diff options
Diffstat (limited to 'attic/muse_qt4_evolution/synti/zynaddsubfx/Synth')
14 files changed, 0 insertions, 4284 deletions
diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/ADnote.C b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/ADnote.C deleted file mode 100644 index 574e2bea..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/ADnote.C +++ /dev/null @@ -1,984 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - ADnote.C - The "additive" synthesizer - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -*/ - -#include <math.h> -#include <stdlib.h> -#include <stdio.h> - - -#include "../globals.h" -#include "../Misc/Util.h" -#include "ADnote.h" - - -ADnote::ADnote(ADnoteParameters *pars,Controller *ctl_,REALTYPE freq,REALTYPE velocity,int portamento_,int midinote_){ - ready=0; - - tmpwave=new REALTYPE [SOUND_BUFFER_SIZE]; - bypassl=new REALTYPE [SOUND_BUFFER_SIZE]; - bypassr=new REALTYPE [SOUND_BUFFER_SIZE]; - - partparams=pars; - ctl=ctl_; - portamento=portamento_; - midinote=midinote_; - NoteEnabled=ON; - basefreq=freq; - if (velocity>1.0) velocity=1.0; - this->velocity=velocity; - time=0.0; - stereo=pars->GlobalPar.PStereo; - - NoteGlobalPar.Detune=getdetune(pars->GlobalPar.PDetuneType - ,pars->GlobalPar.PCoarseDetune,pars->GlobalPar.PDetune); - bandwidthDetuneMultiplier=pars->getBandwidthDetuneMultiplier(); - - if (pars->GlobalPar.PPanning==0) NoteGlobalPar.Panning=RND; - else NoteGlobalPar.Panning=pars->GlobalPar.PPanning/128.0; - - - NoteGlobalPar.FilterCenterPitch=pars->GlobalPar.GlobalFilter->getfreq()+//center freq - pars->GlobalPar.PFilterVelocityScale/127.0*6.0* //velocity sensing - (VelF(velocity,pars->GlobalPar.PFilterVelocityScaleFunction)-1); - - if (pars->GlobalPar.PPunchStrength!=0) { - NoteGlobalPar.Punch.Enabled=1; - NoteGlobalPar.Punch.t=1.0;//start from 1.0 and to 0.0 - NoteGlobalPar.Punch.initialvalue=( (pow(10,1.5*pars->GlobalPar.PPunchStrength/127.0)-1.0) - *VelF(velocity,pars->GlobalPar.PPunchVelocitySensing) ); - REALTYPE time=pow(10,3.0*pars->GlobalPar.PPunchTime/127.0)/10000.0;//0.1 .. 100 ms - REALTYPE stretch=pow(440.0/freq,pars->GlobalPar.PPunchStretch/64.0); - NoteGlobalPar.Punch.dt=1.0/(time*SAMPLE_RATE*stretch); - } else NoteGlobalPar.Punch.Enabled=0; - - for (int nvoice=0;nvoice<NUM_VOICES;nvoice++){ - pars->VoicePar[nvoice].OscilSmp->newrandseed(rand()); - NoteVoicePar[nvoice].OscilSmp=NULL; - NoteVoicePar[nvoice].FMSmp=NULL; - NoteVoicePar[nvoice].VoiceOut=NULL; - - NoteVoicePar[nvoice].FMVoice=-1; - - if (pars->VoicePar[nvoice].Enabled==0) { - NoteVoicePar[nvoice].Enabled=OFF; - continue; //the voice is disabled - }; - - NoteVoicePar[nvoice].Enabled=ON; - NoteVoicePar[nvoice].fixedfreq=pars->VoicePar[nvoice].Pfixedfreq; - NoteVoicePar[nvoice].fixedfreqET=pars->VoicePar[nvoice].PfixedfreqET; - - //use the Globalpars.detunetype if the detunetype is 0 - if (pars->VoicePar[nvoice].PDetuneType!=0){ - NoteVoicePar[nvoice].Detune=getdetune(pars->VoicePar[nvoice].PDetuneType - ,pars->VoicePar[nvoice].PCoarseDetune,8192);//coarse detune - NoteVoicePar[nvoice].FineDetune=getdetune(pars->VoicePar[nvoice].PDetuneType - ,0,pars->VoicePar[nvoice].PDetune);//fine detune - } else { - NoteVoicePar[nvoice].Detune=getdetune(pars->GlobalPar.PDetuneType - ,pars->VoicePar[nvoice].PCoarseDetune,8192);//coarse detune - NoteVoicePar[nvoice].FineDetune=getdetune(pars->GlobalPar.PDetuneType - ,0,pars->VoicePar[nvoice].PDetune);//fine detune - }; - if (pars->VoicePar[nvoice].PFMDetuneType!=0){ - NoteVoicePar[nvoice].FMDetune=getdetune(pars->VoicePar[nvoice].PFMDetuneType - ,pars->VoicePar[nvoice].PFMCoarseDetune,pars->VoicePar[nvoice].PFMDetune); - } else { - NoteVoicePar[nvoice].FMDetune=getdetune(pars->GlobalPar.PDetuneType - ,pars->VoicePar[nvoice].PFMCoarseDetune,pars->VoicePar[nvoice].PFMDetune); - }; - - oscposhi[nvoice]=0;oscposlo[nvoice]=0.0; - oscposhiFM[nvoice]=0;oscposloFM[nvoice]=0.0; - - NoteVoicePar[nvoice].OscilSmp=new REALTYPE[OSCIL_SIZE+OSCIL_SMP_EXTRA_SAMPLES];//the extra points contains the first point - - //Get the voice's oscil or external's voice oscil - int vc=nvoice; - if (pars->VoicePar[nvoice].Pextoscil!=-1) vc=pars->VoicePar[nvoice].Pextoscil; - if (!pars->GlobalPar.Hrandgrouping) pars->VoicePar[vc].OscilSmp->newrandseed(rand()); - oscposhi[nvoice]=pars->VoicePar[vc].OscilSmp->get(NoteVoicePar[nvoice].OscilSmp,getvoicebasefreq(nvoice), - pars->VoicePar[nvoice].Presonance); - - //I store the first elments to the last position for speedups - for (int i=0;i<OSCIL_SMP_EXTRA_SAMPLES;i++) NoteVoicePar[nvoice].OscilSmp[OSCIL_SIZE+i]=NoteVoicePar[nvoice].OscilSmp[i]; - - oscposhi[nvoice]+=(int)((pars->VoicePar[nvoice].Poscilphase-64.0)/128.0*OSCIL_SIZE+OSCIL_SIZE*4); - oscposhi[nvoice]%=OSCIL_SIZE; - - - NoteVoicePar[nvoice].FreqLfo=NULL; - NoteVoicePar[nvoice].FreqEnvelope=NULL; - - NoteVoicePar[nvoice].AmpLfo=NULL; - NoteVoicePar[nvoice].AmpEnvelope=NULL; - - NoteVoicePar[nvoice].VoiceFilter=NULL; - NoteVoicePar[nvoice].FilterEnvelope=NULL; - NoteVoicePar[nvoice].FilterLfo=NULL; - - NoteVoicePar[nvoice].FilterCenterPitch=pars->VoicePar[nvoice].VoiceFilter->getfreq(); - NoteVoicePar[nvoice].filterbypass=pars->VoicePar[nvoice].Pfilterbypass; - - switch(pars->VoicePar[nvoice].PFMEnabled){ - case 1:NoteVoicePar[nvoice].FMEnabled=MORPH;break; - case 2:NoteVoicePar[nvoice].FMEnabled=RING_MOD;break; - case 3:NoteVoicePar[nvoice].FMEnabled=PHASE_MOD;break; - case 4:NoteVoicePar[nvoice].FMEnabled=FREQ_MOD;break; - case 5:NoteVoicePar[nvoice].FMEnabled=PITCH_MOD;break; - default:NoteVoicePar[nvoice].FMEnabled=NONE; - }; - - NoteVoicePar[nvoice].FMVoice=pars->VoicePar[nvoice].PFMVoice; - NoteVoicePar[nvoice].FMFreqEnvelope=NULL; - NoteVoicePar[nvoice].FMAmpEnvelope=NULL; - - //Compute the Voice's modulator volume (incl. damping) - REALTYPE fmvoldamp=pow(440.0/getvoicebasefreq(nvoice),pars->VoicePar[nvoice].PFMVolumeDamp/64.0-1.0); - switch (NoteVoicePar[nvoice].FMEnabled){ - case PHASE_MOD:fmvoldamp=pow(440.0/getvoicebasefreq(nvoice),pars->VoicePar[nvoice].PFMVolumeDamp/64.0); - NoteVoicePar[nvoice].FMVolume=(exp(pars->VoicePar[nvoice].PFMVolume/127.0*FM_AMP_MULTIPLIER)-1.0)*fmvoldamp*4.0; - break; - case FREQ_MOD:NoteVoicePar[nvoice].FMVolume=(exp(pars->VoicePar[nvoice].PFMVolume/127.0*FM_AMP_MULTIPLIER)-1.0)*fmvoldamp*4.0; - break; - // case PITCH_MOD:NoteVoicePar[nvoice].FMVolume=(pars->VoicePar[nvoice].PFMVolume/127.0*8.0)*fmvoldamp;//??????????? - // break; - default:if (fmvoldamp>1.0) fmvoldamp=1.0; - NoteVoicePar[nvoice].FMVolume=pars->VoicePar[nvoice].PFMVolume/127.0*fmvoldamp; - }; - - //Voice's modulator velocity sensing - NoteVoicePar[nvoice].FMVolume*=VelF(velocity,partparams->VoicePar[nvoice].PFMVelocityScaleFunction); - - FMoldsmp[nvoice]=0.0;//this is for FM (integration) - - firsttick[nvoice]=1; - NoteVoicePar[nvoice].DelayTicks=(int)((exp(pars->VoicePar[nvoice].PDelay/127.0*log(50.0))-1.0)/SOUND_BUFFER_SIZE/10.0*SAMPLE_RATE); - }; - - initparameters(); - ready=1; -}; - - -/* - * Kill a voice of ADnote - */ -void ADnote::KillVoice(int nvoice){ - - delete (NoteVoicePar[nvoice].OscilSmp); - - if (NoteVoicePar[nvoice].FreqEnvelope!=NULL) delete(NoteVoicePar[nvoice].FreqEnvelope); - NoteVoicePar[nvoice].FreqEnvelope=NULL; - - if (NoteVoicePar[nvoice].FreqLfo!=NULL) delete(NoteVoicePar[nvoice].FreqLfo); - NoteVoicePar[nvoice].FreqLfo=NULL; - - if (NoteVoicePar[nvoice].AmpEnvelope!=NULL) delete (NoteVoicePar[nvoice].AmpEnvelope); - NoteVoicePar[nvoice].AmpEnvelope=NULL; - - if (NoteVoicePar[nvoice].AmpLfo!=NULL) delete (NoteVoicePar[nvoice].AmpLfo); - NoteVoicePar[nvoice].AmpLfo=NULL; - - if (NoteVoicePar[nvoice].VoiceFilter!=NULL) delete (NoteVoicePar[nvoice].VoiceFilter); - NoteVoicePar[nvoice].VoiceFilter=NULL; - - if (NoteVoicePar[nvoice].FilterEnvelope!=NULL) delete (NoteVoicePar[nvoice].FilterEnvelope); - NoteVoicePar[nvoice].FilterEnvelope=NULL; - - if (NoteVoicePar[nvoice].FilterLfo!=NULL) delete (NoteVoicePar[nvoice].FilterLfo); - NoteVoicePar[nvoice].FilterLfo=NULL; - - if (NoteVoicePar[nvoice].FMFreqEnvelope!=NULL) delete (NoteVoicePar[nvoice].FMFreqEnvelope); - NoteVoicePar[nvoice].FMFreqEnvelope=NULL; - - if (NoteVoicePar[nvoice].FMAmpEnvelope!=NULL) delete (NoteVoicePar[nvoice].FMAmpEnvelope); - NoteVoicePar[nvoice].FMAmpEnvelope=NULL; - - if ((NoteVoicePar[nvoice].FMEnabled!=NONE)&&(NoteVoicePar[nvoice].FMVoice<0)) delete NoteVoicePar[nvoice].FMSmp; - - if (NoteVoicePar[nvoice].VoiceOut!=NULL) - for (int i=0;i<SOUND_BUFFER_SIZE;i++) NoteVoicePar[nvoice].VoiceOut[i]=0.0;//do not delete, yet: perhaps is used by another voice - - NoteVoicePar[nvoice].Enabled=OFF; -}; - -/* - * Kill the note - */ -void ADnote::KillNote(){ - int nvoice; - for (nvoice=0;nvoice<NUM_VOICES;nvoice++){ - if (NoteVoicePar[nvoice].Enabled==ON) KillVoice(nvoice); - - //delete VoiceOut - if (NoteVoicePar[nvoice].VoiceOut!=NULL) delete(NoteVoicePar[nvoice].VoiceOut); - NoteVoicePar[nvoice].VoiceOut=NULL; - }; - - delete (NoteGlobalPar.FreqEnvelope); - delete (NoteGlobalPar.FreqLfo); - delete (NoteGlobalPar.AmpEnvelope); - delete (NoteGlobalPar.AmpLfo); - delete (NoteGlobalPar.GlobalFilterL); - if (stereo!=0) delete (NoteGlobalPar.GlobalFilterR); - delete (NoteGlobalPar.FilterEnvelope); - delete (NoteGlobalPar.FilterLfo); - - NoteEnabled=OFF; -}; - -ADnote::~ADnote(){ - if (NoteEnabled==ON) KillNote(); - delete [] tmpwave; - delete [] bypassl; - delete [] bypassr; -}; - - -/* - * Init the parameters - */ -void ADnote::initparameters(){ - int nvoice,i,tmp[NUM_VOICES]; - - // Global Parameters - NoteGlobalPar.FreqEnvelope=new Envelope(partparams->GlobalPar.FreqEnvelope,basefreq); - NoteGlobalPar.FreqLfo=new LFO(partparams->GlobalPar.FreqLfo,basefreq); - - NoteGlobalPar.AmpEnvelope=new Envelope(partparams->GlobalPar.AmpEnvelope,basefreq); - NoteGlobalPar.AmpLfo=new LFO(partparams->GlobalPar.AmpLfo,basefreq); - - NoteGlobalPar.Volume=4.0*pow(0.1,3.0*(1.0-partparams->GlobalPar.PVolume/96.0))//-60 dB .. 0 dB - *VelF(velocity,partparams->GlobalPar.PAmpVelocityScaleFunction);//velocity sensing - - NoteGlobalPar.AmpEnvelope->envout_dB();//discard the first envelope output - globalnewamplitude=NoteGlobalPar.Volume*NoteGlobalPar.AmpEnvelope->envout_dB()*NoteGlobalPar.AmpLfo->amplfoout(); - - NoteGlobalPar.GlobalFilterL=new Filter(partparams->GlobalPar.GlobalFilter); - if (stereo!=0) NoteGlobalPar.GlobalFilterR=new Filter(partparams->GlobalPar.GlobalFilter); - - NoteGlobalPar.FilterEnvelope=new Envelope(partparams->GlobalPar.FilterEnvelope,basefreq); - NoteGlobalPar.FilterLfo=new LFO(partparams->GlobalPar.FilterLfo,basefreq); - NoteGlobalPar.FilterQ=partparams->GlobalPar.GlobalFilter->getq(); - NoteGlobalPar.FilterFreqTracking=partparams->GlobalPar.GlobalFilter->getfreqtracking(basefreq); - - // Forbids the Modulation Voice to be greater or equal than voice - for (i=0;i<NUM_VOICES;i++) if (NoteVoicePar[i].FMVoice>=i) NoteVoicePar[i].FMVoice=-1; - - // Voice Parameter init - for (nvoice=0;nvoice<NUM_VOICES;nvoice++){ - if (NoteVoicePar[nvoice].Enabled==0) continue; - - NoteVoicePar[nvoice].noisetype=partparams->VoicePar[nvoice].Type; - /* Voice Amplitude Parameters Init */ - NoteVoicePar[nvoice].Volume=pow(0.1,3.0*(1.0-partparams->VoicePar[nvoice].PVolume/127.0)) // -60 dB .. 0 dB - *VelF(velocity,partparams->VoicePar[nvoice].PAmpVelocityScaleFunction);//velocity - - if (partparams->VoicePar[nvoice].PVolumeminus!=0) NoteVoicePar[nvoice].Volume=-NoteVoicePar[nvoice].Volume; - - if (partparams->VoicePar[nvoice].PPanning==0) - NoteVoicePar[nvoice].Panning=RND;// random panning - else NoteVoicePar[nvoice].Panning=partparams->VoicePar[nvoice].PPanning/128.0; - - newamplitude[nvoice]=1.0; - if (partparams->VoicePar[nvoice].PAmpEnvelopeEnabled!=0) { - NoteVoicePar[nvoice].AmpEnvelope=new Envelope(partparams->VoicePar[nvoice].AmpEnvelope,basefreq); - NoteVoicePar[nvoice].AmpEnvelope->envout_dB();//discard the first envelope sample - newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpEnvelope->envout_dB(); - }; - - if (partparams->VoicePar[nvoice].PAmpLfoEnabled!=0){ - NoteVoicePar[nvoice].AmpLfo=new LFO(partparams->VoicePar[nvoice].AmpLfo,basefreq); - newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpLfo->amplfoout(); - }; - - /* Voice Frequency Parameters Init */ - if (partparams->VoicePar[nvoice].PFreqEnvelopeEnabled!=0) - NoteVoicePar[nvoice].FreqEnvelope=new Envelope(partparams->VoicePar[nvoice].FreqEnvelope,basefreq); - - if (partparams->VoicePar[nvoice].PFreqLfoEnabled!=0) NoteVoicePar[nvoice].FreqLfo=new LFO(partparams->VoicePar[nvoice].FreqLfo,basefreq); - - /* Voice Filter Parameters Init */ - if (partparams->VoicePar[nvoice].PFilterEnabled!=0){ - NoteVoicePar[nvoice].VoiceFilter=new Filter(partparams->VoicePar[nvoice].VoiceFilter); - }; - - if (partparams->VoicePar[nvoice].PFilterEnvelopeEnabled!=0) - NoteVoicePar[nvoice].FilterEnvelope=new Envelope(partparams->VoicePar[nvoice].FilterEnvelope,basefreq); - - if (partparams->VoicePar[nvoice].PFilterLfoEnabled!=0) - NoteVoicePar[nvoice].FilterLfo=new LFO(partparams->VoicePar[nvoice].FilterLfo,basefreq); - - NoteVoicePar[nvoice].FilterFreqTracking=partparams->VoicePar[nvoice].VoiceFilter->getfreqtracking(basefreq); - - /* Voice Modulation Parameters Init */ - if ((NoteVoicePar[nvoice].FMEnabled!=NONE)&&(NoteVoicePar[nvoice].FMVoice<0)){ - partparams->VoicePar[nvoice].FMSmp->newrandseed(rand()); - NoteVoicePar[nvoice].FMSmp=new REALTYPE[OSCIL_SIZE+OSCIL_SMP_EXTRA_SAMPLES]; - - //Perform Anti-aliasing only on MORPH or RING MODULATION - - int vc=nvoice; - if (partparams->VoicePar[nvoice].PextFMoscil!=-1) vc=partparams->VoicePar[nvoice].PextFMoscil; - - REALTYPE tmp=1.0; - if ((partparams->VoicePar[vc].FMSmp->Padaptiveharmonics!=0)|| - (NoteVoicePar[nvoice].FMEnabled==MORPH)|| - (NoteVoicePar[nvoice].FMEnabled==RING_MOD)){ - tmp=getFMvoicebasefreq(nvoice); - }; - if (!partparams->GlobalPar.Hrandgrouping) partparams->VoicePar[vc].FMSmp->newrandseed(rand()); - - oscposhiFM[nvoice]=(oscposhi[nvoice]+partparams->VoicePar[vc].FMSmp->get(NoteVoicePar[nvoice].FMSmp,tmp)) % OSCIL_SIZE; - for (int i=0;i<OSCIL_SMP_EXTRA_SAMPLES;i++) NoteVoicePar[nvoice].FMSmp[OSCIL_SIZE+i]=NoteVoicePar[nvoice].FMSmp[i]; - oscposhiFM[nvoice]+=(int)((partparams->VoicePar[nvoice].PFMoscilphase-64.0)/128.0*OSCIL_SIZE+OSCIL_SIZE*4); - oscposhiFM[nvoice]%=OSCIL_SIZE; - }; - - if (partparams->VoicePar[nvoice].PFMFreqEnvelopeEnabled!=0) - NoteVoicePar[nvoice].FMFreqEnvelope=new Envelope(partparams->VoicePar[nvoice].FMFreqEnvelope,basefreq); - - FMnewamplitude[nvoice]=NoteVoicePar[nvoice].FMVolume*ctl->fmamp.relamp; - - if (partparams->VoicePar[nvoice].PFMAmpEnvelopeEnabled!=0){ - NoteVoicePar[nvoice].FMAmpEnvelope=new Envelope(partparams->VoicePar[nvoice].FMAmpEnvelope,basefreq); - FMnewamplitude[nvoice]*=NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB(); - }; - }; - - for (nvoice=0;nvoice<NUM_VOICES;nvoice++){ - for (i=nvoice+1;i<NUM_VOICES;i++) tmp[i]=0; - for (i=nvoice+1;i<NUM_VOICES;i++) - if ((NoteVoicePar[i].FMVoice==nvoice)&&(tmp[i]==0)){ - NoteVoicePar[nvoice].VoiceOut=new REALTYPE[SOUND_BUFFER_SIZE]; - tmp[i]=1; - }; - if (NoteVoicePar[nvoice].VoiceOut!