From b0546e5e7f7044019892543c6c82029db8d564a7 Mon Sep 17 00:00:00 2001
From: Robert Jonsson <spamatica@gmail.com>
Date: Thu, 15 Sep 2011 12:14:55 +0000
Subject: moved attic to a branch of it's own

---
 .../synti/zynaddsubfx/DSP/AnalogFilter.C           | 358 ---------------------
 1 file changed, 358 deletions(-)
 delete mode 100644 attic/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.C

(limited to 'attic/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.C')

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
index 5e461a0b..00000000
--- a/attic/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));
-};
-
-- 
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