clean3

11 files changed, 0 insertions, 1202 deletions

diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.C b/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.C
deleted file mode 100644
index 5e461a0b..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.C
+++ /dev/null
@@ -1,358 +0,0 @@
-/*
-  ZynAddSubFX - a software synthesizer
- 
-  AnalogFilter.C - Several analog filters (lowpass, highpass...)
-  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 <stdio.h>
-#include "AnalogFilter.h"
-
-AnalogFilter::AnalogFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages){
-    stages=Fstages;
-    for (int i=0;i<3;i++){
-	oldc[i]=0.0;oldd[i]=0.0;
-	c[i]=0.0;d[i]=0.0;
-    };
-    type=Ftype;
-    freq=Ffreq;
-    q=Fq;
-    gain=1.0;
-    if (stages>=MAX_FILTER_STAGES) stages=MAX_FILTER_STAGES;
-    cleanup();
-    firsttime=0;
-    abovenq=0;oldabovenq=0;
-    setfreq_and_q(Ffreq,Fq);
-    firsttime=1;
-    d[0]=0;//this is not used
-    outgain=1.0;
-};
-
-AnalogFilter::~AnalogFilter(){
-};
-
-void AnalogFilter::cleanup(){
-    for (int i=0;i<MAX_FILTER_STAGES+1;i++){
-	x[i].c1=0.0;x[i].c2=0.0;
-	y[i].c1=0.0;y[i].c2=0.0;
-	oldx[i]=x[i];
-	oldy[i]=y[i];
-    };
-    needsinterpolation=0;
-};
-
-void AnalogFilter::computefiltercoefs(){
-    REALTYPE tmp;
-    REALTYPE omega,sn,cs,alpha,beta;
-    int zerocoefs=0;//this is used if the freq is too high
-
-    //do not allow frequencies bigger than samplerate/2
-    REALTYPE freq=this->freq;
-    if (freq>(SAMPLE_RATE/2-500.0)) {
-	freq=SAMPLE_RATE/2-500.0;
-	zerocoefs=1;
-    };
-    if (freq<0.1) freq=0.1;
-    //do not allow bogus Q
-    if (q<0.0) q=0.0;
-    REALTYPE tmpq,tmpgain;
-    if (stages==0) {
-	tmpq=q;
-	tmpgain=gain;
-    } else {
-        tmpq=(q>1.0 ? pow(q,1.0/(stages+1)) : q);
-	tmpgain=pow(gain,1.0/(stages+1));
-    };
-    
-    //most of theese are implementations of 
-    //the "Cookbook formulae for audio EQ" by Robert Bristow-Johnson
-    //The original location of the Cookbook is:
-    //http://www.harmony-central.com/Computer/Programming/Audio-EQ-Cookbook.txt
-    switch(type){
-	case 0://LPF 1 pole	    
-	    if (zerocoefs==0) tmp=exp(-2.0*PI*freq/SAMPLE_RATE);
-		else tmp=0.0;
-	    c[0]=1.0-tmp;c[1]=0.0;c[2]=0.0;
-	    d[1]=tmp;d[2]=0.0;
-	    order=1;
-	    break;
-	case 1://HPF 1 pole
-	    if (zerocoefs==0) tmp=exp(-2.0*PI*freq/SAMPLE_RATE);
-		else tmp=0.0;
-	    c[0]=(1.0+tmp)/2.0;c[1]=-(1.0+tmp)/2.0;c[2]=0.0;
-	    d[1]=tmp;d[2]=0.0;
-	    order=1;
-	    break;
-	case 2://LPF 2 poles 
-	    if (zerocoefs==0){
-		omega=2*PI*freq/SAMPLE_RATE;
-	        sn=sin(omega);
-		cs=cos(omega);
-		alpha=sn/(2*tmpq);
-		tmp=1+alpha;
-		c[0]=(1.0-cs)/2.0/tmp;
-		c[1]=(1.0-cs)/tmp;
-		c[2]=(1.0-cs)/2.0/tmp;
-		d[1]=-2*cs/tmp*(-1);
-		d[2]=(1-alpha)/tmp*(-1);
-	    } else {
-		c[0]=1.0;c[1]=0.0;c[2]=0.0;
-		d[1]=0.0;d[2]=0.0;
-	    };	    
-	    order=2;
-	    break;
-	case 3://HPF 2 poles 
-	    if (zerocoefs==0){
-		omega=2*PI*freq/SAMPLE_RATE;
-    		sn=sin(omega);
-		cs=cos(omega);
-		alpha=sn/(2*tmpq);