=NULL) for (i=0;i<SOUND_BUFFER_SIZE;i++) NoteVoicePar[nvoice].VoiceOut[i]=0.0; - }; -}; - - - -/* - * Computes the frequency of an oscillator - */ -void ADnote::setfreq(int nvoice,REALTYPE freq){ - REALTYPE speed; - freq=fabs(freq); - speed=freq*REALTYPE(OSCIL_SIZE)/(REALTYPE) SAMPLE_RATE; - if (speed>OSCIL_SIZE) speed=OSCIL_SIZE; - - F2I(speed,oscfreqhi[nvoice]); - oscfreqlo[nvoice]=speed-floor(speed); -}; - -/* - * Computes the frequency of an modullator oscillator - */ -void ADnote::setfreqFM(int nvoice,REALTYPE freq){ - REALTYPE speed; - freq=fabs(freq); - speed=freq*REALTYPE(OSCIL_SIZE)/(REALTYPE) SAMPLE_RATE; - if (speed>OSCIL_SIZE) speed=OSCIL_SIZE; - - F2I(speed,oscfreqhiFM[nvoice]); - oscfreqloFM[nvoice]=speed-floor(speed); -}; - -/* - * Get Voice base frequency - */ -REALTYPE ADnote::getvoicebasefreq(int nvoice){ - REALTYPE detune=NoteVoicePar[nvoice].Detune/100.0+ - NoteVoicePar[nvoice].FineDetune/100.0*ctl->bandwidth.relbw*bandwidthDetuneMultiplier+ - NoteGlobalPar.Detune/100.0; - - if (NoteVoicePar[nvoice].fixedfreq==0) return(this->basefreq*pow(2,detune/12.0)); - else {//the fixed freq is enabled - REALTYPE fixedfreq=440.0; - int fixedfreqET=NoteVoicePar[nvoice].fixedfreqET; - if (fixedfreqET!=0) {//if the frequency varies according the keyboard note - REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0); - if (fixedfreqET<=64) fixedfreq*=pow(2.0,tmp); - else fixedfreq*=pow(3.0,tmp); - }; - return(fixedfreq*pow(2.0,detune/12.0)); - }; -}; - -/* - * Get Voice's Modullator base frequency - */ -REALTYPE ADnote::getFMvoicebasefreq(int nvoice){ - REALTYPE detune=NoteVoicePar[nvoice].FMDetune/100.0; - return(getvoicebasefreq(nvoice)*pow(2,detune/12.0)); -}; - -/* - * Computes all the parameters for each tick - */ -void ADnote::computecurrentparameters(){ - int nvoice; - REALTYPE voicefreq,voicepitch,filterpitch,filterfreq,FMfreq,FMrelativepitch,globalpitch,globalfilterpitch; - globalpitch=0.01*(NoteGlobalPar.FreqEnvelope->envout()+ - NoteGlobalPar.FreqLfo->lfoout()*ctl->modwheel.relmod); - globaloldamplitude=globalnewamplitude; - globalnewamplitude=NoteGlobalPar.Volume*NoteGlobalPar.AmpEnvelope->envout_dB()*NoteGlobalPar.AmpLfo->amplfoout(); - - globalfilterpitch=NoteGlobalPar.FilterEnvelope->envout()+NoteGlobalPar.FilterLfo->lfoout() - +NoteGlobalPar.FilterCenterPitch; - - REALTYPE tmpfilterfreq=globalfilterpitch+ctl->filtercutoff.relfreq - +NoteGlobalPar.FilterFreqTracking; - - tmpfilterfreq=NoteGlobalPar.GlobalFilterL->getrealfreq(tmpfilterfreq); - - REALTYPE globalfilterq=NoteGlobalPar.FilterQ*ctl->filterq.relq; - NoteGlobalPar.GlobalFilterL->setfreq_and_q(tmpfilterfreq,globalfilterq); - if (stereo!=0) NoteGlobalPar.GlobalFilterR->setfreq_and_q(tmpfilterfreq,globalfilterq); - - //compute the portamento, if it is used by this note - REALTYPE portamentofreqrap=1.0; - if (portamento!=0){//this voice use portamento - portamentofreqrap=ctl->portamento.freqrap; - if (ctl->portamento.used==0){//the portamento has finished - portamento=0;//this note is no longer "portamented" - }; - }; - - //compute parameters for all voices - for (nvoice=0;nvoice<NUM_VOICES;nvoice++){ - if (NoteVoicePar[nvoice].Enabled!=ON) continue; - NoteVoicePar[nvoice].DelayTicks-=1; - if (NoteVoicePar[nvoice].DelayTicks>0) continue; - - /*******************/ - /* Voice Amplitude */ - /*******************/ - oldamplitude[nvoice]=newamplitude[nvoice]; - newamplitude[nvoice]=1.0; - - if (NoteVoicePar[nvoice].AmpEnvelope!=NULL) - newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpEnvelope->envout_dB(); - - if (NoteVoicePar[nvoice].AmpLfo!=NULL) - newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpLfo->amplfoout(); - - /****************/ - /* Voice Filter */ - /****************/ - if (NoteVoicePar[nvoice].VoiceFilter!=NULL){ - filterpitch=NoteVoicePar[nvoice].FilterCenterPitch; - - if (NoteVoicePar[nvoice].FilterEnvelope!=NULL) - filterpitch+=NoteVoicePar[nvoice].FilterEnvelope->envout(); - - if (NoteVoicePar[nvoice].FilterLfo!=NULL) - filterpitch+=NoteVoicePar[nvoice].FilterLfo->lfoout(); - - filterfreq=filterpitch+NoteVoicePar[nvoice].FilterFreqTracking; - filterfreq=NoteVoicePar[nvoice].VoiceFilter->getrealfreq(filterfreq); - - NoteVoicePar[nvoice].VoiceFilter->setfreq(filterfreq); - }; - - if (NoteVoicePar[nvoice].noisetype==0){//compute only if the voice isn't noise - - /*******************/ - /* Voice Frequency */ - /*******************/ - voicepitch=0.0; - if (NoteVoicePar[nvoice].FreqLfo!=NULL) - voicepitch+=NoteVoicePar[nvoice].FreqLfo->lfoout()/100.0 - *ctl->bandwidth.relbw; - - if (NoteVoicePar[nvoice].FreqEnvelope!=NULL) voicepitch+=NoteVoicePar[nvoice].FreqEnvelope->envout()/100.0; - voicefreq=getvoicebasefreq(nvoice)*pow(2,(voicepitch+globalpitch)/12.0);//Hz frequency - voicefreq*=ctl->pitchwheel.relfreq;//change the frequency by the controller - setfreq(nvoice,voicefreq*portamentofreqrap); - - /***************/ - /* Modulator */ - /***************/ - if (NoteVoicePar[nvoice].FMEnabled!=NONE){ - FMrelativepitch=NoteVoicePar[nvoice].FMDetune/100.0; - if (NoteVoicePar[nvoice].FMFreqEnvelope!=NULL) FMrelativepitch+=NoteVoicePar[nvoice].FMFreqEnvelope->envout()/100; - FMfreq=pow(2.0,FMrelativepitch/12.0)*voicefreq*portamentofreqrap; - setfreqFM(nvoice,FMfreq); - - FMoldamplitude[nvoice]=FMnewamplitude[nvoice]; - FMnewamplitude[nvoice]=NoteVoicePar[nvoice].FMVolume*ctl->fmamp.relamp; - if (NoteVoicePar[nvoice].FMAmpEnvelope!=NULL) - FMnewamplitude[nvoice]*=NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB(); - }; - }; - - }; - time+=(REALTYPE)SOUND_BUFFER_SIZE/(REALTYPE)SAMPLE_RATE; -}; - - -/* - * Fadein in a way that removes clicks but keep sound "punchy" - */ -inline void ADnote::fadein(REALTYPE *smps){ - int zerocrossings=0; - for (int i=1;i<SOUND_BUFFER_SIZE;i++) - if ((smps[i-1]<0.0) && (smps[i]>0.0)) zerocrossings++;//this is only the possitive crossings - - REALTYPE tmp=(SOUND_BUFFER_SIZE-1.0)/(zerocrossings+1)/3.0; - if (tmp<8.0) tmp=8.0; - - int n; - F2I(tmp,n);//how many samples is the fade-in - if (n>SOUND_BUFFER_SIZE) n=SOUND_BUFFER_SIZE; - for (int i=0;i<n;i++) {//fade-in - REALTYPE tmp=0.5-cos((REALTYPE)i/(REALTYPE) n*PI)*0.5; - smps[i]*=tmp; - }; -}; - -/* - * Computes the Oscillator (Without Modulation) - LinearInterpolation - */ -inline void ADnote::ComputeVoiceOscillator_LinearInterpolation(int nvoice){ - int i,poshi; - REALTYPE poslo; - - poshi=oscposhi[nvoice]; - poslo=oscposlo[nvoice]; - REALTYPE *smps=NoteVoicePar[nvoice].OscilSmp; - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - tmpwave[i]=smps[poshi]*(1.0-poslo)+smps[poshi+1]*poslo; - poslo+=oscfreqlo[nvoice]; - if (poslo>=1.0) { - poslo-=1.0; - poshi++; - }; - poshi+=oscfreqhi[nvoice]; - poshi&=OSCIL_SIZE-1; - }; - oscposhi[nvoice]=poshi; - oscposlo[nvoice]=poslo; -}; - - - -/* - * Computes the Oscillator (Without Modulation) - CubicInterpolation - * - The differences from the Linear are to little to deserve to be used. This is because I am using a large OSCIL_SIZE (>512) -inline void ADnote::ComputeVoiceOscillator_CubicInterpolation(int nvoice){ - int i,poshi; - REALTYPE poslo; - - poshi=oscposhi[nvoice]; - poslo=oscposlo[nvoice]; - REALTYPE *smps=NoteVoicePar[nvoice].OscilSmp; - REALTYPE xm1,x0,x1,x2,a,b,c; - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - xm1=smps[poshi]; - x0=smps[poshi+1]; - x1=smps[poshi+2]; - x2=smps[poshi+3]; - a=(3.0 * (x0-x1) - xm1 + x2) / 2.0; - b = 2.0*x1 + xm1 - (5.0*x0 + x2) / 2.0; - c = (x1 - xm1) / 2.0; - tmpwave[i]=(((a * poslo) + b) * poslo + c) * poslo + x0; - printf("a\n"); - //tmpwave[i]=smps[poshi]*(1.0-poslo)+smps[poshi+1]*poslo; - poslo+=oscfreqlo[nvoice]; - if (poslo>=1.0) { - poslo-=1.0; - poshi++; - }; - poshi+=oscfreqhi[nvoice]; - poshi&=OSCIL_SIZE-1; - }; - oscposhi[nvoice]=poshi; - oscposlo[nvoice]=poslo; -}; -*/ -/* - * Computes the Oscillator (Morphing) - */ -inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice){ - int i; - REALTYPE amp; - ComputeVoiceOscillator_LinearInterpolation(nvoice); - if (FMnewamplitude[nvoice]>1.0) FMnewamplitude[nvoice]=1.0; - if (FMoldamplitude[nvoice]>1.0) FMoldamplitude[nvoice]=1.0; - - if (NoteVoicePar[nvoice].FMVoice>=0){ - //if I use VoiceOut[] as modullator - int FMVoice=NoteVoicePar[nvoice].FMVoice; - for (i=0;i<SOUND_BUFFER_SIZE;i++) { - amp=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice] - ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE); - tmpwave[i]=tmpwave[i]*(1.0-amp)+amp*NoteVoicePar[FMVoice].VoiceOut[i]; - }; - } else { - int poshiFM=oscposhiFM[nvoice]; - REALTYPE posloFM=oscposloFM[nvoice]; - - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - amp=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice] - ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE); - tmpwave[i]=tmpwave[i]*(1.0-amp)+amp - *(NoteVoicePar[nvoice].FMSmp[poshiFM]*(1-posloFM) - +NoteVoicePar[nvoice].FMSmp[poshiFM+1]*posloFM); - posloFM+=oscfreqloFM[nvoice]; - if (posloFM>=1.0) { - posloFM-=1.0; - poshiFM++; - }; - poshiFM+=oscfreqhiFM[nvoice]; - poshiFM&=OSCIL_SIZE-1; - }; - oscposhiFM[nvoice]=poshiFM; - oscposloFM[nvoice]=posloFM; - }; -}; - -/* - * Computes the Oscillator (Ring Modulation) - */ -inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice){ - int i; - REALTYPE amp; - ComputeVoiceOscillator_LinearInterpolation(nvoice); - if (FMnewamplitude[nvoice]>1.0) FMnewamplitude[nvoice]=1.0; - if (FMoldamplitude[nvoice]>1.0) FMoldamplitude[nvoice]=1.0; - if (NoteVoicePar[nvoice].FMVoice>=0){ - // if I use VoiceOut[] as modullator - for (i=0;i<SOUND_BUFFER_SIZE;i++) { - amp=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice] - ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE); - int FMVoice=NoteVoicePar[nvoice].FMVoice; - for (i=0;i<SOUND_BUFFER_SIZE;i++) - tmpwave[i]*=(1.0-amp)+amp*NoteVoicePar[FMVoice].VoiceOut[i]; - }; - } else { - int poshiFM=oscposhiFM[nvoice]; - REALTYPE posloFM=oscposloFM[nvoice]; - - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - amp=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice] - ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE); - tmpwave[i]*=( NoteVoicePar[nvoice].FMSmp[poshiFM]*(1.0-posloFM) - +NoteVoicePar[nvoice].FMSmp[poshiFM+1]*posloFM)*amp - +(1.0-amp); - posloFM+=oscfreqloFM[nvoice]; - if (posloFM>=1.0) { - posloFM-=1.0; - poshiFM++; - }; - poshiFM+=oscfreqhiFM[nvoice]; - poshiFM&=OSCIL_SIZE-1; - }; - oscposhiFM[nvoice]=poshiFM; - oscposloFM[nvoice]=posloFM; - }; -}; - - - -/* - * Computes the Oscillator (Phase Modulation or Frequency Modulation) - */ -inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,int FMmode){ - int carposhi; - int i,FMmodfreqhi; - REALTYPE FMmodfreqlo,carposlo; - - if (NoteVoicePar[nvoice].FMVoice>=0){ - //if I use VoiceOut[] as modulator - for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]=NoteVoicePar[NoteVoicePar[nvoice].FMVoice].VoiceOut[i]; - } else { - //Compute the modulator and store it in tmpwave[] - int poshiFM=oscposhiFM[nvoice]; - REALTYPE posloFM=oscposloFM[nvoice]; - - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - tmpwave[i]=(NoteVoicePar[nvoice].FMSmp[poshiFM]*(1.0-posloFM) - +NoteVoicePar[nvoice].FMSmp[poshiFM+1]*posloFM); - posloFM+=oscfreqloFM[nvoice]; - if (posloFM>=1.0) { - posloFM=fmod(posloFM,1.0); - poshiFM++; - }; - poshiFM+=oscfreqhiFM[nvoice]; - poshiFM&=OSCIL_SIZE-1; - }; - oscposhiFM[nvoice]=poshiFM; - oscposloFM[nvoice]=posloFM; - }; - // Amplitude interpolation - if (ABOVE_AMPLITUDE_THRESHOLD(FMoldamplitude[nvoice],FMnewamplitude[nvoice])){ - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - tmpwave[i]*=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice] - ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE); - }; - } else for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]*=FMnewamplitude[nvoice]; - - - //normalize makes all sample-rates, oscil_sizes toproduce same sound - if (FMmode!=0){//Frequency modulation - REALTYPE normalize=OSCIL_SIZE/262144.0*44100.0/(REALTYPE)SAMPLE_RATE; - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - FMoldsmp[nvoice]=fmod(FMoldsmp[nvoice]+tmpwave[i]*normalize,OSCIL_SIZE); - tmpwave[i]=FMoldsmp[nvoice]; - }; - } else {//Phase modulation - REALTYPE normalize=OSCIL_SIZE/262144.0; - for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]*=normalize; - }; - - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - F2I(tmpwave[i],FMmodfreqhi); - FMmodfreqlo=fmod(tmpwave[i]+0.0000000001,1.0); - if (FMmodfreqhi<0) FMmodfreqlo++; - - //carrier - carposhi=oscposhi[nvoice]+FMmodfreqhi; - carposlo=oscposlo[nvoice]+FMmodfreqlo; - - if (carposlo>=1.0) { - carposhi++; - carposlo=fmod(carposlo,1.0); - }; - carposhi&=(OSCIL_SIZE-1); - - tmpwave[i]=NoteVoicePar[nvoice].OscilSmp[carposhi]*(1.0-carposlo) - +NoteVoicePar[nvoice].OscilSmp[carposhi+1]*carposlo; - - oscposlo[nvoice]+=oscfreqlo[nvoice]; - if (oscposlo[nvoice]>=1.0) { - oscposlo[nvoice]=fmod(oscposlo[nvoice],1.0); - oscposhi[nvoice]++; - }; - - oscposhi[nvoice]+=oscfreqhi[nvoice]; - oscposhi[nvoice]&=OSCIL_SIZE-1; - }; -}; - - -/*Calculeaza Oscilatorul cu PITCH MODULATION*/ -inline void ADnote::ComputeVoiceOscillatorPitchModulation(int nvoice){ -//TODO -}; - -/* - * Computes the Noise - */ -inline void ADnote::ComputeVoiceNoise(int nvoice){ - for (int i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]=RND*2.0-1.0; -}; - - - -/* - * Compute the ADnote samples - * Returns 0 if the note is finished - */ -int ADnote::noteout(REALTYPE *outl,REALTYPE *outr){ - int i,nvoice; - - for (i=0;i<SOUND_BUFFER_SIZE;i++) { - outl[i]=denormalkillbuf[i]; - outr[i]=denormalkillbuf[i]; - }; - - if (NoteEnabled==OFF) return(0); - - for (i=0;i<SOUND_BUFFER_SIZE;i++) { - bypassl[i]=0.0; - bypassr[i]=0.0; - }; - - computecurrentparameters(); - - for (nvoice=0;nvoice<NUM_VOICES;nvoice++){ - if ((NoteVoicePar[nvoice].Enabled!=ON) || (NoteVoicePar[nvoice].DelayTicks>0)) continue; - if (NoteVoicePar[nvoice].noisetype==0){//voice mode=sound - switch (NoteVoicePar[nvoice].FMEnabled){ - case MORPH:ComputeVoiceOscillatorMorph(nvoice);break; - case RING_MOD:ComputeVoiceOscillatorRingModulation(nvoice);break; - case PHASE_MOD:ComputeVoiceOscillatorFrequencyModulation(nvoice,0);break; - case FREQ_MOD:ComputeVoiceOscillatorFrequencyModulation(nvoice,1);break; - //case PITCH_MOD:ComputeVoiceOscillatorPitchModulation(nvoice);break; - default:ComputeVoiceOscillator_LinearInterpolation(nvoice); - //if (config.cfg.Interpolation) ComputeVoiceOscillator_CubicInterpolation(nvoice); - - }; - } else ComputeVoiceNoise(nvoice); - // Voice Processing - - // Amplitude - if (ABOVE_AMPLITUDE_THRESHOLD(oldamplitude[nvoice],newamplitude[nvoice])){ - int rest=SOUND_BUFFER_SIZE; - //test if the amplitude if raising and the difference is high - if ((newamplitude[nvoice]>oldamplitude[nvoice])&&((newamplitude[nvoice]-oldamplitude[nvoice])>0.25)){ - rest=10; - if (rest>SOUND_BUFFER_SIZE) rest=SOUND_BUFFER_SIZE; - for (int i=0;i<SOUND_BUFFER_SIZE-rest;i++) tmpwave[i]*=oldamplitude[nvoice]; - }; - // Amplitude interpolation - for (i=0;i<rest;i++){ - tmpwave[i+(SOUND_BUFFER_SIZE-rest)]*=INTERPOLATE_AMPLITUDE(oldamplitude[nvoice] - ,newamplitude[nvoice],i,rest); - }; - } else for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]*=newamplitude[nvoice]; - - // Fade in - if (firsttick[nvoice]!