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airwindows/plugins/MacVST/Stonefire/source/StonefireProc.cpp
Christopher Johnson d8efeb53eb Stonefire
2024-04-06 19:54:41 -04:00

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/* ========================================
* Stonefire - Stonefire.h
* Copyright (c) airwindows, Airwindows uses the MIT license
* ======================================== */
#ifndef __Stonefire_H
#include "Stonefire.h"
#endif
void Stonefire::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
VstInt32 inFramesToProcess = sampleFrames; //vst doesn't give us this as a separate variable so we'll make it
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
trebleGainA = trebleGainB; trebleGainB = A*2.0;
midGainA = midGainB; midGainB = B*2.0;
bassGainA = bassGainB; bassGainB = C*2.0;
//simple three band to adjust
double kalman = 1.0-pow(D,2);
//crossover frequency between mid/bass
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;
double temp = (double)sampleFrames/inFramesToProcess;
double trebleGain = (trebleGainA*temp)+(trebleGainB*(1.0-temp));
if (trebleGain > 1.0) trebleGain = pow(trebleGain,3.0+sqrt(overallscale));
if (trebleGain < 1.0) trebleGain = 1.0-pow(1.0-trebleGain,2);
double midGain = (midGainA*temp)+(midGainB*(1.0-temp));
if (midGain > 1.0) midGain *= midGain;
if (midGain < 1.0) midGain = 1.0-pow(1.0-midGain,2);
double bassGain = (bassGainA*temp)+(bassGainB*(1.0-temp));
if (bassGain > 1.0) bassGain *= bassGain;
if (bassGain < 1.0) bassGain = 1.0-pow(1.0-bassGain,2);
//begin Air3L
air[pvSL4] = air[pvAL4] - air[pvAL3]; air[pvSL3] = air[pvAL3] - air[pvAL2];
air[pvSL2] = air[pvAL2] - air[pvAL1]; air[pvSL1] = air[pvAL1] - inputSampleL;
air[accSL3] = air[pvSL4] - air[pvSL3]; air[accSL2] = air[pvSL3] - air[pvSL2];
air[accSL1] = air[pvSL2] - air[pvSL1];
air[acc2SL2] = air[accSL3] - air[accSL2]; air[acc2SL1] = air[accSL2] - air[accSL1];
air[outAL] = -(air[pvAL1] + air[pvSL3] + air[acc2SL2] - ((air[acc2SL2] + air[acc2SL1])*0.5));
air[gainAL] *= 0.5; air[gainAL] += fabs(drySampleL-air[outAL])*0.5;
if (air[gainAL] > 0.3*sqrt(overallscale)) air[gainAL] = 0.3*sqrt(overallscale);
air[pvAL4] = air[pvAL3]; air[pvAL3] = air[pvAL2];
air[pvAL2] = air[pvAL1]; air[pvAL1] = (air[gainAL] * air[outAL]) + drySampleL;
double midL = drySampleL - ((air[outAL]*0.5)+(drySampleL*(0.457-(0.017*overallscale))));
temp = (midL + air[gndavgL])*0.5; air[gndavgL] = midL; midL = temp;
double trebleL = drySampleL-midL;
inputSampleL = midL;
//end Air3L
//begin Air3R
air[pvSR4] = air[pvAR4] - air[pvAR3]; air[pvSR3] = air[pvAR3] - air[pvAR2];
air[pvSR2] = air[pvAR2] - air[pvAR1]; air[pvSR1] = air[pvAR1] - inputSampleR;
air[accSR3] = air[pvSR4] - air[pvSR3]; air[accSR2] = air[pvSR3] - air[pvSR2];
air[accSR1] = air[pvSR2] - air[pvSR1];
air[acc2SR2] = air[accSR3] - air[accSR2]; air[acc2SR1] = air[accSR2] - air[accSR1];
air[outAR] = -(air[pvAR1] + air[pvSR3] + air[acc2SR2] - ((air[acc2SR2] + air[acc2SR1])*0.5));
air[gainAR] *= 0.5; air[gainAR] += fabs(drySampleR-air[outAR])*0.5;
if (air[gainAR] > 0.3*sqrt(overallscale)) air[gainAR] = 0.3*sqrt(overallscale);
air[pvAR4] = air[pvAR3]; air[pvAR3] = air[pvAR2];
air[pvAR2] = air[pvAR1]; air[pvAR1] = (air[gainAR] * air[outAR]) + drySampleR;
double midR = drySampleR - ((air[outAR]*0.5)+(drySampleR*(0.457-(0.017*overallscale))));
temp = (midR + air[gndavgR])*0.5; air[gndavgR] = midR; midR = temp;
double trebleR = drySampleR-midR;
inputSampleR = midR;
//end Air3R
//begin KalmanL
temp = 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(temp-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] += (temp*(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])*temp);
//feed the chain of previous samples
if (kal[prevSampL1] > 1.0) kal[prevSampL1] = 1.0; if (kal[prevSampL1] < -1.0) kal[prevSampL1] = -1.0;
double bassL = kal[kalOutL]*0.777;
midL -= bassL;
//end KalmanL
//begin KalmanR
temp = 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(temp-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] += (temp*(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])*temp);
//feed the chain of previous samples
if (kal[prevSampR1] > 1.0) kal[prevSampR1] = 1.0; if (kal[prevSampR1] < -1.0) kal[prevSampR1] = -1.0;
double bassR = kal[kalOutR]*0.777;
midR -= bassR;
//end KalmanR
inputSampleL = (bassL*bassGain) + (midL*midGain) + (trebleL*trebleGain);
