/* ========================================
* NCSeventeen - NCSeventeen.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __NCSeventeen_H
#include "NCSeventeen.h"
#endif
void NCSeventeen::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
double inP2;
double chebyshev;
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
float fpTemp;
long double fpOld = 0.618033988749894848204586; //golden ratio!
long double fpNew = 1.0 - fpOld;
double IIRscaleback = 0.0004716;
double bassScaleback = 0.0002364;
double trebleScaleback = 0.0005484;
double addBassBuss = 0.000243;
double addTrebBuss = 0.000407;
double addShortBuss = 0.000326;
IIRscaleback /= overallscale;
bassScaleback /= overallscale;
trebleScaleback /= overallscale;
addBassBuss /= overallscale;
addTrebBuss /= overallscale;
addShortBuss /= overallscale;
double limitingBass = 0.39;
double limitingTreb = 0.6;
double limiting = 0.36;
double maxfeedBass = 0.972;
double maxfeedTreb = 0.972;
double maxfeed = 0.975;
double bridgerectifier;
long double inputSampleL;
double lowSampleL = 0.0;
double highSampleL;
double distSampleL;
double minusSampleL;
double plusSampleL;
long double inputSampleR;
double lowSampleR = 0.0;
double highSampleR;
double distSampleR;
double minusSampleR;
double plusSampleR;
double gain = pow(10.0,(A*24.0)/20);
double outlevel = B;
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 *= gain;
inputSampleR *= gain;
if (flip)
{
iirSampleAL = (iirSampleAL * 0.9) + (inputSampleL * 0.1);
lowSampleL = iirSampleAL;
iirSampleAR = (iirSampleAR * 0.9) + (inputSampleR * 0.1);
lowSampleR = iirSampleAR;
}
else
{
iirSampleBL = (iirSampleBL * 0.9) + (inputSampleL * 0.1);
lowSampleL = iirSampleBL;
iirSampleBR = (iirSampleBR * 0.9) + (inputSampleR * 0.1);
lowSampleR = iirSampleBR;
}
highSampleL = inputSampleL - lowSampleL;
highSampleR = inputSampleR - lowSampleR;
flip = !flip;
//we now have two bands and the original source
inP2 = lowSampleL * lowSampleL;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= basslevL;
//second harmonic max +1
if (basslevL > 0) basslevL -= bassScaleback;
if (basslevL < 0) basslevL += bassScaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(lowSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (lowSampleL > 0.0) distSampleL = bridgerectifier;
else distSampleL = -bridgerectifier;
minusSampleL = lowSampleL - distSampleL;
plusSampleL = lowSampleL + distSampleL;
if (minusSampleL > maxfeedBass) minusSampleL = maxfeedBass;
if (plusSampleL > maxfeedBass) plusSampleL = maxfeedBass;
if (plusSampleL < -maxfeedBass) plusSampleL = -maxfeedBass;
if (minusSampleL < -maxfeedBass) minusSampleL = -maxfeedBass;
if (lowSampleL > distSampleL) basslevL += (minusSampleL*addBassBuss);
if (lowSampleL < -distSampleL) basslevL -= (plusSampleL*addBassBuss);
if (basslevL > 1.0) basslevL = 1.0;
if (basslevL < -1.0) basslevL = -1.0;
bridgerectifier = fabs(lowSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (lowSampleL > 0.0) lowSampleL = bridgerectifier;
else lowSampleL = -bridgerectifier;
//apply the distortion transform for reals
lowSampleL /= (1.0+fabs(basslevL*limitingBass));
lowSampleL += chebyshev;
//apply the correction measuresL
inP2 = lowSampleR * lowSampleR;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= basslevR;
//second harmonic max +1
if (basslevR > 0) basslevR -= bassScaleback;
if (basslevR < 0) basslevR += bassScaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(lowSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (lowSampleR > 0.0) distSampleR = bridgerectifier;
else distSampleR = -bridgerectifier;
