--- /dev/null
+declare name "Faust Audio Effect Library";
+declare author "Julius O. Smith (jos at ccrma.stanford.edu)";
+declare copyright "Julius O. Smith III";
+declare version "1.33";
+declare license "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license)
+declare reference "https://ccrma.stanford.edu/realsimple/faust_strings/";
+
+import("filter.lib"); // dcblocker*, lowpass, filterbank, ...
+
+// The following utilities (or equivalents) could go in music.lib:
+
+//----------------------- midikey2hz,pianokey2hz ------------------------
+midikey2hz(x) = 440.0*pow(2.0, (x-69.0)/12); // MIDI key 69 = A440
+pianokey2hz(x) = 440.0*pow(2.0, (x-49.0)/12); // piano key 49 = A440
+
+//---------------- cross2, bypass1, bypass2, select2stereo --------------
+//
+cross2 = _,_,_,_ <: _,!,_,!,!,_,!,_;
+
+bypass1(bpc,e) = _ <: select2(bpc,(inswitch:e),_)
+ with {inswitch = select2(bpc,_,0);};
+
+bypass2(bpc,e) = _,_ <: ((inswitch:e),_,_) : select2stereo(bpc) with {
+ inswitch = _,_ : (select2(bpc,_,0), select2(bpc,_,0)) : _,_;
+};
+
+select2stereo(bpc) = cross2 : select2(bpc), select2(bpc) : _,_;
+
+//---------------------- levelfilter, levelfilterN ----------------------
+// Dynamic level lowpass filter:
+//
+// USAGE: levelfilter(L,freq), where
+// L = desired level (in dB) at Nyquist limit (SR/2), e.g., -60
+// freq = corner frequency (-3dB point) usually set to fundamental freq
+//
+// REFERENCE:
+// https://ccrma.stanford.edu/realsimple/faust_strings/Dynamic_Level_Lowpass_Filter.html
+//
+levelfilter(L,freq,x) = (L * L0 * x) + ((1.0-L) * lp2out(x))
+with {
+ L0 = pow(L,1/3);
+ Lw = PI*freq/SR; // = w1 T / 2
+ Lgain = Lw / (1.0 + Lw);
+ Lpole2 = (1.0 - Lw) / (1.0 + Lw);
+ lp2out = *(Lgain) : + ~ *(Lpole2);
+};
+
+levelfilterN(N,freq,L) = seq(i,N,levelfilter((L/N),freq));
+
+//------------------------- speakerbp -------------------------------
+// Dirt-simple speaker simulator (overall bandpass eq with observed
+// roll-offs above and below the passband).
+//
+// Low-frequency speaker model = +12 dB/octave slope breaking to
+// flat near f1. Implemented using two dc blockers in series.
+//
+// High-frequency model = -24 dB/octave slope implemented using a
+// fourth-order Butterworth lowpass.
+//
+// Example based on measured Celestion G12 (12" speaker):
+// speakerbp(130,5000);
+//
+// Requires filter.lib
+//
+speakerbp(f1,f2) = dcblockerat(f1) : dcblockerat(f1) : lowpass(4,f2);
+
+
+//--------------------- cubicnl(drive,offset) -----------------------
+// Cubic nonlinearity distortion
+//
+// USAGE: cubicnl(drive,offset), where
+// drive = distortion amount, between 0 and 1
+// offset = constant added before nonlinearity to give even harmonics
+// Note: offset can introduce a nonzero mean - feed
+// cubicnl output to dcblocker to remove this.
+//
+// REFERENCES:
+// https://ccrma.stanford.edu/~jos/pasp/Cubic_Soft_Clipper.html
+// https://ccrma.stanford.edu/~jos/pasp/Nonlinear_Distortion.html
+//
+cubicnl(drive,offset) = *(pregain) : +(offset) : clip(-1,1) : cubic
+with {
+ pregain = pow(10.0,2*drive);
+ clip(lo,hi) = min(hi) : max(lo);
+ cubic(x) = x - x*x*x/3;
+ postgain = max(1.0,1.0/pregain); // unity gain when nearly linear
+};
+
+cubicnl_nodc(drive,offset) = cubicnl(drive,offset) : dcblocker;
+
+//--------------------------- cubicnl_demo --------------------------
+// USAGE: _ : cubicnl_demo : _;
+//
+cubicnl_demo = bypass1(bp,
+ cubicnl_nodc(drive:smooth(0.999),offset:smooth(0.999)))
+with {
+ cnl_group(x) = vgroup("CUBIC NONLINEARITY cubicnl
+ [tooltip: Reference:
+ https://ccrma.stanford.edu/~jos/pasp/Cubic_Soft_Clipper.html]", x);
+// bypass_group(x) = cnl_group(hgroup("[0]", x));
+ slider_group(x) = cnl_group(hgroup("[1]", x));
+// bp = bypass_group(checkbox("[0] Bypass
+ bp = slider_group(checkbox("[0] Bypass
+ [tooltip: When this is checked, the nonlinearity has no effect]"));
+// drive = slider_group(vslider("[1] Drive [style: knob]
+ drive = slider_group(hslider("[1] Drive
+ [tooltip: Amount of distortion]",
+ 0, 0, 1, 0.01));
+// offset = slider_group(vslider("[2] Offset [style: knob]
+ offset = slider_group(hslider("[2] Offset
+ [tooltip: Brings in even harmonics]",
+ 0, 0, 1, 0.01));
+};
+
+//------------------------- moog_vcf(res,fr) ---------------------------
+// Moog "Voltage Controlled Filter" (VCF) in "analog" form
+//
+// USAGE: moog_vcf(res,fr), where
+// fr = corner-resonance frequency in Hz ( less than SR/6.3 or so )
+// res = Normalized amount of corner-resonance between 0 and 1
+// (0 is no resonance, 1 is maximum)
+// Requires filter.lib.
+//
+// DESCRIPTION: Moog VCF implemented using the same logical block diagram
+// as the classic analog circuit. As such, it neglects the one-sample
+// delay associated with the feedback path around the four one-poles.
+// This extra delay alters the response, especially at high frequencies
+// (see reference [1] for details).
+// See moog_vcf_2b below for a more accurate implementation.
+//
+// REFERENCES:
+// [1] https://ccrma.stanford.edu/~stilti/papers/moogvcf.pdf
+// [2] https://ccrma.stanford.edu/~jos/pasp/vegf.html
+//
+moog_vcf(res,fr) = (+ : seq(i,4,pole(p)) : *(unitygain(p))) ~ *(mk)
+with {
+ p = 1.0 - fr * 2.0 * PI / SR; // good approximation for fr << SR
+ unitygain(p) = pow(1.0-p,4.0); // one-pole unity-gain scaling
+ mk = -4.0*max(0,min(res,0.999999)); // need mk > -4 for stability
+};
+
+//----------------------- moog_vcf_2b[n] ---------------------------
+// Moog "Voltage Controlled Filter" (VCF) as two biquads
+//
+// USAGE:
+// moog_vcf_2b(res,fr)
+// moog_vcf_2bn(res,fr)
+// where
+// fr = corner-resonance frequency in Hz
+// res = Normalized amount of corner-resonance between 0 and 1
+// (0 is min resonance, 1 is maximum)
+//
+// DESCRIPTION: Implementation of the ideal Moog VCF transfer
+// function factored into second-order sections. As a result, it is
+// more accurate than moog_vcf above, but its coefficient formulas are
+// more complex when one or both parameters are varied. Here, res
+// is the fourth root of that in moog_vcf, so, as the sampling rate
+// approaches infinity, moog_vcf(res,fr) becomes equivalent
+// to moog_vcf_2b[n](res^4,fr) (when res and fr are constant).
+//
+// moog_vcf_2b uses two direct-form biquads (tf2)
+// moog_vcf_2bn uses two protected normalized-ladder biquads (tf2np)
+//
+// REQUIRES: filter.lib
+//
+moog_vcf_2b(res,fr) = tf2s(0,0,b0,a11,a01,w1) : tf2s(0,0,b0,a12,a02,w1)
+with {
+ s = 1; // minus the open-loop location of all four poles
+ frl = max(20,min(10000,fr)); // limit fr to reasonable 20-10k Hz range
+ w1 = 2*PI*frl; // frequency-scaling parameter for bilinear xform
+ // Equivalent: w1 = 1; s = 2*PI*frl;
+ kmax = sqrt(2)*0.999; // 0.999 gives stability margin (tf2 is unprotected)
+ k = min(kmax,sqrt(2)*res); // fourth root of Moog VCF feedback gain
+ b0 = s^2;
+ s2k = sqrt(2) * k;
+ a11 = s * (2 + s2k);
+ a12 = s * (2 - s2k);
+ a01 = b0 * (1 + s2k + k^2);
+ a02 = b0 * (1 - s2k + k^2);
+};
+
+moog_vcf_2bn(res,fr) = tf2snp(0,0,b0,a11,a01,w1) : tf2snp(0,0,b0,a12,a02,w1)
+with {
+ s = 1; // minus the open-loop location of all four poles
+ w1 = 2*PI*max(fr,20); // frequency-scaling parameter for bilinear xform
+ k = sqrt(2)*0.999*res; // fourth root of Moog VCF feedback gain
+ b0 = s^2;
+ s2k = sqrt(2) * k;
+ a11 = s * (2 + s2k);
+ a12 = s * (2 - s2k);
+ a01 = b0 * (1 + s2k + k^2);
+ a02 = b0 * (1 - s2k + k^2);
+};
+
+//------------------------- moog_vcf_demo ---------------------------
+// Illustrate and compare all three Moog VCF implementations above
+// (called by <faust>/examples/vcf_wah_pedals.dsp).
