/* ========================================
* Noise - Noise.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __Noise_H
#include "Noise.h"
#endif
void Noise::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
double cutoffL;
double cutoffR;
double cutofftarget = (A*3.5);
double rumblecutoff = cutofftarget * 0.005;
double invcutoffL;
double invcutoffR;
double drySampleL;
double drySampleR;
long double inputSampleL;
long double inputSampleR;
double highpass = C*38.0;
int lowcut = floor(highpass);
int dcut;
if (lowcut > 37) {dcut= 1151;}
if (lowcut == 37) {dcut= 1091;}
if (lowcut == 36) {dcut= 1087;}
if (lowcut == 35) {dcut= 1031;}
if (lowcut == 34) {dcut= 1013;}
if (lowcut == 33) {dcut= 971;}
if (lowcut == 32) {dcut= 907;}
if (lowcut == 31) {dcut= 839;}
if (lowcut == 30) {dcut= 797;}
if (lowcut == 29) {dcut= 733;}
if (lowcut == 28) {dcut= 719;}
if (lowcut == 27) {dcut= 673;}
if (lowcut == 26) {dcut= 613;}
if (lowcut == 25) {dcut= 593;}
if (lowcut == 24) {dcut= 541;}
if (lowcut == 23) {dcut= 479;}
if (lowcut == 22) {dcut= 431;}
if (lowcut == 21) {dcut= 419;}
if (lowcut == 20) {dcut= 373;}
if (lowcut == 19) {dcut= 311;}
if (lowcut == 18) {dcut= 293;}
if (lowcut == 17) {dcut= 233;}
if (lowcut == 16) {dcut= 191;}
if (lowcut == 15) {dcut= 173;}
if (lowcut == 14) {dcut= 131;}
if (lowcut == 13) {dcut= 113;}
if (lowcut == 12) {dcut= 71;}
if (lowcut == 11) {dcut= 53;}
if (lowcut == 10) {dcut= 31;}
if (lowcut == 9) {dcut= 27;}
if (lowcut == 8) {dcut= 23;}
if (lowcut == 7) {dcut= 19;}
if (lowcut == 6) {dcut= 17;}
if (lowcut == 5) {dcut= 13;}
if (lowcut == 4) {dcut= 11;}
if (lowcut == 3) {dcut= 7;}
if (lowcut == 2) {dcut= 5;}
if (lowcut < 2) {dcut= 3;}
highpass = B * 22.0;
lowcut = floor(highpass)+1;
double decay = 0.001 - ((1.0-pow(1.0-D,3))*0.001);
if (decay == 0.001) decay = 0.1;
double wet = F;
double dry = 1.0 - wet;
wet *= 0.01; //correct large gain issue
double correctionSample;
double accumulatorSampleL;
double accumulatorSampleR;
double overallscale = (E*9.0)+1.0;
double gain = overallscale;
if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;}
if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;}
if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;}
if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;}
if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;}
if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;}
if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;}
if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;}
if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;}
if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;}
//there, now we have a neat little moving average with remainders
if (overallscale < 1.0) overallscale = 1.0;
f[0] /= overallscale;
f[1] /= overallscale;
f[2] /= overallscale;
f[3] /= overallscale;
f[4] /= overallscale;
f[5] /= overallscale;
f[6] /= overallscale;
f[7] /= overallscale;
f[8] /= overallscale;
f[9] /= overallscale;
//and now it's neatly scaled, too
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.
