/* ========================================
* VoiceOfTheStarship - VoiceOfTheStarship.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __VoiceOfTheStarship_H
#include "VoiceOfTheStarship.h"
#endif
void VoiceOfTheStarship::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
double cutoff = pow((A*0.89)+0.1,3);
if (cutoff > 1.0) cutoff = 1.0;
double invcutoff = 1.0 - cutoff;
//this is the lowpass
double overallscale = ((1.0-A)*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;}
//this is the 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
int lowcut = floor(B*16.9);
if (lastAlgorithm != lowcut) {
noiseAL = 0.0; noiseBL = 0.0; noiseCL = 0.0;
noiseAR = 0.0; noiseBR = 0.0; noiseCR = 0.0;
for(int count = 0; count < 11; count++) {bL[count] = 0.0; bR[count] = 0.0;}
lastAlgorithm = lowcut;
}
//cuts the noise back to 0 if we are changing algorithms,
//because that also changes gains and can make loud pops.
//We still get pops, but they'd be even worse
int dcut;
if (lowcut > 15) {lowcut = 1151; dcut= 11517;}
if (lowcut == 15) {lowcut = 113; dcut= 1151;}
if (lowcut == 14) {lowcut = 71; dcut= 719;}
if (lowcut == 13) {lowcut = 53; dcut= 541;}
if (lowcut == 12) {lowcut = 31; dcut= 311;}
if (lowcut == 11) {lowcut = 23; dcut= 233;}
if (lowcut == 10) {lowcut = 19; dcut= 191;}
if (lowcut == 9) {lowcut = 17; dcut= 173;}
if (lowcut == 8) {lowcut = 13; dcut= 131;}
if (lowcut == 7) {lowcut = 11; dcut= 113;}
if (lowcut == 6) {lowcut = 7; dcut= 79;}
if (lowcut == 5) {lowcut = 6; dcut= 67;}
if (lowcut == 4) {lowcut = 5; dcut= 59;}
if (lowcut == 3) {lowcut = 4; dcut= 43;}
if (lowcut == 2) {lowcut = 3; dcut= 37;}
if (lowcut == 1) {lowcut = 2; dcut= 23;}
if (lowcut < 1) {lowcut = 1; dcut= 11;}
//this is the mechanism for cutting back subs without filtering
double rumbletrim = sqrt(lowcut);
//this among other things is just to give volume compensation
double inputSampleL;
double inputSampleR;
while (--sampleFrames >= 0)
{
inputSampleL = *in1;
inputSampleR = *in2;
//we then ignore this!
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;}
//every time we come here, we force the random walk to be
//toward the center of the waveform. Without this,
//it's a pure random walk that will generate DC.
}
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);
//here's the guts of the random walk
if (filterflip)
{
noiseBL *= invcutoff; noiseBL += (noiseAL*cutoff);
inputSampleL = noiseBL;
noiseBR *= invcutoff; noiseBR += (noiseAR*cutoff);
inputSampleR = noiseBR;
}
else
{
noiseCL *= invcutoff; noiseCL += (noiseAL*cutoff);
inputSampleL = noiseCL;
noiseCR *= invcutoff; noiseCR += (noiseAR*cutoff);
inputSampleR = noiseCR;
}
//now we have the output of the filter as inputSample.
//this filter is shallower than a straight IIR: it's interleaved
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] = 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] = inputSampleR;
inputSampleL *= f[0];
inputSampleL += (bL[1] * f[1]);
inputSampleL += (bL[2] * f[2]);
inputSampleL += (bL[3] * f[3]);
inputSampleL += (bL[4] * f[4]);
inputSampleL += (bL[5] * f[5]);
inputSampleL += (bL[6] * f[6]);
inputSampleL += (bL[7] * f[7]);
inputSampleL += (bL[8] * f[8]);
inputSampleL += (bL[9] * f[9]);
inputSampleR *= f[0];
inputSampleR += (bR[1] * f[1]);
inputSampleR += (bR[2] * f[2]);
inputSampleR += (bR[3] * f[3]);
inputSampleR += (bR[4] * f[4]);
inputSampleR += (bR[5] * f[5]);
inputSampleR += (bR[6] * f[6]);
inputSampleR += (bR[7] * f[7]);
inputSampleR += (bR[8] * f[8]);
inputSampleR += (bR[9] * f[9]);
inputSampleL *= 0.1;
inputSampleR *= 0.1;
inputSampleL *= invcutoff;
inputSampleR *= invcutoff;
inputSampleL /= rumbletrim;
inputSampleR /= rumbletrim;
flipL = !flipL;
flipR = !flipR;
filterflip = !filterflip;
//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 VoiceOfTheStarship::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
double cutoff = pow((A*0.89)+0.1,3);
if (cutoff > 1.0) cutoff = 1.0;
double invcutoff = 1.0 - cutoff;
//this is the lowpass
double overallscale = ((1.0-A)*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;}
//this is the 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
int lowcut = floor(B*16.9);
if (lastAlgorithm != lowcut) {
noiseAL = 0.0; noiseBL = 0.0; noiseCL = 0.0;
noiseAR = 0.0; noiseBR = 0.0; noiseCR = 0.0;
for(int count = 0; count < 11; count++) {bL[count] = 0.0; bR[count] = 0.0;}
lastAlgorithm = lowcut;
}
//cuts the noise back to 0 if we are changing algorithms,
//because that also changes gains and can make loud pops.
