/* ========================================
* ADT - ADT.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __ADT_H
#include "ADT.h"
#endif
void ADT::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
double gain = A * 1.272;
double targetA = pow(B,4) * 4790.0;
double fractionA;
double minusA;
double intensityA = C-0.5;
//first delay
double targetB = (pow(D,4) * 4790.0);
double fractionB;
double minusB;
double intensityB = E-0.5;
//second delay
double output = F*2.0;
while (--sampleFrames >= 0)
{
long double inputSampleL = *in1;
long double inputSampleR = *in2;
static int noisesourceL = 0;
static int noisesourceR = 850010;
int residue;
double applyresidue;
noisesourceL = noisesourceL % 1700021; noisesourceL++;
residue = noisesourceL * noisesourceL;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleL += applyresidue;
if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
inputSampleL -= applyresidue;
}
noisesourceR = noisesourceR % 1700021; noisesourceR++;
residue = noisesourceR * noisesourceR;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleR += applyresidue;
if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
inputSampleR -= applyresidue;
}
//for live air, we always apply the dither noise. Then, if our result is
//effectively digital black, we'll subtract it again. We want a 'air' hiss
if (fabs(offsetA - targetA) > 1000) offsetA = targetA;
offsetA = ((offsetA*999.0)+targetA)/1000.0;
fractionA = offsetA - floor(offsetA);
minusA = 1.0 - fractionA;
if (fabs(offsetB - targetB) > 1000) offsetB = targetB;
offsetB = ((offsetB*999.0)+targetB)/1000.0;
fractionB = offsetB - floor(offsetB);
minusB = 1.0 - fractionB;
//chase delay taps for smooth action
if (gain > 0) {inputSampleL /= gain; inputSampleR /= gain;}
if (inputSampleL > 1.2533141373155) inputSampleL = 1.2533141373155;
if (inputSampleL < -1.2533141373155) inputSampleL = -1.2533141373155;
if (inputSampleR > 1.2533141373155) inputSampleR = 1.2533141373155;
if (inputSampleR < -1.2533141373155) inputSampleR = -1.2533141373155;
inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL));
inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR));
//Spiral: lean out the sound a little when decoded by ConsoleBuss
if (gcount < 1 || gcount > 4800) {gcount = 4800;}
int count = gcount;
double totalL = 0.0;
double totalR = 0.0;
double tempL;
double tempR;
pL[count+4800] = pL[count] = inputSampleL;
pR[count+4800] = pR[count] = inputSampleR;
//double buffer
if (intensityA != 0.0)
{
count = (int)(gcount+floor(offsetA));
tempL = (pL[count] * minusA); //less as value moves away from .0
tempL += pL[count+1]; //we can assume always using this in one way or another?
tempL += (pL[count+2] * fractionA); //greater as value moves away from .0
tempL -= (((pL[count]-pL[count+1])-(pL[count+1]-pL[count+2]))/50); //interpolation hacks 'r us
totalL += (tempL * intensityA);
tempR = (pR[count] * minusA); //less as value moves away from .0
tempR += pR[count+1]; //we can assume always using this in one way or another?
tempR += (pR[count+2] * fractionA); //greater as value moves away from .0
tempR -= (((pR[count]-pR[count+1])-(pR[count+1]-pR[count+2]))/50); //interpolation hacks 'r us
totalR += (tempR * intensityA);
}
if (intensityB != 0.0)
{
count = (int)(gcount+floor(offsetB));
tempL = (pL[count] * minusB); //less as value moves away from .0
tempL += pL[count+1]; //we can assume always using this in one way or another?
tempL += (pL[count+2] * fractionB); //greater as value moves away from .0
tempL -= (((pL[count]-pL[count+1])-(pL[count+1]-pL[count+2]))/50); //interpolation hacks 'r us
totalL += (tempL * intensityB);
tempR = (pR[count] * minusB); //less as value moves away from .0
tempR += pR[count+1]; //we can assume always using this in one way or another?
tempR += (pR[count+2] * fractionB); //greater as value moves away from .0
tempR -= (((pR[count]-pR[count+1])-(pR[count+1]-pR[count+2]))/50); //interpolation hacks 'r us
totalR += (tempR * intensityB);
}
gcount--;
//still scrolling through the samples, remember
inputSampleL += totalL;
inputSampleR += totalR;
if (inputSampleL > 1.0) inputSampleL = 1.0;
if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0;
if (inputSampleR < -1.0) inputSampleR = -1.0;
//without this, you can get a NaN condition where it spits out DC offset at full blast!
