commit d49da093a8a60afb1e971f08312b6eada9b14312 parent dced94a695741c349d4cfa660809e1c00a934e62 Author: fundamental <[email protected]> Date: Sun, 23 Mar 2014 17:03:05 -0400 Merge Globals Cleanup Patch Merge patch from falktx which: 1. make effects *never* use synth->... 2. make SYNTH_T::numRandom static, so it can be used 3. replace getTmpBuffer() with "float 4. minor s/final/final_/ var name change, just for safety because "final" is c++11 keyword In addition this fixes the Echo Test as it needed to be updated to the new API Diffstat:
36 files changed, 244 insertions(+), 165 deletions(-)
diff --git a/AUTHORS.txt b/AUTHORS.txt @@ -26,4 +26,5 @@ Contributors: Johannes Lorenz (Effect Documentation) Ilario Glasgo (Italian Doc Translation) Christopher Oliver (Unison Regrssion Fix, presets fix) + Filipe Coelho (Globals Cleanup) diff --git a/src/DSP/AnalogFilter.cpp b/src/DSP/AnalogFilter.cpp @@ -32,8 +32,10 @@ AnalogFilter::AnalogFilter(unsigned char Ftype, float Ffreq, float Fq, - unsigned char Fstages) - :type(Ftype), + unsigned char Fstages, + unsigned int srate, int bufsize) + :Filter(srate, bufsize), + type(Ftype), stages(Fstages), freq(Ffreq), q(Fq), @@ -75,8 +77,8 @@ void AnalogFilter::computefiltercoefs(void) //do not allow frequencies bigger than samplerate/2 float freq = this->freq; - if(freq > (synth->halfsamplerate_f - 500.0f)) { - freq = synth->halfsamplerate_f - 500.0f; + if(freq > (halfsamplerate_f - 500.0f)) { + freq = halfsamplerate_f - 500.0f; zerocoefs = true; } if(freq < 0.1f) @@ -99,7 +101,7 @@ void AnalogFilter::computefiltercoefs(void) float *d = coeff.d; //General Constants - const float omega = 2 * PI * freq / synth->samplerate_f; + const float omega = 2 * PI * freq / samplerate_f; const float sn = sinf(omega), cs = cosf(omega); float alpha, beta; @@ -110,7 +112,7 @@ void AnalogFilter::computefiltercoefs(void) switch(type) { case 0: //LPF 1 pole if(!zerocoefs) - tmp = expf(-2.0f * PI * freq / synth->samplerate_f); + tmp = expf(-2.0f * PI * freq / samplerate_f); else tmp = 0.0f; c[0] = 1.0f - tmp; @@ -122,7 +124,7 @@ void AnalogFilter::computefiltercoefs(void) break; case 1: //HPF 1 pole if(!zerocoefs) - tmp = expf(-2.0f * PI * freq / synth->samplerate_f); + tmp = expf(-2.0f * PI * freq / samplerate_f); else tmp = 0.0f; c[0] = (1.0f + tmp) / 2.0f; @@ -277,7 +279,7 @@ void AnalogFilter::setfreq(float frequency) rap = 1.0f / rap; oldabovenq = abovenq; - abovenq = frequency > (synth->halfsamplerate_f - 500.0f); + abovenq = frequency > (halfsamplerate_f - 500.0f); bool nyquistthresh = (abovenq ^ oldabovenq); @@ -353,9 +355,9 @@ inline void AnalogBiquadFilterB(const float coeff[5], float &src, float work[4]) void AnalogFilter::singlefilterout(float *smp, fstage &hist, const Coeff &coeff) { - assert((synth->buffersize % 8) == 0); + assert((buffersize % 8) == 0); if(order == 1) { //First order filter - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { float y0 = smp[i] * coeff.c[0] + hist.x1 * coeff.c[1] + hist.y1 * coeff.d[1]; hist.y1 = y0; @@ -365,7 +367,7 @@ void AnalogFilter::singlefilterout(float *smp, fstage &hist, } else if(order == 2) {//Second order filter const float coeff_[5] = {coeff.c[0], coeff.c[1], coeff.c[2], coeff.d[1], coeff.d[2]}; float work[4] = {hist.x1, hist.x2, hist.y1, hist.y2}; - for(int i = 0; i < synth->buffersize; i+=8) { + for(int i = 0; i < buffersize; i+=8) { AnalogBiquadFilterA(coeff_, smp[i + 0], work); AnalogBiquadFilterB(coeff_, smp[i + 1], work); AnalogBiquadFilterA(coeff_, smp[i + 2], work); @@ -389,27 +391,26 @@ void AnalogFilter::filterout(float *smp) if(needsinterpolation) { //Merge Filter at old coeff with new coeff - float *ismp = getTmpBuffer(); - memcpy(ismp, smp, synth->bufferbytes); + float ismp[buffersize]; + memcpy(ismp, smp, bufferbytes); for(int i = 0; i < stages + 1; ++i) singlefilterout(ismp, oldHistory[i], oldCoeff); - for(int i = 0; i < synth->buffersize; ++i) { - float x = (float)i / synth->buffersize_f; + for(int i = 0; i < buffersize; ++i) { + float x = (float)i / buffersize_f; smp[i] = ismp[i] * (1.0f - x) + smp[i] * x; } - returnTmpBuffer(ismp); needsinterpolation = false; } - for(int i = 0; i < synth->buffersize; ++i) + for(int i = 0; i < buffersize; ++i) smp[i] *= outgain; } float AnalogFilter::H(float freq) { - float fr = freq / synth->samplerate_f * PI * 2.0f; + float fr = freq / samplerate_f * PI * 2.0f; float x = coeff.c[0], y = 0.0f; for(int n = 1; n < 3; ++n) { x += cosf(n * fr) * coeff.c[n]; diff --git a/src/DSP/AnalogFilter.h b/src/DSP/AnalogFilter.h @@ -35,7 +35,7 @@ class AnalogFilter:public Filter { public: AnalogFilter(unsigned char Ftype, float Ffreq, float Fq, - unsigned char Fstages); + unsigned char Fstages, unsigned int srate, int bufsize); ~AnalogFilter(); void filterout(float *smp); void setfreq(float frequency); diff --git