commit 5c3b6e6e097b18c033f36331bb85f3188a2baffa
parent eb427f941f0f6d60845a66ec678d3db7b2b52802
Author: Jonathan Moore Liles <[email protected]>
Date: Fri, 11 Dec 2020 17:37:41 -0800
Revert "SVFilter: Use Value_Smoothing_Filter to smooth cutoff frequency changes."
Still has zipper artifact, needs more work.
Diffstat:
2 files changed, 69 insertions(+), 30 deletions(-)
diff --git a/src/DSP/SVFilter.cpp b/src/DSP/SVFilter.cpp
@@ -26,6 +26,12 @@
namespace zyn {
+enum FilterInterpolationType {
+ INTERPOLATE_EXTREME = 0x01,
+ INTERPOLATE_NON_ZERO,
+ INTERPOLATE_NONE
+};
+
SVFilter::SVFilter(unsigned char Ftype, float Ffreq, float Fq,
unsigned char Fstages, unsigned int srate, int bufsize)
:Filter(srate, bufsize),
@@ -33,15 +39,15 @@ SVFilter::SVFilter(unsigned char Ftype, float Ffreq, float Fq,
stages(Fstages),
freq(Ffreq),
q(Fq),
- gain(1.0f)
+ gain(1.0f),
+ needsinterpolation(INTERPOLATE_NONE),
+ firsttime(true)
{
if(stages >= MAX_FILTER_STAGES)
stages = MAX_FILTER_STAGES;
outgain = 1.0f;
cleanup();
setfreq_and_q(Ffreq, Fq);
- freq_smoothing.reset(Ffreq);
- freq_smoothing.sample_rate(bufsize);
}
SVFilter::~SVFilter()
@@ -51,6 +57,8 @@ void SVFilter::cleanup()
{
for(int i = 0; i < MAX_FILTER_STAGES + 1; ++i)
st[i].low = st[i].high = st[i].band = st[i].notch = 0.0f;
+ oldabovenq = false;
+ abovenq = false;
}
SVFilter::response::response(float b0, float b1, float b2,
@@ -119,8 +127,26 @@ void SVFilter::setfreq(float frequency)
if(rap < 1.0f)
rap = 1.0f / rap;
+ oldabovenq = abovenq;
+ abovenq = frequency > (samplerate_f / 2 - 500.0f);
+
+ bool nyquistthresh = (abovenq ^ oldabovenq);
+
+ //if the frequency is changed fast, it needs interpolation
+ if((rap > 3.0f) || nyquistthresh) { //(now, filter and coefficients backup)
+ if(!firsttime)
+ needsinterpolation = INTERPOLATE_EXTREME;
+ ipar = par;
+ } else if(rap != 1.0) {
+ if (!firsttime)
+ needsinterpolation = INTERPOLATE_NON_ZERO;
+ ipar = par;
+ } else {
+ needsinterpolation = INTERPOLATE_NONE;
+ }
freq = frequency;
computefiltercoefs();
+ firsttime = false;
}
void SVFilter::setfreq_and_q(float frequency, float q_)
@@ -178,6 +204,21 @@ float *SVFilter::getfilteroutfortype(SVFilter::fstage &x) {
return out;
}
+void SVFilter::singlefilterout_with_par_interpolation(float *smp, fstage &x, parameters &par1, parameters &par2)
+{
+ float *out = getfilteroutfortype(x);
+ for(int i = 0; i < buffersize; ++i) {
+ float p = i / buffersize_f;
+ float f = par1.f + (par2.f - par1.f) * p;
+ float q = par1.q + (par2.q - par1.q) * p;
+ float q_sqrt = sqrtf(q);
+ x.low = x.low + f * x.band;
+ x.high = q_sqrt * smp[i] - x.low - q * x.band;
+ x.band = f * x.high + x.band;
+ x.notch = x.high + x.low;
+ smp[i] = *out;
+ }
+}
// simplifying the responses
// xl = xl*z(-1) + pf*xb*z(-1)
@@ -191,7 +232,7 @@ float *SVFilter::getfilteroutfortype(SVFilter::fstage &x) {
-void SVFilter::singlefilterout(float *smp, SVFilter::fstage &x, SVFilter::parameters &par, int buffersize )
+void SVFilter::singlefilterout(float *smp, SVFilter::fstage &x, SVFilter::parameters &par)
{
float *out = getfilteroutfortype(x);
for(int i = 0; i < buffersize; ++i) {
@@ -205,28 +246,24 @@ void SVFilter::singlefilterout(float *smp, SVFilter::fstage &x, SVFilter::parame
void SVFilter::filterout(float *smp)
{
- assert((buffersize % 8) == 0);
-
- float freqbuf[buffersize];
-
- if ( freq_smoothing.apply( freqbuf, buffersize, freq ) )
- {
- /* 8 sample chunks seems to work OK for AnalogFilter, so do that here too. */
- for ( int i = 0; i < buffersize; i += 8 )
- {
- freq = freqbuf[i];
- computefiltercoefs();
-
- for(int j = 0; j < stages + 1; ++j)
- singlefilterout(smp + i, st[j], par, 8 );
- }
-
- freq = freqbuf[buffersize - 1];
- computefiltercoefs();
+ if (needsinterpolation == INTERPOLATE_EXTREME) {
+ float ismp[buffersize];
+ for(int i = 0; i < stages + 1; ++i)
+ singlefilterout(smp, st[i], par);
+ memcpy(ismp, smp, bufferbytes);
+ for(int i = 0; i < stages + 1; ++i)
+ singlefilterout(ismp, st[i], ipar);
+ for(int i = 0; i < buffersize; ++i) {
+ float x = i / buffersize_f;
+ smp[i] = ismp[i] * (1.0f - x) + smp[i] * x;
+ }
+ } else if (needsinterpolation == INTERPOLATE_NON_ZERO) {
+ for(int i = 0; i < stages + 1; ++i)
+ singlefilterout_with_par_interpolation(smp, st[i], ipar, par);
+ } else {
+ for(int i = 0; i < stages + 1; ++i)
+ singlefilterout(smp, st[i], par);
}
- else
- for(int i = 0; i < stages + 1; ++i)
- singlefilterout(smp, st[i], par, buffersize );
for(int i = 0; i < buffersize; ++i)
smp[i] *= outgain;
diff --git a/src/DSP/SVFilter.h b/src/DSP/SVFilter.h
@@ -16,7 +16,6 @@
#include "../globals.h"
#include "Filter.h"
-#include "Value_Smoothing_Filter.h"
namespace zyn {
@@ -58,16 +57,19 @@ class SVFilter:public Filter
} par, ipar;
float *getfilteroutfortype(SVFilter::fstage &x);
- void singlefilterout(float *smp, fstage &x, parameters &par, int buffersize );
-
- void computefiltercoefs(void);
+ void singlefilterout(float *smp, fstage &x, parameters &par);
+ void singlefilterout_with_par_interpolation(float *smp, fstage &x, parameters &par1, parameters &par2);
+ void computefiltercoefs(void);
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.)
float freq; // Frequency given in Hz
float q; // Q factor (resonance or Q factor)
float gain; // the gain of the filter (if are shelf/peak) filters
- Value_Smoothing_Filter freq_smoothing;
+ bool abovenq, //if the frequency is above the nyquist
+ oldabovenq;
+ int needsinterpolation;
+ bool firsttime;
};
}
