter,breathPressure : instrumentBody) ~
(delay1 : NLFM) : !,_);
//==================== GUI SPECIFICATION ================
freq = hslider("h:Instrument/Frequency[unit:Hz][acc:1 1 -12 0 10]", 110,80,880,1):si.smooth(0.9999):min(880):max(80);
gate = pulsaxo.gate;
pressure = 0.83;
reedStiffness = 0.53;
blowPosition = 0.43;
noiseGain = 0.0001;
typeModulation = 4;
nonLinearity = 0.36;
frequencyMod = 20;
nonLinAttack = 0.12;
vibratoFreq = hslider("h:Parameters/Vibrato Frequency[style:knob][unit:Hz][acc:0 1 -10 0 10]", 6,1,15,0.1):si.smooth(0.999);
vibratoGain = 0.2;
vibratoBegin = 0.05;
vibratoAttack = 0.03;
vibratoRelease = 0.1;
envelopeAttack = 0.58;
envelopeRelease = 0.1;
//==================== SIGNAL PROCESSING ================
//----------------------- Pulsar --------------------------------------
pulsaxo = environment{
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
fade = (0.5); // min > 0 pour eviter division par 0
speed = hslider ("h:[2]Pulse/[1]Speed (Granulator)[unit:Hz][style:knob][acc:0 1 -10 0 10]", 4,0.001,7,0.0001):fi.lowpass(1,1);
proba = hslider ("h:[2]Pulse/[2]Probability (Granulator)[unit:%][style:knob][acc:1 0 -10 0 10]", 88,75,100,1)*(0.01):fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
//----------------------- Nonlinear filter ----------------------------
//nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
//nonlinear filter order
nlfOrder = 6;
//attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
envelopeMod = en.asr(nonLinAttack,1,envelopeRelease,gate);
//nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
//for using it with waveguide instruments
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//reed table parameters
reedTableOffset = 0.7;
reedTableSlope = 0.1 + (0.4*reedStiffness);
//the reed function is declared in instrument.lib
reedTable = instrument.reed(reedTableOffset,reedTableSlope);
//Delay lines length in number of samples
fdel1 = (1-blowPosition) * (ma.SR/freq - 3);
fdel2 = (ma.SR/freq - 3)*blowPosition +1 ;
//Delay lines
delay1 = de.fdelay(4096,fdel1);
delay2 = de.fdelay(4096,fdel2);
//Breath pressure is controlled by an attack / sustain / release envelope (en.asr is declared in instrument.lib)
envelope = (0.55+pressure*0.3)*en.asr(pressure*envelopeAttack,1,pressure*envelopeRelease,gate);
breath = envelope + envelope*noiseGain*no.noise;
//instrument.envVibrato is decalred in instrument.lib
vibrato = vibratoGain*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*os.osc(vibratoFreq);
breathPressure = breath + breath*vibratoGain*os.osc(vibratoFreq);
//Body filter is a one zero filter (declared in instrument.lib)
bodyFilter = *(gain) : instrument.oneZero1(b0,b1)
with {
gain = -0.95;
b0 = 0.5;
b1 = 0.5;
};
instrumentBody(delay1FeedBack,breathP) = delay1FeedBack <: -(delay2) <:
((breathP - _ <: breathP - _*reedTable) - delay1FeedBack),_;
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SPulsaxophone.dsp | faust | From Saxophone by Romain Michon;
=============== DESCRIPTION ================= :
- Pulsing saxophone
- Head = High frequencies
- Bottom = Low frequencies
- Right = Fast
- Left = Slow
- Head to Right, Head to Left = interesting transitions
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
==================== SIGNAL PROCESSING ================
----------------------- Pulsar --------------------------------------
min > 0 pour eviter division par 0
----------------------- Nonlinear filter ----------------------------
nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
nonlinear filter order
attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
for using it with waveguide instruments
----------------------- Synthesis parameters computing and functions declaration ----------------------------
reed table parameters
the reed function is declared in instrument.lib
Delay lines length in number of samples
Delay lines
Breath pressure is controlled by an attack / sustain / release envelope (en.asr is declared in instrument.lib)
instrument.envVibrato is decalred in instrument.lib
Body filter is a one zero filter (declared in instrument.lib) | declare name "Solo Pulsaxophone";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("PULSAXO",
(bodyFilter,breathPressure : instrumentBody) ~
(delay1 : NLFM) : !,_);
freq = hslider("h:Instrument/Frequency[unit:Hz][acc:1 1 -12 0 10]", 110,80,880,1):si.smooth(0.9999):min(880):max(80);
gate = pulsaxo.gate;
pressure = 0.83;
reedStiffness = 0.53;
blowPosition = 0.43;
noiseGain = 0.0001;
typeModulation = 4;
nonLinearity = 0.36;
frequencyMod = 20;
nonLinAttack = 0.12;
vibratoFreq = hslider("h:Parameters/Vibrato Frequency[style:knob][unit:Hz][acc:0 1 -10 0 10]", 6,1,15,0.1):si.smooth(0.999);
vibratoGain = 0.2;
vibratoBegin = 0.05;
vibratoAttack = 0.03;
vibratoRelease = 0.1;
envelopeAttack = 0.58;
envelopeRelease = 0.1;
pulsaxo = environment{
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
speed = hslider ("h:[2]Pulse/[1]Speed (Granulator)[unit:Hz][style:knob][acc:0 1 -10 0 10]", 4,0.001,7,0.0001):fi.lowpass(1,1);
proba = hslider ("h:[2]Pulse/[2]Probability (Granulator)[unit:%][style:knob][acc:1 0 -10 0 10]", 88,75,100,1)*(0.01):fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
nlfOrder = 6;
envelopeMod = en.asr(nonLinAttack,1,envelopeRelease,gate);
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
reedTableOffset = 0.7;
reedTableSlope = 0.1 + (0.4*reedStiffness);
reedTable = instrument.reed(reedTableOffset,reedTableSlope);
fdel1 = (1-blowPosition) * (ma.SR/freq - 3);
fdel2 = (ma.SR/freq - 3)*blowPosition +1 ;
delay1 = de.fdelay(4096,fdel1);
delay2 = de.fdelay(4096,fdel2);
envelope = (0.55+pressure*0.3)*en.asr(pressure*envelopeAttack,1,pressure*envelopeRelease,gate);
breath = envelope + envelope*noiseGain*no.noise;
vibrato = vibratoGain*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*os.osc(vibratoFreq);
breathPressure = breath + breath*vibratoGain*os.osc(vibratoFreq);
bodyFilter = *(gain) : instrument.oneZero1(b0,b1)
with {
gain = -0.95;
b0 = 0.5;
b1 = 0.5;
};
instrumentBody(delay1FeedBack,breathP) = delay1FeedBack <: -(delay2) <:
((breathP - _ <: breathP - _*reedTable) - delay1FeedBack),_;
|
9b3efb99b60d3f1ebaef5f76a11190176dd16826ff96b8101457d521ec73b05b | RuolunWeng/ruolunweng.github.io | Clarinet.dsp | declare name "Clarinet";
declare description "Nonlinear WaveGuide Clarinet";
declare author "Romain Michon";
declare copyright "Romain Michon ([email protected])";
declare version "1.0";
declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
declare description "A simple clarinet physical model, as discussed by Smith (1986), McIntyre, Schumacher, Woodhouse (1983), and others.";
declare reference "https://ccrma.stanford.edu/~jos/pasp/Woodwinds.html";
//Modification Grame July 2015
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- Clarinet responding to vigorous gestures
- Turn ON the Clarinet
- Head = High frequencies/Reverberation/Silence when hold still
- Tilt = very soft sound
- Bottom = Low frequencies
- Right = Breathy clarinet
- Fishing rod (vigorous mouvements) :
==> Downward = to reach lower frequencies
==> Upward = To 'through' the sound in the air = vanishes, comes back when Tilt
- Rocking = from full sound to breathy sound
- Shaking in right position = no.noise impulses
*/
//==================== INSTRUMENT =======================
process = vgroup("CLARINET",
//Commuted Loss Filtering
(_,(breathPressure <: _,_) : (filter*-0.95 - _ <:
//Non-Linear Scattering
*(reedTable)) + _) ~
//Delay with Feedback
(delayLine):// : NLFM) :
//scaling and stereo
*(gain)*1.5 <: instrReverbCla);
//==================== GUI SPECIFICATION ================
freq = hslider("h:[2]Instrument/Frequency[unit:Hz][tooltip:Tone frequency][acc:1 1 -14 0 12]", 440,110,1300,0.01):si.smooth(0.999);
gain = 1;
gate = hslider("[1]ON/OFF (ASR Envelope)",0,0,1,1);
reedStiffness = hslider("h:[3]Parameters/Instrument Stiffness[style:knob][acc:0 1 -12 0 12]", 0.25,0.01,1,0.01);
noiseGain = hslider("h:[3]Parameters/Breath Noise[style:knob][acc:0 1 -12 0 12]", 0.02,0,0.12,0.01);
pressure = hslider("h:[3]Parameters/ Pressure[style:knob][acc:1 0 -5 0 10]", 0.8,0.65,1,0.01);
vibratoFreq = 5;
vibratoGain = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.01;
envelopeAttack = 0.1;
envelopeDecay = 0.05;
envelopeRelease = 0.1;
//==================== SIGNAL PROCESSING ======================
//----------------------- Synthesis PARAMETERS computing and functions declaration ----------------------------
//instrument.reed table PARAMETERS
reedTableOffset = 0.7;
reedTableSlope = -0.44 + (0.26*reedStiffness);
//the instrument.reed function is declared in INSTRUMENT.lib
reedTable = instrument.reed(reedTableOffset,reedTableSlope);
//de.delay line with a length adapted in function of the order of nonlinear filter
delayLength = ma.SR/freq*0.5 - 1.5;// - (nlfOrder*nonLinearity)*(typeModulation < 2);
delayLine = de.fdelay(4096,delayLength);
//one zero filter used as a allpass: pole is set to -1
filter = instrument.oneZero0(0.5,0.5);
//----------------------- Algorithm implementation ----------------------------
//Breath pressure + vibrato + breath no.noise + envelope (Attack / Decay / Sustain / Release)
envelope = en.adsr(envelopeAttack,envelopeDecay,1,envelopeRelease,gate)*pressure*0.9;
vibrato = os.osc(vibratoFreq)*vibratoGain*
instrument.envVibrato(0.1*2*vibratoAttack,0.9*2*vibratoAttack,100,vibratoRelease,gate);
breath = envelope + envelope*no.noise*noiseGain;
breathPressure = breath + breath*vibrato;
//----------------------- INSTRREVERB ----------------------------
instrReverbCla = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[4]Reverb/ Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -15 0 15]", 0.137,0.05,1,0.01) :si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[4]Reverb/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -15 0 15]", 0.45,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/Clarinet.dsp | faust | Synthesis Tool Kit 4.3 (MIT style license);
Modification Grame July 2015
=============== DESCRIPTION ================= :
- Clarinet responding to vigorous gestures
- Turn ON the Clarinet
- Head = High frequencies/Reverberation/Silence when hold still
- Tilt = very soft sound
- Bottom = Low frequencies
- Right = Breathy clarinet
- Fishing rod (vigorous mouvements) :
==> Downward = to reach lower frequencies
==> Upward = To 'through' the sound in the air = vanishes, comes back when Tilt
- Rocking = from full sound to breathy sound
- Shaking in right position = no.noise impulses
==================== INSTRUMENT =======================
Commuted Loss Filtering
Non-Linear Scattering
Delay with Feedback
: NLFM) :
scaling and stereo
==================== GUI SPECIFICATION ================
==================== SIGNAL PROCESSING ======================
----------------------- Synthesis PARAMETERS computing and functions declaration ----------------------------
instrument.reed table PARAMETERS
the instrument.reed function is declared in INSTRUMENT.lib
de.delay line with a length adapted in function of the order of nonlinear filter
- (nlfOrder*nonLinearity)*(typeModulation < 2);
one zero filter used as a allpass: pole is set to -1
----------------------- Algorithm implementation ----------------------------
Breath pressure + vibrato + breath no.noise + envelope (Attack / Decay / Sustain / Release)
----------------------- INSTRREVERB ---------------------------- | declare name "Clarinet";
declare description "Nonlinear WaveGuide Clarinet";
declare author "Romain Michon";
declare copyright "Romain Michon ([email protected])";
declare version "1.0";
declare description "A simple clarinet physical model, as discussed by Smith (1986), McIntyre, Schumacher, Woodhouse (1983), and others.";
declare reference "https://ccrma.stanford.edu/~jos/pasp/Woodwinds.html";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("CLARINET",
(_,(breathPressure <: _,_) : (filter*-0.95 - _ <:
*(reedTable)) + _) ~
*(gain)*1.5 <: instrReverbCla);
freq = hslider("h:[2]Instrument/Frequency[unit:Hz][tooltip:Tone frequency][acc:1 1 -14 0 12]", 440,110,1300,0.01):si.smooth(0.999);
gain = 1;
gate = hslider("[1]ON/OFF (ASR Envelope)",0,0,1,1);
reedStiffness = hslider("h:[3]Parameters/Instrument Stiffness[style:knob][acc:0 1 -12 0 12]", 0.25,0.01,1,0.01);
noiseGain = hslider("h:[3]Parameters/Breath Noise[style:knob][acc:0 1 -12 0 12]", 0.02,0,0.12,0.01);
pressure = hslider("h:[3]Parameters/ Pressure[style:knob][acc:1 0 -5 0 10]", 0.8,0.65,1,0.01);
vibratoFreq = 5;
vibratoGain = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.01;
envelopeAttack = 0.1;
envelopeDecay = 0.05;
envelopeRelease = 0.1;
reedTableOffset = 0.7;
reedTableSlope = -0.44 + (0.26*reedStiffness);
reedTable = instrument.reed(reedTableOffset,reedTableSlope);
delayLine = de.fdelay(4096,delayLength);
filter = instrument.oneZero0(0.5,0.5);
envelope = en.adsr(envelopeAttack,envelopeDecay,1,envelopeRelease,gate)*pressure*0.9;
vibrato = os.osc(vibratoFreq)*vibratoGain*
instrument.envVibrato(0.1*2*vibratoAttack,0.9*2*vibratoAttack,100,vibratoRelease,gate);
breath = envelope + envelope*no.noise*noiseGain;
breathPressure = breath + breath*vibrato;
instrReverbCla = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[4]Reverb/ Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -15 0 15]", 0.137,0.05,1,0.01) :si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[4]Reverb/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -15 0 15]", 0.45,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
efc8381e3ffab53ef243bb0dad2151ae3ad488ba7a353cfa84fc26085d66a0dd | RuolunWeng/ruolunweng.github.io | Pulsaxophone.dsp | declare name "Pulsaxophone";
declare author "ER"; //From Saxophone by Romain Michon;
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ================= :
- Pulsing saxophone
- Head = High frequencies + Reverb
- Bottom = Low frequencies
- Right = Fast
- Left = Slow
- Head to Right, Head to Left = interesting transitions
*/
//==================== INSTRUMENT =======================
process = vgroup("Pulsaxo",
(bodyFilter,breathPressure : instrumentBody) ~
(delay1 : NLFM) : !,fi.lowpass(1,1000)
//Scaling Output and stereo
*(gain) :>_<: instrReverbAccel);
//==================== GUI SPECIFICATION ================
freq = hslider("[1]Frequency[unit:Hz][acc:1 1 -12 0 10]", 110,80,880,1):si.smooth(0.9999):min(880):max(80);
gain = 0.8;
gate = pulsaxo.gate;
pressure = 0.83;
reedStiffness = 0.53;
blowPosition = 0.43;
noiseGain = 0.0001;
typeModulation = 4;
nonLinearity = 0.36;
frequencyMod = 20;
nonLinAttack = 0.12;
vibratoFreq = hslider("[3]Vibrato Frequency[style:knob][unit:Hz][acc:0 1 -10 0 10]", 6,1,15,0.1):si.smooth(0.999);
vibratoGain = 0.2;
vibratoBegin = 0.05;
vibratoAttack = 0.03;
vibratoRelease = 0.1;
envelopeAttack = 0.58;
envelopeRelease = 0.1;
//==================== SIGNAL PROCESSING ================
//----------------------- Pulsar --------------------------------------
pulsaxo = environment {
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
fade = (0.5); // min > 0 pour eviter division par 0
speed = hslider ("h:[2]Pulse/[1]Speed (Granulator)[unit:Hz][style:knob][acc:0 1 -10 0 10]", 4,0.001,7,0.0001):fi.lowpass(1,1);
proba = hslider ("h:[2]Pulse/[2]Probability (Granulator)[unit:%][style:knob][acc:1 0 -10 0 10]", 88,75,100,1)*(0.01):fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
//----------------------- Nonlinear filter ----------------------------
//nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
//nonlinear filter order
nlfOrder = 6;
//attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
envelopeMod = en.asr(nonLinAttack,1,envelopeRelease,gate);
//nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
//for using it with waveguide instruments
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//instrument.reed table parameters
reedTableOffset = 0.7;
reedTableSlope = 0.1 + (0.4*reedStiffness);
//the instrument.reed function is declared in instrument.lib
reedTable = instrument.reed(reedTableOffset,reedTableSlope);
//Delay lines length in number of samples
fdel1 = (1-blowPosition) * (ma.SR/freq - 3);
fdel2 = (ma.SR/freq - 3)*blowPosition +1 ;
//Delay lines
delay1 = de.fdelay(4096,fdel1);
delay2 = de.fdelay(4096,fdel2);
//Breath pressure is controlled by an attack / sustain / release envelope (en.asr is declared in instrument.lib)
envelope = (0.55+pressure*0.3)*en.asr(pressure*envelopeAttack,1,pressure*envelopeRelease,gate);
breath = envelope + envelope*noiseGain*no.noise;
//instrument.envVibrato is decalred in instrument.lib
vibrato = vibratoGain*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*os.osc(vibratoFreq);
breathPressure = breath + breath*vibratoGain*os.osc(vibratoFreq);
//Body filter is a one zero filter (declared in instrument.lib)
bodyFilter = *(gain) : instrument.oneZero1(b0,b1)
with {
gain = -0.95;
b0 = 0.5;
b1 = 0.5;
};
instrumentBody(delay1FeedBack,breathP) = delay1FeedBack <: -(delay2) <:
((breathP - _ <: breathP - _*reedTable) - delay1FeedBack),_;
instrReverbAccel = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
roomSize = hslider("h:[4]Reverb/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -15 0 12]", 0.72,0.1,1.7,0.01):min(1.7):max(0.1);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/Pulsaxophone.dsp | faust | From Saxophone by Romain Michon;
=============== DESCRIPTION ================= :
- Pulsing saxophone
- Head = High frequencies + Reverb
- Bottom = Low frequencies
- Right = Fast
- Left = Slow
- Head to Right, Head to Left = interesting transitions
==================== INSTRUMENT =======================
Scaling Output and stereo
==================== GUI SPECIFICATION ================
==================== SIGNAL PROCESSING ================
----------------------- Pulsar --------------------------------------
min > 0 pour eviter division par 0
----------------------- Nonlinear filter ----------------------------
nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
nonlinear filter order
attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
for using it with waveguide instruments
----------------------- Synthesis parameters computing and functions declaration ----------------------------
instrument.reed table parameters
the instrument.reed function is declared in instrument.lib
Delay lines length in number of samples
Delay lines
Breath pressure is controlled by an attack / sustain / release envelope (en.asr is declared in instrument.lib)
instrument.envVibrato is decalred in instrument.lib
Body filter is a one zero filter (declared in instrument.lib) | declare name "Pulsaxophone";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("Pulsaxo",
(bodyFilter,breathPressure : instrumentBody) ~
(delay1 : NLFM) : !,fi.lowpass(1,1000)
*(gain) :>_<: instrReverbAccel);
freq = hslider("[1]Frequency[unit:Hz][acc:1 1 -12 0 10]", 110,80,880,1):si.smooth(0.9999):min(880):max(80);
gain = 0.8;
gate = pulsaxo.gate;
pressure = 0.83;
reedStiffness = 0.53;
blowPosition = 0.43;
noiseGain = 0.0001;
typeModulation = 4;
nonLinearity = 0.36;
frequencyMod = 20;
nonLinAttack = 0.12;
vibratoFreq = hslider("[3]Vibrato Frequency[style:knob][unit:Hz][acc:0 1 -10 0 10]", 6,1,15,0.1):si.smooth(0.999);
vibratoGain = 0.2;
vibratoBegin = 0.05;
vibratoAttack = 0.03;
vibratoRelease = 0.1;
envelopeAttack = 0.58;
envelopeRelease = 0.1;
pulsaxo = environment {
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
speed = hslider ("h:[2]Pulse/[1]Speed (Granulator)[unit:Hz][style:knob][acc:0 1 -10 0 10]", 4,0.001,7,0.0001):fi.lowpass(1,1);
proba = hslider ("h:[2]Pulse/[2]Probability (Granulator)[unit:%][style:knob][acc:1 0 -10 0 10]", 88,75,100,1)*(0.01):fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
nlfOrder = 6;
envelopeMod = en.asr(nonLinAttack,1,envelopeRelease,gate);
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
reedTableOffset = 0.7;
reedTableSlope = 0.1 + (0.4*reedStiffness);
reedTable = instrument.reed(reedTableOffset,reedTableSlope);
fdel1 = (1-blowPosition) * (ma.SR/freq - 3);
fdel2 = (ma.SR/freq - 3)*blowPosition +1 ;
delay1 = de.fdelay(4096,fdel1);
delay2 = de.fdelay(4096,fdel2);
envelope = (0.55+pressure*0.3)*en.asr(pressure*envelopeAttack,1,pressure*envelopeRelease,gate);
breath = envelope + envelope*noiseGain*no.noise;
vibrato = vibratoGain*instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*os.osc(vibratoFreq);
breathPressure = breath + breath*vibratoGain*os.osc(vibratoFreq);
bodyFilter = *(gain) : instrument.oneZero1(b0,b1)
with {
gain = -0.95;
b0 = 0.5;
b1 = 0.5;
};
instrumentBody(delay1FeedBack,breathP) = delay1FeedBack <: -(delay2) <:
((breathP - _ <: breathP - _*reedTable) - delay1FeedBack),_;
instrReverbAccel = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
roomSize = hslider("h:[4]Reverb/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -15 0 12]", 0.72,0.1,1.7,0.01):min(1.7):max(0.1);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
53b6a7e4fdcf4b39886b4fc03dbf9b0cf900e686ed072a6bec0fcc2121d58635 | RuolunWeng/ruolunweng.github.io | Flute.dsp | declare name "Flute";
declare description "Nonlinear WaveGuide Flute";
declare author "Romain Michon ([email protected])";
declare copyright "Romain Michon";
declare version "1.0";
declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
declare description "A simple flute based on Smith algorythm: https://ccrma.stanford.edu/~jos/pasp/Flutes_Recorders_Pipe_Organs.html";
//Modifications GRAME July 2015
/* =========== DESCRITPION ===========
- Flute
- Turn ON flute (0=OFF, 1=ON)
- Head = High frequencies/ Reverberation/ Silence
- Bottom = Low frequencies
- Left = No vibrato
- Right = Fast vibrato
- Front = Full sound
- Back = Breathy sound
*/
import("stdfaust.lib");
instrument = library("instruments.lib");
//==================== INSTRUMENT =======================
flute = (_ <: (flow + *(feedBack1) : embouchureDelay: poly) + *(feedBack2) : reflexionFilter)~(boreDelay) : NLFM : *(env2)*gain;
process = flute <:instrReverbFlute;
//==================== GUI SPECIFICATION ================
freq = hslider("[1]Frequency[acc:1 1 -10 0 10]", 440,247,1200,1):si.smooth(0.999);
pressure = hslider("[2]Pressure[style:knob][acc:1 0 -10 0 10]", 0.96, 0.2, 0.99, 0.01):si.smooth(0.999):min(0.99):max(0.2);
breathAmp = hslider("[3]Breath Noise[style:knob][acc:2 0 -10 0 10]", 0.02, 0.01, 0.2, 0.01):si.smooth(0.999):min(0.2):max(0.01);
gate = hslider("[0]ON/OFF (ASR Envelope)",0,0,1,1);
vibratoFreq = hslider("[4]Vibrato Freq (Vibrato Envelope)[style:knob][unit:Hz][acc:0 1 -10 0 10]", 4,0.5,8,0.1);
env1Attack = 0.1;//hslider("h:Parameters/Press_Env_Attack[unit:s][style:knob][acc:1 0 -10 0 10][tooltip:Pressure envelope attack duration]",0.05,0.05,0.2,0.01);
//-------------------- Non-Variable Parameters -----------
gain = 1;
typeModulation = 0;
nonLinearity = 0;
frequencyMod = 220;
nonLinAttack = 0.1;
vibratoGain = 0.05;
vibratoBegin = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
env1Release = 0.5;
//==================== SIGNAL PROCESSING ================
//----------------------- Nonlinear filter ----------------------------
//nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
//nonlinear filter order
nlfOrder = 6;
//attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
envelopeMod = en.asr(nonLinAttack,1,0.1,gate);
//nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
//for using it with waveguide instruments
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//Loops feedbacks gains
feedBack1 = 0.4;
feedBack2 = 0.4;
//Delay Lines
embouchureDelayLength = (ma.SR/freq)/2-2;
boreDelayLength = ma.SR/freq-2;
embouchureDelay = de.fdelay(4096,embouchureDelayLength);
boreDelay = de.fdelay(4096,boreDelayLength);
//Polinomial
poly = _ <: _ - _*_*_;
//jet filter is a lowwpass filter (declared in filter.lib)
reflexionFilter = fi.lowpass(1,2000);
//----------------------- Algorithm implementation ----------------------------
//Pressure envelope
env1 = en.adsr(env1Attack,env1Decay,0.9,env1Release,(gate | pressureEnvelope))*pressure*1.1;
//Global envelope
env2 = en.asr(env2Attack,1,env2Release,gate)*0.5;
//Vibrato Envelope
vibratoEnvelope = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*vibratoGain;
vibrato = os.osc(vibratoFreq)*vibratoEnvelope;
breath = no.noise*env1;
flow = env1 + breath*breathAmp + vibrato;
//------------------------ InstrReverb ----------------------------------------
instrReverbFlute = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[5]Reverb/[1]Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -30 0 13]", 0.2,0.05,1,0.01) : si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[5]Reverb/[2]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -30 0 13]", 0.72,0.05,1.7,0.01):min(1.7):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/Flute.dsp | faust | Synthesis Tool Kit 4.3 (MIT style license);
Modifications GRAME July 2015
=========== DESCRITPION ===========
- Flute
- Turn ON flute (0=OFF, 1=ON)
- Head = High frequencies/ Reverberation/ Silence
- Bottom = Low frequencies
- Left = No vibrato
- Right = Fast vibrato
- Front = Full sound
- Back = Breathy sound
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
hslider("h:Parameters/Press_Env_Attack[unit:s][style:knob][acc:1 0 -10 0 10][tooltip:Pressure envelope attack duration]",0.05,0.05,0.2,0.01);
-------------------- Non-Variable Parameters -----------
==================== SIGNAL PROCESSING ================
----------------------- Nonlinear filter ----------------------------
nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
nonlinear filter order
attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
for using it with waveguide instruments
----------------------- Synthesis parameters computing and functions declaration ----------------------------
Loops feedbacks gains
Delay Lines
Polinomial
jet filter is a lowwpass filter (declared in filter.lib)
----------------------- Algorithm implementation ----------------------------
Pressure envelope
Global envelope
Vibrato Envelope
------------------------ InstrReverb ---------------------------------------- | declare name "Flute";
declare description "Nonlinear WaveGuide Flute";
declare author "Romain Michon ([email protected])";
declare copyright "Romain Michon";
declare version "1.0";
declare description "A simple flute based on Smith algorythm: https://ccrma.stanford.edu/~jos/pasp/Flutes_Recorders_Pipe_Organs.html";
import("stdfaust.lib");
instrument = library("instruments.lib");
flute = (_ <: (flow + *(feedBack1) : embouchureDelay: poly) + *(feedBack2) : reflexionFilter)~(boreDelay) : NLFM : *(env2)*gain;
process = flute <:instrReverbFlute;
freq = hslider("[1]Frequency[acc:1 1 -10 0 10]", 440,247,1200,1):si.smooth(0.999);
pressure = hslider("[2]Pressure[style:knob][acc:1 0 -10 0 10]", 0.96, 0.2, 0.99, 0.01):si.smooth(0.999):min(0.99):max(0.2);
breathAmp = hslider("[3]Breath Noise[style:knob][acc:2 0 -10 0 10]", 0.02, 0.01, 0.2, 0.01):si.smooth(0.999):min(0.2):max(0.01);
gate = hslider("[0]ON/OFF (ASR Envelope)",0,0,1,1);
vibratoFreq = hslider("[4]Vibrato Freq (Vibrato Envelope)[style:knob][unit:Hz][acc:0 1 -10 0 10]", 4,0.5,8,0.1);
gain = 1;
typeModulation = 0;
nonLinearity = 0;
frequencyMod = 220;
nonLinAttack = 0.1;
vibratoGain = 0.05;
vibratoBegin = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
env1Release = 0.5;
nlfOrder = 6;
envelopeMod = en.asr(nonLinAttack,1,0.1,gate);
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
feedBack1 = 0.4;
feedBack2 = 0.4;
embouchureDelayLength = (ma.SR/freq)/2-2;
boreDelayLength = ma.SR/freq-2;
embouchureDelay = de.fdelay(4096,embouchureDelayLength);
boreDelay = de.fdelay(4096,boreDelayLength);
poly = _ <: _ - _*_*_;
reflexionFilter = fi.lowpass(1,2000);
env1 = en.adsr(env1Attack,env1Decay,0.9,env1Release,(gate | pressureEnvelope))*pressure*1.1;
env2 = en.asr(env2Attack,1,env2Release,gate)*0.5;
vibratoEnvelope = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*vibratoGain;
vibrato = os.osc(vibratoFreq)*vibratoEnvelope;
breath = no.noise*env1;
flow = env1 + breath*breathAmp + vibrato;
instrReverbFlute = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("h:[5]Reverb/[1]Reverberation Volume (InstrReverb)[style:knob][acc:1 1 -30 0 13]", 0.2,0.05,1,0.01) : si.smooth(0.999):min(1):max(0.05);
roomSize = hslider("h:[5]Reverb/[2]Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -30 0 13]", 0.72,0.05,1.7,0.01):min(1.7):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
50a2a649e9e081b2315e221843bf39c7ad2cf1d3f3a6af1b51cae980e14a1cea | RuolunWeng/ruolunweng.github.io | RandomFlute.dsp | declare name "Random Flute";
declare author "ER";//Adapted from "Nonlinear WaveGuide Flute" by Romain Michon ([email protected]);
import("stdfaust.lib");
instrument = library("instruments.lib");
/* ============== DESCRIPTION ================
- Random frequency flute
- Left = Slow rhythm/long notes/silence
- Right = Fast rhythm/short note
- Head = Reverberation
- Back = Echo
*/
//==================== INSTRUMENT =======================
flute = (_ <: (flow + *(feedBack1) : embouchureDelay: poly) + *(feedBack2) : reflexionFilter)~(boreDelay) : NLFM : *(env2)*gain:_;
process = flute : echo <: instrReverbFlute;
//==================== GUI SPECIFICATION ================
pressure = 1;
breathAmp = hslider("h:[3]Parameters/Breath Noise[style:knob][acc:0 1 -10 0 10]", 0.02, 0.01, 0.05, 0.0001):si.smooth(0.999):min(0.05):max(0.01);
gate = pulsaflute.gate;
vibratoFreq = 5;
env1Attack = 0.05;
//--------------------------- Random Frequency ---------------------------
freq = gate : randfreq : si.smooth(0.99) : fi.lowpass (1, 3000);
randfreq(g) = no.noise : sampleAndhold(sahgate(g))*(1500)+(100)
with{
sampleAndhold(t) = select2(t) ~_;
sahgate(g) = g : upfront : counter -(3) <=(0);
upfront(x) = abs(x-x')>0.5;
counter(g) = (+(1):*(1-g))~_;
};
//----------------------- Echo ----------------------------------------
echo = +~ @(22050) *(feedback);
feedback = hslider("h:Parameters/Echo Intensity[style:knob][acc:2 0 -10 0 10]", 0.001, 0.001, 0.65, 0.001):si.smooth(0.999);
//----------------------- Pulsar --------------------------------------
pulsaflute = environment {
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
fade = (0.5); // min > 0 pour eviter division par 0
speed = hslider ("h:[1]Pulse/[1]Speed (Granulator)[unit:Hz][style:knob][acc:0 1 -10 0 10]", 3,1,6,0.0001):fi.lowpass(1,1);
proba = hslider ("h:[1]Pulse/[2]Probability (Granulator)[unit:%][style:knob][acc:0 1 -10 0 10]", 88,60,100,1) *(0.01):fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
//-------------------- Non-Variable Parameters -----------
N = 27;
gain = 1;
typeModulation = 0;
nonLinearity = 0;
frequencyMod = 220;
nonLinAttack = 0.1;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
env1Release = 0.5;
//==================== SIGNAL PROCESSING ================
//----------------------- Nonlinear filter ----------------------------
//nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
//nonlinear filter order
nlfOrder = 6;
//attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
envelopeMod = en.asr(nonLinAttack,1,0.1,gate);
//nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
//for using it with waveguide instruments
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//Loops feedbacks gains
feedBack1 = 0.4;
feedBack2 = 0.4;
//Delay Lines
embouchureDelayLength = (ma.SR/freq)/2-2;
boreDelayLength = ma.SR/freq-2;
embouchureDelay = de.fdelay(4096,embouchureDelayLength);
boreDelay = de.fdelay(4096,boreDelayLength);
//Polinomial
poly = _ <: _ - _*_*_;
//jet filter is a lowwpass filter (declared in filter.lib)
reflexionFilter = fi.lowpass(1,2000);
//----------------------- Algorithm implementation ----------------------------
//Pressure envelope
env1 = en.adsr(env1Attack,env1Decay,0.9,env1Release,(gate | pressureEnvelope))*pressure*1.1;
//Global envelope
env2 = en.asr(env2Attack,1,env2Release,gate)*0.5;
//Vibrato Envelope
vibratoEnvelope = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*vibratoGain;
vibrato = os.osc(vibratoFreq)*vibratoEnvelope;
breath = no.noise*env1;
flow = env1 + breath*breathAmp + vibrato;
instrReverbFlute = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
roomSize = hslider("h:Reverb/Reverberation Room Size [style:knob][acc:1 1 -10 0 10]", 0.2,0.01,1.7,0.01);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/RandomFlute.dsp | faust | Adapted from "Nonlinear WaveGuide Flute" by Romain Michon ([email protected]);
============== DESCRIPTION ================
- Random frequency flute
- Left = Slow rhythm/long notes/silence
- Right = Fast rhythm/short note
- Head = Reverberation
- Back = Echo
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
--------------------------- Random Frequency ---------------------------
----------------------- Echo ----------------------------------------
----------------------- Pulsar --------------------------------------
min > 0 pour eviter division par 0
-------------------- Non-Variable Parameters -----------
==================== SIGNAL PROCESSING ================
----------------------- Nonlinear filter ----------------------------
nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
nonlinear filter order
attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
for using it with waveguide instruments
----------------------- Synthesis parameters computing and functions declaration ----------------------------
Loops feedbacks gains
Delay Lines
Polinomial
jet filter is a lowwpass filter (declared in filter.lib)
----------------------- Algorithm implementation ----------------------------
Pressure envelope
Global envelope
Vibrato Envelope | declare name "Random Flute";
import("stdfaust.lib");
instrument = library("instruments.lib");
flute = (_ <: (flow + *(feedBack1) : embouchureDelay: poly) + *(feedBack2) : reflexionFilter)~(boreDelay) : NLFM : *(env2)*gain:_;
process = flute : echo <: instrReverbFlute;
pressure = 1;
breathAmp = hslider("h:[3]Parameters/Breath Noise[style:knob][acc:0 1 -10 0 10]", 0.02, 0.01, 0.05, 0.0001):si.smooth(0.999):min(0.05):max(0.01);
gate = pulsaflute.gate;
vibratoFreq = 5;
env1Attack = 0.05;
freq = gate : randfreq : si.smooth(0.99) : fi.lowpass (1, 3000);
randfreq(g) = no.noise : sampleAndhold(sahgate(g))*(1500)+(100)
with{
sampleAndhold(t) = select2(t) ~_;
sahgate(g) = g : upfront : counter -(3) <=(0);
upfront(x) = abs(x-x')>0.5;
counter(g) = (+(1):*(1-g))~_;
};
echo = +~ @(22050) *(feedback);
feedback = hslider("h:Parameters/Echo Intensity[style:knob][acc:2 0 -10 0 10]", 0.001, 0.001, 0.65, 0.001):si.smooth(0.999);
pulsaflute = environment {
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
speed = hslider ("h:[1]Pulse/[1]Speed (Granulator)[unit:Hz][style:knob][acc:0 1 -10 0 10]", 3,1,6,0.0001):fi.lowpass(1,1);
proba = hslider ("h:[1]Pulse/[2]Probability (Granulator)[unit:%][style:knob][acc:0 1 -10 0 10]", 88,60,100,1) *(0.01):fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
N = 27;
gain = 1;
typeModulation = 0;
nonLinearity = 0;
frequencyMod = 220;
nonLinAttack = 0.1;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.1;
env2Release = 0.1;
env1Release = 0.5;
nlfOrder = 6;
envelopeMod = en.asr(nonLinAttack,1,0.1,gate);
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
feedBack1 = 0.4;
feedBack2 = 0.4;
embouchureDelayLength = (ma.SR/freq)/2-2;
boreDelayLength = ma.SR/freq-2;
embouchureDelay = de.fdelay(4096,embouchureDelayLength);
boreDelay = de.fdelay(4096,boreDelayLength);
poly = _ <: _ - _*_*_;
reflexionFilter = fi.lowpass(1,2000);
env1 = en.adsr(env1Attack,env1Decay,0.9,env1Release,(gate | pressureEnvelope))*pressure*1.1;
env2 = en.asr(env2Attack,1,env2Release,gate)*0.5;
vibratoEnvelope = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*vibratoGain;
vibrato = os.osc(vibratoFreq)*vibratoEnvelope;
breath = no.noise*env1;
flow = env1 + breath*breathAmp + vibrato;
instrReverbFlute = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
roomSize = hslider("h:Reverb/Reverberation Room Size [style:knob][acc:1 1 -10 0 10]", 0.2,0.01,1.7,0.01);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
bc688dec1a8f903f1d2316b3c8ef6acd02dfb540de983a2c10633f951e037d52 | RuolunWeng/ruolunweng.github.io | TibetanBowl.dsp | declare name "Tibetan Bowl";
declare description "Banded Waveguide Modeld Tibetan Bowl";
declare author "Romain Michon";
declare copyright "Romain Michon ([email protected])";
declare version "1.0";
declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
declare description "This instrument uses banded waveguide. For more information, see Essl, G. and Cook, P. Banded Waveguides: Towards Physical Modelling of Bar Percussion Instruments, Proceedings of the 1999 International Computer Music Conference.";
//Modification GRAME July 2015
import("stdfaust.lib");
instrument = library("instruments.lib");
/* ============ DESCRIPTION =============
- Tibetan Bowl
- Set the frequency manually
- Fishing rod/Front shaking = Ringing the bowl
- Right = maximum modulation
- Rocking = modulating the sound
- Head = Reverb
*/
//==================== INSTRUMENT =======================
process =
(((select-1)*-1) <:
//nModes resonances with nModes feedbacks for bow table look-up
par(i,nModes,(resonance(i)~_))):>+://~par(i,nModes,_) :> + :
//Signal Scaling and stereo
NLFM : stereo : instrReverbAccel:
*(vol),*(vol);
//==================== GUI SPECIFICATION ================
vol = 0.8;
freq = hslider("[1]Frequency[unit:Hz][tooltip:Tone frequency]",440,180,780,1);
gain = 0.5;
gate = 0;
select = hslider("[0]Play[tooltip:0=Bow; 1=Strike] [acc:2 1 -10 0 10]", 0,0,1,1);
baseGain = 0.5;
typeModulation = 3;
nonLinearity = hslider("[2]Modulation[acc:0 1 -10 0 10][tooltip:Nonlinearity factor (value between 0 and 1)]",0.02,0,0.1,0.001):si.smooth(0.999);
frequencyMod = hslider("[3]Modulation Frequency[unit:Hz][acc:0 0 -10 0 10]", 220,150,500,0.1):si.smooth(0.999);
nonLinAttack = 0.1;
//==================== MODAL PARAMETERS ================
preset = 0;
nMode(0) = 12;
modes(0,0) = 0.996108344;
basegains(0,0) = 0.999925960128219;
excitation(0,0) = 11.900357 / 10;
modes(0,1) = 1.0038916562;
basegains(0,1) = 0.999925960128219;
excitation(0,1) = 11.900357 / 10;
modes(0,2) = 2.979178;
basegains(0,2) = 0.999982774366897;
excitation(0,2) = 10.914886 / 10;
modes(0,3) = 2.99329767;
basegains(0,3) = 0.999982774366897;
excitation(0,3) = 10.914886 / 10;
modes(0,4) = 5.704452;
basegains(0,4) = 1.0;
excitation(0,4) = 42.995041 / 10;
modes(0,5) = 5.704452;
basegains(0,5) = 1.0;
excitation(0,5) = 42.995041 / 10;
modes(0,6) = 8.9982;
basegains(0,6) = 1.0;
excitation(0,6) = 40.063034 / 10;
modes(0,7) = 9.01549726;
basegains(0,7) = 1.0;
excitation(0,7) = 40.063034 / 10;
modes(0,8) = 12.83303;
basegains(0,8) = 0.999965497558225;
excitation(0,8) = 7.063034 / 10;
modes(0,9) = 12.807382;
basegains(0,9) = 0.999965497558225;
excitation(0,9) = 7.063034 / 10;
modes(0,10) = 17.2808219;
basegains(0,10) = 0.9999999999999999999965497558225;
excitation(0,10) = 57.063034 / 10;
modes(0,11) = 21.97602739726;
basegains(0,11) = 0.999999999999999965497558225;
excitation(0,11) = 57.063034 / 10;
//==================== SIGNAL PROCESSING ================
//----------------------- Nonlinear filter ----------------------------
//nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
//nonlinear filter order
nlfOrder = 6;
//nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
//for using it with waveguide instruments
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),1,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//the number of modes depends on the preset being used
nModes = nMode(preset);
delayLengthBase = ma.SR/freq;
//de.delay lengths in number of samples
delayLength(x) = delayLengthBase/modes(preset,x);
//de.delay lines
delayLine(x) = de.delay(4096,delayLength(x));
//Filter bank: fi.bandpass filters (declared in instrument.lib)
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x) = instrument.bandPass(freq*modes(preset,x),radius);
stereoo(periodDuration) = _ <: _,widthdelay : stereopanner
with {
W = 0.5;
A = 0.6;
widthdelay = de.delay(4096,W*periodDuration/2);