+//
+// USAGE: _ : moog_vcf_demo : _;
+
+moog_vcf_demo = bypass1(bp,vcf) with {
+ mvcf_group(x) = hgroup("MOOG VCF (Voltage Controlled Filter)
+ [tooltip: See Faust's effect.lib for info and references]",x);
+
+ meter_group(x) = mvcf_group(vgroup("[0]",x));
+ cb_group(x) = meter_group(hgroup("[0]",x));
+
+ bp = cb_group(checkbox("[0] Bypass [tooltip: When this is checked, the Moog VCF has no effect]"));
+ archsw = cb_group(checkbox("[1] Use Biquads
+ [tooltip: Select moog_vcf_2b (two-biquad) implementation, instead of the default moog_vcf (analog style) implementation]"));
+ bqsw = cb_group(checkbox("[2] Normalized Ladders
+ [tooltip: If using biquads, make them normalized ladders (moog_vcf_2bn)]"));
+
+ freq = mvcf_group(hslider("[1] Corner Frequency [unit:PK] [style:knob]
+ [tooltip: The VCF resonates at the corner frequency (specified in PianoKey (PK) units, with A440 = 49 PK). The VCF response is flat below the corner frequency, and rolls off -24 dB per octave above.]",
+ 25, 1, 88, 0.01) : pianokey2hz) : smooth(0.999);
+
+ res = mvcf_group(hslider("[2] Corner Resonance [style:knob]
+ [tooltip: Amount of resonance near VCF corner frequency (specified between 0 and 1)]",
+ 0.9, 0, 1, 0.01));
+
+ outgain = meter_group(hslider("[1] VCF Output Level [unit:dB]
+ [tooltip: output level in decibels]",
+ 5, -60, 20, 0.1)) : smooth(0.999)
+ : component("music.lib").db2linear;
+
+ vcfbq = _ <: select2(bqsw, moog_vcf_2b(res,freq), moog_vcf_2bn(res,freq));
+ vcfarch = _ <: select2(archsw, moog_vcf(res^4,freq), vcfbq);
+ vcf = vcfarch : *(outgain);
+};
+
+//-------------------------- wah4(fr) -------------------------------
+// Wah effect, 4th order
+// USAGE: wah4(fr), where fr = resonance frequency in Hz
+// REFERENCE "https://ccrma.stanford.edu/~jos/pasp/vegf.html";
+//
+wah4(fr) = 4*moog_vcf((3.2/4),fr:smooth(0.999));
+
+//------------------------- wah4_demo ---------------------------
+// USAGE: _ : wah4_demo : _;
+
+wah4_demo = bypass1(bp, wah4(fr)) with {
+ wah4_group(x) = hgroup("WAH4
+ [tooltip: Fourth-order wah effect made using moog_vcf]", x);
+ bp = wah4_group(checkbox("[0] Bypass
+ [tooltip: When this is checked, the wah pedal has no effect]"));
+ fr = wah4_group(hslider("[1] Resonance Frequency
+ [tooltip: wah resonance frequency in Hz]",
+ 200,100,2000,1));
+// Avoid dc with the moog_vcf (amplitude too high when freq comes up from dc)
+// Also, avoid very high resonance frequencies (e.g., 5kHz or above).
+};
+
+//------------------------ autowah(level) -----------------------------
+// Auto-wah effect
+// USAGE: _ : autowah(level) : _;
+// where level = amount of effect desired (0 to 1).
+//
+autowah(level,x) = level * crybaby(amp_follower(0.1,x),x) + (1.0-level)*x;
+
+//-------------------------- crybaby(wah) -----------------------------
+// Digitized CryBaby wah pedal
+// USAGE: _ : crybaby(wah) : _;
+// where wah = "pedal angle" from 0 to 1.
+// REFERENCE: https://ccrma.stanford.edu/~jos/pasp/vegf.html
+//
+crybaby(wah) = *(gs) : tf2(1,-1,0,a1s,a2s)
+with {
+ Q = pow(2.0,(2.0*(1.0-wah)+1.0)); // Resonance "quality factor"
+ fr = 450.0*pow(2.0,2.3*wah); // Resonance tuning
+ g = 0.1*pow(4.0,wah); // gain (optional)
+
+ // Biquad fit using z = exp(s T) ~ 1 + sT for low frequencies:
+ frn = fr/SR; // Normalized pole frequency (cycles per sample)
+ R = 1 - PI*frn/Q; // pole radius
+ theta = 2*PI*frn; // pole angle
+ a1 = 0-2.0*R*cos(theta); // biquad coeff
+ a2 = R*R; // biquad coeff
+
+ // dezippering of slider-driven signals:
+ s = 0.999; // smoothing parameter (one-pole pole location)
+ a1s = a1 : smooth(s);
+ a2s = a2 : smooth(s);
+ gs = g : smooth(s);
+
+ tf2 = component("filter.lib").tf2;
+};
+
+//------------------------- crybaby_demo ---------------------------
+// USAGE: _ : crybaby_demo : _ ;
+
+crybaby_demo = bypass1(bp, crybaby(wah)) with {
+ crybaby_group(x) = hgroup("CRYBABY [tooltip: Reference: https://ccrma.stanford.edu/~jos/pasp/vegf.html]", x);
+ bp = crybaby_group(checkbox("[0] Bypass [tooltip: When this is checked, the wah pedal has no effect]"));
+ wah = crybaby_group(hslider("[1] Wah parameter [tooltip: wah pedal angle between 0 (rocked back) and 1 (rocked forward)]",0.8,0,1,0.01));
+};
+
+//------------ apnl(a1,a2) ---------------
+// Passive Nonlinear Allpass:
+// switch between allpass coefficient a1 and a2 at signal zero crossings
+// REFERENCE:
+// "A Passive Nonlinear Digital Filter Design ..."
+// by John R. Pierce and Scott A. Van Duyne,
+// JASA, vol. 101, no. 2, pp. 1120-1126, 1997
+// Written by Romain Michon and JOS based on Pierce switching springs idea:
+ apnl(a1,a2,x) = nonLinFilter
+ with{
+ condition = _>0;
+ nonLinFilter = (x - _ <: _*(condition*a1 + (1-condition)*a2),_')~_ :> +;
+ };
+
+//------------ piano_dispersion_filter(M,B,f0) ---------------
+// Piano dispersion allpass filter in closed form
+//
+// ARGUMENTS:
+// M = number of first-order allpass sections (compile-time only)
+// Keep below 20. 8 is typical for medium-sized piano strings.
+// B = string inharmonicity coefficient (0.0001 is typical)
+// f0 = fundamental frequency in Hz
+//
+// INPUT:
+// Signal to be filtered by the allpass chain
+//
+// OUTPUTS:
+// 1. MINUS the estimated delay at f0 of allpass chain in samples,
+// provided in negative form to facilitate subtraction
+// from delay-line length (see USAGE below).
+// 2. Output signal from allpass chain
+//
+// USAGE:
+// piano_dispersion_filter(1,B,f0) : +(totalDelay),_ : fdelay(maxDelay)
+//
+// REFERENCE:
+// "Dispersion Modeling in Waveguide Piano Synthesis
+// Using Tunable Allpass Filters",
+// by Jukka Rauhala and Vesa Valimaki, DAFX-2006, pp. 71-76
+// URL: http://www.dafx.ca/proceedings/papers/p_071.pdf
+// NOTE: An erratum in Eq. (7) is corrected in Dr. Rauhala's
+// encompassing dissertation (and below).
