From e40fc849149dd97c248866a4a1d026dda5e57b62 Mon Sep 17 00:00:00 2001
From: Robert Jonsson <spamatica@gmail.com>
Date: Mon, 7 Mar 2011 19:01:11 +0000
Subject: clean3

---
 .../synti/zynaddsubfx/DSP/SVFilter.C               | 152 +++++++++++++++++++++
 1 file changed, 152 insertions(+)
 create mode 100644 attic/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C

(limited to 'attic/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C')

diff --git a/attic/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C b/attic/muse_qt4_evolution/synti/zynaddsubfx/DSP/SVFilter.C
new file mode 100644
index 00000000..8c0e16b2
--- /dev/null
+++ b/attic/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;
+
+};
+
-- 
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