stereopanner = _,_ : *(1.0-A), *(A);
};
stereo = stereoo(delayLengthBase);
//----------------------- Algorithm implementation ----------------------------
//One resonance
resonance(x) = + : + (excitation(preset,x)*select) : delayLine(x) : *(basegains(preset,x)) : bandPassFilter(x);
//----------------------- Reverb (ajout accelerometre 05/2015) ----------------
instrReverbAccel = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("v:[4]Reverb/[1]Reverberation Volume (InstrReverb) [acc:1 1 -10 0 10]",0.2,0.02,1,0.01) : si.smooth(0.999) :min(1):max(0.02);
roomSize = hslider("v:[4]Reverb/[2]Reverberation Room Size (InstrReverb)[acc:1 1 -10 0 10]", 0.2,0.02,2,0.01):min(2):max(0.02);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/TibetanBowl.dsp | faust | Synthesis Tool Kit 4.3 (MIT style license);
Modification GRAME July 2015
============ DESCRIPTION =============
- Tibetan Bowl
- Set the frequency manually
- Fishing rod/Front shaking = Ringing the bowl
- Right = maximum modulation
- Rocking = modulating the sound
- Head = Reverb
==================== INSTRUMENT =======================
nModes resonances with nModes feedbacks for bow table look-up
~par(i,nModes,_) :> + :
Signal Scaling and stereo
==================== GUI SPECIFICATION ================
==================== MODAL PARAMETERS ================
==================== SIGNAL PROCESSING ================
----------------------- Nonlinear filter ----------------------------
nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
nonlinear filter order
nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
for using it with waveguide instruments
----------------------- Synthesis parameters computing and functions declaration ----------------------------
the number of modes depends on the preset being used
de.delay lengths in number of samples
de.delay lines
Filter bank: fi.bandpass filters (declared in instrument.lib)
----------------------- Algorithm implementation ----------------------------
One resonance
----------------------- Reverb (ajout accelerometre 05/2015) ---------------- | declare name "Tibetan Bowl";
declare description "Banded Waveguide Modeld Tibetan Bowl";
declare author "Romain Michon";
declare copyright "Romain Michon ([email protected])";
declare version "1.0";
declare description "This instrument uses banded waveguide. For more information, see Essl, G. and Cook, P. Banded Waveguides: Towards Physical Modelling of Bar Percussion Instruments, Proceedings of the 1999 International Computer Music Conference.";
import("stdfaust.lib");
instrument = library("instruments.lib");
process =
(((select-1)*-1) <:
NLFM : stereo : instrReverbAccel:
*(vol),*(vol);
vol = 0.8;
freq = hslider("[1]Frequency[unit:Hz][tooltip:Tone frequency]",440,180,780,1);
gain = 0.5;
gate = 0;
select = hslider("[0]Play[tooltip:0=Bow; 1=Strike] [acc:2 1 -10 0 10]", 0,0,1,1);
baseGain = 0.5;
typeModulation = 3;
nonLinearity = hslider("[2]Modulation[acc:0 1 -10 0 10][tooltip:Nonlinearity factor (value between 0 and 1)]",0.02,0,0.1,0.001):si.smooth(0.999);
frequencyMod = hslider("[3]Modulation Frequency[unit:Hz][acc:0 0 -10 0 10]", 220,150,500,0.1):si.smooth(0.999);
nonLinAttack = 0.1;
preset = 0;
nMode(0) = 12;
modes(0,0) = 0.996108344;
basegains(0,0) = 0.999925960128219;
excitation(0,0) = 11.900357 / 10;
modes(0,1) = 1.0038916562;
basegains(0,1) = 0.999925960128219;
excitation(0,1) = 11.900357 / 10;
modes(0,2) = 2.979178;
basegains(0,2) = 0.999982774366897;
excitation(0,2) = 10.914886 / 10;
modes(0,3) = 2.99329767;
basegains(0,3) = 0.999982774366897;
excitation(0,3) = 10.914886 / 10;
modes(0,4) = 5.704452;
basegains(0,4) = 1.0;
excitation(0,4) = 42.995041 / 10;
modes(0,5) = 5.704452;
basegains(0,5) = 1.0;
excitation(0,5) = 42.995041 / 10;
modes(0,6) = 8.9982;
basegains(0,6) = 1.0;
excitation(0,6) = 40.063034 / 10;
modes(0,7) = 9.01549726;
basegains(0,7) = 1.0;
excitation(0,7) = 40.063034 / 10;
modes(0,8) = 12.83303;
basegains(0,8) = 0.999965497558225;
excitation(0,8) = 7.063034 / 10;
modes(0,9) = 12.807382;
basegains(0,9) = 0.999965497558225;
excitation(0,9) = 7.063034 / 10;
modes(0,10) = 17.2808219;
basegains(0,10) = 0.9999999999999999999965497558225;
excitation(0,10) = 57.063034 / 10;
modes(0,11) = 21.97602739726;
basegains(0,11) = 0.999999999999999965497558225;
excitation(0,11) = 57.063034 / 10;
nlfOrder = 6;
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),1,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
nModes = nMode(preset);
delayLengthBase = ma.SR/freq;
delayLength(x) = delayLengthBase/modes(preset,x);
delayLine(x) = de.delay(4096,delayLength(x));
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x) = instrument.bandPass(freq*modes(preset,x),radius);
stereoo(periodDuration) = _ <: _,widthdelay : stereopanner
with {
W = 0.5;
A = 0.6;
widthdelay = de.delay(4096,W*periodDuration/2);
stereopanner = _,_ : *(1.0-A), *(A);
};
stereo = stereoo(delayLengthBase);
resonance(x) = + : + (excitation(preset,x)*select) : delayLine(x) : *(basegains(preset,x)) : bandPassFilter(x);
instrReverbAccel = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("v:[4]Reverb/[1]Reverberation Volume (InstrReverb) [acc:1 1 -10 0 10]",0.2,0.02,1,0.01) : si.smooth(0.999) :min(1):max(0.02);
roomSize = hslider("v:[4]Reverb/[2]Reverberation Room Size (InstrReverb)[acc:1 1 -10 0 10]", 0.2,0.02,2,0.01):min(2):max(0.02);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
ac99a90a15d053cdb7e79d388eb01b1bf76a20453074d3ebdc9945bf9f6381e2 | RuolunWeng/ruolunweng.github.io | FlappyFlute.dsp | declare name "Flappy Flute";
declare author "ER";// Adapted from "Nonlinear WaveGuide Flute" by Romain Michon ([email protected])
import("stdfaust.lib");
instrument = library("instruments.lib");
/* =============== DESCRIPTION ======================== :
- Flute turning into a flapping bird
- Head = Reverberation / High frequencies
- Tilting and jerking = looking for different sounds around head and back
- Bottom = Low frequencies
- Swing (bottom to head) = glissando (nice when followed with Back)
- Back = Echo
- Left = Slow rhythm/Silence
- Right = Flapping bird
*/
//==================== INSTRUMENT =======================
flute = (_ <: (flow + *(feedBack1) : embouchureDelay: poly) + *(feedBack2) : reflexionFilter)~(boreDelay) : NLFM : *(env2)*gain:_;
process = vgroup("Flappy Flute", flute : echo <: instrReverbFlute);
//==================== GUI SPECIFICATION ================
freq = hslider("[1]Frequency[unit:Hz][tooltip:Tone frequency][acc:1 1 -10 0 10]", 440,247,1200,1):si.smooth(0.999);
pressure = hslider("h:[3]Parameters/ Pressure[style:knob][acc:0 0 -10 0 10]", 1, 0.6, 1, 0.01):si.smooth(0.999):min(1):max(0.6);
breathAmp = hslider("h:[3]Parameters/Breath Noise[style:knob][acc:0 1 -10 0 10]", 0.01, 0.01, 0.2, 0.01):si.smooth(0.999):min(0.2):max(0.01);
gate = pulsaflute.gate;
vibratoFreq = 5;//hslider("h:Parameters/Vibrato Frequency[style:knob][unit:Hz]",5,1,15,0.1);
env1Attack = 0.05;//hslider("h:Parameters/Envelope Attack[unit:s][style:knob][tooltip:Pressure envelope attack duration]",0.05,0.05,0.2,0.01);
//----------------------- Echo ----------------------------------------
echo = +~ @(22050) *(feedback);
feedback = hslider("h:[4]Reverberation/Echo Intensity[style:knob][acc:2 0 -10 10 0 0.001] ", 0.001, 0.001, 0.65, 0.001):si.smooth(0.999):min(0.65):max(0.001);
//----------------------- Pulsar --------------------------------------
pulsaflute = environment{
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
fade = (0.5);
speed = hslider ("h:[2]Instrument/[2]Speed (Granulator)[style:knob][acc:0 1 -10 0 10]", 4,1,16,0.0001):fi.lowpass(1,1);
proba = hslider ("h:[2]Instrument/[3]Probability (Granulator)[unit:%][style:knob][acc:1 0 -10 0 10]", 88,60,100,1) *(0.01) : fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
//-------------------- Non-Variable Parameters -----------
N = 27;
gain = hslider("h:[2]Instrument/[1]Volume[style:knob][acc:0 1 -12 0 12]", 1, 0.75, 4, 0.01):min(4):max(0.75);
typeModulation = 0;
nonLinearity = 0;
frequencyMod = 220;
nonLinAttack = 0.1;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.05;
env2Release = 0.05;
env1Release = 0.05;
//==================== SIGNAL PROCESSING ================
//----------------------- Nonlinear filter ----------------------------
//nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
//nonlinear filter order
nlfOrder = 6;
//attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
envelopeMod = en.asr(nonLinAttack,1,0.1,gate);
//nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
//for using it with waveguide instruments
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//Loops feedbacks gains
feedBack1 = 0.4;
feedBack2 = 0.4;
//Delay Lines
embouchureDelayLength = (ma.SR/freq)/2-2;
boreDelayLength = ma.SR/freq-2;
embouchureDelay = de.fdelay(4096,embouchureDelayLength);
boreDelay = de.fdelay(4096,boreDelayLength);
//Polinomial
poly = _ <: _ - _*_*_;
//jet filter is a lowwpass filter (declared in filter.lib)
reflexionFilter = fi.lowpass(1,2000);
//----------------------- Algorithm implementation ----------------------------
//Pressure envelope
env1 = en.adsr(env1Attack,env1Decay,0.9,env1Release,(gate | pressureEnvelope))*pressure*1.1;
//Global envelope
env2 = en.asr(env2Attack,1,env2Release,gate)*0.5;
//Vibrato Envelope
vibratoEnvelope = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*vibratoGain;
vibrato = os.osc(vibratoFreq)*vibratoEnvelope;
breath = no.noise*env1;
flow = env1 + breath*breathAmp + vibrato;
instrReverbFlute = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
roomSize = hslider("h:[4]Reverberation/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -30 0 16]", 0.72,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/FlappyFlute.dsp | faust | Adapted from "Nonlinear WaveGuide Flute" by Romain Michon ([email protected])
=============== DESCRIPTION ======================== :
- Flute turning into a flapping bird
- Head = Reverberation / High frequencies
- Tilting and jerking = looking for different sounds around head and back
- Bottom = Low frequencies
- Swing (bottom to head) = glissando (nice when followed with Back)
- Back = Echo
- Left = Slow rhythm/Silence
- Right = Flapping bird
==================== INSTRUMENT =======================
==================== GUI SPECIFICATION ================
hslider("h:Parameters/Vibrato Frequency[style:knob][unit:Hz]",5,1,15,0.1);
hslider("h:Parameters/Envelope Attack[unit:s][style:knob][tooltip:Pressure envelope attack duration]",0.05,0.05,0.2,0.01);
----------------------- Echo ----------------------------------------
----------------------- Pulsar --------------------------------------
-------------------- Non-Variable Parameters -----------
==================== SIGNAL PROCESSING ================
----------------------- Nonlinear filter ----------------------------
nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
nonlinear filter order
attack - sustain - release envelope for nonlinearity (declared in instrument.lib)
nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
for using it with waveguide instruments
----------------------- Synthesis parameters computing and functions declaration ----------------------------
Loops feedbacks gains
Delay Lines
Polinomial
jet filter is a lowwpass filter (declared in filter.lib)
----------------------- Algorithm implementation ----------------------------
Pressure envelope
Global envelope
Vibrato Envelope | declare name "Flappy Flute";
import("stdfaust.lib");
instrument = library("instruments.lib");
flute = (_ <: (flow + *(feedBack1) : embouchureDelay: poly) + *(feedBack2) : reflexionFilter)~(boreDelay) : NLFM : *(env2)*gain:_;
process = vgroup("Flappy Flute", flute : echo <: instrReverbFlute);
freq = hslider("[1]Frequency[unit:Hz][tooltip:Tone frequency][acc:1 1 -10 0 10]", 440,247,1200,1):si.smooth(0.999);
pressure = hslider("h:[3]Parameters/ Pressure[style:knob][acc:0 0 -10 0 10]", 1, 0.6, 1, 0.01):si.smooth(0.999):min(1):max(0.6);
breathAmp = hslider("h:[3]Parameters/Breath Noise[style:knob][acc:0 1 -10 0 10]", 0.01, 0.01, 0.2, 0.01):si.smooth(0.999):min(0.2):max(0.01);
gate = pulsaflute.gate;
echo = +~ @(22050) *(feedback);
feedback = hslider("h:[4]Reverberation/Echo Intensity[style:knob][acc:2 0 -10 10 0 0.001] ", 0.001, 0.001, 0.65, 0.001):si.smooth(0.999):min(0.65):max(0.001);
pulsaflute = environment{
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
fade = (0.5);
speed = hslider ("h:[2]Instrument/[2]Speed (Granulator)[style:knob][acc:0 1 -10 0 10]", 4,1,16,0.0001):fi.lowpass(1,1);
proba = hslider ("h:[2]Instrument/[3]Probability (Granulator)[unit:%][style:knob][acc:1 0 -10 0 10]", 88,60,100,1) *(0.01) : fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
N = 27;
gain = hslider("h:[2]Instrument/[1]Volume[style:knob][acc:0 1 -12 0 12]", 1, 0.75, 4, 0.01):min(4):max(0.75);
typeModulation = 0;
nonLinearity = 0;
frequencyMod = 220;
nonLinAttack = 0.1;
vibratoGain = 0.1;
vibratoBegin = 0.1;
vibratoAttack = 0.5;
vibratoRelease = 0.2;
pressureEnvelope = 0;
env1Decay = 0.2;
env2Attack = 0.05;
env2Release = 0.05;
env1Release = 0.05;
nlfOrder = 6;
envelopeMod = en.asr(nonLinAttack,1,0.1,gate);
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),envelopeMod,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
feedBack1 = 0.4;
feedBack2 = 0.4;
embouchureDelayLength = (ma.SR/freq)/2-2;
boreDelayLength = ma.SR/freq-2;
embouchureDelay = de.fdelay(4096,embouchureDelayLength);
boreDelay = de.fdelay(4096,boreDelayLength);
poly = _ <: _ - _*_*_;
reflexionFilter = fi.lowpass(1,2000);
env1 = en.adsr(env1Attack,env1Decay,0.9,env1Release,(gate | pressureEnvelope))*pressure*1.1;
env2 = en.asr(env2Attack,1,env2Release,gate)*0.5;
vibratoEnvelope = instrument.envVibrato(vibratoBegin,vibratoAttack,100,vibratoRelease,gate)*vibratoGain;
vibrato = os.osc(vibratoFreq)*vibratoEnvelope;
breath = no.noise*env1;
flow = env1 + breath*breathAmp + vibrato;
instrReverbFlute = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
roomSize = hslider("h:[4]Reverberation/Reverberation Room Size (InstrReverb)[style:knob][acc:1 1 -30 0 16]", 0.72,0.05,2,0.01):min(2):max(0.05);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
a57e71383c2493c609a81da5281d6b93fe93d0e6434ed63c0c8ff898abfa308b | RuolunWeng/ruolunweng.github.io | TibetanBowlMulti.dsp | declare name "Tibetan Bowl";
declare description "Banded Waveguide Modeld Tibetan Bowl";
declare author "Romain Michon";
declare copyright "Romain Michon ([email protected])";
declare version "1.0";
declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
declare description "This instrument uses banded waveguide. For more information, see Essl, G. and Cook, P. Banded Waveguides: Towards Physical Modelling of Bar Percussion Instruments, Proceedings of the 1999 International Computer Music Conference.";
import("stdfaust.lib");
instrument = library("instruments.lib");
/* ============ DESCRIPTION ==============
- Multiple Tibetan Bowls
- Head = Note Still/Reverberation
- Head + Light circling on the spot/slight rotation = reverberated rolling ball in the bowl
- Rocking = Ringing all bowls from low to high frequencies
- Back = Modulation
- Back + Light circling on the spot/slight rotation = rolling ball in the bowl
*/
//==================== INSTRUMENT =======================
process =
(((select-1)*-1) <:
//nModes resonances with nModes feedbacks for bow table look-up
par(i,nModes,(resonance(i)~_))):>+://~par(i,nModes,_) :> + :
//Signal Scaling and stereo
NLFM : stereo : instrReverbAccel:
*(vol),*(vol);
//==================== GUI SPECIFICATION ================
vol = 0.8;
freq = hslider("[1]Frequency[unit:Hz][acc:0 1 -10 0 10]", 440,180,780,1);
gain = 0.5;
gate = 0;
select = hslider("[0]Play[tooltip:0=Bow; 1=Strike][acc:2 1 -10 0 10]", 0,0,1,1);
integrationConstant = 0.01;
baseGain = 0.5;
typeModulation = 3;
nonLinearity = hslider("[2]Modulation[acc:2 0 -10 0 15][tooltip:Nonlinearity factor (value between 0 and 1)]",0.02,0,0.1,0.001);
frequencyMod = hslider("[3]Modulation Frequency[3][unit:Hz][acc:2 0 -10 0 15]", 220,150,500,0.1);
nonLinAttack = 0.1;
//==================== MODAL PARAMETERS ================
preset = 0;
nMode(0) = 12;
modes(0,0) = 0.996108344;
basegains(0,0) = 0.999925960128219;
excitation(0,0) = 11.900357 / 10;
modes(0,1) = 1.0038916562;
basegains(0,1) = 0.999925960128219;
excitation(0,1) = 11.900357 / 10;
modes(0,2) = 2.979178;
basegains(0,2) = 0.999982774366897;
excitation(0,2) = 10.914886 / 10;
modes(0,3) = 2.99329767;
basegains(0,3) = 0.999982774366897;
excitation(0,3) = 10.914886 / 10;
modes(0,4) = 5.704452;
basegains(0,4) = 1.0;
excitation(0,4) = 42.995041 / 10;
modes(0,5) = 5.704452;
basegains(0,5) = 1.0;
excitation(0,5) = 42.995041 / 10;
modes(0,6) = 8.9982;
basegains(0,6) = 1.0;
excitation(0,6) = 40.063034 / 10;
modes(0,7) = 9.01549726;
basegains(0,7) = 1.0;
excitation(0,7) = 40.063034 / 10;
modes(0,8) = 12.83303;
basegains(0,8) = 0.999965497558225;
excitation(0,8) = 7.063034 / 10;
modes(0,9) = 12.807382;
basegains(0,9) = 0.999965497558225;
excitation(0,9) = 7.063034 / 10;
modes(0,10) = 17.2808219;
basegains(0,10) = 0.9999999999999999999965497558225;
excitation(0,10) = 57.063034 / 10;
modes(0,11) = 21.97602739726;
basegains(0,11) = 0.999999999999999965497558225;
excitation(0,11) = 57.063034 / 10;
//==================== SIGNAL PROCESSING ================
//----------------------- Nonlinear filter ----------------------------
//nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
//nonlinear filter order
nlfOrder = 6;
//nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
//for using it with waveguide instruments
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),1,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
//----------------------- Synthesis parameters computing and functions declaration ----------------------------
//the number of modes depends on the preset being used
nModes = nMode(preset);
//bow table parameters
tableOffset = 0;
tableSlope = 10 - (9*bowPressure);
delayLengthBase = ma.SR/freq;
//de.delay lengths in number of samples
delayLength(x) = delayLengthBase/modes(preset,x);
//de.delay lines
delayLine(x) = de.delay(4096,delayLength(x));
//Filter bank: fi.bandpass filters (declared in instrument.lib)
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x) = instrument.bandPass(freq*modes(preset,x),radius);
//Delay lines feedback for bow table lookup control
baseGainApp = 0.8999999999999999 + (0.1*baseGain);
velocityInputApp = integrationConstant;
velocityInput = velocityInputApp + _*baseGainApp,par(i,(nModes-1),(_*baseGainApp)) :> +;
//Bow velocity is controled by an ADSR envelope
maxVelocity = 0.03 + 0.1*gain;
bowVelocity = maxVelocity*en.adsr(0.02,0.005,0.9,0.01,gate);
stereoo(periodDuration) = _ <: _,widthdelay : stereopanner
with{
//W = hslider("v:Spat/spatial width", 0.5, 0, 1, 0.01);
W = 0.5;
//A = hslider("v:Spat/pan angle", 0.6, 0, 1, 0.01);
A = 0.6;
widthdelay = de.delay(4096,W*periodDuration/2);
stereopanner = _,_ : *(1.0-A), *(A);
};
stereo = stereoo(delayLengthBase);
//----------------------- Algorithm implementation ----------------------------
//One resonance
resonance(x) = + : + (excitation(preset,x)*select) : delayLine(x) : *(basegains(preset,x)) : bandPassFilter(x);
//----------------------- Reverb (ajout accelerometre 05/2015) ----------------
instrReverbAccel = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("v:[4]Reverb/[1]Reverberation Volume (InstrReverb)[acc:1 1 -10 0 10]",0.2,0.02,1,0.01) : si.smooth(0.999):min(1):max(0.02);
roomSize = hslider("v:[4]Reverb/[2]Reverberation Room Size (InstrReverb)[acc:1 1 -10 0 10]", 0.2,0.02,1.3,0.01) : min(1.3) : max(0.02);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/TibetanBowlMulti.dsp | faust | Synthesis Tool Kit 4.3 (MIT style license);
============ DESCRIPTION ==============
- Multiple Tibetan Bowls
- Head = Note Still/Reverberation
- Head + Light circling on the spot/slight rotation = reverberated rolling ball in the bowl
- Rocking = Ringing all bowls from low to high frequencies
- Back = Modulation
- Back + Light circling on the spot/slight rotation = rolling ball in the bowl
==================== INSTRUMENT =======================
nModes resonances with nModes feedbacks for bow table look-up
~par(i,nModes,_) :> + :
Signal Scaling and stereo
==================== GUI SPECIFICATION ================
==================== MODAL PARAMETERS ================
==================== SIGNAL PROCESSING ================
----------------------- Nonlinear filter ----------------------------
nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
nonlinear filter order
nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib
for using it with waveguide instruments
----------------------- Synthesis parameters computing and functions declaration ----------------------------
the number of modes depends on the preset being used
bow table parameters
de.delay lengths in number of samples
de.delay lines
Filter bank: fi.bandpass filters (declared in instrument.lib)
Delay lines feedback for bow table lookup control
Bow velocity is controled by an ADSR envelope
W = hslider("v:Spat/spatial width", 0.5, 0, 1, 0.01);
A = hslider("v:Spat/pan angle", 0.6, 0, 1, 0.01);
----------------------- Algorithm implementation ----------------------------
One resonance
----------------------- Reverb (ajout accelerometre 05/2015) ---------------- | declare name "Tibetan Bowl";
declare description "Banded Waveguide Modeld Tibetan Bowl";
declare author "Romain Michon";
declare copyright "Romain Michon ([email protected])";
declare version "1.0";
declare description "This instrument uses banded waveguide. For more information, see Essl, G. and Cook, P. Banded Waveguides: Towards Physical Modelling of Bar Percussion Instruments, Proceedings of the 1999 International Computer Music Conference.";
import("stdfaust.lib");
instrument = library("instruments.lib");
process =
(((select-1)*-1) <:
NLFM : stereo : instrReverbAccel:
*(vol),*(vol);
vol = 0.8;
freq = hslider("[1]Frequency[unit:Hz][acc:0 1 -10 0 10]", 440,180,780,1);
gain = 0.5;
gate = 0;
select = hslider("[0]Play[tooltip:0=Bow; 1=Strike][acc:2 1 -10 0 10]", 0,0,1,1);
integrationConstant = 0.01;
baseGain = 0.5;
typeModulation = 3;
nonLinearity = hslider("[2]Modulation[acc:2 0 -10 0 15][tooltip:Nonlinearity factor (value between 0 and 1)]",0.02,0,0.1,0.001);
frequencyMod = hslider("[3]Modulation Frequency[3][unit:Hz][acc:2 0 -10 0 15]", 220,150,500,0.1);
nonLinAttack = 0.1;
preset = 0;
nMode(0) = 12;
modes(0,0) = 0.996108344;
basegains(0,0) = 0.999925960128219;
excitation(0,0) = 11.900357 / 10;
modes(0,1) = 1.0038916562;
basegains(0,1) = 0.999925960128219;
excitation(0,1) = 11.900357 / 10;
modes(0,2) = 2.979178;
basegains(0,2) = 0.999982774366897;
excitation(0,2) = 10.914886 / 10;
modes(0,3) = 2.99329767;
basegains(0,3) = 0.999982774366897;
excitation(0,3) = 10.914886 / 10;
modes(0,4) = 5.704452;
basegains(0,4) = 1.0;
excitation(0,4) = 42.995041 / 10;
modes(0,5) = 5.704452;
basegains(0,5) = 1.0;
excitation(0,5) = 42.995041 / 10;
modes(0,6) = 8.9982;
basegains(0,6) = 1.0;
excitation(0,6) = 40.063034 / 10;
modes(0,7) = 9.01549726;
basegains(0,7) = 1.0;
excitation(0,7) = 40.063034 / 10;
modes(0,8) = 12.83303;
basegains(0,8) = 0.999965497558225;
excitation(0,8) = 7.063034 / 10;
modes(0,9) = 12.807382;
basegains(0,9) = 0.999965497558225;
excitation(0,9) = 7.063034 / 10;
modes(0,10) = 17.2808219;
basegains(0,10) = 0.9999999999999999999965497558225;
excitation(0,10) = 57.063034 / 10;
modes(0,11) = 21.97602739726;
basegains(0,11) = 0.999999999999999965497558225;
excitation(0,11) = 57.063034 / 10;
nlfOrder = 6;
NLFM = instrument.nonLinearModulator((nonLinearity : si.smooth(0.999)),1,freq,
typeModulation,(frequencyMod : si.smooth(0.999)),nlfOrder);
nModes = nMode(preset);
tableOffset = 0;
tableSlope = 10 - (9*bowPressure);
delayLengthBase = ma.SR/freq;
delayLength(x) = delayLengthBase/modes(preset,x);
delayLine(x) = de.delay(4096,delayLength(x));
radius = 1 - ma.PI*32/ma.SR;
bandPassFilter(x) = instrument.bandPass(freq*modes(preset,x),radius);
baseGainApp = 0.8999999999999999 + (0.1*baseGain);
velocityInputApp = integrationConstant;
velocityInput = velocityInputApp + _*baseGainApp,par(i,(nModes-1),(_*baseGainApp)) :> +;
maxVelocity = 0.03 + 0.1*gain;
bowVelocity = maxVelocity*en.adsr(0.02,0.005,0.9,0.01,gate);
stereoo(periodDuration) = _ <: _,widthdelay : stereopanner
with{
W = 0.5;
A = 0.6;
widthdelay = de.delay(4096,W*periodDuration/2);
stereopanner = _,_ : *(1.0-A), *(A);
};
stereo = stereoo(delayLengthBase);
resonance(x) = + : + (excitation(preset,x)*select) : delayLine(x) : *(basegains(preset,x)) : bandPassFilter(x);
instrReverbAccel = _,_ <: *(reverbGain),*(reverbGain),*(1 - reverbGain),*(1 - reverbGain) :
re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax),_,_ <: _,!,_,!,!,_,!,_ : +,+
with {
reverbGain = hslider("v:[4]Reverb/[1]Reverberation Volume (InstrReverb)[acc:1 1 -10 0 10]",0.2,0.02,1,0.01) : si.smooth(0.999):min(1):max(0.02);
roomSize = hslider("v:[4]Reverb/[2]Reverberation Room Size (InstrReverb)[acc:1 1 -10 0 10]", 0.2,0.02,1.3,0.01) : min(1.3) : max(0.02);
rdel = 20;
f1 = 200;
f2 = 6000;
t60dc = roomSize*3;
t60m = roomSize*2;
fsmax = 48000;
};
|
f69fb158515df4cde82959ec3d9de1816c2f5fcaa0cf7d005f9a0daeed1cd5f3 | RuolunWeng/ruolunweng.github.io | SBird.dsp | declare name "bird";
declare author "Pierre Cochard";
//Modifications by Grame July 2014, June 2015;
/* =============== DESCRIPTION ================= :
- Bird singing generator.
- Right = maximum speed of whistles.
- Left = minimum speed/Rare birds, nearly silence.
*/
import("stdfaust.lib");
// PROCESS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
process = hgroup("Bird", mainOsc(noteTrig : rdm(72,94) : mtof , noteTrig) * envWrapper(noteTrig, ampEnv, amp_xp(2510)) : fi.lowpass(1, 2500) *(0.8) <: _,_);
// AUTO TRIGGER
autoTrig = ba.beat(t) * (abs(no.noise) <= p) : trigger(48)
with {
t = hslider("[1]Speed (Granulator)[style:knob][acc:0 1 -10 0 10]", 240, 120, 480, 0.1) : si.smooth(0.999);
p = hslider("[2]Probability (Granulator)[unit:%][style:knob][acc:0 1 -10 0 10]", 50, 25, 100, 1)*(0.01) : si.smooth(0.999);
trigger(n) = upfront : release(n) : >(0.0)
with {
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
};
};
// BIRD TRIGGER
noteTrig = autoTrig;
// OSCILLATORS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
/* base */
carrierOsc(freq) = os.osci(freq);
modOsc(freq) = os.triangleN(3,freq);
/* fm oscillator */
mainOsc(freq,trig) = freq <: +(*(harmRatio <: +(*(envWrapper(trig,harmEnv,harm_xp(1700))))) : modOsc : *(modIndex <: +(*(envWrapper(trig,modIndexEnv,modIndex_xp(550)))))) <: +(*(envWrapper(trig,freqEnv,freq_xp(943)))) : carrierOsc;
envWrapper(trig,env,sus) = trig : mstosamps(rdm(100,3000)), sus : hitLength : env;
// FIXED PARAMETERS - - - - - - - - - - - - - - - - - - - - - - - - - - -
/* fm */
harmRatio = 0.063;
modIndex = 3.24;
// TIME FUNCTIONS - - - - - - - - - - - - - - - - - - - - - - - - - - - -
metro(ms) = (%(+(1),mstosamps(ms))) ~_ : ==(1);
mstosamps(ms) = ms : /(1000) * ma.SR : int;
rdmInc = _ <: @(1), @(2) : + : *(2994.2313) : int : +(38125);
rdm(rdmin,rdmax) = _,(fmod(_,rdmax - rdmin : int) ~ rdmInc : +(rdmin)) : gater : -(1) : abs;
gater = (_,_,_ <: !,_,!,_,!,!,!,!,_ : select2) ~_;
// MIDI RELATED - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
/* midi pitch */
mtof(midinote) = pow(2,(midinote - 69) / 12) * 440;
// ENVELOPPES - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
/* envelope "reader" (phaser) */
hitLength(length,sustain) = *((==(length,@(length,1))), +(1))~_ <: gater(<(sustain));
/* amplitude envelope */
ampEnvbpf = ba.bpf.start(0, 0) :
ba.bpf.point(amp_xp(60.241), 1.) :
ba.bpf.point(amp_xp(461.847), 0.) :
ba.bpf.point(amp_xp(582.329), 0.928) :
ba.bpf.point(amp_xp(682.731), 0.5) :
ba.bpf.point(amp_xp(983.936), 0.) :
ba.bpf.point(amp_xp(1064.257), 0.) :
ba.bpf.point(amp_xp(1345.382), 0.) :
ba.bpf.point(amp_xp(1526.105), 0.) :
ba.bpf.point(amp_xp(1746.988), 0.) :
ba.bpf.point(amp_xp(1827.309), 0.) :
ba.bpf.point(amp_xp(2088.353), 0.) :
ba.bpf.point(amp_xp(2188.755), 0.) : /* sustain point */
ba.bpf.end(amp_xp(2510.040), 0.);
ampEnv = ampEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
amp_xp(x) = x * ma.SR / 1000. * ampEnv_speed;
ampEnv_speed = noteTrig : rdm(0,2000) : /(1000);
/* freq envelope */
freqEnvbpf = ba.bpf.start(0, 0) :
ba.bpf.point(freq_xp(147.751), 1.) :
ba.bpf.point(freq_xp(193.213), 0.) :
ba.bpf.point(freq_xp(318.233), yp) :
ba.bpf.point(freq_xp(431.888), 0.) :
ba.bpf.point(freq_xp(488.715), 0.434) :
ba.bpf.point(freq_xp(613.735), yp) :
ba.bpf.point(freq_xp(659.197), 1.) :
ba.bpf.point(freq_xp(716.024), yp) :
ba.bpf.point(freq_xp(806.948), 1.) :
ba.bpf.point(freq_xp(829.679), yp) : /* sustain point */
ba.bpf.end(freq_xp(943.333), 0.);
freqEnv = freqEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
freq_xp(x) = x * ma.SR / 1000. * freqEnv_speed;
freqEnv_speed = noteTrig : rdm(0,2000) : /(1000);
yp = noteTrig : rdm(0,1000) : /(1000);
/* harmRatio envelope */
harmEnvbpf = ba.bpf.start(0, 0.) :
ba.bpf.point(harm_xp(863.454), 0.490) :
ba.bpf.point(harm_xp(865), 0.) :
ba.bpf.point (harm_xp(1305.221), 1.) :
ba.bpf.point(harm_xp(1646.586), 0.) : /* sustain point */
ba.bpf.end(harm_xp(1700), 0.);
harmEnv = harmEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
harm_xp(x) = x * ma.SR / 1000. * harmEnv_speed;
harmEnv_speed = noteTrig : rdm(0,2000) : /(1000);
/* modIndex envelope */
modIndexEnvbpf = ba.bpf.start(0, 0.) :
ba.bpf.point(modIndex_xp(240.964), 0.554) :
ba.bpf.point(modIndex_xp(502.068), 0.) : /* sustain point */
ba.bpf.end(modIndex_xp(550), 0.);
modIndexEnv = modIndexEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
modIndex_xp(x) = x * ma.SR / 1000. * modIndexEnv_speed;
modIndexEnv_speed = noteTrig : rdm(0,2000) : /(1000);
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SBird.dsp | faust | Modifications by Grame July 2014, June 2015;
=============== DESCRIPTION ================= :
- Bird singing generator.
- Right = maximum speed of whistles.
- Left = minimum speed/Rare birds, nearly silence.
PROCESS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
AUTO TRIGGER
BIRD TRIGGER
OSCILLATORS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
base
fm oscillator
FIXED PARAMETERS - - - - - - - - - - - - - - - - - - - - - - - - - - -
fm
TIME FUNCTIONS - - - - - - - - - - - - - - - - - - - - - - - - - - - -
MIDI RELATED - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
midi pitch
ENVELOPPES - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
envelope "reader" (phaser)
amplitude envelope
sustain point
freq envelope
sustain point
harmRatio envelope
sustain point
modIndex envelope
sustain point | declare name "bird";
declare author "Pierre Cochard";
import("stdfaust.lib");
process = hgroup("Bird", mainOsc(noteTrig : rdm(72,94) : mtof , noteTrig) * envWrapper(noteTrig, ampEnv, amp_xp(2510)) : fi.lowpass(1, 2500) *(0.8) <: _,_);
autoTrig = ba.beat(t) * (abs(no.noise) <= p) : trigger(48)
with {
t = hslider("[1]Speed (Granulator)[style:knob][acc:0 1 -10 0 10]", 240, 120, 480, 0.1) : si.smooth(0.999);
p = hslider("[2]Probability (Granulator)[unit:%][style:knob][acc:0 1 -10 0 10]", 50, 25, 100, 1)*(0.01) : si.smooth(0.999);
trigger(n) = upfront : release(n) : >(0.0)
with {
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
};
};
noteTrig = autoTrig;
carrierOsc(freq) = os.osci(freq);
modOsc(freq) = os.triangleN(3,freq);
mainOsc(freq,trig) = freq <: +(*(harmRatio <: +(*(envWrapper(trig,harmEnv,harm_xp(1700))))) : modOsc : *(modIndex <: +(*(envWrapper(trig,modIndexEnv,modIndex_xp(550)))))) <: +(*(envWrapper(trig,freqEnv,freq_xp(943)))) : carrierOsc;
envWrapper(trig,env,sus) = trig : mstosamps(rdm(100,3000)), sus : hitLength : env;
harmRatio = 0.063;
modIndex = 3.24;
metro(ms) = (%(+(1),mstosamps(ms))) ~_ : ==(1);
mstosamps(ms) = ms : /(1000) * ma.SR : int;
rdmInc = _ <: @(1), @(2) : + : *(2994.2313) : int : +(38125);
rdm(rdmin,rdmax) = _,(fmod(_,rdmax - rdmin : int) ~ rdmInc : +(rdmin)) : gater : -(1) : abs;
gater = (_,_,_ <: !,_,!,_,!,!,!,!,_ : select2) ~_;
mtof(midinote) = pow(2,(midinote - 69) / 12) * 440;
hitLength(length,sustain) = *((==(length,@(length,1))), +(1))~_ <: gater(<(sustain));
ampEnvbpf = ba.bpf.start(0, 0) :
ba.bpf.point(amp_xp(60.241), 1.) :
ba.bpf.point(amp_xp(461.847), 0.) :
ba.bpf.point(amp_xp(582.329), 0.928) :
ba.bpf.point(amp_xp(682.731), 0.5) :
ba.bpf.point(amp_xp(983.936), 0.) :
ba.bpf.point(amp_xp(1064.257), 0.) :
ba.bpf.point(amp_xp(1345.382), 0.) :
ba.bpf.point(amp_xp(1526.105), 0.) :
ba.bpf.point(amp_xp(1746.988), 0.) :
ba.bpf.point(amp_xp(1827.309), 0.) :
ba.bpf.point(amp_xp(2088.353), 0.) :
ba.bpf.end(amp_xp(2510.040), 0.);
ampEnv = ampEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
amp_xp(x) = x * ma.SR / 1000. * ampEnv_speed;
ampEnv_speed = noteTrig : rdm(0,2000) : /(1000);
freqEnvbpf = ba.bpf.start(0, 0) :
ba.bpf.point(freq_xp(147.751), 1.) :
ba.bpf.point(freq_xp(193.213), 0.) :
ba.bpf.point(freq_xp(318.233), yp) :
ba.bpf.point(freq_xp(431.888), 0.) :
ba.bpf.point(freq_xp(488.715), 0.434) :
ba.bpf.point(freq_xp(613.735), yp) :
ba.bpf.point(freq_xp(659.197), 1.) :
ba.bpf.point(freq_xp(716.024), yp) :
ba.bpf.point(freq_xp(806.948), 1.) :
ba.bpf.end(freq_xp(943.333), 0.);
freqEnv = freqEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
freq_xp(x) = x * ma.SR / 1000. * freqEnv_speed;
freqEnv_speed = noteTrig : rdm(0,2000) : /(1000);
yp = noteTrig : rdm(0,1000) : /(1000);
harmEnvbpf = ba.bpf.start(0, 0.) :
ba.bpf.point(harm_xp(863.454), 0.490) :
ba.bpf.point(harm_xp(865), 0.) :
ba.bpf.point (harm_xp(1305.221), 1.) :
ba.bpf.end(harm_xp(1700), 0.);
harmEnv = harmEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
harm_xp(x) = x * ma.SR / 1000. * harmEnv_speed;
harmEnv_speed = noteTrig : rdm(0,2000) : /(1000);
modIndexEnvbpf = ba.bpf.start(0, 0.) :
ba.bpf.point(modIndex_xp(240.964), 0.554) :
ba.bpf.end(modIndex_xp(550), 0.);
modIndexEnv = modIndexEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
modIndex_xp(x) = x * ma.SR / 1000. * modIndexEnv_speed;
modIndexEnv_speed = noteTrig : rdm(0,2000) : /(1000);
|
11787a313434864de1211d25d232336a4352243ce635d8d84669a849f6677b65 | RuolunWeng/ruolunweng.github.io | SCameleonKeyboard.dsp | declare name "Cameleon Keyboard";
declare author "ER";
import("stdfaust.lib");
//From John Chowning Turenas envelops
/* =============== DESCRIPTION ================= :
- Multiple envelope keyboard
- Pick an envelope
- Rocking = striking across the keyboard from low frequencies (Left) to high frequencies (Right)
- Back + Rotation = long notes
- Front + Rotation = short notes
*/
//--------------------------------- INSTRUMENT ---------------------------------
marimkey(n) = os.osc(octave(n)) * (0.1)
*(trigger(n+1) : envelope : fi.lowpass(1,500));
process = hand <: par(i, 10, marimkey(i)) :> *(3);
//---------------------------------- UI ----------------------------------------
hand = hslider("[1]Instrument Hand[acc:1 0 -10 0 10]", 5, 0, 10, 1);
hight = hslider("[2]Hight[acc:0 1 -10 0 30]", 5, 1, 10, 0.3) : si.smooth(0.99):min(12):max(1);
envsize = hslider("[3]Note Duration (BPF Envelope)[unit:s][acc:2 0 -10 0 10]", 0.2, 0.1, 0.5, 0.01) * (ma.SR) : si.smooth(0.999): min(44100) : max(4410) : int;
//---------------------------------- FREQUENCY TABLE ---------------------------
freq(0) = 164.81;
freq(1) = 174.61;
freq(d) = freq(d-2);
octave(d) = freq(d)* hight;
//------------------------------------ TRIGGER ---------------------------------
upfront(x) = x>x';
counter(g)= (+(1):*(1-g))~_;
position(a,x) = abs(x - a) < 0.5;
trigger(p) = position(p) : upfront : counter;
//----------------------------------- ENVELOPPES ------------------------------
/* envelope */
typeEnv = vslider("[4]Envelope Type (BPF Envelope)[style:radio{'f9':0;'f11':1;'f15':2;'f17':3}]",0,0,3,1):int;
envelope = _<:sum(i, 4, tabchowning.env(i) * (abs(typeEnv - (i)) < 0.5));
/* Tables Chowning */
tabchowning = environment
{
// percussives envelops have been smmothed to avoid clicks.
corres(x) = int(x*envsize/1024);
// f9 0 1024 7 1 248 0.25 259 0.1 259 0.05 258 0
env(0) = f9;
f9 = ba.bpf.start(0, 0):
ba.bpf.point(corres(2), 0.25):
ba.bpf.point(corres(4), 0.5):
ba.bpf.point(corres(10), 0.9):
ba.bpf.point(corres(248), 0.25):
ba.bpf.point(corres(507), 0.1):
ba.bpf.point(corres(766), 0.05):
ba.bpf.end(corres(1024), 0);
/*
//f10 0 1024 7 0.5 197 1 310 0.1 259 0.02 258 0
env(1) = f10;
f10 = ba.bpf.start(0, 0):
ba.bpf.point(corres(2), 0.25):
ba.bpf.point(corres(4), 0.5):
ba.bpf.point(corres(197), 0.99):
ba.bpf.point(corres(507), 0.1):
ba.bpf.point(corres(766), 0.02):
ba.bpf.end(corres(1024), 0);
*/
//f11 0 1024 7 0 93 0.02 52 0.1 103 0.5 52 0.95 31 1 538 0.95 52 0.9 52 0.05 51 0
env(1) = f11;
f11 = ba.bpf.start(0, 0):
ba.bpf.point(corres(93), 0.02):
ba.bpf.point(corres(145), 0.1):
ba.bpf.point(corres(248), 0.5):
ba.bpf.point(corres(300), 0.95):
ba.bpf.point(corres(331), 0.99):
ba.bpf.point(corres(869), 0.95):
ba.bpf.point(corres(921), 0.9):
ba.bpf.point(corres(973), 0.05):
ba.bpf.end(corres(1024), 0);
/*
//f12 0 1024 7 0 93 0.02 52 0.1 103 0.5 52 0.95 31 1 693 0.95
env(3) = f12;
f12 = ba.bpf.start(0, 0):
ba.bpf.point(corres(93), 0.02):
ba.bpf.point(corres(145), 0.1):
ba.bpf.point(corres(248), 0.5):
ba.bpf.point(corres(300), 0.95):
ba.bpf.point(corres(331), 0.99):
ba.bpf.point(corres(1018), 0.95):
ba.bpf.point(corres(1020), 0.25):
ba.bpf.point(corres(1022), 0.125):
ba.bpf.end(corres(1024), 0);
//f13 0 1024 7 0 41 0.5 155 1 310 0.2 259 0.02 259 0 0
env(4) = f13;
f13 = ba.bpf.start(0, 0):
ba.bpf.point(corres(41), 0.5):
ba.bpf.point(corres(196), 0.99):
ba.bpf.point(corres(506), 0.2):
ba.bpf.point(corres(765), 0.02):
ba.bpf.end(corres(1024), 0);
//f14 0 1024 7 1 114 0.75 134 1 259 0.25 259 0.05 258 0
env(5) = f14;
f14 = ba.bpf.start(0, 0):
ba.bpf.point(corres(2), 0.25):
ba.bpf.point(corres(4), 0.5):
ba.bpf.point(corres(6), 0.99):
ba.bpf.point(corres(114), 0.75):
ba.bpf.point(corres(248), 0.99):
ba.bpf.point(corres(507), 0.25):
ba.bpf.point(corres(766), 0.05):
ba.bpf.end(corres(1024), 0);
*/
//f15 0 1024 7 1 41 0.1 52 0.02 155 0 776 0
env(2) = f15;
f15 = ba.bpf.start(0, 0):
ba.bpf.point(corres(2), 0.25):
ba.bpf.point(corres(4), 0.5):
ba.bpf.point(corres(6), 0.99):
ba.bpf.point(corres(41), 0.1):
ba.bpf.point(corres(93), 0.02):
ba.bpf.point(corres(248), 0):
ba.bpf.end(corres(1024), 0);
/*
//f16 0 1024 7 0 145 0.1 155 0.5 103 0.95 103 1 103 0.96 103 0.5 155 0.1 157 0
env(7) = f16;
f16 = ba.bpf.start(0, 0):
ba.bpf.point(corres(145), 0.1):
ba.bpf.point(corres(300), 0.5):
ba.bpf.point(corres(403), 0.95):
ba.bpf.point(corres(506), 0.99):
ba.bpf.point(corres(609), 0.96):
ba.bpf.point(corres(712), 0.5):
ba.bpf.point(corres(867), 0.1):
ba.bpf.end(corres(1024), 0);
*/
//f17 0 1024 7 1 103 0.5 145 0.25 259 0.1 259 0.05 259 0
env(3) = f17;
f17 = ba.bpf.start(0, 0):
ba.bpf.point(corres(2), 0.25):
ba.bpf.point(corres(4), 0.5):
ba.bpf.point(corres(6), 0.99):
ba.bpf.point(corres(103), 0.5):
ba.bpf.point(corres(248), 0.25):
ba.bpf.point(corres(403), 0.1):
ba.bpf.point(corres(507), 0.05):
ba.bpf.point(corres(766), 0):
ba.bpf.end(corres(1024), 0);
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/generators/SCameleonKeyboard.dsp | faust | From John Chowning Turenas envelops
=============== DESCRIPTION ================= :
- Multiple envelope keyboard
- Pick an envelope
- Rocking = striking across the keyboard from low frequencies (Left) to high frequencies (Right)
- Back + Rotation = long notes
- Front + Rotation = short notes
--------------------------------- INSTRUMENT ---------------------------------
---------------------------------- UI ----------------------------------------
---------------------------------- FREQUENCY TABLE ---------------------------
------------------------------------ TRIGGER ---------------------------------
----------------------------------- ENVELOPPES ------------------------------
envelope
Tables Chowning
percussives envelops have been smmothed to avoid clicks.
