diff options
Diffstat (limited to 'muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C')
| -rw-r--r-- | muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C | 152 |
1 files changed, 152 insertions, 0 deletions
diff --git a/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C b/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C new file mode 100644 index 00000000..8c0e16b2 --- /dev/null +++ b/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C @@ -0,0 +1,152 @@ +/* + 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; + +}; + |