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airwindows/plugins/MacVST/ConsoleLAChannel/source/ConsoleLAChannelProc.cpp
Christopher Johnson 5ee0c5711f ZOutputStage
2023-12-16 18:20:00 -05:00

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/* ========================================
* ConsoleLAChannel - ConsoleLAChannel.h
* Copyright (c) airwindows, Airwindows uses the MIT license
* ======================================== */
#ifndef __ConsoleLAChannel_H
#include "ConsoleLAChannel.h"
#endif
void ConsoleLAChannel::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();
int cycleEnd = floor(overallscale);
if (cycleEnd < 1) cycleEnd = 1;
if (cycleEnd > 4) cycleEnd = 4;
int limit = 4*cycleEnd;
double divisor = 2.0/limit;
double treble = (A*6.0)-3.0;
midA = midB;
midB = (B*6.0)-3.0;
bassA = bassB;
bassB = (C*6.0)-3.0;
//these should stack to go up to -3.0 to 3.0
if (treble < 0.0) treble /= 3.0;
if (midB < 0.0) midB /= 3.0;
if (bassB < 0.0) bassB /= 3.0;
//and then become -1.0 to 3.0;
//there will be successive sin/cos stages w. dry/wet in these
double freqMid = 0.024;
switch (cycleEnd)
{
case 1: //base sample rate, no change
break;
case 2: //96k tier
freqMid = 0.012;
break;
case 3: //192k tier
freqMid = 0.006;
break;
}
int bitshiftL = 0;
int bitshiftR = 0;
double panControl = (D*2.0)-1.0; //-1.0 to 1.0
double panAttenuation = (1.0-fabs(panControl));
int panBits = 20; //start centered
if (panAttenuation > 0.0) panBits = floor(1.0 / panAttenuation);
if (panControl > 0.25) bitshiftL += panBits;
if (panControl < -0.25) bitshiftR += panBits;
if (bitshiftL < 0) bitshiftL = 0; if (bitshiftL > 17) bitshiftL = 17;
if (bitshiftR < 0) bitshiftR = 0; if (bitshiftR > 17) bitshiftR = 17;
double gainL = 1.0;
double gainR = 1.0;
switch (bitshiftL)
{
case 17: gainL = 0.0; break;
case 16: gainL = 0.0000152587890625; break;
case 15: gainL = 0.000030517578125; break;
case 14: gainL = 0.00006103515625; break;
case 13: gainL = 0.0001220703125; break;
case 12: gainL = 0.000244140625; break;
case 11: gainL = 0.00048828125; break;
case 10: gainL = 0.0009765625; break;
case 9: gainL = 0.001953125; break;
case 8: gainL = 0.00390625; break;
case 7: gainL = 0.0078125; break;
case 6: gainL = 0.015625; break;
case 5: gainL = 0.03125; break;
case 4: gainL = 0.0625; break;
case 3: gainL = 0.125; break;
case 2: gainL = 0.25; break;
case 1: gainL = 0.5; break;
case 0: break;
}
switch (bitshiftR)
{
case 17: gainR = 0.0; break;
case 16: gainR = 0.0000152587890625; break;
case 15: gainR = 0.000030517578125; break;
case 14: gainR = 0.00006103515625; break;
case 13: gainR = 0.0001220703125; break;
case 12: gainR = 0.000244140625; break;
case 11: gainR = 0.00048828125; break;
case 10: gainR = 0.0009765625; break;
case 9: gainR = 0.001953125; break;
case 8: gainR = 0.00390625; break;
case 7: gainR = 0.0078125; break;
case 6: gainR = 0.015625; break;
case 5: gainR = 0.03125; break;