minusSampleR = lowSampleR - distSampleR;
plusSampleR = lowSampleR + distSampleR;
if (minusSampleR > maxfeedBass) minusSampleR = maxfeedBass;
if (plusSampleR > maxfeedBass) plusSampleR = maxfeedBass;
if (plusSampleR < -maxfeedBass) plusSampleR = -maxfeedBass;
if (minusSampleR < -maxfeedBass) minusSampleR = -maxfeedBass;
if (lowSampleR > distSampleR) basslevR += (minusSampleR*addBassBuss);
if (lowSampleR < -distSampleR) basslevR -= (plusSampleR*addBassBuss);
if (basslevR > 1.0) basslevR = 1.0;
if (basslevR < -1.0) basslevR = -1.0;
bridgerectifier = fabs(lowSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (lowSampleR > 0.0) lowSampleR = bridgerectifier;
else lowSampleR = -bridgerectifier;
//apply the distortion transform for reals
lowSampleR /= (1.0+fabs(basslevR*limitingBass));
lowSampleR += chebyshev;
//apply the correction measuresR
inP2 = highSampleL * highSampleL;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= treblevL;
//second harmonic max +1
if (treblevL > 0) treblevL -= trebleScaleback;
if (treblevL < 0) treblevL += trebleScaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(highSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (highSampleL > 0.0) distSampleL = bridgerectifier;
else distSampleL = -bridgerectifier;
minusSampleL = highSampleL - distSampleL;
plusSampleL = highSampleL + distSampleL;
if (minusSampleL > maxfeedTreb) minusSampleL = maxfeedTreb;
if (plusSampleL > maxfeedTreb) plusSampleL = maxfeedTreb;
if (plusSampleL < -maxfeedTreb) plusSampleL = -maxfeedTreb;
if (minusSampleL < -maxfeedTreb) minusSampleL = -maxfeedTreb;
if (highSampleL > distSampleL) treblevL += (minusSampleL*addTrebBuss);
if (highSampleL < -distSampleL) treblevL -= (plusSampleL*addTrebBuss);
if (treblevL > 1.0) treblevL = 1.0;
if (treblevL < -1.0) treblevL = -1.0;
bridgerectifier = fabs(highSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (highSampleL > 0.0) highSampleL = bridgerectifier;
else highSampleL = -bridgerectifier;
//apply the distortion transform for reals
highSampleL /= (1.0+fabs(treblevL*limitingTreb));
highSampleL += chebyshev;
//apply the correction measuresL
inP2 = highSampleR * highSampleR;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= treblevR;
//second harmonic max +1
if (treblevR > 0) treblevR -= trebleScaleback;
if (treblevR < 0) treblevR += trebleScaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(highSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (highSampleR > 0.0) distSampleR = bridgerectifier;
else distSampleR = -bridgerectifier;
minusSampleR = highSampleR - distSampleR;
plusSampleR = highSampleR + distSampleR;
if (minusSampleR > maxfeedTreb) minusSampleR = maxfeedTreb;
if (plusSampleR > maxfeedTreb) plusSampleR = maxfeedTreb;
if (plusSampleR < -maxfeedTreb) plusSampleR = -maxfeedTreb;
if (minusSampleR < -maxfeedTreb) minusSampleR = -maxfeedTreb;
if (highSampleR > distSampleR) treblevR += (minusSampleR*addTrebBuss);
if (highSampleR < -distSampleR) treblevR -= (plusSampleR*addTrebBuss);
if (treblevR > 1.0) treblevR = 1.0;
if (treblevR < -1.0) treblevR = -1.0;
bridgerectifier = fabs(highSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (highSampleR > 0.0) highSampleR = bridgerectifier;
else highSampleR = -bridgerectifier;
//apply the distortion transform for reals
highSampleR /= (1.0+fabs(treblevR*limitingTreb));
highSampleR += chebyshev;
//apply the correction measuresR
inputSampleL = lowSampleL + highSampleL;
inputSampleR = lowSampleR + highSampleR;
inP2 = inputSampleL * inputSampleL;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= cheblevL;