+// See also: http://www.acoustics.hut.fi/research/asp/piano/
+//
+piano_dispersion_filter(M,B,f0) = -Df0*M,seq(i,M,tf1(a1,1,a1))
+with {
+ a1 = (1-D)/(1+D); // By Eq. 3, have D >= 0, hence a1 >= 0 also
+ D = exp(Cd - Ikey(f0)*kd);
+ trt = pow(2.0,1.0/12.0); // 12th root of 2
+ logb(b,x) = log(x) / log(b); // log-base-b of x
+ Ikey(f0) = logb(trt,f0*trt/27.5);
+ Bc = max(B,0.000001);
+ kd = exp(k1*log(Bc)*log(Bc) + k2*log(Bc)+k3);
+ Cd = exp((m1*log(M)+m2)*log(Bc)+m3*log(M)+m4);
+ k1 = -0.00179;
+ k2 = -0.0233;
+ k3 = -2.93;
+ m1 = 0.0126;
+ m2 = 0.0606;
+ m3 = -0.00825;
+ m4 = 1.97;
+ wT = 2*PI*f0/SR;
+ polydel(a) = atan(sin(wT)/(a+cos(wT)))/wT;
+ Df0 = polydel(a1) - polydel(1.0/a1);
+};
+
+//===================== Phasing and Flanging Effects ====================
+
+//--------------- flanger_mono, flanger_stereo, flanger_demo -------------
+// Flanging effect
+//
+// USAGE:
+// _ : flanger_mono(dmax,curdel,depth,fb,invert) : _;
+// _,_ : flanger_stereo(dmax,curdel1,curdel2,depth,fb,invert) : _,_;
+// _,_ : flanger_demo : _,_;
+//
+// ARGUMENTS:
+// dmax = maximum delay-line length (power of 2) - 10 ms typical
+// curdel = current dynamic delay (not to exceed dmax)
+// depth = effect strength between 0 and 1 (1 typical)
+// fb = feedback gain between 0 and 1 (0 typical)
+// invert = 0 for normal, 1 to invert sign of flanging sum
+//
+// REFERENCE:
+// https://ccrma.stanford.edu/~jos/pasp/Flanging.html
+//
+flanger_mono(dmax,curdel,depth,fb,invert)
+ = _ <: _, (-:fdelay(dmax,curdel)) ~ *(fb) : _,
+ *(select2(invert,depth,0-depth))
+ : + : *(0.5);
+
+flanger_stereo(dmax,curdel1,curdel2,depth,fb,invert)
+ = flanger_mono(dmax,curdel1,depth,fb,invert),
+ flanger_mono(dmax,curdel2,depth,fb,invert);
+
+//------------------------- flanger_demo ---------------------------
+// USAGE: _,_ : flanger_demo : _,_;
+//
+flanger_demo = bypass2(fbp,flanger_stereo_demo) with {
+ flanger_group(x) =
+ vgroup("FLANGER [tooltip: Reference: https://ccrma.stanford.edu/~jos/pasp/Flanging.html]", x);
+ meter_group(x) = flanger_group(hgroup("[0]", x));
+ ctl_group(x) = flanger_group(hgroup("[1]", x));
+ del_group(x) = flanger_group(hgroup("[2] Delay Controls", x));
+ lvl_group(x) = flanger_group(hgroup("[3]", x));
+
+ fbp = meter_group(checkbox(
+ "[0] Bypass [tooltip: When this is checked, the flanger has no effect]"));
+ invert = meter_group(checkbox("[1] Invert Flange Sum"));
+
+ // FIXME: This should be an amplitude-response display:
+ flangeview = lfor(freq) + lfol(freq) : meter_group(hbargraph(
+ "[2] Flange LFO [style: led] [tooltip: Display sum of flange delays]", -1.5,+1.5));
+
+ flanger_stereo_demo(x,y) = attach(x,flangeview),y :
+ *(level),*(level) : flanger_stereo(dmax,curdel1,curdel2,depth,fb,invert);
+
+ lfol = component("oscillator.lib").oscrs; // sine for left channel
+ lfor = component("oscillator.lib").oscrc; // cosine for right channel
+ dmax = 2048;
+ dflange = 0.001 * SR *
+ del_group(hslider("[1] Flange Delay [unit:ms] [style:knob]", 10, 0, 20, 0.001));
+ odflange = 0.001 * SR *
+ del_group(hslider("[2] Delay Offset [unit:ms] [style:knob]", 1, 0, 20, 0.001));
+ freq = ctl_group(hslider("[1] Speed [unit:Hz] [style:knob]", 0.5, 0, 10, 0.01));
+ depth = ctl_group(hslider("[2] Depth [style:knob]", 1, 0, 1, 0.001));
+ fb = ctl_group(hslider("[3] Feedback [style:knob]", 0, -0.999, 0.999, 0.001));
+ level = lvl_group(hslider("Flanger Output Level [unit:dB]", 0, -60, 10, 0.1)) : db2linear;
+ curdel1 = odflange+dflange*(1 + lfol(freq))/2;
+ curdel2 = odflange+dflange*(1 + lfor(freq))/2;
+};
+
+//------- phaser2_mono, phaser2_stereo, phaser2_demo -------
+// Phasing effect
+//
+// USAGE:
+// _ : phaser2_mono(Notches,width,frqmin,fratio,frqmax,speed,depth,fb,invert) : _;
+// _,_ : phaser2_stereo(") : _,_;
+// _,_ : phaser2_demo : _,_;
+//
+// ARGUMENTS:
+// Notches = number of spectral notches (MACRO ARGUMENT - not a signal)
+// width = approximate width of spectral notches in Hz
+// frqmin = approximate minimum frequency of first spectral notch in Hz
+// fratio = ratio of adjacent notch frequencies
+// frqmax = approximate maximum frequency of first spectral notch in Hz
+// speed = LFO frequency in Hz (rate of periodic notch sweep cycles)
+// depth = effect strength between 0 and 1 (1 typical) (aka "intensity")
+// when depth=2, "vibrato mode" is obtained (pure allpass chain)
+// fb = feedback gain between -1 and 1 (0 typical)
+// invert = 0 for normal, 1 to invert sign of flanging sum
+//
+// REFERENCES:
+// https://ccrma.stanford.edu/~jos/pasp/Phasing.html
+// http://www.geofex.com/Article_Folders/phasers/phase.html
+// 'An Allpass Approach to Digital Phasing and Flanging', Julius O. Smith III,
+// Proc. Int. Computer Music Conf. (ICMC-84), pp. 103-109, Paris, 1984.
+// CCRMA Tech. Report STAN-M-21: https://ccrma.stanford.edu/STANM/stanms/stanm21/
+
+vibrato2_mono(sections,phase01,fb,width,frqmin,fratio,frqmax,speed) =
+ (+ : seq(i,sections,ap2p(R,th(i)))) ~ *(fb)
+with {
+ tf2 = component("filter.lib").tf2;
+ // second-order resonant digital allpass given pole radius and angle:
+ ap2p(R,th) = tf2(a2,a1,1,a1,a2) with {
+ a2 = R^2;
+ a1 = -2*R*cos(th);
+ };
+ SR = component("music.lib").SR;
+ R = exp(-pi*width/SR);
+ cososc = component("oscillator.lib").oscrc;
+ sinosc = component("oscillator.lib").oscrs;
+ osc = cososc(speed) * phase01 + sinosc(speed) * (1-phase01);
+ lfo = (1-osc)/2; // in [0,1]
+ pi = 4*atan(1);
+ thmin = 2*pi*frqmin/SR;
+ thmax = 2*pi*frqmax/SR;
+ th1 = thmin + (thmax-thmin)*lfo;
+ th(i) = (fratio^(i+1))*th1;
+};
+
+phaser2_mono(Notches,phase01,width,frqmin,fratio,frqmax,speed,depth,fb,invert) =
+ _ <: *(g1) + g2mi*vibrato2_mono(Notches,phase01,fb,width,frqmin,fratio,frqmax,speed)
+with { // depth=0 => direct-signal only
+ g1 = 1-depth/2; // depth=1 => phaser mode (equal sum of direct and allpass-chain)
+ g2 = depth/2; // depth=2 => vibrato mode (allpass-chain signal only)
+ g2mi = select2(invert,g2,-g2); // inversion negates the allpass-chain signal
+};
+
+phaser2_stereo(Notches,width,frqmin,fratio,frqmax,speed,depth,fb,invert)
+ = phaser2_mono(Notches,0,width,frqmin,fratio,frqmax,speed,depth,fb,invert),
+ phaser2_mono(Notches,1,width,frqmin,fratio,frqmax,speed,depth,fb,invert);
+
+//------------------------- phaser2_demo ---------------------------
+// USAGE: _,_ : phaser2_demo : _,_;
+//
+phaser2_demo = bypass2(pbp,phaser2_stereo_demo) with {
+ phaser2_group(x) =
+ vgroup("PHASER2 [tooltip: Reference: https://ccrma.stanford.edu/~jos/pasp/Flanging.html]", x);
+ meter_group(x) = phaser2_group(hgroup("[0]", x));
+ ctl_group(x) = phaser2_group(hgroup("[1]", x));
+ nch_group(x) = phaser2_group(hgroup("[2]", x));
+ lvl_group(x) = phaser2_group(hgroup("[3]", x));
+
+ pbp = meter_group(checkbox(
+ "[0] Bypass [tooltip: When this is checked, the phaser has no effect]"));
+ invert = meter_group(checkbox("[1] Invert Internal Phaser Sum"));
+ vibr = meter_group(checkbox("[2] Vibrato Mode")); // In this mode you can hear any "Doppler"
+
+ // FIXME: This should be an amplitude-response display:
+ //flangeview = phaser2_amp_resp : meter_group(hspectrumview("[2] Phaser Amplitude Response", 0,1));
+ //phaser2_stereo_demo(x,y) = attach(x,flangeview),y : ...