=0){ - fadein(&tmpwave[0]); - firsttick[nvoice]=0; - }; - - - // Filter - if (NoteVoicePar[nvoice].VoiceFilter!=NULL) NoteVoicePar[nvoice].VoiceFilter->filterout(&tmpwave[0]); - - //check if the amplitude envelope is finished, if yes, the voice will be fadeout - if (NoteVoicePar[nvoice].AmpEnvelope!=NULL) { - if (NoteVoicePar[nvoice].AmpEnvelope->finished()!=0) - for (i=0;i<SOUND_BUFFER_SIZE;i++) - tmpwave[i]*=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE; - //the voice is killed later - }; - - - // Put the ADnote samples in VoiceOut (without appling Global volume, because I wish to use this voice as a modullator) - if (NoteVoicePar[nvoice].VoiceOut!=NULL) - for (i=0;i<SOUND_BUFFER_SIZE;i++) NoteVoicePar[nvoice].VoiceOut[i]=tmpwave[i]; - - - // Add the voice that do not bypass the filter to out - if (NoteVoicePar[nvoice].filterbypass==0){//no bypass - if (stereo==0) for (i=0;i<SOUND_BUFFER_SIZE;i++) outl[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume;//mono - else for (i=0;i<SOUND_BUFFER_SIZE;i++) {//stereo - outl[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume*NoteVoicePar[nvoice].Panning*2.0; - outr[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume*(1.0-NoteVoicePar[nvoice].Panning)*2.0; - }; - } else {//bypass the filter - if (stereo==0) for (i=0;i<SOUND_BUFFER_SIZE;i++) bypassl[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume;//mono - else for (i=0;i<SOUND_BUFFER_SIZE;i++) {//stereo - bypassl[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume*NoteVoicePar[nvoice].Panning*2.0; - bypassr[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume*(1.0-NoteVoicePar[nvoice].Panning)*2.0; - }; - }; - // chech if there is necesary to proces the voice longer (if the Amplitude envelope isn't finished) - if (NoteVoicePar[nvoice].AmpEnvelope!=NULL) { - if (NoteVoicePar[nvoice].AmpEnvelope->finished()!=0) KillVoice(nvoice); - }; - }; - - - //Processing Global parameters - NoteGlobalPar.GlobalFilterL->filterout(&outl[0]); - - if (stereo==0) { - for (i=0;i<SOUND_BUFFER_SIZE;i++) {//set the right channel=left channel - outr[i]=outl[i]; - bypassr[i]=bypassl[i]; - } - } else NoteGlobalPar.GlobalFilterR->filterout(&outr[0]); - - for (i=0;i<SOUND_BUFFER_SIZE;i++) { - outl[i]+=bypassl[i]; - outr[i]+=bypassr[i]; - }; - - if (ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude,globalnewamplitude)){ - // Amplitude Interpolation - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - REALTYPE tmpvol=INTERPOLATE_AMPLITUDE(globaloldamplitude - ,globalnewamplitude,i,SOUND_BUFFER_SIZE); - outl[i]*=tmpvol*NoteGlobalPar.Panning; - outr[i]*=tmpvol*(1.0-NoteGlobalPar.Panning); - }; - } else { - for (i=0;i<SOUND_BUFFER_SIZE;i++) { - outl[i]*=globalnewamplitude*NoteGlobalPar.Panning; - outr[i]*=globalnewamplitude*(1.0-NoteGlobalPar.Panning); - }; - }; - - //Apply the punch - if (NoteGlobalPar.Punch.Enabled!=0){ - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - REALTYPE punchamp=NoteGlobalPar.Punch.initialvalue*NoteGlobalPar.Punch.t+1.0; - outl[i]*=punchamp; - outr[i]*=punchamp; - NoteGlobalPar.Punch.t-=NoteGlobalPar.Punch.dt; - if (NoteGlobalPar.Punch.t<0.0) { - NoteGlobalPar.Punch.Enabled=0; - break; - }; - }; - }; - - // Check if the global amplitude is finished. - // If it does, disable the note - if (NoteGlobalPar.AmpEnvelope->finished()!=0) { - for (i=0;i<SOUND_BUFFER_SIZE;i++) {//fade-out - REALTYPE tmp=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE; - outl[i]*=tmp; - outr[i]*=tmp; - }; - KillNote(); - }; - return(1); -}; - - -/* - * Relase the key (NoteOff) - */ -void ADnote::relasekey(){ -int nvoice; - for (nvoice=0;nvoice<NUM_VOICES;nvoice++){ - if (NoteVoicePar[nvoice].Enabled==0) continue; - if (NoteVoicePar[nvoice].AmpEnvelope!=NULL) NoteVoicePar[nvoice].AmpEnvelope->relasekey(); - if (NoteVoicePar[nvoice].FreqEnvelope!=NULL) NoteVoicePar[nvoice].FreqEnvelope->relasekey(); - if (NoteVoicePar[nvoice].FilterEnvelope!=NULL) NoteVoicePar[nvoice].FilterEnvelope->relasekey(); - if (NoteVoicePar[nvoice].FMFreqEnvelope!=NULL) NoteVoicePar[nvoice].FMFreqEnvelope->relasekey(); - if (NoteVoicePar[nvoice].FMAmpEnvelope!=NULL) NoteVoicePar[nvoice].FMAmpEnvelope->relasekey(); - }; - NoteGlobalPar.FreqEnvelope->relasekey(); - NoteGlobalPar.FilterEnvelope->relasekey(); - NoteGlobalPar.AmpEnvelope->relasekey(); - -}; - -/* - * Check if the note is finished - */ -int ADnote::finished(){ - if (NoteEnabled==ON) return(0); - else return(1); -}; - - - diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/ADnote.h b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/ADnote.h deleted file mode 100644 index 28c18975..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/ADnote.h +++ /dev/null @@ -1,258 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - ADnote.h - The "additive" synthesizer - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - -*/ - -#ifndef AD_NOTE_H -#define AD_NOTE_H - -#include "../globals.h" -#include "Envelope.h" -#include "LFO.h" -#include "../DSP/Filter.h" -#include "../Params/ADnoteParameters.h" -#include "../Params/Controller.h" - -//Globals - -//FM amplitude tune -#define FM_AMP_MULTIPLIER 14.71280603 - -#define OSCIL_SMP_EXTRA_SAMPLES 5 - -class ADnote{ //ADDitive note - public: - ADnote(ADnoteParameters *pars,Controller *ctl_,REALTYPE freq,REALTYPE velocity,int portamento_,int midinote_); - ~ADnote(); - int noteout(REALTYPE *outl,REALTYPE *outr); - void relasekey(); - int finished(); - - - /*ready - this is 0 if it is not ready (the parameters has to be computed) - or other value if the parameters has been computed and if it is ready to output*/ - char ready; - - private: - - void setfreq(int nvoice,REALTYPE freq); - void setfreqFM(int nvoice,REALTYPE freq); - void computecurrentparameters(); - void initparameters(); - void KillVoice(int nvoice); - void KillNote(); - inline REALTYPE getvoicebasefreq(int nvoice); - inline REALTYPE getFMvoicebasefreq(int nvoice); - inline void ComputeVoiceOscillator_LinearInterpolation(int nvoice); - inline void ComputeVoiceOscillator_CubicInterpolation(int nvoice); - inline void ComputeVoiceOscillatorMorph(int nvoice); - inline void ComputeVoiceOscillatorRingModulation(int nvoice); - inline void ComputeVoiceOscillatorFrequencyModulation(int nvoice,int FMmode);//FMmode=0 for phase modulation, 1 for Frequency modulation -// inline void ComputeVoiceOscillatorFrequencyModulation(int nvoice); - inline void ComputeVoiceOscillatorPitchModulation(int nvoice); - - inline void ComputeVoiceNoise(int nvoice); - - inline void fadein(REALTYPE *smps); - - - //GLOBALS - ADnoteParameters *partparams; - unsigned char stereo;//if the note is stereo (allows note Panning) - int midinote; - REALTYPE velocity,basefreq; - - ONOFFTYPE NoteEnabled; - Controller *ctl; - - /*****************************************************************/ - /* GLOBAL PARAMETERS */ - /*****************************************************************/ - - struct ADnoteGlobal{ - /****************************************** - * FREQUENCY GLOBAL PARAMETERS * - ******************************************/ - REALTYPE Detune;//cents - - Envelope *FreqEnvelope; - LFO *FreqLfo; - - /******************************************** - * AMPLITUDE GLOBAL PARAMETERS * - ********************************************/ - REALTYPE Volume;// [ 0 .. 1 ] - - REALTYPE Panning;// [ 0 .. 1 ] - - Envelope *AmpEnvelope; - LFO *AmpLfo; - - struct { - int Enabled; - REALTYPE initialvalue,dt,t; - } Punch; - - /****************************************** - * FILTER GLOBAL PARAMETERS * - ******************************************/ - Filter *GlobalFilterL,*GlobalFilterR; - - REALTYPE FilterCenterPitch;//octaves - REALTYPE FilterQ; - REALTYPE FilterFreqTracking; - - Envelope *FilterEnvelope; - - LFO *FilterLfo; - } NoteGlobalPar; - - - - /***********************************************************/ - /* VOICE PARAMETERS */ - /***********************************************************/ - struct ADnoteVoice{ - /* If the voice is enabled */ - ONOFFTYPE Enabled; - - /* Voice Type (sound/noise)*/ - int noisetype; - - /* Filter Bypass */ - int filterbypass; - - /* Delay (ticks) */ - int DelayTicks; - - /* Waveform of the Voice */ - REALTYPE *OscilSmp; - - /************************************ - * FREQUENCY PARAMETERS * - ************************************/ - int fixedfreq;//if the frequency is fixed to 440 Hz - int fixedfreqET;//if the "fixed" frequency varies according to the note (ET) - - // cents = basefreq*VoiceDetune - REALTYPE Detune,FineDetune; - - Envelope *FreqEnvelope; - LFO *FreqLfo; - - - /*************************** - * AMPLITUDE PARAMETERS * - ***************************/ - - /* Panning 0.0=left, 0.5 - center, 1.0 = right */ - REALTYPE Panning; - REALTYPE Volume;// [-1.0 .. 1.0] - - Envelope *AmpEnvelope; - LFO *AmpLfo; - - /************************* - * FILTER PARAMETERS * - *************************/ - - Filter *VoiceFilter; - - REALTYPE FilterCenterPitch;/* Filter center Pitch*/ - REALTYPE FilterFreqTracking; - - Envelope *FilterEnvelope; - LFO *FilterLfo; - - - /**************************** - * MODULLATOR PARAMETERS * - ****************************/ - - FMTYPE FMEnabled; - - int FMVoice; - - // Voice Output used by other voices if use this as modullator - REALTYPE *VoiceOut; - - /* Wave of the Voice */ - REALTYPE *FMSmp; - - REALTYPE FMVolume; - REALTYPE FMDetune; //in cents - - Envelope *FMFreqEnvelope; - Envelope *FMAmpEnvelope; - } NoteVoicePar[NUM_VOICES]; - - - /********************************************************/ - /* INTERNAL VALUES OF THE NOTE AND OF THE VOICES */ - /********************************************************/ - - //time from the start of the note - REALTYPE time; - - //fractional part (skip) - REALTYPE oscposlo[NUM_VOICES],oscfreqlo[NUM_VOICES]; - - //integer part (skip) - int oscposhi[NUM_VOICES],oscfreqhi[NUM_VOICES]; - - //fractional part (skip) of the Modullator - REALTYPE oscposloFM[NUM_VOICES],oscfreqloFM[NUM_VOICES]; - - //integer part (skip) of the Modullator - unsigned short int oscposhiFM[NUM_VOICES],oscfreqhiFM[NUM_VOICES]; - - //used to compute and interpolate the amplitudes of voices and modullators - REALTYPE oldamplitude[NUM_VOICES], - newamplitude[NUM_VOICES], - FMoldamplitude[NUM_VOICES], - FMnewamplitude[NUM_VOICES]; - - //used by Frequency Modulation (for integration) - REALTYPE FMoldsmp[NUM_VOICES]; - - //temporary buffer - REALTYPE *tmpwave; - - //Filter bypass samples - REALTYPE *bypassl,*bypassr; - - //interpolate the amplitudes - REALTYPE globaloldamplitude,globalnewamplitude; - - //1 - if it is the fitst tick (used to fade in the sound) - char firsttick[NUM_VOICES]; - - //1 if the note has portamento - int portamento; - - //how the fine detunes are made bigger or smaller - REALTYPE bandwidthDetuneMultiplier; -}; - -#endif - - - - diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Envelope.C b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Envelope.C deleted file mode 100644 index a0194022..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Envelope.C +++ /dev/null @@ -1,165 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - Envelope.C - Envelope implementation - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - -*/ - -#include <stdio.h> -#include "Envelope.h" - -Envelope::Envelope(EnvelopeParams *envpars,REALTYPE basefreq){ - int i; - envpoints=envpars->Penvpoints; - if (envpoints>MAX_ENVELOPE_POINTS) envpoints=MAX_ENVELOPE_POINTS; - envsustain=(envpars->Penvsustain==0)?-1:envpars->Penvsustain; - forcedrelase=envpars->Pforcedrelease; - envstretch=pow(440.0/basefreq,envpars->Penvstretch/64.0); - linearenvelope=envpars->Plinearenvelope; - - if (envpars->Pfreemode==0) envpars->converttofree(); - - REALTYPE bufferdt=SOUND_BUFFER_SIZE/(REALTYPE)SAMPLE_RATE; - - int mode=envpars->Envmode; - - //for amplitude envelopes - if ((mode==1)&&(linearenvelope==0)) mode=2;//change to log envelope - if ((mode==2)&&(linearenvelope!=0)) mode=1;//change to linear - - for (i=0;i<MAX_ENVELOPE_POINTS;i++) { - REALTYPE tmp=envpars->getdt(i)/1000.0*envstretch; - if (tmp>bufferdt) envdt[i]=bufferdt/tmp; - else envdt[i]=2.0;//any value larger than 1 - - switch (mode){ - case 2:envval[i]=(1.0-envpars->Penvval[i]/127.0)*MIN_ENVELOPE_DB; - break; - case 3:envval[i]=(pow(2,6.0*fabs(envpars->Penvval[i]-64.0)/64.0)-1.0)*100.0; - if (envpars->Penvval[i]<64) envval[i]=-envval[i]; - break; - case 4:envval[i]=(envpars->Penvval[i]-64.0)/64.0*6.0;//6 octaves (filtru) - break; - case 5:envval[i]=(envpars->Penvval[i]-64.0)/64.0*10; - break; - default:envval[i]=envpars->Penvval[i]/127.0; - }; - - }; - - envdt[0]=1.0; - - currentpoint=1;//the envelope starts from 1 - keyreleased=0; - t=0.0; - envfinish=0; - inct=envdt[1]; - envoutval=0.0; -}; - -Envelope::~Envelope(){ -}; - - -/* - * Relase the key (note envelope) - */ -void Envelope::relasekey(){ - if (keyreleased==1) return; - keyreleased=1; - if (forcedrelase!=0) t=0.0; -}; - -/* - * Envelope Output - */ -REALTYPE Envelope::envout(){ - REALTYPE out; - - if (envfinish!=0) {//if the envelope is finished - envoutval=envval[envpoints-1]; - return(envoutval); - }; - if ((currentpoint==envsustain+1)&&(keyreleased==0)) {//if it is sustaining now - envoutval=envval[envsustain]; - return(envoutval); - }; - - if ((keyreleased!=0) && (forcedrelase!=0)){//do the forced release - - int tmp=(envsustain<0) ? (envpoints-1):(envsustain+1);//if there is no sustain point, use the last point for release - - if (envdt[tmp]<0.00000001) out=envval[tmp]; - else out=envoutval+(envval[tmp]-envoutval)*t; - t+=envdt[tmp]*envstretch; - - if (t>=1.0) { - currentpoint=envsustain+2; - forcedrelase=0; - t=0.0; - inct=envdt[currentpoint]; - if ((currentpoint>=envpoints)||(envsustain<0)) envfinish=1; - }; - return(out); - }; - if (inct>=1.0) out=envval[currentpoint]; - else out=envval[currentpoint-1]+(envval[currentpoint]-envval[currentpoint-1])*t; - - t+=inct; - if (t>=1.0){ - if (currentpoint>=envpoints-1) envfinish=1; - else currentpoint++; - t=0.0; - inct=envdt[currentpoint]; - }; - - envoutval=out; - return (out); -}; - -/* - * Envelope Output (dB) - */ -REALTYPE Envelope::envout_dB(){ - REALTYPE out; - if (linearenvelope!=0) return (envout()); - - if ((currentpoint==1)&&((keyreleased==0)||(forcedrelase==0))) {//first point is always lineary interpolated - REALTYPE v1=dB2rap(envval[0]); - REALTYPE v2=dB2rap(envval[1]); - out=v1+(v2-v1)*t; - - t+=inct; - if (t>=1.0) { - t=0.0; - inct=envdt[2]; - currentpoint++; - out=v2; - }; - - if (out>0.001) envoutval=rap2dB(out); - else envoutval=-40.0; - } else out=dB2rap(envout()); - - return(out); -}; - -int Envelope::finished(){ - return(envfinish); -}; - diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Envelope.h b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Envelope.h deleted file mode 100644 index d78eb16d..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Envelope.h +++ /dev/null @@ -1,58 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - Envelope.h - Envelope implementation - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - -*/ - -#ifndef ENVELOPE_H -#define ENVELOPE_H - -#include <math.h> -#include "../globals.h" -#include "../Params/EnvelopeParams.h" - -class Envelope{ -public: - Envelope(EnvelopeParams *envpars,REALTYPE basefreq); - ~Envelope(); - void relasekey(); - REALTYPE envout(); - REALTYPE envout_dB(); - int finished();//returns 1 if the envelope is finished -private: - int envpoints; - int envsustain;//"-1" means disabled - REALTYPE envdt[MAX_ENVELOPE_POINTS];//millisecons - REALTYPE envval[MAX_ENVELOPE_POINTS];// [0.0 .. 