//We still get pops, but they'd be even worse
int dcut;
if (lowcut > 15) {lowcut = 1151; dcut= 11517;}
if (lowcut == 15) {lowcut = 113; dcut= 1151;}
if (lowcut == 14) {lowcut = 71; dcut= 719;}
if (lowcut == 13) {lowcut = 53; dcut= 541;}
if (lowcut == 12) {lowcut = 31; dcut= 311;}
if (lowcut == 11) {lowcut = 23; dcut= 233;}
if (lowcut == 10) {lowcut = 19; dcut= 191;}
if (lowcut == 9) {lowcut = 17; dcut= 173;}
if (lowcut == 8) {lowcut = 13; dcut= 131;}
if (lowcut == 7) {lowcut = 11; dcut= 113;}
if (lowcut == 6) {lowcut = 7; dcut= 79;}
if (lowcut == 5) {lowcut = 6; dcut= 67;}
if (lowcut == 4) {lowcut = 5; dcut= 59;}
if (lowcut == 3) {lowcut = 4; dcut= 43;}
if (lowcut == 2) {lowcut = 3; dcut= 37;}
if (lowcut == 1) {lowcut = 2; dcut= 23;}
if (lowcut < 1) {lowcut = 1; dcut= 11;}
//this is the mechanism for cutting back subs without filtering
double rumbletrim = sqrt(lowcut);
//this among other things is just to give volume compensation
double inputSampleL;
double inputSampleR;
while (--sampleFrames >= 0)
{
inputSampleL = *in1;
inputSampleR = *in2;
//we then ignore this!
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;}
//every time we come here, we force the random walk to be
//toward the center of the waveform. Without this,
//it's a pure random walk that will generate DC.
}
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);
//here's the guts of the random walk
if (filterflip)
{
noiseBL *= invcutoff; noiseBL += (noiseAL*cutoff);
inputSampleL = noiseBL;
noiseBR *= invcutoff; noiseBR += (noiseAR*cutoff);
inputSampleR = noiseBR;
}
else
{
noiseCL *= invcutoff; noiseCL += (noiseAL*cutoff);
inputSampleL = noiseCL;
noiseCR *= invcutoff; noiseCR += (noiseAR*cutoff);
inputSampleR = noiseCR;
}
//now we have the output of the filter as inputSample.
//this filter is shallower than a straight IIR: it's interleaved
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] = 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] = inputSampleR;
inputSampleL *= f[0];
inputSampleL += (bL[1] * f[1]);
inputSampleL += (bL[2] * f[2]);
inputSampleL += (bL[3] * f[3]);
inputSampleL += (bL[4] * f[4]);
inputSampleL += (bL[5] * f[5]);
inputSampleL += (bL[6] * f[6]);
inputSampleL += (bL[7] * f[7]);
inputSampleL += (bL[8] * f[8]);
inputSampleL += (bL[9] * f[9]);
inputSampleR *= f[0];
inputSampleR += (bR[1] * f[1]);
inputSampleR += (bR[2] * f[2]);
inputSampleR += (bR[3] * f[3]);
inputSampleR += (bR[4] * f[4]);
inputSampleR += (bR[5] * f[5]);
inputSampleR += (bR[6] * f[6]);
inputSampleR += (bR[7] * f[7]);
inputSampleR += (bR[8] * f[8]);
inputSampleR += (bR[9] * f[9]);
inputSampleL *= 0.1;
inputSampleR *= 0.1;
inputSampleL *= invcutoff;
inputSampleR *= invcutoff;
inputSampleL /= rumbletrim;
inputSampleR /= rumbletrim;
flipL = !flipL;
flipR = !flipR;
filterflip = !filterflip;
//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++;
}
}
