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airwindows/plugins/MacSignedVST/Mastering2/source/Mastering2Proc.cpp
Christopher Johnson 912d209fc7 Bugfixes for Sinew Users
2025-03-17 19:22:44 -04:00

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/* ========================================
* Mastering2 - Mastering2.h
* Copyright (c) airwindows, Airwindows uses the MIT license
* ======================================== */
#ifndef __Mastering2_H
#include "Mastering2.h"
#endif
void Mastering2::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();
double iirSide = pow(A,3) * (0.1 / overallscale);
double threshSinew = (0.25+((1.0-B)*0.333))/overallscale;
double depthSinew = 1.0-pow(1.0-B,2.0);
double trebleZoom = C-0.5;
long double trebleGain = (trebleZoom*fabs(trebleZoom))+1.0;
if (trebleGain > 1.0) trebleGain = pow(trebleGain,3.0+sqrt(overallscale));
//this boost is necessary to adapt to higher sample rates
double midZoom = D-0.5;
long double midGain = (midZoom*fabs(midZoom))+1.0;
double kalMid = 0.35-(D*0.25); //crossover frequency between mid/bass
double kalSub = 0.45+(D*0.25); //crossover frequency between bass/sub
double bassZoom = (E*0.5)-0.25;
long double bassGain = (-bassZoom*fabs(bassZoom))+1.0; //control inverted
long double subGain = ((E*0.25)-0.125)+1.0;
if (subGain < 1.0) subGain = 1.0; //very small sub shift, only pos.
long double driveIn = (F-0.5)+1.0;
long double driveOut = (-(F-0.5)*fabs(F-0.5))+1.0;
int spacing = floor(overallscale); //should give us working basic scaling, usually 2 or 4
if (spacing < 1) spacing = 1; if (spacing > 16) spacing = 16;
int dither = (int) (G*5.999);
int depth = (int)(17.0*overallscale);
if (depth < 3) depth = 3; if (depth > 98) depth = 98; //for Dark
while (--sampleFrames >= 0)
{
long double inputSampleL = *in1;
long 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 *= driveIn;
inputSampleR *= driveIn;
long double mid = inputSampleL + inputSampleR;
long double side = inputSampleL - inputSampleR;
//assign mid and side.Between these sections, you can do mid/side processing
long double temp = side;
if (fpFlip) {
iirA = (iirA * (1.0 - iirSide)) + (temp * iirSide); temp = iirA;
} else {
iirB = (iirB * (1.0 - iirSide)) + (temp * iirSide); temp = iirB;
}
iirC = (iirC * (1.0 - iirSide)) + (temp * iirSide); temp = iirC;
side -= sin(temp);
fpFlip = !fpFlip;
inputSampleL = (mid+side)/2.0;
inputSampleR = (mid-side)/2.0;
//unassign mid and side
long double drySampleL = inputSampleL;
long double drySampleR = inputSampleR;
//begin Air3L
air[pvSL4] = air[pvAL4] - air[pvAL3]; air[pvSL3] = air[pvAL3] - air[pvAL2];
air[pvSL2] = air[pvAL2] - air[pvAL1]; air[pvSL1] = air[pvAL1] - inputSampleL;
air[accSL3] = air[pvSL4] - air[pvSL3]; air[accSL2] = air[pvSL3] - air[pvSL2];
air[accSL1] = air[pvSL2] - air[pvSL1];
air[acc2SL2] = air[accSL3] - air[accSL2]; air[acc2SL1] = air[accSL2] - air[accSL1];
air[outAL] = -(air[pvAL1] + air[pvSL3] + air[acc2SL2] - ((air[acc2SL2] + air[acc2SL1])*0.5));
