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airwindows/plugins/WinVST/Console8ChannelHype/Console8ChannelHypeProc.cpp
Christopher Johnson f06250c082 Clarifying Licensing in boilerplate code generation
2022-11-21 09:20:21 -05:00

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/* ========================================
* Console8ChannelHype - Console8ChannelHype.h
* Copyright (c) 2016 airwindows, Airwindows uses the MIT license
* ======================================== */
#ifndef __Console8ChannelHype_H
#include "Console8ChannelHype.h"
#endif
void Console8ChannelHype::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
double iirAmountA = 12.66/getSampleRate();
//this is our distributed unusual highpass, which is
//adding subtle harmonics to the really deep stuff to define it
if (fabs(iirAL)<1.18e-37) iirAL = 0.0;
if (fabs(iirBL)<1.18e-37) iirBL = 0.0;
if (fabs(iirAR)<1.18e-37) iirAR = 0.0;
if (fabs(iirBR)<1.18e-37) iirBR = 0.0;
//catch denormals early and only check once per buffer
if (getSampleRate() > 49000.0) hsr = true;
else hsr = false;
fix[fix_freq] = 24000.0 / getSampleRate();
fix[fix_reso] = 0.76352112;
double K = tan(M_PI * fix[fix_freq]); //lowpass
double norm = 1.0 / (1.0 + K / fix[fix_reso] + K * K);
fix[fix_a0] = K * K * norm;
fix[fix_a1] = 2.0 * fix[fix_a0];
fix[fix_a2] = fix[fix_a0];
fix[fix_b1] = 2.0 * (K * K - 1.0) * norm;
fix[fix_b2] = (1.0 - K / fix[fix_reso] + K * K) * norm;
//this is the fixed biquad distributed anti-aliasing filter
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
cycleEnd = floor(overallscale);
if (cycleEnd < 1) cycleEnd = 1;
if (cycleEnd == 3) cycleEnd = 4;
if (cycleEnd > 4) cycleEnd = 4;
//this is going to be 2 for 88.1 or 96k, 4 for 176 or 192k
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;
iirAL = (iirAL * (1.0 - iirAmountA)) + (inputSampleL * iirAmountA);
double iirAmountBL = fabs(iirAL)+0.00001;
iirBL = (iirBL * (1.0 - iirAmountBL)) + (iirAL * iirAmountBL);
inputSampleL -= iirBL;
iirAR = (iirAR * (1.0 - iirAmountA)) + (inputSampleR * iirAmountA);
double iirAmountBR = fabs(iirAR)+0.00001;
iirBR = (iirBR * (1.0 - iirAmountBR)) + (iirAR * iirAmountBR);
inputSampleR -= iirBR;
//Console8 highpass
if (cycleEnd == 4) {
softL[8] = softL[7]; softL[7] = softL[6]; softL[6] = softL[5]; softL[5] = softL[4];
softL[4] = softL[3]; softL[3] = softL[2]; softL[2] = softL[1]; softL[1] = softL[0];
softL[0] = inputSampleL;
softR[8] = softR[7]; softR[7] = softR[6]; softR[6] = softR[5]; softR[5] = softR[4];
softR[4] = softR[3]; softR[3] = softR[2]; softR[2] = softR[1]; softR[1] = softR[0];
softR[0] = inputSampleR;
}
if (cycleEnd == 2) {
softL[8] = softL[6]; softL[6] = softL[4];
softL[4] = softL[2]; softL[2] = softL[0];
softL[0] = inputSampleL;
softR[8] = softR[6]; softR[6] = softR[4];
softR[4] = softR[2]; softR[2] = softR[0];
softR[0] = inputSampleR;
}
if (cycleEnd == 1) {
softL[8] = softL[4];
softL[4] = softL[0];
softL[0] = inputSampleL;
softR[8] = softR[4];
softR[4] = softR[0];
softR[0] = inputSampleR;
}
softL[9] = ((softL[0]-softL[4])-(softL[4]-softL[8]));
if (softL[9] < -1.57079633) softL[9] = -1.57079633;
if (softL[9] > 1.57079633) softL[9] = 1.57079633;
inputSampleL = softL[8]+(sin(softL[9])*0.61803398);
softR[9] = ((softR[0]-softR[4])-(softR[4]-softR[8]));
if (softR[9] < -1.57079633) softR[9] = -1.57079633;
if (softR[9] > 1.57079633) softR[9] = 1.57079633;
inputSampleR = softR[8]+(sin(softR[9])*0.61803398);
//treble softening effect ended up being an aural exciter
if (hsr){
double outSample = (inputSampleL * fix[fix_a0]) + fix[fix_sL1];
fix[fix_sL1] = (inputSampleL * fix[fix_a1]) - (outSample * fix[fix_b1]) + fix[fix_sL2];
fix[fix_sL2] = (inputSampleL * fix[fix_a2]) - (outSample * fix[fix_b2]);
inputSampleL = outSample;
outSample = (inputSampleR * fix[fix_a0]) + fix[fix_sR1];
fix[fix_sR1] = (inputSampleR * fix[fix_a1]) - (outSample * fix[fix_b1]) + fix[fix_sR2];
fix[fix_sR2] = (inputSampleR * fix[fix_a2]) - (outSample * fix[fix_b2]);
inputSampleR = outSample;
} //fixed biquad filtering ultrasonics