1.0] - REALTYPE envstretch; - int linearenvelope; - - int currentpoint; //current envelope point (starts from 1) - int forcedrelase; - char keyreleased; //if the key was released - char envfinish; - REALTYPE t; // the time from the last point - REALTYPE inct;// the time increment - REALTYPE envoutval;//used to do the forced release -}; - - -#endif - - diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/LFO.C b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/LFO.C deleted file mode 100644 index 4ae548c1..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/LFO.C +++ /dev/null @@ -1,145 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - LFO.C - LFO implementation - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - -*/ - -#include <stdlib.h> -#include <stdio.h> -#include <math.h> - -#include "LFO.h" - - -LFO::LFO(LFOParams *lfopars,REALTYPE basefreq){ - if (lfopars->Pstretch==0) lfopars->Pstretch=1; - REALTYPE lfostretch=pow(basefreq/440.0,(lfopars->Pstretch-64.0)/63.0);//max 2x/octave - - REALTYPE lfofreq=(pow(2,lfopars->Pfreq*10.0)-1.0)/12.0*lfostretch; - incx=fabs(lfofreq)*(REALTYPE)SOUND_BUFFER_SIZE/(REALTYPE)SAMPLE_RATE; - - if (lfopars->Pcontinous==0){ - if (lfopars->Pstartphase==0) x=RND; - else x=fmod((lfopars->Pstartphase-64.0)/127.0+1.0,1.0); - } else { - REALTYPE tmp=fmod(lfopars->time*incx,1.0); - x=fmod((lfopars->Pstartphase-64.0)/127.0+1.0+tmp,1.0); - }; - - //Limit the Frequency(or else...) - if (incx>0.49999999) incx=0.499999999; - - - lfornd=lfopars->Prandomness/127.0; - if (lfornd<0.0) lfornd=0.0; else if (lfornd>1.0) lfornd=1.0; - -// lfofreqrnd=pow(lfopars->Pfreqrand/127.0,2.0)*2.0*4.0; - lfofreqrnd=pow(lfopars->Pfreqrand/127.0,2.0)*4.0; - - switch (lfopars->fel){ - case 1:lfointensity=lfopars->Pintensity/127.0;break; - case 2:lfointensity=lfopars->Pintensity/127.0*4.0;break;//in octave - default:lfointensity=pow(2,lfopars->Pintensity/127.0*11.0)-1.0;//in centi - x-=0.25;//chance the starting phase - break; - }; - - amp1=(1-lfornd)+lfornd*RND; - amp2=(1-lfornd)+lfornd*RND; - lfotype=lfopars->PLFOtype; - lfodelay=lfopars->Pdelay/127.0*4.0;//0..4 sec - incrnd=nextincrnd=1.0; - freqrndenabled=(lfopars->Pfreqrand!=0); - computenextincrnd(); - computenextincrnd();//twice because I want incrnd & nextincrnd to be random -}; - -LFO::~LFO(){ -}; - -/* - * LFO out - */ -REALTYPE LFO::lfoout(){ - REALTYPE out; - switch (lfotype){ - case 1: //LFO_TRIANGLE - if ((x>=0.0)&&(x<0.25)) out=4.0*x; - else if ((x>0.25)&&(x<0.75)) out=2-4*x; - else out=4.0*x-4.0; - break; - case 2: //LFO_SQUARE - if (x<0.5) out=-1; - else out=1; - break; - case 3: //LFO_RAMPUP - out=(x-0.5)*2.0; - break; - case 4: //LFO_RAMPDOWN - out=(0.5-x)*2.0; - break; - case 5: //LFO_EXP_DOWN 1 - out=pow(0.05,x)*2.0-1.0; - break; - case 6: //LFO_EXP_DOWN 2 - out=pow(0.001,x)*2.0-1.0; - break; - default:out=cos(x*2.0*PI);//LFO_SINE - }; - - - if ((lfotype==0)||(lfotype==1)) out*=lfointensity*(amp1+x*(amp2-amp1)); - else out*=lfointensity*amp2; - if (lfodelay<0.00001) { - if (freqrndenabled==0) x+=incx; - else { - float tmp=(incrnd*(1.0-x)+nextincrnd*x); - if (tmp>1.0) tmp=1.0; - else if (tmp<0.0) tmp=0.0; - x+=incx*tmp; - }; - if (x>=1) { - x=fmod(x,1.0); - amp1=amp2; - amp2=(1-lfornd)+lfornd*RND; - - computenextincrnd(); - }; - } else lfodelay-=(REALTYPE)SOUND_BUFFER_SIZE/(REALTYPE)SAMPLE_RATE; - return(out); -}; - -/* - * LFO out (for amplitude) - */ -REALTYPE LFO::amplfoout(){ - REALTYPE out; - out=1.0-lfointensity+lfoout(); - if (out<-1.0) out=-1.0; - else if (out>1.0) out=1.0; - return(out); -}; - - -void LFO::computenextincrnd(){ - if (freqrndenabled==0) return; - incrnd=nextincrnd; - nextincrnd=pow(0.5,lfofreqrnd)+RND*(pow(2.0,lfofreqrnd)-1.0); -}; - diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/LFO.h b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/LFO.h deleted file mode 100644 index 30d04f10..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/LFO.h +++ /dev/null @@ -1,52 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - LFO.h - LFO implementation - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - -*/ - -#ifndef LFO_H -#define LFO_H - -#include "../globals.h" -#include "../Params/LFOParams.h" - - -class LFO{ - public: - LFO(LFOParams *lfopars, REALTYPE basefreq); - ~LFO(); - REALTYPE lfoout(); - REALTYPE amplfoout(); - private: - REALTYPE x; - REALTYPE incx,incrnd,nextincrnd; - REALTYPE amp1,amp2;// used for randomness - REALTYPE lfointensity; - REALTYPE lfornd,lfofreqrnd; - REALTYPE lfodelay; - char lfotype; - int freqrndenabled; - - - void computenextincrnd(); - -}; - - -#endif diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/OscilGen.C b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/OscilGen.C deleted file mode 100644 index 4e6a4dd3..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/OscilGen.C +++ /dev/null @@ -1,1182 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - OscilGen.C - Waveform generator for ADnote - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - -*/ - -#include <stdlib.h> -#include <math.h> -#include <stdio.h> - -#include "OscilGen.h" -#include "../Effects/Distorsion.h" - -REALTYPE *OscilGen::tmpsmps;//this array stores some termporary data and it has SOUND_BUFFER_SIZE elements -FFTFREQS OscilGen::outoscilFFTfreqs; - - -OscilGen::OscilGen(FFTwrapper *fft_,Resonance *res_):Presets(){ - setpresettype("Poscilgen"); - fft=fft_; - res=res_; - newFFTFREQS(&oscilFFTfreqs,OSCIL_SIZE/2); - newFFTFREQS(&basefuncFFTfreqs,OSCIL_SIZE/2); - - randseed=1; - ADvsPAD=false; - - defaults(); -}; - -OscilGen::~OscilGen(){ - deleteFFTFREQS(&basefuncFFTfreqs); - deleteFFTFREQS(&oscilFFTfreqs); -}; - - -void OscilGen::defaults(){ - - oldbasefunc=0;oldbasepar=64;oldhmagtype=0;oldwaveshapingfunction=0;oldwaveshaping=64; - oldbasefuncmodulation=0;oldharmonicshift=0;oldbasefuncmodulationpar1=0;oldbasefuncmodulationpar2=0;oldbasefuncmodulationpar3=0; - oldmodulation=0;oldmodulationpar1=0;oldmodulationpar2=0;oldmodulationpar3=0; - - for (int i=0;i<MAX_AD_HARMONICS;i++){ - hmag[i]=0.0; - hphase[i]=0.0; - Phmag[i]=64; - Phphase[i]=64; - }; - Phmag[0]=127; - Phmagtype=0; - if (ADvsPAD) Prand=127;//max phase randomness (usefull if the oscil will be imported to a ADsynth from a PADsynth - else Prand=64;//no randomness - - Pcurrentbasefunc=0; - Pbasefuncpar=64; - - Pbasefuncmodulation=0; - Pbasefuncmodulationpar1=64; - Pbasefuncmodulationpar2=64; - Pbasefuncmodulationpar3=32; - - Pmodulation=0; - Pmodulationpar1=64; - Pmodulationpar2=64; - Pmodulationpar3=32; - - Pwaveshapingfunction=0; - Pwaveshaping=64; - Pfiltertype=0; - Pfilterpar1=64; - Pfilterpar2=64; - Pfilterbeforews=0; - Psatype=0; - Psapar=64; - - Pamprandpower=64; - Pamprandtype=0; - - Pharmonicshift=0; - Pharmonicshiftfirst=0; - - Padaptiveharmonics=0; - Padaptiveharmonicspower=100; - Padaptiveharmonicsbasefreq=128; - Padaptiveharmonicspar=50; - - for (int i=0;i<OSCIL_SIZE/2;i++) { - oscilFFTfreqs.s[i]=0.0; - oscilFFTfreqs.c[i]=0.0; - basefuncFFTfreqs.s[i]=0.0; - basefuncFFTfreqs.c[i]=0.0; - }; - oscilprepared=0; - oldfilterpars=0;oldsapars=0; - prepare(); -}; - -void OscilGen::convert2sine(int magtype){ - REALTYPE mag[MAX_AD_HARMONICS],phase[MAX_AD_HARMONICS]; - REALTYPE oscil[OSCIL_SIZE]; - FFTFREQS freqs; - newFFTFREQS(&freqs,OSCIL_SIZE/2); - - get(oscil,-1.0); - FFTwrapper *fft=new FFTwrapper(OSCIL_SIZE); - fft->smps2freqs(oscil,freqs); - delete(fft); - - REALTYPE max=0.0; - - mag[0]=0; - phase[0]=0; - for (int i=0;i<MAX_AD_HARMONICS;i++){ - mag[i]=sqrt(pow(freqs.s[i+1],2)+pow(freqs.c[i+1],2.0)); - phase[i]=atan2(freqs.c[i+1],freqs.s[i+1]); - if (max<mag[i]) max=mag[i]; - }; - if (max<0.00001) max=1.0; - - defaults(); - - for (int i=0;i<MAX_AD_HARMONICS-1;i++){ - REALTYPE newmag=mag[i]/max; - REALTYPE newphase=phase[i]; - - Phmag[i]=(int) ((newmag)*64.0)+64; - - Phphase[i]=64-(int) (64.0*newphase/PI); - if (Phphase[i]>127) Phphase[i]=127; - - if (Phmag[i]==64) Phphase[i]=64; - }; - deleteFFTFREQS(&freqs); - prepare(); -}; - -/* - * Base Functions - START - */ -REALTYPE OscilGen::basefunc_pulse(REALTYPE x,REALTYPE a){ - return((fmod(x,1.0)<a)?-1.0:1.0); -}; - -REALTYPE OscilGen::basefunc_saw(REALTYPE x,REALTYPE a){ - if (a<0.00001) a=0.00001; - else if (a>0.99999) a=0.99999; - x=fmod(x,1); - if (x<a) return(x/a*2.0-1.0); - else return((1.0-x)/(1.0-a)*2.0-1.0); -}; - -REALTYPE OscilGen::basefunc_triangle(REALTYPE x,REALTYPE a){ - x=fmod(x+0.25,1); - a=1-a; - if (a<0.00001) a=0.00001; - if (x<0.5) x=x*4-1.0; - else x=(1.0-x)*4-1.0; - x/=-a; - if (x<-1.0) x=-1.0; - if (x>1.0) x=1.0; - return(x); -}; - -REALTYPE OscilGen::basefunc_power(REALTYPE x,REALTYPE a){ - x=fmod(x,1); - if (a<0.00001) a=0.00001; - else if (a>0.99999) a=0.99999; - return(pow(x,exp((a-0.5)*10.0))*2.0-1.0); -}; - -REALTYPE OscilGen::basefunc_gauss(REALTYPE x,REALTYPE a){ - x=fmod(x,1)*2.0-1.0; - if (a<0.00001) a=0.00001; - return(exp(-x*x*(exp(a*8)+5.0))*2.0-1.0); -}; - -REALTYPE OscilGen::basefunc_diode(REALTYPE x,REALTYPE a){ - if (a<0.00001) a=0.00001; - else if (a>0.99999) a=0.99999; - a=a*2.0-1.0; - x=cos((x+0.5)*2.0*PI)-a; - if (x<0.0) x=0.0; - return(x/(1.0-a)*2-1.0); -}; - -REALTYPE OscilGen::basefunc_abssine(REALTYPE x,REALTYPE a){ - x=fmod(x,1); - if (a<0.00001) a=0.00001; - else if (a>0.99999) a=0.99999; - return(sin(pow(x,exp((a-0.5)*5.0))*PI)*2.0-1.0); -}; - -REALTYPE OscilGen::basefunc_pulsesine(REALTYPE x,REALTYPE a){ - if (a<0.00001) a=0.00001; - x=(fmod(x,1)-0.5)*exp((a-0.5)*log(128)); - if (x<-0.5) x=-0.5; - else if (x>0.5) x=0.5; - x=sin(x*PI*2.0); - return(x); -}; - -REALTYPE OscilGen::basefunc_stretchsine(REALTYPE x,REALTYPE a){ - x=fmod(x+0.5,1)*2.0-1.0; - a=(a-0.5)*4;if (a>0.0) a*=2; - a=pow(3.0,a); - REALTYPE b=pow(fabs(x),a); - if (x<0) b=-b; - return(-sin(b*PI)); -}; - -REALTYPE OscilGen::basefunc_chirp(REALTYPE x,REALTYPE a){ - x=fmod(x,1.0)*2.0*PI; - a=(a-0.5)*4;if (a<0.0) a*=2.0; - a=pow(3.0,a); - return(sin(x/2.0)*sin(a*x*x)); -}; - -REALTYPE OscilGen::basefunc_absstretchsine(REALTYPE x,REALTYPE a){ - x=fmod(x+0.5,1)*2.0-1.0; - a=(a-0.5)*9; - a=pow(3.0,a); - REALTYPE b=pow(fabs(x),a); - if (x<0) b=-b; - return(-pow(sin(b*PI),2)); -}; - -REALTYPE OscilGen::basefunc_chebyshev(REALTYPE x,REALTYPE a){ - a=a*a*a*30.0+1.0; - return(cos(acos(x*2.0-1.0)*a)); -}; - -REALTYPE OscilGen::basefunc_sqr(REALTYPE x,REALTYPE a){ - a=a*a*a*a*160.0+0.001; - return(-atan(sin(x*2.0*PI)*a)); -}; -/* - * Base Functions - END - */ - - -/* - * Get the base function - */ -void OscilGen::getbasefunction(REALTYPE *smps){ - int i; - REALTYPE par=(Pbasefuncpar+0.5)/128.0; - if (Pbasefuncpar==64) par=0.5; - - REALTYPE basefuncmodulationpar1=Pbasefuncmodulationpar1/127.0, - basefuncmodulationpar2=Pbasefuncmodulationpar2/127.0, - basefuncmodulationpar3=Pbasefuncmodulationpar3/127.0; - - switch(Pbasefuncmodulation){ - case 1:basefuncmodulationpar1=(pow(2,basefuncmodulationpar1*5.0)-1.0)/10.0; - basefuncmodulationpar3=floor((pow(2,basefuncmodulationpar3*5.0)-1.0)); - if (basefuncmodulationpar3<0.9999) basefuncmodulationpar3=-1.0; - break; - case 2:basefuncmodulationpar1=(pow(2,basefuncmodulationpar1*5.0)-1.0)/10.0; - basefuncmodulationpar3=1.0+floor((pow(2,basefuncmodulationpar3*5.0)-1.0)); - break; - case 3:basefuncmodulationpar1=(pow(2,basefuncmodulationpar1*7.0)-1.0)/10.0; - basefuncmodulationpar3=0.01+(pow(2,basefuncmodulationpar3*16.0)-1.0)/10.0; - break; - }; - -// printf("%.5f %.5f\n",basefuncmodulationpar1,basefuncmodulationpar3); - - for (i=0;i<OSCIL_SIZE;i++) { - REALTYPE t=i*1.0/OSCIL_SIZE; - - switch(Pbasefuncmodulation){ - case 1:t=t*basefuncmodulationpar3+sin((t+basefuncmodulationpar2)*2.0*PI)*basefuncmodulationpar1;//rev - break; - case 2:t=t+sin((t*basefuncmodulationpar3+basefuncmodulationpar2)*2.0*PI)*basefuncmodulationpar1;//sine - break; - case 3:t=t+pow((1.0-cos((t+basefuncmodulationpar2)*2.0*PI))*0.5,basefuncmodulationpar3)*basefuncmodulationpar1;//power - break; - }; - - t=t-floor(t); - - switch (Pcurrentbasefunc){ - case 1:smps[i]=basefunc_triangle(t,par); - break; - case 2:smps[i]=basefunc_pulse(t,par); - break; - case 3:smps[i]=basefunc_saw(t,par); - break; - case 4:smps[i]=basefunc_power(t,par); - break; - case 5:smps[i]=basefunc_gauss(t,par); - break; - case 6:smps[i]=basefunc_diode(t,par); - break; - case 7:smps[i]=basefunc_abssine(t,par); - break; - case 8:smps[i]=basefunc_pulsesine(t,par); - break; - case 9:smps[i]=basefunc_stretchsine(t,par); - break; - case 10:smps[i]=basefunc_chirp(t,par); - break; - case 11:smps[i]=basefunc_absstretchsine(t,par); - break; - case 12:smps[i]=basefunc_chebyshev(t,par); - break; - case 13:smps[i]=basefunc_sqr(t,par); - break; - default:smps[i]=-sin(2.0*PI*i/OSCIL_SIZE); - }; - }; -}; - -/* - * Filter the oscillator - */ -void OscilGen::oscilfilter(){ - if (Pfiltertype==0) return; - REALTYPE par=1.0-Pfilterpar1/128.0; - REALTYPE par2=Pfilterpar2/127.0; - REALTYPE max=0.0,tmp=0.0,p2,x; - for (int i=1;i<OSCIL_SIZE/2;i++){ - REALTYPE gain=1.0; - switch(Pfiltertype){ - case 1: gain=pow(1.0-par*par*par*0.99,i);//lp - tmp=par2*par2*par2*par2*0.5+0.0001; - if (gain<tmp) gain=pow(gain,10.0)/pow(tmp,9.0); - break; - case 2: gain=1.0-pow(1.0-par*par,i+1);//hp1 - gain=pow(gain,par2*2.0+0.1); - break; - case 3: if (par<0.2) par=par*0.25+0.15; - gain=1.0-pow(1.0-par*par*0.999+0.001,i*0.05*i+1.0);//hp1b - tmp=pow(5.0,par2*2.0); - gain=pow(gain,tmp); - break; - case 4: gain=i+1-pow(2,(1.0-par)*7.5);//bp1 - gain=1.0/(1.0+gain*gain/(i+1.0)); - tmp=pow(5.0,par2*2.0); - gain=pow(gain,tmp); - if (gain<1e-5) gain=1e-5; - break; - case 5: gain=i+1-pow(2,(1.0-par)*7.5);//bs1 - gain=pow(atan(gain/(i/10.0+1))/1.57,6); - gain=pow(gain,par2*par2*3.9+0.1); - break; - case 6: tmp=pow(par2,0.33); - gain=(i+1>pow(2,(1.0-par)*10)?0.0:1.0)*par2+(1.0-par2);//lp2 - break; - case 7: tmp=pow(par2,0.33); - //tmp=1.0-(1.0-par2)*(1.0-par2); - gain=(i+1>pow(2,(1.0-par)*7)?1.0:0.0)*par2+(1.0-par2);//hp2 - if (Pfilterpar1==0) gain=1.0; - break; - case 8: tmp=pow(par2,0.33); - //tmp=1.0-(1.0-par2)*(1.0-par2); - gain=(fabs(pow(2,(1.0-par)*7)-i)>i/2+1?0.0:1.0)*par2+(1.0-par2);//bp2 - break; - case 9: tmp=pow(par2,0.33); - gain=(fabs(pow(2,(1.0-par)*7)-i)<i/2+1?0.0:1.0)*par2+(1.0-par2);//bs2 - break; - case 10:tmp=pow(5.0,par2*2.0-1.0); - tmp=pow(i/32.0,tmp)*32.0; - if (Pfilterpar2==64) tmp=i; - gain=cos(par*par*PI/2.0*tmp);//cos - gain*=gain; - break; - case 11:tmp=pow(5.0,par2*2.0-1.0); - tmp=pow(i/32.0,tmp)*32.0; - if (Pfilterpar2==64) tmp=i; - gain=sin(par*par*PI/2.0*tmp);//sin - gain*=gain; - break; - case 12:p2=1.0-par+0.2; - x=i/(64.0*p2*p2); - if (x<0.0) x=0.0; - else if (x>1.0) x=1.0; - tmp=pow(1.0-par2,2.0); - gain=cos(x*PI)*(1.0-tmp)+1.01+tmp;//low shelf - break; - case 13:tmp=(int) (pow(2.0,(1.0-par)*7.2)); - gain=1.0; - if (i==(int) (tmp)) gain=pow(2.0,par2*par2*8.0); - break; - }; - - - oscilFFTfreqs.s[i]*=gain; - oscilFFTfreqs.c[i]*=gain; - REALTYPE tmp=oscilFFTfreqs.s[i]*oscilFFTfreqs.s[i]+ - oscilFFTfreqs.c[i]*oscilFFTfreqs.c[i]; - if (max<tmp) max=tmp; - }; - - max=sqrt(max); - if (max<1e-10) max=1.0; - REALTYPE imax=1.0/max; - for (int i=1;i<OSCIL_SIZE/2;i++) { - oscilFFTfreqs.s[i]*=imax; - oscilFFTfreqs.c[i]*=imax; - }; -}; - -/* - * Change the base function - */ -void OscilGen::changebasefunction(){ - if (Pcurrentbasefunc!