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-rw-r--r--muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.C358
-rw-r--r--muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.h72
-rw-r--r--muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.C99
-rw-r--r--muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.h59
-rw-r--r--muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.C72
-rw-r--r--muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.h51
-rw-r--r--muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter_.h42
-rw-r--r--muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.C163
-rw-r--r--muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.h67
-rw-r--r--muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C152
-rw-r--r--muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.h67
11 files changed, 0 insertions, 1202 deletions
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.C b/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.C
deleted file mode 100644
index 5e461a0b..00000000
--- a/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));
-};
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.h b/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.h
deleted file mode 100644
index 2e9fe68b..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/AnalogFilter.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- ZynAddSubFX - a software synthesizer
-
- Analog Filter.h - 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
-
-*/
-
-#ifndef ANALOG_FILTER_H
-#define ANALOG_FILTER_H
-
-#include "../globals.h"
-#include "Filter_.h"
-class AnalogFilter:public Filter_{
- public:
- AnalogFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages);
- ~AnalogFilter();
- void filterout(REALTYPE *smp);
- void setfreq(REALTYPE frequency);
- void setfreq_and_q(REALTYPE frequency,REALTYPE q_);
- void setq(REALTYPE q_);
-
- void settype(int type_);
- void setgain(REALTYPE dBgain);
- void setstages(int stages_);
- void cleanup();
-
- REALTYPE H(REALTYPE freq);//Obtains the response for a given frequency
-
- private:
- struct fstage{
- REALTYPE c1,c2;
- } x[MAX_FILTER_STAGES+1],y[MAX_FILTER_STAGES+1],
- oldx[MAX_FILTER_STAGES+1],oldy[MAX_FILTER_STAGES+1];
-
- void singlefilterout(REALTYPE *smp,fstage &x,fstage &y,REALTYPE *c,REALTYPE *d);
- void computefiltercoefs();
- int type;//The type of the filter (LPF1,HPF1,LPF2,HPF2...)
- int stages;//how many times the filter is applied (0->1,1->2,etc.)
- REALTYPE freq;//Frequency given in Hz
- REALTYPE q; //Q factor (resonance or Q factor)
- REALTYPE gain;//the gain of the filter (if are shelf/peak) filters
-
- int order;//the order of the filter (number of poles)
-
- REALTYPE c[3],d[3];//coefficients
-
- REALTYPE oldc[3],oldd[3];//old coefficients(used only if some filter paremeters changes very fast, and it needs interpolation)
-
- REALTYPE xd[3],yd[3];//used if the filter is applied more times
- int needsinterpolation,firsttime;
- int abovenq;//this is 1 if the frequency is above the nyquist
- int oldabovenq;//if the last time was above nyquist (used to see if it needs interpolation)
-};
-
-
-#endif
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.C b/muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.C