+
+ phaser2_stereo_demo = *(level),*(level) :
+ phaser2_stereo(Notches,width,frqmin,fratio,frqmax,speed,mdepth,fb,invert);
+
+ Notches = 4; // Compile-time parameter: 2 is typical for analog phaser stomp-boxes
+
+ // FIXME: Add tooltips
+ speed = ctl_group(hslider("[1] Speed [unit:Hz] [style:knob]", 0.5, 0, 10, 0.001));
+ depth = ctl_group(hslider("[2] Notch Depth (Intensity) [style:knob]", 1, 0, 1, 0.001));
+ fb = ctl_group(hslider("[3] Feedback Gain [style:knob]", 0, -0.999, 0.999, 0.001));
+
+ width = nch_group(hslider("[1] Notch width [unit:Hz] [style:knob]", 1000, 10, 5000, 1));
+ frqmin = nch_group(hslider("[2] Min Notch1 Freq [unit:Hz] [style:knob]", 100, 20, 5000, 1));
+ frqmax = nch_group(hslider("[3] Max Notch1 Freq [unit:Hz] [style:knob]", 800, 20, 10000, 1)) : max(frqmin);
+ fratio = nch_group(hslider("[4] Notch Freq Ratio: NotchFreq(n+1)/NotchFreq(n) [style:knob]", 1.5, 1.1, 4, 0.001));
+
+ level = lvl_group(hslider("Phaser Output Level [unit:dB]", 0, -60, 10, 0.1)) : component("music.lib").db2linear;
+
+ mdepth = select2(vibr,depth,2); // Improve "ease of use"
+};
+
+//------------------------- stereo_width(w) ---------------------------
+// Stereo Width effect using the Blumlein Shuffler technique.
+//
+// USAGE: "_,_ : stereo_width(w) : _,_", where
+// w = stereo width between 0 and 1
+//
+// At w=0, the output signal is mono ((left+right)/2 in both channels).
+// At w=1, there is no effect (original stereo image).
+// Thus, w between 0 and 1 varies stereo width from 0 to "original".
+//
+// REFERENCE:
+// "Applications of Blumlein Shuffling to Stereo Microphone Techniques"
+// Michael A. Gerzon, JAES vol. 42, no. 6, June 1994
+//
+stereo_width(w) = shuffle : *(mgain),*(sgain) : shuffle
+with {
+ shuffle = _,_ <: +,-; // normally scaled by 1/sqrt(2) for orthonormality,
+ mgain = 1-w/2; // but we pick up the needed normalization here.
+ sgain = w/2;
+};
+
+//--------------------------- amp_follower ---------------------------
+// Classic analog audio envelope follower with infinitely fast rise and
+// exponential decay. The amplitude envelope instantaneously follows
+// the absolute value going up, but then floats down exponentially.
+//
+// USAGE:
+// _ : amp_follower(rel) : _
+//
+// where
+// rel = release time = amplitude-envelope time-constant (sec) going down
+//
+// REFERENCES:
+// Musical Engineer's Handbook, Bernie Hutchins, Ithaca NY, 1975
+// Elecronotes Newsletter, Bernie Hutchins
+
+amp_follower(rel) = abs : env with {
+ p = tau2pole(rel);
+ env(x) = x * (1.0 - p) : + ~ max(x,_) * p;
+};
+
+//--------------------------- amp_follower_ud ---------------------------
+// Envelope follower with different up and down time-constants
+//
+// USAGE:
+// _ : amp_follower_ud(att,rel) : _
+//
+// where
+// att = attack time = amplitude-envelope time constant (sec) going up
+// rel = release time = amplitude-envelope time constant (sec) going down
+//
+// For audio, att should be faster (smaller) than rel (e.g., 0.001 and 0.01)
+
+amp_follower_ud(att,rel) = amp_follower(rel) : smooth(tau2pole(att));
+
+//=============== Gates, Limiters, and Dynamic Range Compression ============
+
+//----------------- gate_mono, gate_stereo -------------------
+// Mono and stereo signal gates
+//
+// USAGE:
+// _ : gate_mono(thresh,att,hold,rel) : _
+// or
+// _,_ : gate_stereo(thresh,att,hold,rel) : _,_
+//
+// where
+// thresh = dB level threshold above which gate opens (e.g., -60 dB)
+// att = attack time = time constant (sec) for gate to open (e.g., 0.0001 s = 0.1 ms)
+// hold = hold time = time (sec) gate stays open after signal level < thresh (e.g., 0.1 s)
+// rel = release time = time constant (sec) for gate to close (e.g., 0.020 s = 20 ms)
+//
+// REFERENCES:
+// - http://en.wikipedia.org/wiki/Noise_gate
+// - http://www.soundonsound.com/sos/apr01/articles/advanced.asp
+// - http://en.wikipedia.org/wiki/Gating_(sound_engineering)
+
+gate_mono(thresh,att,hold,rel,x) = x * gate_gain_mono(thresh,att,hold,rel,x);
+
+gate_stereo(thresh,att,hold,rel,x,y) = ggm*x, ggm*y with {
+ ggm = gate_gain_mono(thresh,att,hold,rel,abs(x)+abs(y));
+};
+
+gate_gain_mono(thresh,att,hold,rel,x) = extendedrawgate : amp_follower_ud(att,rel) with {
+ extendedrawgate = max(rawgatesig,holdsig);
+ rawgatesig = inlevel(x) > db2linear(thresh);
+ inlevel(x) = amp_follower_ud(att/2,rel/2,x);
+ holdsig = ((max(holdreset & holdsamps,_) ~-(1)) > 0);
+ holdreset = rawgatesig > rawgatesig'; // reset hold when raw gate falls
+ holdsamps = int(hold*SR);
+};
+
+//-------------------- compressor_mono, compressor_stereo ----------------------
+// Mono and stereo dynamic range compressor_s
+//
+// USAGE:
+// _ : compressor_mono(ratio,thresh,att,rel) : _
+// or
+// _,_ : compressor_stereo(ratio,thresh,att,rel) : _,_
+//
+// where
+// ratio = compression ratio (1 = no compression, >1 means compression)
+// thresh = dB level threshold above which compression kicks in
+// att = attack time = time constant (sec) when level & compression going up
+// rel = release time = time constant (sec) coming out of compression
+//
+// REFERENCES:
+// - http://en.wikipedia.org/wiki/Dynamic_range_compression
+// - https://ccrma.stanford.edu/~jos/filters/Nonlinear_Filter_Example_Dynamic.html
+// - Albert Graef's <faust2pd>/examples/synth/compressor_.dsp
+//
+
+compressor_mono(ratio,thresh,att,rel,x) = x * compression_gain_mono(ratio,thresh,att,rel,x);
+
+compressor_stereo(ratio,thresh,att,rel,x,y) = cgm*x, cgm*y with {
+ cgm = compression_gain_mono(ratio,thresh,att,rel,abs(x)+abs(y));
+};
+
+compression_gain_mono(ratio,thresh,att,rel) =
+ amp_follower_ud(att,rel) : linear2db : outminusindb(ratio,thresh) :
+ kneesmooth(att) : db2linear
+with {
+ // kneesmooth(att) installs a "knee" in the dynamic-range compression,
+ // where knee smoothness is set equal to half that of the compression-attack.
+ // A general 'knee' parameter could be used instead of tying it to att/2:
+ kneesmooth(att) = smooth(tau2pole(att/2.0));
+ // compression gain in dB:
+ outminusindb(ratio,thresh,level) = max(level-thresh,0) * (1/float(ratio)-1);
+ // Note: "float(ratio)" REQUIRED when ratio is an integer > 1!