air[gainAL] *= 0.5; air[gainAL] += fabs(drySampleL-air[outAL])*0.5;
if (air[gainAL] > 0.3*sqrt(overallscale)) air[gainAL] = 0.3*sqrt(overallscale);
air[pvAL4] = air[pvAL3]; air[pvAL3] = air[pvAL2];
air[pvAL2] = air[pvAL1]; air[pvAL1] = (air[gainAL] * air[outAL]) + drySampleL;
long double midL = drySampleL - ((air[outAL]*0.5)+(drySampleL*(0.457-(0.017*overallscale))));
temp = (midL + air[gndavgL])*0.5; air[gndavgL] = midL; midL = temp;
long double trebleL = drySampleL-midL;
//end Air3L
//begin Air3R
air[pvSR4] = air[pvAR4] - air[pvAR3]; air[pvSR3] = air[pvAR3] - air[pvAR2];
air[pvSR2] = air[pvAR2] - air[pvAR1]; air[pvSR1] = air[pvAR1] - inputSampleR;
air[accSR3] = air[pvSR4] - air[pvSR3]; air[accSR2] = air[pvSR3] - air[pvSR2];
air[accSR1] = air[pvSR2] - air[pvSR1];
air[acc2SR2] = air[accSR3] - air[accSR2]; air[acc2SR1] = air[accSR2] - air[accSR1];
air[outAR] = -(air[pvAR1] + air[pvSR3] + air[acc2SR2] - ((air[acc2SR2] + air[acc2SR1])*0.5));
air[gainAR] *= 0.5; air[gainAR] += fabs(drySampleR-air[outAR])*0.5;
if (air[gainAR] > 0.3*sqrt(overallscale)) air[gainAR] = 0.3*sqrt(overallscale);
air[pvAR4] = air[pvAR3]; air[pvAR3] = air[pvAR2];
air[pvAR2] = air[pvAR1]; air[pvAR1] = (air[gainAR] * air[outAR]) + drySampleR;
long double midR = drySampleR - ((air[outAR]*0.5)+(drySampleR*(0.457-(0.017*overallscale))));
temp = (midR + air[gndavgR])*0.5; air[gndavgR] = midR; midR = temp;
long double trebleR = drySampleR-midR;
//end Air3R
//begin KalmanML
temp = midL;
kalM[prevSlewL3] += kalM[prevSampL3] - kalM[prevSampL2]; kalM[prevSlewL3] *= 0.5;
kalM[prevSlewL2] += kalM[prevSampL2] - kalM[prevSampL1]; kalM[prevSlewL2] *= 0.5;
kalM[prevSlewL1] += kalM[prevSampL1] - midL; kalM[prevSlewL1] *= 0.5;
//make slews from each set of samples used
kalM[accSlewL2] += kalM[prevSlewL3] - kalM[prevSlewL2]; kalM[accSlewL2] *= 0.5;
kalM[accSlewL1] += kalM[prevSlewL2] - kalM[prevSlewL1]; kalM[accSlewL1] *= 0.5;
//differences between slews: rate of change of rate of change
kalM[accSlewL3] += (kalM[accSlewL2] - kalM[accSlewL1]); kalM[accSlewL3] *= 0.5;
//entering the abyss, what even is this
kalM[kalOutL] += kalM[prevSampL1] + kalM[prevSlewL2] + kalM[accSlewL3]; kalM[kalOutL] *= 0.5;
//resynthesizing predicted result (all iir smoothed)
kalM[kalGainL] += fabs(temp-kalM[kalOutL])*kalMid*8.0; kalM[kalGainL] *= 0.5;
//madness takes its toll. Kalman Gain: how much dry to retain
if (kalM[kalGainL] > kalMid*0.5) kalM[kalGainL] = kalMid*0.5;
//attempts to avoid explosions
kalM[kalOutL] += (temp*(1.0-(0.68+(kalMid*0.157))));
//this is for tuning a really complete cancellation up around Nyquist
kalM[prevSampL3] = kalM[prevSampL2]; kalM[prevSampL2] = kalM[prevSampL1];
kalM[prevSampL1] = (kalM[kalGainL] * kalM[kalOutL]) + ((1.0-kalM[kalGainL])*temp);
//feed the chain of previous samples
long double bassL = (kalM[kalOutL]+kalM[kalAvgL])*0.5;
kalM[kalAvgL] = kalM[kalOutL];
midL -= bassL;
//end KalmanML
//begin KalmanMR
temp = midR;
kalM[prevSlewR3] += kalM[prevSampR3] - kalM[prevSampR2]; kalM[prevSlewR3] *= 0.5;