=0) { - getbasefunction(tmpsmps); - fft->smps2freqs(tmpsmps,basefuncFFTfreqs); - basefuncFFTfreqs.c[0]=0.0; - } else { - for (int i=0;i<OSCIL_SIZE/2;i++){ - basefuncFFTfreqs.s[i]=0.0; - basefuncFFTfreqs.c[i]=0.0; - }; - //in this case basefuncFFTfreqs_ are not used - } - oscilprepared=0; - oldbasefunc=Pcurrentbasefunc; - oldbasepar=Pbasefuncpar; - oldbasefuncmodulation=Pbasefuncmodulation; - oldbasefuncmodulationpar1=Pbasefuncmodulationpar1; - oldbasefuncmodulationpar2=Pbasefuncmodulationpar2; - oldbasefuncmodulationpar3=Pbasefuncmodulationpar3; -}; - -/* - * Waveshape - */ -void OscilGen::waveshape(){ - int i; - - oldwaveshapingfunction=Pwaveshapingfunction; - oldwaveshaping=Pwaveshaping; - if (Pwaveshapingfunction==0) return; - - oscilFFTfreqs.c[0]=0.0;//remove the DC - //reduce the amplitude of the freqs near the nyquist - for (i=1;i<OSCIL_SIZE/8;i++) { - REALTYPE tmp=i/(OSCIL_SIZE/8.0); - oscilFFTfreqs.s[OSCIL_SIZE/2-i]*=tmp; - oscilFFTfreqs.c[OSCIL_SIZE/2-i]*=tmp; - }; - fft->freqs2smps(oscilFFTfreqs,tmpsmps); - - //Normalize - REALTYPE max=0.0; - for (i=0;i<OSCIL_SIZE;i++) - if (max<fabs(tmpsmps[i])) max=fabs(tmpsmps[i]); - if (max<0.00001) max=1.0; - max=1.0/max;for (i=0;i<OSCIL_SIZE;i++) tmpsmps[i]*=max; - - //Do the waveshaping - waveshapesmps(OSCIL_SIZE,tmpsmps,Pwaveshapingfunction,Pwaveshaping); - - fft->smps2freqs(tmpsmps,oscilFFTfreqs);//perform FFT -}; - - -/* - * Do the Frequency Modulation of the Oscil - */ -void OscilGen::modulation(){ - int i; - - oldmodulation=Pmodulation; - oldmodulationpar1=Pmodulationpar1; - oldmodulationpar2=Pmodulationpar2; - oldmodulationpar3=Pmodulationpar3; - if (Pmodulation==0) return; - - - REALTYPE modulationpar1=Pmodulationpar1/127.0, - modulationpar2=0.5-Pmodulationpar2/127.0, - modulationpar3=Pmodulationpar3/127.0; - - switch(Pmodulation){ - case 1:modulationpar1=(pow(2,modulationpar1*7.0)-1.0)/100.0; - modulationpar3=floor((pow(2,modulationpar3*5.0)-1.0)); - if (modulationpar3<0.9999) modulationpar3=-1.0; - break; - case 2:modulationpar1=(pow(2,modulationpar1*7.0)-1.0)/100.0; - modulationpar3=1.0+floor((pow(2,modulationpar3*5.0)-1.0)); - break; - case 3:modulationpar1=(pow(2,modulationpar1*9.0)-1.0)/100.0; - modulationpar3=0.01+(pow(2,modulationpar3*16.0)-1.0)/10.0; - break; - }; - - oscilFFTfreqs.c[0]=0.0;//remove the DC - //reduce the amplitude of the freqs near the nyquist - for (i=1;i<OSCIL_SIZE/8;i++) { - REALTYPE tmp=i/(OSCIL_SIZE/8.0); - oscilFFTfreqs.s[OSCIL_SIZE/2-i]*=tmp; - oscilFFTfreqs.c[OSCIL_SIZE/2-i]*=tmp; - }; - fft->freqs2smps(oscilFFTfreqs,tmpsmps); - int extra_points=2; - REALTYPE *in=new REALTYPE[OSCIL_SIZE+extra_points]; - - //Normalize - REALTYPE max=0.0; - for (i=0;i<OSCIL_SIZE;i++) if (max<fabs(tmpsmps[i])) max=fabs(tmpsmps[i]); - if (max<0.00001) max=1.0; - max=1.0/max;for (i=0;i<OSCIL_SIZE;i++) in[i]=tmpsmps[i]*max; - for (i=0;i<extra_points;i++) in[i+OSCIL_SIZE]=tmpsmps[i]*max; - - //Do the modulation - for (i=0;i<OSCIL_SIZE;i++) { - REALTYPE t=i*1.0/OSCIL_SIZE; - - switch(Pmodulation){ - case 1:t=t*modulationpar3+sin((t+modulationpar2)*2.0*PI)*modulationpar1;//rev - break; - case 2:t=t+sin((t*modulationpar3+modulationpar2)*2.0*PI)*modulationpar1;//sine - break; - case 3:t=t+pow((1.0-cos((t+modulationpar2)*2.0*PI))*0.5,modulationpar3)*modulationpar1;//power - break; - }; - - t=(t-floor(t))*OSCIL_SIZE; - - int poshi=(int) t; - REALTYPE poslo=t-floor(t); - - tmpsmps[i]=in[poshi]*(1.0-poslo)+in[poshi+1]*poslo; - }; - - delete(in); - fft->smps2freqs(tmpsmps,oscilFFTfreqs);//perform FFT -}; - - - -/* - * Adjust the spectrum - */ -void OscilGen::spectrumadjust(){ - if (Psatype==0) return; - REALTYPE par=Psapar/127.0; - switch(Psatype){ - case 1: par=1.0-par*2.0; - if (par>=0.0) par=pow(5.0,par); - else par=pow(8.0,par); - break; - case 2: par=pow(10.0,(1.0-par)*3.0)*0.25; - break; - case 3: par=pow(10.0,(1.0-par)*3.0)*0.25; - break; - }; - - - REALTYPE max=0.0; - for (int i=0;i<OSCIL_SIZE/2;i++){ - REALTYPE tmp=pow(oscilFFTfreqs.c[i],2)+pow(oscilFFTfreqs.s[i],2.0); - if (max<tmp) max=tmp; - }; - max=sqrt(max)/OSCIL_SIZE*2.0; - if (max<1e-8) max=1.0; - - - for (int i=0;i<OSCIL_SIZE/2;i++){ - REALTYPE mag=sqrt(pow(oscilFFTfreqs.s[i],2)+pow(oscilFFTfreqs.c[i],2.0))/max; - REALTYPE phase=atan2(oscilFFTfreqs.s[i],oscilFFTfreqs.c[i]); - - switch (Psatype){ - case 1: mag=pow(mag,par); - break; - case 2: if (mag<par) mag=0.0; - break; - case 3: mag/=par; - if (mag>1.0) mag=1.0; - break; - }; - oscilFFTfreqs.c[i]=mag*cos(phase); - oscilFFTfreqs.s[i]=mag*sin(phase); - }; - -}; - -void OscilGen::shiftharmonics(){ - if (Pharmonicshift==0) return; - - REALTYPE hc,hs; - int harmonicshift=-Pharmonicshift; - - if (harmonicshift>0){ - for (int i=OSCIL_SIZE/2-2;i>=0;i--){ - int oldh=i-harmonicshift; - if (oldh<0){ - hc=0.0; - hs=0.0; - } else { - hc=oscilFFTfreqs.c[oldh+1]; - hs=oscilFFTfreqs.s[oldh+1]; - }; - oscilFFTfreqs.c[i+1]=hc; - oscilFFTfreqs.s[i+1]=hs; - }; - } else { - for (int i=0;i<OSCIL_SIZE/2-1;i++){ - int oldh=i+abs(harmonicshift); - if (oldh>=(OSCIL_SIZE/2-1)){ - hc=0.0; - hs=0.0; - } else { - hc=oscilFFTfreqs.c[oldh+1]; - hs=oscilFFTfreqs.s[oldh+1]; - if (fabs(hc)<0.000001) hc=0.0; - if (fabs(hs)<0.000001) hs=0.0; - }; - - oscilFFTfreqs.c[i+1]=hc; - oscilFFTfreqs.s[i+1]=hs; - }; - }; - - oscilFFTfreqs.c[0]=0.0; -}; - -/* - * Prepare the Oscillator - */ -void OscilGen::prepare(){ - int i,j,k; - REALTYPE a,b,c,d,hmagnew; - - if ((oldbasepar!=Pbasefuncpar)||(oldbasefunc!=Pcurrentbasefunc)|| - (oldbasefuncmodulation!=Pbasefuncmodulation)|| - (oldbasefuncmodulationpar1!=Pbasefuncmodulationpar1)|| - (oldbasefuncmodulationpar2!=Pbasefuncmodulationpar2)|| - (oldbasefuncmodulationpar3!=Pbasefuncmodulationpar3)) - changebasefunction(); - - for (i=0;i<MAX_AD_HARMONICS;i++) hphase[i]=(Phphase[i]-64.0)/64.0*PI/(i+1); - - for (i=0;i<MAX_AD_HARMONICS;i++){ - hmagnew=1.0-fabs(Phmag[i]/64.0-1.0); - switch(Phmagtype){ - case 1:hmag[i]=exp(hmagnew*log(0.01)); break; - case 2:hmag[i]=exp(hmagnew*log(0.001));break; - case 3:hmag[i]=exp(hmagnew*log(0.0001));break; - case 4:hmag[i]=exp(hmagnew*log(0.00001));break; - default:hmag[i]=1.0-hmagnew; - break; - }; - - if (Phmag[i]<64) hmag[i]=-hmag[i]; - }; - - //remove the harmonics where Phmag[i]==64 - for (i=0;i<MAX_AD_HARMONICS;i++) if (Phmag[i]==64) hmag[i]=0.0; - - - for (i=0;i<OSCIL_SIZE/2;i++) { - oscilFFTfreqs.c[i]=0.0; - oscilFFTfreqs.s[i]=0.0; - }; - if (Pcurrentbasefunc==0) {//the sine case - for (i=0;i<MAX_AD_HARMONICS;i++){ - oscilFFTfreqs.c[i+1]=-hmag[i]*sin(hphase[i]*(i+1))/2.0; - oscilFFTfreqs.s[i+1]=hmag[i]*cos(hphase[i]*(i+1))/2.0; - }; - } else { - for (j=0;j<MAX_AD_HARMONICS;j++){ - if (Phmag[j]==64) continue; - for (i=1;i<OSCIL_SIZE/2;i++){ - k=i*(j+1);if (k>=OSCIL_SIZE/2) break; - a=basefuncFFTfreqs.c[i]; - b=basefuncFFTfreqs.s[i]; - c=hmag[j]*cos(hphase[j]*k); - d=hmag[j]*sin(hphase[j]*k); - oscilFFTfreqs.c[k]+=a*c-b*d; - oscilFFTfreqs.s[k]+=a*d+b*c; - }; - }; - - }; - - if (Pharmonicshiftfirst!=0) shiftharmonics(); - - - - if (Pfilterbeforews==0){ - waveshape(); - oscilfilter(); - } else { - oscilfilter(); - waveshape(); - }; - - modulation(); - spectrumadjust(); - if (Pharmonicshiftfirst==0) shiftharmonics(); - - oscilFFTfreqs.c[0]=0.0; - - oldhmagtype=Phmagtype; - oldharmonicshift=Pharmonicshift+Pharmonicshiftfirst*256; - - oscilprepared=1; -}; - -void OscilGen::adaptiveharmonic(FFTFREQS f,REALTYPE freq){ - if ((Padaptiveharmonics==0)/*||(freq<1.0)*/) return; - if (freq<1.0) freq=440.0; - - FFTFREQS inf; - newFFTFREQS(&inf,OSCIL_SIZE/2); - for (int i=0;i<OSCIL_SIZE/2;i++) { - inf.s[i]=f.s[i]; - inf.c[i]=f.c[i]; - f.s[i]=0.0; - f.c[i]=0.0; - }; - inf.c[0]=0.0;inf.s[0]=0.0; - - REALTYPE hc=0.0,hs=0.0; - REALTYPE basefreq=30.0*pow(10.0,Padaptiveharmonicsbasefreq/128.0); - REALTYPE power=(Padaptiveharmonicspower+1.0)/101.0; - - REALTYPE rap=freq/basefreq; - - rap=pow(rap,power); - - bool down=false; - if (rap>1.0) { - rap=1.0/rap; - down=true; - }; - - for (int i=0;i<OSCIL_SIZE/2-2;i++){ - REALTYPE h=i*rap; - int high=(int)(i*rap); - REALTYPE low=fmod(h,1.0); - - if (high>=(OSCIL_SIZE/2-2)){ - break; - } else { - if (down){ - f.c[high]+=inf.c[i]*(1.0-low); - f.s[high]+=inf.s[i]*(1.0-low); - f.c[high+1]+=inf.c[i]*low; - f.s[high+1]+=inf.s[i]*low; - } else { - hc=inf.c[high]*(1.0-low)+inf.c[high+1]*low; - hs=inf.s[high]*(1.0-low)+inf.s[high+1]*low; - }; - if (fabs(hc)<0.000001) hc=0.0; - if (fabs(hs)<0.000001) hs=0.0; - }; - - if (!down){ - if (i==0) {//corect the aplitude of the first harmonic - hc*=rap; - hs*=rap; - }; - f.c[i]=hc; - f.s[i]=hs; - }; - }; - - f.c[1]+=f.c[0];f.s[1]+=f.s[0]; - f.c[0]=0.0;f.s[0]=0.0; - deleteFFTFREQS(&inf); -}; - -void OscilGen::adaptiveharmonicpostprocess(REALTYPE *f,int size){ - if (Padaptiveharmonics<=1) return; - REALTYPE *inf=new REALTYPE[size]; - REALTYPE par=Padaptiveharmonicspar*0.01; - par=1.0-pow((1.0-par),1.5); - - for (int i=0;i<size;i++) { - inf[i]=f[i]*par; - f[i]=f[i]*(1.0-par); - }; - - - if (Padaptiveharmonics==2){//2n+1 - for (int i=0;i<size;i++) if ((i%2)==0) f[i]+=inf[i];//i=0 pt prima armonica,etc. - } else{//celelalte moduri - int nh=(Padaptiveharmonics-3)/2+2; - int sub_vs_add=(Padaptiveharmonics-3)%2; - if (sub_vs_add==0){ - for (int i=0;i<size;i++) { - if (((i+1)%nh)==0){ - f[i]+=inf[i]; - }; - }; - } else { - for (int i=0;i<size/nh-1;i++) { - f[(i+1)*nh-1]+=inf[i]; - }; - }; - }; - - delete(inf); -}; - - - -/* - * Get the oscillator function - */ -short int OscilGen::get(REALTYPE *smps,REALTYPE freqHz){ - return(this->get(smps,freqHz,0)); -}; - -void OscilGen::newrandseed(unsigned int randseed){ - this->randseed=randseed; -}; - -/* - * Get the oscillator function - */ -short int OscilGen::get(REALTYPE *smps,REALTYPE freqHz,int resonance){ - int i; - int nyquist,outpos; - - if ((oldbasepar!=Pbasefuncpar)||(oldbasefunc!=Pcurrentbasefunc)||(oldhmagtype!=Phmagtype) - ||(oldwaveshaping!=Pwaveshaping)||(oldwaveshapingfunction!=Pwaveshapingfunction)) oscilprepared=0; - if (oldfilterpars!=Pfiltertype*256+Pfilterpar1+Pfilterpar2*65536+Pfilterbeforews*16777216){ - oscilprepared=0; - oldfilterpars=Pfiltertype*256+Pfilterpar1+Pfilterpar2*65536+Pfilterbeforews*16777216; - }; - if (oldsapars!=Psatype*256+Psapar){ - oscilprepared=0; - oldsapars=Psatype*256+Psapar; - }; - - if ((oldbasefuncmodulation!=Pbasefuncmodulation)|| - (oldbasefuncmodulationpar1!=Pbasefuncmodulationpar1)|| - (oldbasefuncmodulationpar2!=Pbasefuncmodulationpar2)|| - (oldbasefuncmodulationpar3!=Pbasefuncmodulationpar3)) - oscilprepared=0; - - if ((oldmodulation!=Pmodulation)|| - (oldmodulationpar1!=Pmodulationpar1)|| - (oldmodulationpar2!=Pmodulationpar2)|| - (oldmodulationpar3!=Pmodulationpar3)) - oscilprepared=0; - - if (oldharmonicshift!=Pharmonicshift+Pharmonicshiftfirst*256) oscilprepared=0; - - if (oscilprepared!=1) prepare(); - - outpos=(int)((RND*2.0-1.0)*(REALTYPE) OSCIL_SIZE*(Prand-64.0)/64.0); - outpos=(outpos+2*OSCIL_SIZE) % OSCIL_SIZE; - - - for (i=0;i<OSCIL_SIZE/2;i++){ - outoscilFFTfreqs.c[i]=0.0; - outoscilFFTfreqs.s[i]=0.0; - }; - - nyquist=(int)(0.5*SAMPLE_RATE/fabs(freqHz))+2; - if (ADvsPAD) nyquist=(int)(OSCIL_SIZE/2); - if (nyquist>OSCIL_SIZE/2) nyquist=OSCIL_SIZE/2; - - - int realnyquist=nyquist; - - if (Padaptiveharmonics!=0) nyquist=OSCIL_SIZE/2; - for (i=1;i<nyquist-1;i++) { - outoscilFFTfreqs.c[i]=oscilFFTfreqs.c[i]; - outoscilFFTfreqs.s[i]=oscilFFTfreqs.s[i]; - }; - - adaptiveharmonic(outoscilFFTfreqs,freqHz); - adaptiveharmonicpostprocess(&outoscilFFTfreqs.c[1],OSCIL_SIZE/2-1); - adaptiveharmonicpostprocess(&outoscilFFTfreqs.s[1],OSCIL_SIZE/2-1); - - nyquist=realnyquist; - if (Padaptiveharmonics){//do the antialiasing in the case of adaptive harmonics - for (i=nyquist;i<OSCIL_SIZE/2;i++) { - outoscilFFTfreqs.s[i]=0; - outoscilFFTfreqs.c[i]=0; - }; - }; - - // Randomness (each harmonic), the block type is computed - // in ADnote by setting start position according to this setting - if ((Prand>64)&&(freqHz>=0.0)&&(!ADvsPAD)){ - REALTYPE rnd,angle,a,b,c,d; - rnd=PI*pow((Prand-64.0)/64.0,2.0); - for (i=1;i<nyquist-1;i++){//to Nyquist only for AntiAliasing - angle=rnd*i*RND; - a=outoscilFFTfreqs.c[i]; - b=outoscilFFTfreqs.s[i]; - c=cos(angle); - d=sin(angle); - outoscilFFTfreqs.c[i]=a*c-b*d; - outoscilFFTfreqs.s[i]=a*d+b*c; - }; - }; - - //Harmonic Amplitude Randomness - if ((freqHz>0.1)&&(!ADvsPAD)) { - unsigned int realrnd=rand(); - srand(randseed); - REALTYPE power=Pamprandpower/127.0; - REALTYPE normalize=1.0/(1.2-power); - switch (Pamprandtype){ - case 1: power=power*2.0-0.5; - power=pow(15.0,power); - for (i=1;i<nyquist-1;i++){ - REALTYPE amp=pow(RND,power)*normalize; - outoscilFFTfreqs.c[i]*=amp; - outoscilFFTfreqs.s[i]*=amp; - }; - break; - case 2: power=power*2.0-0.5; - power=pow(15.0,power)*2.0; - REALTYPE rndfreq=2*PI*RND; - for (i=1;i<nyquist-1;i++){ - REALTYPE amp=pow(fabs(sin(i*rndfreq)),power)*normalize; - outoscilFFTfreqs.c[i]*=amp; - outoscilFFTfreqs.s[i]*=amp; - }; - break; - }; - srand(realrnd+1); - }; - - if ((freqHz>0.1)&&(resonance!=0)) res->applyres(nyquist-1,outoscilFFTfreqs,freqHz); - - //Full RMS normalize - REALTYPE sum=0; - for (int j=1;j<OSCIL_SIZE/2;j++) { - REALTYPE term=outoscilFFTfreqs.c[j]*outoscilFFTfreqs.c[j] - +outoscilFFTfreqs.s[j]*outoscilFFTfreqs.s[j]; - sum+=term; - }; - if (sum<0.000001) sum=1.0; - sum=1.0/sqrt(sum); - for (int j=1;j<OSCIL_SIZE/2;j++) { - outoscilFFTfreqs.c[j]*=sum; - outoscilFFTfreqs.s[j]*=sum; - }; - - - if ((ADvsPAD)&&(freqHz>0.1)){//in this case the smps will contain the freqs - for (i=1;i<OSCIL_SIZE/2;i++) smps[i-1]=sqrt(outoscilFFTfreqs.c[i]*outoscilFFTfreqs.c[i] - +outoscilFFTfreqs.s[i]*outoscilFFTfreqs.s[i]); - } else { - fft->freqs2smps(outoscilFFTfreqs,smps); - for (i=0;i<OSCIL_SIZE;i++) smps[i]*=0.25;//correct the amplitude - }; - - if (Prand<64) return(outpos); - else return(0); -}; - - -/* - * Get the spectrum of the oscillator for the UI - */ -void OscilGen::getspectrum(int n, REALTYPE *spc,int what){ - if (n>OSCIL_SIZE/2) n=OSCIL_SIZE/2; - - for (int i=1;i<n;i++){ - if (what==0){ - spc[i-1]=sqrt(oscilFFTfreqs.c[i]*oscilFFTfreqs.c[i] - +oscilFFTfreqs.s[i]*oscilFFTfreqs.s[i]); - } else { - if (Pcurrentbasefunc==0) spc[i-1]=((i==1)?(1.0):(0.0)); - else spc[i-1]=sqrt(basefuncFFTfreqs.c[i]*basefuncFFTfreqs.c[i]+ - basefuncFFTfreqs.s[i]*basefuncFFTfreqs.s[i]); - }; - }; - - if (what==0) { - for (int i=0;i<n;i++) outoscilFFTfreqs.s[i]=outoscilFFTfreqs.c[i]=spc[i+1]; - for (int i=n;i<OSCIL_SIZE/2;i++) outoscilFFTfreqs.s[i]=outoscilFFTfreqs.c[i]=0.0; - adaptiveharmonic(outoscilFFTfreqs,0.0); - for (int i=1;i<n;i++) spc[i-1]=outoscilFFTfreqs.s[i]; - adaptiveharmonicpostprocess(spc,n-1); - }; -}; - - -/* - * Convert the oscillator as base function - */ -void OscilGen::useasbase(){ - int i; - - for (i=0;i<OSCIL_SIZE/2;i++) { - basefuncFFTfreqs.c[i]=oscilFFTfreqs.c[i]; - basefuncFFTfreqs.s[i]=oscilFFTfreqs.s[i]; - }; - - oldbasefunc=Pcurrentbasefunc=127; - - prepare(); -}; - - -/* - * Get the base function for UI - */ -void OscilGen::getcurrentbasefunction(REALTYPE *smps){ - if (Pcurrentbasefunc!