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diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.C b/attic/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.C
deleted file mode 100644
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--- a/attic/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.C
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@@ -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));
-};
-
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