+};
+
+//---------------------------- gate_demo -------------------------
+// USAGE: _,_ : gate_demo : _,_;
+//
+gate_demo = bypass2(gbp,gate_stereo_demo) with {
+
+ gate_group(x) = vgroup("GATE [tooltip: Reference: http://en.wikipedia.org/wiki/Noise_gate]", x);
+ meter_group(x) = gate_group(hgroup("[0]", x));
+ knob_group(x) = gate_group(hgroup("[1]", x));
+
+ gbp = meter_group(checkbox("[0] Bypass [tooltip: When this is checked, the gate has no effect]"));
+
+ gateview = gate_gain_mono(gatethr,gateatt,gatehold,gaterel) : linear2db :
+ meter_group(hbargraph("[1] Gate Gain [unit:dB] [tooltip: Current gain of the gate in dB]",
+ -50,+10)); // [style:led]
+
+ gate_stereo_demo(x,y) = attach(x,gateview(abs(x)+abs(y))),y :
+ gate_stereo(gatethr,gateatt,gatehold,gaterel);
+
+ gatethr = knob_group(hslider("[1] Threshold [unit:dB] [style:knob] [tooltip: When the signal level falls below the Threshold (expressed in dB), the signal is muted]",
+ -30, -120, 0, 0.1));
+
+ gateatt = knob_group(hslider("[2] Attack [unit:us] [style:knob] [tooltip: Time constant in MICROseconds (1/e smoothing time) for the gate gain to go (exponentially) from 0 (muted) to 1 (unmuted)]",
+ 10, 10, 10000, 1)) : *(0.000001) : max(1/SR);
+
+ gatehold = knob_group(hslider("[3] Hold [unit:ms] [style:knob] [tooltip: Time in ms to keep the gate open (no muting) after the signal level falls below the Threshold]",
+ 200, 0, 1000, 1)) : *(0.001) : max(1/SR);
+
+ gaterel = knob_group(hslider("[4] Release [unit:ms] [style:knob] [tooltip: Time constant in ms (1/e smoothing time) for the gain to go (exponentially) from 1 (unmuted) to 0 (muted)]",
+ 100, 0, 1000, 1)) : *(0.001) : max(1/SR);
+};
+
+//---------------------------- compressor_demo -------------------------
+// USAGE: _,_ : compressor_demo : _,_;
+//
+compressor_demo = bypass2(cbp,compressor_stereo_demo) with {
+
+ comp_group(x) = vgroup("COMPRESSOR [tooltip: Reference: http://en.wikipedia.org/wiki/Dynamic_range_compression]", x);
+
+ meter_group(x) = comp_group(hgroup("[0]", x));
+ knob_group(x) = comp_group(hgroup("[1]", x));
+
+ cbp = meter_group(checkbox("[0] Bypass [tooltip: When this is checked, the compressor has no effect]"));
+
+ gainview =
+ compression_gain_mono(ratio,threshold,attack,release) : linear2db :
+ meter_group(hbargraph("[1] Compressor Gain [unit:dB] [tooltip: Current gain of the compressor in dB]",
+ -50,+10));
+
+ displaygain = _,_ <: _,_,(abs,abs:+) : _,_,gainview : _,attach;
+
+ compressor_stereo_demo =
+ displaygain(compressor_stereo(ratio,threshold,attack,release)) :
+ *(makeupgain), *(makeupgain);
+
+ ctl_group(x) = knob_group(hgroup("[3] Compression Control", x));
+
+ ratio = ctl_group(hslider("[0] Ratio [style:knob] [tooltip: A compression Ratio of N means that for each N dB increase in input signal level above Threshold, the output level goes up 1 dB]",
+ 5, 1, 20, 0.1));
+
+ threshold = ctl_group(hslider("[1] Threshold [unit:dB] [style:knob] [tooltip: When the signal level exceeds the Threshold (in dB), its level is compressed according to the Ratio]",
+ -30, -100, 10, 0.1));
+
+ env_group(x) = knob_group(hgroup("[4] Compression Response", x));
+
+ attack = env_group(hslider("[1] Attack [unit:ms] [style:knob] [tooltip: Time constant in ms (1/e smoothing time) for the compression gain to approach (exponentially) a new lower target level (the compression `kicking in')]",
+ 50, 0, 500, 0.1)) : *(0.001) : max(1/SR);
+
+ release = env_group(hslider("[2] Release [unit:ms] [style: knob] [tooltip: Time constant in ms (1/e smoothing time) for the compression gain to approach (exponentially) a new higher target level (the compression 'releasing')]",
+ 500, 0, 1000, 0.1)) : *(0.001) : max(1/SR);
+
+ makeupgain = comp_group(hslider("[5] Makeup Gain [unit:dB] [tooltip: The compressed-signal output level is increased by this amount (in dB) to make up for the level lost due to compression]",
+ 40, -96, 96, 0.1)) : db2linear;
+};
+
+//------------------------------- limiter_* ------------------------------------
+// USAGE:
+// _ : limiter_1176_R4_mono : _;
+// _,_ : limiter_1176_R4_stereo : _,_;
+//
+// DESCRIPTION:
+// A limiter guards against hard-clipping. It can be can be
+// implemented as a compressor having a high threshold (near the
+// clipping level), fast attack and release, and high ratio. Since
+// the ratio is so high, some knee smoothing is
+// desirable ("soft limiting"). This example is intended
+// to get you started using compressor_* as a limiter, so all
+// parameters are hardwired to nominal values here.
+//
+// REFERENCE: http://en.wikipedia.org/wiki/1176_Peak_Limiter
+// Ratios: 4 (moderate compression), 8 (severe compression),
+// 12 (mild limiting), or 20 to 1 (hard limiting)
+// Att: 20-800 MICROseconds (Note: scaled by ratio in the 1176)
+// Rel: 50-1100 ms (Note: scaled by ratio in the 1176)
+// Mike Shipley likes 4:1 (Grammy-winning mixer for Queen, Tom Petty, etc.)
+// Faster attack gives "more bite" (e.g. on vocals)
+// He hears a bright, clear eq effect as well (not implemented here)
+//
+limiter_1176_R4_mono = compressor_mono(4,-6,0.0008,0.5);
+limiter_1176_R4_stereo = compressor_stereo(4,-6,0.0008,0.5);
+
+//========================== Schroeder Reverberators ======================
+
+//------------------------------ jcrev,satrev ------------------------------
+// USAGE:
+// _ : jcrev : _,_,_,_
+// _ : satrev : _,_
+//
+// DESCRIPTION:
+// These artificial reverberators take a mono signal and output stereo
+// (satrev) and quad (jcrev). They were implemented by John Chowning
+// in the MUS10 computer-music language (descended from Music V by Max
+// Mathews). They are Schroeder Reverberators, well tuned for their size.
+// Nowadays, the more expensive freeverb is more commonly used (see the
+// Faust examples directory).
+
+// The reverb below was made from a listing of "RV", dated April 14, 1972,
+// which was recovered from an old SAIL DART backup tape.
+// John Chowning thinks this might be the one that became the
+// well known and often copied JCREV:
+
+jcrev = *(0.06) : allpass_chain <: comb_bank :> _ <: mix_mtx with {
+
+ rev1N = component("filter.lib").rev1;
+
+ rev12(len,g) = rev1N(2048,len,g);
+ rev14(len,g) = rev1N(4096,len,g);
+
+ allpass_chain =
+ rev2(512,347,0.7) :
+ rev2(128,113,0.7) :
+ rev2( 64, 37,0.7);
+
+ comb_bank =
+ rev12(1601,.802),
+ rev12(1867,.773),
+ rev14(2053,.753),
+ rev14(2251,.733);
+
+ mix_mtx = _,_,_,_ <: psum, -psum, asum, -asum : _,_,_,_ with {
+ psum = _,_,_,_ :> _;
+ asum = *(-1),_,*(-1),_ :> _;
+ };
+};
+
+// The reverb below was made from a listing of "SATREV", dated May 15, 1971,
+// which was recovered from an old SAIL DART backup tape.
+// John Chowning thinks this might be the one used on his
+// often-heard brass canon sound examples, one of which can be found at
+// https://ccrma.stanford.edu/~jos/wav/FM_BrassCanon2.wav
+
+satrev = *(0.2) <: comb_bank :> allpass_chain <: _,*(-1) with {
+
+ rev1N = component("filter.lib").rev1;
+
+ rev11(len,g) = rev1N(1024,len,g);
+ rev12(len,g) = rev1N(2048,len,g);
+
+ comb_bank =
+ rev11( 778,.827),
+ rev11( 901,.805),
+ rev11(1011,.783),
+ rev12(1123,.764);
+
+ rev2N = component("filter.lib").rev2;
+
+ allpass_chain =
+ rev2N(128,125,0.7) :
+ rev2N( 64, 42,0.7) :
+ rev2N( 16, 12,0.7);
+};
+
+//-------------------------------- freeverb --------------------------------
+// Freeverb is a widely used, free, open-source Schroeder reverb contributed
+// by ``Jezar at Dreampoint.'' See <faust_distribution>/examples/freeverb.dsp
+
+//=============== Feedback Delay Network (FDN) Reverberators ==============
+
+//-------------------------------- fdnrev0 ---------------------------------
+// Pure Feedback Delay Network Reverberator (generalized for easy scaling).
