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 /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 /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 /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 . 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); };