-		tmp=1+alpha;
-		c[0]=(1.0+cs)/2.0/tmp;
-		c[1]=-(1.0+cs)/tmp;
-		c[2]=(1.0+cs)/2.0/tmp;
-		d[1]=-2*cs/tmp*(-1);
-		d[2]=(1-alpha)/tmp*(-1);
-	    } else {
-		c[0]=0.0;c[1]=0.0;c[2]=0.0;
-		d[1]=0.0;d[2]=0.0;
-	    };
-	    order=2;
-	    break;
-	case 4://BPF 2 poles 
-	    if (zerocoefs==0){
-		omega=2*PI*freq/SAMPLE_RATE;
-    		sn=sin(omega);
-		cs=cos(omega);
-		alpha=sn/(2*tmpq);
-		tmp=1+alpha;
-		c[0]=alpha/tmp*sqrt(tmpq+1);
-		c[1]=0;
-		c[2]=-alpha/tmp*sqrt(tmpq+1);
-		d[1]=-2*cs/tmp*(-1);
-		d[2]=(1-alpha)/tmp*(-1);
-	    } else {
-		c[0]=0.0;c[1]=0.0;c[2]=0.0;
-		d[1]=0.0;d[2]=0.0;
-	    };
-	    order=2;
-	    break;
-	case 5://NOTCH 2 poles 
-	    if (zerocoefs==0){		
-		omega=2*PI*freq/SAMPLE_RATE;
-    		sn=sin(omega);
-		cs=cos(omega);
-		alpha=sn/(2*sqrt(tmpq));
-		tmp=1+alpha;
-		c[0]=1/tmp;
-		c[1]=-2*cs/tmp;
-		c[2]=1/tmp;		
-		d[1]=-2*cs/tmp*(-1);
-		d[2]=(1-alpha)/tmp*(-1);
-	    } else {
-		c[0]=1.0;c[1]=0.0;c[2]=0.0;
-		d[1]=0.0;d[2]=0.0;
-	    };	    
-	    order=2;
-	    break;
-	case 6://PEAK (2 poles)
-	    if (zerocoefs==0){
-		omega=2*PI*freq/SAMPLE_RATE;
-	        sn=sin(omega);
-		cs=cos(omega);
-		tmpq*=3.0;
-		alpha=sn/(2*tmpq);
-		tmp=1+alpha/tmpgain;
-		c[0]=(1.0+alpha*tmpgain)/tmp;
-		c[1]=(-2.0*cs)/tmp;
-		c[2]=(1.0-alpha*tmpgain)/tmp;
-		d[1]=-2*cs/tmp*(-1);
-		d[2]=(1-alpha/tmpgain)/tmp*(-1);
-	    } else {
-		c[0]=1.0;c[1]=0.0;c[2]=0.0;
-		d[1]=0.0;d[2]=0.0;
-	    };
-	    order=2;
-	    break;
-	case 7://Low Shelf - 2 poles
-	    if (zerocoefs==0){
-		omega=2*PI*freq/SAMPLE_RATE;
-	        sn=sin(omega);
-		cs=cos(omega);
-		tmpq=sqrt(tmpq);
-		alpha=sn/(2*tmpq);
-		beta=sqrt(tmpgain)/tmpq;
-		tmp=(tmpgain+1.0)+(tmpgain-1.0)*cs+beta*sn;
-
-		c[0]=tmpgain*((tmpgain+1.0)-(tmpgain-1.0)*cs+beta*sn)/tmp;		
-		c[1]=2.0*tmpgain*((tmpgain-1.0)-(tmpgain+1.0)*cs)/tmp;
-		c[2]=tmpgain*((tmpgain+1.0)-(tmpgain-1.0)*cs-beta*sn)/tmp;		
-		d[1]=-2.0*((tmpgain-1.0)+(tmpgain+1.0)*cs)/tmp*(-1);
-		d[2]=((tmpgain+1.0)+(tmpgain-1.0)*cs-beta*sn)/tmp*(-1);
-	    } else {
-		c[0]=tmpgain;c[1]=0.0;c[2]=0.0;
-		d[1]=0.0;d[2]=0.0;
-	    };
-	    order=2;
-	    break;
-	case 8://High Shelf - 2 poles
-	    if (zerocoefs==0){
-		omega=2*PI*freq/SAMPLE_RATE;
-	        sn=sin(omega);
-		cs=cos(omega);
-		tmpq=sqrt(tmpq);
-		alpha=sn/(2*tmpq);
-		beta=sqrt(tmpgain)/tmpq;
-		tmp=(tmpgain+1.0)-(tmpgain-1.0)*cs+beta*sn;
-
-		c[0]=tmpgain*((tmpgain+1.0)+(tmpgain-1.0)*cs+beta*sn)/tmp;		
-		c[1]=-2.0*tmpgain*((tmpgain-1.0)+(tmpgain+1.0)*cs)/tmp;
-		c[2]=tmpgain*((tmpgain+1.0)+(tmpgain-1.0)*cs-beta*sn)/tmp;		
-		d[1]=2.0*((tmpgain-1.0)-(tmpgain+1.0)*cs)/tmp*(-1);
-		d[2]=((tmpgain+1.0)-(tmpgain-1.0)*cs-beta*sn)/tmp*(-1);
-	    } else {
-		c[0]=1.0;c[1]=0.0;c[2]=0.0;
-		d[1]=0.0;d[2]=0.0;
-	    };
-	    order=2;
-	    break;
-	default://wrong type
-	 type=0;
-	 computefiltercoefs();
-	break;
-    };
-};
-
-
-void AnalogFilter::setfreq(REALTYPE frequency){
-    if (frequency<0.1) frequency=0.1;
-    REALTYPE rap=freq/frequency;if (rap<1.0) rap=1.0/rap;
-    
