/* ========================================
* Hermepass - Hermepass.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __Hermepass_H
#include "Hermepass.h"
#endif
void Hermepass::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();
long double fpOld = 0.618033988749894848204586; //golden ratio!
long double fpNew = 1.0 - fpOld;
double rangescale = 0.1 / overallscale;
double cutoff = pow(A,3);
double slope = pow(B,3) * 6.0;
double newA = cutoff * rangescale;
double newB = newA; //other part of interleaved IIR is the same
double newC = cutoff * rangescale; //first extra pole is the same
cutoff = (cutoff * fpOld) + (0.00001 * fpNew);
double newD = cutoff * rangescale;
cutoff = (cutoff * fpOld) + (0.00001 * fpNew);
double newE = cutoff * rangescale;
cutoff = (cutoff * fpOld) + (0.00001 * fpNew);
double newF = cutoff * rangescale;
cutoff = (cutoff * fpOld) + (0.00001 * fpNew);
double newG = cutoff * rangescale;
cutoff = (cutoff * fpOld) + (0.00001 * fpNew);
double newH = cutoff * rangescale;
//converge toward the unvarying fixed cutoff value
double oldA = 1.0 - newA;
double oldB = 1.0 - newB;
double oldC = 1.0 - newC;
double oldD = 1.0 - newD;
double oldE = 1.0 - newE;
double oldF = 1.0 - newF;
double oldG = 1.0 - newG;
double oldH = 1.0 - newH;
double polesC;
double polesD;
double polesE;
double polesF;
double polesG;
double polesH;
polesC = slope; if (slope > 1.0) polesC = 1.0; slope -= 1.0; if (slope < 0.0) slope = 0.0;
polesD = slope; if (slope > 1.0) polesD = 1.0; slope -= 1.0; if (slope < 0.0) slope = 0.0;
polesE = slope; if (slope > 1.0) polesE = 1.0; slope -= 1.0; if (slope < 0.0) slope = 0.0;
polesF = slope; if (slope > 1.0) polesF = 1.0; slope -= 1.0; if (slope < 0.0) slope = 0.0;
polesG = slope; if (slope > 1.0) polesG = 1.0; slope -= 1.0; if (slope < 0.0) slope = 0.0;
polesH = slope; if (slope > 1.0) polesH = 1.0; slope -= 1.0; if (slope < 0.0) slope = 0.0;
//each one will either be 0.0, the fractional slope value, or 1
long double inputSampleL;
long double inputSampleR;
double tempSampleL;
double tempSampleR;
double correction;
while (--sampleFrames >= 0)
{
inputSampleL = *in1;
inputSampleR = *in2;
if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
static int noisesource = 0;
//this declares a variable before anything else is compiled. It won't keep assigning
//it to 0 for every sample, it's as if the declaration doesn't exist in this context,
//but it lets me add this denormalization fix in a single place rather than updating
//it in three different locations. The variable isn't thread-safe but this is only
//a random seed and we can share it with whatever.
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleL = applyresidue;
}
if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
static int noisesource = 0;
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleR = applyresidue;
//this denormalization routine produces a white noise at -300 dB which the noise
//shaping will interact with to produce a bipolar output, but the noise is actually
//all positive. That should stop any variables from going denormal, and the routine
//only kicks in if digital black is input. As a final touch, if you save to 24-bit
//the silence will return to being digital black again.
