/* ========================================
* Thunder - Thunder.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __Thunder_H
#include "Thunder.h"
#endif
void Thunder::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
double thunder = A * 0.4;
double threshold = 1.0 - (thunder * 2.0);
if (threshold < 0.01) threshold = 0.01;
double muMakeupGain = 1.0 / threshold;
double release = pow((1.28-thunder),5)*32768.0;
release /= overallscale;
double fastest = sqrt(release);
double EQ = ((0.0275 / getSampleRate())*32000.0);
double dcblock = EQ / 300.0;
double basstrim = (0.01/EQ)+1.0;
//FF parameters also ride off Speed
double outputGain = B;
double coefficient;
double inputSense;
double resultL;
double resultR;
double resultM;
double resultML;
double resultMR;
long double inputSampleL;
long double inputSampleR;
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.
}
inputSampleL = inputSampleL * muMakeupGain;
inputSampleR = inputSampleR * muMakeupGain;
if (gateL < fabs(inputSampleL)) gateL = inputSampleL;
else gateL -= dcblock;
if (gateR < fabs(inputSampleR)) gateR = inputSampleR;
else gateR -= dcblock;
//setting up gated DC blocking to control the tendency for rumble and offset
//begin three FathomFive stages
iirSampleAL += (inputSampleL * EQ * thunder);
iirSampleAL -= (iirSampleAL * iirSampleAL * iirSampleAL * EQ);
if (iirSampleAL > gateL) iirSampleAL -= dcblock;
if (iirSampleAL < -gateL) iirSampleAL += dcblock;
resultL = iirSampleAL*basstrim;
iirSampleBL = (iirSampleBL * (1 - EQ)) + (resultL * EQ);
resultL = iirSampleBL;
iirSampleAR += (inputSampleR * EQ * thunder);
iirSampleAR -= (iirSampleAR * iirSampleAR * iirSampleAR * EQ);
if (iirSampleAR > gateR) iirSampleAR -= dcblock;
if (iirSampleAR < -gateR) iirSampleAR += dcblock;
resultR = iirSampleAR*basstrim;
iirSampleBR = (iirSampleBR * (1 - EQ)) + (resultR * EQ);
resultR = iirSampleBR;
iirSampleAM += ((inputSampleL + inputSampleR) * EQ * thunder);
iirSampleAM -= (iirSampleAM * iirSampleAM * iirSampleAM * EQ);
resultM = iirSampleAM*basstrim;
iirSampleBM = (iirSampleBM * (1 - EQ)) + (resultM * EQ);
resultM = iirSampleBM;
iirSampleCM = (iirSampleCM * (1 - EQ)) + (resultM * EQ);
resultM = fabs(iirSampleCM);
resultML = fabs(resultL);
resultMR = fabs(resultR);
if (resultM > resultML) resultML = resultM;
if (resultM > resultMR) resultMR = resultM;
//trying to restrict the buzziness
if (resultML > 1.0) resultML = 1.0;
if (resultMR > 1.0) resultMR = 1.0;
//now we have result L, R and M the trigger modulator which must be 0-1
//begin compressor section
inputSampleL -= (iirSampleBL * thunder);
inputSampleR -= (iirSampleBR * thunder);
//highpass the comp section by sneaking out what will be the reinforcement
inputSense = fabs(inputSampleL);
if (fabs(inputSampleR) > inputSense)
inputSense = fabs(inputSampleR);
//we will take the greater of either channel and just use that, then apply the result
//to both stereo channels.
if (flip)
{
if (inputSense > threshold)
{
muVary = threshold / inputSense;
muAttack = sqrt(fabs(muSpeedA));
muCoefficientA = muCoefficientA * (muAttack-1.0);
if (muVary < threshold)
{
muCoefficientA = muCoefficientA + threshold;
}
else
{
muCoefficientA = muCoefficientA + muVary;
}
muCoefficientA = muCoefficientA / muAttack;
}
else
{
muCoefficientA = muCoefficientA * ((muSpeedA * muSpeedA)-1.0);
muCoefficientA = muCoefficientA + 1.0;
muCoefficientA = muCoefficientA / (muSpeedA * muSpeedA);
}
muNewSpeed = muSpeedA * (muSpeedA-1);
muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest;
muSpeedA = muNewSpeed / muSpeedA;
}
else
{
if (inputSense > threshold)
{
muVary = threshold / inputSense;
muAttack = sqrt(fabs(muSpeedB));
muCoefficientB = muCoefficientB * (muAttack-1);
if (muVary < threshold)
{
muCoefficientB = muCoefficientB + threshold;
}
else
{
muCoefficientB = muCoefficientB + muVary;
}
muCoefficientB = muCoefficientB / muAttack;
}
else
{
muCoefficientB = muCoefficientB * ((muSpeedB * muSpeedB)-1.0);
muCoefficientB = muCoefficientB + 1.0;
muCoefficientB = muCoefficientB / (muSpeedB * muSpeedB);
}
muNewSpeed = muSpeedB * (muSpeedB-1);
muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest;
muSpeedB = muNewSpeed / muSpeedB;
}
//got coefficients, adjusted speeds
if (flip)
{
coefficient = pow(muCoefficientA,2);
inputSampleL *= coefficient;
inputSampleR *= coefficient;
}
else
{
coefficient = pow(muCoefficientB,2);
inputSampleL *= coefficient;
inputSampleR *= coefficient;
}
//applied compression with vari-vari-µ-µ-µ-µ-µ-µ-is-the-kitten-song o/~
//applied gain correction to control output level- tends to constrain sound rather than inflate it
inputSampleL += (resultL * resultM);
inputSampleR += (resultR * resultM);
//combine the two by adding the summed channnel of lows
if (outputGain != 1.0) {
inputSampleL *= outputGain;
inputSampleR *= outputGain;
}
//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 Thunder::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
double thunder = A * 0.4;
double threshold = 1.0 - (thunder * 2.0);
if (threshold < 0.01) threshold = 0.01;
double muMakeupGain = 1.0 / threshold;
double release = pow((1.28-thunder),5)*32768.0;
release /= overallscale;
double fastest = sqrt(release);
double EQ = ((0.0275 / getSampleRate())*32000.0);
double dcblock = EQ / 300.0;
double basstrim = (0.01/EQ)+1.0;
//FF parameters also ride off Speed
double outputGain = B;
double coefficient;
double inputSense;
double resultL;
double resultR;
double resultM;
double resultML;
double resultMR;
long double inputSampleL;
long double inputSampleR;
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.
