mirror of
https://github.com/airwindows/airwindows.git
synced 2026-05-16 06:05:55 -06:00
214 lines
9.6 KiB
C++
Executable file
214 lines
9.6 KiB
C++
Executable file
/* ========================================
|
|
* Kalman - Kalman.h
|
|
* Copyright (c) airwindows, Airwindows uses the MIT license
|
|
* ======================================== */
|
|
|
|
#ifndef __Kalman_H
|
|
#include "Kalman.h"
|
|
#endif
|
|
|
|
void Kalman::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
|
|
{
|
|
float* in1 = inputs[0];
|
|
float* in2 = inputs[1];
|
|
float* out1 = outputs[0];
|
|
float* out2 = outputs[1];
|
|
|
|
double overallscale = 1.0;
|
|
overallscale /= 44100.0;
|
|
overallscale *= getSampleRate();
|
|
|
|
double kalman = 1.0-pow(A,2);
|
|
double wet = (B*2.0)-1.0; //inv-dry-wet for highpass
|
|
double dry = 2.0-(B*2.0);
|
|
if (dry > 1.0) dry = 1.0; //full dry for use with inv, to 0.0 at full wet
|
|
|
|
while (--sampleFrames >= 0)
|
|
{
|
|
double inputSampleL = *in1;
|
|
double inputSampleR = *in2;
|
|
if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
|
|
if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
|
|
double drySampleL = inputSampleL;
|
|
double drySampleR = inputSampleR;
|
|
|
|
//begin Kalman Filter
|
|
double dryKal = inputSampleL = inputSampleL*(1.0-kalman)*0.777;
|
|
inputSampleL *= (1.0-kalman);
|
|
//set up gain levels to control the beast
|
|
kal[prevSlewL3] += kal[prevSampL3] - kal[prevSampL2]; kal[prevSlewL3] *= 0.5;
|
|
kal[prevSlewL2] += kal[prevSampL2] - kal[prevSampL1]; kal[prevSlewL2] *= 0.5;
|
|
kal[prevSlewL1] += kal[prevSampL1] - inputSampleL; kal[prevSlewL1] *= 0.5;
|
|
//make slews from each set of samples used
|
|
kal[accSlewL2] += kal[prevSlewL3] - kal[prevSlewL2]; kal[accSlewL2] *= 0.5;
|
|
kal[accSlewL1] += kal[prevSlewL2] - kal[prevSlewL1]; kal[accSlewL1] *= 0.5;
|
|
//differences between slews: rate of change of rate of change
|
|
kal[accSlewL3] += (kal[accSlewL2] - kal[accSlewL1]); kal[accSlewL3] *= 0.5;
|
|
//entering the abyss, what even is this
|
|
kal[kalOutL] += kal[prevSampL1] + kal[prevSlewL2] + kal[accSlewL3]; kal[kalOutL] *= 0.5;
|
|
//resynthesizing predicted result (all iir smoothed)
|
|
kal[kalGainL] += fabs(dryKal-kal[kalOutL])*kalman*8.0; kal[kalGainL] *= 0.5;
|
|
//madness takes its toll. Kalman Gain: how much dry to retain
|
|
if (kal[kalGainL] > kalman*0.5) kal[kalGainL] = kalman*0.5;
|
|
//attempts to avoid explosions
|
|
kal[kalOutL] += (dryKal*(1.0-(0.68+(kalman*0.157))));
|
|
//this is for tuning a really complete cancellation up around Nyquist
|
|
kal[prevSampL3] = kal[prevSampL2];
|
|
kal[prevSampL2] = kal[prevSampL1];
|
|
kal[prevSampL1] = (kal[kalGainL] * kal[kalOutL]) + ((1.0-kal[kalGainL])*dryKal);
|
|
//feed the chain of previous samples
|
|
if (kal[prevSampL1] > 1.0) kal[prevSampL1] = 1.0;
|
|
if (kal[prevSampL1] < -1.0) kal[prevSampL1] = -1.0;
|
|
//end Kalman Filter, except for trim on output
|
|
inputSampleL = (drySampleL*dry)+(kal[kalOutL]*wet*0.777);
|
|
//now the right channel
|
|
dryKal = inputSampleR = inputSampleR*(1.0-kalman)*0.777;
|
|
inputSampleR *= (1.0-kalman);
|
|
//set up gain levels to control the beast
|
|
kal[prevSlewR3] += kal[prevSampR3] - kal[prevSampR2]; kal[prevSlewR3] *= 0.5;
|
|
kal[prevSlewR2] += kal[prevSampR2] - kal[prevSampR1]; kal[prevSlewR2] *= 0.5;
|
|
kal[prevSlewR1] += kal[prevSampR1] - inputSampleR; kal[prevSlewR1] *= 0.5;