}
tempSampleL = inputSampleL;
tempSampleR = inputSampleR;
//begin L channel
if (fpFlip) {
iirAL = (iirAL * oldA) + (tempSampleL * newA); tempSampleL -= iirAL; correction = iirAL;
} else {
iirBL = (iirBL * oldB) + (tempSampleL * newB); tempSampleL -= iirBL; correction = iirBL;
}
iirCL = (iirCL * oldC) + (tempSampleL * newC); tempSampleL -= iirCL;
iirDL = (iirDL * oldD) + (tempSampleL * newD); tempSampleL -= iirDL;
iirEL = (iirEL * oldE) + (tempSampleL * newE); tempSampleL -= iirEL;
iirFL = (iirFL * oldF) + (tempSampleL * newF); tempSampleL -= iirFL;
iirGL = (iirGL * oldG) + (tempSampleL * newG); tempSampleL -= iirGL;
iirHL = (iirHL * oldH) + (tempSampleL * newH); tempSampleL -= iirHL;
//set up all the iir filters in case they are used
if (polesC == 1.0) correction += iirCL; if (polesC > 0.0 && polesC < 1.0) correction += (iirCL * polesC);
if (polesD == 1.0) correction += iirDL; if (polesD > 0.0 && polesD < 1.0) correction += (iirDL * polesD);
if (polesE == 1.0) correction += iirEL; if (polesE > 0.0 && polesE < 1.0) correction += (iirEL * polesE);
if (polesF == 1.0) correction += iirFL; if (polesF > 0.0 && polesF < 1.0) correction += (iirFL * polesF);
if (polesG == 1.0) correction += iirGL; if (polesG > 0.0 && polesG < 1.0) correction += (iirGL * polesG);
if (polesH == 1.0) correction += iirHL; if (polesH > 0.0 && polesH < 1.0) correction += (iirHL * polesH);
//each of these are added directly if they're fully engaged,
//multiplied by 0-1 if they are the interpolated one, or skipped if they are beyond the interpolated one.
//the purpose is to do all the math at the floating point exponent nearest to the tiny value in use.
//also, it's formatted that way to easily substitute the next variable: this could be written as a loop
//with everything an array value. However, this makes just as much sense for this few poles.
inputSampleL -= correction;
//end L channel
//begin R channel
if (fpFlip) {
iirAR = (iirAR * oldA) + (tempSampleR * newA); tempSampleR -= iirAR; correction = iirAR;
} else {
iirBR = (iirBR * oldB) + (tempSampleR * newB); tempSampleR -= iirBR; correction = iirBR;
}
iirCR = (iirCR * oldC) + (tempSampleR * newC); tempSampleR -= iirCR;
iirDR = (iirDR * oldD) + (tempSampleR * newD); tempSampleR -= iirDR;
iirER = (iirER * oldE) + (tempSampleR * newE); tempSampleR -= iirER;
iirFR = (iirFR * oldF) + (tempSampleR * newF); tempSampleR -= iirFR;
iirGR = (iirGR * oldG) + (tempSampleR * newG); tempSampleR -= iirGR;
iirHR = (iirHR * oldH) + (tempSampleR * newH); tempSampleR -= iirHR;
//set up all the iir filters in case they are used
if (polesC == 1.0) correction += iirCR; if (polesC > 0.0 && polesC < 1.0) correction += (iirCR * polesC);
if (polesD == 1.0) correction += iirDR; if (polesD > 0.0 && polesD < 1.0) correction += (iirDR * polesD);
if (polesE == 1.0) correction += iirER; if (polesE > 0.0 && polesE < 1.0) correction += (iirER * polesE);
if (polesF == 1.0) correction += iirFR; if (polesF > 0.0 && polesF < 1.0) correction += (iirFR * polesF);
if (polesG == 1.0) correction += iirGR; if (polesG > 0.0 && polesG < 1.0) correction += (iirGR * polesG);
if (polesH == 1.0) correction += iirHR; if (polesH > 0.0 && polesH < 1.0) correction += (iirHR * polesH);
//each of these are added directly if they're fully engaged,
//multiplied by 0-1 if they are the interpolated one, or skipped if they are beyond the interpolated one.
//the purpose is to do all the math at the floating point exponent nearest to the tiny value in use.
//also, it's formatted that way to easily substitute the next variable: this could be written as a loop
//with everything an array value. However, this makes just as much sense for this few poles.
inputSampleR -= correction;
//end R channel
fpFlip = !fpFlip;
//stereo 32 bit dither, made small and tidy.
int expon; frexpf((float)inputSampleL, &expon);
long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
inputSampleL += (dither-fpNShapeL); fpNShapeL = dither;
frexpf((float)inputSampleR, &expon);
dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
inputSampleR += (dither-fpNShapeR); fpNShapeR = dither;
//end 32 bit dither
*out1 = inputSampleL;
*out2 = inputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
}
void Hermepass::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();
long double fpOld = 0.618033988749894848204586; //golden ratio!