case 4: gainR = 0.0625; break;
case 3: gainR = 0.125; break;
case 2: gainR = 0.25; break;
case 1: gainR = 0.5; break;
case 0: break;
}
gainA = gainB;
gainB = E*2.0; //smoothed master fader from Z2 filters
//BitShiftGain pre gain trim goes here
double subTrim = 0.0011 / 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;
double temp = (double)sampleFrames/inFramesToProcess;
double gain = (gainA*temp)+(gainB*(1.0-temp));
double mid = (midA*temp)+(midB*(1.0-temp));
double bass = (bassA*temp)+(bassB*(1.0-temp));
//begin Hull2 Treble
hullp--; if (hullp < 0) hullp += 60;
hullL[hullp] = hullL[hullp+60] = inputSampleL;
hullR[hullp] = hullR[hullp+60] = inputSampleR;
int x = hullp;
double bassL = 0.0;
double bassR = 0.0;
while (x < hullp+(limit/2)) {
bassL += hullL[x] * divisor;
bassR += hullR[x] * divisor;
x++;
}
bassL += bassL * 0.125;
bassR += bassR * 0.125;
while (x < hullp+limit) {
bassL -= hullL[x] * 0.125 * divisor;
bassR -= hullR[x] * 0.125 * divisor;
x++;
}
hullL[hullp+20] = hullL[hullp+80] = bassL;
hullR[hullp+20] = hullR[hullp+80] = bassR;
x = hullp+20;
bassL = bassR = 0.0;
while (x < hullp+20+(limit/2)) {
bassL += hullL[x] * divisor;
bassR += hullR[x] * divisor;
x++;
}
bassL += bassL * 0.125;
bassR += bassR * 0.125;
while (x < hullp+20+limit) {
bassL -= hullL[x] * 0.125 * divisor;
bassR -= hullR[x] * 0.125 * divisor;
x++;
}
hullL[hullp+40] = hullL[hullp+100] = bassL;
hullR[hullp+40] = hullR[hullp+100] = bassR;
x = hullp+40;
bassL = bassR = 0.0;
while (x < hullp+40+(limit/2)) {
bassL += hullL[x] * divisor;
bassR += hullR[x] * divisor;
x++;
}
bassL += bassL * 0.125;
bassR += bassR * 0.125;
while (x < hullp+40+limit) {
bassL -= hullL[x] * 0.125 * divisor;
bassR -= hullR[x] * 0.125 * divisor;
x++;
}
double trebleL = inputSampleL - bassL; inputSampleL = bassL;
double trebleR = inputSampleR - bassR; inputSampleR = bassR;
//end Hull2 treble
//begin Pear filter stages
//at this point 'bass' is actually still mid and bass
double slew = ((bassL - pearB[0]) + pearB[1])*freqMid*0.5;
pearB[0] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[0] + pearB[1]));
pearB[1] = slew; slew = ((bassR - pearB[2]) + pearB[3])*freqMid*0.5;
pearB[2] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[2] + pearB[3]));
pearB[3] = slew; slew = ((bassL - pearB[4]) + pearB[5])*freqMid*0.5;
pearB[4] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[4] + pearB[5]));
pearB[5] = slew; slew = ((bassR - pearB[6]) + pearB[7])*freqMid*0.5;
pearB[6] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[6] + pearB[7]));
pearB[7] = slew; slew = ((bassL - pearB[8]) + pearB[9])*freqMid*0.5;
pearB[8] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[8] + pearB[9]));
pearB[9] = slew; slew = ((bassR - pearB[10]) + pearB[11])*freqMid*0.5;
pearB[10] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[10] + pearB[11]));
pearB[11] = slew; slew = ((bassL - pearB[12]) + pearB[13])*freqMid*0.5;
