/* ========================================
* ToneSlant - ToneSlant.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __ToneSlant_H
#include "ToneSlant.h"
#endif
void ToneSlant::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
double inputSampleL;
double inputSampleR;
double correctionSampleL;
double correctionSampleR;
double accumulatorSampleL;
double accumulatorSampleR;
double drySampleL;
double drySampleR;
double overallscale = (A*99.0)+1.0;
double applySlant = (B*2.0)-1.0;
f[0] = 1.0 / overallscale;
//count to f(gain) which will be 0. f(0) is x1
for (int count = 1; count < 102; count++) {
if (count <= overallscale) {
f[count] = (1.0 - (count / overallscale)) / overallscale;
//recalc the filter and don't change the buffer it'll apply to
} else {
bL[count] = 0.0; //blank the unused buffer so when we return to it, no pops
bR[count] = 0.0; //blank the unused buffer so when we return to it, no pops
}
}
while (--sampleFrames >= 0)
{
for (int count = overallscale; count >= 0; count--) {
bL[count+1] = bL[count];
bR[count+1] = bR[count];
}
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.
}
bL[0] = accumulatorSampleL = drySampleL = inputSampleL;
bR[0] = accumulatorSampleR = drySampleR = inputSampleR;
accumulatorSampleL *= f[0];
accumulatorSampleR *= f[0];
for (int count = 1; count < overallscale; count++) {
accumulatorSampleL += (bL[count] * f[count]);
accumulatorSampleR += (bR[count] * f[count]);
}
correctionSampleL = inputSampleL - (accumulatorSampleL*2.0);
correctionSampleR = inputSampleR - (accumulatorSampleR*2.0);
//we're gonna apply the total effect of all these calculations as a single subtract
inputSampleL += (correctionSampleL * applySlant);
inputSampleR += (correctionSampleR * applySlant);
//our one math operation on the input data coming in
//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 ToneSlant::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
double inputSampleL;
double inputSampleR;
double correctionSampleL;
double correctionSampleR;
double accumulatorSampleL;
double accumulatorSampleR;
double drySampleL;
double drySampleR;
double overallscale = (A*99.0)+1.0;
double applySlant = (B*2.0)-1.0;
f[0] = 1.0 / overallscale;
//count to f(gain) which will be 0. f(0) is x1
for (int count = 1; count < 102; count++) {
if (count <= overallscale) {
f[count] = (1.0 - (count / overallscale)) / overallscale;
//recalc the filter and don't change the buffer it'll apply to
} else {
bL[count] = 0.0; //blank the unused buffer so when we return to it, no pops
bR[count] = 0.0; //blank the unused buffer so when we return to it, no pops
}
}
while (--sampleFrames >= 0)
{
for (int count = overallscale; count >= 0; count--) {
bL[count+1] = bL[count];
bR[count+1] = bR[count];
}
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.
}
bL[0] = accumulatorSampleL = drySampleL = inputSampleL;
bR[0] = accumulatorSampleR = drySampleR = inputSampleR;
accumulatorSampleL *= f[0];
accumulatorSampleR *= f[0];
for (int count = 1; count < overallscale; count++) {
accumulatorSampleL += (bL[count] * f[count]);
accumulatorSampleR += (bR[count] * f[count]);
}
correctionSampleL = inputSampleL - (accumulatorSampleL*2.0);
correctionSampleR = inputSampleR - (accumulatorSampleR*2.0);
//we're gonna apply the total effect of all these calculations as a single subtract
inputSampleL += (correctionSampleL * applySlant);
inputSampleR += (correctionSampleR * applySlant);
//our one math operation on the input data coming in
//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++;
}
}