a/src/DSP/Filter.cpp b/src/DSP/Filter.cpp @@ -29,24 +29,37 @@ #include "SVFilter.h" #include "../Params/FilterParams.h" -Filter *Filter::generate(FilterParams *pars) +Filter::Filter(unsigned int srate, int bufsize) + : outgain(1.0f), + samplerate(srate), + buffersize(bufsize) { + alias(); +} + +Filter *Filter::generate(FilterParams *pars, unsigned int srate, int bufsize) +{ + if (srate == 0) + srate = synth->samplerate; + if (bufsize == 0) + bufsize = synth->buffersize; + unsigned char Ftype = pars->Ptype; unsigned char Fstages = pars->Pstages; Filter *filter; switch(pars->Pcategory) { case 1: - filter = new FormantFilter(pars); + filter = new FormantFilter(pars, srate, bufsize); break; case 2: - filter = new SVFilter(Ftype, 1000.0f, pars->getq(), Fstages); + filter = new SVFilter(Ftype, 1000.0f, pars->getq(), Fstages, srate, bufsize); filter->outgain = dB2rap(pars->getgain()); if(filter->outgain > 1.0f) filter->outgain = sqrt(filter->outgain); break; default: - filter = new AnalogFilter(Ftype, 1000.0f, pars->getq(), Fstages); + filter = new AnalogFilter(Ftype, 1000.0f, pars->getq(), Fstages, srate, bufsize); if((Ftype >= 6) && (Ftype <= 8)) filter->setgain(pars->getgain()); else diff --git a/src/DSP/Filter.h b/src/DSP/Filter.h @@ -29,8 +29,9 @@ class Filter { public: static float getrealfreq(float freqpitch); - static Filter *generate(class FilterParams * pars); + static Filter *generate(class FilterParams * pars, unsigned int srate = 0, int bufsize = 0); + Filter(unsigned int srate, int bufsize); virtual ~Filter() {} virtual void filterout(float *smp) = 0; virtual void setfreq(float frequency) = 0; @@ -40,6 +41,24 @@ class Filter protected: float outgain; + + // current setup + unsigned int samplerate; + int buffersize; + + // alias for above terms + float samplerate_f; + float halfsamplerate_f; + float buffersize_f; + int bufferbytes; + + inline void alias() + { + samplerate_f = samplerate; + halfsamplerate_f = samplerate_f / 2.0f; + buffersize_f = buffersize; + bufferbytes = buffersize * sizeof(float); + } }; #endif diff --git a/src/DSP/FormantFilter.cpp b/src/DSP/FormantFilter.cpp @@ -27,11 +27,12 @@ #include "AnalogFilter.h" #include "../Params/FilterParams.h" -FormantFilter::FormantFilter(FilterParams *pars) +FormantFilter::FormantFilter(FilterParams *pars, unsigned int srate, int bufsize) + : Filter(srate, bufsize) { numformants = pars->Pnumformants; for(int i = 0; i < numformants; ++i) - formant[i] = new AnalogFilter(4 /*BPF*/, 1000.0f, 10.0f, pars->Pstages); + formant[i] = new AnalogFilter(4 /*BPF*/, 1000.0f, 10.0f, pars->Pstages, srate, bufsize); cleanup(); for(int j = 0; j < FF_MAX_VOWELS; ++j) @@ -203,29 +204,27 @@ void FormantFilter::setfreq_and_q(float frequency, float q_) void FormantFilter::filterout(float *smp) { - float *inbuffer = getTmpBuffer(); + float inbuffer[buffersize]; - memcpy(inbuffer, smp, synth->bufferbytes); - memset(smp, 0, synth->bufferbytes); + memcpy(inbuffer, smp, bufferbytes); + memset(smp, 0, bufferbytes); for(int j = 0; j < numformants; ++j) { - float *tmpbuf = getTmpBuffer(); - for(int i = 0; i < synth->buffersize; ++i) + float tmpbuf[buffersize]; + for(int i = 0; i < buffersize; ++i) tmpbuf[i] = inbuffer[i] * outgain; formant[j]->filterout(tmpbuf); if(ABOVE_AMPLITUDE_THRESHOLD(oldformantamp[j], currentformants[j].amp)) - for(int i = 0; i < synth->buffersize; ++i) + for(int i = 0; i < buffersize; ++i) smp[i] += tmpbuf[i] * INTERPOLATE_AMPLITUDE(oldformantamp[j], currentformants[j].amp, i, - synth->buffersize); + buffersize); else - for(int i = 0; i < synth->buffersize; ++i) + for(int i = 0; i < buffersize; ++i) smp[i] += tmpbuf[i] * currentformants[j].amp; - returnTmpBuffer(tmpbuf); oldformantamp[j] = currentformants[j].amp; } - returnTmpBuffer(inbuffer); } diff --git a/src/DSP/FormantFilter.h b/src/DSP/FormantFilter.h @@ -30,7 +30,7 @@ class FormantFilter:public Filter { public: - FormantFilter(class FilterParams *pars); + FormantFilter(class FilterParams *pars, unsigned int srate, int bufsize); ~FormantFilter(); void filterout(float *smp); void setfreq(float frequency); diff --git a/src/DSP/SVFilter.cpp b/src/DSP/SVFilter.cpp @@ -29,8 +29,9 @@ #include "SVFilter.h" SVFilter::SVFilter(unsigned char Ftype, float Ffreq, float Fq, - unsigned char Fstages) - :type(Ftype), + unsigned char Fstages, unsigned int srate, int bufsize) + :Filter(srate, bufsize), + type(Ftype), stages(Fstages), freq(Ffreq), q(Fq), @@ -58,7 +59,7 @@ void SVFilter::cleanup() void SVFilter::computefiltercoefs(void) { - par.f = freq / synth->samplerate_f * 4.0f; + par.f = freq / samplerate_f * 4.0f; if(par.f > 0.99999f) par.f = 0.99999f; par.q = 1.0f - atanf(sqrtf(q)) * 2.0f / PI; @@ -76,7 +77,7 @@ void SVFilter::setfreq(float frequency) rap = 1.0f / rap; oldabovenq = abovenq; - abovenq = frequency > (synth->samplerate_f / 2 - 500.0f); + abovenq = frequency > (samplerate_f / 2 - 500.0f); bool nyquistthresh = (abovenq ^ oldabovenq); @@ -144,7 +145,7 @@ void SVFilter::singlefilterout(float *smp, fstage &x, parameters &par) errx(1, "Impossible SVFilter type encountered [%d]", type); } - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { x.low = x.low + par.f * x.band; x.high = par.q_sqrt * smp[i] - x.low - par.q * x.band; x.band = par.f * x.high + x.band; @@ -159,20 +160,19 @@ void SVFilter::filterout(float *smp) singlefilterout(smp, st[i], par); if(needsinterpolation) { - float *ismp = getTmpBuffer(); - memcpy(ismp, smp, synth->bufferbytes); + float ismp[buffersize]; + memcpy(ismp, smp, bufferbytes); for(int i = 0; i < stages + 1; ++i) singlefilterout(ismp, st[i], ipar); - for(int i = 0; i < synth->buffersize; ++i) { - float x = i / synth->buffersize_f; + for(int i = 0; i < buffersize; ++i) { + float x = i / buffersize_f; smp[i] = ismp[i] * (1.0f - x) + smp[i] * x; } - returnTmpBuffer(ismp); needsinterpolation = false; } - for(int i = 0; i < synth->buffersize; ++i) + for(int i = 0; i < buffersize; ++i) smp[i] *= outgain; } diff --git a/src/DSP/SVFilter.h b/src/DSP/SVFilter.h @@ -32,7 +32,8 @@ class SVFilter:public Filter SVFilter(unsigned char Ftype, float Ffreq, float Fq, - unsigned char Fstages); + unsigned char Fstages, + unsigned int srate, int bufsize); ~SVFilter(); void filterout(float *smp); void setfreq(float frequency); diff --git a/src/DSP/Unison.cpp b/src/DSP/Unison.cpp @@ -25,18 +25,19 @@ #include "Unison.h" -Unison::Unison(int update_period_samples_, float max_delay_sec_) +Unison::Unison(int update_period_samples_, float max_delay_sec_, float srate_f) :unison_size(0), base_freq(1.0f), uv(NULL), update_period_samples(update_period_samples_), update_period_sample_k(0), - max_delay((int)(synth->samplerate_f * max_delay_sec_) + 1), + max_delay((int)(srate_f * max_delay_sec_) + 1), delay_k(0), first_time(false), delay_buffer(NULL), unison_amplitude_samples(0.0f), - unison_bandwidth_cents(10.0f) + unison_bandwidth_cents(10.0f), + samplerate_f(srate_f) { if(max_delay < 10) max_delay = 10; @@ -87,15 +88,15 @@ void Unison::updateParameters(void) { if(!uv) return; - float increments_per_second = synth->samplerate_f + float increments_per_second = samplerate_f / (float) update_period_samples; // printf("#%g, %g\n",increments_per_second,base_freq); for(int i = 0; i < unison_size; ++i) { - float base = powf(UNISON_FREQ_SPAN, synth->numRandom() * 2.0f - 1.0f); + float base = powf(UNISON_FREQ_SPAN, SYNTH_T::numRandom() * 2.0f - 1.0f); uv[i].relative_amplitude = base; float period = base / base_freq; float m = 4.0f / (period * increments_per_second); - if(synth->numRandom() < 0.5f) + if(SYNTH_T::numRandom() < 0.5f) m = -m; uv[i].step = m; // printf("%g %g\n",uv[i].relative_amplitude,period); @@ -103,7 +104,7 @@ void Unison::updateParameters(void) float max_speed = powf(2.0f, unison_bandwidth_cents / 1200.0f); unison_amplitude_samples = 0.125f * (max_speed - 1.0f) - * synth->samplerate_f / base_freq; + * samplerate_f / base_freq; //If functions exceed this limit, they should have requested a bigguer delay //and thus are buggy diff --git a/src/DSP/Unison.h b/src/DSP/Unison.h @@ -30,7 +30,7 @@ class Unison { public: - Unison(int update_period_samples_, float max_delay_sec_); + Unison(int update_period_samples_, float max_delay_sec_, float srate_f); ~Unison(); void setSize(int new_size); @@ -69,5 +69,8 @@ class Unison float *delay_buffer; float unison_amplitude_samples; float unison_bandwidth_cents; + + // current setup + float samplerate_f; }; #endif diff --git a/src/Effects/Alienwah.cpp b/src/Effects/Alienwah.cpp @@ -23,8 +23,9 @@ #include <cmath> #include "Alienwah.h" -Alienwah::Alienwah(bool insertion_, float *efxoutl_, float *efxoutr_) - :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0), +Alienwah::Alienwah(bool insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize) + :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0, srate, bufsize), + lfo(srate, bufsize), oldl(NULL), oldr(NULL) { @@ -58,8 +59,8 @@ void Alienwah::out(const Stereo<float *> &smp) clfol = complex<float>(cosf(lfol + phase) * fb, sinf(lfol + phase) * fb); //rework clfor = complex<float>(cosf(lfor + phase) * fb, sinf(lfor + phase) * fb); //rework - for(int i = 0; i < synth->buffersize; ++i) { - float x = ((float) i) / synth->buffersize_f; + for(int i = 0; i < buffersize; ++i) { + float x = ((float) i) / buffersize_f; float x1 = 1.0f - x; //left complex<float> tmp = clfol * x + oldclfol * x1; diff --git a/src/Effects/Alienwah.h b/src/Effects/Alienwah.h @@ -44,7 +44,8 @@ class Alienwah:public Effect */ Alienwah(bool insertion_, float *const efxoutl_, - float *const efxoutr_); + float *const efxoutr_, + unsigned int srate, int bufsize); ~Alienwah(); void