//third harmonic max -1
if (cheblevL > 0) cheblevL -= (IIRscaleback);
if (cheblevL < 0) cheblevL += (IIRscaleback);
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(inputSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (inputSampleL > 0.0) distSampleL = bridgerectifier;
else distSampleL = -bridgerectifier;
minusSampleL = inputSampleL - distSampleL;
plusSampleL = inputSampleL + distSampleL;
if (minusSampleL > maxfeed) minusSampleL = maxfeed;
if (plusSampleL > maxfeed) plusSampleL = maxfeed;
if (plusSampleL < -maxfeed) plusSampleL = -maxfeed;
if (minusSampleL < -maxfeed) minusSampleL = -maxfeed;
if (inputSampleL > distSampleL) cheblevL += (minusSampleL*addShortBuss);
if (inputSampleL < -distSampleL) cheblevL -= (plusSampleL*addShortBuss);
if (cheblevL > 1.0) cheblevL = 1.0;
if (cheblevL < -1.0) cheblevL = -1.0;
bridgerectifier = fabs(inputSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (inputSampleL > 0.0) inputSampleL = bridgerectifier;
else inputSampleL = -bridgerectifier;
//apply the distortion transform for reals
inputSampleL /= (1.0+fabs(cheblevL*limiting));
inputSampleL += chebyshev;
//apply the correction measuresL
inP2 = inputSampleR * inputSampleR;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= cheblevR;
//third harmonic max -1
if (cheblevR > 0) cheblevR -= IIRscaleback;
if (cheblevR < 0) cheblevR += IIRscaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(inputSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (inputSampleR > 0.0) distSampleR = bridgerectifier;
else distSampleR = -bridgerectifier;
minusSampleR = inputSampleR - distSampleR;
plusSampleR = inputSampleR + distSampleR;
if (minusSampleR > maxfeed) minusSampleR = maxfeed;
if (plusSampleR > maxfeed) plusSampleR = maxfeed;
if (plusSampleR < -maxfeed) plusSampleR = -maxfeed;
if (minusSampleR < -maxfeed) minusSampleR = -maxfeed;
if (inputSampleR > distSampleR) cheblevR += (minusSampleR*addShortBuss);
if (inputSampleR < -distSampleR) cheblevR -= (plusSampleR*addShortBuss);
if (cheblevR > 1.0) cheblevR = 1.0;
if (cheblevR < -1.0) cheblevR = -1.0;
bridgerectifier = fabs(inputSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (inputSampleR > 0.0) inputSampleR = bridgerectifier;
else inputSampleR = -bridgerectifier;
//apply the distortion transform for reals
inputSampleR /= (1.0+fabs(cheblevR*limiting));
inputSampleR += chebyshev;
//apply the correction measuresR
if (outlevel < 1.0) {
inputSampleL *= outlevel;
inputSampleR *= outlevel;
}
if (inputSampleL > 0.95) inputSampleL = 0.95;
if (inputSampleL < -0.95) inputSampleL = -0.95;
if (inputSampleR > 0.95) inputSampleR = 0.95;
if (inputSampleR < -0.95) inputSampleR = -0.95;
//iron bar
//noise shaping to 32-bit floating point
if (fpFlip) {
fpTemp = inputSampleL;
fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew);
inputSampleL += fpNShapeLA;
fpTemp = inputSampleR;
fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew);
inputSampleR += fpNShapeRA;
}
else {
fpTemp = inputSampleL;
fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew);
inputSampleL += fpNShapeLB;
fpTemp = inputSampleR;
fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew);
inputSampleR += fpNShapeRB;
}
fpFlip = !fpFlip;
//end noise shaping on 32 bit output
*out1 = inputSampleL;
*out2 = inputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
}
void NCSeventeen::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
double inP2;
double chebyshev;
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
double fpTemp;
long double fpOld = 0.618033988749894848204586; //golden ratio!