+//
+// USAGE:
+// <1,2,4,...,N signals> <:
+// fdnrev0(MAXDELAY,delays,BBSO,freqs,durs,loopgainmax,nonl) :>
+// <1,2,4,...,N signals>
+//
+// WHERE
+// N = 2, 4, 8, ... (power of 2)
+// MAXDELAY = power of 2 at least as large as longest delay-line length
+// delays = N delay lines, N a power of 2, lengths perferably coprime
+// BBSO = odd positive integer = order of bandsplit desired at freqs
+// freqs = NB-1 crossover frequencies separating desired frequency bands
+// durs = NB decay times (t60) desired for the various bands
+// loopgainmax = scalar gain between 0 and 1 used to "squelch" the reverb
+// nonl = nonlinearity (0 to 0.999..., 0 being linear)
+//
+// REFERENCE:
+// https://ccrma.stanford.edu/~jos/pasp/FDN_Reverberation.html
+//
+// DEPENDENCIES: filter.lib (filterbank)
+
+fdnrev0(MAXDELAY, delays, BBSO, freqs, durs, loopgainmax, nonl)
+ = (bus(2*N) :> bus(N) : delaylines(N)) ~
+ (delayfilters(N,freqs,durs) : feedbackmatrix(N))
+with {
+ N = count(delays);
+ NB = count(durs);
+//assert(count(freqs)+1==NB);
+ delayval(i) = take(i+1,delays);
+ dlmax(i) = MAXDELAY; // must hardwire this from argument for now
+//dlmax(i) = 2^max(1,nextpow2(delayval(i))) // try when slider min/max is known
+// with { nextpow2(x) = ceil(log(x)/log(2.0)); };
+// -1 is for feedback delay:
+ delaylines(N) = par(i,N,(delay(dlmax(i),(delayval(i)-1))));
+ delayfilters(N,freqs,durs) = par(i,N,filter(i,freqs,durs));
+ feedbackmatrix(N) = bhadamard(N);
+ vbutterfly(n) = bus(n) <: (bus(n):>bus(n/2)) , ((bus(n/2),(bus(n/2):par(i,n/2,*(-1)))) :> bus(n/2));
+ bhadamard(2) = bus(2) <: +,-;
+ bhadamard(n) = bus(n) <: (bus(n):>bus(n/2)) , ((bus(n/2),(bus(n/2):par(i,n/2,*(-1)))) :> bus(n/2))
+ : (bhadamard(n/2) , bhadamard(n/2));
+
+ // Experimental nonlinearities:
+ // nonlinallpass = apnl(nonl,-nonl);
+ // s = nonl*PI;
+ // nonlinallpass(x) = allpassnn(3,(s*x,s*x*x,s*x*x*x)); // filter.lib
+ nonlinallpass = _; // disabled by default (rather expensive)
+
+ filter(i,freqs,durs) = filterbank(BBSO,freqs) : par(j,NB,*(g(j,i)))
+ :> *(loopgainmax) / sqrt(N) : nonlinallpass
+ with {
+ dur(j) = take(j+1,durs);
+ n60(j) = dur(j)*SR; // decay time in samples
+ g(j,i) = exp(-3.0*log(10.0)*delayval(i)/n60(j));
+ // ~ 1.0 - 6.91*delayval(i)/(SR*dur(j)); // valid for large dur(j)
+ };
+};
+
+// ---------- prime_power_delays -----
+// Prime Power Delay Line Lengths
+//
+// USAGE:
+// bus(N) : prime_power_delays(N,pathmin,pathmax) : bus(N);
+//
+// WHERE
+// N = positive integer up to 16
+// (for higher powers of 2, extend 'primes' array below.)
+// pathmin = minimum acoustic ray length in the reverberator (in meters)
+// pathmax = maximum acoustic ray length (meters) - think "room size"
+//
+// DEPENDENCIES:
+// math.lib (SR, selector, take)
+// music.lib (db2linear)
+//
+// REFERENCE:
+// https://ccrma.stanford.edu/~jos/pasp/Prime_Power_Delay_Line.html
+//
+prime_power_delays(N,pathmin,pathmax) = par(i,N,delayvals(i)) with {
+ Np = 16;
+ primes = 2,3,5,7,11,13,17,19,23,29,31,37,41,43,47,53;
+ prime(n) = primes : selector(n,Np); // math.lib
+
+ // Prime Power Bounds [matlab: floor(log(maxdel)./log(primes(53)))]
+ maxdel=8192; // more than 63 meters at 44100 samples/sec & 343 m/s
+ ppbs = 13,8,5,4, 3,3,3,3, 2,2,2,2, 2,2,2,2; // 8192 is enough for all
+ ppb(i) = take(i+1,ppbs);
+
+ // Approximate desired delay-line lengths using powers of distinct primes:
+ c = 343; // soundspeed in m/s at 20 degrees C for dry air
+ dmin = SR*pathmin/c;
+ dmax = SR*pathmax/c;
+ dl(i) = dmin * (dmax/dmin)^(i/float(N-1)); // desired delay in samples
+ ppwr(i) = floor(0.5+log(dl(i))/log(prime(i))); // best prime power
+ delayvals(i) = prime(i)^ppwr(i); // each delay a power of a distinct prime
+};
+
+//--------------------- stereo_reverb_tester --------------------
+// Handy test inputs for reverberator demos below.
+
+stereo_reverb_tester(revin_group,x,y) = inx,iny with {
+ ck_group(x) = revin_group(vgroup("[1] Input Config",x));
+ mutegain = 1 - ck_group(checkbox("[1] Mute Ext Inputs
+ [tooltip: When this is checked, the stereo external audio inputs are disabled (good for hearing the impulse response or pink-noise response alone)]"));
+ pinkin = ck_group(checkbox("[2] Pink Noise
+ [tooltip: Pink Noise (or 1/f noise) is Constant-Q Noise (useful for adjusting the EQ sections)]"));
+
+ impulsify = _ <: _,mem : - : >(0);
+ imp_group(x) = revin_group(hgroup("[2] Impulse Selection",x));
+ pulseL = imp_group(button("[1] Left
+ [tooltip: Send impulse into LEFT channel]")) : impulsify;
+ pulseC = imp_group(button("[2] Center
+ [tooltip: Send impulse into LEFT and RIGHT channels]")) : impulsify;
+ pulseR = imp_group(button("[3] Right
+ [tooltip: Send impulse into RIGHT channel]")) : impulsify;
+
+ inx = x*mutegain + (pulseL+pulseC) + pn;
+ iny = y*mutegain + (pulseR+pulseC) + pn;
+ pn = 0.1*pinkin*component("oscillator.lib").pink_noise;
+};
+
+//------------------------- fdnrev0_demo ---------------------------
+// USAGE: _,_ : fdnrev0_demo(N,NB,BBSO) : _,_
+// WHERE
+// N = Feedback Delay Network (FDN) order
+// = number of delay lines used = order of feedback matrix
+// = 2, 4, 8, or 16 [extend primes array below for 32, 64, ...]
+// NB = number of frequency bands
+// = number of (nearly) independent T60 controls
+// = integer 3 or greater
+// BBSO = Butterworth band-split order
+// = order of lowpass/highpass bandsplit used at each crossover freq
+// = odd positive integer
+
+fdnrev0_demo(N,NB,BBSO,x,y) = stereo_reverb_tester(revin_group,x,y)
+ <: fdnrev0(MAXDELAY,delays,BBSO,freqs,durs,loopgainmax,nonl)
+ :> *(gain),*(gain)
+with {
+ MAXDELAY = 8192; // sync w delays and prime_power_delays above
+ defdurs = (8.4,6.5,5.0,3.8,2.7); // NB default durations (sec)
+ deffreqs = (500,1000,2000,4000); // NB-1 default crossover frequencies (Hz)
+ deflens = (56.3,63.0); // 2 default min and max path lengths
+
+ fdn_group(x) = vgroup("FEEDBACK DELAY NETWORK (FDN) REVERBERATOR, ORDER 16
+ [tooltip: See Faust's effect.lib for documentation and references]", x);
+
+ freq_group(x) = fdn_group(vgroup("[1] Band Crossover Frequencies", x));
+ t60_group(x) = fdn_group(hgroup("[2] Band Decay Times (T60)", x));
+ path_group(x) = fdn_group(vgroup("[3] Room Dimensions", x));
+ revin_group(x) = fdn_group(hgroup("[4] Input Controls", x));
+ nonl_group(x) = revin_group(vgroup("[4] Nonnlinearity",x));
+ quench_group(x) = revin_group(vgroup("[3] Reverb State",x));
+
+ nonl = nonl_group(hslider("[style:knob] [tooltip: nonlinear mode coupling]",
+ 0, -0.999, 0.999, 0.001));
+ loopgainmax = 1.0-0.5*quench_group(button("[1] Quench
+ [tooltip: Hold down 'Quench' to clear the reverberator]"));
+
+ pathmin = path_group(hslider("[1] min acoustic ray length [unit:m]
+ [tooltip: This length (in meters) determines the shortest delay-line used in the FDN reverberator.
+ Think of it as the shortest wall-to-wall separation in the room.]",
+ 46, 0.1, 63, 0.1));
+ pathmax = path_group(hslider("[2] max acoustic ray length [unit:m]
+ [tooltip: This length (in meters) determines the longest delay-line used in the FDN reverberator.
+ Think of it as the largest wall-to-wall separation in the room.]",
+ 63, 0.1, 63, 0.1));
+
+ durvals(i) = t60_group(vslider("[%i] %i [unit:s]
+ [tooltip: T60 is the 60dB decay-time in seconds. For concert halls, an overall reverberation time (T60) near 1.9 seconds is typical [Beranek 2004]. Here we may set T60 independently in each frequency band. In real rooms, higher frequency bands generally decay faster due to absorption and scattering.]",
+ take(i+1,defdurs), 0.1, 10, 0.1));
+ durs = par(i,NB,durvals(NB-1-i));
+
+ freqvals(i) = freq_group(hslider("[%i] Band %i upper edge in Hz [unit:Hz]
+ [tooltip: Each delay-line signal is split into frequency-bands for separate decay-time control in each band]",
+ take(i+1,deffreqs), 100, 10000, 1));
+ freqs = par(i,NB-1,freqvals(i));
+
+ delays = prime_power_delays(N,pathmin,pathmax);
+
+ gain = hslider("[3] Output Level (dB) [unit:dB]
+ [tooltip: Output scale factor]", -40, -70, 20, 0.1) : db2linear;
+ // (can cause infinite loop:) with { db2linear(x) = pow(10, x/20.0); };
+};
+
+//------------------------------- zita_rev_fdn -------------------------------
+// Internal 8x8 late-reverberation FDN used in the FOSS Linux reverb zita-rev1
+// by Fons Adriaensen <fons@linuxaudio.org>. This is an FDN reverb with
+// allpass comb filters in each feedback delay in addition to the
+// damping filters.