|
|
//make slews from each set of samples used
|
|
kal[accSlewR2] += kal[prevSlewR3] - kal[prevSlewR2]; kal[accSlewR2] *= 0.5;
|
|
kal[accSlewR1] += kal[prevSlewR2] - kal[prevSlewR1]; kal[accSlewR1] *= 0.5;
|
|
//differences between slews: rate of change of rate of change
|
|
kal[accSlewR3] += (kal[accSlewR2] - kal[accSlewR1]); kal[accSlewR3] *= 0.5;
|
|
//entering the abyss, what even is this
|
|
kal[kalOutR] += kal[prevSampR1] + kal[prevSlewR2] + kal[accSlewR3]; kal[kalOutR] *= 0.5;
|
|
//resynthesizing predicted result (all iir smoothed)
|
|
kal[kalGainR] += fabs(dryKal-kal[kalOutR])*kalman*8.0; kal[kalGainR] *= 0.5;
|
|
//madness takes its toll. Kalman Gain: how much dry to retain
|
|
if (kal[kalGainR] > kalman*0.5) kal[kalGainR] = kalman*0.5;
|
|
//attempts to avoid explosions
|
|
kal[kalOutR] += (dryKal*(1.0-(0.68+(kalman*0.157))));
|
|
//this is for tuning a really complete cancellation up around Nyquist
|
|
kal[prevSampR3] = kal[prevSampR2];
|
|
kal[prevSampR2] = kal[prevSampR1];
|
|
kal[prevSampR1] = (kal[kalGainR] * kal[kalOutR]) + ((1.0-kal[kalGainR])*dryKal);
|
|
//feed the chain of previous samples
|
|
if (kal[prevSampR1] > 1.0) kal[prevSampR1] = 1.0;
|
|
if (kal[prevSampR1] < -1.0) kal[prevSampR1] = -1.0;
|
|
//end Kalman Filter, except for trim on output
|
|
inputSampleR = (drySampleR*dry)+(kal[kalOutR]*wet*0.777);
|
|
|
|
//begin 32 bit stereo floating point dither
|
|
int expon; frexpf((float)inputSampleL, &expon);
|
|
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
|
|
inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
|
|
frexpf((float)inputSampleR, &expon);
|
|
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
|
|
inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
|
|
//end 32 bit stereo floating point dither
|
|
|
|
*out1 = inputSampleL;
|
|
*out2 = inputSampleR;
|
|
|
|
in1++;
|
|
in2++;
|
|
out1++;
|
|
out2++;
|
|
}
|
|
}
|
|
|
|
void Kalman::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
|
|
{
|
|
double* in1 = inputs[0];
|
|
double* in2 = inputs[1];
|
|
double* out1 = outputs[0];
|
|
double* out2 = outputs[1];
|
|
|
|
double overallscale = 1.0;
|
|
overallscale /= 44100.0;
|
|
overallscale *= getSampleRate();
|
|
|
|
double kalman = 1.0-pow(A,2);
|
|
double wet = (B*2.0)-1.0; //inv-dry-wet for highpass
|
|
double dry = 2.0-(B*2.0);
|
|
if (dry > 1.0) dry = 1.0; //full dry for use with inv, to 0.0 at full wet
|
|
|
|
while (--sampleFrames >= 0)
|
|
{
|
|
double inputSampleL = *in1;
|
|
double inputSampleR = *in2;
|
|
if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
|
|
if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
|
|
double drySampleL = inputSampleL;
|
|
double drySampleR = inputSampleR;
|
|
|
|
//begin Kalman Filter
|
|
double dryKal = inputSampleL = inputSampleL*(1.0-kalman)*0.777;
|
|
inputSampleL *= (1.0-kalman);
|
|
//set up gain levels to control the beast
|
|
kal[prevSlewL3] += kal[prevSampL3] - kal[prevSampL2]; kal[prevSlewL3] *= 0.5;
|
|
kal[prevSlewL2] += kal[prevSampL2] - kal[prevSampL1]; kal[prevSlewL2] *= 0.5;
|
|
kal[prevSlewL1] += kal[prevSampL1] - inputSampleL; kal[prevSlewL1] *= 0.5;
|
|
//make slews from each set of samples used
|
|
kal[accSlewL2] += kal[prevSlewL3] - kal[prevSlewL2]; kal[accSlewL2] *= 0.5;
|
|
kal[accSlewL1] += kal[prevSlewL2] - kal[prevSlewL1]; kal[accSlewL1] *= 0.5;
|
|
//differences between slews: rate of change of rate of change