=0) { - fft->freqs2smps(basefuncFFTfreqs,smps); - } else getbasefunction(smps);//the sine case -}; - - -void OscilGen::add2XML(XMLwrapper *xml){ - xml->addpar("harmonic_mag_type",Phmagtype); - - xml->addpar("base_function",Pcurrentbasefunc); - xml->addpar("base_function_par",Pbasefuncpar); - xml->addpar("base_function_modulation",Pbasefuncmodulation); - xml->addpar("base_function_modulation_par1",Pbasefuncmodulationpar1); - xml->addpar("base_function_modulation_par2",Pbasefuncmodulationpar2); - xml->addpar("base_function_modulation_par3",Pbasefuncmodulationpar3); - - xml->addpar("modulation",Pmodulation); - xml->addpar("modulation_par1",Pmodulationpar1); - xml->addpar("modulation_par2",Pmodulationpar2); - xml->addpar("modulation_par3",Pmodulationpar3); - - xml->addpar("wave_shaping",Pwaveshaping); - xml->addpar("wave_shaping_function",Pwaveshapingfunction); - - xml->addpar("filter_type",Pfiltertype); - xml->addpar("filter_par1",Pfilterpar1); - xml->addpar("filter_par2",Pfilterpar2); - xml->addpar("filter_before_wave_shaping",Pfilterbeforews); - - xml->addpar("spectrum_adjust_type",Psatype); - xml->addpar("spectrum_adjust_par",Psapar); - - xml->addpar("rand",Prand); - xml->addpar("amp_rand_type",Pamprandtype); - xml->addpar("amp_rand_power",Pamprandpower); - - xml->addpar("harmonic_shift",Pharmonicshift); - xml->addparbool("harmonic_shift_first",Pharmonicshiftfirst); - - xml->addpar("adaptive_harmonics",Padaptiveharmonics); - xml->addpar("adaptive_harmonics_base_frequency",Padaptiveharmonicsbasefreq); - xml->addpar("adaptive_harmonics_power",Padaptiveharmonicspower); - - xml->beginbranch("HARMONICS"); - for (int n=0;n<MAX_AD_HARMONICS;n++){ - if ((Phmag[n]==64)&&(Phphase[n]==64)) continue; - xml->beginbranch("HARMONIC",n+1); - xml->addpar("mag",Phmag[n]); - xml->addpar("phase",Phphase[n]); - xml->endbranch(); - }; - xml->endbranch(); - - if (Pcurrentbasefunc==127){ - REALTYPE max=0.0; - - for (int i=0;i<OSCIL_SIZE/2;i++){ - if (max<fabs(basefuncFFTfreqs.c[i])) max=fabs(basefuncFFTfreqs.c[i]); - if (max<fabs(basefuncFFTfreqs.s[i])) max=fabs(basefuncFFTfreqs.s[i]); - }; - if (max<0.00000001) max=1.0; - - xml->beginbranch("BASE_FUNCTION"); - for (int i=1;i<OSCIL_SIZE/2;i++){ - REALTYPE xc=basefuncFFTfreqs.c[i]/max; - REALTYPE xs=basefuncFFTfreqs.s[i]/max; - if ((fabs(xs)>0.00001)&&(fabs(xs)>0.00001)){ - xml->beginbranch("BF_HARMONIC",i); - xml->addparreal("cos",xc); - xml->addparreal("sin",xs); - xml->endbranch(); - }; - }; - xml->endbranch(); - }; -}; - - -void OscilGen::getfromXML(XMLwrapper *xml){ - - Phmagtype=xml->getpar127("harmonic_mag_type",Phmagtype); - - Pcurrentbasefunc=xml->getpar127("base_function",Pcurrentbasefunc); - Pbasefuncpar=xml->getpar127("base_function_par",Pbasefuncpar); - - Pbasefuncmodulation=xml->getpar127("base_function_modulation",Pbasefuncmodulation); - Pbasefuncmodulationpar1=xml->getpar127("base_function_modulation_par1",Pbasefuncmodulationpar1); - Pbasefuncmodulationpar2=xml->getpar127("base_function_modulation_par2",Pbasefuncmodulationpar2); - Pbasefuncmodulationpar3=xml->getpar127("base_function_modulation_par3",Pbasefuncmodulationpar3); - - Pmodulation=xml->getpar127("modulation",Pmodulation); - Pmodulationpar1=xml->getpar127("modulation_par1",Pmodulationpar1); - Pmodulationpar2=xml->getpar127("modulation_par2",Pmodulationpar2); - Pmodulationpar3=xml->getpar127("modulation_par3",Pmodulationpar3); - - Pwaveshaping=xml->getpar127("wave_shaping",Pwaveshaping); - Pwaveshapingfunction=xml->getpar127("wave_shaping_function",Pwaveshapingfunction); - - Pfiltertype=xml->getpar127("filter_type",Pfiltertype); - Pfilterpar1=xml->getpar127("filter_par1",Pfilterpar1); - Pfilterpar2=xml->getpar127("filter_par2",Pfilterpar2); - Pfilterbeforews=xml->getpar127("filter_before_wave_shaping",Pfilterbeforews); - - Psatype=xml->getpar127("spectrum_adjust_type",Psatype); - Psapar=xml->getpar127("spectrum_adjust_par",Psapar); - - Prand=xml->getpar127("rand",Prand); - Pamprandtype=xml->getpar127("amp_rand_type",Pamprandtype); - Pamprandpower=xml->getpar127("amp_rand_power",Pamprandpower); - - Pharmonicshift=xml->getpar("harmonic_shift",Pharmonicshift,-64,64); - Pharmonicshiftfirst=xml->getparbool("harmonic_shift_first",Pharmonicshiftfirst); - - Padaptiveharmonics=xml->getpar("adaptive_harmonics",Padaptiveharmonics,0,127); - Padaptiveharmonicsbasefreq=xml->getpar("adaptive_harmonics_base_frequency",Padaptiveharmonicsbasefreq,0,255); - Padaptiveharmonicspower=xml->getpar("adaptive_harmonics_power",Padaptiveharmonicspower,0,200); - - - if (xml->enterbranch("HARMONICS")){ - Phmag[0]=64;Phphase[0]=64; - for (int n=0;n<MAX_AD_HARMONICS;n++){ - if (xml->enterbranch("HARMONIC",n+1)==0) continue; - Phmag[n]=xml->getpar127("mag",64); - Phphase[n]=xml->getpar127("phase",64); - xml->exitbranch(); - }; - xml->exitbranch(); - }; - - if (Pcurrentbasefunc!=0) changebasefunction(); - - - if (xml->enterbranch("BASE_FUNCTION")){ - for (int i=1;i<OSCIL_SIZE/2;i++){ - if (xml->enterbranch("BF_HARMONIC",i)){ - basefuncFFTfreqs.c[i]=xml->getparreal("cos",0.0); - basefuncFFTfreqs.s[i]=xml->getparreal("sin",0.0); - xml->exitbranch(); - }; - - - }; - xml->exitbranch(); - - REALTYPE max=0.0; - - basefuncFFTfreqs.c[0]=0.0; - for (int i=0;i<OSCIL_SIZE/2;i++) { - if (max<fabs(basefuncFFTfreqs.c[i])) max=fabs(basefuncFFTfreqs.c[i]); - if (max<fabs(basefuncFFTfreqs.s[i])) max=fabs(basefuncFFTfreqs.s[i]); - }; - if (max<0.00000001) max=1.0; - - for (int i=0;i<OSCIL_SIZE/2;i++) { - if (basefuncFFTfreqs.c[i]) basefuncFFTfreqs.c[i]/=max; - if (basefuncFFTfreqs.s[i]) basefuncFFTfreqs.s[i]/=max; - }; - }; -}; - - - diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/OscilGen.h b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/OscilGen.h deleted file mode 100644 index 1d9980a9..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/OscilGen.h +++ /dev/null @@ -1,176 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - OscilGen.h - Waveform generator for ADnote - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - -*/ - -#ifndef OSCIL_GEN_H -#define OSCIL_GEN_H - -#include "../globals.h" -#include "../Misc/XMLwrapper.h" -#include "Resonance.h" -#include "../DSP/FFTwrapper.h" -#include "../Params/Presets.h" - -class OscilGen:public Presets{ - public: - OscilGen(FFTwrapper *fft_,Resonance *res_); - ~OscilGen(); - - //computes the full spectrum of oscil from harmonics,phases and basefunc - void prepare(); - - //do the antialiasing(cut off higher freqs.),apply randomness and do a IFFT - short get(REALTYPE *smps,REALTYPE freqHz);//returns where should I start getting samples, used in block type randomness - short get(REALTYPE *smps,REALTYPE freqHz,int resonance); - //if freqHz is smaller than 0, return the "un-randomized" sample for UI - - void getbasefunction(REALTYPE *smps); - - //called by UI - void getspectrum(int n,REALTYPE *spc,int what);//what=0 pt. oscil,1 pt. basefunc - void getcurrentbasefunction(REALTYPE *smps); - void useasbase();//convert oscil to base function - - void add2XML(XMLwrapper *xml); - void defaults(); - void getfromXML(XMLwrapper *xml); - - void convert2sine(int magtype); - - //Parameters - - /* - The hmag and hphase starts counting from 0, so the first harmonic(1) has the index 0, - 2-nd harmonic has index 1, ..the 128 harminic has index 127 - */ - unsigned char Phmag[MAX_AD_HARMONICS],Phphase[MAX_AD_HARMONICS];//the MIDI parameters for mag. and phases - - - /*The Type of magnitude: - 0 - Linear - 1 - dB scale (-40) - 2 - dB scale (-60) - 3 - dB scale (-80) - 4 - dB scale (-100)*/ - unsigned char Phmagtype; - - unsigned char Pcurrentbasefunc;//The base function used - 0=sin, 1=... - unsigned char Pbasefuncpar;//the parameter of the base function - - unsigned char Pbasefuncmodulation;//what modulation is applied to the basefunc - unsigned char Pbasefuncmodulationpar1,Pbasefuncmodulationpar2,Pbasefuncmodulationpar3;//the parameter of the base function modulation - - /*the Randomness: - 64=no randomness - 63..0 - block type randomness - 0 is maximum - 65..127 - each harmonic randomness - 127 is maximum*/ - unsigned char Prand; - unsigned char Pwaveshaping,Pwaveshapingfunction; - unsigned char Pfiltertype,Pfilterpar1,Pfilterpar2; - unsigned char Pfilterbeforews; - unsigned char Psatype,Psapar;//spectrum adjust - - unsigned char Pamprandpower, Pamprandtype;//amplitude randomness - int Pharmonicshift;//how the harmonics are shifted - int Pharmonicshiftfirst;//if the harmonic shift is done before waveshaping and filter - - unsigned char Padaptiveharmonics;//the adaptive harmonics status (off=0,on=1,etc..) - unsigned char Padaptiveharmonicsbasefreq;//the base frequency of the adaptive harmonic (30..3000Hz) - unsigned char Padaptiveharmonicspower;//the strength of the effect (0=off,100=full) - unsigned char Padaptiveharmonicspar;//the parameters in 2,3,4.. modes of adaptive harmonics - - unsigned char Pmodulation;//what modulation is applied to the oscil - unsigned char Pmodulationpar1,Pmodulationpar2,Pmodulationpar3;//the parameter of the parameters - - - //makes a new random seed for Amplitude Randomness - //this should be called every note on event - void newrandseed(unsigned int randseed); - - bool ADvsPAD;//if it is used by ADsynth or by PADsynth - - static REALTYPE *tmpsmps;//this array stores some termporary data and it has SOUND_BUFFER_SIZE elements - static FFTFREQS outoscilFFTfreqs; - - private: - - REALTYPE hmag[MAX_AD_HARMONICS],hphase[MAX_AD_HARMONICS];//the magnituides and the phases of the sine/nonsine harmonics -// private: - FFTwrapper *fft; - //computes the basefunction and make the FFT; newbasefunc<0 = same basefunc - void changebasefunction(); - //Waveshaping - void waveshape(); - - //Filter the oscillator accotding to Pfiltertype and Pfilterpar - void oscilfilter(); - - //Adjust the spectrum - void spectrumadjust(); - - //Shift the harmonics - void shiftharmonics(); - - //Do the oscil modulation stuff - void modulation(); - - //Do the adaptive harmonic stuff - void adaptiveharmonic(FFTFREQS f,REALTYPE freq); - - //Do the adaptive harmonic postprocessing (2n+1,2xS,2xA,etc..) - //this function is called even for the user interface - //this can be called for the sine and components, and for the spectrum - //(that's why the sine and cosine components should be processed with a separate call) - void adaptiveharmonicpostprocess(REALTYPE *f, int size); - - //Basic/base functions (Functiile De Baza) - REALTYPE basefunc_pulse(REALTYPE x,REALTYPE a); - REALTYPE basefunc_saw(REALTYPE x,REALTYPE a); - REALTYPE basefunc_triangle(REALTYPE x,REALTYPE a); - REALTYPE basefunc_power(REALTYPE x,REALTYPE a); - REALTYPE basefunc_gauss(REALTYPE x,REALTYPE a); - REALTYPE basefunc_diode(REALTYPE x,REALTYPE a); - REALTYPE basefunc_abssine(REALTYPE x,REALTYPE a); - REALTYPE basefunc_pulsesine(REALTYPE x,REALTYPE a); - REALTYPE basefunc_stretchsine(REALTYPE x,REALTYPE a); - REALTYPE basefunc_chirp(REALTYPE x,REALTYPE a); - REALTYPE basefunc_absstretchsine(REALTYPE x,REALTYPE a); - REALTYPE basefunc_chebyshev(REALTYPE x,REALTYPE a); - REALTYPE basefunc_sqr(REALTYPE x,REALTYPE a); - - //Internal Data - unsigned char oldbasefunc,oldbasepar,oldhmagtype,oldwaveshapingfunction,oldwaveshaping; - int oldfilterpars,oldsapars,oldbasefuncmodulation,oldbasefuncmodulationpar1,oldbasefuncmodulationpar2,oldbasefuncmodulationpar3,oldharmonicshift; - int oldmodulation,oldmodulationpar1,oldmodulationpar2,oldmodulationpar3; - - - FFTFREQS basefuncFFTfreqs;//Base Function Frequencies - FFTFREQS oscilFFTfreqs;//Oscillator Frequencies - this is different than the hamonics set-up by the user, it may contains time-domain data if the antialiasing is turned off - int oscilprepared;//1 if the oscil is prepared, 0 if it is not prepared and is need to call ::prepare() before ::get() - - Resonance *res; - - unsigned int randseed; - -}; - - -#endif diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/PADnote.C b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/PADnote.C deleted file mode 100644 index 9ecc8877..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/PADnote.C +++ /dev/null @@ -1,342 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - PADnote.C - The "pad" synthesizer - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -*/ -#include <math.h> -#include "PADnote.h" -#include "../Misc/Config.h" - -PADnote::PADnote(PADnoteParameters *parameters, Controller *ctl_,REALTYPE freq, REALTYPE velocity, int portamento_, int midinote){ - ready=0; - pars=parameters; - portamento=portamento_; - ctl=ctl_; - this->velocity=velocity; - finished_=false; - - - if (pars->Pfixedfreq==0) basefreq=freq; - else { - basefreq=440.0; - int fixedfreqET=pars->PfixedfreqET; - if (fixedfreqET!=0) {//if the frequency varies according the keyboard note - REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0); - if (fixedfreqET<=64) basefreq*=pow(2.0,tmp); - else basefreq*=pow(3.0,tmp); - }; - - }; - - firsttime=true; - released=false; - realfreq=basefreq; - NoteGlobalPar.Detune=getdetune(pars->PDetuneType - ,pars->PCoarseDetune,pars->PDetune); - - - //find out the closest note - REALTYPE logfreq=log(basefreq*pow(2.0,NoteGlobalPar.Detune/1200.0)); - REALTYPE mindist=fabs(logfreq-log(pars->sample[0].basefreq+0.0001)); - nsample=0; - for (int i=1;i<PAD_MAX_SAMPLES;i++){ - if (pars->sample[i].smp==NULL) break; - REALTYPE dist=fabs(logfreq-log(pars->sample[i].basefreq+0.0001)); -// printf("(mindist=%g) %i %g %g\n",mindist,i,dist,pars->sample[i].basefreq); - - if (dist<mindist){ - nsample=i; - mindist=dist; - }; - }; - - int size=pars->sample[nsample].size; - if (size==0) size=1; - - - poshi_l=(int)(RND*(size-1)); - if (pars->PStereo!=0) poshi_r=(poshi_l+size/2)%size; - else poshi_r=poshi_l; - poslo=0.0; - - tmpwave=new REALTYPE [SOUND_BUFFER_SIZE]; - - - - if (pars->PPanning==0) NoteGlobalPar.Panning=RND; - else NoteGlobalPar.Panning=pars->PPanning/128.0; - - NoteGlobalPar.FilterCenterPitch=pars->GlobalFilter->getfreq()+//center freq - pars->PFilterVelocityScale/127.0*6.0* //velocity sensing - (VelF(velocity,pars->PFilterVelocityScaleFunction)-1); - - if (pars->PPunchStrength!=0) { - NoteGlobalPar.Punch.Enabled=1; - NoteGlobalPar.Punch.t=1.0;//start from 1.0 and to 0.0 - NoteGlobalPar.Punch.initialvalue=( (pow(10,1.5*pars->PPunchStrength/127.0)-1.0) - *VelF(velocity,pars->PPunchVelocitySensing) ); - REALTYPE time=pow(10,3.0*pars->PPunchTime/127.0)/10000.0;//0.1 .. 100 ms - REALTYPE stretch=pow(440.0/freq,pars->PPunchStretch/64.0); - NoteGlobalPar.Punch.dt=1.0/(time*SAMPLE_RATE*stretch); - } else NoteGlobalPar.Punch.Enabled=0; - - - - NoteGlobalPar.FreqEnvelope=new Envelope(pars->FreqEnvelope,basefreq); - NoteGlobalPar.FreqLfo=new LFO(pars->FreqLfo,basefreq); - - NoteGlobalPar.AmpEnvelope=new Envelope(pars->AmpEnvelope,basefreq); - NoteGlobalPar.AmpLfo=new LFO(pars->AmpLfo,basefreq); - - NoteGlobalPar.Volume=4.0*pow(0.1,3.0*(1.0-pars->PVolume/96.0))//-60 dB .. 