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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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