inputSampleR = (bassR*bassGain) + (midR*midGain) + (trebleR*trebleGain);
//applies pan section, and smoothed fader gain
//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 Stonefire::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
VstInt32 inFramesToProcess = sampleFrames; //vst doesn't give us this as a separate variable so we'll make it
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
trebleGainA = trebleGainB; trebleGainB = A*2.0;
midGainA = midGainB; midGainB = B*2.0;
bassGainA = bassGainB; bassGainB = C*2.0;
//simple three band to adjust
double kalman = 1.0-pow(D,2);
//crossover frequency between mid/bass
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;
double temp = (double)sampleFrames/inFramesToProcess;
double trebleGain = (trebleGainA*temp)+(trebleGainB*(1.0-temp));
if (trebleGain > 1.0) trebleGain = pow(trebleGain,3.0+sqrt(overallscale));
if (trebleGain < 1.0) trebleGain = 1.0-pow(1.0-trebleGain,2);
double midGain = (midGainA*temp)+(midGainB*(1.0-temp));
if (midGain > 1.0) midGain *= midGain;
if (midGain < 1.0) midGain = 1.0-pow(1.0-midGain,2);
double bassGain = (bassGainA*temp)+(bassGainB*(1.0-temp));
if (bassGain > 1.0) bassGain *= bassGain;
if (bassGain < 1.0) bassGain = 1.0-pow(1.0-bassGain,2);
//begin Air3L
air[pvSL4] = air[pvAL4] - air[pvAL3]; air[pvSL3] = air[pvAL3] - air[pvAL2];
air[pvSL2] = air[pvAL2] - air[pvAL1]; air[pvSL1] = air[pvAL1] - inputSampleL;
air[accSL3] = air[pvSL4] - air[pvSL3]; air[accSL2] = air[pvSL3] - air[pvSL2];
air[accSL1] = air[pvSL2] - air[pvSL1];
air[acc2SL2] = air[accSL3] - air[accSL2]; air[acc2SL1] = air[accSL2] - air[accSL1];
air[outAL] = -(air[pvAL1] + air[pvSL3] + air[acc2SL2] - ((air[acc2SL2] + air[acc2SL1])*0.5));
air[gainAL] *= 0.5; air[gainAL] += fabs(drySampleL-air[outAL])*0.5;
if (air[gainAL] > 0.3*sqrt(overallscale)) air[gainAL] = 0.3*sqrt(overallscale);
air[pvAL4] = air[pvAL3]; air[pvAL3] = air[pvAL2];
air[pvAL2] = air[pvAL1]; air[pvAL1] = (air[gainAL] * air[outAL]) + drySampleL;
double midL = drySampleL - ((air[outAL]*0.5)+(drySampleL*(0.457-(0.017*overallscale))));
temp = (midL + air[gndavgL])*0.5; air[gndavgL] = midL; midL = temp;
double trebleL = drySampleL-midL;
inputSampleL = midL;
//end Air3L
//begin Air3R
air[pvSR4] = air[pvAR4] - air[pvAR3]; air[pvSR3] = air[pvAR3] - air[pvAR2];
air[pvSR2] = air[pvAR2] - air[pvAR1]; air[pvSR1] = air[pvAR1] - inputSampleR;
air[accSR3] = air[pvSR4] - air[pvSR3]; air[accSR2] = air[pvSR3] - air[pvSR2];
air[accSR1] = air[pvSR2] - air[pvSR1];
air[acc2SR2] = air[accSR3] - air[accSR2]; air[acc2SR1] = air[accSR2] - air[accSR1];
air[outAR] = -(air[pvAR1] + air[pvSR3] + air[acc2SR2] - ((air[acc2SR2] + air[acc2SR1])*0.5));
air[gainAR] *= 0.5; air[gainAR] += fabs(drySampleR-air[outAR])*0.5;
if (air[gainAR] > 0.3*sqrt(overallscale)) air[gainAR] = 0.3*sqrt(overallscale);
air[pvAR4] = air[pvAR3]; air[pvAR3] = air[pvAR2];
air[pvAR2] = air[pvAR1]; air[pvAR1] = (air[gainAR] * air[outAR]) + drySampleR;
double midR = drySampleR - ((air[outAR]*0.5)+(drySampleR*(0.457-(0.017*overallscale))));
temp = (midR + air[gndavgR])*0.5; air[gndavgR] = midR; midR = temp;
double trebleR = drySampleR-midR;
inputSampleR = midR;
//end Air3R
//begin KalmanL
temp = 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(temp-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] += (temp*(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])*temp);
//feed the chain of previous samples
if (kal[prevSampL1] > 1.0) kal[prevSampL1] = 1.0; if (kal[prevSampL1] < -1.0) kal[prevSampL1] = -1.0;
double bassL = kal[kalOutL]*0.777;
midL -= bassL;
//end KalmanL
//begin KalmanR
temp = 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(temp-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] += (temp*(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])*temp);
//feed the chain of previous samples
if (kal[prevSampR1] > 1.0) kal[prevSampR1] = 1.0; if (kal[prevSampR1] < -1.0) kal[prevSampR1] = -1.0;
double bassR = kal[kalOutR]*0.777;
midR -= bassR;
//end KalmanR
inputSampleL = (bassL*bassGain) + (midL*midGain) + (trebleL*trebleGain);
inputSampleR = (bassR*bassGain) + (midR*midGain) + (trebleR*trebleGain);
//applies pan section, and smoothed fader gain
//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++;
}
}
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