-    oldabovenq=abovenq;abovenq=frequency>(SAMPLE_RATE/2-500.0);
-    
-    int nyquistthresh=(abovenq^oldabovenq);
-
-
-    if ((rap>3.0)||(nyquistthresh!=0)){//if the frequency is changed fast, it needs interpolation (now, filter and coeficients backup)
-	for (int i=0;i<3;i++){
-	    oldc[i]=c[i];oldd[i]=d[i];
-	};
-	for (int i=0;i<MAX_FILTER_STAGES+1;i++){
-	    oldx[i]=x[i];
-	    oldy[i]=y[i];
-	};
-	if (firsttime==0) needsinterpolation=1;
-    };
-    freq=frequency;
-    computefiltercoefs();
-    firsttime=0;
-
-};
-
-void AnalogFilter::setfreq_and_q(REALTYPE frequency,REALTYPE q_){
-    q=q_;
-    setfreq(frequency);
-};
-
-void AnalogFilter::setq(REALTYPE q_){
-    q=q_;
-    computefiltercoefs();
-};
-
-void AnalogFilter::settype(int type_){
-    type=type_;
-    computefiltercoefs();
-};
-
-void AnalogFilter::setgain(REALTYPE dBgain){
-    gain=dB2rap(dBgain);
-    computefiltercoefs();
-};
-
-void AnalogFilter::setstages(int stages_){
-    if (stages_>=MAX_FILTER_STAGES) stages_=MAX_FILTER_STAGES-1;
-    stages=stages_;
-    cleanup();
-    computefiltercoefs();
-};
-
-void AnalogFilter::singlefilterout(REALTYPE *smp,fstage &x,fstage &y,REALTYPE *c,REALTYPE *d){
-    int i;
-    REALTYPE y0;
-    if (order==1) {//First order filter
-	for (i=0;i<SOUND_BUFFER_SIZE;i++){
-	    y0=smp[i]*c[0]+x.c1*c[1]+y.c1*d[1];
-	    y.c1=y0;
-	    x.c1=smp[i];
-	    //output
-	    smp[i]=y0;
-	};
-    };
-    if (order==2) {//Second order filter
-	for (i=0;i<SOUND_BUFFER_SIZE;i++){
-	    y0=smp[i]*c[0]+x.c1*c[1]+x.c2*c[2]+y.c1*d[1]+y.c2*d[2];
-	    y.c2=y.c1;
-	    y.c1=y0;
-	    x.c2=x.c1;
-	    x.c1=smp[i];
-	    //output
-	    smp[i]=y0;
-	};
-    };
-};
-void AnalogFilter::filterout(REALTYPE *smp){
-    REALTYPE *ismp=NULL;//used if it needs interpolation
-    int i;
-    if (needsinterpolation!=0){
-	ismp=new REALTYPE[SOUND_BUFFER_SIZE];
-	for (i=0;i<SOUND_BUFFER_SIZE;i++) ismp[i]=smp[i];
-	for (i=0;i<stages+1;i++) singlefilterout(ismp,oldx[i],oldy[i],oldc,oldd);
-    };
-    
-    for (i=0;i<stages+1;i++) singlefilterout(smp,x[i],y[i],c,d);
-
-    if (needsinterpolation!=0){
-	for (i=0;i<SOUND_BUFFER_SIZE;i++) {
-	    REALTYPE x=i/(REALTYPE) SOUND_BUFFER_SIZE;
-	    smp[i]=ismp[i]*(1.0-x)+smp[i]*x;
-	};
-	delete (ismp);
-	needsinterpolation=0;
-    };
-
-    for (i=0;i<SOUND_BUFFER_SIZE;i++) smp[i]*=outgain;
-};
-
-REALTYPE AnalogFilter::H(REALTYPE freq){
-    REALTYPE fr=freq/SAMPLE_RATE*PI*2.0;
-    REALTYPE x=c[0],y=0.0;
-    for (int n=1;n<3;n++){
-	x+=cos(n*fr)*c[n];
-	y-=sin(n*fr)*c[n];
-    };
-    REALTYPE h=x*x+y*y;
-    x=1.0;y=0.0;
-    for (int n=1;n<3;n++){
-	x-=cos(n*fr)*d[n];
-	y+=sin(n*fr)*d[n];
-    };
-    h=h/(x*x+y*y);
-    return(pow(h,(stages+1.0)/2.0));
-};
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.h b/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.h
deleted file mode 100644
index 2e9fe68b..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
-  ZynAddSubFX - a software synthesizer
- 
-  Analog Filter.h - Several analog filters (lowpass, highpass...)