pearB[12] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[12] + pearB[13]));
pearB[13] = slew; slew = ((bassR - pearB[14]) + pearB[15])*freqMid*0.5;
pearB[14] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[14] + pearB[15]));
pearB[15] = slew; slew = ((bassL - pearB[16]) + pearB[17])*freqMid*0.5;
pearB[16] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[16] + pearB[17]));
pearB[17] = slew; slew = ((bassR - pearB[18]) + pearB[19])*freqMid*0.5;
pearB[18] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[18] + pearB[19]));
pearB[19] = slew;
double midL = inputSampleL - bassL;
double midR = inputSampleR - bassR;
//we now have three bands out of hull and pear filters
double w = 0.0; //filter into bands, apply the sin/cos to each band
if (treble > 0.0) {
w = treble; if (w > 1.0) w = 1.0;
trebleL = (trebleL*(1.0-w)) + (sin(trebleL*M_PI_2)*treble);
trebleR = (trebleR*(1.0-w)) + (sin(trebleR*M_PI_2)*treble);
}
if (treble < 0.0) {
if (trebleL > 1.0) trebleL = 1.0; if (trebleL < -1.0) trebleL = -1.0;
if (trebleR > 1.0) trebleR = 1.0; if (trebleR < -1.0) trebleR = -1.0;
w = -treble; if (w > 1.0) w = 1.0;
if (trebleL > 0) trebleL = (trebleL*(1.0-w))+((1.0-cos(trebleL*w))*(1.0-w));
else trebleL = (trebleL*(1.0-w))+((-1.0+cos(-trebleL*w))*(1.0-w));
if (trebleR > 0) trebleR = (trebleR*(1.0-w))+((1.0-cos(trebleR*w))*(1.0-w));
else trebleR = (trebleR*(1.0-w))+((-1.0+cos(-trebleR*w))*(1.0-w));
} //cosine stages for EQ or expansion
if (midL > 1.0) midL = 1.0; if (midL < -1.0) midL = -1.0;
if (midR > 1.0) midR = 1.0; if (midR < -1.0) midR = -1.0;
if (mid > 0.0) {
w = mid; if (w > 1.0) w = 1.0;
midL = (midL*(1.0-w)) + (sin(midL*M_PI_2)*mid);
midR = (midR*(1.0-w)) + (sin(midR*M_PI_2)*mid);
}
if (mid < 0.0) {
w = -mid; if (w > 1.0) w = 1.0;
if (midL > 0) midL = (midL*(1.0-w))+((1.0-cos(midL*w))*(1.0-w));
else midL = (midL*(1.0-w))+((-1.0+cos(-midL*w))*(1.0-w));
if (midR > 0) midR = (midR*(1.0-w))+((1.0-cos(midR*w))*(1.0-w));
else midR = (midR*(1.0-w))+((-1.0+cos(-midR*w))*(1.0-w));
} //cosine stages for EQ or expansion
if (bassL > 1.0) bassL = 1.0; if (bassL < -1.0) bassL = -1.0;
if (bassR > 1.0) bassR = 1.0; if (bassR < -1.0) bassR = -1.0;
if (bass > 0.0) {
w = bass; if (w > 1.0) w = 1.0;
bassL = (bassL*(1.0-w)) + (sin(bassL*M_PI_2)*bass);
bassR = (bassR*(1.0-w)) + (sin(bassR*M_PI_2)*bass);
}
if (bass < 0.0) {
w = -bass; if (w > 1.0) w = 1.0;
if (bassL > 0) bassL = (bassL*(1.0-w))+((1.0-cos(bassL*w))*(1.0-w));
else bassL = (bassL*(1.0-w))+((-1.0+cos(-bassL*w))*(1.0-w));
if (bassR > 0) bassR = (bassR*(1.0-w))+((1.0-cos(bassR*w))*(1.0-w));
else bassR = (bassR*(1.0-w))+((-1.0+cos(-bassR*w))*(1.0-w));
} //cosine stages for EQ or expansion
inputSampleL = (bassL + midL + trebleL)*gainL*gain;
inputSampleR = (bassR + midR + trebleR)*gainR*gain;
//applies BitShiftPan pan section, and smoothed fader gain
//begin SubTight section
double subSampleL = inputSampleL * subTrim;
