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airwindows/plugins/MacVST/Console8BussOut/source/Console8BussOutProc.cpp
Christopher Johnson f06250c082 Clarifying Licensing in boilerplate code generation
2022-11-21 09:20:21 -05:00

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/* ========================================
* Console8BussOut - Console8BussOut.h
* Copyright (c) 2016 airwindows, Airwindows uses the MIT license
* ======================================== */
#ifndef __Console8BussOut_H
#include "Console8BussOut.h"
#endif
void Console8BussOut::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
inTrimA = inTrimB; inTrimB = A*2.0;
//0.5 is unity gain, and we can attenuate to silence or boost slightly over 12dB
//into softclipping overdrive.
if (getSampleRate() > 49000.0) hsr = true; else hsr = false;
fix[fix_freq] = 24000.0 / getSampleRate();
fix[fix_reso] = 0.52110856;
double K = tan(M_PI * fix[fix_freq]); //lowpass
double norm = 1.0 / (1.0 + K / fix[fix_reso] + K * K);
fix[fix_a0] = K * K * norm;
fix[fix_a1] = 2.0 * fix[fix_a0];
fix[fix_a2] = fix[fix_a0];
fix[fix_b1] = 2.0 * (K * K - 1.0) * norm;
fix[fix_b2] = (1.0 - K / fix[fix_reso] + K * K) * norm;
//this is the fixed biquad distributed anti-aliasing filter
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
spacing = floor(overallscale); //should give us working basic scaling, usually 2 or 4
if (spacing < 1) spacing = 1; if (spacing > 16) spacing = 16;
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 position = (double)sampleFrames/inFramesToProcess;
double inTrim = (inTrimA*position)+(inTrimB*(1.0-position));
//input trim smoothed to cut out zipper noise
inputSampleL *= inTrim;
if (inputSampleL > 1.57079633) inputSampleL = 1.57079633; if (inputSampleL < -1.57079633) inputSampleL = -1.57079633;
inputSampleL = sin(inputSampleL);
//Console8 gain stage clips at exactly 1.0 post-sin()
inputSampleR *= inTrim;
if (inputSampleR > 1.57079633) inputSampleR = 1.57079633; if (inputSampleR < -1.57079633) inputSampleR = -1.57079633;
inputSampleR = sin(inputSampleR);
//Console8 gain stage clips at exactly 1.0 post-sin()
if (hsr){
double outSample = (inputSampleL * fix[fix_a0]) + fix[fix_sL1];
fix[fix_sL1] = (inputSampleL * fix[fix_a1]) - (outSample * fix[fix_b1]) + fix[fix_sL2];
fix[fix_sL2] = (inputSampleL * fix[fix_a2]) - (outSample * fix[fix_b2]);
inputSampleL = outSample;
outSample = (inputSampleR * fix[fix_a0]) + fix[fix_sR1];
fix[fix_sR1] = (inputSampleR * fix[fix_a1]) - (outSample * fix[fix_b1]) + fix[fix_sR2];
fix[fix_sR2] = (inputSampleR * fix[fix_a2]) - (outSample * fix[fix_b2]);
inputSampleR = outSample;
} //fixed biquad filtering ultrasonics
inputSampleL *= inTrim; inputSampleR *= inTrim;
//the final output fader, before ClipOnly2 and dithering
//begin ClipOnly2 stereo as a little, compressed chunk that can be dropped into code
if (inputSampleL > 4.0) inputSampleL = 4.0; if (inputSampleL < -4.0) inputSampleL = -4.0;
if (wasPosClipL == true) { //current will be over
if (inputSampleL<lastSampleL) lastSampleL=0.7058208+(inputSampleL*0.2609148);
else lastSampleL = 0.2491717+(lastSampleL*0.7390851);
} wasPosClipL = false;
if (inputSampleL>0.9549925859) {wasPosClipL=true;inputSampleL=0.7058208+(lastSampleL*0.2609148);}
if (wasNegClipL == true) { //current will be -over
if (inputSampleL > lastSampleL) lastSampleL=-0.7058208+(inputSampleL*0.2609148);
else lastSampleL=-0.2491717+(lastSampleL*0.7390851);
} wasNegClipL = false;
if (inputSampleL<-0.9549925859) {wasNegClipL=true;inputSampleL=-0.7058208+(lastSampleL*0.2609148);}
intermediateL[spacing] = inputSampleL;
inputSampleL = lastSampleL; //Latency is however many samples equals one 44.1k sample
for (int x = spacing; x > 0; x--) intermediateL[x-1] = intermediateL[x];
lastSampleL = intermediateL[0]; //run a little buffer to handle this
if (inputSampleR > 4.0) inputSampleR = 4.0; if (inputSampleR < -4.0) inputSampleR = -4.0;
if (wasPosClipR == true) { //current will be over
if (inputSampleR<lastSampleR) lastSampleR=0.7058208+(inputSampleR*0.2609148);
else lastSampleR = 0.2491717+(lastSampleR*0.7390851);
} wasPosClipR = false;
if (inputSampleR>0.9549925859) {wasPosClipR=true;inputSampleR=0.7058208+(lastSampleR*0.2609148);}
if (wasNegClipR == true) { //current will be -over
if (inputSampleR > lastSampleR) lastSampleR=-0.7058208+(inputSampleR*0.2609148);
else lastSampleR=-0.2491717+(lastSampleR*0.7390851);
} wasNegClipR = false;
if (inputSampleR<-0.9549925859) {wasNegClipR=true;inputSampleR=-0.7058208+(lastSampleR*0.2609148);}
intermediateR[spacing] = inputSampleR;
inputSampleR = lastSampleR; //Latency is however many samples equals one 44.1k sample
for (int x = spacing; x > 0; x--) intermediateR[x-1] = intermediateR[x];
lastSampleR = intermediateR[0]; //run a little buffer to handle this
//end ClipOnly2 stereo as a little, compressed chunk that can be dropped into code
//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 Console8BussOut::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
inTrimA = inTrimB; inTrimB = A*2.0;
//0.5 is unity gain, and we can attenuate to silence or boost slightly over 12dB
//into softclipping overdrive.