long double fpNew = 1.0 - fpOld;
double rangescale = 0.1 / overallscale;
double cutoff = pow(A,3);
double slope = pow(B,3) * 6.0;
double newA = cutoff * rangescale;
double newB = newA; //other part of interleaved IIR is the same
double newC = cutoff * rangescale; //first extra pole is the same
cutoff = (cutoff * fpOld) + (0.00001 * fpNew);
double newD = cutoff * rangescale;
cutoff = (cutoff * fpOld) + (0.00001 * fpNew);
double newE = cutoff * rangescale;
cutoff = (cutoff * fpOld) + (0.00001 * fpNew);
double newF = cutoff * rangescale;
cutoff = (cutoff * fpOld) + (0.00001 * fpNew);
double newG = cutoff * rangescale;
cutoff = (cutoff * fpOld) + (0.00001 * fpNew);
double newH = cutoff * rangescale;
//converge toward the unvarying fixed cutoff value
double oldA = 1.0 - newA;
double oldB = 1.0 - newB;
double oldC = 1.0 - newC;
double oldD = 1.0 - newD;
double oldE = 1.0 - newE;
double oldF = 1.0 - newF;
double oldG = 1.0 - newG;
double oldH = 1.0 - newH;
double polesC;
double polesD;
double polesE;
double polesF;
double polesG;
double polesH;
polesC = slope; if (slope > 1.0) polesC = 1.0; slope -= 1.0; if (slope < 0.0) slope = 0.0;
polesD = slope; if (slope > 1.0) polesD = 1.0; slope -= 1.0; if (slope < 0.0) slope = 0.0;
polesE = slope; if (slope > 1.0) polesE = 1.0; slope -= 1.0; if (slope < 0.0) slope = 0.0;
polesF = slope; if (slope > 1.0) polesF = 1.0; slope -= 1.0; if (slope < 0.0) slope = 0.0;
polesG = slope; if (slope > 1.0) polesG = 1.0; slope -= 1.0; if (slope < 0.0) slope = 0.0;
polesH = slope; if (slope > 1.0) polesH = 1.0; slope -= 1.0; if (slope < 0.0) slope = 0.0;
//each one will either be 0.0, the fractional slope value, or 1
long double inputSampleL;
long double inputSampleR;
double tempSampleL;
double tempSampleR;
double correction;
while (--sampleFrames >= 0)
{
inputSampleL = *in1;
inputSampleR = *in2;
if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
static int noisesource = 0;
//this declares a variable before anything else is compiled. It won't keep assigning
//it to 0 for every sample, it's as if the declaration doesn't exist in this context,
//but it lets me add this denormalization fix in a single place rather than updating
//it in three different locations. The variable isn't thread-safe but this is only
//a random seed and we can share it with whatever.
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleL = applyresidue;
}
if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
static int noisesource = 0;
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleR = applyresidue;
//this denormalization routine produces a white noise at -300 dB which the noise
//shaping will interact with to produce a bipolar output, but the noise is actually
//all positive. That should stop any variables from going denormal, and the routine
//only kicks in if digital black is input. As a final touch, if you save to 24-bit
//the silence will return to being digital black again.
