Hello Stéphane and all, here is the finalized code. ( Pickup paramter hidden, highpass removed, reverb replaced with parameterless Dattorro default, more clear parameter names and description, rearranged order of convolution tabs and distortion saves CPU, and a resonant EQ added for weak formants).
From my side it could now be added to the examples, or used and altered in any way. Gabriel [code] // Modulation synthesis with sparse convolution filter and distortions. // ------------------------------------------------------------------- // // A "3D" oscillator oscillating on x,y,z axis, with the radius of x,y used as waveform, // very similar and related to FM / AM. // // The y axis oscillation is set by MIDI pitch, x and z are detuned by simple just tuned ratios. // Feedback acts on the individual sine oscillations (giving a sawtooth like waveform). // // Three weighted copies with time varying shifts are summed in a lossy integrator // (sparse convolution), followed by a peak resonance filter and shaped by // an internal pick-up like distortion and an asymmetric polynomial. // // The convolution tabs give a (variyng) triangle impulse response if integrated twice, // with a -12 dB/octave rolloff and regular notches. // Here only one integrator is used. // // The envelope is hard wired to the oscillation amplitudes and the rise time of the filter. // // An LFO is wired to pitch. // // A resonant EQ and the Dattoro Reverb from the Faust libary are added as effect on the sum. // // // Inspired by the history of sound synthesis, namely Trautonium, Mini Moog, Phase Modulation Synthesis, // Variophon Wind Instrument Synthesizer, Physical Modeling, and the work of Thomas D. Rossing. // // References: // Kot, Vítězslav. (2006). DIGITAL SOUND EFFECTS ECHO AND REVERB BASED ON NON EXPONENTIALLY DECAYING COMB FILTER. // https://en.wikipedia.org/wiki/Variophon // Parker, Julian & Zavalishin, Vadim & Le Bivic, Efflam. (2016). Reducing The Aliasing Of Nonlinear Waveshaping Using Continuous-Time Convolution. // Nicholas G. Horton, Thomas R. Moore. (2008). Modelling The Magnetic Pickup Of An Electric Guitar. // https://www.musicdsp.org/en/latest/Effects/86-waveshaper-gloubi-boulga.html, see comment from 2005-09-22 01:07:58 // Frei, Beat. Digital Sound Generation I & II, ICST Zurich University of the Arts // Smith, J.O. Physical Audio Signal Processing,http://ccrma.stanford.edu/~jos/pasp/, online book, 2010 edition declare options "[midi:on][nvoices:8]"; declare options "[-vec]"; declare name "Paradigma_9"; declare version "1.0"; declare author "gabriel"; import("stdfaust.lib"); // Frequency Ratios table frtonum = waveform{1,16,9,6,5,4,7,3,8,5,7,15}; frtodiv = waveform{1,15,8,5,4,3,5,2,5,3,4, 8}; // MIDI // minimum velocity minvelo = 1 / 32; midigrp(x) = hgroup("[1]MIDI",x); f = nentry("freq[hidden:1]",200,40,2000,0.1); kmidi = nentry("key[hidden:1]",69,0,127,1); bend = ba.semi2ratio(hslider("bend[hidden:1][midi:pitchwheel][style: knob]",0,-2,2,0.01)); gain = nentry("gain[hidden:1]",0.6,0,1,0.01)<:* : _*(1-minvelo):_+ minvelo; master = hslider("volume[midi:ctrl 7]",0.6,0,1,0.01); gate = button("gate[hidden:1]") ; // Oscillator Parameter rtogrp(x) = hgroup("[2]Oscillator",x); rto1sel = rtogrp(hslider("[1]x[style:knob]",-12,-36,36,1)); rto2sel = rtogrp(hslider("[2]z[style:knob]",19,-36,36,1)); fbka = rtogrp(hslider("[3]Feedback[style:knob]",0.15,0,1,0.01)<:*:*(1/ma.PI)); detune = rtogrp(hslider("[4]Detune[style:knob]",0.125,0,0.5,0.005)/ma.SR); // LFO and Envelope Parameter lfogrp(x) = hgroup("[3]Envelope & LFO",x); enva = (lfogrp(ba.db2linear(hslider("[1]A[style:knob]",20,15,66,1) )/1000)); envd = (lfogrp(ba.db2linear(hslider("[2]D[style:knob]",74,26,100,1) )/1000)*envpscal); envs = (lfogrp(hslider("[3]S[style:knob]",0,0,1,0.01) )); envr = (lfogrp(ba.db2linear(hslider("[4]R[style:knob]",50,26,100,1) )/1000)*envpscal); lfof = lfogrp(hslider("[5]LFO Hz[style:knob]",3,0.1,12,0.1)); lfvibra = lfogrp(hslider("[6]Vibrato[style:knob]",0.125,0,1,0.01))<:*; env = en.adsre(enva,envd*envpscal,envs,envr*envpscal,gate); envg = env:_* gain; lfosn = qsin(mphasor(lfof/ma.SR)); // Triangular Filter Parameter fltgrp(x) = hgroup("[4]Filter",x); wid = fltgrp(hslider("[1]Rise[style:knob]",3,1,9,0.001)):2^_:1/_; edge = fltgrp(hslider("[2]Fall[style:knob]",6,1,9,0.001)):2^_:1/_; fiq = fltgrp(hslider("[3]Q[style:knob]",1,0.5,3.87,0.01))<:*; drive = fltgrp(hslider("[4]Drive[style:knob]",-12,-12,30,0.1)):_/20.0:10^_; // Modulation Frequency Ratios rto1oct = rto1sel / 12 : floor; rto1semi = rto1sel + 36 : _% 12; rto1a = frtonum, rto1semi : rdtable; rto1b = frtodiv, rto1semi : rdtable; rto1 = (rto1a/rto1b)*(2^rto1oct); rto1r = min((1/ rto1),1); rto2oct = rto2sel / 12 : floor; rto2semi = rto2sel + 36 : _% 12; rto2a = frtonum, rto2semi : rdtable; rto2b = frtodiv, rto2semi : rdtable; rto2 = rto1*(rto2a/rto2b)*(2^rto2oct); rto2r = min((1 / rto2),1); // Pitch lg2f = ma.log2(f/440); stretch = 0.0333*lg2f; envpscal = ( - 3 * lg2f ):ba.db2linear; fplus = f*bend + lfosn* lfvibra*f * 0.5/12*envg + stretch; w = f/ma.SR; w2 = rto1 * w; w3 = rto2 * w; wplus = fplus/ma.SR; fbk1 = fbka*(0.5 -w)^4; fbk2 = fbka*(0.5 - w2)^4*rto1r; fbk3 = fbka*(0.5 - w3)^4*rto2r; // Modulation Reduction Per Frequency redux1 = ((3.3 -((rto1+1)*w) )/3.3),0: max:_^3; redux2 = ((3.3 -((rto2+1)*w) )/3.3),0: max:_^3; modep = envg; modep1 = envg * redux1 *rto1r * gain ; modep2 = envg * redux2 *rto2r * gain ; // Sine Oscillator wrap(n) = n-( floor( n +0.5)) ; // Bhaskara I based approximate sine curve qsincurve(x) = 1 - ( (x*x)<: *(1.2253517*16),(_<:*:* (-3.60562732*16)):>_ ); qsin(x) = x+(0.5): wrap <: (abs:-(0.25):qsincurve),_:ma.copysign; // Feedback Depth Reduction Curve fbcurve (x)= x:abs:-(1) <:^(3):_,(x):ma.copysign; // Oscillator mphasor(fw) = (+(fw) ~ (wrap)); oscsn(fw, off) = mphasor(fw) + off:qsin:+~*(0.5); osc1(fw, off) = ((fw),+(off):(oscsn)) ~ (*(fbk2):fi.pole(0.5):_*fbcurve(fw)); // 3D to 2D radius oscy(fw, off) = (osc1(fw, off )*osc1(fw*rto2+2*detune,0.75 + off)*modep2)*modep; oscx(fw, off) = (osc1(fw*rto1+detune,0.25 + off)*osc1(fw*rto2+2*detune,0.25 + off)*modep2)*modep1; oscxy(fw, off) = (oscy(fw, off)<:*),(oscx(fw, off)<:*):+:sqrt; // Pick-Up like Distortion // distance : pickd = 0.25; pickup(x, pickd) = x, // normal for in < 1.2e-4 ( x,(x^2:_+pickd:_^(3/2)):/ ), // ILO: ( pickd^(3/2) / ( sqrt(x*x + 1)):ma.neg:_+ pickd^(3/2) ): // select ba.if( (_:abs:_<= 1.2e-4), _, _ ):_*(pickd^(4/3)); // Basic Synthvoice, modulated Oscillations synthvox(fw, ph2, ph3, g1, g2, g3) = (oscxy(fw, 0):_*g1), (oscxy(fw, ph2):_*g2),(oscxy(fw, ph3):_*g3):>_ : fi.zero(1.0)<:_,(pickd):pickup; // Triangle // reduce width with frequency widredux = w <:+:_^3:1.0-_; // diff to max f in octaves, reduced for higher octaves dwo = ( 0.25 / wid ):max(_, 1): ma.log2: ma.inv: _*widredux: ma.inv; // falling edge egderto = edge / wid; wid2 = wid * (2^(dwo * (1-envg ))): _* (2^(dwo * (1- gain ) )): min( _, 0.25): max( _, 4 / (ma.SR/fplus)); wid2e = edge: min( _, 0.25): max( _, 4 /(ma.SR/fplus)); fiw = wplus/wid2; fiwtail = wplus/wid2e; // triangle coefficients apg0 = fiw; apg1 = - apg0 - fiwtail; apg2 = fiwtail; // integration freq igpole = 1.0-5.0/ma.SR; resf = (fplus /( wid2 +wid2e) ): min( _, (0.249 * ma.SR)); // Asymmetric Shaper x - 0.15x²-0.15x³ tubicclip = _:min(_, (1.19419)):max(_,(-1.86086)); tubicilo(x) = x, // normal