if (getSampleRate() > 49000.0) hsr = true; else hsr = false;
fix[fix_freq] = 24000.0 / getSampleRate();
fix[fix_reso] = 0.52110856;
double K = tan(M_PI * fix[fix_freq]); //lowpass
double norm = 1.0 / (1.0 + K / fix[fix_reso] + K * K);
fix[fix_a0] = K * K * norm;
fix[fix_a1] = 2.0 * fix[fix_a0];
fix[fix_a2] = fix[fix_a0];
fix[fix_b1] = 2.0 * (K * K - 1.0) * norm;
fix[fix_b2] = (1.0 - K / fix[fix_reso] + K * K) * norm;
//this is the fixed biquad distributed anti-aliasing filter
double overallscale = 1.0;
overallscale /= 44100.0;
overallscale *= getSampleRate();
spacing = floor(overallscale); //should give us working basic scaling, usually 2 or 4
if (spacing < 1) spacing = 1; if (spacing > 16) spacing = 16;
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 position = (double)sampleFrames/inFramesToProcess;
double inTrim = (inTrimA*position)+(inTrimB*(1.0-position));
//input trim smoothed to cut out zipper noise
inputSampleL *= inTrim;
if (inputSampleL > 1.57079633) inputSampleL = 1.57079633; if (inputSampleL < -1.57079633) inputSampleL = -1.57079633;
inputSampleL = sin(inputSampleL);
//Console8 gain stage clips at exactly 1.0 post-sin()
inputSampleR *= inTrim;
if (inputSampleR > 1.57079633) inputSampleR = 1.57079633; if (inputSampleR < -1.57079633) inputSampleR = -1.57079633;
inputSampleR = sin(inputSampleR);
//Console8 gain stage clips at exactly 1.0 post-sin()
if (hsr){
double outSample = (inputSampleL * fix[fix_a0]) + fix[fix_sL1];
fix[fix_sL1] = (inputSampleL * fix[fix_a1]) - (outSample * fix[fix_b1]) + fix[fix_sL2];
fix[fix_sL2] = (inputSampleL * fix[fix_a2]) - (outSample * fix[fix_b2]);
inputSampleL = outSample;
outSample = (inputSampleR * fix[fix_a0]) + fix[fix_sR1];
fix[fix_sR1] = (inputSampleR * fix[fix_a1]) - (outSample * fix[fix_b1]) + fix[fix_sR2];
fix[fix_sR2] = (inputSampleR * fix[fix_a2]) - (outSample * fix[fix_b2]);
inputSampleR = outSample;
} //fixed biquad filtering ultrasonics
inputSampleL *= inTrim; inputSampleR *= inTrim;
//the final output fader, before ClipOnly2 and dithering
//begin ClipOnly2 stereo as a little, compressed chunk that can be dropped into code
if (inputSampleL > 4.0) inputSampleL = 4.0; if (inputSampleL < -4.0) inputSampleL = -4.0;
if (wasPosClipL == true) { //current will be over
if (inputSampleL<lastSampleL) lastSampleL=0.7058208+(inputSampleL*0.2609148);
else lastSampleL = 0.2491717+(lastSampleL*0.7390851);
} wasPosClipL = false;
if (inputSampleL>0.9549925859) {wasPosClipL=true;inputSampleL=0.7058208+(lastSampleL*0.2609148);}
if (wasNegClipL == true) { //current will be -over
if (inputSampleL > lastSampleL) lastSampleL=-0.7058208+(inputSampleL*0.2609148);
else lastSampleL=-0.2491717+(lastSampleL*0.7390851);
} wasNegClipL = false;
if (inputSampleL<-0.9549925859) {wasNegClipL=true;inputSampleL=-0.7058208+(lastSampleL*0.2609148);}
intermediateL[spacing] = inputSampleL;
inputSampleL = lastSampleL; //Latency is however many samples equals one 44.1k sample
for (int x = spacing; x > 0; x--) intermediateL[x-1] = intermediateL[x];
lastSampleL = intermediateL[0]; //run a little buffer to handle this
if (inputSampleR > 4.0) inputSampleR = 4.0; if (inputSampleR < -4.0) inputSampleR = -4.0;
if (wasPosClipR == true) { //current will be over
if (inputSampleR<lastSampleR) lastSampleR=0.7058208+(inputSampleR*0.2609148);
else lastSampleR = 0.2491717+(lastSampleR*0.7390851);
} wasPosClipR = false;
if (inputSampleR>0.9549925859) {wasPosClipR=true;inputSampleR=0.7058208+(lastSampleR*0.2609148);}
if (wasNegClipR == true) { //current will be -over
if (inputSampleR > lastSampleR) lastSampleR=-0.7058208+(inputSampleR*0.2609148);
else lastSampleR=-0.2491717+(lastSampleR*0.7390851);
} wasNegClipR = false;
if (inputSampleR<-0.9549925859) {wasNegClipR=true;inputSampleR=-0.7058208+(lastSampleR*0.2609148);}
intermediateR[spacing] = inputSampleR;
inputSampleR = lastSampleR; //Latency is however many samples equals one 44.1k sample
for (int x = spacing; x > 0; x--) intermediateR[x-1] = intermediateR[x];
lastSampleR = intermediateR[0]; //run a little buffer to handle this
//end ClipOnly2 stereo as a little, compressed chunk that can be dropped into code
//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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