/* ========================================
* GuitarConditioner - GuitarConditioner.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __GuitarConditioner_H
#include "GuitarConditioner.h"
#endif
void GuitarConditioner::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 inputSampleL;
long double inputSampleR;
long double trebleL;
long double bassL;
long double trebleR;
long double bassR;
double iirTreble = 0.287496/overallscale; //tight is -1
double iirBass = 0.085184/overallscale; //tight is 1
iirTreble += iirTreble;
iirBass += iirBass; //simple double when tight is -1 or 1
double tightBass = 0.6666666666;
double tightTreble = -0.3333333333;
double offset;
double clamp;
double threshTreble = 0.0081/overallscale;
double threshBass = 0.0256/overallscale;
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.
}
trebleL = bassL = inputSampleL;
trebleR = bassR = inputSampleR;
trebleL += trebleL; //+3dB on treble as the highpass is higher
trebleR += trebleR; //+3dB on treble as the highpass is higher
offset = (1 + tightTreble) + ((1-fabs(trebleL))*tightTreble); //treble HP
if (offset < 0) offset = 0;
if (offset > 1) offset = 1; //made offset for HP
if (fpFlip) {
iirSampleTAL = (iirSampleTAL * (1 - (offset * iirTreble))) + (trebleL * (offset * iirTreble));
trebleL = trebleL - iirSampleTAL;
} else {
iirSampleTBL = (iirSampleTBL * (1 - (offset * iirTreble))) + (trebleL * (offset * iirTreble));
trebleL = trebleL - iirSampleTBL;
} //done trebleL HP
offset = (1 + tightTreble) + ((1-fabs(trebleR))*tightTreble); //treble HP
if (offset < 0) offset = 0;
if (offset > 1) offset = 1; //made offset for HP
if (fpFlip) {
iirSampleTAR = (iirSampleTAR * (1 - (offset * iirTreble))) + (trebleR * (offset * iirTreble));
trebleR = trebleR - iirSampleTAR;
} else {
iirSampleTBR = (iirSampleTBR * (1 - (offset * iirTreble))) + (trebleR * (offset * iirTreble));
trebleR = trebleR - iirSampleTBR;
} //done trebleR HP
offset = (1 - tightBass) + (fabs(bassL)*tightBass); //bass HP
if (offset < 0) offset = 0;
if (offset > 1) offset = 1;
if (fpFlip) {
iirSampleBAL = (iirSampleBAL * (1 - (offset * iirBass))) + (bassL * (offset * iirBass));
bassL = bassL - iirSampleBAL;
} else {
iirSampleBBL = (iirSampleBBL * (1 - (offset * iirBass))) + (bassL * (offset * iirBass));
bassL = bassL - iirSampleBBL;
} //done bassL HP
offset = (1 - tightBass) + (fabs(bassR)*tightBass); //bass HP
if (offset < 0) offset = 0;
if (offset > 1) offset = 1;
if (fpFlip) {
iirSampleBAR = (iirSampleBAR * (1 - (offset * iirBass))) + (bassR * (offset * iirBass));
bassR = bassR - iirSampleBAR;
} else {
iirSampleBBR = (iirSampleBBR * (1 - (offset * iirBass))) + (bassR * (offset * iirBass));
bassR = bassR - iirSampleBBR;
} //done bassR HP
inputSampleL = trebleL;
clamp = inputSampleL - lastSampleTL;
if (clamp > threshTreble)
trebleL = lastSampleTL + threshTreble;
if (-clamp > threshTreble)
trebleL = lastSampleTL - threshTreble;
lastSampleTL = trebleL; //trebleL slew
inputSampleR = trebleR;
clamp = inputSampleR - lastSampleTR;
if (clamp > threshTreble)
trebleR = lastSampleTR + threshTreble;
if (-clamp > threshTreble)
trebleR = lastSampleTR - threshTreble;
lastSampleTR = trebleR; //trebleR slew
inputSampleL = bassL;
clamp = inputSampleL - lastSampleBL;
if (clamp > threshBass)
bassL = lastSampleBL + threshBass;
if (-clamp > threshBass)
bassL = lastSampleBL - threshBass;
lastSampleBL = bassL; //bassL slew
inputSampleR = bassR;
clamp = inputSampleR - lastSampleBR;
if (clamp > threshBass)
bassR = lastSampleBR + threshBass;
if (-clamp > threshBass)
bassR = lastSampleBR - threshBass;
lastSampleBR = bassR; //bassR slew
inputSampleL = trebleL + bassL; //final merge
inputSampleR = trebleR + bassR; //final merge
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 GuitarConditioner::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 inputSampleL;
long double inputSampleR;
long double trebleL;
long double bassL;
long double trebleR;
long double bassR;
double iirTreble = 0.287496/overallscale; //tight is -1
double iirBass = 0.085184/overallscale; //tight is 1
iirTreble += iirTreble;
iirBass += iirBass; //simple double when tight is -1 or 1
double tightBass = 0.6666666666;
double tightTreble = -0.3333333333;
double offset;
double clamp;
double threshTreble = 0.0081/overallscale;
double threshBass = 0.0256/overallscale;
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.