kalM[prevSlewR2] += kalM[prevSampR2] - kalM[prevSampR1]; kalM[prevSlewR2] *= 0.5;
kalM[prevSlewR1] += kalM[prevSampR1] - midR; kalM[prevSlewR1] *= 0.5;
//make slews from each set of samples used
kalM[accSlewR2] += kalM[prevSlewR3] - kalM[prevSlewR2]; kalM[accSlewR2] *= 0.5;
kalM[accSlewR1] += kalM[prevSlewR2] - kalM[prevSlewR1]; kalM[accSlewR1] *= 0.5;
//differences between slews: rate of change of rate of change
kalM[accSlewR3] += (kalM[accSlewR2] - kalM[accSlewR1]); kalM[accSlewR3] *= 0.5;
//entering the abyss, what even is this
kalM[kalOutR] += kalM[prevSampR1] + kalM[prevSlewR2] + kalM[accSlewR3]; kalM[kalOutR] *= 0.5;
//resynthesizing predicted result (all iir smoothed)
kalM[kalGainR] += fabs(temp-kalM[kalOutR])*kalMid*8.0; kalM[kalGainR] *= 0.5;
//madness takes its toll. Kalman Gain: how much dry to retain
if (kalM[kalGainR] > kalMid*0.5) kalM[kalGainR] = kalMid*0.5;
//attempts to avoid explosions
kalM[kalOutR] += (temp*(1.0-(0.68+(kalMid*0.157))));
//this is for tuning a really complete cancellation up around Nyquist
kalM[prevSampR3] = kalM[prevSampR2]; kalM[prevSampR2] = kalM[prevSampR1];
kalM[prevSampR1] = (kalM[kalGainR] * kalM[kalOutR]) + ((1.0-kalM[kalGainR])*temp);
//feed the chain of previous samples
long double bassR = (kalM[kalOutR]+kalM[kalAvgR])*0.5;
kalM[kalAvgR] = kalM[kalOutR];
midR -= bassR;
//end KalmanMR
//begin KalmanSL
temp = bassL;
kalS[prevSlewL3] += kalS[prevSampL3] - kalS[prevSampL2]; kalS[prevSlewL3] *= 0.5;
kalS[prevSlewL2] += kalS[prevSampL2] - kalS[prevSampL1]; kalS[prevSlewL2] *= 0.5;
kalS[prevSlewL1] += kalS[prevSampL1] - bassL; kalS[prevSlewL1] *= 0.5;
//make slews from each set of samples used
kalS[accSlewL2] += kalS[prevSlewL3] - kalS[prevSlewL2]; kalS[accSlewL2] *= 0.5;
kalS[accSlewL1] += kalS[prevSlewL2] - kalS[prevSlewL1]; kalS[accSlewL1] *= 0.5;
//differences between slews: rate of change of rate of change
kalS[accSlewL3] += (kalS[accSlewL2] - kalS[accSlewL1]); kalS[accSlewL3] *= 0.5;
//entering the abyss, what even is this
kalS[kalOutL] += kalS[prevSampL1] + kalS[prevSlewL2] + kalS[accSlewL3]; kalS[kalOutL] *= 0.5;
//resynthesizing predicted result (all iir smoothed)
kalS[kalGainL] += fabs(temp-kalS[kalOutL])*kalSub*8.0; kalS[kalGainL] *= 0.5;
//madness takes its toll. Kalman Gain: how much dry to retain
if (kalS[kalGainL] > kalSub*0.5) kalS[kalGainL] = kalSub*0.5;
//attempts to avoid explosions
kalS[kalOutL] += (temp*(1.0-(0.68+(kalSub*0.157))));
//this is for tuning a really complete cancellation up around Nyquist
kalS[prevSampL3] = kalS[prevSampL2]; kalS[prevSampL2] = kalS[prevSampL1];
kalS[prevSampL1] = (kalS[kalGainL] * kalS[kalOutL]) + ((1.0-kalS[kalGainL])*temp);
//feed the chain of previous samples
long double subL = (kalS[kalOutL]+kalS[kalAvgL])*0.5;
kalS[kalAvgL] = kalS[kalOutL];
bassL -= subL;
//end KalmanSL
//begin KalmanSR
temp = bassR;
kalS[prevSlewR3] += kalS[prevSampR3] - kalS[prevSampR2]; kalS[prevSlewR3] *= 0.5;
