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+/*
+ 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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