deleted file mode 100644
index 7c67e631..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.C
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- ZynAddSubFX - a software synthesizer
-
- FFTwrapper.c - A wrapper for Fast Fourier Transforms
- 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 "FFTwrapper.h"
-
-FFTwrapper::FFTwrapper(int fftsize_){
- fftsize=fftsize_;
- tmpfftdata1=new fftw_real[fftsize];
- tmpfftdata2=new fftw_real[fftsize];
-#ifdef FFTW_VERSION_2
- planfftw=rfftw_create_plan(fftsize,FFTW_REAL_TO_COMPLEX,FFTW_ESTIMATE|FFTW_IN_PLACE);
- planfftw_inv=rfftw_create_plan(fftsize,FFTW_COMPLEX_TO_REAL,FFTW_ESTIMATE|FFTW_IN_PLACE);
-#else
- planfftw=fftw_plan_r2r_1d(fftsize,tmpfftdata1,tmpfftdata1,FFTW_R2HC,FFTW_ESTIMATE);
- planfftw_inv=fftw_plan_r2r_1d(fftsize,tmpfftdata2,tmpfftdata2,FFTW_HC2R,FFTW_ESTIMATE);
-#endif
-};
-
-FFTwrapper::~FFTwrapper(){
-#ifdef FFTW_VERSION_2
- rfftw_destroy_plan(planfftw);
- rfftw_destroy_plan(planfftw_inv);
-#else
- fftw_destroy_plan(planfftw);
- fftw_destroy_plan(planfftw_inv);
-#endif
-
- delete [] tmpfftdata1;
- delete [] tmpfftdata2;
-};
-
-/*
- * do the Fast Fourier Transform
- */
-void FFTwrapper::smps2freqs(REALTYPE *smps,FFTFREQS freqs){
-#ifdef FFTW_VERSION_2
- for (int i=0;i<fftsize;i++) tmpfftdata1[i]=smps[i];
- rfftw_one(planfftw,tmpfftdata1,tmpfftdata2);
- for (int i=0;i<fftsize/2;i++) {
- freqs.c[i]=tmpfftdata2[i];
- if (i!=0) freqs.s[i]=tmpfftdata2[fftsize-i];
- };
-#else
- for (int i=0;i<fftsize;i++) tmpfftdata1[i]=smps[i];
- fftw_execute(planfftw);
- for (int i=0;i<fftsize/2;i++) {
- freqs.c[i]=tmpfftdata1[i];
- if (i!=0) freqs.s[i]=tmpfftdata1[fftsize-i];
- };
-#endif
- tmpfftdata2[fftsize/2]=0.0;
-};
-
-/*
- * do the Inverse Fast Fourier Transform
- */
-void FFTwrapper::freqs2smps(FFTFREQS freqs,REALTYPE *smps){
- tmpfftdata2[fftsize/2]=0.0;
-#ifdef FFTW_VERSION_2
- for (int i=0;i<fftsize/2;i++) {
- tmpfftdata1[i]=freqs.c[i];
- if (i!=0) tmpfftdata1[fftsize-i]=freqs.s[i];
- };
- rfftw_one(planfftw_inv,tmpfftdata1,tmpfftdata2);
- for (int i=0;i<fftsize;i++) smps[i]=tmpfftdata2[i];
-#else
- for (int i=0;i<fftsize/2;i++) {
- tmpfftdata2[i]=freqs.c[i];
- if (i!=0) tmpfftdata2[fftsize-i]=freqs.s[i];
- };
- fftw_execute(planfftw_inv);
- for (int i=0;i<fftsize;i++) smps[i]=tmpfftdata2[i];
-#endif
-
-};
-
-
-
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.h b/muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.h
deleted file mode 100644
index df8cad7a..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FFTwrapper.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- ZynAddSubFX - a software synthesizer
-
- FFTwrapper.h - A wrapper for Fast Fourier Transforms
- 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
-
-*/
-
-#ifndef FFT_WRAPPER_H
-#define FFT_WRAPPER_H
-
-#include "../globals.h"
-
-#ifdef FFTW_VERSION_2
-
-#include <fftw.h>
-
-/* If you got error messages about rfftw.h, replace the next include line with "#include <srfftw.h>"
-or with "#include <drfftw.h> (if one doesn't work try the other). It may be necessary to replace
-the <fftw.h> with <dfftw.h> or <sfftw.h>. If the neither one doesn't work,
-please install latest version of fftw(recomanded from the sources) from www.fftw.org.