kalS[prevSlewR2] += kalS[prevSampR2] - kalS[prevSampR1]; kalS[prevSlewR2] *= 0.5;
kalS[prevSlewR1] += kalS[prevSampR1] - bassR; kalS[prevSlewR1] *= 0.5;
//make slews from each set of samples used
kalS[accSlewR2] += kalS[prevSlewR3] - kalS[prevSlewR2]; kalS[accSlewR2] *= 0.5;
kalS[accSlewR1] += kalS[prevSlewR2] - kalS[prevSlewR1]; kalS[accSlewR1] *= 0.5;
//differences between slews: rate of change of rate of change
kalS[accSlewR3] += (kalS[accSlewR2] - kalS[accSlewR1]); kalS[accSlewR3] *= 0.5;
//entering the abyss, what even is this
kalS[kalOutR] += kalS[prevSampR1] + kalS[prevSlewR2] + kalS[accSlewR3]; kalS[kalOutR] *= 0.5;
//resynthesizing predicted result (all iir smoothed)
kalS[kalGainR] += fabs(temp-kalS[kalOutR])*kalSub*8.0; kalS[kalGainR] *= 0.5;
//madness takes its toll. Kalman Gain: how much dry to retain
if (kalS[kalGainR] > kalSub*0.5) kalS[kalGainR] = kalSub*0.5;
//attempts to avoid explosions
kalS[kalOutR] += (temp*(1.0-(0.68+(kalSub*0.157))));
//this is for tuning a really complete cancellation up around Nyquist
kalS[prevSampR3] = kalS[prevSampR2]; kalS[prevSampR2] = kalS[prevSampR1];
kalS[prevSampR1] = (kalS[kalGainR] * kalS[kalOutR]) + ((1.0-kalS[kalGainR])*temp);
//feed the chain of previous samples
long double subR = (kalS[kalOutR]+kalS[kalAvgR])*0.5;
kalS[kalAvgR] = kalS[kalOutR];
bassR -= subR;
//end KalmanSR
inputSampleL = (subL*subGain);
inputSampleR = (subR*subGain);
if (bassZoom > 0.0) {
double closer = bassL * 1.57079633;
if (closer > 1.57079633) closer = 1.57079633;
if (closer < -1.57079633) closer = -1.57079633;
bassL = (bassL*(1.0-bassZoom))+(sin(closer)*bassZoom);
closer = bassR * 1.57079633;
if (closer > 1.57079633) closer = 1.57079633;
if (closer < -1.57079633) closer = -1.57079633;
bassR = (bassR*(1.0-bassZoom))+(sin(closer)*bassZoom);
} //zooming in will make the body of the sound louder: it's just Density
if (bassZoom < 0.0) {
double farther = fabs(bassL) * 1.57079633;
if (farther > 1.57079633) farther = 1.0;
else farther = 1.0-cos(farther);
if (bassL > 0.0) bassL = (bassL*(1.0+bassZoom))-(farther*bassZoom*1.57079633);
if (bassL < 0.0) bassL = (bassL*(1.0+bassZoom))+(farther*bassZoom*1.57079633);
farther = fabs(bassR) * 1.57079633;
if (farther > 1.57079633) farther = 1.0;
else farther = 1.0-cos(farther);
if (bassR > 0.0) bassR = (bassR*(1.0+bassZoom))-(farther*bassZoom*1.57079633);
if (bassR < 0.0) bassR = (bassR*(1.0+bassZoom))+(farther*bassZoom*1.57079633);
} //zooming out boosts the hottest peaks but cuts back softer stuff
inputSampleL += (bassL*bassGain);
inputSampleR += (bassR*bassGain);
if (midZoom > 0.0) {
double closer = midL * 1.57079633;
if (closer > 1.57079633) closer = 1.57079633;
if (closer < -1.57079633) closer = -1.57079633;
midL = (midL*(1.0-midZoom))+(sin(closer)*midZoom);
closer = midR * 1.57079633;
if (closer > 1.57079633) closer = 1.57079633;
if (closer < -1.57079633) closer = -1.57079633;
midR = (midR*(1.0-midZoom))+(sin(closer)*midZoom);
} //zooming in will make the body of the sound louder: it's just Density