for in < 1.2e-4 ( x - 0.15*(x^2)-0.15*(x^3) ), // ILO: (( 0.5*(x^2) - 0.05*(x^3) - 0.0375*(x^4) ),(x <:_,_':- :_<:(abs:max(_,1.2e-4)),(ma.signum):ma.copysign):/): // select ba.if( (_:abs:_<= 1.2e-4), _, _ ):fi.dcblockerat(10.0); // Sound process = synthvox(wplus, wid2, wid2e, apg0, apg1, apg2): fi.dcblockerat(10.0): fi.pole(igpole) : fi.svf.peak( resf, fiq) : _*drive: tubicclip: tubicilo:_*(1/drive); effect = _ * master:preeq( lsf,lsgain, b1f, eqq2, eqg2, b2f, eqq3, eqg3, hsf, hsgain) <:_,_:re.dattorro_rev_default; // --------------------------------------------------------------------------------------------------------------- // Resonant EQ for instrument corpus, based on SVF eqgrp(x) = hgroup("[5]EQ",x); lsgain = eqgrp(hslider("[1]Low Gain[style:knob]",3,-18,18,0.25)); b1f = eqgrp(hslider("[2]Split F Low[style:knob]",-0.5,-1,1,0.05)) :(2.0)^_:_*360; b1gain = eqgrp(hslider("[3]Band 1 Gain[style:knob]",4.5,-18,18,0.25)); b2f = eqgrp(hslider("[4]Split F Hi[style:knob]",-0.5,-1,1,0.05)):(2.0)^_:_*720; b2gain = eqgrp(hslider("[5]Band 2 Gain[style:knob]",4.5,-18,18,0.25)); hsgain = eqgrp(hslider("[6]Hi Gain[style:knob]",-3,-18,18,0.25)); // Q and gain of middle bands are scaled simultanously, with saturation curve on gain, // max gain is reduced from the output. Bands are spaced in octaves by default. lsf = b1f *0.5; hsf = b2f * 2; gcurve( gain, gainrange) = abs(gain/gainrange) <:*:1-_:_+1:_*0.5; qscal( gain, gainrange) = 1.414 * ( abs(gain/ gainrange)) :_+ 1.414; eqq2 = qscal( b1gain, 18.0); eqg2 = b1gain * gcurve( b1gain, 18.0); eqq3 = qscal( b2gain, 18.0); eqg3 = b2gain * gcurve( b2gain, 18.0); eqredux = max( lsgain, eqg2):max(_, eqg3):max(_,hsgain):ba.db2linear: ma.inv; preeq( f1,g1,f2,q2,g2,f3,q3,g3,f4,g4) = _*eqredux:fi.svf.bell( f1, 1.414, g1):fi.svf.bell( f2, q2, g2): fi.svf.bell( f3, q3, g3):fi.svf.hs( f4, 1.414, g4); [/code] On Thu, Sep 11, 2025 at 3:25 PM ga <gvoxangel...@gmail.com> wrote: > > Thanks, this would be a great solution. > > Meanwhile I collected the references for the relevant parts (below, if > someone is interested ). > The unfinished reverb will be omitted for simplicity and clarity, pickup > replaced. > I will take some time for the changes, if someone finds a part in the code > the should be either more concise or > more verbose, or a msitake, let me know. > I also realized that I rewrote some things that seem to have equivalents in > the library, I will most likely replace them. > > Gabriel > > // Modulation synthesis with sparse convolution filter and distortions. > // ==================================================================== > // > // A "3D" oscillator oscillating on x,y,z axis, with the radius of x,y going > // into a hyperbolic distortion. > // > // The y axis oscillation is set by MIDI pitch, x and z are detuned by simple > just tuned ratios. > // Feedback acts on the individual sine oscillations. > // > // > // Three weighted copies with time varying shifts are summed in a lossy > integrator > // ( sparse convolution ), followed by a peak filter and shaped by an > asymmetric polynomial. > // > // The convolution tabs would give a (variyng) triangle impulse response if > integrated twice, > // with a -12 dB/octave rolloff and varying regular notches. > // Here only one integrator is used. > // > // > // The envelope is hard wired to the oscillation amplitudes and the rise time > of the filter. > // > // An LFO is wired to pitch. > // > // > // Inspired by the history of sound synthesis, namely Trautonium, Mini Moog, > Phase Modulation Synthesis, > // Variophon Wind Instrument Synthesizer, Physical Modeling, and the work of > Thams