long double fpNew = 1.0 - fpOld;
double IIRscaleback = 0.0004716;
double bassScaleback = 0.0002364;
double trebleScaleback = 0.0005484;
double addBassBuss = 0.000243;
double addTrebBuss = 0.000407;
double addShortBuss = 0.000326;
IIRscaleback /= overallscale;
bassScaleback /= overallscale;
trebleScaleback /= overallscale;
addBassBuss /= overallscale;
addTrebBuss /= overallscale;
addShortBuss /= overallscale;
double limitingBass = 0.39;
double limitingTreb = 0.6;
double limiting = 0.36;
double maxfeedBass = 0.972;
double maxfeedTreb = 0.972;
double maxfeed = 0.975;
double bridgerectifier;
long double inputSampleL;
double lowSampleL = 0.0;
double highSampleL;
double distSampleL;
double minusSampleL;
double plusSampleL;
long double inputSampleR;
double lowSampleR = 0.0;
double highSampleR;
double distSampleR;
double minusSampleR;
double plusSampleR;
double gain = pow(10.0,(A*24.0)/20);
double outlevel = B;
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 *= gain;
inputSampleR *= gain;
if (flip)
{
iirSampleAL = (iirSampleAL * 0.9) + (inputSampleL * 0.1);
lowSampleL = iirSampleAL;
iirSampleAR = (iirSampleAR * 0.9) + (inputSampleR * 0.1);
lowSampleR = iirSampleAR;
}
else
{
iirSampleBL = (iirSampleBL * 0.9) + (inputSampleL * 0.1);
lowSampleL = iirSampleBL;
iirSampleBR = (iirSampleBR * 0.9) + (inputSampleR * 0.1);
lowSampleR = iirSampleBR;
}
highSampleL = inputSampleL - lowSampleL;
highSampleR = inputSampleR - lowSampleR;
flip = !flip;
//we now have two bands and the original source
inP2 = lowSampleL * lowSampleL;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= basslevL;
//second harmonic max +1
if (basslevL > 0) basslevL -= bassScaleback;
if (basslevL < 0) basslevL += bassScaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(lowSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (lowSampleL > 0.0) distSampleL = bridgerectifier;
else distSampleL = -bridgerectifier;
minusSampleL = lowSampleL - distSampleL;
plusSampleL = lowSampleL + distSampleL;
if (minusSampleL > maxfeedBass) minusSampleL = maxfeedBass;
if (plusSampleL > maxfeedBass) plusSampleL = maxfeedBass;
if (plusSampleL < -maxfeedBass) plusSampleL = -maxfeedBass;
if (minusSampleL < -maxfeedBass) minusSampleL = -maxfeedBass;
if (lowSampleL > distSampleL) basslevL += (minusSampleL*addBassBuss);
if (lowSampleL < -distSampleL) basslevL -= (plusSampleL*addBassBuss);
if (basslevL > 1.0) basslevL = 1.0;
if (basslevL < -1.0) basslevL = -1.0;
bridgerectifier = fabs(lowSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (lowSampleL > 0.0) lowSampleL = bridgerectifier;
else lowSampleL = -bridgerectifier;
//apply the distortion transform for reals
lowSampleL /= (1.0+fabs(basslevL*limitingBass));
lowSampleL += chebyshev;
//apply the correction measuresL
inP2 = lowSampleR * lowSampleR;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= basslevR;
//second harmonic max +1
if (basslevR > 0) basslevR -= bassScaleback;
if (basslevR < 0) basslevR += bassScaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(lowSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (lowSampleR > 0.0) distSampleR = bridgerectifier;
else distSampleR = -bridgerectifier;
minusSampleR = lowSampleR - distSampleR;
plusSampleR = lowSampleR + distSampleR;
if (minusSampleR > maxfeedBass) minusSampleR = maxfeedBass;
if (plusSampleR > maxfeedBass) plusSampleR = maxfeedBass;
if (plusSampleR < -maxfeedBass) plusSampleR = -maxfeedBass;
