Great, thanks a lot. On Sun, 28 Sept 2025, 21:39 Stéphane Letz, <[email protected]> wrote:
> Thanks, added in Faust repo here: > > > https://github.com/grame-cncm/faust/commit/71b8f236e0aaa89e172d90906fd18c26e29cfbad > > Stéphane > > > Le 16 sept. 2025 à 15:52, ga <[email protected]> a écrit : > > > > 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 <[email protected]> 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 <[email protected]> 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 <[email protected]> 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 <[email protected]> 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 <[email protected]> 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 > >>>>> [email protected] > >>>>> https://lists.sourceforge.net/lists/listinfo/faudiostream-users > >>>> > >>> > >
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