-  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 ANALOG_FILTER_H
-#define ANALOG_FILTER_H
-
-#include "../globals.h"
-#include "Filter_.h"
-class AnalogFilter:public Filter_{
- public:
-    AnalogFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages);
-    ~AnalogFilter();	
-    void filterout(REALTYPE *smp);
-    void setfreq(REALTYPE frequency);
-    void setfreq_and_q(REALTYPE frequency,REALTYPE q_);
-    void setq(REALTYPE q_);
-
-    void settype(int type_);
-    void setgain(REALTYPE dBgain);
-    void setstages(int stages_);
-    void cleanup();
-    
-    REALTYPE H(REALTYPE freq);//Obtains the response for a given frequency
-    
- private:
-    struct fstage{
-      REALTYPE c1,c2;
-    } x[MAX_FILTER_STAGES+1],y[MAX_FILTER_STAGES+1],
-      oldx[MAX_FILTER_STAGES+1],oldy[MAX_FILTER_STAGES+1];
-
-    void singlefilterout(REALTYPE *smp,fstage &x,fstage &y,REALTYPE *c,REALTYPE *d);
-    void computefiltercoefs();
-    int type;//The type of the filter (LPF1,HPF1,LPF2,HPF2...)
-    int stages;//how many times the filter is applied (0->1,1->2,etc.)
-    REALTYPE freq;//Frequency given in Hz
-    REALTYPE q; //Q factor (resonance or Q factor)
-    REALTYPE gain;//the gain of the filter (if are shelf/peak) filters
-    
-    int order;//the order of the filter (number of poles)
-
-    REALTYPE c[3],d[3];//coefficients
-
-    REALTYPE oldc[3],oldd[3];//old coefficients(used only if some filter paremeters changes very fast, and it needs interpolation)
-
-    REALTYPE xd[3],yd[3];//used if the filter is applied more times
-    int needsinterpolation,firsttime;
-    int abovenq;//this is 1 if the frequency is above the nyquist
-    int oldabovenq;//if the last time was above nyquist (used to see if it needs interpolation)
-};
-
-
-#endif
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.C b/muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.C
deleted file mode 100644
index 7c67e631..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.C
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
-  ZynAddSubFX - a software synthesizer
- 
-  FFTwrapper.c  -  A wrapper for Fast Fourier Transforms
-  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 "FFTwrapper.h"
-
-FFTwrapper::FFTwrapper(int fftsize_){
-    fftsize=fftsize_;
-    tmpfftdata1=new fftw_real[fftsize];
-    tmpfftdata2=new fftw_real[fftsize];
-#ifdef FFTW_VERSION_2
-    planfftw=rfftw_create_plan(fftsize,FFTW_REAL_TO_COMPLEX,FFTW_ESTIMATE|FFTW_IN_PLACE);
-    planfftw_inv=rfftw_create_plan(fftsize,FFTW_COMPLEX_TO_REAL,FFTW_ESTIMATE|FFTW_IN_PLACE);
-#else
-    planfftw=fftw_plan_r2r_1d(fftsize,tmpfftdata1,tmpfftdata1,FFTW_R2HC,FFTW_ESTIMATE);
-    planfftw_inv=fftw_plan_r2r_1d(fftsize,tmpfftdata2,tmpfftdata2,FFTW_HC2R,FFTW_ESTIMATE);
-#endif
-};
-
-FFTwrapper::~FFTwrapper(){
-#ifdef FFTW_VERSION_2
-    rfftw_destroy_plan(planfftw);
-    rfftw_destroy_plan(planfftw_inv);
-#else 
-    fftw_destroy_plan(planfftw);
-    fftw_destroy_plan(planfftw_inv);
-#endif
-
-    delete [] tmpfftdata1;
-    delete [] tmpfftdata2;
-};
-
-/*
- * do the Fast Fourier Transform
- */
-void FFTwrapper::smps2freqs(REALTYPE *smps,FFTFREQS freqs){
-#ifdef FFTW_VERSION_2
-    for (int i=0;i<fftsize;i++) tmpfftdata1[i]=smps[i];
-    rfftw_one(planfftw,tmpfftdata1,tmpfftdata2);
-    for (int i=0;i<fftsize/2;i++) {
-	freqs.c[i]=tmpfftdata2[i];
-	if (i!=0) freqs.s[i]=tmpfftdata2[fftsize-i];
-    };
-#else
-    for (int i=0;i<fftsize;i++) tmpfftdata1[i]=smps[i];
-    fftw_execute(planfftw);
-    for (int i=0;i<fftsize/2;i++) {
-	freqs.c[i]=tmpfftdata1[i];
-	if (i!=0) freqs.s[i]=tmpfftdata1[fftsize-i];
-    };
-#endif
-    tmpfftdata2[fftsize/2]=0.0;
-};
-
-/*
- * do the Inverse Fast Fourier Transform
- */
-void FFTwrapper::freqs2smps(FFTFREQS freqs,REALTYPE *smps){
-    tmpfftdata2[fftsize/2]=0.0;
-#ifdef FFTW_VERSION_2
-    for (int i=0;i<fftsize/2;i++) {
-	tmpfftdata1[i]=freqs.c[i];
-	if (i!=0) tmpfftdata1[fftsize-i]=freqs.s[i];
-    };
-    rfftw_one(planfftw_inv,tmpfftdata1,tmpfftdata2);
-    for (int i=0;i<fftsize;i++) smps[i]=tmpfftdata2[i];
-#else
-    for (int i=0;i<fftsize/2;i++) {
-	tmpfftdata2[i]=freqs.c[i];
-	if (i!=0) tmpfftdata2[fftsize-i]=freqs.s[i];
-    };
-    fftw_execute(planfftw_inv);
-    for (int i=0;i<fftsize;i++) smps[i]=tmpfftdata2[i];
-#endif
-
-};
-
-
-
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.h b/muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.h
deleted file mode 100644
index df8cad7a..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
-  ZynAddSubFX - a software synthesizer
- 
-  FFTwrapper.h  -  A wrapper for Fast Fourier Transforms
-  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 FFT_WRAPPER_H
-#define FFT_WRAPPER_H
-
-#include "../globals.h"
-
-#ifdef FFTW_VERSION_2
-
-#include <fftw.h>
-
-/* If you got error messages about rfftw.h, replace the next include  line with "#include <srfftw.h>"
-or with "#include <drfftw.h> (if one doesn't work try the other). It may be necessary to replace
-the <fftw.h> with <dfftw.h> or <sfftw.h>. If the neither one doesn't work, 
-please install latest version of fftw(recomanded from the sources) from www.fftw.org.