out(const Stereo<float *> &smp); diff --git a/src/Effects/Chorus.cpp b/src/Effects/Chorus.cpp @@ -26,9 +26,10 @@ using namespace std; -Chorus::Chorus(bool insertion_, float *const efxoutl_, float *efxoutr_) - :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0), - maxdelay((int)(MAX_CHORUS_DELAY / 1000.0f * synth->samplerate_f)), +Chorus::Chorus(bool insertion_, float *const efxoutl_, float *efxoutr_, unsigned int srate, int bufsize) + :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0, srate, bufsize), + lfo(srate, bufsize), + maxdelay((int)(MAX_CHORUS_DELAY / 1000.0f * samplerate_f)), delaySample(new float[maxdelay], new float[maxdelay]) { dlk = 0; @@ -51,7 +52,7 @@ Chorus::~Chorus() float Chorus::getdelay(float xlfo) { float result = - (Pflangemode) ? 0 : (delay + xlfo * depth) * synth->samplerate_f; + (Pflangemode) ? 0 : (delay + xlfo * depth) * samplerate_f; //check if delay is too big (caused by bad setdelay() and setdepth() if((result + 0.5f) >= maxdelay) { @@ -75,7 +76,7 @@ void Chorus::out(const Stereo<float *> &input) dl2 = getdelay(lfol); dr2 = getdelay(lfor); - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { float inL = input.l[i]; float inR = input.r[i]; //LRcross @@ -87,7 +88,7 @@ void Chorus::out(const Stereo<float *> &input) //compute the delay in samples using linear interpolation between the lfo delays float mdel = - (dl1 * (synth->buffersize - i) + dl2 * i) / synth->buffersize_f; + (dl1 * (buffersize - i) + dl2 * i) / buffersize_f; if(++dlk >= maxdelay) dlk = 0; float tmp = dlk - mdel + maxdelay * 2.0f; //where should I get the sample from @@ -105,7 +106,7 @@ void Chorus::out(const Stereo<float *> &input) //Right channel //compute the delay in samples using linear interpolation between the lfo delays - mdel = (dr1 * (synth->buffersize - i) + dr2 * i) / synth->buffersize_f; + mdel = (dr1 * (buffersize - i) + dr2 * i) / buffersize_f; if(++drk >= maxdelay) drk = 0; tmp = drk * 1.0f - mdel + maxdelay * 2.0f; //where should I get the sample from @@ -122,12 +123,12 @@ void Chorus::out(const Stereo<float *> &input) } if(Poutsub) - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { efxoutl[i] *= -1.0f; efxoutr[i] *= -1.0f; } - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { efxoutl[i] *= pangainL; efxoutr[i] *= pangainR; } diff --git a/src/Effects/Chorus.h b/src/Effects/Chorus.h @@ -32,7 +32,7 @@ class Chorus:public Effect { public: - Chorus(bool insertion_, float *efxoutl_, float *efxoutr_); + Chorus(bool insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize); /**Destructor*/ ~Chorus(); void out(const Stereo<float *> &input); diff --git a/src/Effects/Distorsion.cpp b/src/Effects/Distorsion.cpp @@ -25,8 +25,8 @@ #include "../Misc/WaveShapeSmps.h" #include <cmath> -Distorsion::Distorsion(bool insertion_, float *efxoutl_, float *efxoutr_) - :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0), +Distorsion::Distorsion(bool insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize) + :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0, srate, bufsize), Pvolume(50), Pdrive(90), Plevel(64), @@ -37,10 +37,10 @@ Distorsion::Distorsion(bool insertion_, float *efxoutl_, float *efxoutr_) Pstereo(0), Pprefiltering(0) { - lpfl = new AnalogFilter(2, 22000, 1, 0); - lpfr = new AnalogFilter(2, 22000, 1, 0); - hpfl = new AnalogFilter(3, 20, 1, 0); - hpfr = new AnalogFilter(3, 20, 1, 0); + lpfl = new AnalogFilter(2, 22000, 1, 0, srate, bufsize); + lpfr = new AnalogFilter(2, 22000, 1, 0, srate, bufsize); + hpfl = new AnalogFilter(3, 20, 1, 0, srate, bufsize); + hpfr = new AnalogFilter(3, 20, 1, 0, srate, bufsize); setpreset(Ppreset); cleanup(); } @@ -83,29 +83,29 @@ void Distorsion::out(const Stereo<float *> &smp) inputvol *= -1.0f; if(Pstereo) //Stereo - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { efxoutl[i] = smp.l[i] * inputvol * pangainL; efxoutr[i] = smp.r[i] * inputvol * pangainR; } else //Mono - for(int i = 0; i < synth->buffersize; ++i) + for(int i = 0; i < buffersize; ++i) efxoutl[i] = (smp.l[i] * pangainL + smp.r[i] * pangainR) * inputvol; if(Pprefiltering) applyfilters(efxoutl, efxoutr); - waveShapeSmps(synth->buffersize, efxoutl, Ptype + 1, Pdrive); + waveShapeSmps(buffersize, efxoutl, Ptype + 1, Pdrive); if(Pstereo) - waveShapeSmps(synth->buffersize, efxoutr, Ptype + 1, Pdrive); + waveShapeSmps(buffersize, efxoutr, Ptype + 1, Pdrive); if(!Pprefiltering) applyfilters(efxoutl, efxoutr); if(!Pstereo) - memcpy(efxoutr, efxoutl, synth->bufferbytes); + memcpy(efxoutr, efxoutl, bufferbytes); float level = dB2rap(60.0f * Plevel / 127.0f - 40.0f); - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { float lout = efxoutl[i]; float rout = efxoutr[i]; float l = lout * (1.0f - lrcross) + rout * lrcross; diff --git a/src/Effects/Distorsion.h b/src/Effects/Distorsion.h @@ -29,7 +29,7 @@ class Distorsion:public Effect { public: - Distorsion(bool insertion, float *efxoutl_, float *efxoutr_); + Distorsion(bool insertion, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize); ~Distorsion(); void out(const Stereo<float *> &smp); void setpreset(unsigned char npreset); diff --git a/src/Effects/DynamicFilter.cpp b/src/Effects/DynamicFilter.cpp @@ -24,8 +24,9 @@ #include "DynamicFilter.h" #include "../DSP/Filter.h" -DynamicFilter::DynamicFilter(bool insertion_, float *efxoutl_, float *efxoutr_) - :Effect(insertion_, efxoutl_, efxoutr_, new FilterParams(0, 64, 64), 0), +DynamicFilter::DynamicFilter(bool insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize) + :Effect(insertion_, efxoutl_, efxoutr_, new FilterParams(0, 64, 64), 0, srate, bufsize), + lfo(srate, bufsize), Pvolume(110), Pdepth(0), Pampsns(90), @@ -61,7 +62,7 @@ void DynamicFilter::out(const Stereo<float *> &smp) const float freq = filterpars->getfreq(); const float q = filterpars->getq(); - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { efxoutl[i] = smp.l[i]; efxoutr[i] = smp.r[i]; @@ -85,7 +86,7 @@ void DynamicFilter::out(const Stereo<float *> &smp) filterr->filterout(efxoutr); //panning - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { efxoutl[i] *= pangainL; efxoutr[i] *= pangainR; } @@ -130,8 +131,8 @@ void DynamicFilter::reinitfilter(void) { delete filterl; delete filterr; - filterl = Filter::generate(filterpars); - filterr = Filter::generate(filterpars); + filterl = Filter::generate(filterpars, samplerate, buffersize); + filterr = Filter::generate(filterpars, samplerate, buffersize); } void DynamicFilter::setpreset(unsigned char npreset) diff --git a/src/Effects/DynamicFilter.h b/src/Effects/DynamicFilter.h @@ -30,7 +30,7 @@ class DynamicFilter:public Effect { public: - DynamicFilter(bool insertion_, float *efxoutl_, float *efxoutr_); + DynamicFilter(bool insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize); ~DynamicFilter(); void out(const Stereo<float *> &smp); diff --git a/src/Effects/EQ.cpp b/src/Effects/EQ.cpp @@ -24,8 +24,8 @@ #include "EQ.h" #include "../DSP/AnalogFilter.h" -EQ::EQ(bool insertion_, float *efxoutl_, float *efxoutr_) - :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0) +EQ::EQ(bool insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize) + :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0, srate, bufsize) { for(int i = 0; i < MAX_EQ_BANDS; ++i) { filter[i].Ptype = 0; @@ -33,8 +33,8 @@ EQ::EQ(bool insertion_, float *efxoutl_, float *efxoutr_) filter[i].Pgain = 64; filter[i].Pq = 64; filter[i].Pstages = 0; - filter[i].l = new AnalogFilter(6, 1000.0f, 1.0f, 0); - filter[i].r = new AnalogFilter(6, 1000.0f, 1.0f, 0); + filter[i].l = new AnalogFilter(6, 1000.0f, 1.0f, 0, srate, bufsize); + filter[i].r = new AnalogFilter(6, 1000.0f, 1.0f, 0, srate, bufsize); } //default values Pvolume = 50; @@ -56,7 +56,7 @@ void EQ::cleanup(void) //Effect output void EQ::out(const Stereo<float *> &smp) { - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { efxoutl[i] = smp.l[i] * volume; efxoutr[i] = smp.r[i] * volume; } diff --git a/src/Effects/EQ.h b/src/Effects/EQ.h @@ -29,7 +29,7 @@ class EQ:public Effect { public: - EQ(bool insertion_, float *efxoutl_, float *efxoutr_); + EQ(bool insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize); ~EQ() {} void out(const Stereo<float *> &smp); void setpreset(unsigned char npreset); diff --git a/src/Effects/Echo.cpp b/src/Effects/Echo.cpp @@ -27,8 +27,8 @@ #define MAX_DELAY 2 -Echo::Echo(bool insertion_, float *efxoutl_, float *efxoutr_) - :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0), +Echo::Echo(bool insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize) + :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0, srate, bufsize), Pvolume(50), Pdelay(60), Plrdelay(100), @@ -37,8 +37,8 @@ Echo::Echo(bool insertion_, float *efxoutl_, float *efxoutr_) delayTime(1), lrdelay(0), avgDelay(0), - delay(new float[(int)(MAX_DELAY * synth->samplerate)], - new float[(int)(MAX_DELAY * synth->samplerate)]), + delay(new float[(int)(MAX_DELAY * srate)], + new float[(int)(MAX_DELAY * srate)]), old(0.0f), pos(0), delta(1), @@ -57,8 +57,8 @@ Echo::~Echo() //Cleanup the effect void Echo::cleanup(void) { - memset(delay.l, 0, MAX_DELAY * synth->samplerate * sizeof(float)); - memset(delay.r, 0, MAX_DELAY * synth->samplerate * sizeof(float)); + memset(delay.l, 0, MAX_DELAY * samplerate * sizeof(float)); + memset(delay.r, 0, MAX_DELAY * samplerate * sizeof(float)); old = Stereo<float>(0.0f); } @@ -77,14 +77,14 @@ void Echo::initdelays(void) //number of seconds to delay right chan float dr = avgDelay + lrdelay; - ndelta.l = max(1, (int) (dl * synth->samplerate)); - ndelta.r = max(1, (int) (dr * synth->samplerate)); + ndelta.l = max(1, (int) (dl * samplerate)); + ndelta.r = max(1, (int) (dr * samplerate)); } //Effect output void Echo::out(const Stereo<float *> &input) { - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { float ldl = delay.l[pos.l]; float rdl = delay.r[pos.r]; ldl = ldl * (1.0f - lrcross) + rdl * lrcross; @@ -97,9 +97,9 @@ void Echo::out(const Stereo<float *> &input) rdl = input.r[i] * pangainR - rdl * fb; //LowPass Filter - old.l = delay.l[(pos.l + delta.l) % (MAX_DELAY * synth->samplerate)] = + old.l = delay.l[(pos.l + delta.l) % (MAX_DELAY * samplerate)] = ldl * hidamp + old.l * (1.0f - hidamp); - old.r = delay.r[(pos.r + delta.r) % (MAX_DELAY * synth->samplerate)] = + old.r = delay.r[(pos.r + delta.r) % (MAX_DELAY * samplerate)] = rdl * hidamp + old.r * (1.0f - hidamp); //increment @@ -107,8 +107,8 @@ void Echo::out(const Stereo<float *> &input) ++pos.r; // += delta.r; //ensure that pos is still in bounds - pos.l %= MAX_DELAY * synth->samplerate; - pos.r %= MAX_DELAY * synth->samplerate; + pos.l %= MAX_DELAY * samplerate; + pos.r %= MAX_DELAY * samplerate; //adjust delay if needed delta.l = (15 * delta.l + ndelta.l) / 16; diff --git a/src/Effects/Echo.h b/src/Effects/Echo.h @@ -30,7 +30,7 @@ class Echo:public Effect { public: - Echo(bool insertion_, float *efxoutl_, float *efxoutr_); + Echo(bool insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize); ~Echo(); void out(const Stereo<float *> &input); diff --git a/src/Effects/Effect.cpp b/src/Effects/Effect.cpp @@ -26,13 +26,18 @@ #include <cmath> Effect::Effect(bool insertion_, float *efxoutl_, float *efxoutr_, - FilterParams *filterpars_, unsigned char Ppreset_) + FilterParams *filterpars_, unsigned char Ppreset_, + unsigned int srate, int bufsize) :Ppreset(Ppreset_), efxoutl(efxoutl_), efxoutr(efxoutr_), filterpars(filterpars_), - insertion(insertion_) -{} + insertion(insertion_), + samplerate(srate), + buffersize(bufsize) +{ + alias(); +} void Effect::out(float *const smpsl, float *const smpsr) { diff --git a/src/Effects/Effect.h b/src/Effects/Effect.h @@ -43,7 +43,8 @@ class Effect * @param Ppreset_ chosen preset * @return Initialized Effect object*/ Effect(bool insertion_, float *efxoutl_, float *efxoutr_, - FilterParams *filterpars_, unsigned char Ppreset_); + FilterParams *filterpars_, unsigned char Ppreset_, + unsigned int srate, int bufsize); virtual ~Effect() {} /** * Choose a preset @@ -100,6 +101,24 @@ class Effect float pangainR; char Plrcross; // L/R mix float lrcross; + + // current setup + unsigned int samplerate; + int buffersize; + + // alias for above terms + float samplerate_f; + float halfsamplerate_f; + float buffersize_f; + int bufferbytes; + + inline void alias() + { + samplerate_f = samplerate; + halfsamplerate_f = samplerate_f / 2.0f; + buffersize_f = buffersize; + bufferbytes = buffersize * sizeof(float); + } }; #endif diff --git a/src/Effects/EffectLFO.cpp b/src/Effects/EffectLFO.cpp @@ -25,7 +25,7 @@ #include <cmath> -EffectLFO::EffectLFO(void) +EffectLFO::EffectLFO(float srate_f, float bufsize_f) :Pfreq(40), Prandomness(0), PLFOtype(0), @@ -36,7 +36,9 @@ EffectLFO::EffectLFO(void) ampl2(RND), ampr1(RND), ampr2(RND), - lfornd(0.0f) + lfornd(0.0f), + samplerate_f(srate_f), + buffersize_f(bufsize_f) { updateparams(); } @@ -47,7 +49,7 @@ EffectLFO::~EffectLFO() {} void EffectLFO::updateparams(void) { float lfofreq = (powf(2.0f, Pfreq / 127.0f * 10.0f) - 1.0f) * 0.03f; - incx = fabsf(lfofreq) * synth->buffersize_f / synth->samplerate_f; + incx = fabsf(lfofreq) * buffersize_f / samplerate_f; if(incx > 0.49999999f) incx = 0.499999999f; //Limit the Frequency diff --git a/src/Effects/EffectLFO.h b/src/Effects/EffectLFO.h @@ -28,7 +28,7 @@ class EffectLFO { public: - EffectLFO(); + EffectLFO(float srate_f, float bufsize_f); ~EffectLFO(); void effectlfoout(float *outl, float *outr); void updateparams(void); @@ -45,6 +45,10 @@ class EffectLFO float lfointensity; float lfornd; char lfotype; + + // current setup + float samplerate_f; + float buffersize_f; }; #endif diff --git a/src/Effects/EffectMgr.cpp b/src/Effects/EffectMgr.cpp @@ -76,28 +76,28 @@ void EffectMgr::changeeffect(int _nefx) delete efx; switch(nefx) { case 1: - efx = new Reverb(insertion, efxoutl, efxoutr); + efx = new Reverb(insertion, efxoutl, efxoutr, synth->samplerate, synth->buffersize); break; case 2: - efx = new Echo(insertion, efxoutl, efxoutr); + efx = new Echo(insertion, efxoutl, efxoutr, synth->samplerate, synth->buffersize); break; case 3: - efx = new Chorus(insertion, efxoutl, efxoutr); + efx = new Chorus(insertion, efxoutl, efxoutr, synth->samplerate, synth->buffersize); break; case 4: - efx = new Phaser(insertion, efxoutl, efxoutr); + efx = new Phaser(insertion, efxoutl, efxoutr, synth->samplerate, synth->buffersize); break; case 5: - efx = new Alienwah(insertion, efxoutl, efxoutr); + efx = new