if (midZoom < 0.0) {
double farther = fabs(midL) * 1.57079633;
if (farther > 1.57079633) farther = 1.0;
else farther = 1.0-cos(farther);
if (midL > 0.0) midL = (midL*(1.0+midZoom))-(farther*midZoom*1.57079633);
if (midL < 0.0) midL = (midL*(1.0+midZoom))+(farther*midZoom*1.57079633);
farther = fabs(midR) * 1.57079633;
if (farther > 1.57079633) farther = 1.0;
else farther = 1.0-cos(farther);
if (midR > 0.0) midR = (midR*(1.0+midZoom))-(farther*midZoom*1.57079633);
if (midR < 0.0) midR = (midR*(1.0+midZoom))+(farther*midZoom*1.57079633);
} //zooming out boosts the hottest peaks but cuts back softer stuff
inputSampleL += (midL*midGain);
inputSampleR += (midR*midGain);
if (trebleZoom > 0.0) {
double closer = trebleL * 1.57079633;
if (closer > 1.57079633) closer = 1.57079633;
if (closer < -1.57079633) closer = -1.57079633;
trebleL = (trebleL*(1.0-trebleZoom))+(sin(closer)*trebleZoom);
closer = trebleR * 1.57079633;
if (closer > 1.57079633) closer = 1.57079633;
if (closer < -1.57079633) closer = -1.57079633;
trebleR = (trebleR*(1.0-trebleZoom))+(sin(closer)*trebleZoom);
} //zooming in will make the body of the sound louder: it's just Density
if (trebleZoom < 0.0) {
double farther = fabs(trebleL) * 1.57079633;
if (farther > 1.57079633) farther = 1.0;
else farther = 1.0-cos(farther);
if (trebleL > 0.0) trebleL = (trebleL*(1.0+trebleZoom))-(farther*trebleZoom*1.57079633);
if (trebleL < 0.0) trebleL = (trebleL*(1.0+trebleZoom))+(farther*trebleZoom*1.57079633);
farther = fabs(trebleR) * 1.57079633;
if (farther > 1.57079633) farther = 1.0;
else farther = 1.0-cos(farther);
if (trebleR > 0.0) trebleR = (trebleR*(1.0+trebleZoom))-(farther*trebleZoom*1.57079633);
if (trebleR < 0.0) trebleR = (trebleR*(1.0+trebleZoom))+(farther*trebleZoom*1.57079633);
} //zooming out boosts the hottest peaks but cuts back softer stuff
inputSampleL += (trebleL*trebleGain);
inputSampleR += (trebleR*trebleGain);
inputSampleL *= driveOut;
inputSampleR *= driveOut;
//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
temp = inputSampleL;
long double sinew = threshSinew * cos(lastSinewL*lastSinewL);
if (inputSampleL - lastSinewL > sinew) temp = lastSinewL + sinew;
if (-(inputSampleL - lastSinewL) > sinew) temp = lastSinewL - sinew;
lastSinewL = temp;
inputSampleL = (inputSampleL * (1.0-depthSinew))+(lastSinewL*depthSinew);
temp = inputSampleR;
sinew = threshSinew * cos(lastSinewR*lastSinewR);
if (inputSampleR - lastSinewR > sinew) temp = lastSinewR + sinew;
if (-(inputSampleR - lastSinewR) > sinew) temp = lastSinewR - sinew;
lastSinewR = temp;
inputSampleR = (inputSampleR * (1.0-depthSinew))+(lastSinewR*depthSinew);
//run Sinew to stop excess slews, but run a dry/wet to allow a range of brights
switch (dither) {
case 1:
//begin Dark
inputSampleL *= 8388608.0;
inputSampleR *= 8388608.0; //we will apply the 24 bit Dark
//We are doing it first Left, then Right, because the loops may run faster if
//they aren't too jammed full of variables. This means re-running code.
