clean3

14 files changed, 4284 insertions, 0 deletions

diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/ADnote.C b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/ADnote.C
new file mode 100644
index 00000000..574e2bea
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/ADnote.C
@@ -0,0 +1,984 @@
+/*
+  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
new file mode 100644
index 00000000..28c18975
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/ADnote.h
@@ -0,0 +1,258 @@
+/*
+  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
new file mode 100644
index 00000000..a0194022
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Envelope.C
@@ -0,0 +1,165 @@
+/*
+  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
new file mode 100644
index 00000000..d78eb16d
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Envelope.h
@@ -0,0 +1,58 @@
+/*
+  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
new file mode 100644
index 00000000..4ae548c1
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/LFO.C
@@ -0,0 +1,145 @@
+/*
+  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
new file mode 100644
index 00000000..30d04f10
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/LFO.h
@@ -0,0 +1,52 @@
+/*
+  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
new file mode 100644
index 00000000..4e6a4dd3
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/OscilGen.C
@@ -0,0 +1,1182 @@
+/*
+  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
new file mode 100644
index 00000000..1d9980a9
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/OscilGen.h
@@ -0,0 +1,176 @@
+/*
+  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
new file mode 100644
index 00000000..9ecc8877
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/PADnote.C
@@ -0,0 +1,342 @@
+/*
+  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
new file mode 100644
index 00000000..2a99577f
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/PADnote.h
@@ -0,0 +1,106 @@
+/*
+  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
new file mode 100644
index 00000000..fb741055
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Resonance.C
@@ -0,0 +1,231 @@
+/*
+  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
new file mode 100644
index 00000000..7b09e295
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/Resonance.h
@@ -0,0 +1,68 @@
+/*
+  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
new file mode 100644
index 00000000..f198ba04
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/SUBnote.C
@@ -0,0 +1,419 @@
+/*
+  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
new file mode 100644
index 00000000..6e4e2991
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/Synth/SUBnote.h
@@ -0,0 +1,98 @@
+/*
+  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
+