-If you'll install fftw3 you need to change the Makefile.inc
-Hope all goes right." */
-#include <rfftw.h>
-
-#else
-
-#include <fftw3.h>
-#define fftw_real double
-#define rfftw_plan fftw_plan
-#endif
-
-class FFTwrapper{
- public:
- FFTwrapper(int fftsize_);
- ~FFTwrapper();
- void smps2freqs(REALTYPE *smps,FFTFREQS freqs);
- void freqs2smps(FFTFREQS freqs,REALTYPE *smps);
- private:
- int fftsize;
- fftw_real *tmpfftdata1,*tmpfftdata2;
- rfftw_plan planfftw,planfftw_inv;
-};
-#endif
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.C b/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.C
deleted file mode 100644
index fccb0265..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.C
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- ZynAddSubFX - a software synthesizer
-
- Filter.C - Filters, uses analog,formant,etc. filters
- 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 "Filter.h"
-
-Filter::Filter(FilterParams *pars){
- unsigned char Ftype=pars->Ptype;
- unsigned char Fstages=pars->Pstages;
-
- category=pars->Pcategory;
-
- switch (category) {
- case 1:filter=new FormantFilter(pars);
- break;
- case 2:filter=new SVFilter(Ftype,1000.0,pars->getq(),Fstages);
- filter->outgain=dB2rap(pars->getgain());
- if (filter->outgain>1.0) filter->outgain=sqrt(filter->outgain);
- break;
- default:filter=new AnalogFilter(Ftype,1000.0,pars->getq(),Fstages);
- if ((Ftype>=6)&&(Ftype<=8)) filter->setgain(pars->getgain());
- else filter->outgain=dB2rap(pars->getgain());
- break;
- };
-};
-
-Filter::~Filter(){
- delete (filter);
-};
-
-void Filter::filterout(REALTYPE *smp){
- filter->filterout(smp);
-};
-
-void Filter::setfreq(REALTYPE frequency){
- filter->setfreq(frequency);
-};
-
-void Filter::setfreq_and_q(REALTYPE frequency,REALTYPE q_){
- filter->setfreq_and_q(frequency,q_);
-};
-
-void Filter::setq(REALTYPE q_){
- filter->setq(q_);
-};
-
-REALTYPE Filter::getrealfreq(REALTYPE freqpitch){
- if ((category==0)||(category==2)) return(pow(2.0,freqpitch+9.96578428));//log2(1000)=9.95748
- else return(freqpitch);
-};
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.h b/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.h
deleted file mode 100644
index dab948c1..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- ZynAddSubFX - a software synthesizer
-
- Filter.h - Filters, uses analog,formant,etc. filters
- 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
-
-*/
-
-#ifndef FILTER_H
-#define FILTER_H
-
-#include "../globals.h"
-
-#include "Filter_.h"
-#include "AnalogFilter.h"
-#include "FormantFilter.h"
-#include "SVFilter.h"
-#include "../Params/FilterParams.h"
-
-class Filter{
- public:
- Filter(FilterParams *pars);
- ~Filter();
- void filterout(REALTYPE *smp);
- void setfreq(REALTYPE frequency);
- void setfreq_and_q(REALTYPE frequency,REALTYPE q_);
- void setq(REALTYPE q_);
-
- REALTYPE getrealfreq(REALTYPE freqpitch);
- private:
- Filter_ *filter;
- unsigned char category;
-};
-
-
-#endif
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter_.h b/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter_.h
deleted file mode 100644
index 66fff867..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/Filter_.h
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- ZynAddSubFX - a software synthesizer
-
- Filter_.h - This class is inherited by filter classes
- 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
-
-*/
-
-#ifndef FILTER__H
-#define FILTER__H
-
-#include "../globals.h"
-
-class Filter_{
- public:
- virtual ~Filter_(){};