f9 0 1024 7 1 248 0.25 259 0.1 259 0.05 258 0
//f10 0 1024 7 0.5 197 1 310 0.1 259 0.02 258 0
env(1) = f10;
f10 = ba.bpf.start(0, 0):
ba.bpf.point(corres(2), 0.25):
ba.bpf.point(corres(4), 0.5):
ba.bpf.point(corres(197), 0.99):
ba.bpf.point(corres(507), 0.1):
ba.bpf.point(corres(766), 0.02):
ba.bpf.end(corres(1024), 0);
f11 0 1024 7 0 93 0.02 52 0.1 103 0.5 52 0.95 31 1 538 0.95 52 0.9 52 0.05 51 0
//f12 0 1024 7 0 93 0.02 52 0.1 103 0.5 52 0.95 31 1 693 0.95
env(3) = f12;
f12 = ba.bpf.start(0, 0):
ba.bpf.point(corres(93), 0.02):
ba.bpf.point(corres(145), 0.1):
ba.bpf.point(corres(248), 0.5):
ba.bpf.point(corres(300), 0.95):
ba.bpf.point(corres(331), 0.99):
ba.bpf.point(corres(1018), 0.95):
ba.bpf.point(corres(1020), 0.25):
ba.bpf.point(corres(1022), 0.125):
ba.bpf.end(corres(1024), 0);
//f13 0 1024 7 0 41 0.5 155 1 310 0.2 259 0.02 259 0 0
env(4) = f13;
f13 = ba.bpf.start(0, 0):
ba.bpf.point(corres(41), 0.5):
ba.bpf.point(corres(196), 0.99):
ba.bpf.point(corres(506), 0.2):
ba.bpf.point(corres(765), 0.02):
ba.bpf.end(corres(1024), 0);
//f14 0 1024 7 1 114 0.75 134 1 259 0.25 259 0.05 258 0
env(5) = f14;
f14 = ba.bpf.start(0, 0):
ba.bpf.point(corres(2), 0.25):
ba.bpf.point(corres(4), 0.5):
ba.bpf.point(corres(6), 0.99):
ba.bpf.point(corres(114), 0.75):
ba.bpf.point(corres(248), 0.99):
ba.bpf.point(corres(507), 0.25):
ba.bpf.point(corres(766), 0.05):
ba.bpf.end(corres(1024), 0);
f15 0 1024 7 1 41 0.1 52 0.02 155 0 776 0
//f16 0 1024 7 0 145 0.1 155 0.5 103 0.95 103 1 103 0.96 103 0.5 155 0.1 157 0
env(7) = f16;
f16 = ba.bpf.start(0, 0):
ba.bpf.point(corres(145), 0.1):
ba.bpf.point(corres(300), 0.5):
ba.bpf.point(corres(403), 0.95):
ba.bpf.point(corres(506), 0.99):
ba.bpf.point(corres(609), 0.96):
ba.bpf.point(corres(712), 0.5):
ba.bpf.point(corres(867), 0.1):
ba.bpf.end(corres(1024), 0);
f17 0 1024 7 1 103 0.5 145 0.25 259 0.1 259 0.05 259 0 | declare name "Cameleon Keyboard";
declare author "ER";
import("stdfaust.lib");
marimkey(n) = os.osc(octave(n)) * (0.1)
*(trigger(n+1) : envelope : fi.lowpass(1,500));
process = hand <: par(i, 10, marimkey(i)) :> *(3);
hand = hslider("[1]Instrument Hand[acc:1 0 -10 0 10]", 5, 0, 10, 1);
hight = hslider("[2]Hight[acc:0 1 -10 0 30]", 5, 1, 10, 0.3) : si.smooth(0.99):min(12):max(1);
envsize = hslider("[3]Note Duration (BPF Envelope)[unit:s][acc:2 0 -10 0 10]", 0.2, 0.1, 0.5, 0.01) * (ma.SR) : si.smooth(0.999): min(44100) : max(4410) : int;
freq(0) = 164.81;
freq(1) = 174.61;
freq(d) = freq(d-2);
octave(d) = freq(d)* hight;
upfront(x) = x>x';
counter(g)= (+(1):*(1-g))~_;
position(a,x) = abs(x - a) < 0.5;
trigger(p) = position(p) : upfront : counter;
typeEnv = vslider("[4]Envelope Type (BPF Envelope)[style:radio{'f9':0;'f11':1;'f15':2;'f17':3}]",0,0,3,1):int;
envelope = _<:sum(i, 4, tabchowning.env(i) * (abs(typeEnv - (i)) < 0.5));
tabchowning = environment
{
corres(x) = int(x*envsize/1024);
env(0) = f9;
f9 = ba.bpf.start(0, 0):
ba.bpf.point(corres(2), 0.25):
ba.bpf.point(corres(4), 0.5):
ba.bpf.point(corres(10), 0.9):
ba.bpf.point(corres(248), 0.25):
ba.bpf.point(corres(507), 0.1):
ba.bpf.point(corres(766), 0.05):
ba.bpf.end(corres(1024), 0);
env(1) = f11;
f11 = ba.bpf.start(0, 0):
ba.bpf.point(corres(93), 0.02):
ba.bpf.point(corres(145), 0.1):
ba.bpf.point(corres(248), 0.5):
ba.bpf.point(corres(300), 0.95):
ba.bpf.point(corres(331), 0.99):
ba.bpf.point(corres(869), 0.95):
ba.bpf.point(corres(921), 0.9):
ba.bpf.point(corres(973), 0.05):
ba.bpf.end(corres(1024), 0);
env(2) = f15;
f15 = ba.bpf.start(0, 0):
ba.bpf.point(corres(2), 0.25):
ba.bpf.point(corres(4), 0.5):
ba.bpf.point(corres(6), 0.99):
ba.bpf.point(corres(41), 0.1):
ba.bpf.point(corres(93), 0.02):
ba.bpf.point(corres(248), 0):
ba.bpf.end(corres(1024), 0);
env(3) = f17;
f17 = ba.bpf.start(0, 0):
ba.bpf.point(corres(2), 0.25):
ba.bpf.point(corres(4), 0.5):
ba.bpf.point(corres(6), 0.99):
ba.bpf.point(corres(103), 0.5):
ba.bpf.point(corres(248), 0.25):
ba.bpf.point(corres(403), 0.1):
ba.bpf.point(corres(507), 0.05):
ba.bpf.point(corres(766), 0):
ba.bpf.end(corres(1024), 0);
};
|
e342634dede97aeb8e5575eecd55391c902c233d674c06d4b07a4c855b791375 | RuolunWeng/ruolunweng.github.io | Whistles.dsp | declare name "Whistles";
declare author "ER";
declare version "1.0";
import("stdfaust.lib");
instrument = library("instruments.lib");
/* ============ Description ============== :
- 3 triple whistles, one per axis.
- Head = reverberation & whistles heard from far away.
- Bottom + rotation = proximity of the whistles.
- Rapid swings trigger volume increases (fishing rod/rocking/swing).
*/
//----------------- INSTRUMENT ------------------//
process = vgroup("Whistles", nOise.white * (0.5) <: par(f, 3, par(i, 3, whistle(f,i)) )):>_<: frEEvErb.fvb :>_;
whistle(f,n) = BP(f,n) : EQ(f,n) : @(10 + (12000*n)) <:Reson(f,0),_*(1.5):> * (vibrato) *(vibratoEnv(f))*(gain(f));
//----------------- NOISES ----------------------//
nOise = environment{
// white no.noise generator:
random = +(12345)~*(1103515245);
white = random/2147483647.0;
//pink no.noise filter:
p = f : (+ ~ g)
with {
f(x) = 0.04957526213389*x - 0.06305581334498*x' + 0.01483220320740*x'';
g(x) = 1.80116083982126*x - 0.80257737639225*x';
};
//pink no.noise generator:
pink = (white : p);
};
//----------------- FILTERS -------------------//
//gain = 1 - (Q * 0.1);
freq(0) = hslider("[1]Frequency 0[unit:Hz][acc:2 1 -10 0 10]", 110, 50, 220, 0.01):si.smooth(0.999);
freq(1) = hslider("[2]Frequency 1[unit:Hz][acc:2 1 -10 0 10]", 400, 220, 660, 0.01):si.smooth(0.999);
freq(2) = hslider("[3]Frequency 2[unit:Hz][acc:2 1 -10 0 10]", 820, 660, 1100, 0.01):si.smooth(0.999);
gain(n) = hslider("[5]Volume %n[style:knob][acc:%n 0 -10 0 20]", 0.2, 0, 2, 0.001):si.smooth(0.999);
hight(f,n) = freq(f)* (n+1);
level = 20;
Lowf(f,n) = hight(f,n) - Q;
Highf(f,n) = hight(f,n) + Q;
Q = 2 : si.smooth(0.999);//hslider("Q - Filter Bandwidth[style:knob][unit:Hz][tooltip: Band width = 2 * Frequency]",2.5,1,10,0.0001):si.smooth(0.999);
BP(f,n) = fi.bandpass(1, Lowf(f,n), Highf(f,n));
EQ(f,n) = fi.peak_eq(level,hight(f,n),Q) : fi.lowpass(1, 6000);
Reson(f,n) = fi.resonbp(hight(f,n),Q,1) : fi.lowpass(1,3000);
//----------------- VIBRATO --------------------//
vibrato = vibratoGain * os.osc(vibratoFreq) + (1-vibratoGain);
vibratoGain = 0.17;//hslider("Vibrato Volume[style:knob][acc:1 0 -10 0 10]", 0.1, 0.05, 0.5, 0.01) : si.smooth(0.999);
vibratoFreq = vfreq; //hslider("Vibrato Frequency[unit:Hz][acc:0 0 -10 0 12]", 5, 0, 10, 0.001) : si.smooth(0.999);
//--------------------------- Random Frequency ---------------------------
vfreq = pulsawhistle.gate : randfreq : si.smooth(0.99) : fi.lowpass (1, 3000);
randfreq(g) = no.noise : sampleAndhold(sahgate(g))*(10)
with {
sampleAndhold(t) = select2(t) ~_;
sahgate(g) = g : upfront : counter -(3) <=(0);
upfront(x) = abs(x-x')>0.5;
counter(g) = (+(1):*(1-g))~_;
};
//----------------------- Pulsar --------------------------------------
pulsawhistle = environment{
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
fade = (0.5); // min > 0 pour eviter division par 0
speed = 0.5;
proba = 0.9; //hslider ("h:Pulse/Probability[unit:%][style:knob][acc:1 1 -10 0 10]", 88,75,100,1) *(0.01):fi.lowpass(1,1);
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
//----------------------- Vibrato Envelope ----------------------------
vibratoEnv(n) = (instrument.envVibrato(b,a,s,r,t(n)))
with {
b = 0.25;
a = 0.1;
s = 100;
r = 0.8;
t(n) = hslider("[4]Envelope ON/OFF %n[acc:%n 0 -12 0 2]", 1, 0, 1, 1);
};
//------------------------ Freeverb ------------------------------------
frEEvErb = environment{
// Freeverb
//---------
fvb = vgroup("[6]Freeverb", fxctrl(fixedgain, wetSlider, stereoReverb(combfeed, allpassfeed, dampSlider, stereospread)));
//======================================================
//
// Freeverb
// Faster version using fixed delays (20% gain)
//
//======================================================
// Constant Parameters
//--------------------
fixedgain = 0.015; //value of the gain of fxctrl
scalewet = 3.0;
scaledry = 2.0;
scaledamp = 0.4;
scaleroom = 0.28;
offsetroom = 0.7;
initialroom = 0.5;
initialdamp = 0.5;
initialwet = 1.0/scalewet;
initialdry = 0;
initialwidth= 1.0;
initialmode = 0.0;
freezemode = 0.5;
stereospread= 23;
allpassfeed = 0.5; //feedback of the delays used in allpass filters
// Filter Parameters
//------------------
combtuningL1 = 1116;
combtuningL2 = 1188;
combtuningL3 = 1277;
combtuningL4 = 1356;
combtuningL5 = 1422;
combtuningL6 = 1491;
combtuningL7 = 1557;
combtuningL8 = 1617;
allpasstuningL1 = 556;
allpasstuningL2 = 441;
allpasstuningL3 = 341;
allpasstuningL4 = 225;
// Control Sliders
//--------------------
// Damp : filters the high frequencies of the echoes (especially active for great values of RoomSize)
// RoomSize : size of the reverberation room
// Dry : original signal
// Wet : reverberated signal
//dampSlider = hslider("Damp",0.5, 0, 1, 0.025)*scaledamp;
dampSlider = 0.7*scaledamp;
roomsizeSlider = hslider("[7]Reverberation Room Size (Freeverb)[style:knob][acc:1 1 -10 0 13]", 0.5, 0.1, 0.9, 0.025) : si.smooth(0.999) : min(0.9) :max(0.1) *scaleroom + offsetroom;
wetSlider = hslider("[6]Reverberation Intensity (Freeverb)[style:knob][acc:1 1 -10 0 15]", 0.3333, 0.1, 0.9, 0.025) : si.smooth(0.999) : min(0.9) :max(0.1);
combfeed = roomsizeSlider;
// Comb and Allpass filters
//-------------------------
allpass(dt,fb) = (_,_ <: (*(fb),_:+:@(dt)), -) ~ _ : (!,_);
comb(dt, fb, damp) = (+:@(dt)) ~ (*(1-damp) : (+ ~ *(damp)) : *(fb));
// Reverb components
//------------------
monoReverb(fb1, fb2, damp, spread)
= _ <: comb(combtuningL1+spread, fb1, damp),
comb(combtuningL2+spread, fb1, damp),
comb(combtuningL3+spread, fb1, damp),
comb(combtuningL4+spread, fb1, damp),
comb(combtuningL5+spread, fb1, damp),
comb(combtuningL6+spread, fb1, damp),
comb(combtuningL7+spread, fb1, damp),
comb(combtuningL8+spread, fb1, damp)
+>
allpass (allpasstuningL1+spread, fb2)
: allpass (allpasstuningL2+spread, fb2)
: allpass (allpasstuningL3+spread, fb2)
: allpass (allpasstuningL4+spread, fb2)
;
stereoReverb(fb1, fb2, damp, spread)
= + <: monoReverb(fb1, fb2, damp, 0), monoReverb(fb1, fb2, damp, spread);
// fxctrl : add an input gain and a wet-dry control to a stereo FX
//----------------------------------------------------------------
fxctrl(g,w,Fx) = _,_ <: (*(g),*(g) : Fx : *(w),*(w)), *(1-w), *(1-w) +> _,_;
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/Whistles.dsp | faust | ============ Description ============== :
- 3 triple whistles, one per axis.
- Head = reverberation & whistles heard from far away.
- Bottom + rotation = proximity of the whistles.
- Rapid swings trigger volume increases (fishing rod/rocking/swing).
----------------- INSTRUMENT ------------------//
----------------- NOISES ----------------------//
white no.noise generator:
pink no.noise filter:
pink no.noise generator:
----------------- FILTERS -------------------//
gain = 1 - (Q * 0.1);
hslider("Q - Filter Bandwidth[style:knob][unit:Hz][tooltip: Band width = 2 * Frequency]",2.5,1,10,0.0001):si.smooth(0.999);
----------------- VIBRATO --------------------//
hslider("Vibrato Volume[style:knob][acc:1 0 -10 0 10]", 0.1, 0.05, 0.5, 0.01) : si.smooth(0.999);
hslider("Vibrato Frequency[unit:Hz][acc:0 0 -10 0 12]", 5, 0, 10, 0.001) : si.smooth(0.999);
--------------------------- Random Frequency ---------------------------
----------------------- Pulsar --------------------------------------
min > 0 pour eviter division par 0
hslider ("h:Pulse/Probability[unit:%][style:knob][acc:1 1 -10 0 10]", 88,75,100,1) *(0.01):fi.lowpass(1,1);
----------------------- Vibrato Envelope ----------------------------
------------------------ Freeverb ------------------------------------
Freeverb
---------
======================================================
Freeverb
Faster version using fixed delays (20% gain)
======================================================
Constant Parameters
--------------------
value of the gain of fxctrl
feedback of the delays used in allpass filters
Filter Parameters
------------------
Control Sliders
--------------------
Damp : filters the high frequencies of the echoes (especially active for great values of RoomSize)
RoomSize : size of the reverberation room
Dry : original signal
Wet : reverberated signal
dampSlider = hslider("Damp",0.5, 0, 1, 0.025)*scaledamp;
Comb and Allpass filters
-------------------------
Reverb components
------------------
fxctrl : add an input gain and a wet-dry control to a stereo FX
---------------------------------------------------------------- | declare name "Whistles";
declare author "ER";
declare version "1.0";
import("stdfaust.lib");
instrument = library("instruments.lib");
process = vgroup("Whistles", nOise.white * (0.5) <: par(f, 3, par(i, 3, whistle(f,i)) )):>_<: frEEvErb.fvb :>_;
whistle(f,n) = BP(f,n) : EQ(f,n) : @(10 + (12000*n)) <:Reson(f,0),_*(1.5):> * (vibrato) *(vibratoEnv(f))*(gain(f));
nOise = environment{
random = +(12345)~*(1103515245);
white = random/2147483647.0;
p = f : (+ ~ g)
with {
f(x) = 0.04957526213389*x - 0.06305581334498*x' + 0.01483220320740*x'';
g(x) = 1.80116083982126*x - 0.80257737639225*x';
};
pink = (white : p);
};
freq(0) = hslider("[1]Frequency 0[unit:Hz][acc:2 1 -10 0 10]", 110, 50, 220, 0.01):si.smooth(0.999);
freq(1) = hslider("[2]Frequency 1[unit:Hz][acc:2 1 -10 0 10]", 400, 220, 660, 0.01):si.smooth(0.999);
freq(2) = hslider("[3]Frequency 2[unit:Hz][acc:2 1 -10 0 10]", 820, 660, 1100, 0.01):si.smooth(0.999);
gain(n) = hslider("[5]Volume %n[style:knob][acc:%n 0 -10 0 20]", 0.2, 0, 2, 0.001):si.smooth(0.999);
hight(f,n) = freq(f)* (n+1);
level = 20;
Lowf(f,n) = hight(f,n) - Q;
Highf(f,n) = hight(f,n) + Q;
BP(f,n) = fi.bandpass(1, Lowf(f,n), Highf(f,n));
EQ(f,n) = fi.peak_eq(level,hight(f,n),Q) : fi.lowpass(1, 6000);
Reson(f,n) = fi.resonbp(hight(f,n),Q,1) : fi.lowpass(1,3000);
vibrato = vibratoGain * os.osc(vibratoFreq) + (1-vibratoGain);
vfreq = pulsawhistle.gate : randfreq : si.smooth(0.99) : fi.lowpass (1, 3000);
randfreq(g) = no.noise : sampleAndhold(sahgate(g))*(10)
with {
sampleAndhold(t) = select2(t) ~_;
sahgate(g) = g : upfront : counter -(3) <=(0);
upfront(x) = abs(x-x')>0.5;
counter(g) = (+(1):*(1-g))~_;
};
pulsawhistle = environment{
gate = phasor_bin(1) :-(0.001):pulsar;
ratio_env = (0.5);
speed = 0.5;
phasor_bin (init) = (+(float(speed)/float(ma.SR)) : fmod(_,1.0)) ~ *(init);
pulsar = _<:(((_)<(ratio_env)):@(100))*((proba)>((_),(no.noise:abs):ba.latch));
};
vibratoEnv(n) = (instrument.envVibrato(b,a,s,r,t(n)))
with {
b = 0.25;
a = 0.1;
s = 100;
r = 0.8;
t(n) = hslider("[4]Envelope ON/OFF %n[acc:%n 0 -12 0 2]", 1, 0, 1, 1);
};
frEEvErb = environment{
fvb = vgroup("[6]Freeverb", fxctrl(fixedgain, wetSlider, stereoReverb(combfeed, allpassfeed, dampSlider, stereospread)));
scalewet = 3.0;
scaledry = 2.0;
scaledamp = 0.4;
scaleroom = 0.28;
offsetroom = 0.7;
initialroom = 0.5;
initialdamp = 0.5;
initialwet = 1.0/scalewet;
initialdry = 0;
initialwidth= 1.0;
initialmode = 0.0;
freezemode = 0.5;
stereospread= 23;
combtuningL1 = 1116;
combtuningL2 = 1188;
combtuningL3 = 1277;
combtuningL4 = 1356;
combtuningL5 = 1422;
combtuningL6 = 1491;
combtuningL7 = 1557;
combtuningL8 = 1617;
allpasstuningL1 = 556;
allpasstuningL2 = 441;
allpasstuningL3 = 341;
allpasstuningL4 = 225;
dampSlider = 0.7*scaledamp;
roomsizeSlider = hslider("[7]Reverberation Room Size (Freeverb)[style:knob][acc:1 1 -10 0 13]", 0.5, 0.1, 0.9, 0.025) : si.smooth(0.999) : min(0.9) :max(0.1) *scaleroom + offsetroom;
wetSlider = hslider("[6]Reverberation Intensity (Freeverb)[style:knob][acc:1 1 -10 0 15]", 0.3333, 0.1, 0.9, 0.025) : si.smooth(0.999) : min(0.9) :max(0.1);
combfeed = roomsizeSlider;
allpass(dt,fb) = (_,_ <: (*(fb),_:+:@(dt)), -) ~ _ : (!,_);
comb(dt, fb, damp) = (+:@(dt)) ~ (*(1-damp) : (+ ~ *(damp)) : *(fb));
monoReverb(fb1, fb2, damp, spread)
= _ <: comb(combtuningL1+spread, fb1, damp),
comb(combtuningL2+spread, fb1, damp),
comb(combtuningL3+spread, fb1, damp),
comb(combtuningL4+spread, fb1, damp),
comb(combtuningL5+spread, fb1, damp),
comb(combtuningL6+spread, fb1, damp),
comb(combtuningL7+spread, fb1, damp),
comb(combtuningL8+spread, fb1, damp)
+>
allpass (allpasstuningL1+spread, fb2)
: allpass (allpasstuningL2+spread, fb2)
: allpass (allpasstuningL3+spread, fb2)
: allpass (allpasstuningL4+spread, fb2)
;
stereoReverb(fb1, fb2, damp, spread)
= + <: monoReverb(fb1, fb2, damp, 0), monoReverb(fb1, fb2, damp, spread);
fxctrl(g,w,Fx) = _,_ <: (*(g),*(g) : Fx : *(w),*(w)), *(1-w), *(1-w) +> _,_;
};
|
be50ffb5deacc0b959268a3be4e43882a9b3efd1e97d11a1c0da6e5228d43bcc | RuolunWeng/Cage | cage.dsp | import("stdfaust.lib");
switch= en.adsre(0.5,0.1,1,1,checkbox("switch"));
switch2= en.adsre(0.5,0.1,1,1,1-checkbox("switch"));
process = part_radio,part_dialog :> _,_;
part_radio = playerPad, playerVoice :> _*(switch),_*(switch);
part_dialog = Ququ , Birds :> _*(switch2),_*(switch2);
////////
selectPart(sec,offset) =no.pink_noise: de.delay(ma.SR*sec, offset) :abs:*(100):int:ba.sAndH(ba.pulse(ma.SR*sec))%3;
selectPart21(sec,offset) =no.pink_noise: de.delay(ma.SR*sec, offset) :abs:*(500):int:ba.sAndH(ba.pulse(ma.SR*sec))%21;
level = hslider("level", 0.5, 0, 1, 0.01);
s1 = soundfile("[url:{'RADIO1.wav';'RADIO2.wav';'RADIO3.wav'}]",1);
sample1 = so.sound(s1, 0);
sample2 = so.sound(s1, 1);
sample3 = so.sound(s1, 2);
s2 = soundfile("[url:{'dialog1.wav';
'dialog2.wav';
'dialog3.wav';
'lenny.wav';
'rg3.wav';
'macron3.wav';
'macron4.wav';
'jiaomaidiao1.wav';
'jiaomaidiao2.wav';
'jiaomaidiao3.wav';
'nainai1.wav';
'nainai2.wav';
'xinwen1.wav';
'xinwen2.wav';
'journal1.wav';
'journal2.wav';
'journal3.wav';
'macron1.wav';
'macron2.wav';
'rg1.wav';
'rg2.wav'}]",1);
volume(sec,offset) = ba.pulse(ma.SR*sec):ba.peakholder(ma.SR*sec/2):ba.ramp(18000): de.delay(ma.SR*sec, offset);
pad1 = volume(20,0) * select3(selectPart(20,0),sample1.loop_speed_level(1,0.3),sample2.loop_speed_level(1, 0.4),sample3.loop_speed_level(1, 0.5));
pad2 = volume(10,ma.SR) * select3(selectPart(10,ma.SR),sample1.loop_speed_level(1,0.3),sample2.loop_speed_level(1, 0.4),sample3.loop_speed_level(1, 0.5));
pad3 = volume(12,ma.SR*5) * select3(selectPart(12,ma.SR*5),sample1.loop_speed_level(1,0.3),sample2.loop_speed_level(1, 0.4),sample3.loop_speed_level(1, 0.5));
pad4 = volume(16,ma.SR*3) * select3(selectPart(16,ma.SR*3),sample1.loop_speed_level(1,0.3),sample2.loop_speed_level(1, 0.4),sample3.loop_speed_level(1, 0.5));
playerPad = pad1 , pad2 , pad3 , pad4 :> _,_ ;
voice1 = volume(4,0) * ba.selectmulti(ma.SR/10, par(i, 18, so.sound(s2, i).loop_speed_level(1,0.5)), selectPart21(4,0)) ;
voice2 = volume(6,ma.SR) * ba.selectmulti(ma.SR/10, par(i, 18, so.sound(s2, i).loop_speed_level(1,0.5)), selectPart21(6,ma.SR)) ;
playerVoice = voice1 , voice2;
// Ququ
Ququ = hgroup("QUQU", os.osc(freq): ringmod : AsrEnvelop <:volumebird);
freq = hslider("Frequency [unit:Hz] ", 4000, 70, 5000, 0.01):si.smooth(0.999);
ringmod = _<:_,*(os.oscs(freq)):drywet
with {
freq = hslider ( "Modulation Frequency[scale:log]", 97,0.001,100,0.001):si.smooth(0.999);
drywet(x,y) = (1-c)*x + c*y;
c = hslider("Modulation intensity[style:knob][unit:%]", 70,0,100,0.01)*(0.01):si.smooth(0.999);
};
autoTrig = ba.beat(t) * (abs(no.noise) <= p) : trigger(4800)
with {
t = hslider("Speed[style:knob][acc:0 1 -10 0 10]", 250, 120, 480, 0.1) : si.smooth(0.999);
p = hslider("Probability[unit:%][style:knob][acc:0 1 -10 0 10]", 90, 25, 100, 1)*(0.01) : si.smooth(0.999);
trigger(n) = upfront : release(n) : >(0.0)
with {
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
};
};
AsrEnvelop = *(en.asr(a,s,r,autoTrig)):_
with {
a = hslider("Envelope Attack[unit:s][style:knob]", 0.03, 0.01, 2, 0.01) : si.smooth(0.999);
s = 1;
r = hslider("Envelope Release[unit:s][style:knob]", 0.04, 0.01, 5, 0.01) : si.smooth(0.999);
//t=ba.pulsen(hslider("Envelope Period", 6000, 0, 44100, 1), hslider("Envelope Length", 15000, 0, 44100, 1));
//t = ba.beat (hslider("Speed [style:knob]", 120, 0, 480, 0.1) );
p = hslider("Probability (Granulator)[unit:%][style:knob][acc:0 1 -10 0 10]", 90, 25, 100, 1)*(0.01) : si.smooth(0.999);
};
volumebird = par(i,2,*(hslider("Volume", 0.02, 0, 1, 0.01):si.smooth(0.999)));
// Birdy from Grame playground
Birds = hgroup("Birds", mainOsc(noteTrig : rdm(72,94) : mtof , noteTrig) * envWrapper(noteTrig, ampEnv, amp_xp(2510)) : fi.lowpass(1, 2000) *(0.8) <: _,_, (rdmPanner : panSte) : panConnect : *,* : reverb);
// AUTO TRIGGER
autoTriger = ba.beat(t) * (abs(no.noise) <= p) : trigger(48) //tempo(2.5*t))
with {
t = hslider("[1]Speed (Granulator)[style:knob][acc:0 1 -10 0 10]", 120, 120, 480, 0.1) : si.smooth(0.999);
p = hslider("[2]Probability (Granulator)[unit:%][style:knob][acc:1 0 -10 0 10]", 30, 25, 100, 1)*(0.01) : si.smooth(0.999);
trigger(n) = upfront : release(n) : >(0.0) with {
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
};
};
// BIRD TRIGGER
noteTrig = autoTriger : min(1.0);
//noteTrig = autoTrig;
// OSCILLATORS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
/* base */
carrierOsc(freq) = os.osci(freq);
modOsc(freq) = os.triangleN(3,freq);
/* fm oscillator */
mainOsc(freq,trig) = freq <: +(*(harmRatio <: +(*(envWrapper(trig,harmEnv,harm_xp(1700))))) : modOsc : *(modIndex <: +(*(envWrapper(trig,modIndexEnv,modIndex_xp(550)))))) <: +(*(envWrapper(trig,freqEnv,freq_xp(943)))) : carrierOsc;
envWrapper(trig,env,sus) = trig : mstosamps(rdm(100,3000)), sus : hitLength : env;
// FIXED PARAMETERS - - - - - - - - - - - - - - - - - - - - - - - - - - -
/* fm */
harmRatio = 0.063;
modIndex = 3.24;
// TIME FUNCTIONS - - - - - - - - - - - - - - - - - - - - - - - - - - - -
metro(ms) = (%(+(1),mstosamps(ms))) ~_ : ==(1);
mstosamps(ms) = ms : /(1000) * ma.SR : int;
rdmInc = _ <: @(1), @(2) : + : *(2994.2313) : int : +(38125);
rdm(rdmin,rdmax) = _,(fmod(_,rdmax - rdmin : int) ~ rdmInc : +(rdmin)) : gater : -(1) : abs;
gater = (_,_,_ <: !,_,!,_,!,!,!,!,_ : select2) ~_;
// MIDI RELATED - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
/* midi pitch */
mtof(midinote) = pow(2,(midinote - 69) / 12) * 440;
// ENVELOPPES - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
/* envelope "reader" (phaser) */
hitLength(length,sustain) = *((==(length,@(length,1))), +(1))~_ <: gater(<(sustain));
/* amplitude envelope */
ampEnvbpf = ba.bpf.start(0, 0) :
ba.bpf.point(amp_xp(60.241), 1.) :
ba.bpf.point(amp_xp(461.847), 0.) :
ba.bpf.point(amp_xp(582.329), 0.928) :
ba.bpf.point(amp_xp(682.731), 0.5) :
ba.bpf.point(amp_xp(983.936), 0.) :
ba.bpf.point(amp_xp(1064.257), 0.) :
ba.bpf.point(amp_xp(1345.382), 0.) :
ba.bpf.point(amp_xp(1526.105), 0.) :
ba.bpf.point(amp_xp(1746.988), 0.) :
ba.bpf.point(amp_xp(1827.309), 0.) :
ba.bpf.point(amp_xp(2088.353), 0.) :
ba.bpf.point(amp_xp(2188.755), 0.) : /* sustain point */
ba.bpf.end(amp_xp(2510.040), 0.);
ampEnv = ampEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
amp_xp(x) = x * ma.SR / 1000. * ampEnv_speed;
ampEnv_speed = noteTrig : rdm(0,2000) : /(1000);
/* freq envelope */
freqEnvbpf = ba.bpf.start(0, 0) :
ba.bpf.point(freq_xp(147.751), 1.) :
ba.bpf.point(freq_xp(193.213), 0.) :
ba.bpf.point(freq_xp(318.233), yp) :
ba.bpf.point(freq_xp(431.888), 0.) :
ba.bpf.point(freq_xp(488.715), 0.434) :
ba.bpf.point(freq_xp(613.735), yp) :
ba.bpf.point(freq_xp(659.197), 1.) :
ba.bpf.point(freq_xp(716.024), yp) :
ba.bpf.point(freq_xp(806.948), 1.) :
ba.bpf.point(freq_xp(829.679), yp) : /* sustain point */
ba.bpf.end(freq_xp(943.333), 0.);
freqEnv = freqEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
freq_xp(x) = x * ma.SR / 1000. * freqEnv_speed;
freqEnv_speed = noteTrig : rdm(0,2000) : /(1000);
yp = noteTrig : rdm(0,1000) : /(1000);
/* harmRatio envelope */
harmEnvbpf = ba.bpf.start(0, 0.) :
ba.bpf.point(harm_xp(863.454), 0.490) :
ba.bpf.point(harm_xp(865), 0.) :
ba.bpf.point (harm_xp(1305.221), 1.) :
ba.bpf.point(harm_xp(1646.586), 0.) : /* sustain point */
ba.bpf.end(harm_xp(1700), 0.);
harmEnv = harmEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
harm_xp(x) = x * ma.SR / 1000. * harmEnv_speed;
harmEnv_speed = noteTrig : rdm(0,2000) : /(1000);
/* modIndex envelope */
modIndexEnvbpf = ba.bpf.start(0, 0.) :
ba.bpf.point(modIndex_xp(240.964), 0.554) :
ba.bpf.point(modIndex_xp(502.068), 0.) : /* sustain point */
ba.bpf.end(modIndex_xp(550), 0.);
modIndexEnv = modIndexEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
modIndex_xp(x) = x * ma.SR / 1000. * modIndexEnv_speed;
modIndexEnv_speed = noteTrig : rdm(0,2000) : /(1000);
// PANNER STEREO - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
panSte = _ <: -(1,_),_ : sqrt,sqrt;
rdmPanner = noteTrig : rdm(0,1000) : /(1000);
/* cable crosser = 1,3 & 2,4 */
panConnect = _,_,_,_ <: _,!,!,!,!,!,_,!,!,_,!,!,!,!,!,_;
// REVERB BASED OF ZITA - - - - - - - - - - - - - - - - - - - - - - - - - -
reverb(x,y) = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax,x,y)
: out_eq : dry_wet(x,y) : out_level
with {
fsmax = 48000.0; // highest sampling rate that will be used
rdel = 60;
f1 = 200;
t60dc = 3;
t60m = 2;
f2 = 6000;
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 {
tpbt = wcT/sqrt(max(0,g)); // tan(ma.PI*B/ma.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
};
eq1f = 315;
eq1l = 0;
eq1q = 3;
eq2f = 1500;
eq2l = 0.0;
eq2q = 3.0;
//out_group(x) = 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;
};
presence = hslider("[3]Proximity (InstrReverb)[style:knob][acc:1 0 -15 0 10]", 0.5, 0, 1, 0.01) : si.smooth(0.999);
drywet = 1 - 2*presence;
out_level = *(gain),*(gain);
//gain = vslider("[5]Reverberation Volume[unit:dB][style:knob]", -20, -70, 20, 0.1)
gain = -30 : +(6*presence) : ba.db2linear : si.smooth(0.999);
};
| https://raw.githubusercontent.com/RuolunWeng/Cage/1a3945e3327f2e4654a8b67c48cb68bf1e8ce1fb/cage/cage.dsp | faust | //////
Ququ
t=ba.pulsen(hslider("Envelope Period", 6000, 0, 44100, 1), hslider("Envelope Length", 15000, 0, 44100, 1));
t = ba.beat (hslider("Speed [style:knob]", 120, 0, 480, 0.1) );
Birdy from Grame playground
AUTO TRIGGER
tempo(2.5*t))
BIRD TRIGGER
noteTrig = autoTrig;
OSCILLATORS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
base
fm oscillator
FIXED PARAMETERS - - - - - - - - - - - - - - - - - - - - - - - - - - -
fm
TIME FUNCTIONS - - - - - - - - - - - - - - - - - - - - - - - - - - - -
MIDI RELATED - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
midi pitch
ENVELOPPES - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
envelope "reader" (phaser)
amplitude envelope
sustain point
freq envelope
sustain point
harmRatio envelope
sustain point
modIndex envelope
sustain point
PANNER STEREO - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
cable crosser = 1,3 & 2,4
REVERB BASED OF ZITA - - - - - - - - - - - - - - - - - - - - - - - - - -
highest sampling rate that will be used
tan(ma.PI*B/ma.SR), B bw in Hz (Q^2 ~ g/4)
peak frequency in rad/sample
peak gain
out_group(x) = x; //fdn_group(hgroup("[5] Output", x));
gain = vslider("[5]Reverberation Volume[unit:dB][style:knob]", -20, -70, 20, 0.1) | import("stdfaust.lib");
switch= en.adsre(0.5,0.1,1,1,checkbox("switch"));
switch2= en.adsre(0.5,0.1,1,1,1-checkbox("switch"));
process = part_radio,part_dialog :> _,_;
part_radio = playerPad, playerVoice :> _*(switch),_*(switch);
part_dialog = Ququ , Birds :> _*(switch2),_*(switch2);
selectPart(sec,offset) =no.pink_noise: de.delay(ma.SR*sec, offset) :abs:*(100):int:ba.sAndH(ba.pulse(ma.SR*sec))%3;
selectPart21(sec,offset) =no.pink_noise: de.delay(ma.SR*sec, offset) :abs:*(500):int:ba.sAndH(ba.pulse(ma.SR*sec))%21;
level = hslider("level", 0.5, 0, 1, 0.01);
s1 = soundfile("[url:{'RADIO1.wav';'RADIO2.wav';'RADIO3.wav'}]",1);
sample1 = so.sound(s1, 0);
sample2 = so.sound(s1, 1);
sample3 = so.sound(s1, 2);
s2 = soundfile("[url:{'dialog1.wav';
'dialog2.wav';
'dialog3.wav';
'lenny.wav';
'rg3.wav';
'macron3.wav';
'macron4.wav';
'jiaomaidiao1.wav';
'jiaomaidiao2.wav';
'jiaomaidiao3.wav';
'nainai1.wav';
'nainai2.wav';
'xinwen1.wav';
'xinwen2.wav';
'journal1.wav';
'journal2.wav';
'journal3.wav';
'macron1.wav';
'macron2.wav';
'rg1.wav';
'rg2.wav'}]",1);
volume(sec,offset) = ba.pulse(ma.SR*sec):ba.peakholder(ma.SR*sec/2):ba.ramp(18000): de.delay(ma.SR*sec, offset);
pad1 = volume(20,0) * select3(selectPart(20,0),sample1.loop_speed_level(1,0.3),sample2.loop_speed_level(1, 0.4),sample3.loop_speed_level(1, 0.5));
pad2 = volume(10,ma.SR) * select3(selectPart(10,ma.SR),sample1.loop_speed_level(1,0.3),sample2.loop_speed_level(1, 0.4),sample3.loop_speed_level(1, 0.5));
pad3 = volume(12,ma.SR*5) * select3(selectPart(12,ma.SR*5),sample1.loop_speed_level(1,0.3),sample2.loop_speed_level(1, 0.4),sample3.loop_speed_level(1, 0.5));
pad4 = volume(16,ma.SR*3) * select3(selectPart(16,ma.SR*3),sample1.loop_speed_level(1,0.3),sample2.loop_speed_level(1, 0.4),sample3.loop_speed_level(1, 0.5));
playerPad = pad1 , pad2 , pad3 , pad4 :> _,_ ;
voice1 = volume(4,0) * ba.selectmulti(ma.SR/10, par(i, 18, so.sound(s2, i).loop_speed_level(1,0.5)), selectPart21(4,0)) ;
voice2 = volume(6,ma.SR) * ba.selectmulti(ma.SR/10, par(i, 18, so.sound(s2, i).loop_speed_level(1,0.5)), selectPart21(6,ma.SR)) ;
playerVoice = voice1 , voice2;
Ququ = hgroup("QUQU", os.osc(freq): ringmod : AsrEnvelop <:volumebird);
freq = hslider("Frequency [unit:Hz] ", 4000, 70, 5000, 0.01):si.smooth(0.999);
ringmod = _<:_,*(os.oscs(freq)):drywet
with {
freq = hslider ( "Modulation Frequency[scale:log]", 97,0.001,100,0.001):si.smooth(0.999);
drywet(x,y) = (1-c)*x + c*y;
c = hslider("Modulation intensity[style:knob][unit:%]", 70,0,100,0.01)*(0.01):si.smooth(0.999);
};
autoTrig = ba.beat(t) * (abs(no.noise) <= p) : trigger(4800)
with {
t = hslider("Speed[style:knob][acc:0 1 -10 0 10]", 250, 120, 480, 0.1) : si.smooth(0.999);
p = hslider("Probability[unit:%][style:knob][acc:0 1 -10 0 10]", 90, 25, 100, 1)*(0.01) : si.smooth(0.999);
trigger(n) = upfront : release(n) : >(0.0)
with {
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
};
};
AsrEnvelop = *(en.asr(a,s,r,autoTrig)):_
with {
a = hslider("Envelope Attack[unit:s][style:knob]", 0.03, 0.01, 2, 0.01) : si.smooth(0.999);
s = 1;
r = hslider("Envelope Release[unit:s][style:knob]", 0.04, 0.01, 5, 0.01) : si.smooth(0.999);
p = hslider("Probability (Granulator)[unit:%][style:knob][acc:0 1 -10 0 10]", 90, 25, 100, 1)*(0.01) : si.smooth(0.999);
};
volumebird = par(i,2,*(hslider("Volume", 0.02, 0, 1, 0.01):si.smooth(0.999)));
Birds = hgroup("Birds", mainOsc(noteTrig : rdm(72,94) : mtof , noteTrig) * envWrapper(noteTrig, ampEnv, amp_xp(2510)) : fi.lowpass(1, 2000) *(0.8) <: _,_, (rdmPanner : panSte) : panConnect : *,* : reverb);
with {
t = hslider("[1]Speed (Granulator)[style:knob][acc:0 1 -10 0 10]", 120, 120, 480, 0.1) : si.smooth(0.999);
p = hslider("[2]Probability (Granulator)[unit:%][style:knob][acc:1 0 -10 0 10]", 30, 25, 100, 1)*(0.01) : si.smooth(0.999);
trigger(n) = upfront : release(n) : >(0.0) with {
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
};
};
noteTrig = autoTriger : min(1.0);
carrierOsc(freq) = os.osci(freq);
modOsc(freq) = os.triangleN(3,freq);
mainOsc(freq,trig) = freq <: +(*(harmRatio <: +(*(envWrapper(trig,harmEnv,harm_xp(1700))))) : modOsc : *(modIndex <: +(*(envWrapper(trig,modIndexEnv,modIndex_xp(550)))))) <: +(*(envWrapper(trig,freqEnv,freq_xp(943)))) : carrierOsc;
envWrapper(trig,env,sus) = trig : mstosamps(rdm(100,3000)), sus : hitLength : env;
harmRatio = 0.063;
modIndex = 3.24;
metro(ms) = (%(+(1),mstosamps(ms))) ~_ : ==(1);
mstosamps(ms) = ms : /(1000) * ma.SR : int;
rdmInc = _ <: @(1), @(2) : + : *(2994.2313) : int : +(38125);
rdm(rdmin,rdmax) = _,(fmod(_,rdmax - rdmin : int) ~ rdmInc : +(rdmin)) : gater : -(1) : abs;
gater = (_,_,_ <: !,_,!,_,!,!,!,!,_ : select2) ~_;
mtof(midinote) = pow(2,(midinote - 69) / 12) * 440;
hitLength(length,sustain) = *((==(length,@(length,1))), +(1))~_ <: gater(<(sustain));
ampEnvbpf = ba.bpf.start(0, 0) :
ba.bpf.point(amp_xp(60.241), 1.) :
ba.bpf.point(amp_xp(461.847), 0.) :
ba.bpf.point(amp_xp(582.329), 0.928) :
ba.bpf.point(amp_xp(682.731), 0.5) :
ba.bpf.point(amp_xp(983.936), 0.) :
ba.bpf.point(amp_xp(1064.257), 0.) :
ba.bpf.point(amp_xp(1345.382), 0.) :
ba.bpf.point(amp_xp(1526.105), 0.) :
ba.bpf.point(amp_xp(1746.988), 0.) :
ba.bpf.point(amp_xp(1827.309), 0.) :
ba.bpf.point(amp_xp(2088.353), 0.) :
ba.bpf.end(amp_xp(2510.040), 0.);
ampEnv = ampEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
amp_xp(x) = x * ma.SR / 1000. * ampEnv_speed;
ampEnv_speed = noteTrig : rdm(0,2000) : /(1000);
freqEnvbpf = ba.bpf.start(0, 0) :
ba.bpf.point(freq_xp(147.751), 1.) :
ba.bpf.point(freq_xp(193.213), 0.) :
ba.bpf.point(freq_xp(318.233), yp) :
ba.bpf.point(freq_xp(431.888), 0.) :
ba.bpf.point(freq_xp(488.715), 0.434) :
ba.bpf.point(freq_xp(613.735), yp) :
ba.bpf.point(freq_xp(659.197), 1.) :
ba.bpf.point(freq_xp(716.024), yp) :
ba.bpf.point(freq_xp(806.948), 1.) :
ba.bpf.end(freq_xp(943.333), 0.);
freqEnv = freqEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
freq_xp(x) = x * ma.SR / 1000. * freqEnv_speed;
freqEnv_speed = noteTrig : rdm(0,2000) : /(1000);
yp = noteTrig : rdm(0,1000) : /(1000);
harmEnvbpf = ba.bpf.start(0, 0.) :
ba.bpf.point(harm_xp(863.454), 0.490) :
ba.bpf.point(harm_xp(865), 0.) :
ba.bpf.point (harm_xp(1305.221), 1.) :
ba.bpf.end(harm_xp(1700), 0.);
harmEnv = harmEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
harm_xp(x) = x * ma.SR / 1000. * harmEnv_speed;
harmEnv_speed = noteTrig : rdm(0,2000) : /(1000);
modIndexEnvbpf = ba.bpf.start(0, 0.) :
ba.bpf.point(modIndex_xp(240.964), 0.554) :
ba.bpf.end(modIndex_xp(550), 0.);
modIndexEnv = modIndexEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
modIndex_xp(x) = x * ma.SR / 1000. * modIndexEnv_speed;
modIndexEnv_speed = noteTrig : rdm(0,2000) : /(1000);
panSte = _ <: -(1,_),_ : sqrt,sqrt;
rdmPanner = noteTrig : rdm(0,1000) : /(1000);
panConnect = _,_,_,_ <: _,!,!,!,!,!,_,!,!,_,!,!,!,!,!,_;
reverb(x,y) = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax,x,y)
: out_eq : dry_wet(x,y) : out_level
with {
rdel = 60;
f1 = 200;
t60dc = 3;
t60m = 2;
f2 = 6000;
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 {
};
eq1f = 315;
eq1l = 0;
eq1q = 3;
eq2f = 1500;
eq2l = 0.0;
eq2q = 3.0;
dry_wet(x,y) = *(wet) + dry*x, *(wet) + dry*y with {
wet = 0.5*(drywet+1.0);
dry = 1.0-wet;
};
presence = hslider("[3]Proximity (InstrReverb)[style:knob][acc:1 0 -15 0 10]", 0.5, 0, 1, 0.01) : si.smooth(0.999);
drywet = 1 - 2*presence;
out_level = *(gain),*(gain);
gain = -30 : +(6*presence) : ba.db2linear : si.smooth(0.999);
};
|
bd010418c721f2647897f15f36fabc24f753a7d42d735c461349c9c4dd06f4a1 | RuolunWeng/ruolunweng.github.io | Birds.dsp | declare name "bird";
declare author "Pierre Cochard";
/* Modifications by Grame July 2014 */
import("stdfaust.lib");
/* =============== DESCRIPTION ================= :
- Bird singing generator.
