airwindows/plugins/WinVST/Sweeten/SweetenProc.cpp
Christopher Johnson b329cc339b Creature
2023-08-19 19:00:00 -04:00

192 lines
9.1 KiB
C++
Executable file

/* ========================================
* Sweeten - Sweeten.h
* Copyright (c) airwindows, Airwindows uses the MIT license
* ======================================== */
#ifndef __Sweeten_H
#include "Sweeten.h"
#endif
void Sweeten::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();
int cycleEnd = floor(overallscale);
if (cycleEnd < 1) cycleEnd = 1;
if (cycleEnd > 4) cycleEnd = 4;
//this is going to be 2 for 88.1 or 96k, 3 for silly people, 4 for 176 or 192k
int sweetBits = 10-floor(A*10.0);
double sweet = 1.0;
switch (sweetBits)
{
case 11: sweet = 0.00048828125; break;
case 10: sweet = 0.0009765625; break;
case 9: sweet = 0.001953125; break;
case 8: sweet = 0.00390625; break;
case 7: sweet = 0.0078125; break;
case 6: sweet = 0.015625; break;
case 5: sweet = 0.03125; break;
case 4: sweet = 0.0625; break;
case 3: sweet = 0.125; break;
case 2: sweet = 0.25; break;
case 1: sweet = 0.5; break;
case 0: sweet = 1.0; break;
case -1: sweet = 2.0; break;
} //now we have our input trim
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;
double sweetSample = inputSampleL;
double sv = sweetSample; sweetSample = (sweetSample + savg[0]) * 0.5; savg[0] = sv;
if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[1]) * 0.5; savg[1] = sv;
if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[2]) * 0.5; savg[2] = sv;
if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[3]) * 0.5; savg[3] = sv;}
} //if undersampling code is present, this gives a simple averaging that stacks up
} //when high sample rates are present, for a more intense aliasing reduction. PRE nonlinearity
sweetSample = (sweetSample*sweetSample*sweet); //second harmonic (nonlinearity)
sv = sweetSample; sweetSample = (sweetSample + savg[4]) * 0.5; savg[4] = sv;
if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[5]) * 0.5; savg[5] = sv;
if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[6]) * 0.5; savg[6] = sv;
if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[7]) * 0.5; savg[7] = sv;}
} //if undersampling code is present, this gives a simple averaging that stacks up
} //when high sample rates are present, for a more intense aliasing reduction. POST nonlinearity
inputSampleL -= sweetSample; //apply the filtered second harmonic correction
sweetSample = inputSampleR;
sv = sweetSample; sweetSample = (sweetSample + savg[8]) * 0.5; savg[8] = sv;
if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[9]) * 0.5; savg[9] = sv;
if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[10]) * 0.5; savg[10] = sv;
if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[11]) * 0.5; savg[11] = sv;}
} //if undersampling code is present, this gives a simple averaging that stacks up
} //when high sample rates are present, for a more intense aliasing reduction. PRE nonlinearity
sweetSample = (sweetSample*sweetSample*sweet); //second harmonic (nonlinearity)
sv = sweetSample; sweetSample = (sweetSample + savg[12]) * 0.5; savg[12] = sv;
if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[13]) * 0.5; savg[13] = sv;
if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[14]) * 0.5; savg[14] = sv;
if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[15]) * 0.5; savg[15] = sv;}
} //if undersampling code is present, this gives a simple averaging that stacks up
} //when high sample rates are present, for a more intense aliasing reduction. POST nonlinearity
inputSampleR -= sweetSample; //apply the filtered second harmonic correction
//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 Sweeten::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();
int cycleEnd = floor(overallscale);
if (cycleEnd < 1) cycleEnd = 1;
if (cycleEnd > 4) cycleEnd = 4;
//this is going to be 2 for 88.1 or 96k, 3 for silly people, 4 for 176 or 192k
int sweetBits = 10-floor(A*10.0);
double sweet = 1.0;
switch (sweetBits)
{
case 11: sweet = 0.00048828125; break;
case 10: sweet = 0.0009765625; break;
case 9: sweet = 0.001953125; break;
case 8: sweet = 0.00390625; break;
case 7: sweet = 0.0078125; break;
case 6: sweet = 0.015625; break;
case 5: sweet = 0.03125; break;
case 4: sweet = 0.0625; break;
case 3: sweet = 0.125; break;
case 2: sweet = 0.25; break;
case 1: sweet = 0.5; break;
case 0: sweet = 1.0; break;
case -1: sweet = 2.0; break;
} //now we have our input trim
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;
double sweetSample = inputSampleL;
double sv = sweetSample; sweetSample = (sweetSample + savg[0]) * 0.5; savg[0] = sv;
if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[1]) * 0.5; savg[1] = sv;
if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[2]) * 0.5; savg[2] = sv;
if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[3]) * 0.5; savg[3] = sv;}
} //if undersampling code is present, this gives a simple averaging that stacks up
} //when high sample rates are present, for a more intense aliasing reduction. PRE nonlinearity
sweetSample = (sweetSample*sweetSample*sweet); //second harmonic (nonlinearity)
sv = sweetSample; sweetSample = (sweetSample + savg[4]) * 0.5; savg[4] = sv;
if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[5]) * 0.5; savg[5] = sv;
if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[6]) * 0.5; savg[6] = sv;
if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[7]) * 0.5; savg[7] = sv;}
} //if undersampling code is present, this gives a simple averaging that stacks up
} //when high sample rates are present, for a more intense aliasing reduction. POST nonlinearity
inputSampleL -= sweetSample; //apply the filtered second harmonic correction
sweetSample = inputSampleR;
sv = sweetSample; sweetSample = (sweetSample + savg[8]) * 0.5; savg[8] = sv;
if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[9]) * 0.5; savg[9] = sv;
if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[10]) * 0.5; savg[10] = sv;
if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[11]) * 0.5; savg[11] = sv;}
} //if undersampling code is present, this gives a simple averaging that stacks up
} //when high sample rates are present, for a more intense aliasing reduction. PRE nonlinearity
sweetSample = (sweetSample*sweetSample*sweet); //second harmonic (nonlinearity)
sv = sweetSample; sweetSample = (sweetSample + savg[12]) * 0.5; savg[12] = sv;
if (cycleEnd > 1) {sv = sweetSample; sweetSample = (sweetSample + savg[13]) * 0.5; savg[13] = sv;
if (cycleEnd > 2) {sv = sweetSample; sweetSample = (sweetSample + savg[14]) * 0.5; savg[14] = sv;
if (cycleEnd > 3) {sv = sweetSample; sweetSample = (sweetSample + savg[15]) * 0.5; savg[15] = sv;}
} //if undersampling code is present, this gives a simple averaging that stacks up
} //when high sample rates are present, for a more intense aliasing reduction. POST nonlinearity
inputSampleR -= sweetSample; //apply the filtered second harmonic correction
//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++;
}
}