0 dB - *VelF(velocity,pars->PAmpVelocityScaleFunction);//velocity sensing - - NoteGlobalPar.AmpEnvelope->envout_dB();//discard the first envelope output - globaloldamplitude=globalnewamplitude=NoteGlobalPar.Volume*NoteGlobalPar.AmpEnvelope->envout_dB()*NoteGlobalPar.AmpLfo->amplfoout(); - - NoteGlobalPar.GlobalFilterL=new Filter(pars->GlobalFilter); - NoteGlobalPar.GlobalFilterR=new Filter(pars->GlobalFilter); - - NoteGlobalPar.FilterEnvelope=new Envelope(pars->FilterEnvelope,basefreq); - NoteGlobalPar.FilterLfo=new LFO(pars->FilterLfo,basefreq); - NoteGlobalPar.FilterQ=pars->GlobalFilter->getq(); - NoteGlobalPar.FilterFreqTracking=pars->GlobalFilter->getfreqtracking(basefreq); - - ready=1;///sa il pun pe asta doar cand e chiar gata - - if (parameters->sample[nsample].smp==NULL){ - finished_=true; - return; - }; -}; - -PADnote::~PADnote(){ - delete (NoteGlobalPar.FreqEnvelope); - delete (NoteGlobalPar.FreqLfo); - delete (NoteGlobalPar.AmpEnvelope); - delete (NoteGlobalPar.AmpLfo); - delete (NoteGlobalPar.GlobalFilterL); - delete (NoteGlobalPar.GlobalFilterR); - delete (NoteGlobalPar.FilterEnvelope); - delete (NoteGlobalPar.FilterLfo); - delete (tmpwave); -}; - - -inline void PADnote::fadein(REALTYPE *smps){ - int zerocrossings=0; - for (int i=1;i<SOUND_BUFFER_SIZE;i++) - if ((smps[i-1]<0.0) && (smps[i]>0.0)) zerocrossings++;//this is only the possitive crossings - - REALTYPE tmp=(SOUND_BUFFER_SIZE-1.0)/(zerocrossings+1)/3.0; - if (tmp<8.0) tmp=8.0; - - int n; - F2I(tmp,n);//how many samples is the fade-in - if (n>SOUND_BUFFER_SIZE) n=SOUND_BUFFER_SIZE; - for (int i=0;i<n;i++) {//fade-in - REALTYPE tmp=0.5-cos((REALTYPE)i/(REALTYPE) n*PI)*0.5; - smps[i]*=tmp; - }; -}; - - -void PADnote::computecurrentparameters(){ - REALTYPE globalpitch,globalfilterpitch; - globalpitch=0.01*(NoteGlobalPar.FreqEnvelope->envout()+ - NoteGlobalPar.FreqLfo->lfoout()*ctl->modwheel.relmod+NoteGlobalPar.Detune); - globaloldamplitude=globalnewamplitude; - globalnewamplitude=NoteGlobalPar.Volume*NoteGlobalPar.AmpEnvelope->envout_dB()*NoteGlobalPar.AmpLfo->amplfoout(); - - globalfilterpitch=NoteGlobalPar.FilterEnvelope->envout()+NoteGlobalPar.FilterLfo->lfoout() - +NoteGlobalPar.FilterCenterPitch; - - REALTYPE tmpfilterfreq=globalfilterpitch+ctl->filtercutoff.relfreq - +NoteGlobalPar.FilterFreqTracking; - - tmpfilterfreq=NoteGlobalPar.GlobalFilterL->getrealfreq(tmpfilterfreq); - - REALTYPE globalfilterq=NoteGlobalPar.FilterQ*ctl->filterq.relq; - NoteGlobalPar.GlobalFilterL->setfreq_and_q(tmpfilterfreq,globalfilterq); - NoteGlobalPar.GlobalFilterR->setfreq_and_q(tmpfilterfreq,globalfilterq); - - //compute the portamento, if it is used by this note - REALTYPE portamentofreqrap=1.0; - if (portamento!=0){//this voice use portamento - portamentofreqrap=ctl->portamento.freqrap; - if (ctl->portamento.used==0){//the portamento has finished - portamento=0;//this note is no longer "portamented" - }; - }; - - realfreq=basefreq*portamentofreqrap*pow(2.0,globalpitch/12.0)*ctl->pitchwheel.relfreq; -}; - - -int PADnote::Compute_Linear(REALTYPE *outl,REALTYPE *outr,int freqhi,REALTYPE freqlo){ - REALTYPE *smps=pars->sample[nsample].smp; - if (smps==NULL){ - finished_=true; - return(1); - }; - int size=pars->sample[nsample].size; - for (int i=0;i<SOUND_BUFFER_SIZE;i++){ - poshi_l+=freqhi; - poshi_r+=freqhi; - poslo+=freqlo; - if (poslo>=1.0){ - poshi_l+=1; - poshi_r+=1; - poslo-=1.0; - }; - if (poshi_l>=size) poshi_l%=size; - if (poshi_r>=size) poshi_r%=size; - - outl[i]=smps[poshi_l]*(1.0-poslo)+smps[poshi_l+1]*poslo; - outr[i]=smps[poshi_r]*(1.0-poslo)+smps[poshi_r+1]*poslo; - }; - return(1); -}; -int PADnote::Compute_Cubic(REALTYPE *outl,REALTYPE *outr,int freqhi,REALTYPE freqlo){ - REALTYPE *smps=pars->sample[nsample].smp; - if (smps==NULL){ - finished_=true; - return(1); - }; - int size=pars->sample[nsample].size; - REALTYPE xm1,x0,x1,x2,a,b,c; - for (int i=0;i<SOUND_BUFFER_SIZE;i++){ - poshi_l+=freqhi; - poshi_r+=freqhi; - poslo+=freqlo; - if (poslo>=1.0){ - poshi_l+=1; - poshi_r+=1; - poslo-=1.0; - }; - if (poshi_l>=size) poshi_l%=size; - if (poshi_r>=size) poshi_r%=size; - - - //left - xm1=smps[poshi_l]; - x0=smps[poshi_l + 1]; - x1=smps[poshi_l + 2]; - x2=smps[poshi_l + 3]; - a = (3.0 * (x0-x1) - xm1 + x2)*0.5; - b = 2.0*x1 + xm1 - (5.0*x0 + x2)*0.5; - c = (x1 - xm1)*0.5; - outl[i] = (((a * poslo) + b) * poslo + c) * poslo + x0; - //right - xm1=smps[poshi_r]; - x0=smps[poshi_r + 1]; - x1=smps[poshi_r + 2]; - x2=smps[poshi_r + 3]; - a = (3.0 * (x0-x1) - xm1 + x2)*0.5; - b = 2.0*x1 + xm1 - (5.0*x0 + x2)*0.5; - c = (x1 - xm1)*0.5; - outr[i] = (((a * poslo) + b) * poslo + c) * poslo + x0; - }; - return(1); -}; - - -int PADnote::noteout(REALTYPE *outl,REALTYPE *outr){ - computecurrentparameters(); - REALTYPE *smps=pars->sample[nsample].smp; - if (smps==NULL){ - for (int i=0;i<SOUND_BUFFER_SIZE;i++){ - outl[i]=0.0; - outr[i]=0.0; - }; - return(1); - }; - REALTYPE smpfreq=pars->sample[nsample].basefreq; - - - REALTYPE freqrap=realfreq/smpfreq; - int freqhi=(int) (floor(freqrap)); - REALTYPE freqlo=freqrap-floor(freqrap); - - - if (config.cfg.Interpolation) Compute_Cubic(outl,outr,freqhi,freqlo); - else Compute_Linear(outl,outr,freqhi,freqlo); - - - if (firsttime){ - fadein(outl); - fadein(outr); - firsttime=false; - }; - - NoteGlobalPar.GlobalFilterL->filterout(outl); - NoteGlobalPar.GlobalFilterR->filterout(outr); - - //Apply the punch - if (NoteGlobalPar.Punch.Enabled!=0){ - for (int i=0;i<SOUND_BUFFER_SIZE;i++){ - REALTYPE punchamp=NoteGlobalPar.Punch.initialvalue*NoteGlobalPar.Punch.t+1.0; - outl[i]*=punchamp; - outr[i]*=punchamp; - NoteGlobalPar.Punch.t-=NoteGlobalPar.Punch.dt; - if (NoteGlobalPar.Punch.t<0.0) { - NoteGlobalPar.Punch.Enabled=0; - break; - }; - }; - }; - - if (ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude,globalnewamplitude)){ - // Amplitude Interpolation - for (int i=0;i<SOUND_BUFFER_SIZE;i++){ - REALTYPE tmpvol=INTERPOLATE_AMPLITUDE(globaloldamplitude,globalnewamplitude,i,SOUND_BUFFER_SIZE); - outl[i]*=tmpvol*NoteGlobalPar.Panning; - outr[i]*=tmpvol*(1.0-NoteGlobalPar.Panning); - }; - } else { - for (int i=0;i<SOUND_BUFFER_SIZE;i++) { - outl[i]*=globalnewamplitude*NoteGlobalPar.Panning; - outr[i]*=globalnewamplitude*(1.0-NoteGlobalPar.Panning); - }; - }; - - - // Check if the global amplitude is finished. - // If it does, disable the note - if (NoteGlobalPar.AmpEnvelope->finished()!=0) { - for (int i=0;i<SOUND_BUFFER_SIZE;i++) {//fade-out - REALTYPE tmp=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE; - outl[i]*=tmp; - outr[i]*=tmp; - }; - finished_=1; - }; - - return(1); -}; - -int PADnote::finished(){ - return(finished_); -}; - -void PADnote::relasekey(){ - NoteGlobalPar.FreqEnvelope->relasekey(); - NoteGlobalPar.FilterEnvelope->relasekey(); - NoteGlobalPar.AmpEnvelope->relasekey(); -}; - diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/PADnote.h b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/PADnote.h deleted file mode 100644 index 2a99577f..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/PADnote.h +++ /dev/null @@ -1,106 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - PADnote.h - The "pad" synthesizer - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -*/ -#ifndef PAD_NOTE_H -#define PAD_NOTE_H - -#include "../globals.h" -#include "../Params/PADnoteParameters.h" -#include "../Params/Controller.h" -#include "Envelope.h" -#include "LFO.h" -#include "../DSP/Filter.h" -#include "../Params/Controller.h" - -class PADnote{ - public: - PADnote(PADnoteParameters *parameters, Controller *ctl_,REALTYPE freq, REALTYPE velocity, int portamento_, int midinote); - ~PADnote(); - - int noteout(REALTYPE *outl,REALTYPE *outr); - int finished(); - void relasekey(); - - int ready; - - private: - void fadein(REALTYPE *smps); - void computecurrentparameters(); - bool finished_; - PADnoteParameters *pars; - - int poshi_l,poshi_r; - REALTYPE poslo; - - REALTYPE basefreq; - bool firsttime,released; - - int nsample,portamento; - - int Compute_Linear(REALTYPE *outl,REALTYPE *outr,int freqhi,REALTYPE freqlo); - int Compute_Cubic(REALTYPE *outl,REALTYPE *outr,int freqhi,REALTYPE freqlo); - - - struct{ - /****************************************** - * FREQUENCY GLOBAL PARAMETERS * - ******************************************/ - REALTYPE Detune;//cents - - Envelope *FreqEnvelope; - LFO *FreqLfo; - - /******************************************** - * AMPLITUDE GLOBAL PARAMETERS * - ********************************************/ - REALTYPE Volume;// [ 0 .. 1 ] - - REALTYPE Panning;// [ 0 .. 1 ] - - Envelope *AmpEnvelope; - LFO *AmpLfo; - - struct { - int Enabled; - REALTYPE initialvalue,dt,t; - } Punch; - - /****************************************** - * FILTER GLOBAL PARAMETERS * - ******************************************/ - Filter *GlobalFilterL,*GlobalFilterR; - - REALTYPE FilterCenterPitch;//octaves - REALTYPE FilterQ; - REALTYPE FilterFreqTracking; - - Envelope *FilterEnvelope; - - LFO *FilterLfo; - } NoteGlobalPar; - - - REALTYPE globaloldamplitude,globalnewamplitude,velocity,realfreq; - REALTYPE *tmpwave; - Controller *ctl; -}; - - -#endif diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Resonance.C b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Resonance.C deleted file mode 100644 index fb741055..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Resonance.C +++ /dev/null @@ -1,231 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - Resonance.C - Resonance - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -*/ - -#include <math.h> -#include <stdlib.h> -#include "Resonance.h" - - -#include <stdio.h> - -Resonance::Resonance():Presets(){ - setpresettype("Presonance"); - defaults(); -}; - -Resonance::~Resonance(){ -}; - - -void Resonance::defaults(){ - Penabled=0; - PmaxdB=20; - Pcenterfreq=64;//1 kHz - Poctavesfreq=64; - Pprotectthefundamental=0; - ctlcenter=1.0; - ctlbw=1.0; - for (int i=0;i<N_RES_POINTS;i++) Prespoints[i]=64; -}; - -/* - * Set a point of resonance function with a value - */ -void Resonance::setpoint(int n,unsigned char p){ - if ((n<0)||(n>=N_RES_POINTS)) return; - Prespoints[n]=p; -}; - -/* - * Apply the resonance to FFT data - */ -void Resonance::applyres(int n,FFTFREQS fftdata,REALTYPE freq){ - if (Penabled==0) return;//if the resonance is disabled - REALTYPE sum=0.0, - l1=log(getfreqx(0.0)*ctlcenter), - l2=log(2.0)*getoctavesfreq()*ctlbw; - - for (int i=0;i<N_RES_POINTS;i++) if (sum<Prespoints[i]) sum=Prespoints[i]; - if (sum<1.0) sum=1.0; - - for (int i=1;i<n;i++){ - REALTYPE x=(log(freq*i)-l1)/l2;//compute where the n-th hamonics fits to the graph - if (x<0.0) x=0.0; - - x*=N_RES_POINTS; - REALTYPE dx=x-floor(x);x=floor(x); - int kx1=(int)x; if (kx1>=N_RES_POINTS) kx1=N_RES_POINTS-1; - int kx2=kx1+1;if (kx2>=N_RES_POINTS) kx2=N_RES_POINTS-1; - REALTYPE y=(Prespoints[kx1]*(1.0-dx)+Prespoints[kx2]*dx)/127.0-sum/127.0; - - y=pow(10.0,y*PmaxdB/20.0); - - if ((Pprotectthefundamental!=0)&&(i==1)) y=1.0; - - fftdata.c[i]*=y; - fftdata.s[i]*=y; - }; -}; - -/* - * Gets the response at the frequency "freq" - */ - -REALTYPE Resonance::getfreqresponse(REALTYPE freq){ - REALTYPE l1=log(getfreqx(0.0)*ctlcenter), - l2=log(2.0)*getoctavesfreq()*ctlbw,sum=0.0; - - for (int i=0;i<N_RES_POINTS;i++) if (sum<Prespoints[i]) sum=Prespoints[i]; - if (sum<1.0) sum=1.0; - - REALTYPE x=(log(freq)-l1)/l2;//compute where the n-th hamonics fits to the graph - if (x<0.0) x=0.0; - x*=N_RES_POINTS; - REALTYPE dx=x-floor(x);x=floor(x); - int kx1=(int)x; if (kx1>=N_RES_POINTS) kx1=N_RES_POINTS-1; - int kx2=kx1+1;if (kx2>=N_RES_POINTS) kx2=N_RES_POINTS-1; - REALTYPE result=(Prespoints[kx1]*(1.0-dx)+Prespoints[kx2]*dx)/127.0-sum/127.0; - result=pow(10.0,result*PmaxdB/20.0); - return(result); -}; - - -/* - * Smooth the resonance function - */ -void Resonance::smooth(){ - REALTYPE old=Prespoints[0]; - for (int i=0;i<N_RES_POINTS;i++){ - old=old*0.4+Prespoints[i]*0.6; - Prespoints[i]=(int) old; - }; - old=Prespoints[N_RES_POINTS-1]; - for (int i=N_RES_POINTS-1;i>0;i--){ - old=old*0.4+Prespoints[i]*0.6; - Prespoints[i]=(int) old+1; - if (Prespoints[i]>127) Prespoints[i]=127; - }; -}; - -/* - * Randomize the resonance function - */ -void Resonance::randomize(int type){ - int r=(int)(RND*127.0); - for (int i=0;i<N_RES_POINTS;i++){ - Prespoints[i]=r; - if ((RND<0.1)&&(type==0)) r=(int)(RND*127.0); - if ((RND<0.3)&&(type==1)) r=(int)(RND*127.0); - if (type==2) r=(int)(RND*127.0); - }; - smooth(); -}; - -/* - * Interpolate the peaks - */ -void Resonance::interpolatepeaks(int type){ - int x1=0,y1=Prespoints[0]; - for (int i=1;i<N_RES_POINTS;i++){ - if ((Prespoints[i]!=64)||(i+1==N_RES_POINTS)){ - int y2=Prespoints[i]; - for (int k=0;k<i-x1;k++){ - float x=(float) k/(i-x1); - if (type==0) x=(1-cos(x*PI))*0.5; - Prespoints[x1+k]=(int)(y1*(1.0-x)+y2*x); - }; - x1=i; - y1=y2; - }; - }; -}; - -/* - * Get the frequency from x, where x is [0..1]; x is the x coordinate - */ -REALTYPE Resonance::getfreqx(REALTYPE x){ - if (x>1.0) x=1.0; - REALTYPE octf=pow(2.0,getoctavesfreq()); - return(getcenterfreq()/sqrt(octf)*pow(octf,x)); -}; - -/* - * Get the x coordinate from frequency (used by the UI) - */ -REALTYPE Resonance::getfreqpos(REALTYPE freq){ - return((log(freq)-log(getfreqx(0.0)))/log(2.0)/getoctavesfreq()); -}; - -/* - * Get the center frequency of the resonance graph - */ -REALTYPE Resonance::getcenterfreq(){ - return(10000.0*pow(10,-(1.0-Pcenterfreq/127.0)*2.0)); -}; - -/* - * Get the number of octave that the resonance functions applies to - */ -REALTYPE Resonance::getoctavesfreq(){ - return(0.25+10.0*Poctavesfreq/127.0); -}; - -void Resonance::sendcontroller(MidiControllers ctl,REALTYPE par){ - if (ctl==C_resonance_center) ctlcenter=par; - else ctlbw=par; -}; - - - - -void Resonance::add2XML(XMLwrapper *xml){ - xml->addparbool("enabled",Penabled); - - if ((Penabled==0)&&(xml->minimal)) return; - - xml->addpar("max_db",PmaxdB); - xml->addpar("center_freq",Pcenterfreq); - xml->addpar("octaves_freq",Poctavesfreq); - xml->addparbool("protect_fundamental_frequency",Pprotectthefundamental); - xml->addpar("resonance_points",N_RES_POINTS); - for (int i=0;i<N_RES_POINTS;i++){ - xml->beginbranch("RESPOINT",i); - xml->addpar("val",Prespoints[i]); - xml->endbranch(); - }; -}; - - -void Resonance::getfromXML(XMLwrapper *xml){ - Penabled=xml->getparbool("enabled",Penabled); - - PmaxdB=xml->getpar127("max_db",PmaxdB); - Pcenterfreq=xml->getpar127("center_freq",Pcenterfreq); - Poctavesfreq=xml->getpar127("octaves_freq",Poctavesfreq); - Pprotectthefundamental=xml->getparbool("protect_fundamental_frequency",Pprotectthefundamental); - for (int i=0;i<N_RES_POINTS;i++){ - if (xml->enterbranch("RESPOINT",i)==0) continue; - Prespoints[i]=xml->getpar127("val",Prespoints[i]); - xml->exitbranch(); - }; -}; - - diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Resonance.h b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Resonance.h deleted file mode 100644 index 7b09e295..