- Head = Reverberation, birds heard from far away.
- Bottom = Maximum proximity of the birds.
- Right = maximum speed of whistles.
- Left = minimum speed, birds rarely heard.
*/
// PROCESS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
process = hgroup("Birds", mainOsc(noteTrig : rdm(72,94) : mtof , noteTrig) * envWrapper(noteTrig, ampEnv, amp_xp(2510)) : fi.lowpass(1, 2000) *(0.8) <: _,_, (rdmPanner : panSte) : panConnect : *,* : reverb);
// AUTO TRIGGER
autoTrig = ba.beat(t) * (abs(no.noise) <= p) : trigger(48) //tempo(2.5*t))
with {
t = hslider("[1]Speed (Granulator)[style:knob][acc:0 1 -10 0 10]", 240, 120, 480, 0.1) : si.smooth(0.999);
p = hslider("[2]Probability (Granulator)[unit:%][style:knob][acc:1 0 -10 0 10]", 50, 25, 100, 1)*(0.01) : si.smooth(0.999);
trigger(n) = upfront : release(n) : >(0.0) with {
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
};
};
// BIRD TRIGGER
noteTrig = autoTrig : min(1.0);
//noteTrig = autoTrig;
// OSCILLATORS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
/* base */
carrierOsc(freq) = os.osci(freq);
modOsc(freq) = os.triangleN(3,freq);
/* fm oscillator */
mainOsc(freq,trig) = freq <: +(*(harmRatio <: +(*(envWrapper(trig,harmEnv,harm_xp(1700))))) : modOsc : *(modIndex <: +(*(envWrapper(trig,modIndexEnv,modIndex_xp(550)))))) <: +(*(envWrapper(trig,freqEnv,freq_xp(943)))) : carrierOsc;
envWrapper(trig,env,sus) = trig : mstosamps(rdm(100,3000)), sus : hitLength : env;
// FIXED PARAMETERS - - - - - - - - - - - - - - - - - - - - - - - - - - -
/* fm */
harmRatio = 0.063;
modIndex = 3.24;
// TIME FUNCTIONS - - - - - - - - - - - - - - - - - - - - - - - - - - - -
metro(ms) = (%(+(1),mstosamps(ms))) ~_ : ==(1);
mstosamps(ms) = ms : /(1000) * ma.SR : int;
rdmInc = _ <: @(1), @(2) : + : *(2994.2313) : int : +(38125);
rdm(rdmin,rdmax) = _,(fmod(_,rdmax - rdmin : int) ~ rdmInc : +(rdmin)) : gater : -(1) : abs;
gater = (_,_,_ <: !,_,!,_,!,!,!,!,_ : select2) ~_;
// MIDI RELATED - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
/* midi pitch */
mtof(midinote) = pow(2,(midinote - 69) / 12) * 440;
// ENVELOPPES - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
/* envelope "reader" (phaser) */
hitLength(length,sustain) = *((==(length,@(length,1))), +(1))~_ <: gater(<(sustain));
/* amplitude envelope */
ampEnvbpf = ba.bpf.start(0, 0) :
ba.bpf.point(amp_xp(60.241), 1.) :
ba.bpf.point(amp_xp(461.847), 0.) :
ba.bpf.point(amp_xp(582.329), 0.928) :
ba.bpf.point(amp_xp(682.731), 0.5) :
ba.bpf.point(amp_xp(983.936), 0.) :
ba.bpf.point(amp_xp(1064.257), 0.) :
ba.bpf.point(amp_xp(1345.382), 0.) :
ba.bpf.point(amp_xp(1526.105), 0.) :
ba.bpf.point(amp_xp(1746.988), 0.) :
ba.bpf.point(amp_xp(1827.309), 0.) :
ba.bpf.point(amp_xp(2088.353), 0.) :
ba.bpf.point(amp_xp(2188.755), 0.) : /* sustain point */
ba.bpf.end(amp_xp(2510.040), 0.);
ampEnv = ampEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
amp_xp(x) = x * ma.SR / 1000. * ampEnv_speed;
ampEnv_speed = noteTrig : rdm(0,2000) : /(1000);
/* freq envelope */
freqEnvbpf = ba.bpf.start(0, 0) :
ba.bpf.point(freq_xp(147.751), 1.) :
ba.bpf.point(freq_xp(193.213), 0.) :
ba.bpf.point(freq_xp(318.233), yp) :
ba.bpf.point(freq_xp(431.888), 0.) :
ba.bpf.point(freq_xp(488.715), 0.434) :
ba.bpf.point(freq_xp(613.735), yp) :
ba.bpf.point(freq_xp(659.197), 1.) :
ba.bpf.point(freq_xp(716.024), yp) :
ba.bpf.point(freq_xp(806.948), 1.) :
ba.bpf.point(freq_xp(829.679), yp) : /* sustain point */
ba.bpf.end(freq_xp(943.333), 0.);
freqEnv = freqEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
freq_xp(x) = x * ma.SR / 1000. * freqEnv_speed;
freqEnv_speed = noteTrig : rdm(0,2000) : /(1000);
yp = noteTrig : rdm(0,1000) : /(1000);
/* harmRatio envelope */
harmEnvbpf = ba.bpf.start(0, 0.) :
ba.bpf.point(harm_xp(863.454), 0.490) :
ba.bpf.point(harm_xp(865), 0.) :
ba.bpf.point (harm_xp(1305.221), 1.) :
ba.bpf.point(harm_xp(1646.586), 0.) : /* sustain point */
ba.bpf.end(harm_xp(1700), 0.);
harmEnv = harmEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
harm_xp(x) = x * ma.SR / 1000. * harmEnv_speed;
harmEnv_speed = noteTrig : rdm(0,2000) : /(1000);
/* modIndex envelope */
modIndexEnvbpf = ba.bpf.start(0, 0.) :
ba.bpf.point(modIndex_xp(240.964), 0.554) :
ba.bpf.point(modIndex_xp(502.068), 0.) : /* sustain point */
ba.bpf.end(modIndex_xp(550), 0.);
modIndexEnv = modIndexEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
modIndex_xp(x) = x * ma.SR / 1000. * modIndexEnv_speed;
modIndexEnv_speed = noteTrig : rdm(0,2000) : /(1000);
// PANNER STEREO - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
panSte = _ <: -(1,_),_ : sqrt,sqrt;
rdmPanner = noteTrig : rdm(0,1000) : /(1000);
/* cable crosser = 1,3 & 2,4 */
panConnect = _,_,_,_ <: _,!,!,!,!,!,_,!,!,_,!,!,!,!,!,_;
// REVERB BASED OF ZITA - - - - - - - - - - - - - - - - - - - - - - - - - -
reverb(x,y) = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax,x,y)
: out_eq : dry_wet(x,y) : out_level
with {
fsmax = 48000.0; // highest sampling rate that will be used
rdel = 60;
f1 = 200;
t60dc = 3;
t60m = 2;
f2 = 6000;
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 {
tpbt = wcT/sqrt(max(0,g)); // tan(ma.PI*B/ma.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
};
eq1f = 315;
eq1l = 0;
eq1q = 3;
eq2f = 1500;
eq2l = 0.0;
eq2q = 3.0;
//out_group(x) = 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;
};
presence = hslider("[3]Proximity (InstrReverb)[style:knob][acc:1 0 -15 0 10]", 0.5, 0, 1, 0.01) : si.smooth(0.999);
drywet = 1 - 2*presence;
out_level = *(gain),*(gain);
//gain = vslider("[5]Reverberation Volume[unit:dB][style:knob]", -20, -70, 20, 0.1)
gain = -10 : +(6*presence) : ba.db2linear : si.smooth(0.999);
};
| https://raw.githubusercontent.com/RuolunWeng/ruolunweng.github.io/035564bb7e36eb4e810ca80077ffa8a9d3e5130b/faustplayground/faust-modules/combined/Birds.dsp | faust | Modifications by Grame July 2014
=============== DESCRIPTION ================= :
- Bird singing generator.
- Head = Reverberation, birds heard from far away.
- Bottom = Maximum proximity of the birds.
- Right = maximum speed of whistles.
- Left = minimum speed, birds rarely heard.
PROCESS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
AUTO TRIGGER
tempo(2.5*t))
BIRD TRIGGER
noteTrig = autoTrig;
OSCILLATORS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
base
fm oscillator
FIXED PARAMETERS - - - - - - - - - - - - - - - - - - - - - - - - - - -
fm
TIME FUNCTIONS - - - - - - - - - - - - - - - - - - - - - - - - - - - -
MIDI RELATED - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
midi pitch
ENVELOPPES - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
envelope "reader" (phaser)
amplitude envelope
sustain point
freq envelope
sustain point
harmRatio envelope
sustain point
modIndex envelope
sustain point
PANNER STEREO - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
cable crosser = 1,3 & 2,4
REVERB BASED OF ZITA - - - - - - - - - - - - - - - - - - - - - - - - - -
highest sampling rate that will be used
tan(ma.PI*B/ma.SR), B bw in Hz (Q^2 ~ g/4)
peak frequency in rad/sample
peak gain
out_group(x) = x; //fdn_group(hgroup("[5] Output", x));
gain = vslider("[5]Reverberation Volume[unit:dB][style:knob]", -20, -70, 20, 0.1) | declare name "bird";
declare author "Pierre Cochard";
import("stdfaust.lib");
process = hgroup("Birds", mainOsc(noteTrig : rdm(72,94) : mtof , noteTrig) * envWrapper(noteTrig, ampEnv, amp_xp(2510)) : fi.lowpass(1, 2000) *(0.8) <: _,_, (rdmPanner : panSte) : panConnect : *,* : reverb);
with {
t = hslider("[1]Speed (Granulator)[style:knob][acc:0 1 -10 0 10]", 240, 120, 480, 0.1) : si.smooth(0.999);
p = hslider("[2]Probability (Granulator)[unit:%][style:knob][acc:1 0 -10 0 10]", 50, 25, 100, 1)*(0.01) : si.smooth(0.999);
trigger(n) = upfront : release(n) : >(0.0) with {
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
release(n) = + ~ decay(n);
};
};
noteTrig = autoTrig : min(1.0);
carrierOsc(freq) = os.osci(freq);
modOsc(freq) = os.triangleN(3,freq);
mainOsc(freq,trig) = freq <: +(*(harmRatio <: +(*(envWrapper(trig,harmEnv,harm_xp(1700))))) : modOsc : *(modIndex <: +(*(envWrapper(trig,modIndexEnv,modIndex_xp(550)))))) <: +(*(envWrapper(trig,freqEnv,freq_xp(943)))) : carrierOsc;
envWrapper(trig,env,sus) = trig : mstosamps(rdm(100,3000)), sus : hitLength : env;
harmRatio = 0.063;
modIndex = 3.24;
metro(ms) = (%(+(1),mstosamps(ms))) ~_ : ==(1);
mstosamps(ms) = ms : /(1000) * ma.SR : int;
rdmInc = _ <: @(1), @(2) : + : *(2994.2313) : int : +(38125);
rdm(rdmin,rdmax) = _,(fmod(_,rdmax - rdmin : int) ~ rdmInc : +(rdmin)) : gater : -(1) : abs;
gater = (_,_,_ <: !,_,!,_,!,!,!,!,_ : select2) ~_;
mtof(midinote) = pow(2,(midinote - 69) / 12) * 440;
hitLength(length,sustain) = *((==(length,@(length,1))), +(1))~_ <: gater(<(sustain));
ampEnvbpf = ba.bpf.start(0, 0) :
ba.bpf.point(amp_xp(60.241), 1.) :
ba.bpf.point(amp_xp(461.847), 0.) :
ba.bpf.point(amp_xp(582.329), 0.928) :
ba.bpf.point(amp_xp(682.731), 0.5) :
ba.bpf.point(amp_xp(983.936), 0.) :
ba.bpf.point(amp_xp(1064.257), 0.) :
ba.bpf.point(amp_xp(1345.382), 0.) :
ba.bpf.point(amp_xp(1526.105), 0.) :
ba.bpf.point(amp_xp(1746.988), 0.) :
ba.bpf.point(amp_xp(1827.309), 0.) :
ba.bpf.point(amp_xp(2088.353), 0.) :
ba.bpf.end(amp_xp(2510.040), 0.);
ampEnv = ampEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
amp_xp(x) = x * ma.SR / 1000. * ampEnv_speed;
ampEnv_speed = noteTrig : rdm(0,2000) : /(1000);
freqEnvbpf = ba.bpf.start(0, 0) :
ba.bpf.point(freq_xp(147.751), 1.) :
ba.bpf.point(freq_xp(193.213), 0.) :
ba.bpf.point(freq_xp(318.233), yp) :
ba.bpf.point(freq_xp(431.888), 0.) :
ba.bpf.point(freq_xp(488.715), 0.434) :
ba.bpf.point(freq_xp(613.735), yp) :
ba.bpf.point(freq_xp(659.197), 1.) :
ba.bpf.point(freq_xp(716.024), yp) :
ba.bpf.point(freq_xp(806.948), 1.) :
ba.bpf.end(freq_xp(943.333), 0.);
freqEnv = freqEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
freq_xp(x) = x * ma.SR / 1000. * freqEnv_speed;
freqEnv_speed = noteTrig : rdm(0,2000) : /(1000);
yp = noteTrig : rdm(0,1000) : /(1000);
harmEnvbpf = ba.bpf.start(0, 0.) :
ba.bpf.point(harm_xp(863.454), 0.490) :
ba.bpf.point(harm_xp(865), 0.) :
ba.bpf.point (harm_xp(1305.221), 1.) :
ba.bpf.end(harm_xp(1700), 0.);
harmEnv = harmEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
harm_xp(x) = x * ma.SR / 1000. * harmEnv_speed;
harmEnv_speed = noteTrig : rdm(0,2000) : /(1000);
modIndexEnvbpf = ba.bpf.start(0, 0.) :
ba.bpf.point(modIndex_xp(240.964), 0.554) :
ba.bpf.end(modIndex_xp(550), 0.);
modIndexEnv = modIndexEnvbpf : si.smooth(0.999) : fi.lowpass(1, 3000);
modIndex_xp(x) = x * ma.SR / 1000. * modIndexEnv_speed;
modIndexEnv_speed = noteTrig : rdm(0,2000) : /(1000);
panSte = _ <: -(1,_),_ : sqrt,sqrt;
rdmPanner = noteTrig : rdm(0,1000) : /(1000);
panConnect = _,_,_,_ <: _,!,!,!,!,!,_,!,!,_,!,!,!,!,!,_;
reverb(x,y) = re.zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax,x,y)
: out_eq : dry_wet(x,y) : out_level
with {
rdel = 60;
f1 = 200;
t60dc = 3;
t60m = 2;
f2 = 6000;
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 {
};
eq1f = 315;
eq1l = 0;
eq1q = 3;
eq2f = 1500;
eq2l = 0.0;
eq2q = 3.0;
dry_wet(x,y) = *(wet) + dry*x, *(wet) + dry*y with {
wet = 0.5*(drywet+1.0);
dry = 1.0-wet;
};
presence = hslider("[3]Proximity (InstrReverb)[style:knob][acc:1 0 -15 0 10]", 0.5, 0, 1, 0.01) : si.smooth(0.999);
drywet = 1 - 2*presence;
out_level = *(gain),*(gain);
gain = -10 : +(6*presence) : ba.db2linear : si.smooth(0.999);
};
|
d2c266aba9e6cb2579af44b241875e9e7e531618b9812a9aec4843250ab31025 | LSSN/2020-01-23-dsp-3a-antoniotestoni | Veri.dsp | // Sintesi sottrattiva significa analizzare attraverso strumenti come spettrogramma,
// oscilloscopio e sonogramma il suono o rumore preso in considerazione seconda vari
// aspetti.
//GS - SOTTRATTIVA, lo dice la parola stessa, significa sottrarre da qualcosa che
//GS - abbia uno spettro complesso, per ottenere un prodotto derivato.
//GS - è un processo molto simile a quello della scultura, scavare, modellare.
// Oscilloscopio è uno strumento che ci permette attraverso il grafico
// ampiezza-tempo di vedere i processi di rarefazione e di condensazione di un suono.
// Lo spettrogramma ci serve a capire attraverso il grafico freq.-tempo l'intensità di
// un suono nel tempo. Il sonogramma ci serve per capire l'altezza di un
// Il rumore è un qualcosa che non trasmette informazioni o che non capiamo,
// quindi un qualcosa di cui non riusciamo a distinguerne ne la frequenza ne l'ampiezza.
// Noi uttilizzando filtri passa-bassa(lowpass) e filtri passa-alta(highpass) possiamo
// distinguere questi 2 parametri. Il filtro lowpass si traduce in passabasso, serve ad
// attenuare l'ampiezza delle frequenze sopra un punto di taglio. Il filtro highpass
// serve a far passare le frequenze alte di un determinato suono o rumore da un punto
// di taglio. Nel linguaggio standard di faust il rumore è scritto con il codice no.noise,
// il filtro bassa è scritto con il codice fi.lowpass
import("stdfaust.lib");
fcut = vslider("cut-off [style:knob][scale:exp]", 1000, 20, 20000, 1);
order = 8;
process = no.noise : fi.lowpass(order,fcut) : fi.highpass(order,fcut);
| https://raw.githubusercontent.com/LSSN/2020-01-23-dsp-3a-antoniotestoni/cce97d07b807f69cbed40a82fccf08079de707de/Veri.dsp | faust | Sintesi sottrattiva significa analizzare attraverso strumenti come spettrogramma,
oscilloscopio e sonogramma il suono o rumore preso in considerazione seconda vari
aspetti.
GS - SOTTRATTIVA, lo dice la parola stessa, significa sottrarre da qualcosa che
GS - abbia uno spettro complesso, per ottenere un prodotto derivato.
GS - è un processo molto simile a quello della scultura, scavare, modellare.
Oscilloscopio è uno strumento che ci permette attraverso il grafico
ampiezza-tempo di vedere i processi di rarefazione e di condensazione di un suono.
Lo spettrogramma ci serve a capire attraverso il grafico freq.-tempo l'intensità di
un suono nel tempo. Il sonogramma ci serve per capire l'altezza di un
Il rumore è un qualcosa che non trasmette informazioni o che non capiamo,
quindi un qualcosa di cui non riusciamo a distinguerne ne la frequenza ne l'ampiezza.
Noi uttilizzando filtri passa-bassa(lowpass) e filtri passa-alta(highpass) possiamo
distinguere questi 2 parametri. Il filtro lowpass si traduce in passabasso, serve ad
attenuare l'ampiezza delle frequenze sopra un punto di taglio. Il filtro highpass
serve a far passare le frequenze alte di un determinato suono o rumore da un punto
di taglio. Nel linguaggio standard di faust il rumore è scritto con il codice no.noise,
il filtro bassa è scritto con il codice fi.lowpass |
import("stdfaust.lib");
fcut = vslider("cut-off [style:knob][scale:exp]", 1000, 20, 20000, 1);
order = 8;
process = no.noise : fi.lowpass(order,fcut) : fi.highpass(order,fcut);
|
503cb9ece6ccfaf157c1ccdb7f662fefd8a46e7da66b2013a1221ca90280b371 | LSSN/2019-11-29-dsp-camillacongiu | basspand.dsp | import("stdfaust.lib");
process= no.noise :fi.bandpass(10,1000,2000) : *(vslider("gain",0,0,1,0.1));
//cicciona <3scusaaaaa<3<3<3<3<3<3<3
//i numeri che indentificano lo slider sono il valore iniziale, valore minimo, valore massimo e step incrementale.
//step incrementale è la precisione di passaggio da uno spettro all'altro
//il valore iniziale è il valore che deve assumere il controllo quando azioniamo il programma.
//il valore inziale può essere solo tra il valore minimo e il valore massimo.
//lo slider in faust può essere sia verticale sia orizzontale, per cambiarlo da verticale (vslider) orizzontale (oslider).
//
| https://raw.githubusercontent.com/LSSN/2019-11-29-dsp-camillacongiu/4cf80dc9f2c40da1129b5dd2a0bdc3e6bed9c3b6/basspand.dsp | faust | cicciona <3scusaaaaa<3<3<3<3<3<3<3
i numeri che indentificano lo slider sono il valore iniziale, valore minimo, valore massimo e step incrementale.
step incrementale è la precisione di passaggio da uno spettro all'altro
il valore iniziale è il valore che deve assumere il controllo quando azioniamo il programma.
il valore inziale può essere solo tra il valore minimo e il valore massimo.
lo slider in faust può essere sia verticale sia orizzontale, per cambiarlo da verticale (vslider) orizzontale (oslider).
| import("stdfaust.lib");
process= no.noise :fi.bandpass(10,1000,2000) : *(vslider("gain",0,0,1,0.1));
|
e4fa87e6a55145431b9b59ef935ecc5b6cf0e96cd17b0ae5d302e8f2d7570c6e | LSSN/2020-01-23-2a-dsp-giulialostia1920 | verifica.dsp | //Crea un file di faust in cui esponi, in forma di commento, i principi della sintesi sottrattiva. Realizza un esempio attraverso un filtraggio passa banda.
//La sintesi sotrattiva sottrae
import("stdfaust.lib");
catof=vslider("cat-of [style:knob]",1000,20,2000,1);
process= fi. lowpass (2,catof) : fi.highpass (2,catof);
| https://raw.githubusercontent.com/LSSN/2020-01-23-2a-dsp-giulialostia1920/b416f87f60b48c539f23462593f95f6ceedd9b15/verifica.dsp | faust | Crea un file di faust in cui esponi, in forma di commento, i principi della sintesi sottrattiva. Realizza un esempio attraverso un filtraggio passa banda.
La sintesi sotrattiva sottrae |
import("stdfaust.lib");
catof=vslider("cat-of [style:knob]",1000,20,2000,1);
process= fi. lowpass (2,catof) : fi.highpass (2,catof);
|
1b7e875ec11b67ecef1b26e9e58bfb537928a645418e3fda5e846b985ab023d4 | LSSN/2020-01-23-dsp-3a-AlessandraVardeu | verifica.dsp | //per sintesi sottrativa si fa rimento ad modello di sinetsi sonora ancora uitilizzata nella musica elettronica che si occupa del processo di creazione del suono e si divide in tre aspetti:
// spettrogramma, sonogramma e forma d'onda.
// questo processo si applica attraverso le sorgenti sonore coovvero gli elemnti che generano il suono come i filtri, amplificatori.
// filtri sono gli oggeti di base epr la sintesi:
//filtro passaalto= fi.highpas
//filtro passabasso=fi.lowpass
// Nello spettrogramma l'ampiezzza si muove sul dominio della frquenza.
//lo spettrogramma è una rappresentazione grafica del suono che si basa su effettive misurazioni del suono in funzione
//delle variabili.
//nel sonogramma la frequenza si muove sul dominio del tempo.
//l sonogramma è una registrazione grafica fornita dal sonografo.
//nella forma d'onda ogni punto è osservabile sul dominio temporale., è la rapprsentazione grafica do un segnale, ampiezza e tempo.
//Prima di process va sempre scritto fcut e order
//ogni riga deve terminare con ;
// Elementi:
//no.noise=rumore
//fcut=frequenza di taglio
//vslider= primo modello di pomello
//i valori numerici di vlisder sono:
//valore di inizializzaione
//alore minimo
//valore massimo
//step=precisione
//order=ordine ;più aumenta l'ordine più il filtro è verticale,più l'ordine abbassa più il filtro è normale.
//gain=regola il volume
//style knob= fa diventare il primo modello di pomello normale ad un pomello circolare
//scale exp pomello esponenziale
//white noise=rumore bianco che ha tutte le frequenze e tutte le ampiezze(rettangolo completamente colorato),rumore banda larga.
//strumento che permette la visione completa delle cose è il microscopio
//analisi di un corpo ossia l'analisi della struttura molecolare
import ("stdfaust.lib");
fcut=vslider("[01] cut-off [scale:exp][style:knob]", 1000,20,20000,1) ;
order=128;
// init min max step
gain=vslider("[02] gain", -20, -96, 0, 0.1) : ba.db2linear ;
process=fi.lowpass(order,fcut) : fi.highpass(order,fcut) : *(gain) ;
| https://raw.githubusercontent.com/LSSN/2020-01-23-dsp-3a-AlessandraVardeu/b32ea6b917e6951725f4f2f5672e4aaa3e391040/verifica.dsp | faust | per sintesi sottrativa si fa rimento ad modello di sinetsi sonora ancora uitilizzata nella musica elettronica che si occupa del processo di creazione del suono e si divide in tre aspetti:
spettrogramma, sonogramma e forma d'onda.
questo processo si applica attraverso le sorgenti sonore coovvero gli elemnti che generano il suono come i filtri, amplificatori.
filtri sono gli oggeti di base epr la sintesi:
filtro passaalto= fi.highpas
filtro passabasso=fi.lowpass
Nello spettrogramma l'ampiezzza si muove sul dominio della frquenza.
lo spettrogramma è una rappresentazione grafica del suono che si basa su effettive misurazioni del suono in funzione
delle variabili.
nel sonogramma la frequenza si muove sul dominio del tempo.
l sonogramma è una registrazione grafica fornita dal sonografo.
nella forma d'onda ogni punto è osservabile sul dominio temporale., è la rapprsentazione grafica do un segnale, ampiezza e tempo.
Prima di process va sempre scritto fcut e order
ogni riga deve terminare con ;
Elementi:
no.noise=rumore
fcut=frequenza di taglio
vslider= primo modello di pomello
i valori numerici di vlisder sono:
valore di inizializzaione
alore minimo
valore massimo
step=precisione
order=ordine ;più aumenta l'ordine più il filtro è verticale,più l'ordine abbassa più il filtro è normale.
gain=regola il volume
style knob= fa diventare il primo modello di pomello normale ad un pomello circolare
scale exp pomello esponenziale
white noise=rumore bianco che ha tutte le frequenze e tutte le ampiezze(rettangolo completamente colorato),rumore banda larga.
strumento che permette la visione completa delle cose è il microscopio
analisi di un corpo ossia l'analisi della struttura molecolare
init min max step |
import ("stdfaust.lib");
fcut=vslider("[01] cut-off [scale:exp][style:knob]", 1000,20,20000,1) ;
order=128;
gain=vslider("[02] gain", -20, -96, 0, 0.1) : ba.db2linear ;
process=fi.lowpass(order,fcut) : fi.highpass(order,fcut) : *(gain) ;
|
adeabdd0084923ed5c6a16b8603077f144bb8f9e4b75baff2f7511019dd90166 | LSSN/2020-01-25-dsp-4a-camillacongiu | iubgbug.dsp | crea un file di faust in cui esponi, in forma di commento, i principi della sintesi sottrattiva.
//la sintesi sottrattiva è un modello di sintesi utilizzata nella musica elettronica dove una sorgente sonora viene filtrata da un punto di visa "spettrale", quindi sottraendo da essa le bande di frequenza. gli aspetti più importanti vengono evidenziati attraverso il processo dinamico, ovvero quando questa operazione di filtraggio si sviluppa durante l'evoluzione temporale del segnale.
il principio su cui si basa è un oscillatore che crea delle forme d'onda e attraverso un filtro alcune componenti dell'onda vengono selezionate, le altre escluse.
// numeri che indentificano lo slider sono il valore iniziale, valore minimo, valore massimo e step incrementale.
//step incrementale è la precisione di passaggio da uno spettro all'altro
//il valore iniziale è il valore che deve assumere il controllo quando azioniamo il programma.
//il valore inziale può essere solo tra il valore minimo e il valore massimo.
//lo slider in faust può essere sia verticale sia orizzontale, per cambiarlo da verticale (vslider) orizzontale (oslider).
// il rumore è un suono che non ha ampiezza e non trasmette informazioni.
// lo spettroscopio ci mostra il contenuto del suono nel dominio della frequenza, sull'asse delle x le frequenze sull'asse delle y l'ampiezza.
//l'oscilloscopio ci mostra il contenuto del suono nel dominio del tempo. sull'asse delle x abbiamo il tempo e sull'asse delle y abbiamo le ampiezze.
//lowpass: filtro che vuol dire "passa basso". è tra i più semplici.
//dato un punto di taglio il filtro di primo ordine attenua 6 dB per ottava.
//la velocità di un filtro è chiamata ordine.
//la frequenza di taglio si chiama "cut off"
import("stdfaust.lib");
gain = vslider("gain",0,0,1,0.1);
process = no.noise : fi.lowpass(8,6000) : fi.highpass(2,6000);
| https://raw.githubusercontent.com/LSSN/2020-01-25-dsp-4a-camillacongiu/637fbc69ad093fe1af96f951b49f2c1216ee63e0/iubgbug.dsp | faust | la sintesi sottrattiva è un modello di sintesi utilizzata nella musica elettronica dove una sorgente sonora viene filtrata da un punto di visa "spettrale", quindi sottraendo da essa le bande di frequenza. gli aspetti più importanti vengono evidenziati attraverso il processo dinamico, ovvero quando questa operazione di filtraggio si sviluppa durante l'evoluzione temporale del segnale.
numeri che indentificano lo slider sono il valore iniziale, valore minimo, valore massimo e step incrementale.
step incrementale è la precisione di passaggio da uno spettro all'altro
il valore iniziale è il valore che deve assumere il controllo quando azioniamo il programma.
il valore inziale può essere solo tra il valore minimo e il valore massimo.
lo slider in faust può essere sia verticale sia orizzontale, per cambiarlo da verticale (vslider) orizzontale (oslider).
il rumore è un suono che non ha ampiezza e non trasmette informazioni.
lo spettroscopio ci mostra il contenuto del suono nel dominio della frequenza, sull'asse delle x le frequenze sull'asse delle y l'ampiezza.
l'oscilloscopio ci mostra il contenuto del suono nel dominio del tempo. sull'asse delle x abbiamo il tempo e sull'asse delle y abbiamo le ampiezze.
lowpass: filtro che vuol dire "passa basso". è tra i più semplici.
dato un punto di taglio il filtro di primo ordine attenua 6 dB per ottava.
la velocità di un filtro è chiamata ordine.
la frequenza di taglio si chiama "cut off" | crea un file di faust in cui esponi, in forma di commento, i principi della sintesi sottrattiva.
il principio su cui si basa è un oscillatore che crea delle forme d'onda e attraverso un filtro alcune componenti dell'onda vengono selezionate, le altre escluse.