double subSampleR = inputSampleR * subTrim;
double scale = 0.5+fabs(subSampleL*0.5);
subSampleL = (subAL+(sin(subAL-subSampleL)*scale));
subAL = subSampleL*scale;
scale = 0.5+fabs(subSampleR*0.5);
subSampleR = (subAR+(sin(subAR-subSampleR)*scale));
subAR = subSampleR*scale;
scale = 0.5+fabs(subSampleL*0.5);
subSampleL = (subBL+(sin(subBL-subSampleL)*scale));
subBL = subSampleL*scale;
scale = 0.5+fabs(subSampleR*0.5);
subSampleR = (subBR+(sin(subBR-subSampleR)*scale));
subBR = subSampleR*scale;
scale = 0.5+fabs(subSampleL*0.5);
subSampleL = (subCL+(sin(subCL-subSampleL)*scale));
subCL = subSampleL*scale;
scale = 0.5+fabs(subSampleR*0.5);
subSampleR = (subCR+(sin(subCR-subSampleR)*scale));
subCR = subSampleR*scale;
if (subSampleL > 0.25) subSampleL = 0.25;
if (subSampleL < -0.25) subSampleL = -0.25;
if (subSampleR > 0.25) subSampleR = 0.25;
if (subSampleR < -0.25) subSampleR = -0.25;
inputSampleL += (subSampleL*16.0);
inputSampleR += (subSampleR*16.0);
//end SubTight section
//begin Console7 Channel processing
if (inputSampleL > 1.097) inputSampleL = 1.097;
if (inputSampleL < -1.097) inputSampleL = -1.097;
if (inputSampleR > 1.097) inputSampleR = 1.097;
if (inputSampleR < -1.097) inputSampleR = -1.097;
inputSampleL = ((sin(inputSampleL*fabs(inputSampleL))/((fabs(inputSampleL) == 0.0) ?1:fabs(inputSampleL)))*0.8)+(sin(inputSampleL)*0.2);
inputSampleR = ((sin(inputSampleR*fabs(inputSampleR))/((fabs(inputSampleR) == 0.0) ?1:fabs(inputSampleR)))*0.8)+(sin(inputSampleR)*0.2);
//this is a version of Spiral blended 80/20 with regular Density.
//It's blending between two different harmonics in the overtones of the algorithm
//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 ConsoleLAChannel::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();
int cycleEnd = floor(overallscale);
if (cycleEnd < 1) cycleEnd = 1;
if (cycleEnd > 4) cycleEnd = 4;
int limit = 4*cycleEnd;
double divisor = 2.0/limit;
double treble = (A*6.0)-3.0;
midA = midB;
midB = (B*6.0)-3.0;
bassA = bassB;
bassB = (C*6.0)-3.0;
//these should stack to go up to -3.0 to 3.0
if (treble < 0.0) treble /= 3.0;
if (midB < 0.0) midB /= 3.0;
if (bassB < 0.0) bassB /= 3.0;
//and then become -1.0 to 3.0;
//there will be successive sin/cos stages w. dry/wet in these
double freqMid = 0.024;
switch (cycleEnd)
{
case 1: //base sample rate, no change
break;
case 2: //96k tier
freqMid = 0.012;
break;
case 3: //192k tier
freqMid = 0.006;
break;
}
int bitshiftL = 0;
int bitshiftR = 0;
double panControl = (D*2.0)-1.0; //-1.0 to 1.0
double panAttenuation = (1.0-fabs(panControl));
int panBits = 20; //start centered
if (panAttenuation > 0.0) panBits = floor(1.0 / panAttenuation);
if (panControl > 0.25) bitshiftL += panBits;
if (panControl < -0.25) bitshiftR += panBits;
if (bitshiftL < 0) bitshiftL = 0; if (bitshiftL > 17) bitshiftL = 17;
if (bitshiftR < 0) bitshiftR = 0; if (bitshiftR > 17) bitshiftR = 17;
double gainL = 1.0;