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Resonance.h +++ /dev/null @@ -1,68 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - Resonance.h - Resonance - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - -*/ -#ifndef RESONANCE_H -#define RESONANCE_H - -#include "../globals.h" -#include "../Misc/Util.h" -#include "../Misc/XMLwrapper.h" -#include "../Params/Presets.h" - -#define N_RES_POINTS 256 - -class Resonance:public Presets{ - public: - Resonance(); - ~Resonance(); - void setpoint(int n,unsigned char p); - void applyres(int n,FFTFREQS fftdata,REALTYPE freq); - void smooth(); - void interpolatepeaks(int type); - void randomize(int type); - - void add2XML(XMLwrapper *xml); - void defaults(); - void getfromXML(XMLwrapper *xml); - - - REALTYPE getfreqpos(REALTYPE freq); - REALTYPE getfreqx(REALTYPE x); - REALTYPE getfreqresponse(REALTYPE freq); - REALTYPE getcenterfreq(); - REALTYPE getoctavesfreq(); - void sendcontroller(MidiControllers ctl,REALTYPE par); - - //parameters - unsigned char Penabled; //if the ressonance is enabled - unsigned char Prespoints[N_RES_POINTS]; //how many points define the resonance function - unsigned char PmaxdB; //how many dB the signal may be amplified - unsigned char Pcenterfreq,Poctavesfreq; //the center frequency of the res. func., and the number of octaves - unsigned char Pprotectthefundamental; //the fundamental (1-st harmonic) is not damped, even it resonance function is low - - //controllers - REALTYPE ctlcenter;//center frequency(relative) - REALTYPE ctlbw;//bandwidth(relative) - - private: -}; - -#endif diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/SUBnote.C b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/SUBnote.C deleted file mode 100644 index f198ba04..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/SUBnote.C +++ /dev/null @@ -1,419 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - SUBnote.C - The "subtractive" synthesizer - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - -*/ - -#include <math.h> -#include <stdlib.h> -#include <stdio.h> -#include "../globals.h" -#include "SUBnote.h" -#include "../Misc/Util.h" - -SUBnote::SUBnote(SUBnoteParameters *parameters,Controller *ctl_,REALTYPE freq,REALTYPE velocity,int portamento_,int midinote){ - ready=0; - - tmpsmp=new REALTYPE[SOUND_BUFFER_SIZE]; - tmprnd=new REALTYPE[SOUND_BUFFER_SIZE]; - - pars=parameters; - ctl=ctl_; - portamento=portamento_; - NoteEnabled=ON; - volume=pow(0.1,3.0*(1.0-pars->PVolume/96.0));//-60 dB .. 0 dB - volume*=VelF(velocity,pars->PAmpVelocityScaleFunction); - if (pars->PPanning!=0) panning=pars->PPanning/127.0; - else panning=RND; - numstages=pars->Pnumstages; - stereo=pars->Pstereo; - start=pars->Pstart; - firsttick=1; - int pos[MAX_SUB_HARMONICS]; - - if (pars->Pfixedfreq==0) basefreq=freq; - else { - basefreq=440.0; - int fixedfreqET=pars->PfixedfreqET; - if (fixedfreqET!=0) {//if the frequency varies according the keyboard note - REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0); - if (fixedfreqET<=64) basefreq*=pow(2.0,tmp); - else basefreq*=pow(3.0,tmp); - }; - - }; - REALTYPE detune=getdetune(pars->PDetuneType,pars->PCoarseDetune,pars->PDetune); - basefreq*=pow(2.0,detune/1200.0);//detune -// basefreq*=ctl->pitchwheel.relfreq;//pitch wheel - - //global filter - GlobalFilterCenterPitch=pars->GlobalFilter->getfreq()+//center freq - (pars->PGlobalFilterVelocityScale/127.0*6.0)* //velocity sensing - (VelF(velocity,pars->PGlobalFilterVelocityScaleFunction)-1); - - GlobalFilterL=NULL;GlobalFilterR=NULL; - GlobalFilterEnvelope=NULL; - - //select only harmonics that desire to compute - numharmonics=0; - for (int n=0;n<MAX_SUB_HARMONICS;n++){ - if (pars->Phmag[n]==0)continue; - if (n*basefreq>SAMPLE_RATE/2.0) break;//remove the freqs above the Nyquist freq - pos[numharmonics++]=n; - }; - - if (numharmonics==0) { - NoteEnabled=OFF; - return; - }; - - - lfilter=new bpfilter[numstages*numharmonics]; - if (stereo!=0) rfilter=new bpfilter[numstages*numharmonics]; - - //how much the amplitude is normalised (because the harmonics) - REALTYPE reduceamp=0.0; - - for (int n=0;n<numharmonics;n++){ - - REALTYPE freq=basefreq*(pos[n]+1); - - //the bandwidth is not absolute(Hz); it is relative to frequency - REALTYPE bw=pow(10,(pars->Pbandwidth-127.0)/127.0*4)*numstages; - - //Bandwidth Scale - bw*=pow(1000/freq,(pars->Pbwscale-64.0)/64.0*3.0); - - //Relative BandWidth - bw*=pow(100,(pars->Phrelbw[pos[n]]-64.0)/64.0); - - if (bw>25.0) bw=25.0; - - //try to keep same amplitude on all freqs and bw. (empirically) - REALTYPE gain=sqrt(1500.0/(bw*freq)); - - REALTYPE hmagnew=1.0-pars->Phmag[pos[n]]/127.0; - REALTYPE hgain; - - switch(pars->Phmagtype){ - case 1:hgain=exp(hmagnew*log(0.01)); break; - case 2:hgain=exp(hmagnew*log(0.001));break; - case 3:hgain=exp(hmagnew*log(0.0001));break; - case 4:hgain=exp(hmagnew*log(0.00001));break; - default:hgain=1.0-hmagnew; - }; - gain*=hgain; - reduceamp+=hgain; - - for (int nph=0;nph<numstages;nph++){ - REALTYPE amp=1.0; - if (nph==0) amp=gain; - initfilter(lfilter[nph+n*numstages],freq,bw,amp,hgain); - if (stereo!=0) initfilter(rfilter[nph+n*numstages],freq,bw,amp,hgain); - }; - }; - - if (reduceamp<0.001) reduceamp=1.0; - volume/=reduceamp; - - oldpitchwheel=0; - oldbandwidth=64; - if (pars->Pfixedfreq==0) initparameters(basefreq); - else initparameters(basefreq/440.0*freq); - - oldamplitude=newamplitude; - ready=1; -}; - -SUBnote::~SUBnote(){ - if (NoteEnabled!=OFF) KillNote(); - delete [] tmpsmp; - delete [] tmprnd; -}; - -/* - * Kill the note - */ -void SUBnote::KillNote(){ - if (NoteEnabled!=OFF){ - delete [] lfilter; - lfilter=NULL; - if (stereo!=0) delete [] rfilter; - rfilter=NULL; - delete(AmpEnvelope); - if (FreqEnvelope!=NULL) delete(FreqEnvelope); - if (BandWidthEnvelope!=NULL) delete(BandWidthEnvelope); - NoteEnabled=OFF; - }; - -}; - - -/* - * Compute the filters coefficients - */ -void SUBnote::computefiltercoefs(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE gain){ - if (freq>SAMPLE_RATE/2.0-200.0) { - freq=SAMPLE_RATE/2.0-200.0; - }; - - REALTYPE omega=2.0*PI*freq/SAMPLE_RATE; - REALTYPE sn=sin(omega);REALTYPE cs=cos(omega); - REALTYPE alpha=sn*sinh(LOG_2/2.0*bw*omega/sn); - - if (alpha>1) alpha=1; - if (alpha>bw) alpha=bw; - - filter.b0=alpha/(1.0+alpha)*filter.amp*gain; - filter.b2=-alpha/(1.0+alpha)*filter.amp*gain; - filter.a1=-2.0*cs/(1.0+alpha); - filter.a2=(1.0-alpha)/(1.0+alpha); - -}; - - -/* - * Initialise the filters - */ -void SUBnote::initfilter(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE amp,REALTYPE mag){ - filter.xn1=0.0;filter.xn2=0.0; - - if (start==0) { - filter.yn1=0.0; - filter.yn2=0.0; - } else { - REALTYPE a=0.1*mag;//empirically - REALTYPE p=RND*2.0*PI; - if (start==1) a*=RND; - filter.yn1=a*cos(p); - filter.yn2=a*cos(p+freq*2.0*PI/SAMPLE_RATE); - - //correct the error of computation the start amplitude - //at very high frequencies - if (freq>SAMPLE_RATE*0.96) { - filter.yn1=0.0; - filter.yn2=0.0; - - }; - }; - - filter.amp=amp; - filter.freq=freq; - filter.bw=bw; - computefiltercoefs(filter,freq,bw,1.0); -}; - -/* - * Do the filtering - */ -void SUBnote::filter(bpfilter &filter,REALTYPE *smps){ - int i; - REALTYPE out; - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - out=smps[i] * filter.b0 + filter.b2 * filter.xn2 - -filter.a1 * filter.yn1 - filter.a2 * filter.yn2; - filter.xn2=filter.xn1; - filter.xn1=smps[i]; - filter.yn2=filter.yn1; - filter.yn1=out; - smps[i]=out; - - }; -}; - -/* - * Init Parameters - */ -void SUBnote::initparameters(REALTYPE freq){ - AmpEnvelope=new Envelope(pars->AmpEnvelope,freq); - if (pars->PFreqEnvelopeEnabled!=0) FreqEnvelope=new Envelope(pars->FreqEnvelope,freq); - else FreqEnvelope=NULL; - if (pars->PBandWidthEnvelopeEnabled!=0) BandWidthEnvelope=new Envelope(pars->BandWidthEnvelope,freq); - else BandWidthEnvelope=NULL; - if (pars->PGlobalFilterEnabled!=0){ - globalfiltercenterq=pars->GlobalFilter->getq(); - GlobalFilterL=new Filter(pars->GlobalFilter); - if (stereo!=0) GlobalFilterR=new Filter(pars->GlobalFilter); - GlobalFilterEnvelope=new Envelope(pars->GlobalFilterEnvelope,freq); - GlobalFilterFreqTracking=pars->GlobalFilter->getfreqtracking(basefreq); - }; - computecurrentparameters(); -}; - - -/* - * Compute Parameters of SUBnote for each tick - */ -void SUBnote::computecurrentparameters(){ - if ((FreqEnvelope!=NULL)||(BandWidthEnvelope!=NULL)|| - (oldpitchwheel!=ctl->pitchwheel.data)|| - (oldbandwidth!=ctl->bandwidth.data)|| - (portamento!=0)){ - REALTYPE envfreq=1.0; - REALTYPE envbw=1.0; - REALTYPE gain=1.0; - - if (FreqEnvelope!=NULL) { - envfreq=FreqEnvelope->envout()/1200; - envfreq=pow(2.0,envfreq); - }; - envfreq*=ctl->pitchwheel.relfreq;//pitch wheel - if (portamento!=0) {//portamento is used - envfreq*=ctl->portamento.freqrap; - if (ctl->portamento.used==0){//the portamento has finished - portamento=0;//this note is no longer "portamented" - }; - }; - - if (BandWidthEnvelope!=NULL) { - envbw=BandWidthEnvelope->envout(); - envbw=pow(2,envbw); - }; - envbw*=ctl->bandwidth.relbw;//bandwidth controller - - REALTYPE tmpgain=1.0/sqrt(envbw*envfreq); - - for (int n=0;n<numharmonics;n++){ - for (int nph=0;nph<numstages;nph++) { - if (nph==0) gain=tmpgain;else gain=1.0; - computefiltercoefs( lfilter[nph+n*numstages], - lfilter[nph+n*numstages].freq*envfreq, - lfilter[nph+n*numstages].bw*envbw,gain); - }; - }; - if (stereo!=0) - for (int n=0;n<numharmonics;n++){ - for (int nph=0;nph<numstages;nph++) { - if (nph==0) gain=tmpgain;else gain=1.0; - computefiltercoefs( rfilter[nph+n*numstages], - rfilter[nph+n*numstages].freq*envfreq, - rfilter[nph+n*numstages].bw*envbw,gain); - }; - }; - oldbandwidth=ctl->bandwidth.data; - oldpitchwheel=ctl->pitchwheel.data; - }; - newamplitude=volume*AmpEnvelope->envout_dB()*2.0; - - //Filter - if (GlobalFilterL!=NULL){ - REALTYPE globalfilterpitch=GlobalFilterCenterPitch+GlobalFilterEnvelope->envout(); - REALTYPE filterfreq=globalfilterpitch+ctl->filtercutoff.relfreq+GlobalFilterFreqTracking; - filterfreq=GlobalFilterL->getrealfreq(filterfreq); - - GlobalFilterL->setfreq_and_q(filterfreq,globalfiltercenterq*ctl->filterq.relq); - if (GlobalFilterR!=NULL) GlobalFilterR->setfreq_and_q(filterfreq,globalfiltercenterq*ctl->filterq.relq); - }; - -}; - -/* - * Note Output - */ -int SUBnote::noteout(REALTYPE *outl,REALTYPE *outr){ - int i; - - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - outl[i]=denormalkillbuf[i]; - outr[i]=denormalkillbuf[i]; - }; - - if (NoteEnabled==OFF) return(0); - - //left channel - for (i=0;i<SOUND_BUFFER_SIZE;i++) tmprnd[i]=RND*2.0-1.0; - for (int n=0;n<numharmonics;n++){ - for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpsmp[i]=tmprnd[i]; - for (int nph=0;nph<numstages;nph++) - filter(lfilter[nph+n*numstages],tmpsmp); - for (i=0;i<SOUND_BUFFER_SIZE;i++) outl[i]+=tmpsmp[i]; - }; - - if (GlobalFilterL!=NULL) GlobalFilterL->filterout(&outl[0]); - - //right channel - if (stereo!=0){ - for (i=0;i<SOUND_BUFFER_SIZE;i++) tmprnd[i]=RND*2.0-1.0; - for (int n=0;n<numharmonics;n++){ - for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpsmp[i]=tmprnd[i]; - for (int nph=0;nph<numstages;nph++) - filter(rfilter[nph+n*numstages],tmpsmp); - for (i=0;i<SOUND_BUFFER_SIZE;i++) outr[i]+=tmpsmp[i]; - }; - if (GlobalFilterR!=NULL) GlobalFilterR->filterout(&outr[0]); - } else for (i=0;i<SOUND_BUFFER_SIZE;i++) outr[i]=outl[i]; - - if (firsttick!=0){ - int n=10;if (n>SOUND_BUFFER_SIZE) n=SOUND_BUFFER_SIZE; - for (i=0;i<n;i++) { - REALTYPE ampfadein=0.5-0.5*cos((REALTYPE) i/(REALTYPE) n*PI); - outl[i]*=ampfadein; - outr[i]*=ampfadein; - }; - firsttick=0; - }; - - if (ABOVE_AMPLITUDE_THRESHOLD(oldamplitude,newamplitude)){ - // Amplitude interpolation - for (i=0;i<SOUND_BUFFER_SIZE;i++){ - REALTYPE tmpvol=INTERPOLATE_AMPLITUDE(oldamplitude - ,newamplitude,i,SOUND_BUFFER_SIZE); - outl[i]*=tmpvol*panning; - outr[i]*=tmpvol*(1.0-panning); - }; - } else { - for (i=0;i<SOUND_BUFFER_SIZE;i++) { - outl[i]*=newamplitude*panning; - outr[i]*=newamplitude*(1.0-panning); - }; - }; - - oldamplitude=newamplitude; - computecurrentparameters(); - - // Check if the note needs to be computed more - if (AmpEnvelope->finished()!=0){ - for (i=0;i<SOUND_BUFFER_SIZE;i++) {//fade-out - REALTYPE tmp=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE; - outl[i]*=tmp; - outr[i]*=tmp; - }; - KillNote(); - }; - return(1); -}; - -/* - * Relase Key (Note Off) - */ -void SUBnote::relasekey(){ - AmpEnvelope->relasekey(); - if (FreqEnvelope!=NULL) FreqEnvelope->relasekey(); - if (BandWidthEnvelope!=NULL) BandWidthEnvelope->relasekey(); - if (GlobalFilterEnvelope!=NULL) GlobalFilterEnvelope->relasekey(); -}; - -/* - * Check if the note is finished - */ -int SUBnote::finished(){ - if (NoteEnabled==OFF) return(1); - else return(0); -}; - diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/SUBnote.h b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/SUBnote.h deleted file mode 100644 index 6e4e2991..00000000 --- a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/SUBnote.h +++ /dev/null @@ -1,98 +0,0 @@ -/* - ZynAddSubFX - a software synthesizer - - SUBnote.h - The subtractive synthesizer - Copyright (C) 2002-2005 Nasca Octavian Paul - Author: Nasca Octavian Paul - - This program is free software; you can redistribute it and/or modify - it under the terms of version 2 of the GNU General Public License - as published by the Free Software Foundation. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License (version 2) for more details. - - You should have received a copy of the GNU General Public License (version 2) - along with this program; if not, write to the Free Software Foundation, - Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - -*/ - -#ifndef SUB_NOTE_H -#define SUB_NOTE_H - -#include "../globals.h" -#include "../Params/SUBnoteParameters.h" -#include "../Params/Controller.h" -#include "Envelope.h" -#include "../DSP/Filter.h" - -class SUBnote{ - public: - SUBnote(SUBnoteParameters *parameters,Controller *ctl_,REALTYPE freq,REALTYPE velocity,int portamento_,int midinote); - ~SUBnote(); - int noteout(REALTYPE *outl,REALTYPE *outr);//note output,return 0 if the note is finished - void relasekey(); - int finished(); - - int ready; //if I can get the sampledata - - private: - - void computecurrentparameters(); - void initparameters(REALTYPE freq); - void KillNote(); - - SUBnoteParameters *pars; - - //parameters - int stereo; - int numstages;//number of stages of filters - int numharmonics;//number of harmonics (after the too higher hamonics are removed) - int start;//how the harmonics start - REALTYPE basefreq; - REALTYPE panning; - Envelope *AmpEnvelope; - Envelope *FreqEnvelope; - Envelope *BandWidthEnvelope; - - Filter *GlobalFilterL,*GlobalFilterR; - - Envelope *GlobalFilterEnvelope; - - //internal values - ONOFFTYPE NoteEnabled; - int firsttick,portamento; - REALTYPE volume,oldamplitude,newamplitude; - - REALTYPE GlobalFilterCenterPitch;//octaves - REALTYPE GlobalFilterFreqTracking; - - struct bpfilter{ - REALTYPE freq,bw,amp; //filter parameters - REALTYPE a1,a2,b0,b2;//filter coefs. b1=0 - REALTYPE xn1,xn2,yn1,yn2; //filter internal values - }; - - void initfilter(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE amp,REALTYPE mag); - void computefiltercoefs(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE gain); - void filter(bpfilter &filter,REALTYPE *smps); - - bpfilter *lfilter,*rfilter; - - REALTYPE *tmpsmp; - REALTYPE *tmprnd;//this is filled with random numbers - - Controller *ctl; - int oldpitchwheel,oldbandwidth; - REALTYPE globalfiltercenterq; - -}; - - - - -#endif - |