- virtual void filterout(REALTYPE */*smp*/){};
- virtual void setfreq(REALTYPE /*frequency*/){};
- virtual void setfreq_and_q(REALTYPE /*frequency*/,REALTYPE /*q_*/){};
- virtual void setq(REALTYPE /*q_*/){};
- virtual void setgain(REALTYPE /*dBgain*/){};
- REALTYPE outgain;
- private:
-};
-
-
-#endif
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.C b/muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.C
deleted file mode 100644
index 482cef91..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.C
+++ /dev/null
@@ -1,163 +0,0 @@
-/*
- ZynAddSubFX - a software synthesizer
-
- FormantFilter.C - formant filters
- 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 "FormantFilter.h"
-
-FormantFilter::FormantFilter(FilterParams *pars){
- numformants=pars->Pnumformants;
- for (int i=0;i<numformants;i++) formant[i]=new AnalogFilter(4/*BPF*/,1000.0,10.0,pars->Pstages);
- cleanup();
- inbuffer=new REALTYPE [SOUND_BUFFER_SIZE];
- tmpbuf=new REALTYPE [SOUND_BUFFER_SIZE];
-
- for (int j=0;j<FF_MAX_VOWELS;j++)
- for (int i=0;i<numformants;i++){
- formantpar[j][i].freq=pars->getformantfreq(pars->Pvowels[j].formants[i].freq);
- formantpar[j][i].amp=pars->getformantamp(pars->Pvowels[j].formants[i].amp);
- formantpar[j][i].q=pars->getformantq(pars->Pvowels[j].formants[i].q);
- };
- for (int i=0;i<FF_MAX_FORMANTS;i++) oldformantamp[i]=1.0;
- for (int i=0;i<numformants;i++){
- currentformants[i].freq=1000.0;
- currentformants[i].amp=1.0;
- currentformants[i].q=2.0;
- };
-
- formantslowness=pow(1.0-(pars->Pformantslowness/128.0),3.0);
-
- sequencesize=pars->Psequencesize;if (sequencesize==0) sequencesize=1;
- for (int k=0;k<sequencesize;k++) sequence[k].nvowel=pars->Psequence[k].nvowel;
-
- vowelclearness=pow(10.0,(pars->Pvowelclearness-32.0)/48.0);
-
- sequencestretch=pow(0.1,(pars->Psequencestretch-32.0)/48.0);
- if (pars->Psequencereversed) sequencestretch*= -1.0;
-
- outgain=dB2rap(pars->getgain());
-
- oldinput=-1.0;
- Qfactor=1.0;oldQfactor=Qfactor;
- firsttime=1;
-};
-
-FormantFilter::~FormantFilter(){
- for (int i=0;i<numformants;i++) delete(formant[i]);
- delete (inbuffer);
- delete (tmpbuf);
-};
-
-
-
-
-void FormantFilter::cleanup(){
- for (int i=0;i<numformants;i++) formant[i]->cleanup();
-};
-
-void FormantFilter::setpos(REALTYPE input){
- int p1,p2;
-
- if (firsttime!=0) slowinput=input;
- else slowinput=slowinput*(1.0-formantslowness)+input*formantslowness;
-
- if ((fabs(oldinput-input)<0.001)&&(fabs(slowinput-input)<0.001)&&
- (fabs(Qfactor-oldQfactor)<0.001)) {
-// oldinput=input; daca setez asta, o sa faca probleme la schimbari foarte lente
- firsttime=0;
- return;
- } else oldinput=input;
-
-
- REALTYPE pos=fmod(input*sequencestretch,1.0);if (pos<0.0) pos+=1.0;
-
- F2I(pos*sequencesize,p2);
- p1=p2-1;if (p1<0) p1+=sequencesize;
-
- pos=fmod(pos*sequencesize,1.0);
- if (pos<0.0) pos=0.0; else if (pos>1.0) pos=1.0;
- pos=(atan((pos*2.0-1.0)*vowelclearness)/atan(vowelclearness)+1.0)*0.5;
-
- p1=sequence[p1].nvowel;
- p2=sequence[p2].nvowel;
-
- if (firsttime!=0) {
- for (int i=0;i<numformants;i++){
- currentformants[i].freq=formantpar[p1][i].freq*(1.0-pos)+formantpar[p2][i].freq*pos;
- currentformants[i].amp=formantpar[p1][i].amp*(1.0-pos)+formantpar[p2][i].amp*pos;
- currentformants[i].q=formantpar[p1][i].q*(1.0-pos)+formantpar[p2][i].q*pos;
- formant[i]->setfreq_and_q(currentformants[i].freq,currentformants[i].q*Qfactor);
- oldformantamp[i]=currentformants[i].amp;
- };
- firsttime=0;
- } else {
- for (int i=0;i<numformants;i++){
- currentformants[i].freq=currentformants[i].freq*(1.0-formantslowness)