import("stdfaust.lib");
gain = vslider("gain",0,0,1,0.1);
process = no.noise : fi.lowpass(8,6000) : fi.highpass(2,6000);
|
a1f0a594c1b86bbdb596aee5d036732fb09c5c61a1fe63b8bc310cd5304b415b | elaforge/karya | guitar.dsp | import("stdfaust.lib");
declare description "Guitar model.";
declare control0_gate "Gate.";
declare control1_pitch "Pitch signal.";
declare control2_dyn "constant:Dynamic signal.";
declare control3_pos "constant:Pluck position.";
declare flags "impulse-gate";
process(gate, pitch, dyn, pluckPosition) =
nylonGuitar(stringLength, pluckPosition, gain, gate) * outGain
with {
outGain = .75;
gain = dyn;
stringLength = freq : pm.f2l;
freq = ba.midikey2hz(pitch);
};
nylonGuitar(stringLength, pluckPosition, gain, trigger) =
pm.pluckString(stringLength, 1,1.5, 1, gain, trigger)
: nylonGuitarModel(stringLength, pluckPosition);
nylonGuitarModel(stringLength, pluckPosition, excitation) = pm.endChain(egChain)
with {
egChain = pm.chain(guitarNuts(brightness, absorption)
: pm.nylonString(stringL, pluckPosition, excitation)
: pm.guitarBridge : pm.guitarBody : pm.out);
stringL = stringLength - lengthTuning;
lengthTuning = 0.11;
// brightness = 0.4;
// absorption = 0.5;
brightness = 0.9;
absorption = 0.2;
};
guitarNuts(brightness, absorption) =
pm.lTermination(0 - pm.bridgeFilter(brightness, absorption), pm.basicBlock);
/*
pm.bridgeFilter(brightness, absorption)
pluckString(stringLength, 1, 1.5, 1, gain, trigger)
: nylonGuitarModel(stringLength, pluckPosition);
*/
| https://raw.githubusercontent.com/elaforge/karya/471a2131f5a68b3b10b1a138e6f9ed1282980a18/Synth/Faust/dsp/guitar.dsp | faust | brightness = 0.4;
absorption = 0.5;
pm.bridgeFilter(brightness, absorption)
pluckString(stringLength, 1, 1.5, 1, gain, trigger)
: nylonGuitarModel(stringLength, pluckPosition);
| import("stdfaust.lib");
declare description "Guitar model.";
declare control0_gate "Gate.";
declare control1_pitch "Pitch signal.";
declare control2_dyn "constant:Dynamic signal.";
declare control3_pos "constant:Pluck position.";
declare flags "impulse-gate";
process(gate, pitch, dyn, pluckPosition) =
nylonGuitar(stringLength, pluckPosition, gain, gate) * outGain
with {
outGain = .75;
gain = dyn;
stringLength = freq : pm.f2l;
freq = ba.midikey2hz(pitch);
};
nylonGuitar(stringLength, pluckPosition, gain, trigger) =
pm.pluckString(stringLength, 1,1.5, 1, gain, trigger)
: nylonGuitarModel(stringLength, pluckPosition);
nylonGuitarModel(stringLength, pluckPosition, excitation) = pm.endChain(egChain)
with {
egChain = pm.chain(guitarNuts(brightness, absorption)
: pm.nylonString(stringL, pluckPosition, excitation)
: pm.guitarBridge : pm.guitarBody : pm.out);
stringL = stringLength - lengthTuning;
lengthTuning = 0.11;
brightness = 0.9;
absorption = 0.2;
};
guitarNuts(brightness, absorption) =
pm.lTermination(0 - pm.bridgeFilter(brightness, absorption), pm.basicBlock);
|
8b2310833810d10dc46b30c6d10384892b87f98a97d0daaf91d7c43e568650dc | elaforge/karya | blow_bottle.dsp | declare name "blowBottle";
declare description "Blown Bottle Instrument";
declare author "Romain Michon ([email protected])";
declare copyright "Romain Michon";
declare version "1.0";
declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
declare description "This object implements a helmholtz resonator (biquad
filter) with a polynomial jet excitation (a la Cook).";
import("stdfaust.lib");
inst = library("instruments.lib");
//==================== GUI SPECIFICATION ================
freq = nentry(
"h:_Basic/freq [1][unit:Hz] [tooltip:Tone frequency]",
440, 20, 20000,1);
gain = nentry(
"h:_Basic/gain [1][tooltip:Gain (value between 0 and 1)]",
1, 0, 1, 0.01);
gate = button("h:_Basic/gate [1][tooltip:noteOn = 1, noteOff = 0]");
noiseGain = hslider(
"h:_Physical_and_Nonlinearity/v:_Physical/noise_gain
[2][tooltip:Breath noise gain (value between 0 and 1)]",
0.5, 0, 1, 0.01)*2;
pressure = hslider(
"h:_Physical_and_Nonlinearity/v:_Physical/pressure
[2][tooltip:Breath pressure (value bewteen 0 and 1)]", 1, 0, 1, 0.01);
typeModulation = nentry(
"h:_Physical_and_Nonlinearity/v:_Nonlinear_Filter/modulation_type
[3][tooltip: 0=theta is modulated by the incoming signal;
1=theta is modulated by the averaged incoming signal;
2=theta is modulated by the squared incoming signal;
3=theta is modulated by a sine wave of frequency freqMod;
4=theta is modulated by a sine wave of frequency freq;]",
0, 0, 4, 1);
nonLinearity = hslider(
"h:_Physical_and_Nonlinearity/v:_Nonlinear_Filter/nonlinearity
[3][tooltip:Nonlinearity factor (value between 0 and 1)]",
0, 0, 1, 0.01);
frequencyMod = hslider(
"h:_Physical_and_Nonlinearity/v:_Nonlinear_Filter/modulation_frequency
[3][unit:Hz][tooltip:Frequency of the sine wave for the modulation of
theta (works if Modulation Type=3)]", 220, 20, 1000, 0.1);
nonLinAttack = hslider(
"h:_Physical_and_Nonlinearity/v:_Nonlinear_Filter/nonlinearity_attack
[3][unit:s][Attack duration of the nonlinearity]",
0.1, 0, 2, 0.01);
envelopeAttack = hslider(
"h:_Envelopes_and_Vibrato/v:_Envelope/envelope_attack
[5][unit:s][tooltip:Envelope attack duration]",
0.01, 0, 2, 0.01);
envelopeDecay = hslider(
"h:_Envelopes_and_Vibrato/v:_Envelope/envelope_decay
[5][unit:s][tooltip:Envelope decay duration]",
0.01, 0, 2, 0.01);
envelopeRelease = hslider(
"h:_Envelopes_and_Vibrato/v:_Envelope/envelope_release
[5][unit:s][tooltip:Envelope release duration]",
0.5, 0, 2, 0.01);
//==================== SIGNAL PROCESSING ================
// Nonlinear filter
//nonlinearities are created by the nonlinear passive allpass ladder filter
//declared in miscfilter.lib
//nonlinear filter order
nlfOrder = 6;
//attack - sustain - release envelope for nonlinearity (declared in
//instruments.lib)
envelopeMod = inst.en.asr(nonLinAttack, 1, envelopeRelease, gate);
//nonLinearModultor is declared in instruments.lib, it adapts allpassnn from
//miscfilter.lib for using it with waveguide instruments
nlfm = inst.nonLinearModulator((nonLinearity : si.smoo), envelopeMod, freq,
typeModulation, (frequencyMod : si.smoo), nlfOrder);
// Synthesis parameters computing and functions declaration
//botlle radius
bottleRadius = 0.999;
stereo = stereoizer(ma.SR / freq);
bandPassFilter = inst.bandPass(freq, bottleRadius);
// Algorithm implementation
//global envelope is of type attack - decay - sustain - release
envelopeG = gain * inst.en.adsr(
gain * envelopeAttack, envelopeDecay, 1, envelopeRelease, gate);
//pressure envelope is also ADSR
envelope = pressure * inst.en.adsr(gain*0.02, 0.01, 1, gain * 0.2, gate);
// breath pressure
breathPressure = envelope;
//breath noise
randPressure = noiseGain * no.noise * breathPressure;
process =
// differential pressure
(-(breathPressure)
<: ((+(1)) * randPressure : +(breathPressure)) - *(inst.jetTable), _
: bandPassFilter,_) ~ nlfm : !, _
//signal scaling
: fi.dcblocker * envelopeG * 0.5 : stereo; // : inst.instrReverb;
// stereoizer is declared in instruments.lib and implement a stereo
// spatialisation in function of the frequency period in number of samples
stereoizer(periodDuration) = _ <: _,widthdelay : stereopanner
with {
W = hslider("v:_Spat/spatial-width", 0.5, 0, 1, 0.01);
A = hslider("v:_Spat/pan-angle", 0.6, 0, 1, 0.01);
widthdelay = de.delay(4096,W*periodDuration/2);
stereopanner = _,_ : *(1.0-A), *(A);
};
| https://raw.githubusercontent.com/elaforge/karya/471a2131f5a68b3b10b1a138e6f9ed1282980a18/Synth/Faust/dsp/blow_bottle.dsp | faust | Synthesis Tool Kit 4.3 (MIT style license);
==================== GUI SPECIFICATION ================
==================== SIGNAL PROCESSING ================
Nonlinear filter
nonlinearities are created by the nonlinear passive allpass ladder filter
declared in miscfilter.lib
nonlinear filter order
attack - sustain - release envelope for nonlinearity (declared in
instruments.lib)
nonLinearModultor is declared in instruments.lib, it adapts allpassnn from
miscfilter.lib for using it with waveguide instruments
Synthesis parameters computing and functions declaration
botlle radius
Algorithm implementation
global envelope is of type attack - decay - sustain - release
pressure envelope is also ADSR
breath pressure
breath noise
differential pressure
signal scaling
: inst.instrReverb;
stereoizer is declared in instruments.lib and implement a stereo
spatialisation in function of the frequency period in number of samples | declare name "blowBottle";
declare description "Blown Bottle Instrument";
declare author "Romain Michon ([email protected])";
declare copyright "Romain Michon";
declare version "1.0";
declare description "This object implements a helmholtz resonator (biquad
filter) with a polynomial jet excitation (a la Cook).";
import("stdfaust.lib");
inst = library("instruments.lib");
freq = nentry(
"h:_Basic/freq [1][unit:Hz] [tooltip:Tone frequency]",
440, 20, 20000,1);
gain = nentry(
"h:_Basic/gain [1][tooltip:Gain (value between 0 and 1)]",
1, 0, 1, 0.01);
gate = button("h:_Basic/gate [1][tooltip:noteOn = 1, noteOff = 0]");
noiseGain = hslider(
"h:_Physical_and_Nonlinearity/v:_Physical/noise_gain
[2][tooltip:Breath noise gain (value between 0 and 1)]",
0.5, 0, 1, 0.01)*2;
pressure = hslider(
"h:_Physical_and_Nonlinearity/v:_Physical/pressure
[2][tooltip:Breath pressure (value bewteen 0 and 1)]", 1, 0, 1, 0.01);
typeModulation = nentry(
"h:_Physical_and_Nonlinearity/v:_Nonlinear_Filter/modulation_type
[3][tooltip: 0=theta is modulated by the incoming signal;
1=theta is modulated by the averaged incoming signal;
2=theta is modulated by the squared incoming signal;
3=theta is modulated by a sine wave of frequency freqMod;
4=theta is modulated by a sine wave of frequency freq;]",
0, 0, 4, 1);
nonLinearity = hslider(
"h:_Physical_and_Nonlinearity/v:_Nonlinear_Filter/nonlinearity
[3][tooltip:Nonlinearity factor (value between 0 and 1)]",
0, 0, 1, 0.01);
frequencyMod = hslider(
"h:_Physical_and_Nonlinearity/v:_Nonlinear_Filter/modulation_frequency
[3][unit:Hz][tooltip:Frequency of the sine wave for the modulation of
theta (works if Modulation Type=3)]", 220, 20, 1000, 0.1);
nonLinAttack = hslider(
"h:_Physical_and_Nonlinearity/v:_Nonlinear_Filter/nonlinearity_attack
[3][unit:s][Attack duration of the nonlinearity]",
0.1, 0, 2, 0.01);
envelopeAttack = hslider(
"h:_Envelopes_and_Vibrato/v:_Envelope/envelope_attack
[5][unit:s][tooltip:Envelope attack duration]",
0.01, 0, 2, 0.01);
envelopeDecay = hslider(
"h:_Envelopes_and_Vibrato/v:_Envelope/envelope_decay
[5][unit:s][tooltip:Envelope decay duration]",
0.01, 0, 2, 0.01);
envelopeRelease = hslider(
"h:_Envelopes_and_Vibrato/v:_Envelope/envelope_release
[5][unit:s][tooltip:Envelope release duration]",
0.5, 0, 2, 0.01);
nlfOrder = 6;
envelopeMod = inst.en.asr(nonLinAttack, 1, envelopeRelease, gate);
nlfm = inst.nonLinearModulator((nonLinearity : si.smoo), envelopeMod, freq,
typeModulation, (frequencyMod : si.smoo), nlfOrder);
bottleRadius = 0.999;
stereo = stereoizer(ma.SR / freq);
bandPassFilter = inst.bandPass(freq, bottleRadius);
envelopeG = gain * inst.en.adsr(
gain * envelopeAttack, envelopeDecay, 1, envelopeRelease, gate);
envelope = pressure * inst.en.adsr(gain*0.02, 0.01, 1, gain * 0.2, gate);
breathPressure = envelope;
randPressure = noiseGain * no.noise * breathPressure;
process =
(-(breathPressure)
<: ((+(1)) * randPressure : +(breathPressure)) - *(inst.jetTable), _
: bandPassFilter,_) ~ nlfm : !, _
stereoizer(periodDuration) = _ <: _,widthdelay : stereopanner
with {
W = hslider("v:_Spat/spatial-width", 0.5, 0, 1, 0.01);
A = hslider("v:_Spat/pan-angle", 0.6, 0, 1, 0.01);
widthdelay = de.delay(4096,W*periodDuration/2);
stereopanner = _,_ : *(1.0-A), *(A);
};
|
a72a7ca3cbf87e7889760c8f4fdc81fb89a9bdb42999c23c008aa52053f4a5d0 | aravind-sadharani/puretones-music-room | tanpura-string.dsp | import("stdfaust.lib");
PureTonesString(coarsefreq,period,finetune,variance,delay) = string(freq*(1+delta)/2,gamma) + string(freq*(1-delta)/2,gamma) : *(gain)
with {
freq = coarsefreq;
delta = vslider("[04]Variance",variance,0,20,0.1)/10000;
gamma = 0.5;
gate = vgroup("[00]Play String",os.lf_pulsetrainpos(1/period,0.3):@(ma.SR*delay*period));
gain = 10^((vslider("[08]Gain",0,-20,20,0.1)-6) : /(20));
envelope1 = en.adsr(0.1*period,0.3*period,0.2,0.3*period,gate);
envelope2 = en.adsr(0.2*period,0.4*period,0.4,0.4*period,gate);
envelope3 = en.adsr(0.1*period,0.5*period,0.6,0.5*period,gate);
fullstring(f,n1,n2,g) = ((g^(n2+2-n1))*os.osc(f*n2) + os.osc(f*n1) - (g^(n2+1-n1))*os.osc(f*(n2+1)) - g*os.osc(f*(n1-1)))/(1+g^2-2*g*os.osccos(f));
octave1gain = vslider("[11]Octave 1", 5.6,0,10,0.1)*0.04;
octave2gain = vslider("[11]Octave 2", 7.8,0,10,0.1)*0.04;
octave3gain = vslider("[12]Octave 3", 5.6,0,10,0.1)*0.03;
octave4gain = vslider("[13]Octave 4", 1,0,10,0.1)*0.04;
octave5gain = vslider("[14]Octave 5", 0.4,0,10,0.1)*0.01;
octave6gain = vslider("[15]Octave 6", 0.2,0,10,0.1)*0.003;
string1(f,g) = octave6gain*fullstring(f,32,64,g) + octave5gain*fullstring(f,16,32,g) : *(envelope1);
string2(f,g) = octave4gain*fullstring(f,8,16,g) + octave3gain*fullstring(f,4,8,g) : *(envelope2);
string3(f,g) = octave1gain*(os.osc(f)+1.42*os.osc(2*f)) + octave2gain*fullstring(f,2,4,g) : *(envelope3);
string(f,g) = string1(f,g) + string2(f,g) + string3(f,g);
};
commonPitch = hslider("[0][style:radio{'B':14;'A#':13;'A':12;'G#':11;'G':10;'F#':9;'F':8;'E':7;'D#':6;'D':5;'C#':4;'C':3}]Pitch",3,3,14,1);
fineTune = hslider("Fine_Tune",0,-100,100,1);
_voice_1cpitch = 110*(2^(commonPitch/12))*(2^(fineTune/1200))*(2^(0));
period = vslider("[3]Period",7,4,10,0.5);
process = hgroup("Motif", PureTonesString(_voice_1cpitch,period,0,5,0)) <: dm.zita_light;
| https://raw.githubusercontent.com/aravind-sadharani/puretones-music-room/b237ebc23b7b11d7c018716278c2ed702d374e99/src/posts/tanpuraworking-1/tanpura-string.dsp | faust | import("stdfaust.lib");
PureTonesString(coarsefreq,period,finetune,variance,delay) = string(freq*(1+delta)/2,gamma) + string(freq*(1-delta)/2,gamma) : *(gain)
with {
freq = coarsefreq;
delta = vslider("[04]Variance",variance,0,20,0.1)/10000;
gamma = 0.5;
gate = vgroup("[00]Play String",os.lf_pulsetrainpos(1/period,0.3):@(ma.SR*delay*period));
gain = 10^((vslider("[08]Gain",0,-20,20,0.1)-6) : /(20));
envelope1 = en.adsr(0.1*period,0.3*period,0.2,0.3*period,gate);
envelope2 = en.adsr(0.2*period,0.4*period,0.4,0.4*period,gate);
envelope3 = en.adsr(0.1*period,0.5*period,0.6,0.5*period,gate);
fullstring(f,n1,n2,g) = ((g^(n2+2-n1))*os.osc(f*n2) + os.osc(f*n1) - (g^(n2+1-n1))*os.osc(f*(n2+1)) - g*os.osc(f*(n1-1)))/(1+g^2-2*g*os.osccos(f));
octave1gain = vslider("[11]Octave 1", 5.6,0,10,0.1)*0.04;
octave2gain = vslider("[11]Octave 2", 7.8,0,10,0.1)*0.04;
octave3gain = vslider("[12]Octave 3", 5.6,0,10,0.1)*0.03;
octave4gain = vslider("[13]Octave 4", 1,0,10,0.1)*0.04;
octave5gain = vslider("[14]Octave 5", 0.4,0,10,0.1)*0.01;
octave6gain = vslider("[15]Octave 6", 0.2,0,10,0.1)*0.003;
string1(f,g) = octave6gain*fullstring(f,32,64,g) + octave5gain*fullstring(f,16,32,g) : *(envelope1);
string2(f,g) = octave4gain*fullstring(f,8,16,g) + octave3gain*fullstring(f,4,8,g) : *(envelope2);
string3(f,g) = octave1gain*(os.osc(f)+1.42*os.osc(2*f)) + octave2gain*fullstring(f,2,4,g) : *(envelope3);
string(f,g) = string1(f,g) + string2(f,g) + string3(f,g);
};
commonPitch = hslider("[0][style:radio{'B':14;'A#':13;'A':12;'G#':11;'G':10;'F#':9;'F':8;'E':7;'D#':6;'D':5;'C#':4;'C':3}]Pitch",3,3,14,1);
fineTune = hslider("Fine_Tune",0,-100,100,1);
_voice_1cpitch = 110*(2^(commonPitch/12))*(2^(fineTune/1200))*(2^(0));
period = vslider("[3]Period",7,4,10,0.5);
process = hgroup("Motif", PureTonesString(_voice_1cpitch,period,0,5,0)) <: dm.zita_light;
|
|
7ef976725e6761c7483d17e7a7455600e8292e2e5a2a22a37abc769bf5d6b39f | aravind-sadharani/puretones-music-room | Two-close-strings.dsp | import("stdfaust.lib");
commonPitch = hslider("[0][style:radio{'B':14;'A#':13;'A':12;'G#':11;'G':10;'F#':9;'F':8;'E':7;'D#':6;'D':5;'C#':4;'C':3}]Pitch",3,3,14,1);
fineTune = hslider("Fine_Tune",0,-100,100,1);
String2Tone(f,r,g) = StringModel(pm.f2l(f*r),StringPluck) : *(StringEnv)
with {
StringPluck = en.adsr(0.00001,cperiod*0.7,0.9,cperiod*0.3,g);
StringEnv = en.adsr(0.0001,cperiod*0.7,0.9,cperiod*0.4,g);
StringModel(length,excitation) = 2*pm.endChain(egChain)
with{
brightness = 0.6/((length)^(1/3));
stiffness = 25*((length)^(1/3));
pluckPosition = 0.61;
StringBody(stringL,excitation) = reflectance,transmittance,_
with{
c = (0.375*(stringL^(1/4)) - 0.0825);
transmittance = _ <: *(1-c),1*c*fi.resonbp(pm.l2f(stringL),2,1) :> _;
reflectance = _;
};
StringBridge(brightness) = pm.rTermination(pm.basicBlock,reflectance) : _,transmittance,_
with{
reflectance = (-1)*pm.bridgeFilter(brightness,0);
transmittance = _;
};
openStringPick(length,stiffness,pluckPosition,excitation) = strChain
with{
dispersionFilters = par(i,2,si.smooth(stiffness)),_;
maxStringLength = 6;
nti = length*pluckPosition; // length of the upper portion of the string
itb = length*(1-pluckPosition); // length of the lower portion of the string
strChain = pm.chain(
pm.stringSegment(maxStringLength,nti) :
pm.in(excitation) :
dispersionFilters :
pm.stringSegment(maxStringLength,itb)
);
};
lengthTuning = 13*pm.speedOfSound/ma.SR;
stringL = length-lengthTuning;
egChain = pm.chain(
pm.lStringRigidTermination :
openStringPick(stringL,stiffness/1000,pluckPosition,excitation) :
StringBridge(brightness) :
StringBody(length,excitation) :
pm.out
);
};
};
freq = 110*(2^(commonPitch/12))*(2^(fineTune/1200));
delta = 0.25;
sharpness = 0.5;
cperiod = 12;
cgain = 0.4;
gate = os.lf_pulsetrainpos(1/cperiod,0.1);
g1 = gate + (gate : @(ma.SR*cperiod*2/3));
g2 = (gate : @(ma.SR*cperiod/3)) + (gate : @(ma.SR*cperiod*2/3));
env1 = en.adsr(0.01,cperiod/2,0.3,cperiod/8,g1);
env2 = en.adsr(0.01,cperiod/2,0.3,cperiod/8,g2);
string1 = String2Tone(freq-delta,1,g1) : *(env1) : *(cgain);
string2 = String2Tone(freq+delta,1,g2) : *(env2) : *(cgain);
mix(l,r) = 0.7*l+0.3*r,0.3*l+0.7*r;
process = hgroup("Motif",(string1, string2 : mix)) : @(ma.SR*0.01),@(ma.SR*0.01) : dm.zita_light;
| https://raw.githubusercontent.com/aravind-sadharani/puretones-music-room/b237ebc23b7b11d7c018716278c2ed702d374e99/src/posts/tanpuraworking-1/Two-close-strings.dsp | faust | length of the upper portion of the string
length of the lower portion of the string | import("stdfaust.lib");
commonPitch = hslider("[0][style:radio{'B':14;'A#':13;'A':12;'G#':11;'G':10;'F#':9;'F':8;'E':7;'D#':6;'D':5;'C#':4;'C':3}]Pitch",3,3,14,1);
fineTune = hslider("Fine_Tune",0,-100,100,1);
String2Tone(f,r,g) = StringModel(pm.f2l(f*r),StringPluck) : *(StringEnv)
with {
StringPluck = en.adsr(0.00001,cperiod*0.7,0.9,cperiod*0.3,g);
StringEnv = en.adsr(0.0001,cperiod*0.7,0.9,cperiod*0.4,g);
StringModel(length,excitation) = 2*pm.endChain(egChain)
with{
brightness = 0.6/((length)^(1/3));
stiffness = 25*((length)^(1/3));
pluckPosition = 0.61;
StringBody(stringL,excitation) = reflectance,transmittance,_
with{
c = (0.375*(stringL^(1/4)) - 0.0825);
transmittance = _ <: *(1-c),1*c*fi.resonbp(pm.l2f(stringL),2,1) :> _;
reflectance = _;
};
StringBridge(brightness) = pm.rTermination(pm.basicBlock,reflectance) : _,transmittance,_
with{
reflectance = (-1)*pm.bridgeFilter(brightness,0);
transmittance = _;
};
openStringPick(length,stiffness,pluckPosition,excitation) = strChain
with{
dispersionFilters = par(i,2,si.smooth(stiffness)),_;
maxStringLength = 6;
strChain = pm.chain(
pm.stringSegment(maxStringLength,nti) :
pm.in(excitation) :
dispersionFilters :
pm.stringSegment(maxStringLength,itb)
);
};
lengthTuning = 13*pm.speedOfSound/ma.SR;
stringL = length-lengthTuning;
egChain = pm.chain(
pm.lStringRigidTermination :
openStringPick(stringL,stiffness/1000,pluckPosition,excitation) :
StringBridge(brightness) :
StringBody(length,excitation) :
pm.out
);
};
};
freq = 110*(2^(commonPitch/12))*(2^(fineTune/1200));
delta = 0.25;
sharpness = 0.5;
cperiod = 12;
cgain = 0.4;
gate = os.lf_pulsetrainpos(1/cperiod,0.1);
g1 = gate + (gate : @(ma.SR*cperiod*2/3));
g2 = (gate : @(ma.SR*cperiod/3)) + (gate : @(ma.SR*cperiod*2/3));
env1 = en.adsr(0.01,cperiod/2,0.3,cperiod/8,g1);
env2 = en.adsr(0.01,cperiod/2,0.3,cperiod/8,g2);
string1 = String2Tone(freq-delta,1,g1) : *(env1) : *(cgain);
string2 = String2Tone(freq+delta,1,g2) : *(env2) : *(cgain);
mix(l,r) = 0.7*l+0.3*r,0.3*l+0.7*r;
process = hgroup("Motif",(string1, string2 : mix)) : @(ma.SR*0.01),@(ma.SR*0.01) : dm.zita_light;
|
20454a1c81c1343650ef67d27dab802ee47bb563188b7ae8c986fb499ca94d81 | aravind-sadharani/puretones | puretones.dsp | // _____ _ _ _ __ __ _ __ __ _
// / ____| | | | (_) | \/ | (_) \ \ / / | |
// | (___ __ _ __| | |__ __ _ _ __ __ _ _ __ _ | \ / |_ _ ___ _ ___ \ \ /\ / /__ _ __| | _____
// \___ \ / _` |/ _` | '_ \ / _` | '__/ _` | '_ \| | | |\/| | | | / __| |/ __| \ \/ \/ / _ \| '__| |/ / __|
// ____) | (_| | (_| | | | | (_| | | | (_| | | | | | | | | | |_| \__ \ | (__ \ /\ / (_) | | | <\__ \
// |_____/ \__,_|\__,_|_| |_|\__,_|_| \__,_|_| |_|_| |_| |_|\__,_|___/_|\___| \/ \/ \___/|_| |_|\_\___/
//
// PureTones Drone Six - Developed by Aravind Iyer and S Balachander, Sadharani Music Works
// A Six string version which offers a few more advanced features than puretones-drone.dsp which has four strings
//
import("stdfaust.lib");
PureTonesString(coarsefreq,period,finetune,ratio,variance,delay) = string(freq*(1+delta)/2,gamma) + string(freq*(1-delta)/2,gamma) : *(gain)
with {
finecent = vslider("[02]Fine Tune",finetune,-100,100,1);
fineratio = 2^(finecent/1200);
ultrafinecent = vslider("[03]Ultrafine Tune",0,-100,100,1);
ultrafineratio = 2^(ultrafinecent/120000);
ratioselector = vslider("[001][style:radio{'SA':0;'Ni^':1;'Ni_':2;'Dha^':3;'Dha_':4;'Pa':5;'Ma^':6;'Ma_':7;'Ga^':8;'Ga_':9;'Re^':10;'Re_':11;'Sa':12}]Select Note",ratio,0,12,1);
ratioselected = 2,243/128,16/9,27/16,128/81,3/2,729/512,4/3,81/64,32/27,9/8,256/243,1 : ba.selectn(13,ratioselector);
freq = ratioselected*coarsefreq*fineratio*ultrafineratio;
delta = ratioselected*vslider("[04]Variance",variance,0,20,0.1)/10000;
gamma = 0.5;
gate = vgroup("[00]Play String",checkbox("[1]Loop")*os.lf_pulsetrainpos(1/period,0.3):@(ma.SR*delay*period) + button("[0]Once"));
gain = 10^((vslider("[08]Gain",0,-20,20,0.1)-18) : /(20))/ratioselected;
envelope1 = en.adsr(0.1*period,0.3*period,0.2,0.3*period,gate);
envelope2 = en.adsr(0.2*period,0.4*period,0.4,0.4*period,gate);
envelope3 = en.adsr(0.1*period,0.5*period,0.6,0.5*period,gate);
fullstring(f,n1,n2,g) = ((g^(n2+2-n1))*os.osc(f*n2) + os.osc(f*n1) - (g^(n2+1-n1))*os.osc(f*(n2+1)) - g*os.osc(f*(n1-1)))/(1+g^2-2*g*os.osccos(f));
octave1gain = vslider("[11]Octave 1", 5.6,0,10,0.1)*0.04;
octave2gain = vslider("[11]Octave 2", 7.8,0,10,0.1)*0.04;
octave3gain = vslider("[12]Octave 3", 5.6,0,10,0.1)*0.03;
octave4gain = vslider("[13]Octave 4", 1,0,10,0.1)*0.04;
octave5gain = vslider("[14]Octave 5", 0.4,0,10,0.1)*0.01;
octave6gain = vslider("[15]Octave 6", 0.2,0,10,0.1)*0.003;
string1(f,g) = octave6gain*fullstring(f,32,64,g) + octave5gain*fullstring(f,16,32,g) : *(envelope1);
string2(f,g) = octave4gain*fullstring(f,8,16,g) + octave3gain*fullstring(f,4,8,g) : *(envelope2);
string3(f,g) = octave1gain*(os.osc(f)+1.42*os.osc(2*f)) + octave2gain*fullstring(f,2,4,g) : *(envelope3);
string(f,g) = string1(f,g) + string2(f,g) + string3(f,g);
};
PureTones(c,p) = hgroup("[0]",(c,p)) : PureTonesSystem
with {
string1(c,p) = hgroup("[1]1st String",PureTonesString(c,p-0.2,0,5,5,0));
string2(c,p) = hgroup("[2]2nd String",PureTonesString(c,p-0.2,0,0,5,0.3));
string3(c,p) = hgroup("[3]3rd String",PureTonesString(c,p-0.2,0,12,5,0.6));
string4(c,p) = hgroup("[4]4th String",PureTonesString(c,p+0.2,0,5,5,0.5));
string5(c,p) = hgroup("[5]5th String",PureTonesString(c,p+0.2,0,0,5,0.8));
string6(c,p) = hgroup("[6]6th String",PureTonesString(c,p+0.2,0,12,5,0.1));
PureTonesLeft(c,p) = (c,p) <: _,_,_,_,_,_ : tgroup("[1]",string1,string2,string3) :> _;
PureTonesRight(c,p) = (c,p) <: _,_,_,_,_,_ : tgroup("[1]",string4,string5,string6) :> _;
PureTonesSystem(c,p) = PureTonesLeft(c,p), PureTonesRight(c,p);
};
coarseselector = vslider("[0][style:radio{'B':14;'A#':13;'A':12;'G#':11;'G':10;'F#':9;'F':8;'E':7;'D#':6;'D':5;'C#':4;'C':3}]Common Frequency",11,3,14,1);
coarse = 110*(2^(coarseselector/12));
octaveselector = vslider("[1][style:radio{'High':1;'Medium':0;'Low':-1}]Octave Selector",0,-1,1,1);
finecent = vslider("[2]Fine Tune",0,-100,100,1);
fineratio = 2^(finecent/1200);
period = vslider("[3]Period",7,4,10,0.5);
process = hgroup("[00]PureTones v1.0", PureTones(coarse*fineratio*(2^octaveselector),period)) <: dm.zita_light;
| https://raw.githubusercontent.com/aravind-sadharani/puretones/e605d7e8ff393cae6fd5216888528290c0f71c33/dronewebapp/puretones.dsp | faust | _____ _ _ _ __ __ _ __ __ _
/ ____| | | | (_) | \/ | (_) \ \ / / | |
| (___ __ _ __| | |__ __ _ _ __ __ _ _ __ _ | \ / |_ _ ___ _ ___ \ \ /\ / /__ _ __| | _____
\___ \ / _` |/ _` | '_ \ / _` | '__/ _` | '_ \| | | |\/| | | | / __| |/ __| \ \/ \/ / _ \| '__| |/ / __|
____) | (_| | (_| | | | | (_| | | | (_| | | | | | | | | | |_| \__ \ | (__ \ /\ / (_) | | | <\__ \
|_____/ \__,_|\__,_|_| |_|\__,_|_| \__,_|_| |_|_| |_| |_|\__,_|___/_|\___| \/ \/ \___/|_| |_|\_\___/
PureTones Drone Six - Developed by Aravind Iyer and S Balachander, Sadharani Music Works
A Six string version which offers a few more advanced features than puretones-drone.dsp which has four strings
|
import("stdfaust.lib");
PureTonesString(coarsefreq,period,finetune,ratio,variance,delay) = string(freq*(1+delta)/2,gamma) + string(freq*(1-delta)/2,gamma) : *(gain)
with {
finecent = vslider("[02]Fine Tune",finetune,-100,100,1);
fineratio = 2^(finecent/1200);
ultrafinecent = vslider("[03]Ultrafine Tune",0,-100,100,1);
ultrafineratio = 2^(ultrafinecent/120000);
ratioselector = vslider("[001][style:radio{'SA':0;'Ni^':1;'Ni_':2;'Dha^':3;'Dha_':4;'Pa':5;'Ma^':6;'Ma_':7;'Ga^':8;'Ga_':9;'Re^':10;'Re_':11;'Sa':12}]Select Note",ratio,0,12,1);
ratioselected = 2,243/128,16/9,27/16,128/81,3/2,729/512,4/3,81/64,32/27,9/8,256/243,1 : ba.selectn(13,ratioselector);
freq = ratioselected*coarsefreq*fineratio*ultrafineratio;
delta = ratioselected*vslider("[04]Variance",variance,0,20,0.1)/10000;
gamma = 0.5;
gate = vgroup("[00]Play String",checkbox("[1]Loop")*os.lf_pulsetrainpos(1/period,0.3):@(ma.SR*delay*period) + button("[0]Once"));
gain = 10^((vslider("[08]Gain",0,-20,20,0.1)-18) : /(20))/ratioselected;
envelope1 = en.adsr(0.1*period,0.3*period,0.2,0.3*period,gate);
envelope2 = en.adsr(0.2*period,0.4*period,0.4,0.4*period,gate);
envelope3 = en.adsr(0.1*period,0.5*period,0.6,0.5*period,gate);
fullstring(f,n1,n2,g) = ((g^(n2+2-n1))*os.osc(f*n2) + os.osc(f*n1) - (g^(n2+1-n1))*os.osc(f*(n2+1)) - g*os.osc(f*(n1-1)))/(1+g^2-2*g*os.osccos(f));
octave1gain = vslider("[11]Octave 1", 5.6,0,10,0.1)*0.04;
octave2gain = vslider("[11]Octave 2", 7.8,0,10,0.1)*0.04;
octave3gain = vslider("[12]Octave 3", 5.6,0,10,0.1)*0.03;
octave4gain = vslider("[13]Octave 4", 1,0,10,0.1)*0.04;
octave5gain = vslider("[14]Octave 5", 0.4,0,10,0.1)*0.01;
octave6gain = vslider("[15]Octave 6", 0.2,0,10,0.1)*0.003;
string1(f,g) = octave6gain*fullstring(f,32,64,g) + octave5gain*fullstring(f,16,32,g) : *(envelope1);
string2(f,g) = octave4gain*fullstring(f,8,16,g) + octave3gain*fullstring(f,4,8,g) : *(envelope2);
string3(f,g) = octave1gain*(os.osc(f)+1.42*os.osc(2*f)) + octave2gain*fullstring(f,2,4,g) : *(envelope3);
string(f,g) = string1(f,g) + string2(f,g) + string3(f,g);
};
PureTones(c,p) = hgroup("[0]",(c,p)) : PureTonesSystem
with {
string1(c,p) = hgroup("[1]1st String",PureTonesString(c,p-0.2,0,5,5,0));
string2(c,p) = hgroup("[2]2nd String",PureTonesString(c,p-0.2,0,0,5,0.3));
string3(c,p) = hgroup("[3]3rd String",PureTonesString(c,p-0.2,0,12,5,0.6));
string4(c,p) = hgroup("[4]4th String",PureTonesString(c,p+0.2,0,5,5,0.5));
string5(c,p) = hgroup("[5]5th String",PureTonesString(c,p+0.2,0,0,5,0.8));
string6(c,p) = hgroup("[6]6th String",PureTonesString(c,p+0.2,0,12,5,0.1));
PureTonesLeft(c,p) = (c,p) <: _,_,_,_,_,_ : tgroup("[1]",string1,string2,string3) :> _;
PureTonesRight(c,p) = (c,p) <: _,_,_,_,_,_ : tgroup("[1]",string4,string5,string6) :> _;
PureTonesSystem(c,p) = PureTonesLeft(c,p), PureTonesRight(c,p);
};
coarseselector = vslider("[0][style:radio{'B':14;'A#':13;'A':12;'G#':11;'G':10;'F#':9;'F':8;'E':7;'D#':6;'D':5;'C#':4;'C':3}]Common Frequency",11,3,14,1);
coarse = 110*(2^(coarseselector/12));
octaveselector = vslider("[1][style:radio{'High':1;'Medium':0;'Low':-1}]Octave Selector",0,-1,1,1);
finecent = vslider("[2]Fine Tune",0,-100,100,1);
fineratio = 2^(finecent/1200);
period = vslider("[3]Period",7,4,10,0.5);
process = hgroup("[00]PureTones v1.0", PureTones(coarse*fineratio*(2^octaveselector),period)) <: dm.zita_light;
|
cd0e031fb807af2b6a74ab7623baffe3035a294e9295ad065ab8d1a8f36fce4e | aravind-sadharani/puretones | musicscale.dsp | // _____ _ _ _ __ __ _ __ __ _
// / ____| | | | (_) | \/ | (_) \ \ / / | |
// | (___ __ _ __| | |__ __ _ _ __ __ _ _ __ _ | \ / |_ _ ___ _ ___ \ \ /\ / /__ _ __| | _____
// \___ \ / _` |/ _` | '_ \ / _` | '__/ _` | '_ \| | | |\/| | | | / __| |/ __| \ \/ \/ / _ \| '__| |/ / __|
// ____) | (_| | (_| | | | | (_| | | | (_| | | | | | | | | | |_| \__ \ | (__ \ /\ / (_) | | | <\__ \
// |_____/ \__,_|\__,_|_| |_|\__,_|_| \__,_|_| |_|_| |_| |_|\__,_|___/_|\___| \/ \/ \___/|_| |_|\_\___/
//
// PureTones Keys - Developed by Aravind Iyer and S Balachander, Sadharani Music Works
// A tunable keyboard to be used with puretones-drone.dsp or puretones-drone-six.dsp
//
import("stdfaust.lib");
coarseselector = vslider("[000][style:radio{'B':14;'A#':13;'A':12;'G#':11;'G':10;'F#':9;'F':8;'E':7;'D#':6;'D':5;'C#':4;'C':3}]Pitch",11,3,14,1);
coarse = 110*(2^(coarseselector/12));
octaveselector = vslider("[001][style:radio{'High':1;'Medium':0;'Low':-1;'Lowest':-2}]Octave",0,-2,1,1);
finecent = vslider("[002]Fine Tune",0,-100,100,1);
fineratio = 2^(finecent/1200);
commonfreq = coarse*fineratio*(2^octaveselector);
period = 2^(vslider("[003]Period",2,0,3,0.1));
note(freq,ratio,period) = string(freq,tunedratio)*rolloffenv
with {
n = 32;
g = 0.6;
amplitude = 0.4*(1-g)/(1-g^(n+1));
ToneStringModel(f) = ((g^(n+1))*os.osc(f*n) - (g^n)*os.osc(f*(n+1)) + os.osc(f))/((1-g)^2+4*g*os.osc(f/2)*os.osc(f/2)) : *(amplitude);
variance = vslider("[000]Variance",2,0,10,0.1)/10000;
string(f,r) = ToneStringModel(f*r*(1+variance)) + ToneStringModel(f*r*(1-variance));
c = checkbox("Play") : si.smoo;
b = button("Pluck") ;
cent = vslider("[00]Cent", 0,-100,100,1);
ratio10 = 2^(cent/1200);
pointOonecent = vslider("[01]0.01 Cent", 0,-100,100,1);
ratio00 = 2^(pointOonecent/120000);
shakeselector = checkbox("[02]Gamaka");
delta1cent = vslider("[03]Starting Cent", 0,-220,220,1);
delta1pointOonecent = vslider("[04]Starting 0.01 Cent", 0,-100,100,1);
delta1 = (2^(delta1cent/1200))*(2^(delta1pointOonecent/120000))-1;
delta2cent = vslider("[05]Ending Cent", 0,-220,220,1);
delta2pointOonecent = vslider("[06]Ending 0.01 Cent", 0,-100,100,1);
delta2 = (2^(delta2cent/1200))*(2^(delta2pointOonecent/120000))-1;
rate = 2^(vslider("[07]Rate",17,-5,25,0.1)/10);
number = vslider("[08]Number",3.5,0,10,0.01);
phasor(f) = (+(f/ma.SR) ~ ma.decimal);
cphasedcos(x) = phasor(x) - (phasor(x) : ba.latch(gate(period))) : *(2*ma.PI) : cos;
bphasedcos(x) = phasor(x) - (phasor(x) : ba.latch(b)) : *(2*ma.PI) : cos;
ramp(x) = +(x/ma.SR) ~ _;
clockedramp(x) = ramp(x) - (ramp(x) : ba.latch(gate(period)));
blockedramp(x) = ramp(x) - (ramp(x) : ba.latch(b));
shake(d1,d2,r,n,p) = shakeselector*c*(1+((d1+d2)/2+(d1-d2)*cphasedcos(r)/2)*(clockedramp(r) < n)) + shakeselector*(1-c)*(1+((d1+d2)/2+(d1-d2)*bphasedcos(r)/2)*(blockedramp(r) < n))+ (1-shakeselector);
gate(p) = os.lf_pulsetrainpos(1/p,0.3);
env(p) = 4*en.adsr(0.0001,p-1,0.5,p-1,gate(p));
benv = 3.5*en.adsr(0.0001,1,0.8,1,b);
pluck = (period) : env;
tunedratio = ratio*ratio10*ratio00*(delta1,delta2,rate,number,period : shake);
rolloffenv = en.adsr(0.001,period*0.6,0.8,period*0.5,c*gate(period)) + en.adsr(0.0001,0.2,0.8,0.1,b);
};
scale(c,e) = tgroup("[05]12 Note Scale", string01+string02+string03+string04+string05+string06+string07+string08+string09+string10+string11+string12+string13)
with {
string01 = hgroup("[01] Sa ", (c,1,e : note));
string02 = hgroup("[02] re ", (c,256/243,e : note));
string03 = hgroup("[03] Re ", (c,9/8,e : note));
string04 = hgroup("[04] ga ", (c,32/27,e : note));
string05 = hgroup("[05] Ga ", (c,81/64,e : note));
string06 = hgroup("[06] ma ", (c,4/3,e : note));
string07 = hgroup("[07] Ma ", (c,729/512,e : note));
string08 = hgroup("[08] Pa ", (c,3/2,e : note));
string09 = hgroup("[09] dha ", (c,128/81,e : note));
string10 = hgroup("[10] Dha ", (c,27/16,e : note));
string11 = hgroup("[11] ni ", (c,16/9,e : note));
string12 = hgroup("[12] Ni ", (c,243/128,e : note));
string13 = hgroup("[13] SA ", (c,2,e : note));
};
process = hgroup("[0000]Common Parameters",(commonfreq,period) : scale) <: dm.zita_light;
| https://raw.githubusercontent.com/aravind-sadharani/puretones/9799d01cd664fcb3b383c6559ffdf9ece8dbfba9/scalewebapp/musicscale.dsp | faust | _____ _ _ _ __ __ _ __ __ _
/ ____| | | | (_) | \/ | (_) \ \ / / | |
| (___ __ _ __| | |__ __ _ _ __ __ _ _ __ _ | \ / |_ _ ___ _ ___ \ \ /\ / /__ _ __| | _____
\___ \ / _` |/ _` | '_ \ / _` | '__/ _` | '_ \| | | |\/| | | | / __| |/ __| \ \/ \/ / _ \| '__| |/ / __|
____) | (_| | (_| | | | | (_| | | | (_| | | | | | | | | | |_| \__ \ | (__ \ /\ / (_) | | | <\__ \
|_____/ \__,_|\__,_|_| |_|\__,_|_| \__,_|_| |_|_| |_| |_|\__,_|___/_|\___| \/ \/ \___/|_| |_|\_\___/
PureTones Keys - Developed by Aravind Iyer and S Balachander, Sadharani Music Works
A tunable keyboard to be used with puretones-drone.dsp or puretones-drone-six.dsp
|
import("stdfaust.lib");
coarseselector = vslider("[000][style:radio{'B':14;'A#':13;'A':12;'G#':11;'G':10;'F#':9;'F':8;'E':7;'D#':6;'D':5;'C#':4;'C':3}]Pitch",11,3,14,1);
coarse = 110*(2^(coarseselector/12));
octaveselector = vslider("[001][style:radio{'High':1;'Medium':0;'Low':-1;'Lowest':-2}]Octave",0,-2,1,1);
finecent = vslider("[002]Fine Tune",0,-100,100,1);
fineratio = 2^(finecent/1200);
commonfreq = coarse*fineratio*(2^octaveselector);
period = 2^(vslider("[003]Period",2,0,3,0.1));
note(freq,ratio,period) = string(freq,tunedratio)*rolloffenv
with {
n = 32;
g = 0.6;
amplitude = 0.4*(1-g)/(1-g^(n+1));
ToneStringModel(f) = ((g^(n+1))*os.osc(f*n) - (g^n)*os.osc(f*(n+1)) + os.osc(f))/((1-g)^2+4*g*os.osc(f/2)*os.osc(f/2)) : *(amplitude);
variance = vslider("[000]Variance",2,0,10,0.1)/10000;
string(f,r) = ToneStringModel(f*r*(1+variance)) + ToneStringModel(f*r*(1-variance));
c = checkbox("Play") : si.smoo;
b = button("Pluck") ;
cent = vslider("[00]Cent", 0,-100,100,1);
ratio10 = 2^(cent/1200);
pointOonecent = vslider("[01]0.01 Cent", 0,-100,100,1);
ratio00 = 2^(pointOonecent/120000);
shakeselector = checkbox("[02]Gamaka");
delta1cent = vslider("[03]Starting Cent", 0,-220,220,1);
delta1pointOonecent = vslider("[04]Starting 0.01 Cent", 0,-100,100,1);
delta1 = (2^(delta1cent/1200))*(2^(delta1pointOonecent/120000))-1;
delta2cent = vslider("[05]Ending Cent", 0,-220,220,1);
delta2pointOonecent = vslider("[06]Ending 0.01 Cent", 0,-100,100,1);
delta2 = (2^(delta2cent/1200))*(2^(delta2pointOonecent/120000))-1;
rate = 2^(vslider("[07]Rate",17,-5,25,0.1)/10);
number = vslider("[08]Number",3.5,0,10,0.01);
phasor(f) = (+(f/ma.SR) ~ ma.decimal);
cphasedcos(x) = phasor(x) - (phasor(x) : ba.latch(gate(period))) : *(2*ma.PI) : cos;
bphasedcos(x) = phasor(x) - (phasor(x) : ba.latch(b)) : *(2*ma.PI) : cos;
ramp(x) = +(x/ma.SR) ~ _;
clockedramp(x) = ramp(x) - (ramp(x) : ba.latch(gate(period)));
blockedramp(x) = ramp(x) - (ramp(x) : ba.latch(b));
shake(d1,d2,r,n,p) = shakeselector*c*(1+((d1+d2)/2+(d1-d2)*cphasedcos(r)/2)*(clockedramp(r) < n)) + shakeselector*(1-c)*(1+((d1+d2)/2+(d1-d2)*bphasedcos(r)/2)*(blockedramp(r) < n))+ (1-shakeselector);
gate(p) = os.lf_pulsetrainpos(1/p,0.3);
env(p) = 4*en.adsr(0.0001,p-1,0.5,p-1,gate(p));