Alienwah(insertion, efxoutl, efxoutr, synth->samplerate, synth->buffersize); break; case 6: - efx = new Distorsion(insertion, efxoutl, efxoutr); + efx = new Distorsion(insertion, efxoutl, efxoutr, synth->samplerate, synth->buffersize); break; case 7: - efx = new EQ(insertion, efxoutl, efxoutr); + efx = new EQ(insertion, efxoutl, efxoutr, synth->samplerate, synth->buffersize); break; case 8: - efx = new DynamicFilter(insertion, efxoutl, efxoutr); + efx = new DynamicFilter(insertion, efxoutl, efxoutr, synth->samplerate, synth->buffersize); break; //put more effect here default: diff --git a/src/Effects/Phaser.cpp b/src/Effects/Phaser.cpp @@ -39,8 +39,8 @@ using namespace std; #define ONE_ 0.99999f // To prevent LFO ever reaching 1.0f for filter stability purposes #define ZERO_ 0.00001f // Same idea as above. -Phaser::Phaser(const int &insertion_, float *efxoutl_, float *efxoutr_) - :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0), old(NULL), xn1(NULL), +Phaser::Phaser(const int &insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize) + :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0, srate, bufsize), lfo(srate, bufsize), old(NULL), xn1(NULL), yn1(NULL), diff(0.0f), oldgain(0.0f), fb(0.0f) { analog_setup(); @@ -72,16 +72,20 @@ void Phaser::analog_setup() Rmx = Rmin / Rmax; Rconst = 1.0f + Rmx; // Handle parallel resistor relationship C = 0.00000005f; // 50 nF - CFs = 2.0f * synth->samplerate_f * C; - invperiod = 1.0f / synth->buffersize_f; + CFs = 2.0f * samplerate_f * C; + invperiod = 1.0f / buffersize_f; } Phaser::~Phaser() { + if(old.l) + delete[] old.l; if(xn1.l) delete[] xn1.l; if(yn1.l) delete[] yn1.l; + if(old.r) + delete[] old.r; if(xn1.r) delete[] xn1.r; if(yn1.r) @@ -129,7 +133,7 @@ void Phaser::AnalogPhase(const Stereo<float *> &input) g = oldgain; oldgain = mod; - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { g.l += diff.l; // Linear interpolation between LFO samples g.r += diff.r; @@ -151,8 +155,8 @@ void Phaser::AnalogPhase(const Stereo<float *> &input) } if(Poutsub) { - invSignal(efxoutl, synth->buffersize); - invSignal(efxoutr, synth->buffersize); + invSignal(efxoutl, buffersize); + invSignal(efxoutr, buffersize); } } @@ -202,8 +206,8 @@ void Phaser::normalPhase(const Stereo<float *> &input) gain.l = limit(gain.l, ZERO_, ONE_); gain.r = limit(gain.r, ZERO_, ONE_); - for(int i = 0; i < synth->buffersize; ++i) { - float x = (float) i / synth->buffersize_f; + for(int i = 0; i < buffersize; ++i) { + float x = (float) i / buffersize_f; float x1 = 1.0f - x; //TODO think about making panning an external feature Stereo<float> xn(input.l[i] * pangainL + fb.l, @@ -227,8 +231,8 @@ void Phaser::normalPhase(const Stereo<float *> &input) oldgain = gain; if(Poutsub) { - invSignal(efxoutl, synth->buffersize); - invSignal(efxoutr, synth->buffersize); + invSignal(efxoutl, buffersize); + invSignal(efxoutr, buffersize); } } @@ -299,10 +303,14 @@ void Phaser::setoffset(unsigned char Poffset) void Phaser::setstages(unsigned char Pstages) { + if(old.l) + delete[] old.l; if(xn1.l) delete[] xn1.l; if(yn1.l) delete[] yn1.l; + if(old.r) + delete[] old.r; if(xn1.r) delete[] xn1.r; if(yn1.r) diff --git a/src/Effects/Phaser.h b/src/Effects/Phaser.h @@ -35,7 +35,7 @@ class Phaser:public Effect { public: - Phaser(const int &insertion_, float *efxoutl_, float *efxoutr_); + Phaser(const int &insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize); ~Phaser(); void out(const Stereo<float *> &input); void setpreset(unsigned char npreset); diff --git a/src/Effects/Reverb.cpp b/src/Effects/Reverb.cpp @@ -28,8 +28,8 @@ //todo: EarlyReflections, Prdelay, Perbalance -Reverb::Reverb(bool insertion_, float *efxoutl_, float *efxoutr_) - :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0), +Reverb::Reverb(bool insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize) + :Effect(insertion_, efxoutl_, efxoutr_, NULL, 0, srate, bufsize), // defaults Pvolume(48), Ptime(64), @@ -116,7 +116,7 @@ void Reverb::processmono(int ch, float *output, float *inputbuf) const int comblength = comblen[j]; float &lpcombj = lpcomb[j]; - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { float fbout = comb[j][ck] * combfb[j]; fbout = fbout * (1.0f - lohifb) + lpcombj * lohifb; lpcombj = fbout; @@ -132,7 +132,7 @@ void Reverb::processmono(int ch, float *output, float *inputbuf) for(int j = REV_APS * ch; j < REV_APS * (1 + ch); ++j) { int &ak = apk[j]; const int aplength = aplen[j]; - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { float tmp = ap[j][ak]; ap[j][ak] = 0.7f * tmp + output[i]; output[i] = tmp - 0.7f * ap[j][ak]; @@ -148,12 +148,12 @@ void Reverb::out(const Stereo<float *> &smp) if(!Pvolume && insertion) return; - float *inputbuf = getTmpBuffer(); - for(int i = 0; i < synth->buffersize; ++i) + float inputbuf[buffersize]; + for(int