//we can go directly into the first distortion stage of ChannelOut
//with a filtered signal, so its biquad is between stages
//on the input channel we have direct signal, not Console8 decode
//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 Console8ChannelHype::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
double iirAmountA = 12.66/getSampleRate();
//this is our distributed unusual highpass, which is
//adding subtle harmonics to the really deep stuff to define it
if (fabs(iirAL)<1.18e-37) iirAL = 0.0;
if (fabs(iirBL)<1.18e-37) iirBL = 0.0;
if (fabs(iirAR)<1.18e-37) iirAR = 0.0;
if (fabs(iirBR)<1.18e-37) iirBR = 0.0;
//catch denormals early and only check once per buffer
if (getSampleRate() > 49000.0) hsr = true;
else hsr = false;
fix[fix_freq] = 24000.0 / getSampleRate();
fix[fix_reso] = 0.76352112;
double K = tan(M_PI * fix[fix_freq]); //lowpass
double norm = 1.0 / (1.0 + K / fix[fix_reso] + K * K);
fix[fix_a0] = K * K * norm;
fix[fix_a1] = 2.0 * fix[fix_a0];
fix[fix_a2] = fix[fix_a0];
fix[fix_b1] = 2.0 * (K * K - 1.0) * norm;
fix[fix_b2] = (1.0 - K / fix[fix_reso] + K * K) * norm;
//this is the fixed biquad distributed anti-aliasing filter
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
cycleEnd = floor(overallscale);
if (cycleEnd < 1) cycleEnd = 1;
if (cycleEnd == 3) cycleEnd = 4;
if (cycleEnd > 4) cycleEnd = 4;
//this is going to be 2 for 88.1 or 96k, 4 for 176 or 192k
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;
iirAL = (iirAL * (1.0 - iirAmountA)) + (inputSampleL * iirAmountA);
double iirAmountBL = fabs(iirAL)+0.00001;
iirBL = (iirBL * (1.0 - iirAmountBL)) + (iirAL * iirAmountBL);
inputSampleL -= iirBL;
iirAR = (iirAR * (1.0 - iirAmountA)) + (inputSampleR * iirAmountA);
double iirAmountBR = fabs(iirAR)+0.00001;
iirBR = (iirBR * (1.0 - iirAmountBR)) + (iirAR * iirAmountBR);
inputSampleR -= iirBR;
//Console8 highpass
if (cycleEnd == 4) {
softL[8] = softL[7]; softL[7] = softL[6]; softL[6] = softL[5]; softL[5] = softL[4];
softL[4] = softL[3]; softL[3] = softL[2]; softL[2] = softL[1]; softL[1] = softL[0];
softL[0] = inputSampleL;
softR[8] = softR[7]; softR[7] = softR[6]; softR[6] = softR[5]; softR[5] = softR[4];
softR[4] = softR[3]; softR[3] = softR[2]; softR[2] = softR[1]; softR[1] = softR[0];
softR[0] = inputSampleR;
}
if (cycleEnd == 2) {
softL[8] = softL[6]; softL[6] = softL[4];
softL[4] = softL[2]; softL[2] = softL[0];
softL[0] = inputSampleL;
softR[8] = softR[6]; softR[6] = softR[4];
softR[4] = softR[2]; softR[2] = softR[0];
softR[0] = inputSampleR;
}
if (cycleEnd == 1) {
softL[8] = softL[4];
softL[4] = softL[0];
softL[0] = inputSampleL;
softR[8] = softR[4];
softR[4] = softR[0];
softR[0] = inputSampleR;
}
softL[9] = ((softL[0]-softL[4])-(softL[4]-softL[8]));
if (softL[9] < -1.57079633) softL[9] = -1.57079633;
if (softL[9] > 1.57079633) softL[9] = 1.57079633;
inputSampleL = softL[8]+(sin(softL[9])*0.61803398);
softR[9] = ((softR[0]-softR[4])-(softR[4]-softR[8]));
if (softR[9] < -1.57079633) softR[9] = -1.57079633;
if (softR[9] > 1.57079633) softR[9] = 1.57079633;
inputSampleR = softR[8]+(sin(softR[9])*0.61803398);
//treble softening effect ended up being an aural exciter
if (hsr){
double outSample = (inputSampleL * fix[fix_a0]) + fix[fix_sL1];
fix[fix_sL1] = (inputSampleL * fix[fix_a1]) - (outSample * fix[fix_b1]) + fix[fix_sL2];
fix[fix_sL2] = (inputSampleL * fix[fix_a2]) - (outSample * fix[fix_b2]);
inputSampleL = outSample;
outSample = (inputSampleR * fix[fix_a0]) + fix[fix_sR1];
fix[fix_sR1] = (inputSampleR * fix[fix_a1]) - (outSample * fix[fix_b1]) + fix[fix_sR2];
fix[fix_sR2] = (inputSampleR * fix[fix_a2]) - (outSample * fix[fix_b2]);
inputSampleR = outSample;
} //fixed biquad filtering ultrasonics
//we can go directly into the first distortion stage of ChannelOut
//with a filtered signal, so its biquad is between stages
//on the input channel we have direct signal, not Console8 decode
//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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