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airwindows/plugins/MacVST/DubPlate2/source/DubPlate2Proc.cpp
Christopher Johnson 21901acae0 DubPlate2
2025-03-30 15:34:42 -04:00

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/* ========================================
* DubPlate2 - DubPlate2.h
* Copyright (c) airwindows, Airwindows uses the MIT license
* ======================================== */
#ifndef __DubPlate2_H
#include "DubPlate2.h"
#endif
void DubPlate2::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
VstInt32 inFramesToProcess = sampleFrames; //vst doesn't give us this as a separate variable so we'll make it
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
double inputGain = pow(A*2.0,2.0);
double trebleGain = pow(B*2.0,2.0);
double trebleFreq = ((2000.0*trebleGain)+200.0)/getSampleRate();
if (trebleFreq > 0.45) trebleFreq = 0.45;
baxH[bax_freq] = trebleFreq;
baxH[bax_reso] = 0.57735026919; //bessel second order
double K = tan(M_PI * baxH[bax_freq]); //lowpass
double norm = 1.0 / (1.0 + K / baxH[bax_reso] + K * K);
baxH[bax_a0] = K * K * norm;
baxH[bax_a1] = 2.0 * baxH[bax_a0];
baxH[bax_a2] = baxH[bax_a0];
baxH[bax_b1] = 2.0 * (K * K - 1.0) * norm;
baxH[bax_b2] = (1.0 - K / baxH[bax_reso] + K * K) * norm;
//end bax highpass
double bassGain = pow(C*2.0,2.0);
double bassFreq = pow((1.0-C)*2.0,2.0);
bassFreq = ((2000.0*bassFreq)+200.0)/getSampleRate();
if (bassFreq > 0.45) bassFreq = 0.45;
baxL[bax_freq] = bassFreq;
baxL[bax_reso] = 0.57735026919; //bessel second order
K = tan(M_PI * baxL[bax_freq]); //lowpass
norm = 1.0 / (1.0 + K / baxL[bax_reso] + K * K);
baxL[bax_a0] = K * K * norm;
baxL[bax_a1] = 2.0 * baxL[bax_a0];
baxL[bax_a2] = baxL[bax_a0];
baxL[bax_b1] = 2.0 * (K * K - 1.0) * norm;
baxL[bax_b2] = (1.0 - K / baxL[bax_reso] + K * K) * norm;
//end bax lowpass
double iirSide = 0.01862 / overallscale;
double iirMid = 0.01102 / overallscale;
double threshSinewA = 0.1442 / overallscale;
double threshSinewB = 0.0274 / overallscale;
while (--sampleFrames >= 0)
{
double inputSampleL = *in1;
double inputSampleR = *in2;
if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
inputSampleL = sin(fmax(fmin(inputSampleL*inputGain,M_PI_2),-M_PI_2));
inputSampleR = sin(fmax(fmin(inputSampleR*inputGain,M_PI_2),-M_PI_2));
//encode Console5: good cleanness
double trebleSampleL;
double trebleSampleR;
double bassSampleL;
double bassSampleR;
if (flip) {
trebleSampleL = (inputSampleL * baxH[bax_a0]) + baxH[bax_sLA1];
baxH[bax_sLA1] = (inputSampleL * baxH[bax_a1]) - (trebleSampleL * baxH[bax_b1]) + baxH[bax_sLA2];
baxH[bax_sLA2] = (inputSampleL * baxH[bax_a2]) - (trebleSampleL * baxH[bax_b2]);
trebleSampleL = inputSampleL - trebleSampleL;
trebleSampleR = (inputSampleR * baxH[bax_a0]) + baxH[bax_sRA1];
baxH[bax_sRA1] = (inputSampleR * baxH[bax_a1]) - (trebleSampleR * baxH[bax_b1]) + baxH[bax_sRA2];
baxH[bax_sRA2] = (inputSampleR * baxH[bax_a2]) - (trebleSampleR * baxH[bax_b2]);