i = 0; i < buffersize; ++i) inputbuf[i] = (smp.l[i] + smp.r[i]) / 2.0f; if(idelay) - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { //Initial delay r float tmp = inputbuf[i] + idelay[idelayk] * idelayfb; inputbuf[i] = idelay[idelayk]; @@ -164,7 +164,7 @@ void Reverb::out(const Stereo<float *> &smp) } if(bandwidth) - bandwidth->process(synth->buffersize, inputbuf); + bandwidth->process(buffersize, inputbuf); if(lpf) lpf->filterout(inputbuf); @@ -173,7 +173,6 @@ void Reverb::out(const Stereo<float *> &smp) processmono(0, efxoutl, inputbuf); //left processmono(1, efxoutr, inputbuf); //right - returnTmpBuffer(inputbuf); float lvol = rs / REV_COMBS * pangainL; float rvol = rs / REV_COMBS * pangainR; @@ -181,7 +180,7 @@ void Reverb::out(const Stereo<float *> &smp) lvol *= 2.0f; rvol *= 2.0f; } - for(int i = 0; i < synth->buffersize; ++i) { + for(int i = 0; i < buffersize; ++i) { efxoutl[i] *= lvol; efxoutr[i] *= rvol; } @@ -210,7 +209,7 @@ void Reverb::settime(unsigned char _Ptime) for(int i = 0; i < REV_COMBS * 2; ++i) combfb[i] = - -expf((float)comblen[i] / synth->samplerate_f * logf(0.001f) / t); + -expf((float)comblen[i] / samplerate_f * logf(0.001f) / t); //the feedback is negative because it removes the DC } @@ -241,7 +240,7 @@ void Reverb::setidelay(unsigned char _Pidelay) delete [] idelay; idelay = NULL; - idelaylen = (int) (synth->samplerate_f * delay / 1000); + idelaylen = (int) (samplerate_f * delay / 1000); if(idelaylen > 1) { idelayk = 0; idelay = new float[idelaylen]; @@ -266,7 +265,7 @@ void Reverb::sethpf(unsigned char _Phpf) else { float fr = expf(powf(Phpf / 127.0f, 0.5f) * logf(10000.0f)) + 20.0f; if(hpf == NULL) - hpf = new AnalogFilter(3, fr, 1, 0); + hpf = new AnalogFilter(3, fr, 1, 0, samplerate, buffersize); else hpf->setfreq(fr); } @@ -283,7 +282,7 @@ void Reverb::setlpf(unsigned char _Plpf) else { float fr = expf(powf(Plpf / 127.0f, 0.5f) * logf(25000.0f)) + 40.0f; if(!lpf) - lpf = new AnalogFilter(2, fr, 1, 0); + lpf = new AnalogFilter(2, fr, 1, 0, samplerate, buffersize); else lpf->setfreq(fr); } @@ -315,7 +314,7 @@ void Reverb::settype(unsigned char _Ptype) Ptype = NUM_TYPES - 1; // adjust the combs according to the samplerate - float samplerate_adjust = synth->samplerate_f / 44100.0f; + float samplerate_adjust = samplerate_f / 44100.0f; float tmp; for(int i = 0; i < REV_COMBS * 2; ++i) { if(Ptype == 0) @@ -360,7 +359,7 @@ void Reverb::settype(unsigned char _Ptype) //not been verified yet. //As this cannot be resized in a RT context, a good upper bound should //be found - bandwidth = new Unison(synth->buffersize / 4 + 1, 2.0f); + bandwidth = new Unison(buffersize / 4 + 1, 2.0f, samplerate_f); bandwidth->setSize(50); bandwidth->setBaseFrequency(1.0f); } diff --git a/src/Effects/Reverb.h b/src/Effects/Reverb.h @@ -32,7 +32,7 @@ class Reverb:public Effect { public: - Reverb(bool insertion_, float *efxoutl_, float *efxoutr_); + Reverb(bool insertion_, float *efxoutl_, float *efxoutr_, unsigned int srate, int bufsize); ~Reverb(); void out(const Stereo<float *> &smp); void cleanup(void); diff --git a/src/Misc/Part.cpp b/src/Misc/Part.cpp @@ -153,21 +153,21 @@ void Part::defaultsinstrument() /* * Cleanup the part */ -void Part::cleanup(bool final) +void Part::cleanup(bool final_) { for(int k = 0; k < POLIPHONY; ++k) KillNotePos(k); for(int i = 0; i < synth->buffersize; ++i) { - partoutl[i] = final ? 0.0f : denormalkillbuf[i]; - partoutr[i] = final ? 0.0f : denormalkillbuf[i]; + partoutl[i] = final_ ? 0.0f : denormalkillbuf[i]; + partoutr[i] = final_ ? 0.0f : denormalkillbuf[i]; } ctl.resetall(); for(int nefx = 0; nefx < NUM_PART_EFX; ++nefx) partefx[nefx]->cleanup(); for(int n = 0; n < NUM_PART_EFX + 1; ++n) for(int i = 0; i < synth->buffersize; ++i) { - partfxinputl[n][i] = final ? 0.0f : denormalkillbuf[i]; - partfxinputr[n][i] = final ? 0.0f : denormalkillbuf[i]; + partfxinputl[n][i] = final_ ? 0.0f : denormalkillbuf[i]; + partfxinputr[n][i] = final_ ? 0.0f : denormalkillbuf[i]; } } diff --git a/src/Misc/Util.cpp b/src/Misc/Util.cpp @@ -203,7 +203,7 @@ void clearTmpBuffers(void) pool.clear(); } -float SYNTH_T::numRandom() const +float SYNTH_T::numRandom() { return RND; } diff --git a/src/Tests/EchoTest.h b/src/Tests/EchoTest.h @@ -45,7 +45,7 @@ class EchoTest:public CxxTest::TestSuite new float[synth->buffersize]); for(int i = 0; i < synth->buffersize; ++i) input->l[i] = input->r[i] = 0.0f; - testFX = new Echo(true, outL, outR); + testFX = new Echo(true, outL, outR, 44100, 256); } void tearDown() { diff --git a/src/globals.h b/src/globals.h @@ -242,7 +242,7 @@ struct SYNTH_T { bufferbytes = buffersize * sizeof(float); oscilsize_f = oscilsize; } - float numRandom(void) const; //defined in Util.cpp for now + static float numRandom(void); //defined in Util.cpp for now }; extern SYNTH_T *synth;