double gainR = 1.0;
switch (bitshiftL)
{
case 17: gainL = 0.0; break;
case 16: gainL = 0.0000152587890625; break;
case 15: gainL = 0.000030517578125; break;
case 14: gainL = 0.00006103515625; break;
case 13: gainL = 0.0001220703125; break;
case 12: gainL = 0.000244140625; break;
case 11: gainL = 0.00048828125; break;
case 10: gainL = 0.0009765625; break;
case 9: gainL = 0.001953125; break;
case 8: gainL = 0.00390625; break;
case 7: gainL = 0.0078125; break;
case 6: gainL = 0.015625; break;
case 5: gainL = 0.03125; break;
case 4: gainL = 0.0625; break;
case 3: gainL = 0.125; break;
case 2: gainL = 0.25; break;
case 1: gainL = 0.5; break;
case 0: break;
}
switch (bitshiftR)
{
case 17: gainR = 0.0; break;
case 16: gainR = 0.0000152587890625; break;
case 15: gainR = 0.000030517578125; break;
case 14: gainR = 0.00006103515625; break;
case 13: gainR = 0.0001220703125; break;
case 12: gainR = 0.000244140625; break;
case 11: gainR = 0.00048828125; break;
case 10: gainR = 0.0009765625; break;
case 9: gainR = 0.001953125; break;
case 8: gainR = 0.00390625; break;
case 7: gainR = 0.0078125; break;
case 6: gainR = 0.015625; break;
case 5: gainR = 0.03125; break;
case 4: gainR = 0.0625; break;
case 3: gainR = 0.125; break;
case 2: gainR = 0.25; break;
case 1: gainR = 0.5; break;
case 0: break;
}
gainA = gainB;
gainB = E*2.0; //smoothed master fader from Z2 filters
//BitShiftGain pre gain trim goes here
double subTrim = 0.0011 / 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;
double temp = (double)sampleFrames/inFramesToProcess;
double gain = (gainA*temp)+(gainB*(1.0-temp));
double mid = (midA*temp)+(midB*(1.0-temp));
double bass = (bassA*temp)+(bassB*(1.0-temp));
//begin Hull2 Treble
hullp--; if (hullp < 0) hullp += 60;
hullL[hullp] = hullL[hullp+60] = inputSampleL;
hullR[hullp] = hullR[hullp+60] = inputSampleR;
int x = hullp;
double bassL = 0.0;
double bassR = 0.0;
while (x < hullp+(limit/2)) {
bassL += hullL[x] * divisor;
bassR += hullR[x] * divisor;
x++;
}
bassL += bassL * 0.125;
bassR += bassR * 0.125;
while (x < hullp+limit) {
bassL -= hullL[x] * 0.125 * divisor;
bassR -= hullR[x] * 0.125 * divisor;
x++;
}
hullL[hullp+20] = hullL[hullp+80] = bassL;
hullR[hullp+20] = hullR[hullp+80] = bassR;
x = hullp+20;
bassL = bassR = 0.0;
while (x < hullp+20+(limit/2)) {
bassL += hullL[x] * divisor;
bassR += hullR[x] * divisor;
x++;
}
bassL += bassL * 0.125;
bassR += bassR * 0.125;
while (x < hullp+20+limit) {
bassL -= hullL[x] * 0.125 * divisor;
bassR -= hullR[x] * 0.125 * divisor;
x++;
}
hullL[hullp+40] = hullL[hullp+100] = bassL;
hullR[hullp+40] = hullR[hullp+100] = bassR;
x = hullp+40;
bassL = bassR = 0.0;
while (x < hullp+40+(limit/2)) {
bassL += hullL[x] * divisor;
bassR += hullR[x] * divisor;
x++;
}
bassL += bassL * 0.125;
bassR += bassR * 0.125;
while (x < hullp+40+limit) {
bassL -= hullL[x] * 0.125 * divisor;
bassR -= hullR[x] * 0.125 * divisor;
x++;
}