D. Rossing. > // > // References: > // Kot, Vítězslav. (2006). DIGITAL SOUND EFFECTS ECHO AND REVERB BASED ON NON > EXPONENTIALLY DECAYING COMB FILTER. > // https://en.wikipedia.org/wiki/Variophon > // Parker, Julian & Zavalishin, Vadim & Le Bivic, Efflam. (2016). Reducing > The Aliasing Of Nonlinear Waveshaping Using Continuous-Time Convolution. > // Nicholas G. Horton, Thomas R. Moore. (2008). Modelling The Magnetic Pickup > Of An Electric Guitar. > // > https://www.musicdsp.org/en/latest/Effects/86-waveshaper-gloubi-boulga.html, > see comment from 2005-09-22 01:07:58 > // Frei, Beat. Digital Sound Generation I & II, ICST Zurich University of the > Arts > // Smith, J.O. Physical Audio Signal > Processing,http://ccrma.stanford.edu/~jos/pasp/, online book, 2010 edition > > On Thu, Sep 11, 2025 at 10:57 AM Stéphane Letz <l...@grame.fr> wrote: >> >> Hi Gabriel, >> >> I suggest we do it simple for now. If you can cleanup and document the DSP >> code, then I can put in the examples/misc section: >> https://faustdoc.grame.fr/examples/#misc >> >> Thanks. >> >> Stéphane >> >> >> > Le 10 sept. 2025 à 18:43, ga <gvoxangel...@gmail.com> a écrit : >> > >> > Thanks >> > >> > I will look into installing Faust locally, I am bit deterred by the vast >> > amount of dependencies >> > and my little experience with installing such projects. >> > >> > I also don't have much experience with make and compiling and C, >> > but I think faust2rpialsaconsole might be onther option I have to look into >> > as running it on a Pi seems a reasonable solution for hardware. >> > I do have a Pi 400, on which unfortunately the Patch OS which might be a >> > good choice for OS does not run (or I didnt get it to run ). >> > >> > to 4) >> > The code and concept is public and libre from my side, but maybe licenses >> > of third parties have to be considered. >> > >> > So I reused and altered code from the Faust library ( by Julius Smith I >> > think ) for the allpass delay, >> > and the idea for the triangular filter was originally inspired by the >> > historic Variophon triangular oscillator, etc. >> > so at least a proper note with history and references would be desireable. >> > Since the concept has a really long history with many sources and >> > variants, and is floating on my desk since years, >> > it's a bit difficult to be accurate in this regards, and to do this >> > justice. >> > >> > The code also still needs some minor tweaks and cosmetic changes before it >> > is released in a 'final' version. >> > For instance it uses two SVFs in series at the moment with very similar >> > corner frequencies, >> > which could probably be replaced by a single SVF with a 'morphing' output. >> > >> > A previous version had roughly antialiased synched noise (windowed with a >> > quarter sine wave) in a addition to the osciallator, >> > to mimick a corpus impulse response, and to enhance piano and string >> > reminiscent sounds. >> > >> > I now tried to replace this with short allpass delays but it sounds less >> > convincing and "boxed", and setting >> > the length of the allpass chain is also too arbitrary att the moment. >> > >> > Also noise has the interesting property that it has fluctuations, so a >> > seed could be matched >> > to produce a sequence that resembles the derivative (or 2nd derivative) of >> > a real corpus impulse response. >> > >> > I would like to keep the paramter set to 4 though, as the idea for a >> > hardware interface is to have >> > two rows with 4 push and turn encoders each, one row for synth and one for >> > EQ and other effects, >> > with each encoder serving also as a button to select a set of 4 parameters >> > that belong together, like ADSR. >> > ( sketch : >> > https://assets.steadyhq.com/production/post/c0d7b8ae-4d1f-4afa-afe8-8bcce17883ac/uploads/images/5prochhxdb/UI.jpg?auto=compress&w=800&fit=max&dpr=2&fm=webp) >> > >> > (Pressing two encoders in the corners simultanously could be used for >> > saveing and laoding presets.) >> > >> > This is one reason why the noise was omitted in this version. >> > >> > Such a controller should be seperate from the computing hardware and be >> > useful for many things, >> > and could be easy to build from two I²C breakout boards from Adafruit, >> > but I do not have the tools and funds for this at the moment, and it >> > requires >> > additional code for interfacing, which I do not have experience with. >> > >> > The idea defintively is to make it an all open source and somewhat >> > flexible synth concept. >> > >> > An interesting aspect for me is that it touches and fuses many aspects of >> > the history of synthesis, >> > and synthesis approaches, starting with the Trautonium, modulation >> > synthesis, subtractive, >> > aspects and findings of physical modeling, etc, in a very compact but >> > meaningful parameter set and combination. >> > ( less paramter than a Mini Moog I think, from which it also borroughs of >> > course). >> > >> > By this it is also a good simplified model to learn and teach I think, for >> > instance you could >> > examine what makes a sound "pianoide" and then expand on this with real >> > pianos and real accurate modeling >> > of real phyiscal forces etc., and then again examine their perceptual >> > significance and compare to this "cartoon" >> > version, and many similar things. >> > >> > I dont know whats the best way to publish this so others can contribute >> > and expand on this. >> > Maintaining and ovreseeing a project on Sourceforge or similar requires a >> > lot of work and energy and experience >> > which I do not have. >> > So I am also looking for interested people I can hand this idea over, >> > Including it with Faust examples would be interesting in this regards, >> > but I am not sure it is fundamental and also simple enough for this, etc. >> > >> > Gabriel >> > >> > >> > >> > >> > On Wed, Sep 10, 2025 at 3:14 PM Stéphane Letz <l...@grame.fr> wrote: >> > Hi, >> > >> > Thanks for this interesting code. For exporting the code, you have several >> > options: >> > >> > 1) exporting the DSP for a standard plugin format. >> > >> > - you can possibly use the JUCE export for that, as an >> > intermediate step: >> > https://github.com/grame-cncm/faust/tree/master-dev/architecture/juce. For >> > maximal flexibility the best would be to compile and install a local Faust >> > version. >> > >> > - another option is to use the Fadeli project >> > https://github.com/DISTRHO/Fadeli >> > >> > 2) you may find more info on this page >> > https://faust.grame.fr/community/powered-by-faust >> > >> > 3) you can connect to the Faust developer/user community on Discord >> > channel, see https://faust.grame.fr/community/help/ >> > >> > 4) You wrote « I am proposing the attached synthesis engine. » : Is the >> > code public ? Are you interested to contribute it in the Faust examples: >> > https://faustdoc.grame.fr/examples/ >> > >> > Thanks. >> > >> > Stéphane >> > >> > >> > > Le 4 sept. 2025 à 11:53, ga <gvoxangel...