if (minusSampleR < -maxfeedBass) minusSampleR = -maxfeedBass;
if (lowSampleR > distSampleR) basslevR += (minusSampleR*addBassBuss);
if (lowSampleR < -distSampleR) basslevR -= (plusSampleR*addBassBuss);
if (basslevR > 1.0) basslevR = 1.0;
if (basslevR < -1.0) basslevR = -1.0;
bridgerectifier = fabs(lowSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (lowSampleR > 0.0) lowSampleR = bridgerectifier;
else lowSampleR = -bridgerectifier;
//apply the distortion transform for reals
lowSampleR /= (1.0+fabs(basslevR*limitingBass));
lowSampleR += chebyshev;
//apply the correction measuresR
inP2 = highSampleL * highSampleL;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= treblevL;
//second harmonic max +1
if (treblevL > 0) treblevL -= trebleScaleback;
if (treblevL < 0) treblevL += trebleScaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(highSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (highSampleL > 0.0) distSampleL = bridgerectifier;
else distSampleL = -bridgerectifier;
minusSampleL = highSampleL - distSampleL;
plusSampleL = highSampleL + distSampleL;
if (minusSampleL > maxfeedTreb) minusSampleL = maxfeedTreb;
if (plusSampleL > maxfeedTreb) plusSampleL = maxfeedTreb;
if (plusSampleL < -maxfeedTreb) plusSampleL = -maxfeedTreb;
if (minusSampleL < -maxfeedTreb) minusSampleL = -maxfeedTreb;
if (highSampleL > distSampleL) treblevL += (minusSampleL*addTrebBuss);
if (highSampleL < -distSampleL) treblevL -= (plusSampleL*addTrebBuss);
if (treblevL > 1.0) treblevL = 1.0;
if (treblevL < -1.0) treblevL = -1.0;
bridgerectifier = fabs(highSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (highSampleL > 0.0) highSampleL = bridgerectifier;
else highSampleL = -bridgerectifier;
//apply the distortion transform for reals
highSampleL /= (1.0+fabs(treblevL*limitingTreb));
highSampleL += chebyshev;
//apply the correction measuresL
inP2 = highSampleR * highSampleR;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= treblevR;
//second harmonic max +1
if (treblevR > 0) treblevR -= trebleScaleback;
if (treblevR < 0) treblevR += trebleScaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(highSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (highSampleR > 0.0) distSampleR = bridgerectifier;
else distSampleR = -bridgerectifier;
minusSampleR = highSampleR - distSampleR;
plusSampleR = highSampleR + distSampleR;
if (minusSampleR > maxfeedTreb) minusSampleR = maxfeedTreb;
if (plusSampleR > maxfeedTreb) plusSampleR = maxfeedTreb;
if (plusSampleR < -maxfeedTreb) plusSampleR = -maxfeedTreb;
if (minusSampleR < -maxfeedTreb) minusSampleR = -maxfeedTreb;
if (highSampleR > distSampleR) treblevR += (minusSampleR*addTrebBuss);
if (highSampleR < -distSampleR) treblevR -= (plusSampleR*addTrebBuss);
if (treblevR > 1.0) treblevR = 1.0;
if (treblevR < -1.0) treblevR = -1.0;
bridgerectifier = fabs(highSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (highSampleR > 0.0) highSampleR = bridgerectifier;
else highSampleR = -bridgerectifier;
//apply the distortion transform for reals
highSampleR /= (1.0+fabs(treblevR*limitingTreb));
highSampleR += chebyshev;
//apply the correction measuresR
inputSampleL = lowSampleL + highSampleL;
inputSampleR = lowSampleR + highSampleR;
inP2 = inputSampleL * inputSampleL;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= cheblevL;
//third harmonic max -1
if (cheblevL > 0) cheblevL -= (IIRscaleback);
