From e40fc849149dd97c248866a4a1d026dda5e57b62 Mon Sep 17 00:00:00 2001
From: Robert Jonsson <spamatica@gmail.com>
Date: Mon, 7 Mar 2011 19:01:11 +0000
Subject: clean3

---
 .../synti/zynaddsubfx/DSP/AnalogFilter.C           | 358 +++++++++++++++++++++
 1 file changed, 358 insertions(+)
 create 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
new file mode 100644
index 00000000..5e461a0b
--- /dev/null
+++ b/attic/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.C
@@ -0,0 +1,358 @@
+/*
+  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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