inputSampleL = asin(inputSampleL);
inputSampleR = asin(inputSampleR);
//amplitude aspect
inputSampleL *= gain;
inputSampleR *= gain;
if (output < 1.0) {inputSampleL *= output; inputSampleR *= output;}
//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 ADT::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
double gain = A * 1.272;
double targetA = pow(B,4) * 4790.0;
double fractionA;
double minusA;
double intensityA = C-0.5;
//first delay
double targetB = (pow(D,4) * 4790.0);
double fractionB;
double minusB;
double intensityB = E-0.5;
//second delay
double output = F*2.0;
while (--sampleFrames >= 0)
{
long double inputSampleL = *in1;
long double inputSampleR = *in2;
static int noisesourceL = 0;
static int noisesourceR = 850010;
int residue;
double applyresidue;
noisesourceL = noisesourceL % 1700021; noisesourceL++;
residue = noisesourceL * noisesourceL;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleL += applyresidue;
if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
inputSampleL -= applyresidue;
}
noisesourceR = noisesourceR % 1700021; noisesourceR++;
residue = noisesourceR * noisesourceR;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleR += applyresidue;
if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
inputSampleR -= applyresidue;
}
//for live air, we always apply the dither noise. Then, if our result is
//effectively digital black, we'll subtract it again. We want a 'air' hiss
if (fabs(offsetA - targetA) > 1000) offsetA = targetA;
offsetA = ((offsetA*999.0)+targetA)/1000.0;
fractionA = offsetA - floor(offsetA);
minusA = 1.0 - fractionA;
if (fabs(offsetB - targetB) > 1000) offsetB = targetB;
offsetB = ((offsetB*999.0)+targetB)/1000.0;
fractionB = offsetB - floor(offsetB);
minusB = 1.0 - fractionB;
//chase delay taps for smooth action
if (gain > 0) {inputSampleL /= gain; inputSampleR /= gain;}
if (inputSampleL > 1.2533141373155) inputSampleL = 1.2533141373155;
if (inputSampleL < -1.2533141373155) inputSampleL = -1.2533141373155;
if (inputSampleR > 1.2533141373155) inputSampleR = 1.2533141373155;
if (inputSampleR < -1.2533141373155) inputSampleR = -1.2533141373155;
inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL));
inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR));
//Spiral: lean out the sound a little when decoded by ConsoleBuss
if (gcount < 1 || gcount > 4800) {gcount = 4800;}
int count = gcount;
double totalL = 0.0;
double totalR = 0.0;
double tempL;
double tempR;
pL[count+4800] = pL[count] = inputSampleL;
pR[count+4800] = pR[count] = inputSampleR;
//double buffer
if (intensityA != 0.0)
{
count = (int)(gcount+floor(offsetA));
tempL = (pL[count] * minusA); //less as value moves away from .0
tempL += pL[count+1]; //we can assume always using this in one way or another?
tempL += (pL[count+2] * fractionA); //greater as value moves away from .0
tempL -= (((pL[count]-pL[count+1])-(pL[count+1]-pL[count+2]))/50); //interpolation hacks 'r us
totalL += (tempL * intensityA);
tempR = (pR[count] * minusA); //less as value moves away from .0
tempR += pR[count+1]; //we can assume always using this in one way or another?
tempR += (pR[count+2] * fractionA); //greater as value moves away from .0
tempR -= (((pR[count]-pR[count+1])-(pR[count+1]-pR[count+2]))/50); //interpolation hacks 'r us
totalR += (tempR * intensityA);
}
if (intensityB != 0.0)
{
count = (int)(gcount+floor(offsetB));
tempL = (pL[count] * minusB); //less as value moves away from .0
tempL += pL[count+1]; //we can assume always using this in one way or another?
tempL += (pL[count+2] * fractionB); //greater as value moves away from .0
tempL -= (((pL[count]-pL[count+1])-(pL[count+1]-pL[count+2]))/50); //interpolation hacks 'r us
totalL += (tempL * intensityB);
tempR = (pR[count] * minusB); //less as value moves away from .0
tempR += pR[count+1]; //we can assume always using this in one way or another?
tempR += (pR[count+2] * fractionB); //greater as value moves away from .0
tempR -= (((pR[count]-pR[count+1])-(pR[count+1]-pR[count+2]))/50); //interpolation hacks 'r us
totalR += (tempR * intensityB);
}
gcount--;
//still scrolling through the samples, remember
inputSampleL += totalL;
inputSampleR += totalR;
if (inputSampleL > 1.0) inputSampleL = 1.0;
if (inputSampleL < -1.0) inputSampleL = -1.0;
if (inputSampleR > 1.0) inputSampleR = 1.0;
if (inputSampleR < -1.0) inputSampleR = -1.0;
//without this, you can get a NaN condition where it spits out DC offset at full blast!
inputSampleL = asin(inputSampleL);
inputSampleR = asin(inputSampleR);
//amplitude aspect
inputSampleL *= gain;
inputSampleR *= gain;
if (output < 1.0) {inputSampleL *= output; inputSampleR *= output;}
//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++;
}
}