double trebleL = inputSampleL - bassL; inputSampleL = bassL;
double trebleR = inputSampleR - bassR; inputSampleR = bassR;
//end Hull2 treble
//begin Pear filter stages
//at this point 'bass' is actually still mid and bass
double slew = ((bassL - pearB[0]) + pearB[1])*freqMid*0.5;
pearB[0] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[0] + pearB[1]));
pearB[1] = slew; slew = ((bassR - pearB[2]) + pearB[3])*freqMid*0.5;
pearB[2] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[2] + pearB[3]));
pearB[3] = slew; slew = ((bassL - pearB[4]) + pearB[5])*freqMid*0.5;
pearB[4] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[4] + pearB[5]));
pearB[5] = slew; slew = ((bassR - pearB[6]) + pearB[7])*freqMid*0.5;
pearB[6] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[6] + pearB[7]));
pearB[7] = slew; slew = ((bassL - pearB[8]) + pearB[9])*freqMid*0.5;
pearB[8] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[8] + pearB[9]));
pearB[9] = slew; slew = ((bassR - pearB[10]) + pearB[11])*freqMid*0.5;
pearB[10] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[10] + pearB[11]));
pearB[11] = slew; slew = ((bassL - pearB[12]) + pearB[13])*freqMid*0.5;
pearB[12] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[12] + pearB[13]));
pearB[13] = slew; slew = ((bassR - pearB[14]) + pearB[15])*freqMid*0.5;
pearB[14] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[14] + pearB[15]));
pearB[15] = slew; slew = ((bassL - pearB[16]) + pearB[17])*freqMid*0.5;
pearB[16] = bassL = (freqMid * bassL) + ((1.0-freqMid) * (pearB[16] + pearB[17]));
pearB[17] = slew; slew = ((bassR - pearB[18]) + pearB[19])*freqMid*0.5;
pearB[18] = bassR = (freqMid * bassR) + ((1.0-freqMid) * (pearB[18] + pearB[19]));
pearB[19] = slew;
double midL = inputSampleL - bassL;
double midR = inputSampleR - bassR;
//we now have three bands out of hull and pear filters
double w = 0.0; //filter into bands, apply the sin/cos to each band
if (treble > 0.0) {
w = treble; if (w > 1.0) w = 1.0;
trebleL = (trebleL*(1.0-w)) + (sin(trebleL*M_PI_2)*treble);
trebleR = (trebleR*(1.0-w)) + (sin(trebleR*M_PI_2)*treble);
}
if (treble < 0.0) {
if (trebleL > 1.0) trebleL = 1.0; if (trebleL < -1.0) trebleL = -1.0;
if (trebleR > 1.0) trebleR = 1.0; if (trebleR < -1.0) trebleR = -1.0;
w = -treble; if (w > 1.0) w = 1.0;
if (trebleL > 0) trebleL = (trebleL*(1.0-w))+((1.0-cos(trebleL*w))*(1.0-w));
else trebleL = (trebleL*(1.0-w))+((-1.0+cos(-trebleL*w))*(1.0-w));
if (trebleR > 0) trebleR = (trebleR*(1.0-w))+((1.0-cos(trebleR*w))*(1.0-w));
else trebleR = (trebleR*(1.0-w))+((-1.0+cos(-trebleR*w))*(1.0-w));
} //cosine stages for EQ or expansion
if (midL > 1.0) midL = 1.0; if (midL < -1.0) midL = -1.0;
if (midR > 1.0) midR = 1.0; if (midR < -1.0) midR = -1.0;
if (mid > 0.0) {
w = mid; if (w > 1.0) w = 1.0;
midL = (midL*(1.0-w)) + (sin(midL*M_PI_2)*mid);
midR = (midR*(1.0-w)) + (sin(midR*M_PI_2)*mid);
}
if (mid < 0.0) {
w = -mid; if (w > 1.0) w = 1.0;