}
drySampleL = inputSampleL;
drySampleR = inputSampleR;
if (surgeL<fabs(inputSampleL))
{
surgeL += (rand()/(double)RAND_MAX)*(fabs(inputSampleL)-surgeL);
if (surgeL > 1.0) surgeL = 1.0;
}
else
{
surgeL -= ((rand()/(double)RAND_MAX)*(surgeL-fabs(inputSampleL))*decay);
if (surgeL < 0.0) surgeL = 0.0;
}
cutoffL = pow((cutofftarget*surgeL),5);
if (cutoffL > 1.0) cutoffL = 1.0;
invcutoffL = 1.0 - cutoffL;
//set up modified cutoff L
if (surgeR<fabs(inputSampleR))
{
surgeR += (rand()/(double)RAND_MAX)*(fabs(inputSampleR)-surgeR);
if (surgeR > 1.0) surgeR = 1.0;
}
else
{
surgeR -= ((rand()/(double)RAND_MAX)*(surgeR-fabs(inputSampleR))*decay);
if (surgeR < 0.0) surgeR = 0.0;
}
cutoffR = pow((cutofftarget*surgeR),5);
if (cutoffR > 1.0) cutoffR = 1.0;
invcutoffR = 1.0 - cutoffR;
//set up modified cutoff R
flipL = !flipL;
flipR = !flipR;
filterflip = !filterflip;
quadratic -= 1;
if (quadratic < 0)
{
position += 1;
quadratic = position * position;
quadratic = quadratic % 170003; //% is C++ mod operator
quadratic *= quadratic;
quadratic = quadratic % 17011; //% is C++ mod operator
quadratic *= quadratic;
//quadratic = quadratic % 1709; //% is C++ mod operator
//quadratic *= quadratic;
quadratic = quadratic % dcut; //% is C++ mod operator
quadratic *= quadratic;
quadratic = quadratic % lowcut;
//sets density of the centering force
if (noiseAL < 0) {flipL = true;}
else {flipL = false;}
if (noiseAR < 0) {flipR = true;}
else {flipR = false;}
}
if (flipL) noiseAL += (rand()/(double)RAND_MAX);
else noiseAL -= (rand()/(double)RAND_MAX);
if (flipR) noiseAR += (rand()/(double)RAND_MAX);
else noiseAR -= (rand()/(double)RAND_MAX);
if (filterflip)
{
noiseBL *= invcutoffL; noiseBL += (noiseAL*cutoffL);
inputSampleL = noiseBL+noiseCL;
rumbleAL *= (1.0-rumblecutoff);
rumbleAL += (inputSampleL*rumblecutoff);
noiseBR *= invcutoffR; noiseBR += (noiseAR*cutoffR);
inputSampleR = noiseBR+noiseCR;
rumbleAR *= (1.0-rumblecutoff);
rumbleAR += (inputSampleR*rumblecutoff);
}
else
{
noiseCL *= invcutoffL; noiseCL += (noiseAL*cutoffL);
inputSampleL = noiseBL+noiseCL;
rumbleBL *= (1.0-rumblecutoff);
rumbleBL += (inputSampleL*rumblecutoff);
noiseCR *= invcutoffR; noiseCR += (noiseAR*cutoffR);
inputSampleR = noiseBR+noiseCR;
rumbleBR *= (1.0-rumblecutoff);
rumbleBR += (inputSampleR*rumblecutoff);
}
inputSampleL -= (rumbleAL+rumbleBL);
inputSampleL *= (1.0-rumblecutoff);
inputSampleR -= (rumbleAR+rumbleBR);
inputSampleR *= (1.0-rumblecutoff);
inputSampleL *= wet;
inputSampleL += (drySampleL * dry);
inputSampleR *= wet;
inputSampleR += (drySampleR * dry);
//apply the dry to the noise
bL[9] = bL[8]; bL[8] = bL[7]; bL[7] = bL[6]; bL[6] = bL[5];
bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1];
bL[1] = bL[0]; bL[0] = accumulatorSampleL = inputSampleL;
bR[9] = bR[8]; bR[8] = bR[7]; bR[7] = bR[6]; bR[6] = bR[5];
bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1];
bR[1] = bR[0]; bR[0] = accumulatorSampleR = inputSampleR;
accumulatorSampleL *= f[0];
accumulatorSampleL += (bL[1] * f[1]);
accumulatorSampleL += (bL[2] * f[2]);
accumulatorSampleL += (bL[3] * f[3]);
accumulatorSampleL += (bL[4] * f[4]);
accumulatorSampleL += (bL[5] * f[5]);
accumulatorSampleL += (bL[6] * f[6]);
accumulatorSampleL += (bL[7] * f[7]);
accumulatorSampleL += (bL[8] * f[8]);
accumulatorSampleL += (bL[9] * f[9]);
//we are doing our repetitive calculations on a separate value
accumulatorSampleR *= f[0];
accumulatorSampleR += (bR[1] * f[1]);
accumulatorSampleR += (bR[2] * f[2]);
accumulatorSampleR += (bR[3] * f[3]);
accumulatorSampleR += (bR[4] * f[4]);
accumulatorSampleR += (bR[5] * f[5]);
accumulatorSampleR += (bR[6] * f[6]);
accumulatorSampleR += (bR[7] * f[7]);
accumulatorSampleR += (bR[8] * f[8]);
accumulatorSampleR += (bR[9] * f[9]);
//we are doing our repetitive calculations on a separate value
correctionSample = inputSampleL - accumulatorSampleL;
//we're gonna apply the total effect of all these calculations as a single subtract
//(formerly a more complicated algorithm)
inputSampleL -= correctionSample;
//applying the distance calculation to both the dry AND the noise output to blend them
correctionSample = inputSampleR - accumulatorSampleR;
//we're gonna apply the total effect of all these calculations as a single subtract
//(formerly a more complicated algorithm)
inputSampleR -= correctionSample;
//applying the distance calculation to both the dry AND the noise output to blend them
//sometimes I'm really tired and can't do stuff, and I remember trying to simplify this
//and breaking it somehow. So, there ya go, strange obtuse code.