benv = 3.5*en.adsr(0.0001,1,0.8,1,b);
pluck = (period) : env;
tunedratio = ratio*ratio10*ratio00*(delta1,delta2,rate,number,period : shake);
rolloffenv = en.adsr(0.001,period*0.6,0.8,period*0.5,c*gate(period)) + en.adsr(0.0001,0.2,0.8,0.1,b);
};
scale(c,e) = tgroup("[05]12 Note Scale", string01+string02+string03+string04+string05+string06+string07+string08+string09+string10+string11+string12+string13)
with {
string01 = hgroup("[01] Sa ", (c,1,e : note));
string02 = hgroup("[02] re ", (c,256/243,e : note));
string03 = hgroup("[03] Re ", (c,9/8,e : note));
string04 = hgroup("[04] ga ", (c,32/27,e : note));
string05 = hgroup("[05] Ga ", (c,81/64,e : note));
string06 = hgroup("[06] ma ", (c,4/3,e : note));
string07 = hgroup("[07] Ma ", (c,729/512,e : note));
string08 = hgroup("[08] Pa ", (c,3/2,e : note));
string09 = hgroup("[09] dha ", (c,128/81,e : note));
string10 = hgroup("[10] Dha ", (c,27/16,e : note));
string11 = hgroup("[11] ni ", (c,16/9,e : note));
string12 = hgroup("[12] Ni ", (c,243/128,e : note));
string13 = hgroup("[13] SA ", (c,2,e : note));
};
process = hgroup("[0000]Common Parameters",(commonfreq,period) : scale) <: dm.zita_light;
|
576f2a0dd038f3f36526feb23a9b15f6c09fd1efecd9651cb4d825fce4768775 | aravind-sadharani/puretones-music-room | 16harmonics.dsp | import("stdfaust.lib");
commonPitch = hslider("[0][style:radio{'B':14;'A#':13;'A':12;'G#':11;'G':10;'F#':9;'F':8;'E':7;'D#':6;'D':5;'C#':4;'C':3}]Pitch",3,3,14,1);
fineTune = hslider("Fine_Tune",0,-100,100,1);
cperiod = 2^(vslider("[01]Motif Tempo",1.0,-2,4,0.1) - 3);
cgain = 10^(vslider("[02]Motif Gain",0,-20,20,0.1) - 6 : /(20)) : /(sqrt(_voice_1noteratio));
delta = vslider("[04]Shake Variance", 10,0,120,1);
rate = vslider("[05]Shake Rate",11.5,10,25,0.1);
c2v(d) = 2^(d/1200)-1;
l2l(r) = 2^(r/10);
number = vslider("[06]Shake Number",3.4,1,10,0.1);
phasor(f) = ba.period(ma.SR/f) : *(f/ma.SR);
ramp(x) = ba.time : *(x);
fullstring(f,n1,n2,g) = ((g^(n2+2-n1))*os.osc(f*n2) + os.osc(f*n1) - (g^(n2+1-n1))*os.osc(f*(n2+1)) - g*os.osc(f*(n1-1)))/(1+g^2-2*g*os.osccos(f));
envelope(trigger) = en.adsr(0.001,cperiod/3,0.4,cperiod/8,trigger);
_voice_1noteindex = cperiod : _voice_1motifnotes;
_voice_1cpitch = 110*(2^(commonPitch/12))*(2^(fineTune/1200))*(2^(-1));
_voice_1ratio_0 = 1;//(1) * (1/2) * (2^(0/1200)); //Sa'
_voice_1ratio_1 = 2;//(1) * (1) * (2^(0/1200)); //Sa
_voice_1ratio_2 = 3;//(3/2) * (1) * (2^(0/1200)); //Pa
_voice_1ratio_3 = 4;//(1) * (2) * (2^(0/1200)); //Sa"
_voice_1ratio_4 = 5;//(81/64) * (2) * (2^(0/1200)); //Ga"
_voice_1ratio_5 = 6;//(3/2) * (2) * (2^(0/1200)); //Pa"
_voice_1ratio_6 = 7;//(16/9) * (2) * (2^(0/1200)); //ni"
_voice_1ratio_7 = 8;//(2) * (2) * (2^(0/1200)); //SA"
_voice_1ratio_8 = 9;//(9/8) * (2) * (2^(0/1200)); //Re"
_voice_1ratio_9 = 10;//(81/64) * (1) * (2^(0/1200)); //Ga
_voice_1ratio_10 = 11;//(4/3) * (2) * (2^(0/1200)); //ma"
_voice_1ratio_11 = 12;//(3/2) * (1/2) * (2^(0/1200)); //Pa'
_voice_1ratio_12 = 13;//(128/81) * (2) * (2^(0/1200)); //dha"
_voice_1ratio_13 = 14;//(16/9) * (1) * (2^(0/1200)); //ni
_voice_1ratio_14 = 15;//(243/128) * (2) * (2^(0/1200)); //Ni"
_voice_1ratio_15 = 16;//(2) * (1) * (2^(0/1200)); //SA
_voice_1noteratio = _voice_1ratio_0,_voice_1ratio_1,_voice_1ratio_2,_voice_1ratio_3,_voice_1ratio_4,_voice_1ratio_5,_voice_1ratio_6,_voice_1ratio_7,_voice_1ratio_8,_voice_1ratio_9,_voice_1ratio_10,_voice_1ratio_11,_voice_1ratio_12,_voice_1ratio_13,_voice_1ratio_14,_voice_1ratio_15 : ba.selectn(16,_voice_1noteindex);
_voice_1gatewaveform = waveform{1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0};
_voice_1gate(p) = _voice_1gatewaveform,int(2*ba.period(256*p*ma.SR)/(p*ma.SR)) : rdtable;
_voice_1motif = waveform{0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,3,3,3,3,3,3,3,3,3,3,3,3,3,3,3,3,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,9,9,9,9,9,9,9,9,9,9,9,9,9,9,9,9,10,10,10,10,10,10,10,10,10,10,10,10,10,10,10,10,11,11,11,11,11,11,11,11,11,11,11,11,11,11,11,11,12,12,12,12,12,12,12,12,12,12,12,12,12,12,12,12,13,13,13,13,13,13,13,13,13,13,13,13,13,13,13,13,14,14,14,14,14,14,14,14,14,14,14,14,14,14,14,14,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,14,14,14,14,14,14,14,14,14,14,14,14,14,14,14,14,13,13,13,13,13,13,13,13,13,13,13,13,13,13,13,13,12,12,12,12,12,12,12,12,12,12,12,12,12,12,12,12,11,11,11,11,11,11,11,11,11,11,11,11,11,11,11,11,10,10,10,10,10,10,10,10,10,10,10,10,10,10,10,10,9,9,9,9,9,9,9,9,9,9,9,9,9,9,9,9,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,3,3,3,3,3,3,3,3,3,3,3,3,3,3,3,3,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0};
_voice_1motifnotes(p) = _voice_1motif,int(2*ba.period(256*p*ma.SR)/(p*ma.SR)) : rdtable;
_voice_1notes = fullstring(_voice_1cpitch,_voice_1noteratio,_voice_1noteratio+1,0.5) : *(envelope(_voice_1gate(cperiod))) : @(ma.SR*0.1);
_voice_2notes = 0;
_voice_3notes = 0;
mix(a,b) = 0.7*a+0.3*b,0.3*a+0.7*b;
concert = hgroup("[00]Motif",2*cgain*(0.7*_voice_1notes + 0.9*_voice_2notes),2*cgain*(0.7*_voice_1notes + 0.9*_voice_3notes));
process = concert : mix : dm.zita_light;
| https://raw.githubusercontent.com/aravind-sadharani/puretones-music-room/b237ebc23b7b11d7c018716278c2ed702d374e99/src/posts/tanpuraworking-1/16harmonics.dsp | faust | (1) * (1/2) * (2^(0/1200)); //Sa'
(1) * (1) * (2^(0/1200)); //Sa
(3/2) * (1) * (2^(0/1200)); //Pa
(1) * (2) * (2^(0/1200)); //Sa"
(81/64) * (2) * (2^(0/1200)); //Ga"
(3/2) * (2) * (2^(0/1200)); //Pa"
(16/9) * (2) * (2^(0/1200)); //ni"
(2) * (2) * (2^(0/1200)); //SA"
(9/8) * (2) * (2^(0/1200)); //Re"
(81/64) * (1) * (2^(0/1200)); //Ga
(4/3) * (2) * (2^(0/1200)); //ma"
(3/2) * (1/2) * (2^(0/1200)); //Pa'
(128/81) * (2) * (2^(0/1200)); //dha"
(16/9) * (1) * (2^(0/1200)); //ni
(243/128) * (2) * (2^(0/1200)); //Ni"
(2) * (1) * (2^(0/1200)); //SA | import("stdfaust.lib");
commonPitch = hslider("[0][style:radio{'B':14;'A#':13;'A':12;'G#':11;'G':10;'F#':9;'F':8;'E':7;'D#':6;'D':5;'C#':4;'C':3}]Pitch",3,3,14,1);
fineTune = hslider("Fine_Tune",0,-100,100,1);
cperiod = 2^(vslider("[01]Motif Tempo",1.0,-2,4,0.1) - 3);
cgain = 10^(vslider("[02]Motif Gain",0,-20,20,0.1) - 6 : /(20)) : /(sqrt(_voice_1noteratio));
delta = vslider("[04]Shake Variance", 10,0,120,1);
rate = vslider("[05]Shake Rate",11.5,10,25,0.1);
c2v(d) = 2^(d/1200)-1;
l2l(r) = 2^(r/10);
number = vslider("[06]Shake Number",3.4,1,10,0.1);
phasor(f) = ba.period(ma.SR/f) : *(f/ma.SR);
ramp(x) = ba.time : *(x);
fullstring(f,n1,n2,g) = ((g^(n2+2-n1))*os.osc(f*n2) + os.osc(f*n1) - (g^(n2+1-n1))*os.osc(f*(n2+1)) - g*os.osc(f*(n1-1)))/(1+g^2-2*g*os.osccos(f));
envelope(trigger) = en.adsr(0.001,cperiod/3,0.4,cperiod/8,trigger);
_voice_1noteindex = cperiod : _voice_1motifnotes;
_voice_1cpitch = 110*(2^(commonPitch/12))*(2^(fineTune/1200))*(2^(-1));
_voice_1noteratio = _voice_1ratio_0,_voice_1ratio_1,_voice_1ratio_2,_voice_1ratio_3,_voice_1ratio_4,_voice_1ratio_5,_voice_1ratio_6,_voice_1ratio_7,_voice_1ratio_8,_voice_1ratio_9,_voice_1ratio_10,_voice_1ratio_11,_voice_1ratio_12,_voice_1ratio_13,_voice_1ratio_14,_voice_1ratio_15 : ba.selectn(16,_voice_1noteindex);
_voice_1gatewaveform = waveform{1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0};
_voice_1gate(p) = _voice_1gatewaveform,int(2*ba.period(256*p*ma.SR)/(p*ma.SR)) : rdtable;
_voice_1motif = waveform{0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,3,3,3,3,3,3,3,3,3,3,3,3,3,3,3,3,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,9,9,9,9,9,9,9,9,9,9,9,9,9,9,9,9,10,10,10,10,10,10,10,10,10,10,10,10,10,10,10,10,11,11,11,11,11,11,11,11,11,11,11,11,11,11,11,11,12,12,12,12,12,12,12,12,12,12,12,12,12,12,12,12,13,13,13,13,13,13,13,13,13,13,13,13,13,13,13,13,14,14,14,14,14,14,14,14,14,14,14,14,14,14,14,14,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,14,14,14,14,14,14,14,14,14,14,14,14,14,14,14,14,13,13,13,13,13,13,13,13,13,13,13,13,13,13,13,13,12,12,12,12,12,12,12,12,12,12,12,12,12,12,12,12,11,11,11,11,11,11,11,11,11,11,11,11,11,11,11,11,10,10,10,10,10,10,10,10,10,10,10,10,10,10,10,10,9,9,9,9,9,9,9,9,9,9,9,9,9,9,9,9,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,3,3,3,3,3,3,3,3,3,3,3,3,3,3,3,3,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0};
_voice_1motifnotes(p) = _voice_1motif,int(2*ba.period(256*p*ma.SR)/(p*ma.SR)) : rdtable;
_voice_1notes = fullstring(_voice_1cpitch,_voice_1noteratio,_voice_1noteratio+1,0.5) : *(envelope(_voice_1gate(cperiod))) : @(ma.SR*0.1);
_voice_2notes = 0;
_voice_3notes = 0;
mix(a,b) = 0.7*a+0.3*b,0.3*a+0.7*b;
concert = hgroup("[00]Motif",2*cgain*(0.7*_voice_1notes + 0.9*_voice_2notes),2*cgain*(0.7*_voice_1notes + 0.9*_voice_3notes));
process = concert : mix : dm.zita_light;
|
875a75e92b058e24d5a30c03fa1142dbf1721be569e85d21fba8f2b0e7e5499a | aravind-sadharani/puretones-music-room | musicscale.dsp | import("stdfaust.lib");
freq = hslider("freq",200,50,1000,0.01);
bend = ba.semi2ratio(hslider("bend[midi:pitchwheel]",0,-6,6,0.01)) : si.polySmooth(gate,0.999,1);
gain = hslider("gain",0.5,0,1,0.01);
gate = button("gate") : en.adsr(0,0,1,0.01);
cperiod = hslider("Common_Parameters/Period",2,0,3,0.1);
midiKey = ba.hz2midikey(freq);
rootKey = hslider("[0][style:radio{'B':14;'A#':13;'A':12;'G#':11;'G':10;'F#':9;'F':8;'E':7;'D#':6;'D':5;'C#':4;'C':3}]Common_Parameters/Pitch",3,3,14,1) - 3;
octave = hslider("Common_Parameters/Octave",0,-2,2,1);
noteId = (midiKey + octave*12 - rootKey) : %(12);
fineTune = hslider("Common_Parameters/Fine_Tune",0,-100,100,1);
offsetSa = 0 + hslider("Common_Parameters/12_Note_Scale/Sa/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Sa/0.01_Cent",0,-100,100,1);
offsetre = -9.78 + hslider("Common_Parameters/12_Note_Scale/re/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/re/0.01_Cent",0,-100,100,1);
offsetRe = 3.91 + hslider("Common_Parameters/12_Note_Scale/Re/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Re/0.01_Cent",0,-100,100,1);
offsetga = -5.87 + hslider("Common_Parameters/12_Note_Scale/ga/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/ga/0.01_Cent",0,-100,100,1);
offsetGa = 7.82 + hslider("Common_Parameters/12_Note_Scale/Ga/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Ga/0.01_Cent",0,-100,100,1);
offsetma = -1.96 + hslider("Common_Parameters/12_Note_Scale/ma/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/ma/0.01_Cent",0,-100,100,1);
offsetMa = 11.73 + hslider("Common_Parameters/12_Note_Scale/Ma/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Ma/0.01_Cent",0,-100,100,1);
offsetPa = 1.96 + hslider("Common_Parameters/12_Note_Scale/Pa/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Pa/0.01_Cent",0,-100,100,1);
offsetdha = -7.82 + hslider("Common_Parameters/12_Note_Scale/dha/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/dha/0.01_Cent",0,-100,100,1);
offsetDha = 5.87 + hslider("Common_Parameters/12_Note_Scale/Dha/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Dha/0.01_Cent",0,-100,100,1);
offsetni = -3.91 + hslider("Common_Parameters/12_Note_Scale/ni/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/ni/0.01_Cent",0,-100,100,1);
offsetNi = 9.78 + hslider("Common_Parameters/12_Note_Scale/Ni/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Ni/0.01_Cent",0,-100,100,1);
noteOffset = offsetSa,offsetre,offsetRe,offsetga,offsetGa,offsetma,offsetMa,offsetPa,offsetdha,offsetDha,offsetni,offsetNi : ba.selectn(12,noteId);
realFreq = bend*freq*(2^(noteOffset/1200))*(2^(fineTune/1200));
String1Tone(f,g) = StringModel(pm.f2l(f*(1+variance)),0.63,10*StringPluck,0.7,0,40) + StringModel(pm.f2l(f*(1-variance)),0.63,10*StringPluck,0.7,0,40) : *(StringEnv)
with {
variance = hslider("[00]Variance",2,0,4,0.1)/10000;
StringPluck = en.adsr(0.00001,cperiod*0.7,0.9,cperiod*0.3,g);
StringEnv = en.adsr(0.0001,cperiod*0.6,0.5,cperiod*0.5,g);
StringModel(length,pluckPosition,excitation,brightness,damping,stiffness) = 0.1*pm.endChain(egChain)
with{
openStringPick(length,stiffness,pluckPosition,excitation) = strChain
with{
dispersionFilters = par(i,2,si.smooth(stiffness)),_;
maxStringLength = 6;
nti = length*pluckPosition; // length of the upper portion of the string
itb = length*(1-pluckPosition); // length of the lower portion of the string
strChain = pm.chain(
pm.stringSegment(maxStringLength,nti) :
pm.in(excitation) :
pm.out :
dispersionFilters :
pm.stringSegment(maxStringLength,itb)
);
};
lengthTuning = 14*pm.speedOfSound/ma.SR;
stringL = length-lengthTuning;
egChain = pm.chain(
pm.lStringRigidTermination :
openStringPick(stringL,stiffness/1000,pluckPosition,excitation) :
pm.rTermination(pm.basicBlock,(-1)*pm.bridgeFilter(brightness,damping))
);
};
};
process = String1Tone(realFreq,gate)*gain*gate : fi.lowpass(3,4000) <: _,_;
effect = dm.zita_light; | https://raw.githubusercontent.com/aravind-sadharani/puretones-music-room/b237ebc23b7b11d7c018716278c2ed702d374e99/src/data/musicscale.dsp | faust | length of the upper portion of the string
length of the lower portion of the string | import("stdfaust.lib");
freq = hslider("freq",200,50,1000,0.01);
bend = ba.semi2ratio(hslider("bend[midi:pitchwheel]",0,-6,6,0.01)) : si.polySmooth(gate,0.999,1);
gain = hslider("gain",0.5,0,1,0.01);
gate = button("gate") : en.adsr(0,0,1,0.01);
cperiod = hslider("Common_Parameters/Period",2,0,3,0.1);
midiKey = ba.hz2midikey(freq);
rootKey = hslider("[0][style:radio{'B':14;'A#':13;'A':12;'G#':11;'G':10;'F#':9;'F':8;'E':7;'D#':6;'D':5;'C#':4;'C':3}]Common_Parameters/Pitch",3,3,14,1) - 3;
octave = hslider("Common_Parameters/Octave",0,-2,2,1);
noteId = (midiKey + octave*12 - rootKey) : %(12);
fineTune = hslider("Common_Parameters/Fine_Tune",0,-100,100,1);
offsetSa = 0 + hslider("Common_Parameters/12_Note_Scale/Sa/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Sa/0.01_Cent",0,-100,100,1);
offsetre = -9.78 + hslider("Common_Parameters/12_Note_Scale/re/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/re/0.01_Cent",0,-100,100,1);
offsetRe = 3.91 + hslider("Common_Parameters/12_Note_Scale/Re/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Re/0.01_Cent",0,-100,100,1);
offsetga = -5.87 + hslider("Common_Parameters/12_Note_Scale/ga/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/ga/0.01_Cent",0,-100,100,1);
offsetGa = 7.82 + hslider("Common_Parameters/12_Note_Scale/Ga/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Ga/0.01_Cent",0,-100,100,1);
offsetma = -1.96 + hslider("Common_Parameters/12_Note_Scale/ma/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/ma/0.01_Cent",0,-100,100,1);
offsetMa = 11.73 + hslider("Common_Parameters/12_Note_Scale/Ma/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Ma/0.01_Cent",0,-100,100,1);
offsetPa = 1.96 + hslider("Common_Parameters/12_Note_Scale/Pa/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Pa/0.01_Cent",0,-100,100,1);
offsetdha = -7.82 + hslider("Common_Parameters/12_Note_Scale/dha/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/dha/0.01_Cent",0,-100,100,1);
offsetDha = 5.87 + hslider("Common_Parameters/12_Note_Scale/Dha/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Dha/0.01_Cent",0,-100,100,1);
offsetni = -3.91 + hslider("Common_Parameters/12_Note_Scale/ni/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/ni/0.01_Cent",0,-100,100,1);
offsetNi = 9.78 + hslider("Common_Parameters/12_Note_Scale/Ni/Cent",0,-220,220,1) + 0.01*hslider("Common_Parameters/12_Note_Scale/Ni/0.01_Cent",0,-100,100,1);
noteOffset = offsetSa,offsetre,offsetRe,offsetga,offsetGa,offsetma,offsetMa,offsetPa,offsetdha,offsetDha,offsetni,offsetNi : ba.selectn(12,noteId);
realFreq = bend*freq*(2^(noteOffset/1200))*(2^(fineTune/1200));
String1Tone(f,g) = StringModel(pm.f2l(f*(1+variance)),0.63,10*StringPluck,0.7,0,40) + StringModel(pm.f2l(f*(1-variance)),0.63,10*StringPluck,0.7,0,40) : *(StringEnv)
with {
variance = hslider("[00]Variance",2,0,4,0.1)/10000;
StringPluck = en.adsr(0.00001,cperiod*0.7,0.9,cperiod*0.3,g);
StringEnv = en.adsr(0.0001,cperiod*0.6,0.5,cperiod*0.5,g);
StringModel(length,pluckPosition,excitation,brightness,damping,stiffness) = 0.1*pm.endChain(egChain)
with{
openStringPick(length,stiffness,pluckPosition,excitation) = strChain
with{
dispersionFilters = par(i,2,si.smooth(stiffness)),_;
maxStringLength = 6;
strChain = pm.chain(
pm.stringSegment(maxStringLength,nti) :
pm.in(excitation) :
pm.out :
dispersionFilters :
pm.stringSegment(maxStringLength,itb)
);
};
lengthTuning = 14*pm.speedOfSound/ma.SR;
stringL = length-lengthTuning;
egChain = pm.chain(
pm.lStringRigidTermination :
openStringPick(stringL,stiffness/1000,pluckPosition,excitation) :
pm.rTermination(pm.basicBlock,(-1)*pm.bridgeFilter(brightness,damping))
);
};
};
process = String1Tone(realFreq,gate)*gain*gate : fi.lowpass(3,4000) <: _,_;
effect = dm.zita_light; |
c4ff6f577bfc05093d7a4de848871f38f03086cfebc1f349a54d5e639fdd149c | ossia/score-user-library | cryBaby.dsp | declare name "cryBaby";
declare description "Application demonstrating the CryBaby wah pedal emulation";
import("stdfaust.lib");
process = dm.crybaby_demo;
| https://raw.githubusercontent.com/ossia/score-user-library/1de4c36f179105f1728e72b02e96f68d1c9130c2/Presets/Faust/filtering/cryBaby.dsp | faust | declare name "cryBaby";
declare description "Application demonstrating the CryBaby wah pedal emulation";
import("stdfaust.lib");
process = dm.crybaby_demo;
|
|
7b2210fdaa6c5c322265070cbb1f2eac0f74f5b990fdad6f78479cf5317e785c | ossia/score-user-library | parametricEqualizer.dsp | declare name "parametricEqualizer";
declare description "Exercise and compare Parametric Equalizer sections on test signals";
import("stdfaust.lib");
process = dm.parametric_eq_demo;
| https://raw.githubusercontent.com/ossia/score-user-library/1de4c36f179105f1728e72b02e96f68d1c9130c2/Presets/Faust/filtering/parametricEqualizer.dsp | faust | declare name "parametricEqualizer";
declare description "Exercise and compare Parametric Equalizer sections on test signals";
import("stdfaust.lib");
process = dm.parametric_eq_demo;
|
|
e71900fc412def3651c6fc72a5528a5b34aaa4a956b45432503e210d828b16ee | hatchjaw/teensy-wfs | WFS.dsp | declare name "Distributed WFS";
declare description "Basic WFS for a distributed setup consisting of modules that each handle two output channels.";
import("stdfaust.lib");
import("WFS_Params.lib");
// Set which speakers to control.
moduleID = hslider("moduleID", 0, 0, (N_SPEAKERS / SPEAKERS_PER_MODULE) - 1, 1);
// Simulate distance by changing gain and applying a lowpass as a function
// of distance
distanceSim(distance) = *(dGain) : fi.lowpass(2, fc)
with{
// Use inverse square law; I_2/I_1 = (d_1/d_2)^2
// Assume sensible listening distance of 5 m from array.
i1 = 1.; // Intensity 1...
d1 = 5.; // ...at distance 5 m
d2 = d1 + distance;
i2 = i1 * (d1/d2)^2; //
dGain = i2;
// dGain = (MAX_Y_DIST - distance*.5)/(MAX_Y_DIST);
fc = dGain*15000 + 5000;
};
// Create a speaker array *perspective* for one source
// i.e. give each source a distance simulation and a delay
// relative to each speaker.
speakerArray(x, y) = _ <:
par(i, SPEAKERS_PER_MODULE, distanceSim(hypotenuse(i)) : de.fdelay(MAX_DELAY, smallDelay(i)))
with{
// y (front-to-back) is always just y, the longitudinal
// distance of the source from the array.
// Get x between the source and specific speaker in the array
// E.g. for 16 speakers (8 modules), with a spacing, s, of .25 m,
// array width, w = (16-1)*.25 = 3.75,
// let module m = 2 (third module in array)
// let speaker j = 0 (first speaker in module)
// let x = 2.25 (m, relative to left edge of array)
// cx = x - s*(m*2 + j)
// = 2.25 - .25*(2*2 + 0)
// = 1.25
//
// let m = 7, j = 1, x = 2.25
// cx = 2.25 - .25*(7*2 + 1) = -1.5
//
// let m = 0, j = 0, x = 2.25
// cx = 2.25 - .25*(0*2 + 0) = 2.25
cathetusX(k) = x - (SPEAKER_DIST*(k + moduleID*2));
hypotenuse(j) = cathetusX(j)^2 + y^2 : sqrt;
smallDelay(j) = (hypotenuse(j) - y)*SAMPLES_PER_METRE;
};
// Take each source...
sourcesArray(s) = par(i, ba.count(s), ba.take(i + 1, s) :
// ...and distribute it across the speaker array for this module.
speakerArray(x(i), y(i)))
// Merge onto the output speakers.
:> par(i, SPEAKERS_PER_MODULE, _)
with{
// Use normalised input co-ordinate space; scale to dimensions.
// X position lies on the width of the speaker array
// x(p) = hslider("%p/x", 0, 0, 1, 0.001) : si.smoo : *(SPEAKER_DIST*N_SPEAKERS);
x(p) = hslider("%p/x", 0, 0, 1, 0.001) : *(SPEAKER_DIST*N_SPEAKERS);
// Y position is from zero (on the array) to a quasi-arbitrary maximum.
// y(p) = hslider("%p/y", 0, 0, 1, 0.001) : si.smoo : *(MAX_Y_DIST);
y(p) = hslider("%p/y", 0, 0, 1, 0.001) : *(MAX_Y_DIST);
};
// Distribute input channels (i.e. sources) across the sources array.
process = sourcesArray(par(i, N_SOURCES, _)); | https://raw.githubusercontent.com/hatchjaw/teensy-wfs/9d3a67f5bf9d5c26f4b7a389eb912ae7f2542269/src/faust/WFS.dsp | faust | Set which speakers to control.
Simulate distance by changing gain and applying a lowpass as a function
of distance
Use inverse square law; I_2/I_1 = (d_1/d_2)^2
Assume sensible listening distance of 5 m from array.
Intensity 1...
...at distance 5 m
dGain = (MAX_Y_DIST - distance*.5)/(MAX_Y_DIST);
Create a speaker array *perspective* for one source
i.e. give each source a distance simulation and a delay
relative to each speaker.
y (front-to-back) is always just y, the longitudinal
distance of the source from the array.
Get x between the source and specific speaker in the array
E.g. for 16 speakers (8 modules), with a spacing, s, of .25 m,
array width, w = (16-1)*.25 = 3.75,
let module m = 2 (third module in array)
let speaker j = 0 (first speaker in module)
let x = 2.25 (m, relative to left edge of array)
cx = x - s*(m*2 + j)
= 2.25 - .25*(2*2 + 0)
= 1.25
let m = 7, j = 1, x = 2.25
cx = 2.25 - .25*(7*2 + 1) = -1.5
let m = 0, j = 0, x = 2.25
cx = 2.25 - .25*(0*2 + 0) = 2.25
Take each source...
...and distribute it across the speaker array for this module.
Merge onto the output speakers.
Use normalised input co-ordinate space; scale to dimensions.
X position lies on the width of the speaker array
x(p) = hslider("%p/x", 0, 0, 1, 0.001) : si.smoo : *(SPEAKER_DIST*N_SPEAKERS);
Y position is from zero (on the array) to a quasi-arbitrary maximum.
y(p) = hslider("%p/y", 0, 0, 1, 0.001) : si.smoo : *(MAX_Y_DIST);
Distribute input channels (i.e. sources) across the sources array. | declare name "Distributed WFS";
declare description "Basic WFS for a distributed setup consisting of modules that each handle two output channels.";
import("stdfaust.lib");
import("WFS_Params.lib");
moduleID = hslider("moduleID", 0, 0, (N_SPEAKERS / SPEAKERS_PER_MODULE) - 1, 1);
distanceSim(distance) = *(dGain) : fi.lowpass(2, fc)
with{
d2 = d1 + distance;
dGain = i2;
fc = dGain*15000 + 5000;
};
speakerArray(x, y) = _ <:
par(i, SPEAKERS_PER_MODULE, distanceSim(hypotenuse(i)) : de.fdelay(MAX_DELAY, smallDelay(i)))
with{
cathetusX(k) = x - (SPEAKER_DIST*(k + moduleID*2));
hypotenuse(j) = cathetusX(j)^2 + y^2 : sqrt;
smallDelay(j) = (hypotenuse(j) - y)*SAMPLES_PER_METRE;
};
sourcesArray(s) = par(i, ba.count(s), ba.take(i + 1, s) :
speakerArray(x(i), y(i)))
:> par(i, SPEAKERS_PER_MODULE, _)
with{
x(p) = hslider("%p/x", 0, 0, 1, 0.001) : *(SPEAKER_DIST*N_SPEAKERS);
y(p) = hslider("%p/y", 0, 0, 1, 0.001) : *(MAX_Y_DIST);
};
process = sourcesArray(par(i, N_SOURCES, _)); |
ed877654488508b6aa7a418acbca4c4b88266764834728535cf9107d7feffe06 | hatchjaw/springamajig | SpringGrain.dsp | import("stdfaust.lib");
NUMGRAINS = 10;
MAXGRAINDURATION = .75;
TSIZE = 48000 * 2;
grainStart = hslider("Grain start", 754, 0, TSIZE, 1 );
grainSize = hslider("Grain length (s)", .001, .001, MAXGRAINDURATION, .001);
grainSpeed = hslider("Grain speed",1, -3, 3, 0.01);
grainDensity = hslider("Grain density", 2, .1, 25, .1);
grainRegularity = hslider("Rhythm", 0, 0, 1, 0.01);
freezeWrite = checkbox("Freeze");
// Envelopes for gating grains
//------------------------------------------------------------
envTri(d, t) = en.ar(d/2, d/2, t);
//------------------------------------------------------------
envSin(d, t) = (envTri(d, t)*ma.PI/2) : sin;
//------------------------------------------------------------
envSqrt(d, t) = envTri(d, t) : sqrt;
//------------------------------------------------------------
envHann(d, t) = (1-cos(ma.PI*envTri(d, t)))/2;
// Modulo that handles negative numbers
modulo(b, a) = a : %(b) : +(b) : %(b);
//-------------`(sparsePeriodicTrigger)`-----------------
// Emits +1 impulses ("trigger events") at an average frequency,
// with the distribution adjustable from purely periodic to purely random.
//
// ### Usage
// ```
// sparsePeriodicTrigger(f0, periodicity, pnoise) : _
// ```
//
// Where:
//
// * `f0`: average number of triggers per second.
// * `periodicity`: coefficient of distribution noise. 0 <= periodicity <= 1. 0 = random distribution, 1 = regularly spaced pulses.
// * `pnoise`: random source of probability. Pure white noise is good.
//
// Courtesy of https://github.com/myklemykle/weather_organ/blob/master/weatherorgan.dsp
//----------------------------
sparsePeriodicTrigger(f0, periodicity, noise) =
(
+(rate) // add the rate;
<: _, >=(1) // if greater than 1 ...
: _, *(1+w*noise) : - // ... subtract 1+(w*noise)
) ~ _
<: _, _' : < // emit 1 if the value decreased, 0 otherwise.
with {
w = max(0, min(1, 1 - periodicity));
rate = f0/ma.SR;
};
sparseTrigger(noiseIndex) = sparsePeriodicTrigger(grainDensity, grainRegularity, no.noises(NUMGRAINS, noiseIndex));
// A grain generator is a lookup table of samples.
grain(trigger, instance, signal) = rwtable(TSIZE, 0., int(writeIndex), signal, int(readIndex))
// It is windowed.
: *(envHann(duration, trigger))
with {
duration = grainSize;
// Constantly write input to the table...
writePos = _ ~ +(1) : %(TSIZE);
// ...unless 'Freeze' is true.
writeIndex = ba.if(freezeWrite, 0, writePos);
// Convert grain size from seconds to samples
grainLengthSamps = duration * ma.SR;
// The clock starts when the trigger is not zero, and counts down for the length of the grain.
clock = max(grainLengthSamps * (trigger != 0)) ~ (-(1) : max(0));
// Prevent retriggering while the clock is running.
// uniqueTrigger = ba.if(clock == 0, trigger, 0);
// Constantly update the read position, within the bounds of the grain size and table size...
// Set a per-lookup offset.
grainOffset = int(instance*TSIZE/NUMGRAINS);
// Add a bit of inter-grain wobble to the sample increment.
sampleIncrement = grainSpeed * (1 + (instance - (NUMGRAINS/2))/85);
readPos = _ ~ +(sampleIncrement) : modulo(grainSize * ma.SR) : +(grainStart) : +(grainOffset) : modulo(TSIZE);
// ...but only update the read poisition if the clock is running.
readIndex = ba.if(clock > 0, readPos, 0);
};
process = _ <: par(i, NUMGRAINS, grain(sparseTrigger(i), i)) :> _;
| https://raw.githubusercontent.com/hatchjaw/springamajig/e6e350531b7988deb7bff987d589c87c3a59513a/faust/SpringGrain.dsp | faust | Envelopes for gating grains
------------------------------------------------------------
------------------------------------------------------------
------------------------------------------------------------
------------------------------------------------------------
Modulo that handles negative numbers
-------------`(sparsePeriodicTrigger)`-----------------
Emits +1 impulses ("trigger events") at an average frequency,
with the distribution adjustable from purely periodic to purely random.
### Usage
```
sparsePeriodicTrigger(f0, periodicity, pnoise) : _
```
Where:
* `f0`: average number of triggers per second.
* `periodicity`: coefficient of distribution noise. 0 <= periodicity <= 1. 0 = random distribution, 1 = regularly spaced pulses.
* `pnoise`: random source of probability. Pure white noise is good.
Courtesy of https://github.com/myklemykle/weather_organ/blob/master/weatherorgan.dsp
----------------------------
add the rate;
if greater than 1 ...
... subtract 1+(w*noise)
emit 1 if the value decreased, 0 otherwise.
A grain generator is a lookup table of samples.
It is windowed.
Constantly write input to the table...
...unless 'Freeze' is true.
Convert grain size from seconds to samples
The clock starts when the trigger is not zero, and counts down for the length of the grain.
Prevent retriggering while the clock is running.
uniqueTrigger = ba.if(clock == 0, trigger, 0);
Constantly update the read position, within the bounds of the grain size and table size...
Set a per-lookup offset.
Add a bit of inter-grain wobble to the sample increment.
...but only update the read poisition if the clock is running. | import("stdfaust.lib");
NUMGRAINS = 10;
MAXGRAINDURATION = .75;
TSIZE = 48000 * 2;
grainStart = hslider("Grain start", 754, 0, TSIZE, 1 );
grainSize = hslider("Grain length (s)", .001, .001, MAXGRAINDURATION, .001);
grainSpeed = hslider("Grain speed",1, -3, 3, 0.01);
grainDensity = hslider("Grain density", 2, .1, 25, .1);
grainRegularity = hslider("Rhythm", 0, 0, 1, 0.01);
freezeWrite = checkbox("Freeze");
envTri(d, t) = en.ar(d/2, d/2, t);
envSin(d, t) = (envTri(d, t)*ma.PI/2) : sin;
envSqrt(d, t) = envTri(d, t) : sqrt;
envHann(d, t) = (1-cos(ma.PI*envTri(d, t)))/2;
modulo(b, a) = a : %(b) : +(b) : %(b);
sparsePeriodicTrigger(f0, periodicity, noise) =
(
) ~ _
with {
w = max(0, min(1, 1 - periodicity));
rate = f0/ma.SR;
};
sparseTrigger(noiseIndex) = sparsePeriodicTrigger(grainDensity, grainRegularity, no.noises(NUMGRAINS, noiseIndex));
grain(trigger, instance, signal) = rwtable(TSIZE, 0., int(writeIndex), signal, int(readIndex))
: *(envHann(duration, trigger))
with {
duration = grainSize;
writePos = _ ~ +(1) : %(TSIZE);
writeIndex = ba.if(freezeWrite, 0, writePos);
grainLengthSamps = duration * ma.SR;
clock = max(grainLengthSamps * (trigger != 0)) ~ (-(1) : max(0));
grainOffset = int(instance*TSIZE/NUMGRAINS);
sampleIncrement = grainSpeed * (1 + (instance - (NUMGRAINS/2))/85);
readPos = _ ~ +(sampleIncrement) : modulo(grainSize * ma.SR) : +(grainStart) : +(grainOffset) : modulo(TSIZE);
readIndex = ba.if(clock > 0, readPos, 0);
};
process = _ <: par(i, NUMGRAINS, grain(sparseTrigger(i), i)) :> _;
|
d20268953ea88e5487dfe56e883f10394a205933be96177c2923bd4b275f603b | agraef/pd-remote | amp.dsp |
/* Stereo amplifier stage with bass, treble, gain and balance controls and a
dB meter. */
declare name "amp";
declare description "stereo amplifier stage";
declare author "Albert Graef";
declare version "2.0";
import("stdfaust.lib");
/* Fixed bass and treble frequencies. You might want to tune these for your
setup. */
bass_freq = 300;
treble_freq = 1200;
/* Smoothing (lowpass) filter from signals.lib. We use this for the gain and
balance controls to avoid zipper noise. */
smooth = si.smooth(0.99);
/* Bass and treble gain controls in dB. The range of +/-20 corresponds to a
boost/cut factor of 10. */
bass_gain = nentry("[1] bass [midi:ctrl 16] [unit:dB]", 0, -20, 20, 0.1);
treble_gain = nentry("[2] treble [midi:ctrl 17] [unit:dB]", 0, -20, 20, 0.1);
/* Gain and balance controls. */
gain = smooth(ba.db2linear(g)) with {
g = nentry("[3] gain [midi:ctrl 7] [unit:dB]", 0, -60, 10, 0.1);
};
bal = smooth(b) with {
b = hslider("balance [midi:ctrl 8]", 0, -1, 1, 0.001);
};
/* Balance a stereo signal using the constant power pan rule. Note that this
will attenuate the signal by 3 dB in the center position. If you prefer,
you can just apply a makeup gain of sqrt(2) to have unity gain in the
center position, but this will also boost the signal by the same amount
(possibly causing distortion) when panning hard left or right. */
balance = *(l), *(r) with {
p = ma.PI*(bal+1)/4; l = cos(p); r = sin(p);
};
/* Generic biquad filter. */
filter(b0,b1,b2,a0,a1,a2) = f : (+ ~ g)
with {
f(x) = (b0/a0)*x+(b1/a0)*x'+(b2/a0)*x'';
g(y) = 0-(a1/a0)*y-(a2/a0)*y';
};
/* Low and high shelf filters, straight from Robert Bristow-Johnson's "Audio
EQ Cookbook", see http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt. f0
is the shelf midpoint frequency, g the desired gain in dB. S is the shelf
slope parameter, we always set that to 1 here. */
low_shelf(f0,g) = filter(b0,b1,b2,a0,a1,a2)
with {
S = 1;
A = pow(10,g/40);
w0 = 2*ma.PI*f0/ma.SR;
alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/S - 1) + 2 );
b0 = A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha );
b1 = 2*A*( (A-1) - (A+1)*cos(w0) );
b2 = A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha );
a0 = (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha;
a1 = -2*( (A-1) + (A+1)*cos(w0) );
a2 = (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha;
};
high_shelf(f0,g) = filter(b0,b1,b2,a0,a1,a2)
with {
S = 1;
A = pow(10,g/40);
w0 = 2*ma.PI*f0/ma.SR;
alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/S - 1) + 2 );
b0 = A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha );
b1 = -2*A*( (A-1) + (A+1)*cos(w0) );
b2 = A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha );
a0 = (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha;
a1 = 2*( (A-1) - (A+1)*cos(w0) );
a2 = (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha;
};
/* The tone control. We simply run a low and a high shelf in series here. */
tone = low_shelf(bass_freq,bass_gain)
: high_shelf(treble_freq,treble_gain);
/* Envelop follower. This is basically a 1 pole LP with configurable attack/
release time. The result is converted to dB. You have to set the desired
attack/release time in seconds using the t parameter below. */
t = 0.1; // attack/release time in seconds
g = exp(-1/(ma.SR*t)); // corresponding gain factor
env = abs : *(1-g) : + ~ *(g) : ba.linear2db;
/* Use this if you want the RMS instead. Note that this doesn't really
calculate an RMS value (you'd need an FIR for that), but in practice our
simple 1 pole IIR filter works just as well. */
rms = sqr : *(1-g) : + ~ *(g) : sqrt : ba.linear2db;
sqr(x) = x*x;
/* The dB meters for left and right channel. These are passive controls. */
left_meter(x) = attach(x, env(x) : hbargraph("left [midi:ctrl 18] [osc:/left -60 10] [unit:dB]", -60, 10));
right_meter(x) = attach(x, env(x) : hbargraph("right [midi:ctrl 19] [osc:/right -60 10] [unit:dB]", -60, 10));
/* The main program. */
process = hgroup("[1]", (tone, tone) : (_*gain, _*gain))
: vgroup("[2]", balance)
: vgroup("[3]", (left_meter, right_meter));
| https://raw.githubusercontent.com/agraef/pd-remote/4fede0b70ac5f9544a783dd45ddcf4643a29bc63/examples/dsp/amp.dsp | faust | Stereo amplifier stage with bass, treble, gain and balance controls and a
dB meter.
Fixed bass and treble frequencies. You might want to tune these for your
setup.
Smoothing (lowpass) filter from signals.lib. We use this for the gain and
balance controls to avoid zipper noise.
Bass and treble gain controls in dB. The range of +/-20 corresponds to a
boost/cut factor of 10.
Gain and balance controls.
Balance a stereo signal using the constant power pan rule. Note that this
will attenuate the signal by 3 dB in the center position. If you prefer,
you can just apply a makeup gain of sqrt(2) to have unity gain in the
center position, but this will also boost the signal by the same amount
(possibly causing distortion) when panning hard left or right.
Generic biquad filter.
Low and high shelf filters, straight from Robert Bristow-Johnson's "Audio
EQ Cookbook", see http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt. f0
is the shelf midpoint frequency, g the desired gain in dB. S is the shelf
slope parameter, we always set that to 1 here.
The tone control. We simply run a low and a high shelf in series here.
Envelop follower. This is basically a 1 pole LP with configurable attack/
release time. The result is converted to dB. You have to set the desired
attack/release time in seconds using the t parameter below.
attack/release time in seconds
corresponding gain factor
Use this if you want the RMS instead. Note that this doesn't really
calculate an RMS value (you'd need an FIR for that), but in practice our
simple 1 pole IIR filter works just as well.
The dB meters for left and right channel. These are passive controls.