if (midL > 0) midL = (midL*(1.0-w))+((1.0-cos(midL*w))*(1.0-w));
else midL = (midL*(1.0-w))+((-1.0+cos(-midL*w))*(1.0-w));
if (midR > 0) midR = (midR*(1.0-w))+((1.0-cos(midR*w))*(1.0-w));
else midR = (midR*(1.0-w))+((-1.0+cos(-midR*w))*(1.0-w));
} //cosine stages for EQ or expansion
if (bassL > 1.0) bassL = 1.0; if (bassL < -1.0) bassL = -1.0;
if (bassR > 1.0) bassR = 1.0; if (bassR < -1.0) bassR = -1.0;
if (bass > 0.0) {
w = bass; if (w > 1.0) w = 1.0;
bassL = (bassL*(1.0-w)) + (sin(bassL*M_PI_2)*bass);
bassR = (bassR*(1.0-w)) + (sin(bassR*M_PI_2)*bass);
}
if (bass < 0.0) {
w = -bass; if (w > 1.0) w = 1.0;
if (bassL > 0) bassL = (bassL*(1.0-w))+((1.0-cos(bassL*w))*(1.0-w));
else bassL = (bassL*(1.0-w))+((-1.0+cos(-bassL*w))*(1.0-w));
if (bassR > 0) bassR = (bassR*(1.0-w))+((1.0-cos(bassR*w))*(1.0-w));
else bassR = (bassR*(1.0-w))+((-1.0+cos(-bassR*w))*(1.0-w));
} //cosine stages for EQ or expansion
inputSampleL = (bassL + midL + trebleL)*gainL*gain;
inputSampleR = (bassR + midR + trebleR)*gainR*gain;
//applies BitShiftPan pan section, and smoothed fader gain
//begin SubTight section
double subSampleL = inputSampleL * subTrim;
double subSampleR = inputSampleR * subTrim;
double scale = 0.5+fabs(subSampleL*0.5);
subSampleL = (subAL+(sin(subAL-subSampleL)*scale));
subAL = subSampleL*scale;
scale = 0.5+fabs(subSampleR*0.5);
subSampleR = (subAR+(sin(subAR-subSampleR)*scale));
subAR = subSampleR*scale;
scale = 0.5+fabs(subSampleL*0.5);
subSampleL = (subBL+(sin(subBL-subSampleL)*scale));
subBL = subSampleL*scale;
scale = 0.5+fabs(subSampleR*0.5);
subSampleR = (subBR+(sin(subBR-subSampleR)*scale));
subBR = subSampleR*scale;
scale = 0.5+fabs(subSampleL*0.5);
subSampleL = (subCL+(sin(subCL-subSampleL)*scale));
subCL = subSampleL*scale;
scale = 0.5+fabs(subSampleR*0.5);
subSampleR = (subCR+(sin(subCR-subSampleR)*scale));
subCR = subSampleR*scale;
if (subSampleL > 0.25) subSampleL = 0.25;
if (subSampleL < -0.25) subSampleL = -0.25;
if (subSampleR > 0.25) subSampleR = 0.25;
if (subSampleR < -0.25) subSampleR = -0.25;
inputSampleL += (subSampleL*16.0);
inputSampleR += (subSampleR*16.0);
//end SubTight section
//begin Console7 Channel processing
if (inputSampleL > 1.097) inputSampleL = 1.097;
if (inputSampleL < -1.097) inputSampleL = -1.097;
if (inputSampleR > 1.097) inputSampleR = 1.097;
if (inputSampleR < -1.097) inputSampleR = -1.097;
inputSampleL = ((sin(inputSampleL*fabs(inputSampleL))/((fabs(inputSampleL) == 0.0) ?1:fabs(inputSampleL)))*0.8)+(sin(inputSampleL)*0.2);
inputSampleR = ((sin(inputSampleR*fabs(inputSampleR))/((fabs(inputSampleR) == 0.0) ?1:fabs(inputSampleR)))*0.8)+(sin(inputSampleR)*0.2);
//this is a version of Spiral blended 80/20 with regular Density.
//It's blending between two different harmonics in the overtones of the algorithm
//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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