-If you'll install fftw3 you need to change the Makefile.inc
-Hope all goes right." */
-#include <rfftw.h>
-
-#else
-
-#include <fftw3.h>
-#define fftw_real double
-#define rfftw_plan fftw_plan
-#endif
-
-class FFTwrapper{
-    public:
-	FFTwrapper(int fftsize_);
-	~FFTwrapper();
-	void smps2freqs(REALTYPE *smps,FFTFREQS freqs);
-	void freqs2smps(FFTFREQS freqs,REALTYPE *smps);
-    private:
-	int fftsize;
-	fftw_real *tmpfftdata1,*tmpfftdata2;
-	rfftw_plan planfftw,planfftw_inv;
-};
-#endif
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.C b/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.C
deleted file mode 100644
index fccb0265..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.C
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
-  ZynAddSubFX - a software synthesizer
- 
-  Filter.C - Filters, uses analog,formant,etc. filters
-  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 <stdio.h>
-
-#include "Filter.h"
-
-Filter::Filter(FilterParams *pars){
-    unsigned char Ftype=pars->Ptype;
-    unsigned char Fstages=pars->Pstages;
-    
-    category=pars->Pcategory;
-    
-    switch (category) {
-	case 1:filter=new FormantFilter(pars);
-		break;
-	case 2:filter=new SVFilter(Ftype,1000.0,pars->getq(),Fstages);
-	       filter->outgain=dB2rap(pars->getgain());		
-	       if (filter->outgain>1.0) filter->outgain=sqrt(filter->outgain);
-		break;
-	default:filter=new AnalogFilter(Ftype,1000.0,pars->getq(),Fstages);
-		if ((Ftype>=6)&&(Ftype<=8)) filter->setgain(pars->getgain());
-		    else filter->outgain=dB2rap(pars->getgain());		
-		break;
-    };
-};
-
-Filter::~Filter(){
-    delete (filter);
-};
-
-void Filter::filterout(REALTYPE *smp){
-    filter->filterout(smp);
-};
-
-void Filter::setfreq(REALTYPE frequency){
-    filter->setfreq(frequency);
-};
-
-void Filter::setfreq_and_q(REALTYPE frequency,REALTYPE q_){
-    filter->setfreq_and_q(frequency,q_);
-};
-
-void Filter::setq(REALTYPE q_){
-    filter->setq(q_);
-};
-
-REALTYPE Filter::getrealfreq(REALTYPE freqpitch){
-    if ((category==0)||(category==2)) return(pow(2.0,freqpitch+9.96578428));//log2(1000)=9.95748
-	else return(freqpitch);
-};
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.h b/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.h
deleted file mode 100644
index dab948c1..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
-  ZynAddSubFX - a software synthesizer
- 
-  Filter.h - Filters, uses analog,formant,etc. filters
-  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 FILTER_H
-#define FILTER_H
-
-#include "../globals.h"
-
-#include "Filter_.h"
-#include "AnalogFilter.h"
-#include "FormantFilter.h"
-#include "SVFilter.h"
-#include "../Params/FilterParams.h"
-
-class Filter{
- public:
-    Filter(FilterParams *pars);
-    ~Filter();	
-    void filterout(REALTYPE *smp);
-    void setfreq(REALTYPE frequency);
-    void setfreq_and_q(REALTYPE frequency,REALTYPE q_);
-    void setq(REALTYPE q_);
-
-    REALTYPE getrealfreq(REALTYPE freqpitch);
- private:
-    Filter_ *filter;
-    unsigned char category;
-};
-
-
-#endif
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter_.h b/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter_.h
deleted file mode 100644
index 66fff867..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter_.h
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
-  ZynAddSubFX - a software synthesizer
- 
-  Filter_.h - This class is inherited by filter classes
-  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 FILTER__H
-#define FILTER__H
-
-#include "../globals.h"
-
-class Filter_{
- public:
-    virtual ~Filter_(){};	
-    virtual void filterout(REALTYPE */*smp*/){};
-    virtual void setfreq(REALTYPE /*frequency*/){};
-    virtual void setfreq_and_q(REALTYPE /*frequency*/,REALTYPE /*q_*/){};
-    virtual void setq(REALTYPE /*q_*/){};