trebleSampleR = inputSampleR - trebleSampleR;
bassSampleL = (inputSampleL * baxL[bax_a0]) + baxL[bax_sLA1];
baxL[bax_sLA1] = (inputSampleL * baxL[bax_a1]) - (bassSampleL * baxL[bax_b1]) + baxL[bax_sLA2];
baxL[bax_sLA2] = (inputSampleL * baxL[bax_a2]) - (bassSampleL * baxL[bax_b2]);
bassSampleR = (inputSampleR * baxL[bax_a0]) + baxL[bax_sRA1];
baxL[bax_sRA1] = (inputSampleR * baxL[bax_a1]) - (bassSampleR * baxL[bax_b1]) + baxL[bax_sRA2];
baxL[bax_sRA2] = (inputSampleR * baxL[bax_a2]) - (bassSampleR * baxL[bax_b2]);
} else {
trebleSampleL = (inputSampleL * baxH[bax_a0]) + baxH[bax_sLB1];
baxH[bax_sLB1] = (inputSampleL * baxH[bax_a1]) - (trebleSampleL * baxH[bax_b1]) + baxH[bax_sLB2];
baxH[bax_sLB2] = (inputSampleL * baxH[bax_a2]) - (trebleSampleL * baxH[bax_b2]);
trebleSampleL = inputSampleL - trebleSampleL;
trebleSampleR = (inputSampleR * baxH[bax_a0]) + baxH[bax_sRB1];
baxH[bax_sRB1] = (inputSampleR * baxH[bax_a1]) - (trebleSampleR * baxH[bax_b1]) + baxH[bax_sRB2];
baxH[bax_sRB2] = (inputSampleR * baxH[bax_a2]) - (trebleSampleR * baxH[bax_b2]);
trebleSampleR = inputSampleR - trebleSampleR;
bassSampleL = (inputSampleL * baxL[bax_a0]) + baxL[bax_sLB1];
baxL[bax_sLB1] = (inputSampleL * baxL[bax_a1]) - (bassSampleL * baxL[bax_b1]) + baxL[bax_sLB2];
baxL[bax_sLB2] = (inputSampleL * baxL[bax_a2]) - (bassSampleL * baxL[bax_b2]);
bassSampleR = (inputSampleR * baxL[bax_a0]) + baxL[bax_sRB1];
baxL[bax_sRB1] = (inputSampleR * baxL[bax_a1]) - (bassSampleR * baxL[bax_b1]) + baxL[bax_sRB2];
baxL[bax_sRB2] = (inputSampleR * baxL[bax_a2]) - (bassSampleR * baxL[bax_b2]);
}
trebleSampleL *= trebleGain;
trebleSampleR *= trebleGain;
bassSampleL *= bassGain;
bassSampleR *= bassGain;
inputSampleL = bassSampleL + trebleSampleL; //interleaved biquad
inputSampleR = bassSampleR + trebleSampleR; //interleaved biquad
inputSampleL = asin(fmax(fmin(inputSampleL,0.99999),-0.99999));
inputSampleR = asin(fmax(fmin(inputSampleR,0.99999),-0.99999));
//amplitude aspect
double mid = inputSampleL + inputSampleR;
double side = inputSampleL - inputSampleR;
//assign mid and side.Between these sections, you can do mid/side processing
double temp = side;
double correction;
if (flip) {
iirA = (iirA * (1.0 - iirSide)) + (temp * iirSide); temp -= iirA; correction = iirA;
} else {
iirB = (iirB * (1.0 - iirSide)) + (temp * iirSide); temp -= iirB; correction = iirB;
}
iirC = (iirC * (1.0 - iirSide)) + (temp * iirSide); temp -= iirC; correction += (iirC * 0.162);
side -= sin(correction);
iirD = (iirD * (1.0 - iirMid)) + (mid * iirMid);
mid -= sin(iirD);
//gonna cut those lows a hair
inputSampleL = (mid+side)/2.0;
inputSampleR = (mid-side)/2.0;
//unassign mid and side
temp = inputSampleL;
double sinew = threshSinewA * cos(lastSinewAL*lastSinewAL);
if (inputSampleL - lastSinewAL > sinew) temp = lastSinewAL + sinew;
if (-(inputSampleL - lastSinewAL) > sinew) temp = lastSinewAL - sinew;
lastSinewAL = temp;
if (lastSinewAL > 1.0) lastSinewAL = 1.0;
if (lastSinewAL < -1.0) lastSinewAL = -1.0;