+//
+// USAGE:
+// bus(8) : zita_rev_fdn(f1,f2,t60dc,t60m,fsmax) : bus(8)
+//
+// WHERE
+// f1 = crossover frequency (Hz) separating dc and midrange frequencies
+// f2 = frequency (Hz) above f1 where T60 = t60m/2 (see below)
+// t60dc = desired decay time (t60) at frequency 0 (sec)
+// t60m = desired decay time (t60) at midrange frequencies (sec)
+// fsmax = maximum sampling rate to be used (Hz)
+//
+// REFERENCES:
+// http://www.kokkinizita.net/linuxaudio/zita-rev1-doc/quickguide.html
+// https://ccrma.stanford.edu/~jos/pasp/Zita_Rev1.html
+//
+// DEPENDENCIES:
+// filter.lib (allpass_comb, lowpass, smooth)
+// math.lib (hadamard, take, etc.)
+
+zita_rev_fdn(f1,f2,t60dc,t60m,fsmax) =
+ ((bus(2*N) :> allpass_combs(N) : feedbackmatrix(N)) ~
+ (delayfilters(N,freqs,durs) : fbdelaylines(N)))
+with {
+ N = 8;
+
+ // Delay-line lengths in seconds:
+ apdelays = (0.020346, 0.024421, 0.031604, 0.027333, 0.022904,
+ 0.029291, 0.013458, 0.019123); // feedforward delays in seconds
+ tdelays = ( 0.153129, 0.210389, 0.127837, 0.256891, 0.174713,
+ 0.192303, 0.125000, 0.219991); // total delays in seconds
+ tdelay(i) = floor(0.5 + SR*take(i+1,tdelays)); // samples
+ apdelay(i) = floor(0.5 + SR*take(i+1,apdelays));
+ fbdelay(i) = tdelay(i) - apdelay(i);
+ // NOTE: Since SR is not bounded at compile time, we can't use it to
+ // allocate delay lines; hence, the fsmax parameter:
+ tdelaymaxfs(i) = floor(0.5 + fsmax*take(i+1,tdelays));
+ apdelaymaxfs(i) = floor(0.5 + fsmax*take(i+1,apdelays));
+ fbdelaymaxfs(i) = tdelaymaxfs(i) - apdelaymaxfs(i);
+ nextpow2(x) = ceil(log(x)/log(2.0));
+ maxapdelay(i) = int(2.0^max(1.0,nextpow2(apdelaymaxfs(i))));
+ maxfbdelay(i) = int(2.0^max(1.0,nextpow2(fbdelaymaxfs(i))));
+
+ apcoeff(i) = select2(i&1,0.6,-0.6); // allpass comb-filter coefficient
+ allpass_combs(N) =
+ par(i,N,(allpass_comb(maxapdelay(i),apdelay(i),apcoeff(i)))); // filter.lib
+ fbdelaylines(N) = par(i,N,(delay(maxfbdelay(i),(fbdelay(i)))));
+ freqs = (f1,f2); durs = (t60dc,t60m);
+ delayfilters(N,freqs,durs) = par(i,N,filter(i,freqs,durs));
+ feedbackmatrix(N) = hadamard(N); // math.lib
+
+ staynormal = 10.0^(-20); // let signals decay well below LSB, but not to zero
+
+ special_lowpass(g,f) = smooth(p) with {
+ // unity-dc-gain lowpass needs gain g at frequency f => quadratic formula:
+ p = mbo2 - sqrt(max(0,mbo2*mbo2 - 1.0)); // other solution is unstable
+ mbo2 = (1.0 - gs*c)/(1.0 - gs); // NOTE: must ensure |g|<1 (t60m finite)
+ gs = g*g;
+ c = cos(2.0*PI*f/float(SR));
+ };
+
+ filter(i,freqs,durs) = lowshelf_lowpass(i)/sqrt(float(N))+staynormal
+ with {
+ lowshelf_lowpass(i) = gM*low_shelf1_l(g0/gM,f(1)):special_lowpass(gM,f(2));
+ low_shelf1_l(G0,fx,x) = x + (G0-1)*lowpass(1,fx,x); // filter.lib
+ g0 = g(0,i);
+ gM = g(1,i);
+ f(k) = take(k,freqs);
+ dur(j) = take(j+1,durs);
+ n60(j) = dur(j)*SR; // decay time in samples
+ g(j,i) = exp(-3.0*log(10.0)*tdelay(i)/n60(j));
+ };
+};
+
+// Stereo input delay used by zita_rev1 in both stereo and ambisonics mode:
+zita_in_delay(rdel) = zita_delay_mono(rdel), zita_delay_mono(rdel) with {
+ zita_delay_mono(rdel) = delay(8192,SR*rdel*0.001) * 0.3;
+};
+
+// Stereo input mapping used by zita_rev1 in both stereo and ambisonics mode:
+zita_distrib2(N) = _,_ <: fanflip(N) with {
+ fanflip(4) = _,_,*(-1),*(-1);
+ fanflip(N) = fanflip(N/2),fanflip(N/2);
+};
+
+//--------------------------- zita_rev_fdn_demo ------------------------------
+// zita_rev_fdn_demo = zita_rev_fdn (above) + basic GUI
+//
+// USAGE:
+// bus(8) : zita_rev_fdn_demo(f1,f2,t60dc,t60m,fsmax) : bus(8)
+//
+// WHERE
+// (args and references as for zita_rev_fdn above)
+
+zita_rev_fdn_demo = zita_rev_fdn(f1,f2,t60dc,t60m,fsmax)
+with {
+ fsmax = 48000.0;
+ fdn_group(x) = hgroup(
+ "Zita_Rev Internal FDN Reverb [tooltip: ~ Zita_Rev's internal 8x8 Feedback Delay Network (FDN) & Schroeder allpass-comb reverberator. See Faust's effect.lib for documentation and references]",x);
+ t60dc = fdn_group(vslider("[1] Low RT60 [unit:s] [style:knob]
+ [style:knob]
+ [tooltip: T60 = time (in seconds) to decay 60dB in low-frequency band]",
+ 3, 1, 8, 0.1));
+ f1 = fdn_group(vslider("[2] LF X [unit:Hz] [style:knob]
+ [tooltip: Crossover frequency (Hz) separating low and middle frequencies]",
+ 200, 50, 1000, 1));
+ t60m = fdn_group(vslider("[3] Mid RT60 [unit:s] [style:knob]
+ [tooltip: T60 = time (in seconds) to decay 60dB in middle band]",
+ 2, 1, 8, 0.1));
+ f2 = fdn_group(vslider("[4] HF Damping [unit:Hz] [style:knob]
+ [tooltip: Frequency (Hz) at which the high-frequency T60 is half the middle-band's T60]",
+ 6000, 1500, 0.49*fsmax, 1));
+};
+
+//---------------------------- zita_rev1_stereo ---------------------------
+// Extend zita_rev_fdn to include zita_rev1 input/output mapping in stereo mode.
+//
+// USAGE:
+// _,_ : zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax) : _,_
+//
+// WHERE
+// rdel = delay (in ms) before reverberation begins (e.g., 0 to ~100 ms)
+// (remaining args and refs as for zita_rev_fdn above)
+
+zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax) =
+ zita_in_delay(rdel)
+ : zita_distrib2(N)
+ : zita_rev_fdn(f1,f2,t60dc,t60m,fsmax)
+ : output2(N)
+with {
+ N = 8;
+ output2(N) = outmix(N) : *(t1),*(t1);
+ t1 = 0.37; // zita-rev1 linearly ramps from 0 to t1 over one buffer
+ outmix(4) = !,butterfly(2),!; // probably the result of some experimenting!
+ outmix(N) = outmix(N/2),par(i,N/2,!);
+};
+
+//----------------------------- zita_rev1_ambi ---------------------------
+// Extend zita_rev_fdn to include zita_rev1 input/output mapping in
+// "ambisonics mode", as provided in the Linux C++ version.
+//
+// USAGE:
+// _,_ : zita_rev1_ambi(rgxyz,rdel,f1,f2,t60dc,t60m,fsmax) : _,_,_,_
+//
+// WHERE
+// rgxyz = relative gain of lanes 1,4,2 to lane 0 in output (e.g., -9 to 9)
+// (remaining args and references as for zita_rev1_stereo above)
+
+zita_rev1_ambi(rgxyz,rdel,f1,f2,t60dc,t60m,fsmax) =
+ zita_in_delay(rdel)
+ : zita_distrib2(N)
+ : zita_rev_fdn(f1,f2,t60dc,t60m,fsmax)
+ : output4(N) // ambisonics mode
+with {
+ N=8;
+ output4(N) = select4 : *(t0),*(t1),*(t1),*(t1);
+ select4 = _,_,_,!,_,!,!,! : _,_,cross with { cross(x,y) = y,x; };
+ t0 = 1.0/sqrt(2.0);
+ t1 = t0 * 10.0^(0.05 * rgxyz);
+};
+
+//---------------------------------- zita_rev1 ------------------------------
+// Example GUI for zita_rev1_stereo (mostly following the Linux zita-rev1 GUI).
