moved attic to a branch of it's own

14 files changed, 0 insertions, 4284 deletions

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