- +(formantpar[p1][i].freq*(1.0-pos)+formantpar[p2][i].freq*pos)*formantslowness;
-
- currentformants[i].amp=currentformants[i].amp*(1.0-formantslowness)
- +(formantpar[p1][i].amp*(1.0-pos)+formantpar[p2][i].amp*pos)*formantslowness;
-
- currentformants[i].q=currentformants[i].q*(1.0-formantslowness)
- +(formantpar[p1][i].q*(1.0-pos)+formantpar[p2][i].q*pos)*formantslowness;
-
- formant[i]->setfreq_and_q(currentformants[i].freq,currentformants[i].q*Qfactor);
- };
- };
-
- oldQfactor=Qfactor;
-};
-
-void FormantFilter::setfreq(REALTYPE frequency){
- setpos(frequency);
-};
-
-void FormantFilter::setq(REALTYPE q_){
- Qfactor=q_;
- for (int i=0;i<numformants;i++) formant[i]->setq(Qfactor*currentformants[i].q);
-};
-
-void FormantFilter::setfreq_and_q(REALTYPE frequency,REALTYPE q_){
- Qfactor=q_;
- setpos(frequency);
-};
-
-
-void FormantFilter::filterout(REALTYPE *smp){
- int i,j;
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- inbuffer[i]=smp[i];
- smp[i]=0.0;
- };
-
- for (j=0;j<numformants;j++) {
- for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpbuf[i]=inbuffer[i]*outgain;
- formant[j]->filterout(tmpbuf);
-
- if (ABOVE_AMPLITUDE_THRESHOLD(oldformantamp[j],currentformants[j].amp))
- for (i=0;i<SOUND_BUFFER_SIZE;i++) smp[i]+=tmpbuf[i]*
- INTERPOLATE_AMPLITUDE(oldformantamp[j],currentformants[j].amp,i,SOUND_BUFFER_SIZE);
- else for (i=0;i<SOUND_BUFFER_SIZE;i++) smp[i]+=tmpbuf[i]*currentformants[j].amp;
- oldformantamp[j]=currentformants[j].amp;
- };
-};
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.h b/muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.h
deleted file mode 100644
index 7cb52499..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/FormantFilter.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- ZynAddSubFX - a software synthesizer
-
- FormantFilter.h - formant filter
- 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
-
-*/
-
-#ifndef FORMANT_FILTER_H
-#define FORMANT_FILTER_H
-
-#include "../globals.h"
-#include "Filter_.h"
-#include "AnalogFilter.h"
-#include "../Params/FilterParams.h"
-
-
-class FormantFilter:public Filter_{
- public:
- FormantFilter(FilterParams *pars);
- ~FormantFilter();
- void filterout(REALTYPE *smp);
- void setfreq(REALTYPE frequency);
- void setfreq_and_q(REALTYPE frequency,REALTYPE q_);
- void setq(REALTYPE q_);
-
- void cleanup();
- private:
- AnalogFilter *formant[FF_MAX_FORMANTS];
- REALTYPE *inbuffer,*tmpbuf;
-
- struct {
- REALTYPE freq,amp,q;//frequency,amplitude,Q
- } formantpar[FF_MAX_VOWELS][FF_MAX_FORMANTS],currentformants[FF_MAX_FORMANTS];
-
- struct {
- unsigned char nvowel;
- } sequence [FF_MAX_SEQUENCE];
-
- REALTYPE oldformantamp[FF_MAX_FORMANTS];
-
- int sequencesize,numformants,firsttime;
- REALTYPE oldinput,slowinput;
- REALTYPE Qfactor,formantslowness,oldQfactor;
- REALTYPE vowelclearness,sequencestretch;
-
- void setpos(REALTYPE input);
-
-};
-
-
-#endif
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C b/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C
deleted file mode 100644
index 8c0e16b2..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C
+++ /dev/null
@@ -1,152 +0,0 @@
-/*
- ZynAddSubFX - a software synthesizer
-
- SVFilter.C - Several state-variable filters
- 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 "SVFilter.h"
-
-SVFilter::SVFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages){
- stages=Fstages;
- type=Ftype;
- freq=Ffreq;
- q=Fq;
- gain=1.0;
- outgain=1.0;
- needsinterpolation=0;
- firsttime=1;
- if (stages>=MAX_FILTER_STAGES) stages=MAX_FILTER_STAGES;