//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 Noise::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
double cutoffL;
double cutoffR;
double cutofftarget = (A*3.5);
double rumblecutoff = cutofftarget * 0.005;
double invcutoffL;
double invcutoffR;
double drySampleL;
double drySampleR;
long double inputSampleL;
long double inputSampleR;
double highpass = C*38.0;
int lowcut = floor(highpass);
int dcut;
if (lowcut > 37) {dcut= 1151;}
if (lowcut == 37) {dcut= 1091;}
if (lowcut == 36) {dcut= 1087;}
if (lowcut == 35) {dcut= 1031;}
if (lowcut == 34) {dcut= 1013;}
if (lowcut == 33) {dcut= 971;}
if (lowcut == 32) {dcut= 907;}
if (lowcut == 31) {dcut= 839;}
if (lowcut == 30) {dcut= 797;}
if (lowcut == 29) {dcut= 733;}
if (lowcut == 28) {dcut= 719;}
if (lowcut == 27) {dcut= 673;}
if (lowcut == 26) {dcut= 613;}
if (lowcut == 25) {dcut= 593;}
if (lowcut == 24) {dcut= 541;}
if (lowcut == 23) {dcut= 479;}
if (lowcut == 22) {dcut= 431;}
if (lowcut == 21) {dcut= 419;}
if (lowcut == 20) {dcut= 373;}
if (lowcut == 19) {dcut= 311;}
if (lowcut == 18) {dcut= 293;}
if (lowcut == 17) {dcut= 233;}
if (lowcut == 16) {dcut= 191;}
if (lowcut == 15) {dcut= 173;}
if (lowcut == 14) {dcut= 131;}
if (lowcut == 13) {dcut= 113;}
if (lowcut == 12) {dcut= 71;}
if (lowcut == 11) {dcut= 53;}
if (lowcut == 10) {dcut= 31;}
if (lowcut == 9) {dcut= 27;}
if (lowcut == 8) {dcut= 23;}
if (lowcut == 7) {dcut= 19;}
if (lowcut == 6) {dcut= 17;}
if (lowcut == 5) {dcut= 13;}
if (lowcut == 4) {dcut= 11;}
if (lowcut == 3) {dcut= 7;}
if (lowcut == 2) {dcut= 5;}
if (lowcut < 2) {dcut= 3;}
highpass = B * 22.0;
lowcut = floor(highpass)+1;
double decay = 0.001 - ((1.0-pow(1.0-D,3))*0.001);
if (decay == 0.001) decay = 0.1;
double wet = F;
double dry = 1.0 - wet;
wet *= 0.01; //correct large gain issue
double correctionSample;
double accumulatorSampleL;
double accumulatorSampleR;
double overallscale = (E*9.0)+1.0;
double gain = overallscale;
if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;}
if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;}
if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;}
if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;}
if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;}
if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;}
if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;}
if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;}
if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;}
if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;}
//there, now we have a neat little moving average with remainders
if (overallscale < 1.0) overallscale = 1.0;
f[0] /= overallscale;
f[1] /= overallscale;
f[2] /= overallscale;
f[3] /= overallscale;
f[4] /= overallscale;
f[5] /= overallscale;
f[6] /= overallscale;
f[7] /= overallscale;
f[8] /= overallscale;
f[9] /= overallscale;
//and now it's neatly scaled, too
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.