-    virtual void setgain(REALTYPE /*dBgain*/){};
-    REALTYPE outgain;
- private:
-};
-
-
-#endif
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.C b/muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.C
deleted file mode 100644
index 482cef91..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.C
+++ /dev/null
@@ -1,163 +0,0 @@
-/*
-  ZynAddSubFX - a software synthesizer
- 
-  FormantFilter.C - formant filters
-  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 <stdio.h>
-#include "FormantFilter.h"
-
-FormantFilter::FormantFilter(FilterParams *pars){
-    numformants=pars->Pnumformants;
-    for (int i=0;i<numformants;i++) formant[i]=new AnalogFilter(4/*BPF*/,1000.0,10.0,pars->Pstages);
-    cleanup();
-    inbuffer=new REALTYPE [SOUND_BUFFER_SIZE];
-    tmpbuf=new REALTYPE [SOUND_BUFFER_SIZE];
-
-    for (int j=0;j<FF_MAX_VOWELS;j++) 
-	for (int i=0;i<numformants;i++){
-	    formantpar[j][i].freq=pars->getformantfreq(pars->Pvowels[j].formants[i].freq);
-	    formantpar[j][i].amp=pars->getformantamp(pars->Pvowels[j].formants[i].amp);
-	    formantpar[j][i].q=pars->getformantq(pars->Pvowels[j].formants[i].q);
-	};
-    for (int i=0;i<FF_MAX_FORMANTS;i++) oldformantamp[i]=1.0;
-    for (int i=0;i<numformants;i++){
-	currentformants[i].freq=1000.0;
-	currentformants[i].amp=1.0;
-	currentformants[i].q=2.0;
-    };
-
-    formantslowness=pow(1.0-(pars->Pformantslowness/128.0),3.0);
-
-    sequencesize=pars->Psequencesize;if (sequencesize==0) sequencesize=1;
-    for (int k=0;k<sequencesize;k++) sequence[k].nvowel=pars->Psequence[k].nvowel;
-    
-    vowelclearness=pow(10.0,(pars->Pvowelclearness-32.0)/48.0);
-
-    sequencestretch=pow(0.1,(pars->Psequencestretch-32.0)/48.0);
-    if (pars->Psequencereversed) sequencestretch*= -1.0;
-
-    outgain=dB2rap(pars->getgain());
-
-    oldinput=-1.0;
-    Qfactor=1.0;oldQfactor=Qfactor;
-    firsttime=1;
-};
-
-FormantFilter::~FormantFilter(){
-    for (int i=0;i<numformants;i++) delete(formant[i]);
-    delete (inbuffer);
-    delete (tmpbuf);
-};
-
-
-
-
-void FormantFilter::cleanup(){
-    for (int i=0;i<numformants;i++) formant[i]->cleanup();
-};
-
-void FormantFilter::setpos(REALTYPE input){
-    int p1,p2;
-    
-    if (firsttime!=0) slowinput=input;
-	else slowinput=slowinput*(1.0-formantslowness)+input*formantslowness;
-
-    if ((fabs(oldinput-input)<0.001)&&(fabs(slowinput-input)<0.001)&&
-	(fabs(Qfactor-oldQfactor)<0.001)) {
-//	oldinput=input; daca setez asta, o sa faca probleme la schimbari foarte lente
-	firsttime=0;
-	return;
-    } else oldinput=input;
-
-
-    REALTYPE pos=fmod(input*sequencestretch,1.0);if (pos<0.0) pos+=1.0;
-    
-    F2I(pos*sequencesize,p2);
-    p1=p2-1;if (p1<0) p1+=sequencesize;
-
-    pos=fmod(pos*sequencesize,1.0);    
-    if (pos<0.0) pos=0.0; else if (pos>1.0) pos=1.0;
-    pos=(atan((pos*2.0-1.0)*vowelclearness)/atan(vowelclearness)+1.0)*0.5;
-
-    p1=sequence[p1].nvowel;
-    p2=sequence[p2].nvowel;
-    
-    if (firsttime!=0) {
-       for (int i=0;i<numformants;i++){
-	    currentformants[i].freq=formantpar[p1][i].freq*(1.0-pos)+formantpar[p2][i].freq*pos;
-	    currentformants[i].amp=formantpar[p1][i].amp*(1.0-pos)+formantpar[p2][i].amp*pos;
-	    currentformants[i].q=formantpar[p1][i].q*(1.0-pos)+formantpar[p2][i].q*pos;	
-	    formant[i]->setfreq_and_q(currentformants[i].freq,currentformants[i].q*Qfactor);
-	    oldformantamp[i]=currentformants[i].amp;
-	};
-	firsttime=0;
-    } else {
-	for (int i=0;i<numformants;i++){
-	    currentformants[i].freq=currentformants[i].freq*(1.0-formantslowness)
-		+(formantpar[p1][i].freq*(1.0-pos)+formantpar[p2][i].freq*pos)*formantslowness;
-
-	    currentformants[i].amp=currentformants[i].amp*(1.0-formantslowness)
-		+(formantpar[p1][i].amp*(1.0-pos)+formantpar[p2][i].amp*pos)*formantslowness;
-