|
|
kal[accSlewL3] += (kal[accSlewL2] - kal[accSlewL1]); kal[accSlewL3] *= 0.5;
|
|
//entering the abyss, what even is this
|
|
kal[kalOutL] += kal[prevSampL1] + kal[prevSlewL2] + kal[accSlewL3]; kal[kalOutL] *= 0.5;
|
|
//resynthesizing predicted result (all iir smoothed)
|
|
kal[kalGainL] += fabs(dryKal-kal[kalOutL])*kalman*8.0; kal[kalGainL] *= 0.5;
|
|
//madness takes its toll. Kalman Gain: how much dry to retain
|
|
if (kal[kalGainL] > kalman*0.5) kal[kalGainL] = kalman*0.5;
|
|
//attempts to avoid explosions
|
|
kal[kalOutL] += (dryKal*(1.0-(0.68+(kalman*0.157))));
|
|
//this is for tuning a really complete cancellation up around Nyquist
|
|
kal[prevSampL3] = kal[prevSampL2];
|
|
kal[prevSampL2] = kal[prevSampL1];
|
|
kal[prevSampL1] = (kal[kalGainL] * kal[kalOutL]) + ((1.0-kal[kalGainL])*dryKal);
|
|
//feed the chain of previous samples
|
|
if (kal[prevSampL1] > 1.0) kal[prevSampL1] = 1.0;
|
|
if (kal[prevSampL1] < -1.0) kal[prevSampL1] = -1.0;
|
|
//end Kalman Filter, except for trim on output
|
|
inputSampleL = (drySampleL*dry)+(kal[kalOutL]*wet*0.777);
|
|
//now the right channel
|
|
dryKal = inputSampleR = inputSampleR*(1.0-kalman)*0.777;
|
|
inputSampleR *= (1.0-kalman);
|
|
//set up gain levels to control the beast
|
|
kal[prevSlewR3] += kal[prevSampR3] - kal[prevSampR2]; kal[prevSlewR3] *= 0.5;
|
|
kal[prevSlewR2] += kal[prevSampR2] - kal[prevSampR1]; kal[prevSlewR2] *= 0.5;
|
|
kal[prevSlewR1] += kal[prevSampR1] - inputSampleR; kal[prevSlewR1] *= 0.5;
|
|
//make slews from each set of samples used
|
|
kal[accSlewR2] += kal[prevSlewR3] - kal[prevSlewR2]; kal[accSlewR2] *= 0.5;
|
|
kal[accSlewR1] += kal[prevSlewR2] - kal[prevSlewR1]; kal[accSlewR1] *= 0.5;
|
|
//differences between slews: rate of change of rate of change
|
|
kal[accSlewR3] += (kal[accSlewR2] - kal[accSlewR1]); kal[accSlewR3] *= 0.5;
|
|
//entering the abyss, what even is this
|
|
kal[kalOutR] += kal[prevSampR1] + kal[prevSlewR2] + kal[accSlewR3]; kal[kalOutR] *= 0.5;
|
|
//resynthesizing predicted result (all iir smoothed)
|
|
kal[kalGainR] += fabs(dryKal-kal[kalOutR])*kalman*8.0; kal[kalGainR] *= 0.5;
|
|
//madness takes its toll. Kalman Gain: how much dry to retain
|
|
if (kal[kalGainR] > kalman*0.5) kal[kalGainR] = kalman*0.5;
|
|
//attempts to avoid explosions
|
|
kal[kalOutR] += (dryKal*(1.0-(0.68+(kalman*0.157))));
|
|
//this is for tuning a really complete cancellation up around Nyquist
|
|
kal[prevSampR3] = kal[prevSampR2];
|
|
kal[prevSampR2] = kal[prevSampR1];
|
|
kal[prevSampR1] = (kal[kalGainR] * kal[kalOutR]) + ((1.0-kal[kalGainR])*dryKal);
|
|
//feed the chain of previous samples
|
|
if (kal[prevSampR1] > 1.0) kal[prevSampR1] = 1.0;
|
|
if (kal[prevSampR1] < -1.0) kal[prevSampR1] = -1.0;
|
|
//end Kalman Filter, except for trim on output
|
|
inputSampleR = (drySampleR*dry)+(kal[kalOutR]*wet*0.777);
|
|
|
|
//begin 64 bit stereo floating point dither
|
|
//int expon; frexp((double)inputSampleL, &expon);
|
|
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
|
|
//inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
|
|
//frexp((double)inputSampleR, &expon);
|
|
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
|
|
//inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
|
|
//end 64 bit stereo floating point dither
|
|
|
|
*out1 = inputSampleL;
|
|
*out2 = inputSampleR;
|
|
|
|
in1++;
|
|
in2++;
|
|
out1++;
|
|
out2++;
|
|
}
|
|
}
|