- cleanup();
- setfreq_and_q(Ffreq,Fq);
-};
-
-SVFilter::~SVFilter(){
-};
-
-void SVFilter::cleanup(){
- for (int i=0;i<MAX_FILTER_STAGES+1;i++){
- st[i].low=0.0;st[i].high=0.0;
- st[i].band=0.0;st[i].notch=0.0;
- };
- oldabovenq=0;
- abovenq=0;
-};
-
-void SVFilter::computefiltercoefs(){
- par.f=freq / SAMPLE_RATE*4.0;
- if (par.f>0.99999) par.f=0.99999;
- par.q=1.0-atan(sqrt(q))*2.0/PI;
- par.q=pow(par.q,1.0/(stages+1));
- par.q_sqrt=sqrt(par.q);
-};
-
-
-void SVFilter::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)
- if (firsttime==0) needsinterpolation=1;
- ipar=par;
- };
- freq=frequency;
- computefiltercoefs();
- firsttime=0;
-
-};
-
-void SVFilter::setfreq_and_q(REALTYPE frequency,REALTYPE q_){
- q=q_;
- setfreq(frequency);
-};
-
-void SVFilter::setq(REALTYPE q_){
- q=q_;
- computefiltercoefs();
-};
-
-void SVFilter::settype(int type_){
- type=type_;
- computefiltercoefs();
-};
-
-void SVFilter::setgain(REALTYPE dBgain){
- gain=dB2rap(dBgain);
- computefiltercoefs();
-};
-
-void SVFilter::setstages(int stages_){
- if (stages_>=MAX_FILTER_STAGES) stages_=MAX_FILTER_STAGES-1;
- stages=stages_;
- cleanup();
- computefiltercoefs();
-};
-
-void SVFilter::singlefilterout(REALTYPE *smp,fstage &x,parameters &par){
- int i;
- REALTYPE *out=NULL;
- switch(type){
- case 0: out=&x.low;break;
- case 1: out=&x.high;break;
- case 2: out=&x.band;break;
- case 3: out=&x.notch;break;
- };
-
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- x.low = x.low + par.f * x.band;
- x.high = par.q_sqrt * smp[i] - x.low - par.q*x.band;
- x.band = par.f * x.high + x.band;
- x.notch = x.high + x.low;
-
- smp[i]= *out;
- };
-};
-
-void SVFilter::filterout(REALTYPE *smp){
- int i;
- REALTYPE *ismp=NULL;
-
- 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,st[i],ipar);
- };
-
- for (i=0;i<stages+1;i++) singlefilterout(smp,st[i],par);
-
- 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;
-
-};
-
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.h b/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.h
deleted file mode 100644
index 3117e2c9..00000000
--- a/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- ZynAddSubFX - a software synthesizer
-
- SV Filter.h - Several state-variable filters
- 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
-
-*/
-
-#ifndef SV_FILTER_H
-#define SV_FILTER_H
-
-#include "../globals.h"
-#include "Filter_.h"
-class SVFilter:public Filter_{
- public:
- SVFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages);
- ~SVFilter();
- void filterout(REALTYPE *smp);
- void setfreq(REALTYPE frequency);
- void setfreq_and_q(REALTYPE frequency,REALTYPE q_);
- void setq(REALTYPE q_);
-
- void settype(int type_);
- void setgain(REALTYPE dBgain);
- void setstages(int stages_);
- void cleanup();
-
- private:
- struct fstage{
- REALTYPE low,high,band,notch;
- } st[MAX_FILTER_STAGES+1];
-
- struct parameters{
- REALTYPE f,q,q_sqrt;
- }par,ipar;
-
-
- void singlefilterout(REALTYPE *smp,fstage &x,parameters &par);
- void computefiltercoefs();
- int type;//The type of the filter (LPF1,HPF1,LPF2,HPF2...)
- int stages;//how many times the filter is applied (0->1,1->2,etc.)
- REALTYPE freq;//Frequency given in Hz
- REALTYPE q; //Q factor (resonance or Q factor)
- REALTYPE gain;//the gain of the filter (if are shelf/peak) filters
-
- int abovenq;//this is 1 if the frequency is above the nyquist
- int oldabovenq;
- int needsinterpolation,firsttime;
-};
-
-
-#endif
-
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-28 20:05:43 +0000