//begin left
quantA = floor(inputSampleL);
quantB = floor(inputSampleL+1.0);
//to do this style of dither, we quantize in either direction and then
//do a reconstruction of what the result will be for each choice.
//We then evaluate which one we like, and keep a history of what we previously had
expectedSlew = 0;
for(int x = 0; x < depth; x++) {
expectedSlew += (darkSampleL[x+1] - darkSampleL[x]);
}
expectedSlew /= depth; //we have an average of all recent slews
//we are doing that to voice the thing down into the upper mids a bit
//it mustn't just soften the brightest treble, it must smooth high mids too
testA = fabs((darkSampleL[0] - quantA) - expectedSlew);
testB = fabs((darkSampleL[0] - quantB) - expectedSlew);
if (testA < testB) inputSampleL = quantA;
else inputSampleL = quantB;
//select whichever one departs LEAST from the vector of averaged
//reconstructed previous final samples. This will force a kind of dithering
//as it'll make the output end up as smooth as possible
for(int x = depth; x >=0; x--) {
darkSampleL[x+1] = darkSampleL[x];
}
darkSampleL[0] = inputSampleL;
//end Dark left
//begin right
quantA = floor(inputSampleR);
quantB = floor(inputSampleR+1.0);
//to do this style of dither, we quantize in either direction and then
//do a reconstruction of what the result will be for each choice.
//We then evaluate which one we like, and keep a history of what we previously had
expectedSlew = 0;
for(int x = 0; x < depth; x++) {
expectedSlew += (darkSampleR[x+1] - darkSampleR[x]);
}
expectedSlew /= depth; //we have an average of all recent slews
//we are doing that to voice the thing down into the upper mids a bit
//it mustn't just soften the brightest treble, it must smooth high mids too
testA = fabs((darkSampleR[0] - quantA) - expectedSlew);
testB = fabs((darkSampleR[0] - quantB) - expectedSlew);
if (testA < testB) inputSampleR = quantA;
else inputSampleR = quantB;
//select whichever one departs LEAST from the vector of averaged
//reconstructed previous final samples. This will force a kind of dithering
//as it'll make the output end up as smooth as possible
for(int x = depth; x >=0; x--) {
darkSampleR[x+1] = darkSampleR[x];
}
darkSampleR[0] = inputSampleR;
//end Dark right
inputSampleL /= 8388608.0;
inputSampleR /= 8388608.0;
break; //Dark (Monitoring2)
case 2:
//begin Dark for Ten Nines
inputSampleL *= 8388608.0;
inputSampleR *= 8388608.0; //we will apply the 24 bit Dark
//We are doing it first Left, then Right, because the loops may run faster if
//they aren't too jammed full of variables. This means re-running code.
//begin L
correction = 0;
if (flip) {
NSOddL = (NSOddL * 0.9999999999) + prevShapeL;
NSEvenL = (NSEvenL * 0.9999999999) - prevShapeL;
correction = NSOddL;
} else {
NSOddL = (NSOddL * 0.9999999999) - prevShapeL;
NSEvenL = (NSEvenL * 0.9999999999) + prevShapeL;
correction = NSEvenL;
}
shapedSampleL = inputSampleL+correction;
//end Ten Nines L
//begin Dark L
quantA = floor(shapedSampleL);
quantB = floor(shapedSampleL+1.0);
//to do this style of dither, we quantize in either direction and then
//do a reconstruction of what the result will be for each choice.