inputSampleL = (inputSampleL * 0.5)+(lastSinewAL * 0.5);
sinew = threshSinewB * cos(lastSinewBL*lastSinewBL);
if (inputSampleL - lastSinewBL > sinew) temp = lastSinewBL + sinew;
if (-(inputSampleL - lastSinewBL) > sinew) temp = lastSinewBL - sinew;
lastSinewBL = temp;
if (lastSinewBL > 1.0) lastSinewBL = 1.0;
if (lastSinewBL < -1.0) lastSinewBL = -1.0;
inputSampleL = (inputSampleL * 0.5)+(lastSinewBL * 0.5);
temp = inputSampleR;
sinew = threshSinewA * cos(lastSinewAR*lastSinewAR);
if (inputSampleR - lastSinewAR > sinew) temp = lastSinewAR + sinew;
if (-(inputSampleR - lastSinewAR) > sinew) temp = lastSinewAR - sinew;
lastSinewAR = temp;
if (lastSinewAR > 1.0) lastSinewAR = 1.0;
if (lastSinewAR < -1.0) lastSinewAR = -1.0;
inputSampleR = (inputSampleR * 0.5)+(lastSinewAR * 0.5);
sinew = threshSinewB * cos(lastSinewBR*lastSinewBR);
if (inputSampleR - lastSinewBR > sinew) temp = lastSinewBR + sinew;
if (-(inputSampleR - lastSinewBR) > sinew) temp = lastSinewBR - sinew;
lastSinewBR = temp;
if (lastSinewBR > 1.0) lastSinewBR = 1.0;
if (lastSinewBR < -1.0) lastSinewBR = -1.0;
inputSampleR = (inputSampleR * 0.5)+(lastSinewBR * 0.5);
//run Sinew to stop excess slews, two layers to make it more audible
flip = !flip; //both for Baxandall3 and DubPlate highpass
//begin 32 bit stereo floating point dither
int expon; frexpf((float)inputSampleL, &expon);
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
frexpf((float)inputSampleR, &expon);
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
//end 32 bit stereo floating point dither
*out1 = inputSampleL;
*out2 = inputSampleR;
in1++;
in2++;
out1++;
out2++;
}
}
void DubPlate2::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
VstInt32 inFramesToProcess = sampleFrames; //vst doesn't give us this as a separate variable so we'll make it
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
double inputGain = pow(A*2.0,2.0);
double trebleGain = pow(B*2.0,2.0);
double trebleFreq = ((2000.0*trebleGain)+200.0)/getSampleRate();
if (trebleFreq > 0.45) trebleFreq = 0.45;
baxH[bax_freq] = trebleFreq;
baxH[bax_reso] = 0.57735026919; //bessel second order
double K = tan(M_PI * baxH[bax_freq]); //lowpass
double norm = 1.0 / (1.0 + K / baxH[bax_reso] + K * K);
baxH[bax_a0] = K * K * norm;
baxH[bax_a1] = 2.0 * baxH[bax_a0];
baxH[bax_a2] = baxH[bax_a0];
baxH[bax_b1] = 2.0 * (K * K - 1.0) * norm;
baxH[bax_b2] = (1.0 - K / baxH[bax_reso] + K * K) * norm;
//end bax highpass
double bassGain = pow(C*2.0,2.0);
double bassFreq = pow((1.0-C)*2.0,2.0);
bassFreq = ((2000.0*bassFreq)+200.0)/getSampleRate();
if (bassFreq > 0.45) bassFreq = 0.45;
baxL[bax_freq] = bassFreq;
baxL[bax_reso] = 0.57735026919; //bessel second order
K = tan(M_PI * baxL[bax_freq]); //lowpass
norm = 1.0 / (1.0 + K / baxL[bax_reso] + K * K);
baxL[bax_a0] = K * K * norm;
baxL[bax_a1] = 2.0 * baxL[bax_a0];
baxL[bax_a2] = baxL[bax_a0];
baxL[bax_b1] = 2.0 * (K * K - 1.0) * norm;
baxL[bax_b2] = (1.0 - K / baxL[bax_reso] + K * K) * norm;