The main program. |
declare name "amp";
declare description "stereo amplifier stage";
declare author "Albert Graef";
declare version "2.0";
import("stdfaust.lib");
bass_freq = 300;
treble_freq = 1200;
smooth = si.smooth(0.99);
bass_gain = nentry("[1] bass [midi:ctrl 16] [unit:dB]", 0, -20, 20, 0.1);
treble_gain = nentry("[2] treble [midi:ctrl 17] [unit:dB]", 0, -20, 20, 0.1);
gain = smooth(ba.db2linear(g)) with {
g = nentry("[3] gain [midi:ctrl 7] [unit:dB]", 0, -60, 10, 0.1);
};
bal = smooth(b) with {
b = hslider("balance [midi:ctrl 8]", 0, -1, 1, 0.001);
};
balance = *(l), *(r) with {
p = ma.PI*(bal+1)/4; l = cos(p); r = sin(p);
};
filter(b0,b1,b2,a0,a1,a2) = f : (+ ~ g)
with {
f(x) = (b0/a0)*x+(b1/a0)*x'+(b2/a0)*x'';
g(y) = 0-(a1/a0)*y-(a2/a0)*y';
};
low_shelf(f0,g) = filter(b0,b1,b2,a0,a1,a2)
with {
S = 1;
A = pow(10,g/40);
w0 = 2*ma.PI*f0/ma.SR;
alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/S - 1) + 2 );
b0 = A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha );
b1 = 2*A*( (A-1) - (A+1)*cos(w0) );
b2 = A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha );
a0 = (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha;
a1 = -2*( (A-1) + (A+1)*cos(w0) );
a2 = (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha;
};
high_shelf(f0,g) = filter(b0,b1,b2,a0,a1,a2)
with {
S = 1;
A = pow(10,g/40);
w0 = 2*ma.PI*f0/ma.SR;
alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/S - 1) + 2 );
b0 = A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha );
b1 = -2*A*( (A-1) + (A+1)*cos(w0) );
b2 = A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha );
a0 = (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha;
a1 = 2*( (A-1) - (A+1)*cos(w0) );
a2 = (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha;
};
tone = low_shelf(bass_freq,bass_gain)
: high_shelf(treble_freq,treble_gain);
env = abs : *(1-g) : + ~ *(g) : ba.linear2db;
rms = sqr : *(1-g) : + ~ *(g) : sqrt : ba.linear2db;
sqr(x) = x*x;
left_meter(x) = attach(x, env(x) : hbargraph("left [midi:ctrl 18] [osc:/left -60 10] [unit:dB]", -60, 10));
right_meter(x) = attach(x, env(x) : hbargraph("right [midi:ctrl 19] [osc:/right -60 10] [unit:dB]", -60, 10));
process = hgroup("[1]", (tone, tone) : (_*gain, _*gain))
: vgroup("[2]", balance)
: vgroup("[3]", (left_meter, right_meter));
|
3d597511b3ed2c80d54a2f7d684fc0560a93380b310a18a3afddd3509be66cb7 | tonal-glyph/faustus | tp0.dsp | import("stdfaust.lib");
process = 0,_~+(1):soundfile("son[url:{'tango.wav'}]",2):!,!,_,_;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/tests/soundfile/tp0.dsp | faust | import("stdfaust.lib");
process = 0,_~+(1):soundfile("son[url:{'tango.wav'}]",2):!,!,_,_;
|
|
00492b9697017118c2edd8dfd20b50619398ae6d38e51e752a959518fc129089 | tonal-glyph/faustus | filterBank.dsp | declare name "filterBank";
declare description "Graphic Equalizer consisting of a filter-bank driving a bank of faders";
import("stdfaust.lib");
process = dm.filterbank_demo;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/filtering/filterBank.dsp | faust | declare name "filterBank";
declare description "Graphic Equalizer consisting of a filter-bank driving a bank of faders";
import("stdfaust.lib");
process = dm.filterbank_demo;
|
|
008139be97257f7bd5e555d5a4896eb1bf8e017bc1a33ec25550c91a6358ac0c | tonal-glyph/faustus | wahPedal.dsp | declare name "wahPedal";
declare description "Demonstrate the Fourth-Order Wah pedal (similar to the Moog VCF)";
import("stdfaust.lib");
process = dm.wah4_demo;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/filtering/wahPedal.dsp | faust | declare name "wahPedal";
declare description "Demonstrate the Fourth-Order Wah pedal (similar to the Moog VCF)";
import("stdfaust.lib");
process = dm.wah4_demo;
|
|
ac90eb891cfde62e2c62f07c0383dac3b17416dba0275170c19270bc571df3d9 | tonal-glyph/faustus | spectralTilt.dsp | declare name "spectralTilt";
declare description "Demonstrate the Spectral Tilt effect on test signals";
import("stdfaust.lib");
O = 2; // filter order
process = dm.spectral_tilt_demo(2);
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/filtering/spectralTilt.dsp | faust | filter order | declare name "spectralTilt";
declare description "Demonstrate the Spectral Tilt effect on test signals";
import("stdfaust.lib");
process = dm.spectral_tilt_demo(2);
|
09423aa851e966a347fcd8213828efb2f145fce8ecd4342ff4586fb1d5eba895 | tonal-glyph/faustus | effects.dsp | // All effects used by minimoog.dsp
import("stdfaust.lib");
import("layout2.dsp");
process = _,_ : +
: component("echo.dsp")
: component("flanger.dsp")
: component("chorus.dsp")
: component("freeverb.dsp");
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/SAM/effects/effects.dsp | faust | All effects used by minimoog.dsp |
import("stdfaust.lib");
import("layout2.dsp");
process = _,_ : +
: component("echo.dsp")
: component("flanger.dsp")
: component("chorus.dsp")
: component("freeverb.dsp");
|
e1bd3c704e2a49218db2dcab7f9a96a6c2bb40c7f89bba1e5f27eb2f84e8a277 | tonal-glyph/faustus | flute.dsp | declare name "Flute";
declare description "Simple flute physical model with physical parameters.";
declare license "MIT";
declare copyright "(c)Romain Michon, CCRMA (Stanford University), GRAME";
import("stdfaust.lib");
process = pm.flute_ui <: _,_;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/physicalModeling/flute.dsp | faust | declare name "Flute";
declare description "Simple flute physical model with physical parameters.";
declare license "MIT";
declare copyright "(c)Romain Michon, CCRMA (Stanford University), GRAME";
import("stdfaust.lib");
process = pm.flute_ui <: _,_;
|
|
752974d9cc112ca78663c2de3e24fb64a3846ad29b2b7222d8f665d9989acab7 | tonal-glyph/faustus | clarinet.dsp | declare name "Clarinet";
declare description "Simple clarinet physical model with physical parameters.";
declare license "MIT";
declare copyright "(c)Romain Michon, CCRMA (Stanford University), GRAME";
import("stdfaust.lib");
process = pm.clarinet_ui <: _,_;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/physicalModeling/clarinet.dsp | faust | declare name "Clarinet";
declare description "Simple clarinet physical model with physical parameters.";
declare license "MIT";
declare copyright "(c)Romain Michon, CCRMA (Stanford University), GRAME";
import("stdfaust.lib");
process = pm.clarinet_ui <: _,_;
|
|
f3f0354b24c5f289c5695f1e9082a476c9cbc5247c8c260f98349a085ea4b09a | tonal-glyph/faustus | volume.dsp | //---------------------------------------------------------
// Volume control in dB with MIDI control (CC-1, modWheel)
//---------------------------------------------------------
import("stdfaust.lib");
gain = vslider("Volume[midi:ctrl 2] [tooltip CC-1]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo;
process = _,_: *(gain), *(gain);
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/SAM/volume/volume.dsp | faust | ---------------------------------------------------------
Volume control in dB with MIDI control (CC-1, modWheel)
--------------------------------------------------------- |
import("stdfaust.lib");
gain = vslider("Volume[midi:ctrl 2] [tooltip CC-1]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo;
process = _,_: *(gain), *(gain);
|
4809b2fe432a5d026d4f61ba8a9545ac00ab3bc5a16bb6ed3c5aa8c35f7f9e34 | tonal-glyph/faustus | sine_synth.dsp | // tosc.dsp - test simple oscillator + MIDI bindings
import("stdfaust.lib");
process = g * a * os.oscrs(f*b) <: _,_;
a = hslider("gain [midi:ctrl 7]",1,0,1,0.001);
f = hslider("freq",392.0,200.0,450.0,0.01);
b = hslider("bend [midi:pitchwheel]",1,0.1,10,0.001);
g = button("gate");
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/SAM/sine_synth/sine_synth.dsp | faust | tosc.dsp - test simple oscillator + MIDI bindings | import("stdfaust.lib");
process = g * a * os.oscrs(f*b) <: _,_;
a = hslider("gain [midi:ctrl 7]",1,0,1,0.001);
f = hslider("freq",392.0,200.0,450.0,0.01);
b = hslider("bend [midi:pitchwheel]",1,0.1,10,0.001);
g = button("gate");
|
2ac67558ab40c4efc1acf32effdedd45380fb676790bdc31e593791108332b35 | tonal-glyph/faustus | virtualAnalogLab.dsp | declare name "virtualAnalogLab";
import("stdfaust.lib");
process =
vgroup("[1]", dm.virtual_analog_oscillator_demo) :
vgroup("[2]", dm.moog_vcf_demo) :
vgroup("[3]", dm.spectral_level_demo)
// See also: vgroup("[3]", dm.fft_spectral_level_demo(32))
<: _,_;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/generator/virtualAnalogLab.dsp | faust | See also: vgroup("[3]", dm.fft_spectral_level_demo(32)) | declare name "virtualAnalogLab";
import("stdfaust.lib");
process =
vgroup("[1]", dm.virtual_analog_oscillator_demo) :
vgroup("[2]", dm.moog_vcf_demo) :
vgroup("[3]", dm.spectral_level_demo)
<: _,_;
|
90f793ac2bdb9e9d6e9967b76ff7a4256cf19d838c1db11b805bc0f56ce717d2 | tonal-glyph/faustus | filteredSawtooth.dsp | import("stdfaust.lib");
freq = nentry("freq",50,200,1000,0.01) ;
gain = nentry("gain",0.5,0,1,0.01) : si.smoo;
gate = button("gate") : si.smoo;
cutoff = nentry("cutoff",10000,50,10000,0.01) : si.smoo;
process = os.sawtooth(freq)*gain*gate : fi.lowpass(3,cutoff) <: _,_;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/documentation/faust-manual/misc/juce/filteredSawtooth.dsp | faust | import("stdfaust.lib");
freq = nentry("freq",50,200,1000,0.01) ;
gain = nentry("gain",0.5,0,1,0.01) : si.smoo;
gate = button("gate") : si.smoo;
cutoff = nentry("cutoff",10000,50,10000,0.01) : si.smoo;
process = os.sawtooth(freq)*gain*gate : fi.lowpass(3,cutoff) <: _,_;
|
|
a5d16294274fc10d2941c4a963a0252e740fee5080005badecd4bff431acfbc9 | tonal-glyph/faustus | reverbDesigner.dsp | declare name "reverbDesigner";
import("stdfaust.lib");
N = 16; // Feedback Delay Network (FDN) order (power of 2, 2 to 16)
NB = 5; // Number of T60-controlled frequency-bands (3 or more)
BSO = 3; // Order of each lowpass/highpass bandsplit (odd positive integer)
process = dm.fdnrev0_demo(N,NB,BSO);
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/reverb/reverbDesigner.dsp | faust | Feedback Delay Network (FDN) order (power of 2, 2 to 16)
Number of T60-controlled frequency-bands (3 or more)
Order of each lowpass/highpass bandsplit (odd positive integer) | declare name "reverbDesigner";
import("stdfaust.lib");
process = dm.fdnrev0_demo(N,NB,BSO);
|
781e9af137503075b79bdbe210e5b44cb0f17c5b009bc7a88a09403a68268fb5 | tonal-glyph/faustus | graphicEqLab.dsp | declare name "graphicEqLab";
declare description "Signal generators through a filter bank with spectrum analysis display";
import("stdfaust.lib");
process =
// ol.sawtooth_demo : fl.filterbank_demo : fl.spectral_level_demo <: _,_;
vgroup("[1]",dm.sawtooth_demo) :
vgroup("[2]",dm.filterbank_demo) :
vgroup("[3]",dm.spectral_level_demo) <:
_,_;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/filtering/graphicEqLab.dsp | faust | ol.sawtooth_demo : fl.filterbank_demo : fl.spectral_level_demo <: _,_; | declare name "graphicEqLab";
declare description "Signal generators through a filter bank with spectrum analysis display";
import("stdfaust.lib");
process =
vgroup("[1]",dm.sawtooth_demo) :
vgroup("[2]",dm.filterbank_demo) :
vgroup("[3]",dm.spectral_level_demo) <:
_,_;
|
d37f31515cf4cb46e8cb4957a00ca5b7943634932ef198a4eb85712ae5e2cc49 | tonal-glyph/faustus | sawtooth_synth.dsp |
import("stdfaust.lib");
normMIDI(mv) = mv/127.0;
vol = normMIDI(hslider("Ctrl Value IN (Ctrl 1) [midi:ctrl 1]", 60, 0, 127, 1)) ;
f = nentry("freq",200,40,2000,0.01);
bend = nentry("bend",1,0,10,0.01) : si.polySmooth(t,0.999,1);
g = nentry("gain",1,0,1,0.01);
t = button("gate");
freq = f*bend;
envelope = t*g*vol : si.smoo;
process = os.sawtooth(freq)*envelope <: _,_;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/SAM/sawtooth_synth/sawtooth_synth.dsp | faust |
import("stdfaust.lib");
normMIDI(mv) = mv/127.0;
vol = normMIDI(hslider("Ctrl Value IN (Ctrl 1) [midi:ctrl 1]", 60, 0, 127, 1)) ;
f = nentry("freq",200,40,2000,0.01);
bend = nentry("bend",1,0,10,0.01) : si.polySmooth(t,0.999,1);
g = nentry("gain",1,0,1,0.01);
t = button("gate");
freq = f*bend;
envelope = t*g*vol : si.smoo;
process = os.sawtooth(freq)*envelope <: _,_;
|
|
9a811666c849d5057ad5f51f8e743d6237469d6641717b656d2154490ee6677a | tonal-glyph/faustus | parametricEqLab.dsp | declare name "parametricEqLab";
declare description "Demonstrate the Parametric Equalizer sections on test signals with spectrum analysis display";
import("stdfaust.lib");
//process = ol.sawtooth_demo : fl.parametric_eq_demo :
// fl.mth_octave_spectral_level_demo(2) <: _,_;
process =
vgroup("[1]", dm.sawtooth_demo) :
vgroup("[2]", dm.parametric_eq_demo) :
vgroup("[3]", dm.mth_octave_spectral_level_demo(2))
<: _,_;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/filtering/parametricEqLab.dsp | faust | process = ol.sawtooth_demo : fl.parametric_eq_demo :
fl.mth_octave_spectral_level_demo(2) <: _,_; | declare name "parametricEqLab";
declare description "Demonstrate the Parametric Equalizer sections on test signals with spectrum analysis display";
import("stdfaust.lib");
process =
vgroup("[1]", dm.sawtooth_demo) :
vgroup("[2]", dm.parametric_eq_demo) :
vgroup("[3]", dm.mth_octave_spectral_level_demo(2))
<: _,_;
|
796fb0888ad0640692cca2b624ba966caa62d5fe5b23320ecc322b40d7740eb2 | tonal-glyph/faustus | rain.dsp | //----------------------`rain`--------------------------
// A very simple rain simulator
//
// #### Usage
//
// ```
// rain(d,l) : _,_
// ```
//
// Where:
//
// * `d`: is the density of the rain: between 0 and 1
// * `l`: is the level (volume) of the rain: between 0 and 1
//
//----------------------------------------------------------
import("stdfaust.lib");
rain(density,level) = no.multinoise(2) : par(i, 2, drop) : par(i, 2, *(level))
with {
drop = _ <: @(1), (abs < density) : *;
};
process = rain (
hslider("v:rain/density", 300, 0, 1000, 1) / 1000,
hslider("v:rain/volume", 0.5, 0, 1, 0.01)
);
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/gameaudio/rain.dsp | faust | ----------------------`rain`--------------------------
A very simple rain simulator
#### Usage
```
rain(d,l) : _,_
```
Where:
* `d`: is the density of the rain: between 0 and 1
* `l`: is the level (volume) of the rain: between 0 and 1
---------------------------------------------------------- |
import("stdfaust.lib");
rain(density,level) = no.multinoise(2) : par(i, 2, drop) : par(i, 2, *(level))
with {
drop = _ <: @(1), (abs < density) : *;
};
process = rain (
hslider("v:rain/density", 300, 0, 1000, 1) / 1000,
hslider("v:rain/volume", 0.5, 0, 1, 0.01)
);
|
f7b0f9d346cc946fe8b7ac920d2d87155e7b9fd9d25b253242f9c2265189259a | tonal-glyph/faustus | vcfWahLab.dsp | import("stdfaust.lib");
declare description "Demonstrate competing variable-lowpass-filter effects on test signals with spectrum analysis display";
declare name "vcfWahLab";
// process = ol.sawtooth_demo :
// el.crybaby_demo : el.moog_vcf_demo : el.wah4_demo :
// fl.spectral_level_demo <: _,_;
process =
vgroup("[1]", dm.sawtooth_demo) :
vgroup("[2]", dm.crybaby_demo) :
vgroup("[3]", dm.wah4_demo) :
vgroup("[4]", dm.moog_vcf_demo) :
vgroup("[5]", dm.spectral_level_demo) <:
_,_;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/filtering/vcfWahLab.dsp | faust | process = ol.sawtooth_demo :
el.crybaby_demo : el.moog_vcf_demo : el.wah4_demo :
fl.spectral_level_demo <: _,_; | import("stdfaust.lib");
declare description "Demonstrate competing variable-lowpass-filter effects on test signals with spectrum analysis display";
declare name "vcfWahLab";
process =
vgroup("[1]", dm.sawtooth_demo) :
vgroup("[2]", dm.crybaby_demo) :
vgroup("[3]", dm.wah4_demo) :
vgroup("[4]", dm.moog_vcf_demo) :
vgroup("[5]", dm.spectral_level_demo) <:
_,_;
|
dc5fd39dc3615f6e083c77796759c94191b3bc9dfe4d1969b8c601f75a3a72c3 | tonal-glyph/faustus | sound.dsp |
import("stdfaust.lib");
so_loop_speed(s, part, speed) = (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) = float(speed*srate(s))/ma.SR : (+,length(s):fmod)~_ : int;
};
part = nentry("file", 0, 0, 10, 1);
speed = hslider("speed", 1, 0, 4, 0.01);
process = so_loop_speed(soundfile("s1.wav",1), part, speed),
so_loop_speed(soundfile("s2.wav",1), part, speed) : dm.freeverb_demo;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/tests/architecture-tests/sound.dsp | faust |
import("stdfaust.lib");
so_loop_speed(s, part, speed) = (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) = float(speed*srate(s))/ma.SR : (+,length(s):fmod)~_ : int;
};
part = nentry("file", 0, 0, 10, 1);
speed = hslider("speed", 1, 0, 4, 0.01);
process = so_loop_speed(soundfile("s1.wav",1), part, speed),
so_loop_speed(soundfile("s2.wav",1), part, speed) : dm.freeverb_demo;
|
|
e15bf34652b4d0ec016cdd1b45d22a2dd51a54c95368d4c9f9ef020435dfe72d | tonal-glyph/faustus | organ.dsp | // Simple Organ
declare nvoices "8";
import("stdfaust.lib");
// Midi interface
midigate = button ("gate"); // MIDI keyon-keyoff
midifreq = hslider("freq[unit:Hz]", 440, 20, 20000, 1); // MIDI keyon key
midigain = hslider("gain", 0.5, 0, 10, 0.01); // MIDI keyon velocity
process = voice(midigate, midigain, midifreq) * hslider("volume", 0.5, 0, 1, 0.01);
// Implementation
phasor(f) = f/ma.SR : (+,1.0:fmod) ~ _ ;
osc(f) = phasor(f) * 6.28318530718 : sin;
timbre(freq) = osc(freq) + 0.5 * osc(2.0*freq) + 0.25 * osc(3.0*freq);
envelop(gate, gain) = gate * gain : smooth(0.9995)
with { smooth(c) = * (1-c) : + ~ * (c) ; } ;
voice(gate, gain, freq) = envelop(gate, gain) * timbre(freq);
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/tests/architecture-tests/organ.dsp | faust | Simple Organ
Midi interface
MIDI keyon-keyoff
MIDI keyon key
MIDI keyon velocity
Implementation | declare nvoices "8";
import("stdfaust.lib");
process = voice(midigate, midigain, midifreq) * hslider("volume", 0.5, 0, 1, 0.01);
phasor(f) = f/ma.SR : (+,1.0:fmod) ~ _ ;
osc(f) = phasor(f) * 6.28318530718 : sin;
timbre(freq) = osc(freq) + 0.5 * osc(2.0*freq) + 0.25 * osc(3.0*freq);
envelop(gate, gain) = gate * gain : smooth(0.9995)
with { smooth(c) = * (1-c) : + ~ * (c) ; } ;
voice(gate, gain, freq) = envelop(gate, gain) * timbre(freq);
|
71074380f21a6ccddd9bd33d944ee0adaaf558d485a25d91bdd92d2048c90da9 | tonal-glyph/faustus | tp1.dsp | import("stdfaust.lib");
part = nentry("file", 0, 0, 10, 1);
speed = hslider("speed", 1, 0, 4, 0.01);
level = hslider("level", 0.5, 0, 1, 0.01);
//process = (part, +(1)~_) : soundfile("files [url: {'RnB.wav';'tango.wav';'levot.wav'}]",2) :(!,!,_,_);
//process = so.loop(soundfile("files [url: {'/Documents/faust-github-faust2/tests/soundfile/RnB.wav';'/Documents/faust-github-faust2/tests/soundfile/tango.wav';'/Documents/faust-github-faust2/tests/soundfile/levot.wav'}]",2), part);
//process = so.loop_speed(soundfile("files [url: {'RnB.wav';'tango.wav';'levot.wav'}]",2), part, speed);
process = so.loop_speed_level(soundfile("files [url: {'RnB.wav';'tango.wav';'levot.wav'}]",2), part, speed, level);
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/tests/soundfile/tp1.dsp | faust | process = (part, +(1)~_) : soundfile("files [url: {'RnB.wav';'tango.wav';'levot.wav'}]",2) :(!,!,_,_);
process = so.loop(soundfile("files [url: {'/Documents/faust-github-faust2/tests/soundfile/RnB.wav';'/Documents/faust-github-faust2/tests/soundfile/tango.wav';'/Documents/faust-github-faust2/tests/soundfile/levot.wav'}]",2), part);
process = so.loop_speed(soundfile("files [url: {'RnB.wav';'tango.wav';'levot.wav'}]",2), part, speed); | import("stdfaust.lib");
part = nentry("file", 0, 0, 10, 1);
speed = hslider("speed", 1, 0, 4, 0.01);
level = hslider("level", 0.5, 0, 1, 0.01);
process = so.loop_speed_level(soundfile("files [url: {'RnB.wav';'tango.wav';'levot.wav'}]",2), part, speed, level);
|
dd7c13304e8396454e80133294aa64693b86194ddbb312b00b71f911b1eb07f5 | tonal-glyph/faustus | harpe.dsp | //-----------------------------------------------
// Basic harpe simulation with OSC control
// (based on Karplus-Strong)
//
//-----------------------------------------------
declare name "harpe";
declare author "Grame";
import("stdfaust.lib");
process = harpe(11); // an 11 strings harpe
//-----------------------------------------------
// String simulation
//-----------------------------------------------
string(freq, att, level, trig) = no.noise*level
: *(trig : trigger(freq2samples(freq)))
: resonator(freq2samples(freq), att)
with {
resonator(d, a) = (+ : @(d-1)) ~ (average : *(1.0-a));
average(x) = (x+x')/2;
trigger(n) = upfront : + ~ decay(n) : >(0.0);
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
freq2samples(f) = 44100.0/f;
};
//-----------------------------------------------
// Build a N strings harpe
// Each string is triggered by a specific
// position [0..1] of the "hand"
//-----------------------------------------------
harpe(N) = hand <: par(i, N, position((i+0.5)/N)
: string( 440 * 2.0^(i/5.0), att, lvl)
: pan((i+0.5)/N) )
:> _,_
with {
lvl = hslider("level [unit:f][osc:/accxyz/0 -10 10]", 0.5, 0, 1, 0.01)^2;
att = hslider("attenuation [osc:/1/fader3]", 0.005, 0, 0.01, 0.001);
hand = hslider("hand[osc:/accxyz/1 0 20]", 0, 0, 1, 0.01):smooth(0.9);
pan(p) = _ <: *(sqrt(1-p)), *(sqrt(p));
position(a,x) = (min(x,x') < a) & (a < max(x, x'));
smooth(c) = *(1.0-c) : + ~ *(c);
};
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/physicalModeling/old/harpe.dsp | faust | -----------------------------------------------
Basic harpe simulation with OSC control
(based on Karplus-Strong)
-----------------------------------------------
an 11 strings harpe
-----------------------------------------------
String simulation
-----------------------------------------------
-----------------------------------------------
Build a N strings harpe
Each string is triggered by a specific
position [0..1] of the "hand"
----------------------------------------------- |
declare name "harpe";
declare author "Grame";
import("stdfaust.lib");
string(freq, att, level, trig) = no.noise*level
: *(trig : trigger(freq2samples(freq)))
: resonator(freq2samples(freq), att)
with {
resonator(d, a) = (+ : @(d-1)) ~ (average : *(1.0-a));
average(x) = (x+x')/2;
trigger(n) = upfront : + ~ decay(n) : >(0.0);
upfront(x) = (x-x') > 0.0;
decay(n,x) = x - (x>0.0)/n;
freq2samples(f) = 44100.0/f;
};
harpe(N) = hand <: par(i, N, position((i+0.5)/N)
: string( 440 * 2.0^(i/5.0), att, lvl)
: pan((i+0.5)/N) )
:> _,_
with {
lvl = hslider("level [unit:f][osc:/accxyz/0 -10 10]", 0.5, 0, 1, 0.01)^2;
att = hslider("attenuation [osc:/1/fader3]", 0.005, 0, 0.01, 0.001);
hand = hslider("hand[osc:/accxyz/1 0 20]", 0, 0, 1, 0.01):smooth(0.9);
pan(p) = _ <: *(sqrt(1-p)), *(sqrt(p));
position(a,x) = (min(x,x') < a) & (a < max(x, x'));
smooth(c) = *(1.0-c) : + ~ *(c);
};
|
b33c51e76a6d267e235caa47c1af6dac18ec722c9ba98aee6649150b3183c4ea | tonal-glyph/faustus | AdditiveSynth.dsp | import("stdfaust.lib");
///////////////////////////////////////////////////////////////////////////////////////////////////
//
// Additive synthesizer, must be used with OSC message to program sound.
// It as 8 harmonics. Each have it's own volume envelope.
//
///////////////////////////////////////////////////////////////////////////////////////////////////
//
// OSC messages (see BELA console for precise adress)
// For each harmonics (%rang indicate harmonic number, starting at 0) :
// vol%rang : General Volume (vol0 control the volume of the fundamental)
// A%rang : Attack
// D%rang : Decay
// S%rang : Sustain
// R%rang : Release
//
///////////////////////////////////////////////////////////////////////////////////////////////////
// GENERAL
midigate = button("gate");
midifreq = nentry("freq[unit:Hz]", 440, 20, 20000, 1);
midigain = nentry("gain", 0.5, 0, 10, 0.01);
// pitchwheel
pitchwheel = hslider("bend [midi:pitchwheel]",1,0.001,10,0.01);
gFreq = midifreq * pitchwheel;
partiel(rang) = os.oscrs(gFreq*(rang+1))*volume
with {
// UI
vol = hslider("vol%rang", 1, 0, 1, 0.001);
a = 0.01 * hslider("A%rang", 1, 0, 400, 0.001);
d = 0.01 * hslider("D%rang", 1, 0, 400, 0.001);
s = hslider("S%rang", 1, 0, 1, 0.001);
r = 0.01 * hslider("R%rang", 1, 0, 800, 0.001);
volume = ((en.adsr(a,d,s,r,midigate))*vol) : max (0) : min (1);
};
process = par(i, 8, partiel(i)) :> / (8);
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/bela/AdditiveSynth.dsp | faust | /////////////////////////////////////////////////////////////////////////////////////////////////
Additive synthesizer, must be used with OSC message to program sound.
It as 8 harmonics. Each have it's own volume envelope.
/////////////////////////////////////////////////////////////////////////////////////////////////
OSC messages (see BELA console for precise adress)
For each harmonics (%rang indicate harmonic number, starting at 0) :
vol%rang : General Volume (vol0 control the volume of the fundamental)
A%rang : Attack
D%rang : Decay
S%rang : Sustain
R%rang : Release
/////////////////////////////////////////////////////////////////////////////////////////////////
GENERAL
pitchwheel
UI | import("stdfaust.lib");
midigate = button("gate");
midifreq = nentry("freq[unit:Hz]", 440, 20, 20000, 1);
midigain = nentry("gain", 0.5, 0, 10, 0.01);
pitchwheel = hslider("bend [midi:pitchwheel]",1,0.001,10,0.01);
gFreq = midifreq * pitchwheel;
partiel(rang) = os.oscrs(gFreq*(rang+1))*volume
with {
vol = hslider("vol%rang", 1, 0, 1, 0.001);
a = 0.01 * hslider("A%rang", 1, 0, 400, 0.001);
d = 0.01 * hslider("D%rang", 1, 0, 400, 0.001);
s = hslider("S%rang", 1, 0, 1, 0.001);
r = 0.01 * hslider("R%rang", 1, 0, 800, 0.001);
volume = ((en.adsr(a,d,s,r,midigate))*vol) : max (0) : min (1);
};
process = par(i, 8, partiel(i)) :> / (8);
|
7ef7f76dbf84659fbee18748855df1e00f395518866446105a21c70a70406f9d | tonal-glyph/faustus | AdditiveSynth_Analog.dsp | import("stdfaust.lib");
///////////////////////////////////////////////////////////////////////////////////////////////////
//
// Additive synthesizer, must be used with OSC message to program sound.
// It as 8 harmonics. Each have it's own volume envelope.
//
///////////////////////////////////////////////////////////////////////////////////////////////////
// ANALOG IMPLEMENTATION:
//
// ANALOG_0 : vol0 (volum of fundamental)
// ANALOG_1 : vol1
// ...
// ANALOG_7 : vol7
//
// OSC messages (see BELA console for precise adress)
// For each harmonics (%rang indicate harmonic number, starting at 0) :
// A%rang : Attack
// D%rang : Decay
// S%rang : Sustain
// R%rang : Release
//
///////////////////////////////////////////////////////////////////////////////////////////////////
// GENERAL
midigate = button("gate");
midifreq = nentry("freq[unit:Hz]", 440, 20, 20000, 1);
midigain = nentry("gain", 0.5, 0, 10, 0.01);
// pitchwheel
pitchwheel = hslider("bend [midi:pitchwheel]",1,0.001,10,0.01);
gFreq = midifreq * pitchwheel;
partiel(rang) = os.oscrs(gFreq*(rang+1))*volume
with {
// UI
vol = hslider("vol%rang[BELA: ANALOG_%rang]", 1, 0, 1, 0.001);
a = 0.01 * hslider("A%rang", 1, 0, 400, 0.001);
d = 0.01 * hslider("D%rang", 1, 0, 400, 0.001);
s = hslider("S%rang", 1, 0, 1, 0.001);
r = 0.01 * hslider("R%rang", 1, 0, 800, 0.001);
volume = ((en.adsr(a,d,s,r,midigate))*vol) : max (0) : min (1);
};
process = par(i, 8, partiel(i)) :> / (8);
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/bela/AdditiveSynth_Analog.dsp | faust | /////////////////////////////////////////////////////////////////////////////////////////////////
Additive synthesizer, must be used with OSC message to program sound.
It as 8 harmonics. Each have it's own volume envelope.
/////////////////////////////////////////////////////////////////////////////////////////////////
ANALOG IMPLEMENTATION:
ANALOG_0 : vol0 (volum of fundamental)
ANALOG_1 : vol1
...
ANALOG_7 : vol7
OSC messages (see BELA console for precise adress)
For each harmonics (%rang indicate harmonic number, starting at 0) :
A%rang : Attack
D%rang : Decay
S%rang : Sustain
R%rang : Release
/////////////////////////////////////////////////////////////////////////////////////////////////
GENERAL
pitchwheel
UI | import("stdfaust.lib");
midigate = button("gate");
midifreq = nentry("freq[unit:Hz]", 440, 20, 20000, 1);
midigain = nentry("gain", 0.5, 0, 10, 0.01);
pitchwheel = hslider("bend [midi:pitchwheel]",1,0.001,10,0.01);
gFreq = midifreq * pitchwheel;
partiel(rang) = os.oscrs(gFreq*(rang+1))*volume
with {
vol = hslider("vol%rang[BELA: ANALOG_%rang]", 1, 0, 1, 0.001);
a = 0.01 * hslider("A%rang", 1, 0, 400, 0.001);
d = 0.01 * hslider("D%rang", 1, 0, 400, 0.001);
s = hslider("S%rang", 1, 0, 1, 0.001);
r = 0.01 * hslider("R%rang", 1, 0, 800, 0.001);
volume = ((en.adsr(a,d,s,r,midigate))*vol) : max (0) : min (1);
};
process = par(i, 8, partiel(i)) :> / (8);
|
4e90a725d62a7be47d6398e5da457b856699e4f19a94ad75de555af15a80978c | tonal-glyph/faustus | chorus.dsp | import("stdfaust.lib");
import("layout2.dsp");
voices = 8; // MUST BE EVEN
process = ba.bypass1to2(cbp,chorus_mono(dmax,curdel,rate,sigma,do2,voices));
dmax = 8192;
curdel = dmax * ckg(vslider("[0] Delay [midi:ctrl 55] [style:knob]", 0.5, 0, 1, 1)) : si.smooth(0.999);
rateMax = 7.0; // Hz
rateMin = 0.01;
rateT60 = 0.15661;
rate = ckg(vslider("[1] Rate [midi:ctrl 56] [unit:Hz] [style:knob]", 0.5, rateMin, rateMax, 0.0001))
: si.smooth(ba.tau2pole(rateT60/6.91));
depth = ckg(vslider("[4] Depth [midi:ctrl 57] [style:knob]", 0.5, 0, 1, 0.001)) : si.smooth(ba.tau2pole(depthT60/6.91));
depthT60 = 0.15661;
delayPerVoice = 0.5*curdel/voices;
sigma = delayPerVoice * ckg(vslider("[6] Deviation [midi:ctrl 58] [style:knob]",0.5,0,1,0.001)) : si.smooth(0.999);
periodic = 1;
do2 = depth; // use when depth=1 means "multivibrato" effect (no original => all are modulated)
cbp = 1-int(csg(vslider("[0] Enable [midi:ctrl 103][style:knob]",0,0,1,1)));
chorus_mono(dmax,curdel,rate,sigma,do2,voices)
= _ <: (*(1-do2)<:_,_),(*(do2) <: par(i,voices,voice(i)) :> _,_) : ro.interleave(2,2) : +,+
with {
angle(i) = 2*ma.PI*(i/2)/voices + (i%2)*ma.PI/2;
voice(i) = de.fdelay(dmax,min(dmax,del(i))) * cos(angle(i));
del(i) = curdel*(i+1)/voices + dev(i);
rates(i) = rate/float(i+1);
dev(i) = sigma * os.oscp(rates(i),i*2*ma.PI/voices);
};
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/SAM/effects/chorus.dsp | faust | MUST BE EVEN
Hz
use when depth=1 means "multivibrato" effect (no original => all are modulated) | import("stdfaust.lib");
import("layout2.dsp");
process = ba.bypass1to2(cbp,chorus_mono(dmax,curdel,rate,sigma,do2,voices));
dmax = 8192;
curdel = dmax * ckg(vslider("[0] Delay [midi:ctrl 55] [style:knob]", 0.5, 0, 1, 1)) : si.smooth(0.999);
rateMin = 0.01;
rateT60 = 0.15661;
rate = ckg(vslider("[1] Rate [midi:ctrl 56] [unit:Hz] [style:knob]", 0.5, rateMin, rateMax, 0.0001))
: si.smooth(ba.tau2pole(rateT60/6.91));
depth = ckg(vslider("[4] Depth [midi:ctrl 57] [style:knob]", 0.5, 0, 1, 0.001)) : si.smooth(ba.tau2pole(depthT60/6.91));
depthT60 = 0.15661;
delayPerVoice = 0.5*curdel/voices;
sigma = delayPerVoice * ckg(vslider("[6] Deviation [midi:ctrl 58] [style:knob]",0.5,0,1,0.001)) : si.smooth(0.999);
periodic = 1;
cbp = 1-int(csg(vslider("[0] Enable [midi:ctrl 103][style:knob]",0,0,1,1)));
chorus_mono(dmax,curdel,rate,sigma,do2,voices)
= _ <: (*(1-do2)<:_,_),(*(do2) <: par(i,voices,voice(i)) :> _,_) : ro.interleave(2,2) : +,+
with {
angle(i) = 2*ma.PI*(i/2)/voices + (i%2)*ma.PI/2;
voice(i) = de.fdelay(dmax,min(dmax,del(i))) * cos(angle(i));
del(i) = curdel*(i+1)/voices + dev(i);
rates(i) = rate/float(i+1);
dev(i) = sigma * os.oscp(rates(i),i*2*ma.PI/voices);
};
|
6688003c78a23f044d494ab5b0ca3ca6908beb2d10c67ba1d17d0049f3ad7df0 | tonal-glyph/faustus | chorus.dsp | import("stdfaust.lib");
import("layout2.dsp");
voices = 8; // MUST BE EVEN
process = ba.bypass1to2(cbp,chorus_mono(dmax,curdel,rate,sigma,do2,voices));
dmax = 8192;
curdel = dmax * ckg(vslider("[0] Delay [midi:ctrl 4] [style:knob]", 0.5, 0, 1, 1)) : si.smooth(0.999);
rateMax = 7.0; // Hz
rateMin = 0.01;
rateT60 = 0.15661;
rate = ckg(vslider("[1] Rate [midi:ctrl 2] [unit:Hz] [style:knob]", 0.5, rateMin, rateMax, 0.0001))
: si.smooth(ba.tau2pole(rateT60/6.91));
depth = ckg(vslider("[4] Depth [midi:ctrl 3] [style:knob]", 0.5, 0, 1, 0.001)) : si.smooth(ba.tau2pole(depthT60/6.91));
depthT60 = 0.15661;
delayPerVoice = 0.5*curdel/voices;
sigma = delayPerVoice * ckg(vslider("[6] Deviation [midi:ctrl 58] [style:knob]",0.5,0,1,0.001)) : si.smooth(0.999);
periodic = 1;
do2 = depth; // use when depth=1 means "multivibrato" effect (no original => all are modulated)
cbp = 1-int(csg(vslider("[0] Enable [midi:ctrl 105][style:knob]",0,0,1,1)));
chorus_mono(dmax,curdel,rate,sigma,do2,voices)
= _ <: (*(1-do2)<:_,_),(*(do2) <: par(i,voices,voice(i)) :> _,_) : ro.interleave(2,2) : +,+
with {
angle(i) = 2*ma.PI*(i/2)/voices + (i%2)*ma.PI/2;
voice(i) = de.fdelay(dmax,min(dmax,del(i))) * cos(angle(i));
del(i) = curdel*(i+1)/voices + dev(i);
rates(i) = rate/float(i+1);
dev(i) = sigma * os.oscp(rates(i),i*2*ma.PI/voices);
};
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/SAM/chorus/chorus.dsp | faust | MUST BE EVEN
Hz
use when depth=1 means "multivibrato" effect (no original => all are modulated) | import("stdfaust.lib");
import("layout2.dsp");
process = ba.bypass1to2(cbp,chorus_mono(dmax,curdel,rate,sigma,do2,voices));
dmax = 8192;
curdel = dmax * ckg(vslider("[0] Delay [midi:ctrl 4] [style:knob]", 0.5, 0, 1, 1)) : si.smooth(0.999);
rateMin = 0.01;
rateT60 = 0.15661;
rate = ckg(vslider("[1] Rate [midi:ctrl 2] [unit:Hz] [style:knob]", 0.5, rateMin, rateMax, 0.0001))
: si.smooth(ba.tau2pole(rateT60/6.91));
depth = ckg(vslider("[4] Depth [midi:ctrl 3] [style:knob]", 0.5, 0, 1, 0.001)) : si.smooth(ba.tau2pole(depthT60/6.91));
depthT60 = 0.15661;
delayPerVoice = 0.5*curdel/voices;
sigma = delayPerVoice * ckg(vslider("[6] Deviation [midi:ctrl 58] [style:knob]",0.5,0,1,0.001)) : si.smooth(0.999);
periodic = 1;
cbp = 1-int(csg(vslider("[0] Enable [midi:ctrl 105][style:knob]",0,0,1,1)));
chorus_mono(dmax,curdel,rate,sigma,do2,voices)
= _ <: (*(1-do2)<:_,_),(*(do2) <: par(i,voices,voice(i)) :> _,_) : ro.interleave(2,2) : +,+
with {
angle(i) = 2*ma.PI*(i/2)/voices + (i%2)*ma.PI/2;
voice(i) = de.fdelay(dmax,min(dmax,del(i))) * cos(angle(i));
del(i) = curdel*(i+1)/voices + dev(i);
rates(i) = rate/float(i+1);
dev(i) = sigma * os.oscp(rates(i),i*2*ma.PI/voices);
};
|
99446d214604d52e4bd98b1da99d37f4b0b293ef28f2a574d3318b2f8292bbf4 | tonal-glyph/faustus | trumpet.dsp | //################################### trumpet.dsp #####################################
// A simple trumpet app... (for large screens).
//
// ## Compilation Instructions
//
// This Faust code will compile fine with any of the standard Faust targets. However
// it was specifically designed to be used with `faust2smartkeyb`. For best results,
// we recommend to use the following parameters to compile it:
//
// ```
// faust2smartkeyb [-ios/-android] -effect reverb.dsp trumpet.dsp
// ```
//
// ## Version/Licence
//
// Version 0.0, Feb. 2017
// Copyright Romain Michon CCRMA (Stanford University)/GRAME 2017
// MIT Licence: https://opensource.org/licenses/MIT
//########################################################################################
import("stdfaust.lib");
declare interface "SmartKeyboard{
'Number of Keyboards':'5',
'Max Keyboard Polyphony':'1',
'Mono Mode':'1',
'Keyboard 0 - Number of Keys':'13',
'Keyboard 1 - Number of Keys':'13',
'Keyboard 2 - Number of Keys':'13',
'Keyboard 3 - Number of Keys':'13',
'Keyboard 4 - Number of Keys':'13',
'Keyboard 0 - Lowest Key':'77',
'Keyboard 1 - Lowest Key':'72',
'Keyboard 2 - Lowest Key':'67',
'Keyboard 3 - Lowest Key':'62',
'Keyboard 4 - Lowest Key':'57',
'Rounding Mode':'2',
'Keyboard 0 - Send Y':'1',
'Keyboard 1 - Send Y':'1',
'Keyboard 2 - Send Y':'1',
'Keyboard 3 - Send Y':'1',
'Keyboard 4 - Send Y':'1',
}";
// standard parameters
f = hslider("freq",300,50,2000,0.01);
bend = hslider("bend[midi:pitchwheel]",1,0,10,0.01) : si.polySmooth(gate,0.999,1);
gain = hslider("gain",1,0,1,0.01);
s = hslider("sustain[midi:ctrl 64]",0,0,1,1); // for sustain pedal
t = button("gate");
y = hslider("y[midi:ctrl 1]",1,0,1,0.001) : si.smoo;
// fomating parameters
gate = t+s : min(1);
freq = f*bend;
cutoff = y*4000+50;
envelope = gate*gain : si.smoo;
process = os.sawtooth(freq)*envelope : fi.lowpass(3,cutoff) <: _,_;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/smartKeyboard/trumpet.dsp | faust | ################################### trumpet.dsp #####################################
A simple trumpet app... (for large screens).
## Compilation Instructions
This Faust code will compile fine with any of the standard Faust targets. However
it was specifically designed to be used with `faust2smartkeyb`. For best results,
we recommend to use the following parameters to compile it:
```
faust2smartkeyb [-ios/-android] -effect reverb.dsp trumpet.dsp
```
## Version/Licence
Version 0.0, Feb. 2017
Copyright Romain Michon CCRMA (Stanford University)/GRAME 2017
MIT Licence: https://opensource.org/licenses/MIT
########################################################################################
standard parameters
for sustain pedal
fomating parameters |
import("stdfaust.lib");
declare interface "SmartKeyboard{
'Number of Keyboards':'5',
'Max Keyboard Polyphony':'1',
'Mono Mode':'1',
'Keyboard 0 - Number of Keys':'13',
'Keyboard 1 - Number of Keys':'13',
'Keyboard 2 - Number of Keys':'13',
'Keyboard 3 - Number of Keys':'13',
'Keyboard 4 - Number of Keys':'13',
'Keyboard 0 - Lowest Key':'77',
'Keyboard 1 - Lowest Key':'72',
'Keyboard 2 - Lowest Key':'67',
'Keyboard 3 - Lowest Key':'62',
'Keyboard 4 - Lowest Key':'57',
'Rounding Mode':'2',
'Keyboard 0 - Send Y':'1',
'Keyboard 1 - Send Y':'1',
'Keyboard 2 - Send Y':'1',
'Keyboard 3 - Send Y':'1',
'Keyboard 4 - Send Y':'1',
}";
f = hslider("freq",300,50,2000,0.01);
bend = hslider("bend[midi:pitchwheel]",1,0,10,0.01) : si.polySmooth(gate,0.999,1);
gain = hslider("gain",1,0,1,0.01);
t = button("gate");
y = hslider("y[midi:ctrl 1]",1,0,1,0.001) : si.smoo;
gate = t+s : min(1);
freq = f*bend;
cutoff = y*4000+50;
envelope = gate*gain : si.smoo;
process = os.sawtooth(freq)*envelope : fi.lowpass(3,cutoff) <: _,_;
|
859230cb58be7636f486769bbba2f9d9f2c4353f1ca18579ed90d677230d62e0 | tonal-glyph/faustus | flanger.dsp | // Created from flange.dsp 2015/06/21
import("stdfaust.lib");
import("layout2.dsp");
flanger_mono(dmax,curdel,depth,fb,invert,lfoshape)
= _ <: _, (-:de.fdelay(dmax,curdel)) ~ *(fb) : _,
*(select2(invert,depth,0-depth))
: + : *(1/(1+depth)); // ideal for dc and reinforced sinusoids (in-phase summed signals)
process = ba.bypass1(fbp,flanger_mono_gui);
// Kill the groups to save vertical space:
meter_group(x) = flsg(x);
ctl_group(x) = flkg(x);
del_group(x) = flkg(x);
lvl_group(x) = flkf(x);
flangeview = lfo(freq);
flanger_mono_gui = attach(flangeview) : flanger_mono(dmax,curdel,depth,fb,invert,lfoshape);
sinlfo(freq) = (1 + os.oscrs(freq))/2;
trilfo(freq) = 1.0-abs(os.saw1(freq));
lfo(f) = (lfoshape * trilfo(f)) + ((1-lfoshape) * sinlfo(f));
dmax = 2048;
odflange = 44; // ~1 ms at 44.1 kHz = min delay
dflange = ((dmax-1)-odflange)*del_group(vslider("[1] Delay [midi:ctrl 50][style:knob]", 0.22, 0, 1, 1));
freq = ctl_group(vslider("[1] Rate [midi:ctrl 2] [unit:Hz] [style:knob]", 0.5, 0, 10, 0.01)) : si.smooth(ba.tau2pole(freqT60/6.91));
freqT60 = 0.15661;
depth = ctl_group(vslider("[3] Depth [midi:ctrl 3] [style:knob]", .75, 0, 1, 0.001)) : si.smooth(ba.tau2pole(depthT60/6.91));
depthT60 = 0.15661;
fb = ctl_group(vslider("[5] Feedback [midi:ctrl 4] [style:knob]", 0, -0.995, 0.99, 0.001)) : si.smooth(ba.tau2pole(fbT60/6.91));
fbT60 = 0.15661;
lfoshape = ctl_group(vslider("[7] Waveshape [midi:ctrl 54] [style:knob]", 0, 0, 1, 0.001));
curdel = odflange+dflange*lfo(freq);
fbp = 1-int(flsg(vslider("[0] Enable [midi:ctrl 105][style:knob]",0,0,1,1)));
invert = flsg(vslider("[1] Invert [midi:ctrl 49][style:knob]",0,0,1,1):int);
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/SAM/flanger/flanger.dsp | faust | Created from flange.dsp 2015/06/21
ideal for dc and reinforced sinusoids (in-phase summed signals)
Kill the groups to save vertical space:
~1 ms at 44.1 kHz = min delay |
import("stdfaust.lib");
import("layout2.dsp");
flanger_mono(dmax,curdel,depth,fb,invert,lfoshape)
= _ <: _, (-:de.fdelay(dmax,curdel)) ~ *(fb) : _,
*(select2(invert,depth,0-depth))
process = ba.bypass1(fbp,flanger_mono_gui);
meter_group(x) = flsg(x);
ctl_group(x) = flkg(x);
del_group(x) = flkg(x);
lvl_group(x) = flkf(x);
flangeview = lfo(freq);
flanger_mono_gui = attach(flangeview) : flanger_mono(dmax,curdel,depth,fb,invert,lfoshape);
sinlfo(freq) = (1 + os.oscrs(freq))/2;
trilfo(freq) = 1.0-abs(os.saw1(freq));
lfo(f) = (lfoshape * trilfo(f)) + ((1-lfoshape) * sinlfo(f));
dmax = 2048;
dflange = ((dmax-1)-odflange)*del_group(vslider("[1] Delay [midi:ctrl 50][style:knob]", 0.22, 0, 1, 1));
freq = ctl_group(vslider("[1] Rate [midi:ctrl 2] [unit:Hz] [style:knob]", 0.5, 0, 10, 0.01)) : si.smooth(ba.tau2pole(freqT60/6.91));
freqT60 = 0.15661;
depth = ctl_group(vslider("[3] Depth [midi:ctrl 3] [style:knob]", .75, 0, 1, 0.001)) : si.smooth(ba.tau2pole(depthT60/6.91));
depthT60 = 0.15661;
fb = ctl_group(vslider("[5] Feedback [midi:ctrl 4] [style:knob]", 0, -0.995, 0.99, 0.001)) : si.smooth(ba.tau2pole(fbT60/6.91));
fbT60 = 0.15661;
lfoshape = ctl_group(vslider("[7] Waveshape [midi:ctrl 54] [style:knob]", 0, 0, 1, 0.001));
curdel = odflange+dflange*lfo(freq);
fbp = 1-int(flsg(vslider("[0] Enable [midi:ctrl 105][style:knob]",0,0,1,1)));
invert = flsg(vslider("[1] Invert [midi:ctrl 49][style:knob]",0,0,1,1):int);
|
ae7ab641ed7e9e8549d7409c1a121d43c996aaea2fa8189a729a52989b42f6a2 | tonal-glyph/faustus | granulator.dsp | // FROM FAUST DEMO
// Designed to use the Analog Input for parameters contrôles.
//
///////////////////////////////////////////////////////////////////////////////////////////////////
//
// 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);
// la frequence donne la largeur de bande extraite du bruit blanc
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 //////////////////////////////
//Le pulsar permet de creer une 'pulsation' plus ou moins aleatoire (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 pour eviter division par 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/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/bela/granulator.dsp | faust | FROM FAUST DEMO
Designed to use the Analog Input for parameters contrôles.
/////////////////////////////////////////////////////////////////////////////////////////////////
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
la frequence donne la largeur de bande extraite du bruit blanc
maximum P (for de.delay-line allocation)
PHASOR_BIN //////////////////////////////
PULSAR //////////////////////////////
Le pulsar permet de creer une 'pulsation' plus ou moins aleatoire (proba).
min > 0 pour eviter division par 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);
|
f9847c4cdf74a9566e90701febd338bb5d35ffdb9c30f93c97747f2386a23c24 | tonal-glyph/faustus | WaveSynth_Analog.dsp | import("stdfaust.lib");
///////////////////////////////////////////////////////////////////////////////////////////////////
//
// Simple demo of wavetable synthesis. A LFO modulate the interpolation between 4 tables.