-	    currentformants[i].q=currentformants[i].q*(1.0-formantslowness)
-		+(formantpar[p1][i].q*(1.0-pos)+formantpar[p2][i].q*pos)*formantslowness;
-
-	    formant[i]->setfreq_and_q(currentformants[i].freq,currentformants[i].q*Qfactor);
-	};
-    };
-    
-    oldQfactor=Qfactor;
-};
-
-void FormantFilter::setfreq(REALTYPE frequency){
-    setpos(frequency);
-};
-
-void FormantFilter::setq(REALTYPE q_){
-    Qfactor=q_;
-    for (int i=0;i<numformants;i++) formant[i]->setq(Qfactor*currentformants[i].q);
-};
-
-void FormantFilter::setfreq_and_q(REALTYPE frequency,REALTYPE q_){    
-    Qfactor=q_;    
-    setpos(frequency);
-};
-
-
-void FormantFilter::filterout(REALTYPE *smp){
-    int i,j;
-    for (i=0;i<SOUND_BUFFER_SIZE;i++) {
-	inbuffer[i]=smp[i];
-	smp[i]=0.0;
-    };
-    
-    for (j=0;j<numformants;j++) {
-        for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpbuf[i]=inbuffer[i]*outgain;
-	formant[j]->filterout(tmpbuf);
-
-	if (ABOVE_AMPLITUDE_THRESHOLD(oldformantamp[j],currentformants[j].amp))
-    	    for (i=0;i<SOUND_BUFFER_SIZE;i++) smp[i]+=tmpbuf[i]*
-		INTERPOLATE_AMPLITUDE(oldformantamp[j],currentformants[j].amp,i,SOUND_BUFFER_SIZE);
-	else for (i=0;i<SOUND_BUFFER_SIZE;i++) smp[i]+=tmpbuf[i]*currentformants[j].amp;
-	oldformantamp[j]=currentformants[j].amp;
-    };
-};
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.h b/muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.h
deleted file mode 100644
index 7cb52499..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
-  ZynAddSubFX - a software synthesizer
- 
-  FormantFilter.h - formant filter
-  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 FORMANT_FILTER_H
-#define FORMANT_FILTER_H
-
-#include "../globals.h"
-#include "Filter_.h"
-#include "AnalogFilter.h"
-#include "../Params/FilterParams.h"
-
-
-class FormantFilter:public Filter_{
- public:
-    FormantFilter(FilterParams *pars);
-    ~FormantFilter();	
-    void filterout(REALTYPE *smp);
-    void setfreq(REALTYPE frequency);
-    void setfreq_and_q(REALTYPE frequency,REALTYPE q_);
-    void setq(REALTYPE q_);
-
-    void cleanup();
- private:
-    AnalogFilter *formant[FF_MAX_FORMANTS];
-    REALTYPE *inbuffer,*tmpbuf;
-
-    struct {
-	REALTYPE freq,amp,q;//frequency,amplitude,Q
-    } formantpar[FF_MAX_VOWELS][FF_MAX_FORMANTS],currentformants[FF_MAX_FORMANTS];
-
-    struct {
-	unsigned char nvowel;
-    } sequence [FF_MAX_SEQUENCE];
-    
-    REALTYPE oldformantamp[FF_MAX_FORMANTS];
-    
-    int sequencesize,numformants,firsttime;
-    REALTYPE oldinput,slowinput;
-    REALTYPE Qfactor,formantslowness,oldQfactor;
-    REALTYPE vowelclearness,sequencestretch;
-    
-    void setpos(REALTYPE input);
-    
-};
-
-
-#endif
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C b/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C
deleted file mode 100644
index 8c0e16b2..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C
+++ /dev/null
@@ -1,152 +0,0 @@
-/*
-  ZynAddSubFX - a software synthesizer
- 
-  SVFilter.C - Several state-variable filters
-  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 <stdio.h>
-#include "SVFilter.h"
-
-SVFilter::SVFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages){
-    stages=Fstages;
-    type=Ftype;
-    freq=Ffreq;
-    q=Fq;
-    gain=1.0;
-    outgain=1.0;
-    needsinterpolation=0;
-    firsttime=1;
-    if (stages>=MAX_FILTER_STAGES) stages=MAX_FILTER_STAGES;
-    cleanup();
-    setfreq_and_q(Ffreq,Fq);
-};
-
-SVFilter::~SVFilter(){
-};
-
-void SVFilter::cleanup(){
-    for (int i=0;i<MAX_FILTER_STAGES+1;i++){
-	st[i].low=0.0;st[i].high=0.0;
-	st[i].band=0.0;st[i].notch=0.0;
-    };
-    oldabovenq=0;
-    abovenq=0;
-};
-
-void SVFilter::computefiltercoefs(){
-    par.f=freq / SAMPLE_RATE*4.0;
-    if (par.f>0.99999) par.f=0.99999;
-    par.q=1.0-atan(sqrt(q))*2.0/PI;