}
tempSampleL = inputSampleL;
tempSampleR = inputSampleR;
//begin L channel
if (fpFlip) {
iirAL = (iirAL * oldA) + (tempSampleL * newA); tempSampleL -= iirAL; correction = iirAL;
} else {
iirBL = (iirBL * oldB) + (tempSampleL * newB); tempSampleL -= iirBL; correction = iirBL;
}
iirCL = (iirCL * oldC) + (tempSampleL * newC); tempSampleL -= iirCL;
iirDL = (iirDL * oldD) + (tempSampleL * newD); tempSampleL -= iirDL;
iirEL = (iirEL * oldE) + (tempSampleL * newE); tempSampleL -= iirEL;
iirFL = (iirFL * oldF) + (tempSampleL * newF); tempSampleL -= iirFL;
iirGL = (iirGL * oldG) + (tempSampleL * newG); tempSampleL -= iirGL;
iirHL = (iirHL * oldH) + (tempSampleL * newH); tempSampleL -= iirHL;
//set up all the iir filters in case they are used
if (polesC == 1.0) correction += iirCL; if (polesC > 0.0 && polesC < 1.0) correction += (iirCL * polesC);
if (polesD == 1.0) correction += iirDL; if (polesD > 0.0 && polesD < 1.0) correction += (iirDL * polesD);
if (polesE == 1.0) correction += iirEL; if (polesE > 0.0 && polesE < 1.0) correction += (iirEL * polesE);
if (polesF == 1.0) correction += iirFL; if (polesF > 0.0 && polesF < 1.0) correction += (iirFL * polesF);
if (polesG == 1.0) correction += iirGL; if (polesG > 0.0 && polesG < 1.0) correction += (iirGL * polesG);
if (polesH == 1.0) correction += iirHL; if (polesH > 0.0 && polesH < 1.0) correction += (iirHL * polesH);
//each of these are added directly if they're fully engaged,
//multiplied by 0-1 if they are the interpolated one, or skipped if they are beyond the interpolated one.
//the purpose is to do all the math at the floating point exponent nearest to the tiny value in use.
//also, it's formatted that way to easily substitute the next variable: this could be written as a loop
//with everything an array value. However, this makes just as much sense for this few poles.
inputSampleL -= correction;
//end L channel
//begin R channel
if (fpFlip) {
iirAR = (iirAR * oldA) + (tempSampleR * newA); tempSampleR -= iirAR; correction = iirAR;
} else {
iirBR = (iirBR * oldB) + (tempSampleR * newB); tempSampleR -= iirBR; correction = iirBR;
}
iirCR = (iirCR * oldC) + (tempSampleR * newC); tempSampleR -= iirCR;
iirDR = (iirDR * oldD) + (tempSampleR * newD); tempSampleR -= iirDR;
iirER = (iirER * oldE) + (tempSampleR * newE); tempSampleR -= iirER;
iirFR = (iirFR * oldF) + (tempSampleR * newF); tempSampleR -= iirFR;
iirGR = (iirGR * oldG) + (tempSampleR * newG); tempSampleR -= iirGR;
iirHR = (iirHR * oldH) + (tempSampleR * newH); tempSampleR -= iirHR;
//set up all the iir filters in case they are used
if (polesC == 1.0) correction += iirCR; if (polesC > 0.0 && polesC < 1.0) correction += (iirCR * polesC);
if (polesD == 1.0) correction += iirDR; if (polesD > 0.0 && polesD < 1.0) correction += (iirDR * polesD);
if (polesE == 1.0) correction += iirER; if (polesE > 0.0 && polesE < 1.0) correction += (iirER * polesE);
if (polesF == 1.0) correction += iirFR; if (polesF > 0.0 && polesF < 1.0) correction += (iirFR * polesF);
if (polesG == 1.0) correction += iirGR; if (polesG > 0.0 && polesG < 1.0) correction += (iirGR * polesG);
if (polesH == 1.0) correction += iirHR; if (polesH > 0.0 && polesH < 1.0) correction += (iirHR * polesH);
//each of these are added directly if they're fully engaged,
//multiplied by 0-1 if they are the interpolated one, or skipped if they are beyond the interpolated one.
//the purpose is to do all the math at the floating point exponent nearest to the tiny value in use.
//also, it's formatted that way to easily substitute the next variable: this could be written as a loop
//with everything an array value. However, this makes just as much sense for this few poles.
inputSampleR -= correction;
//end R channel
fpFlip = !fpFlip;
//stereo 64 bit dither, made small and tidy.
int expon; frexp((double)inputSampleL, &expon);
long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
dither /= 536870912.0; //needs this to scale to 64 bit zone
inputSampleL += (dither-fpNShapeL); fpNShapeL = dither;
frexp((double)inputSampleR, &expon);
dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
dither /= 536870912.0; //needs this to scale to 64 bit zone
inputSampleR += (dither-fpNShapeR); fpNShapeR = dither;
//end 64 bit dither
*out1 = inputSampleL;
*out2 = inputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
}