//end bax lowpass
double iirSide = 0.01862 / overallscale;
double iirMid = 0.01102 / overallscale;
double threshSinewA = 0.1442 / overallscale;
double threshSinewB = 0.0274 / overallscale;
while (--sampleFrames >= 0)
{
double inputSampleL = *in1;
double inputSampleR = *in2;
if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
inputSampleL = sin(fmax(fmin(inputSampleL*inputGain,M_PI_2),-M_PI_2));
inputSampleR = sin(fmax(fmin(inputSampleR*inputGain,M_PI_2),-M_PI_2));
//encode Console5: good cleanness
double trebleSampleL;
double trebleSampleR;
double bassSampleL;
double bassSampleR;
if (flip) {
trebleSampleL = (inputSampleL * baxH[bax_a0]) + baxH[bax_sLA1];
baxH[bax_sLA1] = (inputSampleL * baxH[bax_a1]) - (trebleSampleL * baxH[bax_b1]) + baxH[bax_sLA2];
baxH[bax_sLA2] = (inputSampleL * baxH[bax_a2]) - (trebleSampleL * baxH[bax_b2]);
trebleSampleL = inputSampleL - trebleSampleL;
trebleSampleR = (inputSampleR * baxH[bax_a0]) + baxH[bax_sRA1];
baxH[bax_sRA1] = (inputSampleR * baxH[bax_a1]) - (trebleSampleR * baxH[bax_b1]) + baxH[bax_sRA2];
baxH[bax_sRA2] = (inputSampleR * baxH[bax_a2]) - (trebleSampleR * baxH[bax_b2]);
trebleSampleR = inputSampleR - trebleSampleR;
bassSampleL = (inputSampleL * baxL[bax_a0]) + baxL[bax_sLA1];
baxL[bax_sLA1] = (inputSampleL * baxL[bax_a1]) - (bassSampleL * baxL[bax_b1]) + baxL[bax_sLA2];
baxL[bax_sLA2] = (inputSampleL * baxL[bax_a2]) - (bassSampleL * baxL[bax_b2]);
bassSampleR = (inputSampleR * baxL[bax_a0]) + baxL[bax_sRA1];
baxL[bax_sRA1] = (inputSampleR * baxL[bax_a1]) - (bassSampleR * baxL[bax_b1]) + baxL[bax_sRA2];
baxL[bax_sRA2] = (inputSampleR * baxL[bax_a2]) - (bassSampleR * baxL[bax_b2]);
} else {
trebleSampleL = (inputSampleL * baxH[bax_a0]) + baxH[bax_sLB1];
baxH[bax_sLB1] = (inputSampleL * baxH[bax_a1]) - (trebleSampleL * baxH[bax_b1]) + baxH[bax_sLB2];
baxH[bax_sLB2] = (inputSampleL * baxH[bax_a2]) - (trebleSampleL * baxH[bax_b2]);
trebleSampleL = inputSampleL - trebleSampleL;
trebleSampleR = (inputSampleR * baxH[bax_a0]) + baxH[bax_sRB1];
baxH[bax_sRB1] = (inputSampleR * baxH[bax_a1]) - (trebleSampleR * baxH[bax_b1]) + baxH[bax_sRB2];
baxH[bax_sRB2] = (inputSampleR * baxH[bax_a2]) - (trebleSampleR * baxH[bax_b2]);
trebleSampleR = inputSampleR - trebleSampleR;
bassSampleL = (inputSampleL * baxL[bax_a0]) + baxL[bax_sLB1];
baxL[bax_sLB1] = (inputSampleL * baxL[bax_a1]) - (bassSampleL * baxL[bax_b1]) + baxL[bax_sLB2];
baxL[bax_sLB2] = (inputSampleL * baxL[bax_a2]) - (bassSampleL * baxL[bax_b2]);
bassSampleR = (inputSampleR * baxL[bax_a0]) + baxL[bax_sRB1];
baxL[bax_sRB1] = (inputSampleR * baxL[bax_a1]) - (bassSampleR * baxL[bax_b1]) + baxL[bax_sRB2];
baxL[bax_sRB2] = (inputSampleR * baxL[bax_a2]) - (bassSampleR * baxL[bax_b2]);
}
trebleSampleL *= trebleGain;
trebleSampleR *= trebleGain;
bassSampleL *= bassGain;
bassSampleR *= bassGain;
inputSampleL = bassSampleL + trebleSampleL; //interleaved biquad
inputSampleR = bassSampleR + trebleSampleR; //interleaved biquad