-    par.q=pow(par.q,1.0/(stages+1));
-    par.q_sqrt=sqrt(par.q);
-};
-
-
-void SVFilter::setfreq(REALTYPE frequency){
-    if (frequency<0.1) frequency=0.1;
-    REALTYPE rap=freq/frequency;if (rap<1.0) rap=1.0/rap;
-    
-    oldabovenq=abovenq;abovenq=frequency>(SAMPLE_RATE/2-500.0);
-    
-    int nyquistthresh=(abovenq^oldabovenq);
-
-
-    if ((rap>3.0)||(nyquistthresh!=0)){//if the frequency is changed fast, it needs interpolation (now, filter and coeficients backup)
-	if (firsttime==0) needsinterpolation=1;
-	ipar=par;
-    };
-    freq=frequency;
-    computefiltercoefs();
-    firsttime=0;
-
-};
-
-void SVFilter::setfreq_and_q(REALTYPE frequency,REALTYPE q_){
-    q=q_;
-    setfreq(frequency);
-};
-
-void SVFilter::setq(REALTYPE q_){
-    q=q_;
-    computefiltercoefs();
-};
-
-void SVFilter::settype(int type_){
-    type=type_;
-    computefiltercoefs();
-};
-
-void SVFilter::setgain(REALTYPE dBgain){
-    gain=dB2rap(dBgain);
-    computefiltercoefs();
-};
-
-void SVFilter::setstages(int stages_){
-    if (stages_>=MAX_FILTER_STAGES) stages_=MAX_FILTER_STAGES-1;
-    stages=stages_;
-    cleanup();
-    computefiltercoefs();
-};
-
-void SVFilter::singlefilterout(REALTYPE *smp,fstage &x,parameters &par){
-    int i;
-    REALTYPE *out=NULL;
-    switch(type){
-	case 0: out=&x.low;break;
-	case 1: out=&x.high;break;
-	case 2: out=&x.band;break;
-	case 3: out=&x.notch;break;
-    };
-    
-    for (i=0;i<SOUND_BUFFER_SIZE;i++){
-	  x.low = x.low + par.f * x.band;
-	  x.high = par.q_sqrt * smp[i] - x.low - par.q*x.band;
-	  x.band = par.f * x.high + x.band;
-	  x.notch = x.high + x.low;
-
-	  smp[i]= *out;
-    };
-};
-
-void SVFilter::filterout(REALTYPE *smp){
-    int i;
-    REALTYPE *ismp=NULL;
-    
-    if (needsinterpolation!=0){
-	ismp=new REALTYPE[SOUND_BUFFER_SIZE];
-	for (i=0;i<SOUND_BUFFER_SIZE;i++) ismp[i]=smp[i];
-	for (i=0;i<stages+1;i++) singlefilterout(ismp,st[i],ipar);
-    };
-
-    for (i=0;i<stages+1;i++) singlefilterout(smp,st[i],par);
-
-    if (needsinterpolation!=0){
-	for (i=0;i<SOUND_BUFFER_SIZE;i++) {
-	    REALTYPE x=i/(REALTYPE) SOUND_BUFFER_SIZE;
-	    smp[i]=ismp[i]*(1.0-x)+smp[i]*x;
-	};
-	delete (ismp);
-	needsinterpolation=0;
-    };
-
-    for (i=0;i<SOUND_BUFFER_SIZE;i++) smp[i]*=outgain;
-
-};
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.h b/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.h
deleted file mode 100644
index 3117e2c9..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
-  ZynAddSubFX - a software synthesizer
- 
-  SV Filter.h - Several state-variable filters
-  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 SV_FILTER_H
-#define SV_FILTER_H
-
-#include "../globals.h"
-#include "Filter_.h"
-class SVFilter:public Filter_{
- public:
-    SVFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages);
-    ~SVFilter();	
-    void filterout(REALTYPE *smp);
-    void setfreq(REALTYPE frequency);
-    void setfreq_and_q(REALTYPE frequency,REALTYPE q_);
-    void setq(REALTYPE q_);
-
-    void settype(int type_);
-    void setgain(REALTYPE dBgain);
-    void setstages(int stages_);
-    void cleanup();
-
- private:
-    struct fstage{
-      REALTYPE low,high,band,notch;
-    } st[MAX_FILTER_STAGES+1];
-
-    struct parameters{
-	REALTYPE f,q,q_sqrt;
-    }par,ipar;
-
-
-    void singlefilterout(REALTYPE *smp,fstage &x,parameters &par);
-    void computefiltercoefs();
-    int type;//The type of the filter (LPF1,HPF1,LPF2,HPF2...)
-    int stages;//how many times the filter is applied (0->1,1->2,etc.)
-    REALTYPE freq;//Frequency given in Hz
-    REALTYPE q; //Q factor (resonance or Q factor)
-    REALTYPE gain;//the gain of the filter (if are shelf/peak) filters
-    
-    int abovenq;//this is 1 if the frequency is above the nyquist
-    int oldabovenq;
-    int needsinterpolation,firsttime;
-};
-
-
-#endif
-