}
drySampleL = inputSampleL;
drySampleR = inputSampleR;
if (surgeL<fabs(inputSampleL))
{
surgeL += (rand()/(double)RAND_MAX)*(fabs(inputSampleL)-surgeL);
if (surgeL > 1.0) surgeL = 1.0;
}
else
{
surgeL -= ((rand()/(double)RAND_MAX)*(surgeL-fabs(inputSampleL))*decay);
if (surgeL < 0.0) surgeL = 0.0;
}
cutoffL = pow((cutofftarget*surgeL),5);
if (cutoffL > 1.0) cutoffL = 1.0;
invcutoffL = 1.0 - cutoffL;
//set up modified cutoff L
if (surgeR<fabs(inputSampleR))
{
surgeR += (rand()/(double)RAND_MAX)*(fabs(inputSampleR)-surgeR);
if (surgeR > 1.0) surgeR = 1.0;
}
else
{
surgeR -= ((rand()/(double)RAND_MAX)*(surgeR-fabs(inputSampleR))*decay);
if (surgeR < 0.0) surgeR = 0.0;
}
cutoffR = pow((cutofftarget*surgeR),5);
if (cutoffR > 1.0) cutoffR = 1.0;
invcutoffR = 1.0 - cutoffR;
//set up modified cutoff R
flipL = !flipL;
flipR = !flipR;
filterflip = !filterflip;
quadratic -= 1;
if (quadratic < 0)
{
position += 1;
quadratic = position * position;
quadratic = quadratic % 170003; //% is C++ mod operator
quadratic *= quadratic;
quadratic = quadratic % 17011; //% is C++ mod operator
quadratic *= quadratic;
//quadratic = quadratic % 1709; //% is C++ mod operator
//quadratic *= quadratic;
quadratic = quadratic % dcut; //% is C++ mod operator
quadratic *= quadratic;
quadratic = quadratic % lowcut;
//sets density of the centering force
if (noiseAL < 0) {flipL = true;}
else {flipL = false;}
if (noiseAR < 0) {flipR = true;}
else {flipR = false;}
}
if (flipL) noiseAL += (rand()/(double)RAND_MAX);
else noiseAL -= (rand()/(double)RAND_MAX);
if (flipR) noiseAR += (rand()/(double)RAND_MAX);
else noiseAR -= (rand()/(double)RAND_MAX);
if (filterflip)
{
noiseBL *= invcutoffL; noiseBL += (noiseAL*cutoffL);
inputSampleL = noiseBL+noiseCL;
rumbleAL *= (1.0-rumblecutoff);
rumbleAL += (inputSampleL*rumblecutoff);
noiseBR *= invcutoffR; noiseBR += (noiseAR*cutoffR);
inputSampleR = noiseBR+noiseCR;
rumbleAR *= (1.0-rumblecutoff);
rumbleAR += (inputSampleR*rumblecutoff);
}
else
{
noiseCL *= invcutoffL; noiseCL += (noiseAL*cutoffL);
inputSampleL = noiseBL+noiseCL;
rumbleBL *= (1.0-rumblecutoff);
rumbleBL += (inputSampleL*rumblecutoff);
noiseCR *= invcutoffR; noiseCR += (noiseAR*cutoffR);
inputSampleR = noiseBR+noiseCR;
rumbleBR *= (1.0-rumblecutoff);
rumbleBR += (inputSampleR*rumblecutoff);
}
inputSampleL -= (rumbleAL+rumbleBL);
inputSampleL *= (1.0-rumblecutoff);
inputSampleR -= (rumbleAR+rumbleBR);
inputSampleR *= (1.0-rumblecutoff);
inputSampleL *= wet;
inputSampleL += (drySampleL * dry);
inputSampleR *= wet;
inputSampleR += (drySampleR * dry);
//apply the dry to the noise
bL[9] = bL[8]; bL[8] = bL[7]; bL[7] = bL[6]; bL[6] = bL[5];
bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1];
bL[1] = bL[0]; bL[0] = accumulatorSampleL = inputSampleL;
bR[9] = bR[8]; bR[8] = bR[7]; bR[7] = bR[6]; bR[6] = bR[5];
bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1];
bR[1] = bR[0]; bR[0] = accumulatorSampleR = inputSampleR;
accumulatorSampleL *= f[0];
accumulatorSampleL += (bL[1] * f[1]);
accumulatorSampleL += (bL[2] * f[2]);
accumulatorSampleL += (bL[3] * f[3]);
accumulatorSampleL += (bL[4] * f[4]);
accumulatorSampleL += (bL[5] * f[5]);
accumulatorSampleL += (bL[6] * f[6]);
accumulatorSampleL += (bL[7] * f[7]);
accumulatorSampleL += (bL[8] * f[8]);
accumulatorSampleL += (bL[9] * f[9]);
//we are doing our repetitive calculations on a separate value
accumulatorSampleR *= f[0];
accumulatorSampleR += (bR[1] * f[1]);
accumulatorSampleR += (bR[2] * f[2]);
accumulatorSampleR += (bR[3] * f[3]);
accumulatorSampleR += (bR[4] * f[4]);
accumulatorSampleR += (bR[5] * f[5]);
accumulatorSampleR += (bR[6] * f[6]);
accumulatorSampleR += (bR[7] * f[7]);
accumulatorSampleR += (bR[8] * f[8]);
accumulatorSampleR += (bR[9] * f[9]);
//we are doing our repetitive calculations on a separate value
correctionSample = inputSampleL - accumulatorSampleL;
//we're gonna apply the total effect of all these calculations as a single subtract
//(formerly a more complicated algorithm)
inputSampleL -= correctionSample;
//applying the distance calculation to both the dry AND the noise output to blend them
correctionSample = inputSampleR - accumulatorSampleR;
//we're gonna apply the total effect of all these calculations as a single subtract
//(formerly a more complicated algorithm)
inputSampleR -= correctionSample;
//applying the distance calculation to both the dry AND the noise output to blend them
//sometimes I'm really tired and can't do stuff, and I remember trying to simplify this
//and breaking it somehow. So, there ya go, strange obtuse code.
//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++;
}
}