+//
+// Only the dry/wet and output level parameters are "dezippered" here. If
+// parameters are to be varied in real time, use "smooth(0.999)" or the like
+// in the same way.
+//
+// REFERENCE:
+// http://www.kokkinizita.net/linuxaudio/zita-rev1-doc/quickguide.html
+//
+// DEPENDENCIES:
+// filter.lib (peak_eq_rm)
+
+zita_rev1(x,y) = 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
+
+ fdn_group(x) = hgroup(
+ "[0] Zita_Rev1 [tooltip: ~ ZITA REV1 FEEDBACK DELAY NETWORK (FDN) & SCHROEDER ALLPASS-COMB REVERBERATOR (8x8). See Faust's effect.lib for documentation and references]", x);
+
+ in_group(x) = fdn_group(hgroup("[1] Input", x));
+
+ rdel = in_group(vslider("[1] In Delay [unit:ms] [style:knob]
+ [tooltip: Delay in ms before reverberation begins]",
+ 60,20,100,1));
+
+ freq_group(x) = fdn_group(hgroup("[2] Decay Times in Bands (see tooltips)", x));
+
+ f1 = freq_group(vslider("[1] LF X [unit:Hz] [style:knob]
+ [tooltip: Crossover frequency (Hz) separating low and middle frequencies]",
+ 200, 50, 1000, 1));
+
+ t60dc = freq_group(vslider("[2] Low RT60 [unit:s] [style:knob]
+ [style:knob] [tooltip: T60 = time (in seconds) to decay 60dB in low-frequency band]",
+ 3, 1, 8, 0.1));
+
+ t60m = freq_group(vslider("[3] Mid RT60 [unit:s] [style:knob]
+ [tooltip: T60 = time (in seconds) to decay 60dB in middle band]",
+ 2, 1, 8, 0.1));
+
+ f2 = freq_group(vslider("[4] HF Damping [unit:Hz] [style:knob]
+ [tooltip: Frequency (Hz) at which the high-frequency T60 is half the middle-band's T60]",
+ 6000, 1500, 0.49*fsmax, 1));
+
+ out_eq = pareq_stereo(eq1f,eq1l,eq1q) : pareq_stereo(eq2f,eq2l,eq2q);
+// Zolzer style peaking eq (not used in zita-rev1) (filter.lib):
+// pareq_stereo(eqf,eql,Q) = peak_eq(eql,eqf,eqf/Q), peak_eq(eql,eqf,eqf/Q);
+// Regalia-Mitra peaking eq with "Q" hard-wired near sqrt(g)/2 (filter.lib):
+ pareq_stereo(eqf,eql,Q) = peak_eq_rm(eql,eqf,tpbt), peak_eq_rm(eql,eqf,tpbt)
+ with {
+ tpbt = wcT/sqrt(max(0,g)); // tan(PI*B/SR), B bw in Hz (Q^2 ~ g/4)
+ wcT = 2*PI*eqf/SR; // peak frequency in rad/sample
+ g = db2linear(eql); // peak gain
+ };
+
+ eq1_group(x) = fdn_group(hgroup("[3] RM Peaking Equalizer 1", x));
+
+ eq1f = eq1_group(vslider("[1] Eq1 Freq [unit:Hz] [style:knob]
+ [tooltip: Center-frequency of second-order Regalia-Mitra peaking equalizer section 1]",
+ 315, 40, 2500, 1));
+
+ eq1l = eq1_group(vslider("[2] Eq1 Level [unit:dB] [style:knob]
+ [tooltip: Peak level in dB of second-order Regalia-Mitra peaking equalizer section 1]",
+ 0, -15, 15, 0.1));
+
+ eq1q = eq1_group(vslider("[3] Eq1 Q [style:knob]
+ [tooltip: Q = centerFrequency/bandwidth of second-order peaking equalizer section 1]",
+ 3, 0.1, 10, 0.1));
+
+ eq2_group(x) = fdn_group(hgroup("[4] RM Peaking Equalizer 2", x));
+
+ eq2f = eq2_group(vslider("[1] Eq2 Freq [unit:Hz] [style:knob]
+ [tooltip: Center-frequency of second-order Regalia-Mitra peaking equalizer section 2]",
+ 315, 40, 2500, 1));
+
+ eq2l = eq2_group(vslider("[2] Eq2 Level [unit:dB] [style:knob]
+ [tooltip: Peak level in dB of second-order Regalia-Mitra peaking equalizer section 2]",
+ 0, -15, 15, 0.1));
+
+ eq2q = eq2_group(vslider("[3] Eq2 Q [style:knob]
+ [tooltip: Q = centerFrequency/bandwidth of second-order peaking equalizer section 2]",
+ 3, 0.1, 10, 0.1));
+
+ out_group(x) = fdn_group(hgroup("[5] Output", x));
+
+ dry_wet(x,y) = *(wet) + dry*x, *(wet) + dry*y with {
+ wet = 0.5*(drywet+1.0);
+ dry = 1.0-wet;
+ };
+
+ drywet = out_group(vslider("[1] Dry/Wet Mix [style:knob]
+ [tooltip: -1 = dry, 1 = wet]",
+ 0, -1.0, 1.0, 0.01)) : smooth(0.999);
+
+ out_level = *(gain),*(gain);
+
+ gain = out_group(vslider("[2] Level [unit:dB] [style:knob]
+ [tooltip: Output scale factor]", -20, -70, 40, 0.1))
+ : smooth(0.999) : db2linear;
+
+};
+
+//---------------------------------- mesh_square ------------------------------
+// Square Rectangular Digital Waveguide Mesh
+//
+// USAGE:
+// bus(4*N) : mesh_square(N) : bus(4*N);
+//
+// WHERE
+// N = number of nodes along each edge - a power of two (1,2,4,8,...)
+//
+// EXAMPLE: Reflectively terminated mesh impulsed at one corner:
+// mesh_square_test(N,x) = mesh_square(N)~(busi(4*N,x)) // input to corner
+// with { busi(N,x) = bus(N) : par(i,N,*(-1)) : par(i,N-1,_), +(x); };
+// process = 1-1' : mesh_square_test(4); // all modes excited forever
+//
+// REQUIRES: math.lib.
+//
+// REFERENCE:
+// https://ccrma.stanford.edu/~jos/pasp/Digital_Waveguide_Mesh.html
+
+// four-port scattering junction:
+mesh_square(1)
+ = bus(4) <: par(i,4,*(-1)), (bus(4) :> (*(.5)) <: bus(4)) :> bus(4);
+
+// rectangular NxN square waveguide mesh:
+mesh_square(N) = bus(4*N) : (route_inputs(N/2) : par(i,4,mesh_square(N/2)))
+ ~(prune_feedback(N/2))
+ : prune_outputs(N/2) : route_outputs(N/2) : bus(4*N)
+with {
+ block(N) = par(i,N,!);
+
+ // select block i of N, block size = M:
+ s(i,N,M) = par(j, M*N, Sv(i, j))
+ with { Sv(i,i) = bus(N); Sv(i,j) = block(N); };
+
+ // prune mesh outputs down to the signals which make it out:
+ prune_outputs(N)
+ = bus(16*N) :
+ block(N), bus(N), block(N), bus(N),
+ block(N), bus(N), bus(N), block(N),
+ bus(N), block(N), block(N), bus(N),
+ bus(N), block(N), bus(N), block(N)
+ : bus(8*N);
+
+ // collect mesh outputs into standard order (N,W,E,S):
+ route_outputs(N)
+ = bus(8*N)
+ <: s(4,N,8),s(5,N,8), s(0,N,8),s(2,N,8),
+ s(3,N,8),s(7,N,8), s(1,N,8),s(6,N,8)
+ : bus(8*N);
+
+ // collect signals used as feedback:
+ prune_feedback(N) = bus(16*N) :
+ bus(N), block(N), bus(N), block(N),
+ bus(N), block(N), block(N), bus(N),
+ block(N), bus(N), bus(N), block(N),
+ block(N), bus(N), block(N), bus(N) :
+ bus(8*N);
+
+ // route mesh inputs (feedback, external inputs):
+ route_inputs(N) = bus(8*N), bus(8*N)
+ <:s(8,N,16),s(4,N,16), s(12,N,16),s(3,N,16),
+ s(9,N,16),s(6,N,16), s(1,N,16),s(14,N,16),
+ s(0,N,16),s(10,N,16), s(13,N,16),s(7,N,16),
+ s(2,N,16),s(11,N,16), s(5,N,16),s(15,N,16)
+ : bus(16*N);
+};
projects / Faustine.git / blobdiff