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/* ========================================
* Average - Average.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __Average_H
#include "Average.h"
#endif
void Average::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
double correctionSample;
double accumulatorSampleL;
double accumulatorSampleR;
double drySampleL;
double drySampleR;
double inputSampleL;
double inputSampleR;
double overallscale = (A * 9.0)+1.0;
double wet = B;
double dry = 1.0 - wet;
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;
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;
//primitive way of doing this: for larger batches of samples, you might
//try using a circular buffer like in a reverb. If you add the new sample
//and subtract the one on the end you can keep a running tally of the samples
//between. Beware of tiny floating-point math errors eventually screwing up
//your system, though!
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]);
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
inputSampleL -= correctionSample;
correctionSample = inputSampleR - accumulatorSampleR;
inputSampleR -= correctionSample;
//our one math operation on the input data coming in
if (wet < 1.0) {
inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
}
//dry/wet control only applies if you're using it. We don't do a multiply by 1.0
//if it 'won't change anything' but our sample might be at a very different scaling
//in the floating point system.
//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 Average::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
double correctionSample;
double accumulatorSampleL;
double accumulatorSampleR;
double drySampleL;
double drySampleR;
double inputSampleL;
double inputSampleR;
double overallscale = (A * 9.0)+1.0;
double wet = B;
double dry = 1.0 - wet;
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;
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;
//primitive way of doing this: for larger batches of samples, you might
//try using a circular buffer like in a reverb. If you add the new sample
//and subtract the one on the end you can keep a running tally of the samples
//between. Beware of tiny floating-point math errors eventually screwing up
//your system, though!
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]);
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
inputSampleL -= correctionSample;
correctionSample = inputSampleR - accumulatorSampleR;
inputSampleR -= correctionSample;
//our one math operation on the input data coming in
if (wet < 1.0) {
inputSampleL = (inputSampleL * wet) + (drySampleL * dry);
inputSampleR = (inputSampleR * wet) + (drySampleR * dry);
}
//dry/wet control only applies if you're using it. We don't do a multiply by 1.0
//if it 'won't change anything' but our sample might be at a very different scaling
//in the floating point system.
//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++;
}
}
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