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