@gmail.com> a écrit : >> > > >> > > Hello >> > > I am proposing the attached synthesis engine. >> > > It uses a "3D" oscillator that oscillates in x,y, z ( similar to FM / AM) >> > > The radius of x,y is fed into a pickup distortion, which goes into a >> > > triangular filter ( 3 phase offset copies going into an integrator) an >> > > asymmetric distortion. >> > > It has only 4× 4 parameter, including classic ADSR and LFO, envelope >> > > hardwired to oscillation amplitudes. >> > > Its capable of a variety of semi- realistic sounds. >> > > Sound demo is here: >> > > https://youtu.be/7CBhMcYDWac?feature=shared >> > > >> > > >> > > I would need some help to streamline the code mor Faustian, >> > > to export including GUI, and to export including the effect, >> > > and maybe ideas how to port this to some small hardware, Pi or Daisy Pod >> > > ( though I doubt it will run there ). as well as opinion on the method >> > > and ideas. >> > > >> > > Code: >> > > >> > > declare options "[midi:on][nvoices:8]"; >> > > declare options "[-vec]"; >> > > declare name "Paradigma_9 v007"; >> > > declare version "0.0.7"; >> > > declare author "gabriel"; >> > > declare copyright "https://steady.page/en/voxangelica/";; >> > > declare license "DWTW"; >> > > // a synthesizer with "philonic" 3D spin oscillator and triangular filter >> > > import("stdfaust.lib"); >> > > import("maths.lib"); >> > > >> > > // frequency ratios table >> > > frtonum = waveform{1,16,9,6,5,4,7,3,8,5,7,15}; >> > > frtodiv = waveform{1,15,8,5,4,3,5,2,5,3,4, 8}; >> > > >> > > // MIDI >> > > midigrp(x) = hgroup("[1]MIDI",x); >> > > f = nentry("freq",200,40,2000,0.1) ; >> > > kmidi = nentry("key",69,0,127,1) ; >> > > bend = ba.semi2ratio(hslider("bend[midi:pitchwheel][style: >> > > knob]",0,-2,2,0.01)) ; >> > > gain = nentry("gain",0.6,0,1,0.01)<:* ; >> > > master = hslider("volume[midi:ctrl 7]",1,0,2,0.01) ; >> > > gate = button("gate") ; >> > > >> > > // spin oscill params >> > > rtogrp(x) = hgroup("[2]philonic",x); >> > > rto1sel = rtogrp(hslider("[1]x[style:knob]",-12,-24,24,1)); >> > > rto2sel = rtogrp(hslider("[2]z[style:knob]",19,-24,24,1)); >> > > fbka = >> > > rtogrp(hslider("[3]excentric[style:knob]",0.4,0,1,0.01)<:*:*(1/ma.PI)); >> > > detune = >> > > rtogrp(hslider("[4]warble[style:knob]",0.125,0,0.5,0.005)/ma.SR); >> > > pickd = >> > > rtogrp(hslider("[5]distance[style:knob]",0.7,0.25,1,0.0625))<:*:si.smoo; >> > > >> > > // LFO and Envelope Parameter >> > > lfogrp(x) = hgroup("[3]envelope & lfo",x); >> > > enva = (lfogrp(ba.db2linear(hslider("[1]A[style:knob]",20,15,66,1) >> > > )/1000)); >> > > envd = (lfogrp(ba.db2linear(hslider("[2]D[style:knob]",74,26,100,1) >> > > )/1000)*envpscal); >> > > envs = (lfogrp(hslider("[3]S[style:knob]",0,0,1,0.01) )); >> > > envr = (lfogrp(ba.db2linear(hslider("[4]R[style:knob]",50,26,100,1) >> > > )/1000)*envpscal); >> > > lfof = lfogrp(hslider("[5]LFO Hz[style:knob]",3,0.1,12,0.1)); >> > > lfvibra = lfogrp(hslider("[6]Vibrato[style:knob]",0.125,0,2,0.01))<:*; >> > > >> > > env = en.adsre(enva,envd*envpscal,envs,envr*envpscal,gate); >> > > envg = env:_* gain; >> > > >> > > lfosn = qsin(mphasor(lfof/ma.SR)); >> > > >> > > // Triangular Filter Parameter >> > > fltgrp(x) = hgroup("[4]triangulation",x); >> > > wid = fltgrp(hslider("[1]rise[style:knob]",4.89,1,9,0.001)):2^_:1/_; >> > > edge = fltgrp(hslider("[2]fall[style:knob]",6,1,9,0.001)):2^_:1/_; >> > > fiq = fltgrp(hslider("[3]q[style:knob]",1.18,0.5,3.87,0.01))<:*; >> > > hpon = fltgrp(checkbox("[4]highpass")); >> > > drive = fltgrp(hslider("[5]drive[style:knob]",0,-6,36,0.1)):_/20.0:10^_; >> > > >> > > rto1oct = rto1sel / 12 : floor; >> > > rto1semi = rto1sel + 24 : _% 12; >> > > rto1a = frtonum, rto1semi : rdtable; >> > > rto1b = frtodiv, rto1semi : rdtable; >> > > rto1 = (rto1a/rto1b)*(2^rto1oct); >> > > rto1r = min((1/ rto1),1); >> > > >> > > rto2oct = rto2sel / 12 : floor; >> > > rto2semi = rto2sel + 24 : _% 12; >> > > rto2a = frtonum, rto2semi : rdtable; >> > > rto2b = frtodiv, rto2semi : rdtable; >> > > rto2 = rto1*(rto2a/rto2b)*(2^rto2oct); >> > > rto2r = min((1 / rto2),1); >> > > >> > > // fve >> > > lg2f = ma.log2(f/440); >> > > stretch = 0.0333*lg2f; >> > > envpscal = ( - 3 * lg2f ):ba.db2linear; >> > > fplus = f*bend + lfosn* lfvibra*f * 0.5/12*envg + stretch; >> > > >> > > w = f/ma.SR; >> > > w2 = rto1 * w; >> > > w3 = rto2 * w; >> > > wplus = fplus/ma.SR; >> > > >> > > fbk1 = fbka*(0.5 -w)^4; >> > > fbk2 = fbka*(0.5 - w2)^4*rto1r; >> > > fbk3 = fbka*(0.5 - w3)^4*rto2r; >> > > >> > > // modulation reduction per frequency >> > > redux1 = ((3.3 -((rto1+1)*w) )/3.3),0: max:_^3; >> > > redux2 = ((3.3 -((rto2+1)*w) )/3.3),0: max:_^3; >> > > modep = envg; >> > > modep1 = envg * redux1 *rto1r * gain ; >> > > modep2 = envg * redux2 *rto2r * gain ; >> > > >> > > // sine oscillator >> > > wrap(n) = n-( floor( n +0.5)) ; >> > > qsincurve(x) = 1 - ( (x*x)<: *(1.2253517*16),(_<:*:* >> > > (-3.60562732*16)):>_ ); >> > > qsin(x) = x+(0.5): wrap <: (abs:-(0.25):qsincurve),_:ma.copysign; >> > > // feedback depth reduction curve >> > > fbcurve (x)= x:abs:-(1) <:^(3):_,(x):ma.copysign; >> > > >> > > // oscillator >> > > mphasor(fw) = (+(fw) ~ (wrap)); >> > > oscsn(fw, off) = mphasor(fw) + off:qsin:+~*(0.5); >> > > osc1(fw, off) = ((fw),+(off):(oscsn)) ~ >> > > (*(fbk2):fi.pole(0.5):_*fbcurve(fw)); >> > > dcrem(x) = x <:_,_': -: +~*(0.999773243); >> > > >> > > // 3D >> > > oscy(fw, off) = (osc1(fw, off )*osc1(fw*rto2+2*detune,0.75 + >> > > off)*modep2)*modep; >> > > oscx(fw, off) = (osc1(fw*rto1+detune,0.25 + >> > > off)*osc1(fw*rto2+2*detune,0.25 + off)*modep2)*modep1; >> > > oscxy(fw, off) = (oscy(fw, off)<:*),(oscx(fw, off)<:*):+:sqrt: >> > > fi.zero(1.0);//dcrem; // >> > > //oscxyb(fw, off) = (oscy(fw, off):fi.zero(1)) <:_,(_^2),((oscx(fw, >> > > off):fi.zero(1):_^2)): _, (_+_):_,(_+0.1:_^(3/2)):_/_; >> > > // with pickup >> > > oscxyc(fw, off) = oscxy(fw, off) <:_,(_^2:_+pickd:_^(3/2)):/; >> > > // >> > > //synthvox(fw, ph2, ph3, g1, g2, g3) = (oscxy(fw, 0):_*g1), (oscxy(fw, >> > > ph2):_*g2),(oscxy(fw, ph3):_*g3):>_ ; >> > > synthvox(fw, ph2, ph3, g1, g2, g3) = (oscxyc(fw, 0):_*g1), (oscxyc(fw, >> > > ph2):_*g2),(oscxyc(fw, ph3):_*g3):>_ ; >> > > // triangulation >> > > widredux = w <:+:_^3:1.0-_; >> > > // diff to max f in octaves, reduced for higher octaves >> > > dwo = ( 0.25 / wid ):max(_, 1): ma.log2: ma.inv: _*widredux: ma.inv; >> > > //edge = 1/7; // falling triangle edge, >> > > egderto = edge / wid; >> > > wid2 = wid * (2^(dwo * (1-envg ))): >> > > _* (2^(dwo * (1- gain ) )): >> > > min( _, 0.25): max( _, 4 / (ma.SR/fplus)); >> > > wid2e = edge: min( _, 0.25): max( _, 4 /(ma.SR/fplus)); >> > > >> > > fiw = wplus/wid2; >> > > fiwtail = wplus/wid2e; >> > > // triangle coefficients >> > > apg0 = fiw; >> > > apg1 = - apg0 - fiwtail; >> > > apg2 = fiwtail; >> > > // integration freq >> > > igpole = 1.0-5.0/ma.SR; >> > > resf = (fplus /( wid2 +wid2e) ): min( _, (0.249 * ma.SR)); >> > > >> > > // shaper >> > > // x - 0.15x²-0.15x³ >> > > tubicclip = _:min(_, (1.19419)):max(_,(-1.86086)); >> > > //tubic(x) = x - 0.15*(x^2)-0.15*(x^3); >> > > tubicilo(x) = x, >> > > // normal for in < 1.2e-4 >> > > ( x - 0.15*(x^2)-0.15*(x^3) ), >> > > // ILO: >> > > (( 0.5*(x^2) - 0.05*(x^3) - 0.0375*(x^4) ),(x <:_,_':- >> > > :_<:(abs:max(_,1.2e-4)),(ma.signum):ma.copysign):/): >> > > // select >> > > ba.if( (_:abs:_<= 1.2e-4), _, _ ):dcrem; >> > > // >> > > superfbp = 1 - sin( 2 * ma.PI * w ); >> > > >> > > // make sound >> > > process = synthvox(wplus, wid2, wid2e, apg0, apg1, apg2): >> > > fi.dcblockerat(10.0): fi.pole(igpole) : fi.svf.peak( resf, fiq) <: >> > > ba.if( hpon, fi.svf.hp( fplus/(wid+wid):min(_, ma.SR*0.249), >> > > 0.707 ),_): >> > > _*drive: tubicclip: tubicilo:_*(1/drive); >> > > effect = _ * master:rev; >> > > >> > > // >> > > ############################################################################################### >> > > // CIELverb >> > > ###################################################################################### >> > > // minimalist reverb >> > > // >> > > // UI >> > > revgrp(x) = hgroup("[5]reverb",x); >> > > sizem = >> > > revgrp(hslider("[1]size[style:knob]",0,-1.5,1.5,0.02)):(2.0)^_:_*16.7:si.smoo; >> > > revt = revgrp(hslider("[2]revTime[style:knob]",60,40,80,0.1)): >> > > ba.db2linear:_*0.001; >> > > bright = revgrp(hslider("[4]brightness[style:knob]",90,52,112,0.1)): >> > > ba.midikey2hz; >> > > earlyl = revgrp(hslider("[5]early/late[style:knob]",0,0,1,0.01)); >> > > drywet = revgrp(hslider("[6]dry/wet[style:knob]",0.5,0,1,0.01)) <:*; >> > > >> > > // reverb settings >> > > // change revt with size >> > > revtadapt = revt * ( 0.161*(sizem^3)/(6*sizem^2 )); >> > > // diffusion delay times >> > > revd0 = ma.SR * (sizem / 334); >> > > revd1 = revd0 * 1 / ( 2 - log(2)); >> > > revd2 = revd0 * 1 / ( 3 - log(2)); >> > > revd3 = revd0 * 1 / ( 4 - log(2)); >> > > revgn = 10^(-3*(( sizem/ 334 )/revtadapt)); >> > > // diffusion allpass coeficients >> > > revc = 0.707;//0.61803; // >> > > revc1 = -revc * 10^(-3*(( revd1/ ma.SR )/revtadapt)); >> > > revc2 = -revc * 10^(-3*(( revd2/ ma.SR )/revtadapt)); >> > > revc3 = -revc * 10^(-3*(( revd3/ ma.SR )/revtadapt)); >> > > // post (early) >> > > revdp = revd3 * 1 / ( 4 - log(2)); >> > > postd1 = revdp * 1 / ( 2 - log(2)); >> > > postd2 = postd1 * 1 / ( 3 - log(2)); >> > > postd3 = postd2 * 1 / ( 4 - log(2)); >> > > postc = 0.382;//1/3;//0.61803; // >> > > postc1 = -postc * 10^(-3*(( postd1/ ma.SR )/(revtadapt * 1 / ( 4 - >> > > log(2))))); >> > > postc2 = -postc * 10^(-3*(( postd2/ ma.SR )/(revtadapt * 1 / ( 4 - >> > > log(2))))); >> > > postc3 = -postc * 10^(-3*(( postd2/ ma.SR )/(revtadapt * 1 / ( 4 - >> > > log(2))))); >> > > >> > > // left right delay time offsets >> > > postdlroff = ma.SR * 0.15 /334 ; >> > > >> > > lfo1 = os.oscsin(0.13)*8.0; >> > > apcomblp(maxdel,N,g) = (+ <: >> > > (de.fdelay1a(maxdel,N-1.5)<:_,_':+:_*0.5),*(g)) ~ *(-g) : mem,_ : +; >> > > // post diffusion (early reflections, placed after reverb loop) >> > > postdiff( in ) = in <: >> > > (apcomblp( 4096, postd1, postc1): apcomblp( 4096, postd2 >> > > + postdlroff, postc2): apcomblp( 4096, postd3 - postdlroff*0.382, >> > > postc3)), >> > > (apcomblp( 4096, postd1 + postdlroff, postc1) :apcomblp( >> > > 4096, postd2 - postdlroff*0.382, postc2): apcomblp( 4096, postd3, >> > > postc3)); >> > > >> > > // feedback filter >> > > dampp = sin( 2 * ma.PI * bright/ma.SR); >> > > fidamp = _*(dampp) : +~*(1-dampp) <:_,_':+:_*0.5: *(revgn); >> > > >> > > >> > > // reverb >> > > rev = _<: +~ (apcomblp( 4096, revd1 - lfo1, revc1):apcomblp( 4096, revd2 >> > > + lfo1, revc2): apcomblp( 4096, revd3,revc3): de.fdelay1a( 4096, revd0 >> > > ):fidamp ),_:(_*drywet:postdiff),((_*(1-drywet))<:_,_):route(4, 4, >> > > (1,1),(2,3),(3,2),(4,4)):>(_+_),(_+_); >> > > >> > > // END REVERB >> > > ############################################################################ >> > > >> > > _______________________________________________ >> > > Faudiostream-users mailing list >> > > Faudiostream-users@lists.sourceforge.net >> > > https://lists.sourceforge.net/lists/listinfo/faudiostream-users >> > >> _______________________________________________ Faudiostream-users mailing list Faudiostream-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/faudiostream-users