inputSampleL = asin(fmax(fmin(inputSampleL,0.99999),-0.99999));
inputSampleR = asin(fmax(fmin(inputSampleR,0.99999),-0.99999));
//amplitude aspect
double mid = inputSampleL + inputSampleR;
double side = inputSampleL - inputSampleR;
//assign mid and side.Between these sections, you can do mid/side processing
double temp = side;
double correction;
if (flip) {
iirA = (iirA * (1.0 - iirSide)) + (temp * iirSide); temp -= iirA; correction = iirA;
} else {
iirB = (iirB * (1.0 - iirSide)) + (temp * iirSide); temp -= iirB; correction = iirB;
}
iirC = (iirC * (1.0 - iirSide)) + (temp * iirSide); temp -= iirC; correction += (iirC * 0.162);
side -= sin(correction);
iirD = (iirD * (1.0 - iirMid)) + (mid * iirMid);
mid -= sin(iirD);
//gonna cut those lows a hair
inputSampleL = (mid+side)/2.0;
inputSampleR = (mid-side)/2.0;
//unassign mid and side
temp = inputSampleL;
double sinew = threshSinewA * cos(lastSinewAL*lastSinewAL);
if (inputSampleL - lastSinewAL > sinew) temp = lastSinewAL + sinew;
if (-(inputSampleL - lastSinewAL) > sinew) temp = lastSinewAL - sinew;
lastSinewAL = temp;
if (lastSinewAL > 1.0) lastSinewAL = 1.0;
if (lastSinewAL < -1.0) lastSinewAL = -1.0;
inputSampleL = (inputSampleL * 0.5)+(lastSinewAL * 0.5);
sinew = threshSinewB * cos(lastSinewBL*lastSinewBL);
if (inputSampleL - lastSinewBL > sinew) temp = lastSinewBL + sinew;
if (-(inputSampleL - lastSinewBL) > sinew) temp = lastSinewBL - sinew;
lastSinewBL = temp;
if (lastSinewBL > 1.0) lastSinewBL = 1.0;
if (lastSinewBL < -1.0) lastSinewBL = -1.0;
inputSampleL = (inputSampleL * 0.5)+(lastSinewBL * 0.5);
temp = inputSampleR;
sinew = threshSinewA * cos(lastSinewAR*lastSinewAR);
if (inputSampleR - lastSinewAR > sinew) temp = lastSinewAR + sinew;
if (-(inputSampleR - lastSinewAR) > sinew) temp = lastSinewAR - sinew;
lastSinewAR = temp;
if (lastSinewAR > 1.0) lastSinewAR = 1.0;
if (lastSinewAR < -1.0) lastSinewAR = -1.0;
inputSampleR = (inputSampleR * 0.5)+(lastSinewAR * 0.5);
sinew = threshSinewB * cos(lastSinewBR*lastSinewBR);
if (inputSampleR - lastSinewBR > sinew) temp = lastSinewBR + sinew;
if (-(inputSampleR - lastSinewBR) > sinew) temp = lastSinewBR - sinew;
lastSinewBR = temp;
if (lastSinewBR > 1.0) lastSinewBR = 1.0;
if (lastSinewBR < -1.0) lastSinewBR = -1.0;
inputSampleR = (inputSampleR * 0.5)+(lastSinewBR * 0.5);
//run Sinew to stop excess slews, two layers to make it more audible
flip = !flip; //both for Baxandall3 and DubPlate highpass
//begin 64 bit stereo floating point dither
//int expon; frexp((double)inputSampleL, &expon);
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
//inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
//frexp((double)inputSampleR, &expon);
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
//inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
//end 64 bit stereo floating point dither
*out1 = inputSampleL;
*out2 = inputSampleR;
in1++;
in2++;
out1++;
out2++;
}
}
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