// It's possible to add more tables step.
//
///////////////////////////////////////////////////////////////////////////////////////////////////
// ANALOG IMPLEMENTATION:
//
// ANALOG_0 : Wave travelling
// ANALOG_1 : LFO Frequency
// ANALOG_2 : LFO Depth (wave travel modulation)
// ANALOG_3 : Release
//
// MIDI:
// CC 73 : Attack
// CC 76 : Decay
// CC 77 : Sustain
//
///////////////////////////////////////////////////////////////////////////////////////////////////
// GENERAL
midigate = button ("gate");
midifreq = nentry("freq[unit:Hz]", 440, 20, 20000, 1);
midigain = nentry("gain", 0.5, 0, 1, 0.01);
waveTravel = hslider("waveTravel[BELA: ANALOG_0]",0,0,1,0.01);
// pitchwheel
pitchwheel = hslider("bend [midi:pitchwheel]",1,0.001,10,0.01);
gFreq = midifreq * pitchwheel;
// LFO
lfoDepth = hslider("lfoDepth[BELA: ANALOG_2]",0,0.,1,0.001):si.smoo;
lfoFreq = hslider("lfoFreq[BELA: ANALOG_1]",0.1,0.01,10,0.001):si.smoo;
moov = ((os.lf_trianglepos(lfoFreq) * lfoDepth) + waveTravel) : min(1) : max(0);
volA = hslider("A[midi:ctrl 73]",0.01,0.01,4,0.01);
volD = hslider("D[midi:ctrl 76]",0.6,0.01,8,0.01);
volS = hslider("S[midi:ctrl 77]",0.2,0,1,0.01);
volR = hslider("R[BELA: ANALOG_3]",0.8,0.01,8,0.01);
envelop = en.adsre(volA,volD,volS,volR,midigate);
// Out Amplitude
vol = envelop * midigain;
WF(tablesize, rang) = abs((fmod ((1+(float(ba.time)*rang)/float(tablesize)), 4.0 ))-2) -1.;
// 4 WF maxi with this version:
scanner(nb, position) = -(_,soustraction) : *(_,coef) : cos : max(0)
with {
coef = 3.14159 * ((nb-1)*0.5);
soustraction = select2( position>0, 0, (position/(nb-1)) );
};
wfosc(freq) = (rdtable(tablesize, wt1, faze)*(moov : scanner(4,0)))+(rdtable(tablesize, wt2, faze)*(moov : scanner(4,1)))
+ (rdtable(tablesize, wt3, faze)*(moov : scanner(4,2)))+(rdtable(tablesize, wt4, faze)*(moov : scanner(4,3)))
with {
tablesize = 1024;
wt1 = WF(tablesize, 16);
wt2 = WF(tablesize, 8);
wt3 = WF(tablesize, 6);
wt4 = WF(tablesize, 4);
faze = int(os.phasor(tablesize,freq));
};
process = wfosc(gFreq) * vol;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/bela/WaveSynth_Analog.dsp | faust | /////////////////////////////////////////////////////////////////////////////////////////////////
Simple demo of wavetable synthesis. A LFO modulate the interpolation between 4 tables.
It's possible to add more tables step.
/////////////////////////////////////////////////////////////////////////////////////////////////
ANALOG IMPLEMENTATION:
ANALOG_0 : Wave travelling
ANALOG_1 : LFO Frequency
ANALOG_2 : LFO Depth (wave travel modulation)
ANALOG_3 : Release
MIDI:
CC 73 : Attack
CC 76 : Decay
CC 77 : Sustain
/////////////////////////////////////////////////////////////////////////////////////////////////
GENERAL
pitchwheel
LFO
Out Amplitude
4 WF maxi with this version: | import("stdfaust.lib");
midigate = button ("gate");
midifreq = nentry("freq[unit:Hz]", 440, 20, 20000, 1);
midigain = nentry("gain", 0.5, 0, 1, 0.01);
waveTravel = hslider("waveTravel[BELA: ANALOG_0]",0,0,1,0.01);
pitchwheel = hslider("bend [midi:pitchwheel]",1,0.001,10,0.01);
gFreq = midifreq * pitchwheel;
lfoDepth = hslider("lfoDepth[BELA: ANALOG_2]",0,0.,1,0.001):si.smoo;
lfoFreq = hslider("lfoFreq[BELA: ANALOG_1]",0.1,0.01,10,0.001):si.smoo;
moov = ((os.lf_trianglepos(lfoFreq) * lfoDepth) + waveTravel) : min(1) : max(0);
volA = hslider("A[midi:ctrl 73]",0.01,0.01,4,0.01);
volD = hslider("D[midi:ctrl 76]",0.6,0.01,8,0.01);
volS = hslider("S[midi:ctrl 77]",0.2,0,1,0.01);
volR = hslider("R[BELA: ANALOG_3]",0.8,0.01,8,0.01);
envelop = en.adsre(volA,volD,volS,volR,midigate);
vol = envelop * midigain;
WF(tablesize, rang) = abs((fmod ((1+(float(ba.time)*rang)/float(tablesize)), 4.0 ))-2) -1.;
scanner(nb, position) = -(_,soustraction) : *(_,coef) : cos : max(0)
with {
coef = 3.14159 * ((nb-1)*0.5);
soustraction = select2( position>0, 0, (position/(nb-1)) );
};
wfosc(freq) = (rdtable(tablesize, wt1, faze)*(moov : scanner(4,0)))+(rdtable(tablesize, wt2, faze)*(moov : scanner(4,1)))
+ (rdtable(tablesize, wt3, faze)*(moov : scanner(4,2)))+(rdtable(tablesize, wt4, faze)*(moov : scanner(4,3)))
with {
tablesize = 1024;
wt1 = WF(tablesize, 16);
wt2 = WF(tablesize, 8);
wt3 = WF(tablesize, 6);
wt4 = WF(tablesize, 4);
faze = int(os.phasor(tablesize,freq));
};
process = wfosc(gFreq) * vol;
|
1484a867ae6ccb43b580d12ade65f11fd83be63bf04313dee3734daa5a58afca | tonal-glyph/faustus | flanger.dsp | // Created from flange.dsp 2015/06/21
import("stdfaust.lib");
import("layout2.dsp");
flanger_mono(dmax,curdel,depth,fb,invert,lfoshape)
= _ <: _, (-:de.fdelay(dmax,curdel)) ~ *(fb) : _,
*(select2(invert,depth,0-depth))
: + : *(1/(1+depth)); // ideal for dc and reinforced sinusoids (in-phase summed signals)
process = ba.bypass1(fbp,flanger_mono_gui);
// Kill the groups to save vertical space:
meter_group(x) = flsg(x);
ctl_group(x) = flkg(x);
del_group(x) = flkg(x);
lvl_group(x) = flkf(x);
flangeview = lfo(freq);
flanger_mono_gui = attach(flangeview) : flanger_mono(dmax,curdel,depth,fb,invert,lfoshape);
sinlfo(freq) = (1 + os.oscrs(freq))/2;
trilfo(freq) = 1.0-abs(os.saw1(freq));
lfo(f) = (lfoshape * trilfo(f)) + ((1-lfoshape) * sinlfo(f));
dmax = 2048;
odflange = 44; // ~1 ms at 44.1 kHz = min delay
dflange = ((dmax-1)-odflange)*del_group(vslider("[1] Delay [midi:ctrl 50][style:knob]", 0.22, 0, 1, 1));
freq = ctl_group(vslider("[1] Rate [midi:ctrl 51] [unit:Hz] [style:knob]", 0.5, 0, 10, 0.01)) : si.smooth(ba.tau2pole(freqT60/6.91));
freqT60 = 0.15661;
depth = ctl_group(vslider("[3] Depth [midi:ctrl 52] [style:knob]", .75, 0, 1, 0.001)) : si.smooth(ba.tau2pole(depthT60/6.91));
depthT60 = 0.15661;
fb = ctl_group(vslider("[5] Feedback [midi:ctrl 53] [style:knob]", 0, -0.995, 0.99, 0.001)) : si.smooth(ba.tau2pole(fbT60/6.91));
fbT60 = 0.15661;
lfoshape = ctl_group(vslider("[7] Waveshape [midi:ctrl 54] [style:knob]", 0, 0, 1, 0.001));
curdel = odflange+dflange*lfo(freq);
fbp = 1-int(flsg(vslider("[0] Enable [midi:ctrl 102][style:knob]",0,0,1,1)));
invert = flsg(vslider("[1] Invert [midi:ctrl 49][style:knob]",0,0,1,1):int);
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/SAM/effects/flanger.dsp | faust | Created from flange.dsp 2015/06/21
ideal for dc and reinforced sinusoids (in-phase summed signals)
Kill the groups to save vertical space:
~1 ms at 44.1 kHz = min delay |
import("stdfaust.lib");
import("layout2.dsp");
flanger_mono(dmax,curdel,depth,fb,invert,lfoshape)
= _ <: _, (-:de.fdelay(dmax,curdel)) ~ *(fb) : _,
*(select2(invert,depth,0-depth))
process = ba.bypass1(fbp,flanger_mono_gui);
meter_group(x) = flsg(x);
ctl_group(x) = flkg(x);
del_group(x) = flkg(x);
lvl_group(x) = flkf(x);
flangeview = lfo(freq);
flanger_mono_gui = attach(flangeview) : flanger_mono(dmax,curdel,depth,fb,invert,lfoshape);
sinlfo(freq) = (1 + os.oscrs(freq))/2;
trilfo(freq) = 1.0-abs(os.saw1(freq));
lfo(f) = (lfoshape * trilfo(f)) + ((1-lfoshape) * sinlfo(f));
dmax = 2048;
dflange = ((dmax-1)-odflange)*del_group(vslider("[1] Delay [midi:ctrl 50][style:knob]", 0.22, 0, 1, 1));
freq = ctl_group(vslider("[1] Rate [midi:ctrl 51] [unit:Hz] [style:knob]", 0.5, 0, 10, 0.01)) : si.smooth(ba.tau2pole(freqT60/6.91));
freqT60 = 0.15661;
depth = ctl_group(vslider("[3] Depth [midi:ctrl 52] [style:knob]", .75, 0, 1, 0.001)) : si.smooth(ba.tau2pole(depthT60/6.91));
depthT60 = 0.15661;
fb = ctl_group(vslider("[5] Feedback [midi:ctrl 53] [style:knob]", 0, -0.995, 0.99, 0.001)) : si.smooth(ba.tau2pole(fbT60/6.91));
fbT60 = 0.15661;
lfoshape = ctl_group(vslider("[7] Waveshape [midi:ctrl 54] [style:knob]", 0, 0, 1, 0.001));
curdel = odflange+dflange*lfo(freq);
fbp = 1-int(flsg(vslider("[0] Enable [midi:ctrl 102][style:knob]",0,0,1,1)));
invert = flsg(vslider("[1] Invert [midi:ctrl 49][style:knob]",0,0,1,1):int);
|
7dd062317e9c78f008804bb296ac26e1190fe3269c2852c8b526f1bd30496622 | tonal-glyph/faustus | simpleFX.dsp | import("stdfaust.lib");
//
///////////////////////////////////////////////////////////////////////////////////////////////////
//
// Simple FX chaine build for a mono synthesizer.
// It controle general volume and pan.
// FX Chaine is:
// Drive
// Flanger
// Reverberation
//
///////////////////////////////////////////////////////////////////////////////////////////////////
// MIDI IMPLEMENTATION:
// (All are available by OSC)
//
// CC 7 : Volume
// CC 10 : Pan
//
// CC 92 : Distortion Drive
//
// CC 13 : Flanger Delay
// CC 93 : Flanger Dry/Wet
// CC 94 : Flanger Feedback
//
// CC 12 : Reverberation Room size
// CC 91 : Reverberation Dry/Wet
// CC 95 : Reverberation Damp
// CC 90 : Reverberation Stereo Width
//
///////////////////////////////////////////////////////////////////////////////////////////////////
// VOLUME:
vol = hslider ("volume[midi:ctrl 7]",1,0,1,0.001);// Should be 7 according to MIDI CC norm.
// EFFECTS /////////////////////////////////////////////
drive = hslider ("drive[midi:ctrl 92]",0.3,0,1,0.001);
// Flanger
curdel = hslider ("flangDel[midi:ctrl 13]",4,0.001,10,0.001);
fb = hslider ("flangFeedback[midi:ctrl 94]",0.7,0,1,0.001);
fldw = hslider ("dryWetFlang[midi:ctrl 93]",0.5,0,1,0.001);
flanger = efx
with {
fldel = (curdel + (os.lf_triangle(1) * 2) ) : min(10);
efx = _ <: _, pf.flanger_mono(10,fldel,1,fb,0) : dry_wet(fldw);
};
// Panoramique:
panno = _ : sp.panner(hslider ("pan[midi:ctrl 10]",0.5,0,1,0.001)) : _,_;
// REVERB (from freeverb_demo)
reverb = _,_ <: (*(g)*fixedgain,*(g)*fixedgain :
re.stereo_freeverb(combfeed, allpassfeed, damping, spatSpread)),
*(1-g), *(1-g) :> _,_
with {
scaleroom = 0.28;
offsetroom = 0.7;
allpassfeed = 0.5;
scaledamp = 0.4;
fixedgain = 0.1;
origSR = 44100;
damping = vslider("Damp[midi:ctrl 95]",0.5, 0, 1, 0.025)*scaledamp*origSR/ma.SR;
combfeed = vslider("RoomSize[midi:ctrl 12]", 0.7, 0, 1, 0.025)*scaleroom*origSR/ma.SR + offsetroom;
spatSpread = vslider("Stereo[midi:ctrl 90]",0.6,0,1,0.01)*46*ma.SR/origSR;
g = vslider("dryWetReverb[midi:ctrl 91]", 0.4, 0, 1, 0.001);
// (g = Dry/Wet)
};
// Dry-Wet (from C. LEBRETON)
dry_wet(dw,x,y) = wet*y + dry*x
with {
wet = 0.5*(dw+1.0);
dry = 1.0-wet;
};
// ALL
effets = _ *(vol) : ef.cubicnl_nodc(drive, 0.1) : flanger : panno : reverb;
process = effets;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/bela/simpleFX.dsp | faust |
/////////////////////////////////////////////////////////////////////////////////////////////////
Simple FX chaine build for a mono synthesizer.
It controle general volume and pan.
FX Chaine is:
Drive
Flanger
Reverberation
/////////////////////////////////////////////////////////////////////////////////////////////////
MIDI IMPLEMENTATION:
(All are available by OSC)
CC 7 : Volume
CC 10 : Pan
CC 92 : Distortion Drive
CC 13 : Flanger Delay
CC 93 : Flanger Dry/Wet
CC 94 : Flanger Feedback
CC 12 : Reverberation Room size
CC 91 : Reverberation Dry/Wet
CC 95 : Reverberation Damp
CC 90 : Reverberation Stereo Width
/////////////////////////////////////////////////////////////////////////////////////////////////
VOLUME:
Should be 7 according to MIDI CC norm.
EFFECTS /////////////////////////////////////////////
Flanger
Panoramique:
REVERB (from freeverb_demo)
(g = Dry/Wet)
Dry-Wet (from C. LEBRETON)
ALL | import("stdfaust.lib");
drive = hslider ("drive[midi:ctrl 92]",0.3,0,1,0.001);
curdel = hslider ("flangDel[midi:ctrl 13]",4,0.001,10,0.001);
fb = hslider ("flangFeedback[midi:ctrl 94]",0.7,0,1,0.001);
fldw = hslider ("dryWetFlang[midi:ctrl 93]",0.5,0,1,0.001);
flanger = efx
with {
fldel = (curdel + (os.lf_triangle(1) * 2) ) : min(10);
efx = _ <: _, pf.flanger_mono(10,fldel,1,fb,0) : dry_wet(fldw);
};
panno = _ : sp.panner(hslider ("pan[midi:ctrl 10]",0.5,0,1,0.001)) : _,_;
reverb = _,_ <: (*(g)*fixedgain,*(g)*fixedgain :
re.stereo_freeverb(combfeed, allpassfeed, damping, spatSpread)),
*(1-g), *(1-g) :> _,_
with {
scaleroom = 0.28;
offsetroom = 0.7;
allpassfeed = 0.5;
scaledamp = 0.4;
fixedgain = 0.1;
origSR = 44100;
damping = vslider("Damp[midi:ctrl 95]",0.5, 0, 1, 0.025)*scaledamp*origSR/ma.SR;
combfeed = vslider("RoomSize[midi:ctrl 12]", 0.7, 0, 1, 0.025)*scaleroom*origSR/ma.SR + offsetroom;
spatSpread = vslider("Stereo[midi:ctrl 90]",0.6,0,1,0.01)*46*ma.SR/origSR;
g = vslider("dryWetReverb[midi:ctrl 91]", 0.4, 0, 1, 0.001);
};
dry_wet(dw,x,y) = wet*y + dry*x
with {
wet = 0.5*(dw+1.0);
dry = 1.0-wet;
};
effets = _ *(vol) : ef.cubicnl_nodc(drive, 0.1) : flanger : panno : reverb;
process = effets;
|
ed79d37b0bac3a0f18c44173772d7f9c369d22f17a2647357bd337bb04dea146 | tonal-glyph/faustus | WaveSynth.dsp | import("stdfaust.lib");
///////////////////////////////////////////////////////////////////////////////////////////////////
//
// Simple demo of wavetable synthesis. A LFO modulate the interpolation between 4 tables.
// It's possible to add more tables step.
//
///////////////////////////////////////////////////////////////////////////////////////////////////
// MIDI IMPLEMENTATION:
//
// CC 1 : LFO Depth (wave travel modulation)
// CC 14 : LFO Frequency
// CC 70 : Wave travelling
//
// CC 73 : Attack
// CC 76 : Decay
// CC 77 : Sustain
// CC 72 : Release
//
///////////////////////////////////////////////////////////////////////////////////////////////////
// GENERAL
midigate = button ("gate");
midifreq = nentry("freq[unit:Hz]", 440, 20, 20000, 1);
midigain = nentry("gain", 0.5, 0, 1, 0.01);
waveTravel = hslider("waveTravel [midi:ctrl ]",0,0,1,0.01);
// pitchwheel
pitchwheel = hslider("bend [midi:pitchwheel]",1,0.001,10,0.01);
gFreq = midifreq * pitchwheel;
// LFO
lfoDepth = hslider ("lfoDepth[midi:ctrl 1]",0,0.,1,0.001):si.smoo;
lfoFreq = hslider ("lfoFreq[midi:ctrl 14]",0.1,0.01,10,0.001):si.smoo;
moov = ((os.lf_trianglepos(lfoFreq) * lfoDepth) + waveTravel) : min(1) : max(0);
volA = hslider("A[midi:ctrl 73]",0.01,0.01,4,0.01);
volD = hslider("D[midi:ctrl 76]",0.6,0.01,8,0.01);
volS = hslider("S[midi:ctrl 77]",0.2,0,1,0.01);
volR = hslider("R[midi:ctrl 72]",0.8,0.01,8,0.01);
envelop = en.adsre(volA,volD,volS,volR,midigate);
// Out Amplitude
vol = envelop * midigain ;
WF(tablesize, rang) = abs((fmod ((1+(float(ba.time)*rang)/float(tablesize)), 4.0 ))-2) -1.;
// 4 WF maxi with this version:
scanner(nb, position) = -(_,soustraction) : *(_,coef) : cos : max(0)
with{
coef = 3.14159 * ((nb-1)*0.5);
soustraction = select2( position>0, 0, (position/(nb-1)) );
};
wfosc(freq) = (rdtable(tablesize, wt1, faze)*(moov : scanner(4,0)))+(rdtable(tablesize, wt2, faze)*(moov : scanner(4,1)))
+ (rdtable(tablesize, wt3, faze)*(moov : scanner(4,2)))+(rdtable(tablesize, wt4, faze)*(moov : scanner(4,3)))
with {
tablesize = 1024;
wt1 = WF(tablesize, 16);
wt2 = WF(tablesize, 8);
wt3 = WF(tablesize, 6);
wt4 = WF(tablesize, 4);
faze = int(os.phasor(tablesize,freq));
};
process = wfosc(gFreq) * vol;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/bela/WaveSynth.dsp | faust | /////////////////////////////////////////////////////////////////////////////////////////////////
Simple demo of wavetable synthesis. A LFO modulate the interpolation between 4 tables.
It's possible to add more tables step.
/////////////////////////////////////////////////////////////////////////////////////////////////
MIDI IMPLEMENTATION:
CC 1 : LFO Depth (wave travel modulation)
CC 14 : LFO Frequency
CC 70 : Wave travelling
CC 73 : Attack
CC 76 : Decay
CC 77 : Sustain
CC 72 : Release
/////////////////////////////////////////////////////////////////////////////////////////////////
GENERAL
pitchwheel
LFO
Out Amplitude
4 WF maxi with this version: | import("stdfaust.lib");
midigate = button ("gate");
midifreq = nentry("freq[unit:Hz]", 440, 20, 20000, 1);
midigain = nentry("gain", 0.5, 0, 1, 0.01);
waveTravel = hslider("waveTravel [midi:ctrl ]",0,0,1,0.01);
pitchwheel = hslider("bend [midi:pitchwheel]",1,0.001,10,0.01);
gFreq = midifreq * pitchwheel;
lfoDepth = hslider ("lfoDepth[midi:ctrl 1]",0,0.,1,0.001):si.smoo;
lfoFreq = hslider ("lfoFreq[midi:ctrl 14]",0.1,0.01,10,0.001):si.smoo;
moov = ((os.lf_trianglepos(lfoFreq) * lfoDepth) + waveTravel) : min(1) : max(0);
volA = hslider("A[midi:ctrl 73]",0.01,0.01,4,0.01);
volD = hslider("D[midi:ctrl 76]",0.6,0.01,8,0.01);
volS = hslider("S[midi:ctrl 77]",0.2,0,1,0.01);
volR = hslider("R[midi:ctrl 72]",0.8,0.01,8,0.01);
envelop = en.adsre(volA,volD,volS,volR,midigate);
vol = envelop * midigain ;
WF(tablesize, rang) = abs((fmod ((1+(float(ba.time)*rang)/float(tablesize)), 4.0 ))-2) -1.;
scanner(nb, position) = -(_,soustraction) : *(_,coef) : cos : max(0)
with{
coef = 3.14159 * ((nb-1)*0.5);
soustraction = select2( position>0, 0, (position/(nb-1)) );
};
wfosc(freq) = (rdtable(tablesize, wt1, faze)*(moov : scanner(4,0)))+(rdtable(tablesize, wt2, faze)*(moov : scanner(4,1)))
+ (rdtable(tablesize, wt3, faze)*(moov : scanner(4,2)))+(rdtable(tablesize, wt4, faze)*(moov : scanner(4,3)))
with {
tablesize = 1024;
wt1 = WF(tablesize, 16);
wt2 = WF(tablesize, 8);
wt3 = WF(tablesize, 6);
wt4 = WF(tablesize, 4);
faze = int(os.phasor(tablesize,freq));
};
process = wfosc(gFreq) * vol;
|
da2f7bfb69ad975035f8135cd7c98684be8834cbaca9c127f69afd0e7d022be8 | tonal-glyph/faustus | frog.dsp | //################################### frog.dsp #####################################
// A simple smart phone abstract instrument than can be controlled using the touch
// screen and the accelerometers of the device.
//
// ## `SmartKeyboard` Use Strategy
//
// The idea here is to use the `SmartKeyboard` interface as an X/Y control pad by just
// creating one keyboard with on key and by retrieving the X and Y position on that single
// key using the `x` and `y` standard parameters. Keyboard mode is deactivated so that
// the color of the pad doesn't change when it is pressed.
//
// ## Compilation Instructions
//
// This Faust code will compile fine with any of the standard Faust targets. However
// it was specifically designed to be used with `faust2smartkeyb`. For best results,
// we recommend to use the following parameters to compile it:
//
// ```
// faust2smartkeyb [-ios/-android] frog.dsp
// ```
//
// ## Version/Licence
//
// Version 0.0, Feb. 2017
// Copyright Romain Michon CCRMA (Stanford University)/GRAME 2017
// MIT Licence: https://opensource.org/licenses/MIT
//########################################################################################
declare name "frog";
import("stdfaust.lib");
//========================= Smart Keyboard Configuration =================================
// (1 keyboards with 1 key configured as a pad.
//========================================================================================
declare interface "SmartKeyboard{
'Number of Keyboards':'1',
'Keyboard 0 - Number of Keys':'1',
'Keyboard 0 - Piano Keyboard':'0',
'Keyboard 0 - Static Mode':'1',
'Keyboard 0 - Send X':'1',
'Keyboard 0 - Send Y':'1'
}";
//================================ Instrument Parameters =================================
// Creates the connection between the synth and the mobile device
//========================================================================================
// SmartKeyboard X parameter
x = hslider("x",0,0,1,0.01);
// SmartKeyboard Y parameter
y = hslider("y",0,0,1,0.01);
// SmartKeyboard gate parameter
gate = button("gate");
// the cutoff frequency of the filter is controlled with the x axis of the accelerometer
cutoff = hslider("cutoff[acc: 0 0 -10 0 10]",2500,50,5000,0.01);
//=================================== Parameters Mapping =================================
//========================================================================================
maxFreq = 100;
minFreq = 1;
freq = x*(maxFreq-minFreq) + minFreq : si.polySmooth(gate,0.999,1);
maxQ = 40;
minQ = 1;
q = (1-y)*(maxQ-minQ) + minQ : si.smoo;
filterCutoff = cutoff : si.smoo;
//============================================ DSP =======================================
//========================================================================================
process = sy.dubDub(freq,filterCutoff,q,gate) <: _,_;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/smartKeyboard/frog.dsp | faust | ################################### frog.dsp #####################################
A simple smart phone abstract instrument than can be controlled using the touch
screen and the accelerometers of the device.
## `SmartKeyboard` Use Strategy
The idea here is to use the `SmartKeyboard` interface as an X/Y control pad by just
creating one keyboard with on key and by retrieving the X and Y position on that single
key using the `x` and `y` standard parameters. Keyboard mode is deactivated so that
the color of the pad doesn't change when it is pressed.
## Compilation Instructions
This Faust code will compile fine with any of the standard Faust targets. However
it was specifically designed to be used with `faust2smartkeyb`. For best results,
we recommend to use the following parameters to compile it:
```
faust2smartkeyb [-ios/-android] frog.dsp
```
## Version/Licence
Version 0.0, Feb. 2017
Copyright Romain Michon CCRMA (Stanford University)/GRAME 2017
MIT Licence: https://opensource.org/licenses/MIT
########################################################################################
========================= Smart Keyboard Configuration =================================
(1 keyboards with 1 key configured as a pad.
========================================================================================
================================ Instrument Parameters =================================
Creates the connection between the synth and the mobile device
========================================================================================
SmartKeyboard X parameter
SmartKeyboard Y parameter
SmartKeyboard gate parameter
the cutoff frequency of the filter is controlled with the x axis of the accelerometer
=================================== Parameters Mapping =================================
========================================================================================
============================================ DSP =======================================
======================================================================================== |
declare name "frog";
import("stdfaust.lib");
declare interface "SmartKeyboard{
'Number of Keyboards':'1',
'Keyboard 0 - Number of Keys':'1',
'Keyboard 0 - Piano Keyboard':'0',
'Keyboard 0 - Static Mode':'1',
'Keyboard 0 - Send X':'1',
'Keyboard 0 - Send Y':'1'
}";
x = hslider("x",0,0,1,0.01);
y = hslider("y",0,0,1,0.01);
gate = button("gate");
cutoff = hslider("cutoff[acc: 0 0 -10 0 10]",2500,50,5000,0.01);
maxFreq = 100;
minFreq = 1;
freq = x*(maxFreq-minFreq) + minFreq : si.polySmooth(gate,0.999,1);
maxQ = 40;
minQ = 1;
q = (1-y)*(maxQ-minQ) + minQ : si.smoo;
filterCutoff = cutoff : si.smoo;
process = sy.dubDub(freq,filterCutoff,q,gate) <: _,_;
|
91a750d46aaf1669029d0959fba254836e436a3095d20ca8f95f8de4b7cb5dce | tonal-glyph/faustus | simpleFX_Analog.dsp | import("stdfaust.lib");
//
///////////////////////////////////////////////////////////////////////////////////////////////////
//
// Simple FX chaine build for a mono synthesizer.
// It controle general volume and pan.
// FX Chaine is:
// Drive
// Flanger
// Reverberation
//
// This version use ANALOG IN to controle some of the parameters.
// Other parameters continue to be available by MIDI or OSC.
//
///////////////////////////////////////////////////////////////////////////////////////////////////
// ANALOG IMPLEMENTATION:
//
// ANALOG_4 : Distortion Drive
// ANALOG_5 : Flanger Dry/Wet
// ANALOG_6 : Reverberation Dry/Wet
// ANALOG_7 : Reverberation Room size
//
// MIDI:
// CC 7 : Volume
// CC 10 : Pan
//
// CC 13 : Flanger Delay
// CC 13 : Flanger Delay
// CC 94 : Flanger Feedback
//
// CC 95 : Reverberation Damp
// CC 90: Reverberation Stereo Width
//
///////////////////////////////////////////////////////////////////////////////////////////////////
// VOLUME:
vol = hslider("volume[midi:ctrl 7]",1,0,1,0.001);// Should be 7 according to MIDI CC norm.
// EFFECTS /////////////////////////////////////////////
drive = hslider ("drive[BELA: ANALOG_4]",0.3,0,1,0.001);
// Flanger
curdel = hslider("flangDel[midi:ctrl 13]",4,0.001,10,0.001);
fb = hslider("flangFeedback[midi:ctrl 94]",0.7,0,1,0.001);
fldw = hslider("dryWetFlang[BELA: ANALOG_5]",0.5,0,1,0.001);
flanger = efx
with {
fldel = (curdel + (os.lf_triangle(1) * 2) ) : min(10);
efx = _ <: _, pf.flanger_mono(10,fldel,1,fb,0) : dry_wet(fldw);
};
// Pannoramique:
panno = _ : sp.panner(hslider ("pan[midi:ctrl 10]",0.5,0,1,0.001)) : _,_;
// REVERB (from freeverb_demo)
reverb = _,_ <: (*(g)*fixedgain,*(g)*fixedgain :
re.stereo_freeverb(combfeed, allpassfeed, damping, spatSpread)),
*(1-g), *(1-g) :> _,_
with {
scaleroom = 0.28;
offsetroom = 0.7;
allpassfeed = 0.5;
scaledamp = 0.4;
fixedgain = 0.1;
origSR = 44100;
damping = vslider("Damp[midi:ctrl 95]",0.5, 0, 1, 0.025)*scaledamp*origSR/ma.SR;
combfeed = vslider("RoomSize[BELA: ANALOG_7]", 0.7, 0, 1, 0.025)*scaleroom*origSR/ma.SR + offsetroom;
spatSpread = vslider("Stereo[midi:ctrl 90]",0.6,0,1,0.01)*46*ma.SR/origSR;
g = vslider("dryWetReverb[BELA: ANALOG_6]", 0.4, 0, 1, 0.001);
// (g = Dry/Wet)
};
// Dry-Wet (from C. LEBRETON)
dry_wet(dw,x,y) = wet*y + dry*x
with {
wet = 0.5*(dw+1.0);
dry = 1.0-wet;
};
// ALL
effets = _ *(vol) : ef.cubicnl_nodc(drive, 0.1) : flanger : panno : reverb;
process = effets;
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/bela/simpleFX_Analog.dsp | faust |
/////////////////////////////////////////////////////////////////////////////////////////////////
Simple FX chaine build for a mono synthesizer.
It controle general volume and pan.
FX Chaine is:
Drive
Flanger
Reverberation
This version use ANALOG IN to controle some of the parameters.
Other parameters continue to be available by MIDI or OSC.
/////////////////////////////////////////////////////////////////////////////////////////////////
ANALOG IMPLEMENTATION:
ANALOG_4 : Distortion Drive
ANALOG_5 : Flanger Dry/Wet
ANALOG_6 : Reverberation Dry/Wet
ANALOG_7 : Reverberation Room size
MIDI:
CC 7 : Volume
CC 10 : Pan
CC 13 : Flanger Delay
CC 13 : Flanger Delay
CC 94 : Flanger Feedback
CC 95 : Reverberation Damp
CC 90: Reverberation Stereo Width
/////////////////////////////////////////////////////////////////////////////////////////////////
VOLUME:
Should be 7 according to MIDI CC norm.
EFFECTS /////////////////////////////////////////////
Flanger
Pannoramique:
REVERB (from freeverb_demo)
(g = Dry/Wet)
Dry-Wet (from C. LEBRETON)
ALL | import("stdfaust.lib");
drive = hslider ("drive[BELA: ANALOG_4]",0.3,0,1,0.001);
curdel = hslider("flangDel[midi:ctrl 13]",4,0.001,10,0.001);
fb = hslider("flangFeedback[midi:ctrl 94]",0.7,0,1,0.001);
fldw = hslider("dryWetFlang[BELA: ANALOG_5]",0.5,0,1,0.001);
flanger = efx
with {
fldel = (curdel + (os.lf_triangle(1) * 2) ) : min(10);
efx = _ <: _, pf.flanger_mono(10,fldel,1,fb,0) : dry_wet(fldw);
};
panno = _ : sp.panner(hslider ("pan[midi:ctrl 10]",0.5,0,1,0.001)) : _,_;
reverb = _,_ <: (*(g)*fixedgain,*(g)*fixedgain :
re.stereo_freeverb(combfeed, allpassfeed, damping, spatSpread)),
*(1-g), *(1-g) :> _,_
with {
scaleroom = 0.28;
offsetroom = 0.7;
allpassfeed = 0.5;
scaledamp = 0.4;
fixedgain = 0.1;
origSR = 44100;
damping = vslider("Damp[midi:ctrl 95]",0.5, 0, 1, 0.025)*scaledamp*origSR/ma.SR;
combfeed = vslider("RoomSize[BELA: ANALOG_7]", 0.7, 0, 1, 0.025)*scaleroom*origSR/ma.SR + offsetroom;
spatSpread = vslider("Stereo[midi:ctrl 90]",0.6,0,1,0.01)*46*ma.SR/origSR;
g = vslider("dryWetReverb[BELA: ANALOG_6]", 0.4, 0, 1, 0.001);
};
dry_wet(dw,x,y) = wet*y + dry*x
with {
wet = 0.5*(dw+1.0);
dry = 1.0-wet;
};
effets = _ *(vol) : ef.cubicnl_nodc(drive, 0.1) : flanger : panno : reverb;
process = effets;
|
2cfbf0390befc308c125b8943cc4b3be1525dd15a555f0d105584024fd47a343 | tonal-glyph/faustus | 16_channel_volume.dsp | //-----------------------------------------------
// MIDI controlled 16 channel volume control in db
//-----------------------------------------------
import("stdfaust.lib");
import("layout.dsp");
channel01 = v01(vslider("Volume-01 [midi:ctrl 1] [tooltip CC-1]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel02 = v02(vslider("Volume-02 [midi:ctrl 2] [tooltip CC-2]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel03 = v03(vslider("Volume-03 [midi:ctrl 3] [tooltip CC-3]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel04 = v04(vslider("Volume-04 [midi:ctrl 4] [tooltip CC-4]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel05 = v05(vslider("Volume-05 [midi:ctrl 5] [tooltip CC-5]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel06 = v06(vslider("Volume-06 [midi:ctrl 6] [tooltip CC-6]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel07 = v07(vslider("Volume-07 [midi:ctrl 7] [tooltip CC-7]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel08 = v08(vslider("Volume-08 [midi:ctrl 8] [tooltip CC-8]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel09 = v09(vslider("Volume-09 [midi:ctrl 9] [tooltip CC-9]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel10 = v10(vslider("Volume-10 [midi:ctrl 10] [tooltip CC-10]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel11 = v11(vslider("Volume-11 [midi:ctrl 11] [tooltip CC-11]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel12 = v12(vslider("Volume-12 [midi:ctrl 12] [tooltip CC-12]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel13 = v13(vslider("Volume-13 [midi:ctrl 13] [tooltip CC-13]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel14 = v14(vslider("Volume-14 [midi:ctrl 14] [tooltip CC-14]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel15 = v15(vslider("Volume-15 [midi:ctrl 15] [tooltip CC-15]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel16 = v16(vslider("Volume-16 [midi:ctrl 16] [tooltip CC-16]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
process = *(channel01), *(channel02), *(channel03), *(channel04), *(channel05), *(channel06), *(channel07), *(channel08), *(channel09), *(channel10), *(channel11), *(channel12), *(channel13), *(channel14), *(channel15), *(channel16);
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/SAM/16_channel_volume/16_channel_volume.dsp | faust | -----------------------------------------------
MIDI controlled 16 channel volume control in db
----------------------------------------------- |
import("stdfaust.lib");
import("layout.dsp");
channel01 = v01(vslider("Volume-01 [midi:ctrl 1] [tooltip CC-1]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel02 = v02(vslider("Volume-02 [midi:ctrl 2] [tooltip CC-2]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel03 = v03(vslider("Volume-03 [midi:ctrl 3] [tooltip CC-3]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel04 = v04(vslider("Volume-04 [midi:ctrl 4] [tooltip CC-4]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel05 = v05(vslider("Volume-05 [midi:ctrl 5] [tooltip CC-5]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel06 = v06(vslider("Volume-06 [midi:ctrl 6] [tooltip CC-6]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel07 = v07(vslider("Volume-07 [midi:ctrl 7] [tooltip CC-7]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel08 = v08(vslider("Volume-08 [midi:ctrl 8] [tooltip CC-8]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel09 = v09(vslider("Volume-09 [midi:ctrl 9] [tooltip CC-9]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel10 = v10(vslider("Volume-10 [midi:ctrl 10] [tooltip CC-10]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel11 = v11(vslider("Volume-11 [midi:ctrl 11] [tooltip CC-11]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel12 = v12(vslider("Volume-12 [midi:ctrl 12] [tooltip CC-12]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel13 = v13(vslider("Volume-13 [midi:ctrl 13] [tooltip CC-13]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel14 = v14(vslider("Volume-14 [midi:ctrl 14] [tooltip CC-14]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel15 = v15(vslider("Volume-15 [midi:ctrl 15] [tooltip CC-15]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
channel16 = v16(vslider("Volume-16 [midi:ctrl 16] [tooltip CC-16]", 0, -70, +4, 0.1) : ba.db2linear : si.smoo);
process = *(channel01), *(channel02), *(channel03), *(channel04), *(channel05), *(channel06), *(channel07), *(channel08), *(channel09), *(channel10), *(channel11), *(channel12), *(channel13), *(channel14), *(channel15), *(channel16);
|
a8bedcfc17eec147a52fad6ba38e07bb6926c09656335d717a73ff5ce602a095 | tonal-glyph/faustus | crossDelay2.dsp | import("stdfaust.lib");
///////////////////////////////////////////////////////////////////////////////////////////////////
//
// Stereo Delay with feedback and crossfeedback (L to R and R to L feedback).
// And pitch shifting on feedback.
// A pre-delay without feedback is added for a wider stereo effect.
//
// Designed to use the Analog Input for parameters controls.
//
///////////////////////////////////////////////////////////////////////////////////////////////////
//
// ANALOG IN:
// ANALOG 0 : Pre-Delay L
// ANALOG 1 : Pre-Delay R
// ANALOG 2 : Delay L
// ANALOG 3 : Delay R
// ANALOG 4 : Cross feedback
// ANALOG 5 : Feedback
// ANALOG 6 : Pitchshifter L
// ANALOG 7 : Pitchshifter R
//
// Available by OSC : (see BELA console for precise adress)
// Feedback filter:
// crossLF : Crossfeedback Lowpass
// crossHF : Crossfeedback Highpass
// feedbLF : Feedback Lowpass
// feedbHF : Feedback Highpass
//
///////////////////////////////////////////////////////////////////////////////////////////////////
preDelL = ba.sec2samp(hslider("delR[BELA: ANALOG_0]", 1,0,2,0.001)):si.smoo;
preDelR = ba.sec2samp(hslider("delR[BELA: ANALOG_1]", 1,0,2,0.001)):si.smoo;
delL = ba.sec2samp(hslider("delL[BELA: ANALOG_2]", 1,0,2,0.001)):si.smoo;
delR = ba.sec2samp(hslider("delR[BELA: ANALOG_3]", 1,0,2,0.001)):si.smoo;
crossLF = hslider("crossLF", 12000, 20, 20000, 0.001);
crossHF = hslider("crossLF", 60, 20, 20000, 0.001);
feedbLF = hslider("feedbLF", 12000, 20, 20000, 0.001);
feedbHF = hslider("feedbHF", 60, 20, 20000, 0.001);
CrossFeedb = hslider("CrossFeedb[BELA: ANALOG_4]", 0.0, 0., 1, 0.001):si.smoo;
feedback = hslider("feedback[BELA: ANALOG_5]", 0.0, 0., 1, 0.001):si.smoo;
pitchL = hslider("shiftL[BELA: ANALOG_6]", 0,-12,12,0.001):si.smoo;
pitchR = hslider("shiftL[BELA: ANALOG_7]", 0,-12,12,0.001):si.smoo;
routeur(a,b,c,d) = ((a*CrossFeedb):fi.lowpass(2,crossLF):fi.highpass(2,crossHF))+((b*feedback):fi.lowpass(2,feedbLF):fi.highpass(2,feedbHF))+c,
((b*CrossFeedb):fi.lowpass(2,crossLF):fi.highpass(2,crossHF))+((a*feedback):fi.lowpass(2,feedbLF):fi.highpass(2,feedbHF))+d;
process = (de.sdelay(65536, 512,preDelL),de.sdelay(65536, 512,preDelR)):(routeur : de.sdelay(65536, 512,delL) , de.sdelay(65536, 512,delR))~(ef.transpose(512, 256, pitchL) , ef.transpose(512, 256, pitchR));
| https://raw.githubusercontent.com/tonal-glyph/faustus/cc887b115aceaee202edbdb37ee0e4087bfb5a33/examples/bela/crossDelay2.dsp | faust | /////////////////////////////////////////////////////////////////////////////////////////////////
Stereo Delay with feedback and crossfeedback (L to R and R to L feedback).
And pitch shifting on feedback.
A pre-delay without feedback is added for a wider stereo effect.
Designed to use the Analog Input for parameters controls.
/////////////////////////////////////////////////////////////////////////////////////////////////
ANALOG IN:
ANALOG 0 : Pre-Delay L
ANALOG 1 : Pre-Delay R
ANALOG 2 : Delay L
ANALOG 3 : Delay R
ANALOG 4 : Cross feedback
ANALOG 5 : Feedback
ANALOG 6 : Pitchshifter L
ANALOG 7 : Pitchshifter R
Available by OSC : (see BELA console for precise adress)
Feedback filter:
crossLF : Crossfeedback Lowpass
crossHF : Crossfeedback Highpass
feedbLF : Feedback Lowpass
feedbHF : Feedback Highpass
///////////////////////////////////////////////////////////////////////////////////////////////// | import("stdfaust.lib");
preDelL = ba.sec2samp(hslider("delR[BELA: ANALOG_0]", 1,0,2,0.001)):si.smoo;
preDelR = ba.sec2samp(hslider("delR[BELA: ANALOG_1]", 1,0,2,0.001)):si.smoo;
delL = ba.sec2samp(hslider("delL[BELA: ANALOG_2]", 1,0,2,0.001)):si.smoo;
delR = ba.sec2samp(hslider("delR[BELA: ANALOG_3]", 1,0,2,0.001)):si.smoo;
crossLF = hslider("crossLF", 12000, 20, 20000, 0.001);
crossHF = hslider("crossLF", 60, 20, 20000, 0.001);
feedbLF = hslider("feedbLF", 12000, 20, 20000, 0.001);
feedbHF = hslider("feedbHF", 60, 20, 20000, 0.001);
CrossFeedb = hslider("CrossFeedb[BELA: ANALOG_4]", 0.0, 0., 1, 0.001):si.smoo;
feedback = hslider("feedback[BELA: ANALOG_5]", 0.0, 0., 1, 0.001):si.smoo;
pitchL = hslider("shiftL[BELA: ANALOG_6]", 0,-12,12,0.001):si.smoo;
pitchR = hslider("shiftL[BELA: ANALOG_7]", 0,-12,12,0.001):si.smoo;
routeur(a,b,c,d) = ((a*CrossFeedb):fi.lowpass(2,crossLF):fi.highpass(2,crossHF))+((b*feedback):fi.lowpass(2,feedbLF):fi.highpass(2,feedbHF))+c,
((b*CrossFeedb):fi.lowpass(2,crossLF):fi.highpass(2,crossHF))+((a*feedback):fi.lowpass(2,feedbLF):fi.highpass(2,feedbHF))+d;
process = (de.sdelay(65536, 512,preDelL),de.sdelay(65536, 512,preDelR)):(routeur : de.sdelay(65536, 512,delL) , de.sdelay(65536, 512,delR))~(ef.transpose(512, 256, pitchL) , ef.transpose(512, 256, pitchR));
|
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