if (cheblevL < 0) cheblevL += (IIRscaleback);
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(inputSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (inputSampleL > 0.0) distSampleL = bridgerectifier;
else distSampleL = -bridgerectifier;
minusSampleL = inputSampleL - distSampleL;
plusSampleL = inputSampleL + distSampleL;
if (minusSampleL > maxfeed) minusSampleL = maxfeed;
if (plusSampleL > maxfeed) plusSampleL = maxfeed;
if (plusSampleL < -maxfeed) plusSampleL = -maxfeed;
if (minusSampleL < -maxfeed) minusSampleL = -maxfeed;
if (inputSampleL > distSampleL) cheblevL += (minusSampleL*addShortBuss);
if (inputSampleL < -distSampleL) cheblevL -= (plusSampleL*addShortBuss);
if (cheblevL > 1.0) cheblevL = 1.0;
if (cheblevL < -1.0) cheblevL = -1.0;
bridgerectifier = fabs(inputSampleL);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (inputSampleL > 0.0) inputSampleL = bridgerectifier;
else inputSampleL = -bridgerectifier;
//apply the distortion transform for reals
inputSampleL /= (1.0+fabs(cheblevL*limiting));
inputSampleL += chebyshev;
//apply the correction measuresL
inP2 = inputSampleR * inputSampleR;
if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0;
chebyshev = (2 * inP2);
chebyshev *= cheblevR;
//third harmonic max -1
if (cheblevR > 0) cheblevR -= IIRscaleback;
if (cheblevR < 0) cheblevR += IIRscaleback;
//this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on
bridgerectifier = fabs(inputSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (inputSampleR > 0.0) distSampleR = bridgerectifier;
else distSampleR = -bridgerectifier;
minusSampleR = inputSampleR - distSampleR;
plusSampleR = inputSampleR + distSampleR;
if (minusSampleR > maxfeed) minusSampleR = maxfeed;
if (plusSampleR > maxfeed) plusSampleR = maxfeed;
if (plusSampleR < -maxfeed) plusSampleR = -maxfeed;
if (minusSampleR < -maxfeed) minusSampleR = -maxfeed;
if (inputSampleR > distSampleR) cheblevR += (minusSampleR*addShortBuss);
if (inputSampleR < -distSampleR) cheblevR -= (plusSampleR*addShortBuss);
if (cheblevR > 1.0) cheblevR = 1.0;
if (cheblevR < -1.0) cheblevR = -1.0;
bridgerectifier = fabs(inputSampleR);
if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633;
//max value for sine function
bridgerectifier = sin(bridgerectifier);
if (inputSampleR > 0.0) inputSampleR = bridgerectifier;
else inputSampleR = -bridgerectifier;
//apply the distortion transform for reals
inputSampleR /= (1.0+fabs(cheblevR*limiting));
inputSampleR += chebyshev;
//apply the correction measuresR
if (outlevel < 1.0) {
inputSampleL *= outlevel;
inputSampleR *= outlevel;
}
if (inputSampleL > 0.95) inputSampleL = 0.95;
if (inputSampleL < -0.95) inputSampleL = -0.95;
if (inputSampleR > 0.95) inputSampleR = 0.95;
if (inputSampleR < -0.95) inputSampleR = -0.95;
//iron bar
//noise shaping to 64-bit floating point
if (fpFlip) {
fpTemp = inputSampleL;
fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew);
inputSampleL += fpNShapeLA;
fpTemp = inputSampleR;
fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew);
inputSampleR += fpNShapeRA;
}
else {
fpTemp = inputSampleL;
fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew);
inputSampleL += fpNShapeLB;
fpTemp = inputSampleR;
fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew);
inputSampleR += fpNShapeRB;
}
fpFlip = !fpFlip;
//end noise shaping on 64 bit output
*out1 = inputSampleL;
*out2 = inputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
}