}
trebleL = bassL = inputSampleL;
trebleR = bassR = inputSampleR;
trebleL += trebleL; //+3dB on treble as the highpass is higher
trebleR += trebleR; //+3dB on treble as the highpass is higher
offset = (1 + tightTreble) + ((1-fabs(trebleL))*tightTreble); //treble HP
if (offset < 0) offset = 0;
if (offset > 1) offset = 1; //made offset for HP
if (fpFlip) {
iirSampleTAL = (iirSampleTAL * (1 - (offset * iirTreble))) + (trebleL * (offset * iirTreble));
trebleL = trebleL - iirSampleTAL;
} else {
iirSampleTBL = (iirSampleTBL * (1 - (offset * iirTreble))) + (trebleL * (offset * iirTreble));
trebleL = trebleL - iirSampleTBL;
} //done trebleL HP
offset = (1 + tightTreble) + ((1-fabs(trebleR))*tightTreble); //treble HP
if (offset < 0) offset = 0;
if (offset > 1) offset = 1; //made offset for HP
if (fpFlip) {
iirSampleTAR = (iirSampleTAR * (1 - (offset * iirTreble))) + (trebleR * (offset * iirTreble));
trebleR = trebleR - iirSampleTAR;
} else {
iirSampleTBR = (iirSampleTBR * (1 - (offset * iirTreble))) + (trebleR * (offset * iirTreble));
trebleR = trebleR - iirSampleTBR;
} //done trebleR HP
offset = (1 - tightBass) + (fabs(bassL)*tightBass); //bass HP
if (offset < 0) offset = 0;
if (offset > 1) offset = 1;
if (fpFlip) {
iirSampleBAL = (iirSampleBAL * (1 - (offset * iirBass))) + (bassL * (offset * iirBass));
bassL = bassL - iirSampleBAL;
} else {
iirSampleBBL = (iirSampleBBL * (1 - (offset * iirBass))) + (bassL * (offset * iirBass));
bassL = bassL - iirSampleBBL;
} //done bassL HP
offset = (1 - tightBass) + (fabs(bassR)*tightBass); //bass HP
if (offset < 0) offset = 0;
if (offset > 1) offset = 1;
if (fpFlip) {
iirSampleBAR = (iirSampleBAR * (1 - (offset * iirBass))) + (bassR * (offset * iirBass));
bassR = bassR - iirSampleBAR;
} else {
iirSampleBBR = (iirSampleBBR * (1 - (offset * iirBass))) + (bassR * (offset * iirBass));
bassR = bassR - iirSampleBBR;
} //done bassR HP
inputSampleL = trebleL;
clamp = inputSampleL - lastSampleTL;
if (clamp > threshTreble)
trebleL = lastSampleTL + threshTreble;
if (-clamp > threshTreble)
trebleL = lastSampleTL - threshTreble;
lastSampleTL = trebleL; //trebleL slew
inputSampleR = trebleR;
clamp = inputSampleR - lastSampleTR;
if (clamp > threshTreble)
trebleR = lastSampleTR + threshTreble;
if (-clamp > threshTreble)
trebleR = lastSampleTR - threshTreble;
lastSampleTR = trebleR; //trebleR slew
inputSampleL = bassL;
clamp = inputSampleL - lastSampleBL;
if (clamp > threshBass)
bassL = lastSampleBL + threshBass;
if (-clamp > threshBass)
bassL = lastSampleBL - threshBass;
lastSampleBL = bassL; //bassL slew
inputSampleR = bassR;
clamp = inputSampleR - lastSampleBR;
if (clamp > threshBass)
bassR = lastSampleBR + threshBass;
if (-clamp > threshBass)
bassR = lastSampleBR - threshBass;
lastSampleBR = bassR; //bassR slew
inputSampleL = trebleL + bassL; //final merge
inputSampleR = trebleR + bassR; //final merge
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++;
}
}