}
inputSampleL = inputSampleL * muMakeupGain;
inputSampleR = inputSampleR * muMakeupGain;
if (gateL < fabs(inputSampleL)) gateL = inputSampleL;
else gateL -= dcblock;
if (gateR < fabs(inputSampleR)) gateR = inputSampleR;
else gateR -= dcblock;
//setting up gated DC blocking to control the tendency for rumble and offset
//begin three FathomFive stages
iirSampleAL += (inputSampleL * EQ * thunder);
iirSampleAL -= (iirSampleAL * iirSampleAL * iirSampleAL * EQ);
if (iirSampleAL > gateL) iirSampleAL -= dcblock;
if (iirSampleAL < -gateL) iirSampleAL += dcblock;
resultL = iirSampleAL*basstrim;
iirSampleBL = (iirSampleBL * (1 - EQ)) + (resultL * EQ);
resultL = iirSampleBL;
iirSampleAR += (inputSampleR * EQ * thunder);
iirSampleAR -= (iirSampleAR * iirSampleAR * iirSampleAR * EQ);
if (iirSampleAR > gateR) iirSampleAR -= dcblock;
if (iirSampleAR < -gateR) iirSampleAR += dcblock;
resultR = iirSampleAR*basstrim;
iirSampleBR = (iirSampleBR * (1 - EQ)) + (resultR * EQ);
resultR = iirSampleBR;
iirSampleAM += ((inputSampleL + inputSampleR) * EQ * thunder);
iirSampleAM -= (iirSampleAM * iirSampleAM * iirSampleAM * EQ);
resultM = iirSampleAM*basstrim;
iirSampleBM = (iirSampleBM * (1 - EQ)) + (resultM * EQ);
resultM = iirSampleBM;
iirSampleCM = (iirSampleCM * (1 - EQ)) + (resultM * EQ);
resultM = fabs(iirSampleCM);
resultML = fabs(resultL);
resultMR = fabs(resultR);
if (resultM > resultML) resultML = resultM;
if (resultM > resultMR) resultMR = resultM;
//trying to restrict the buzziness
if (resultML > 1.0) resultML = 1.0;
if (resultMR > 1.0) resultMR = 1.0;
//now we have result L, R and M the trigger modulator which must be 0-1
//begin compressor section
inputSampleL -= (iirSampleBL * thunder);
inputSampleR -= (iirSampleBR * thunder);
//highpass the comp section by sneaking out what will be the reinforcement
inputSense = fabs(inputSampleL);
if (fabs(inputSampleR) > inputSense)
inputSense = fabs(inputSampleR);
//we will take the greater of either channel and just use that, then apply the result
//to both stereo channels.
if (flip)
{
if (inputSense > threshold)
{
muVary = threshold / inputSense;
muAttack = sqrt(fabs(muSpeedA));
muCoefficientA = muCoefficientA * (muAttack-1.0);
if (muVary < threshold)
{
muCoefficientA = muCoefficientA + threshold;
}
else
{
muCoefficientA = muCoefficientA + muVary;
}
muCoefficientA = muCoefficientA / muAttack;
}
else
{
muCoefficientA = muCoefficientA * ((muSpeedA * muSpeedA)-1.0);
muCoefficientA = muCoefficientA + 1.0;
muCoefficientA = muCoefficientA / (muSpeedA * muSpeedA);
}
muNewSpeed = muSpeedA * (muSpeedA-1);
muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest;
muSpeedA = muNewSpeed / muSpeedA;
}
else
{
if (inputSense > threshold)
{
muVary = threshold / inputSense;
muAttack = sqrt(fabs(muSpeedB));
muCoefficientB = muCoefficientB * (muAttack-1);
if (muVary < threshold)
{
muCoefficientB = muCoefficientB + threshold;
}
else
{
muCoefficientB = muCoefficientB + muVary;
}
muCoefficientB = muCoefficientB / muAttack;
}
else
{
muCoefficientB = muCoefficientB * ((muSpeedB * muSpeedB)-1.0);
muCoefficientB = muCoefficientB + 1.0;
muCoefficientB = muCoefficientB / (muSpeedB * muSpeedB);
}
muNewSpeed = muSpeedB * (muSpeedB-1);
muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest;
muSpeedB = muNewSpeed / muSpeedB;
}
//got coefficients, adjusted speeds
if (flip)
{
coefficient = pow(muCoefficientA,2);
inputSampleL *= coefficient;
inputSampleR *= coefficient;
}
else
{
coefficient = pow(muCoefficientB,2);
inputSampleL *= coefficient;
inputSampleR *= coefficient;
}
//applied compression with vari-vari-µ-µ-µ-µ-µ-µ-is-the-kitten-song o/~
//applied gain correction to control output level- tends to constrain sound rather than inflate it
inputSampleL += (resultL * resultM);
inputSampleR += (resultR * resultM);
//combine the two by adding the summed channnel of lows
if (outputGain != 1.0) {
inputSampleL *= outputGain;
inputSampleR *= outputGain;
}
//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++;
}
}