//We then evaluate which one we like, and keep a history of what we previously had
expectedSlew = 0;
for(int x = 0; x < depth; x++) {
expectedSlew += (darkSampleL[x+1] - darkSampleL[x]);
}
expectedSlew /= depth; //we have an average of all recent slews
//we are doing that to voice the thing down into the upper mids a bit
//it mustn't just soften the brightest treble, it must smooth high mids too
testA = fabs((darkSampleL[0] - quantA) - expectedSlew);
testB = fabs((darkSampleL[0] - quantB) - expectedSlew);
if (testA < testB) inputSampleL = quantA;
else inputSampleL = quantB;
//select whichever one departs LEAST from the vector of averaged
//reconstructed previous final samples. This will force a kind of dithering
//as it'll make the output end up as smooth as possible
for(int x = depth; x >=0; x--) {
darkSampleL[x+1] = darkSampleL[x];
}
darkSampleL[0] = inputSampleL;
//end Dark L
prevShapeL = (floor(shapedSampleL) - inputSampleL)*0.9999999999;
//end Ten Nines L
//begin R
correction = 0;
if (flip) {
NSOddR = (NSOddR * 0.9999999999) + prevShapeR;
NSEvenR = (NSEvenR * 0.9999999999) - prevShapeR;
correction = NSOddR;
} else {
NSOddR = (NSOddR * 0.9999999999) - prevShapeR;
NSEvenR = (NSEvenR * 0.9999999999) + prevShapeR;
correction = NSEvenR;
}
shapedSampleR = inputSampleR+correction;
//end Ten Nines R
//begin Dark R
quantA = floor(shapedSampleR);
quantB = floor(shapedSampleR+1.0);
//to do this style of dither, we quantize in either direction and then
//do a reconstruction of what the result will be for each choice.
//We then evaluate which one we like, and keep a history of what we previously had
expectedSlew = 0;
for(int x = 0; x < depth; x++) {
expectedSlew += (darkSampleR[x+1] - darkSampleR[x]);
}
expectedSlew /= depth; //we have an average of all recent slews
//we are doing that to voice the thing down into the upper mids a bit
//it mustn't just soften the brightest treble, it must smooth high mids too
testA = fabs((darkSampleR[0] - quantA) - expectedSlew);
testB = fabs((darkSampleR[0] - quantB) - expectedSlew);
if (testA < testB) inputSampleR = quantA;
else inputSampleR = quantB;
//select whichever one departs LEAST from the vector of averaged
//reconstructed previous final samples. This will force a kind of dithering
//as it'll make the output end up as smooth as possible
for(int x = depth; x >=0; x--) {
darkSampleR[x+1] = darkSampleR[x];
}
darkSampleR[0] = inputSampleR;
//end Dark R
prevShapeR = (floor(shapedSampleR) - inputSampleR)*0.9999999999;
//end Ten Nines
flip = !flip;
inputSampleL /= 8388608.0;
inputSampleR /= 8388608.0;
break; //Ten Nines (which goes into Dark in Monitoring3)
case 3:
inputSampleL *= 8388608.0;
inputSampleR *= 8388608.0;
ditherL = -1.0;
ditherL += (double(fpdL)/UINT32_MAX);
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
ditherL += (double(fpdL)/UINT32_MAX);
//TPDF: two 0-1 random noises
ditherR = -1.0;
ditherR += (double(fpdR)/UINT32_MAX);
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
ditherR += (double(fpdR)/UINT32_MAX);
//TPDF: two 0-1 random noises
if (fabs(ditherL-ditherR) < 0.5) {
ditherL = -1.0;
ditherL += (double(fpdL)/UINT32_MAX);
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
ditherL += (double(fpdL)/UINT32_MAX);
}
if (fabs(ditherL-ditherR) < 0.5) {
ditherR = -1.0;
ditherR += (double(fpdR)/UINT32_MAX);
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
ditherR += (double(fpdR)/UINT32_MAX);
}
if (fabs(ditherL-ditherR) < 0.5) {
ditherL = -1.0;
ditherL += (double(fpdL)/UINT32_MAX);
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
ditherL += (double(fpdL)/UINT32_MAX);
}
inputSampleL = floor(inputSampleL+ditherL);
inputSampleR = floor(inputSampleR+ditherR);
inputSampleL /= 8388608.0;
inputSampleR /= 8388608.0;
break; //TPDFWide (a good neutral with the width enhancement)
case 4:
inputSampleL *= 8388608.0;
inputSampleR *= 8388608.0;
//Paul Frindle: It's true that the dither itself can sound different
//if it's given a different freq response and you get to hear it.
//The one we use most is triangular single pole high pass dither.
//It's not freq bent enough to sound odd, but is slightly less audible than
//flat dither. It can also be easily made by taking one sample of dither
//away from the previous one - this gives you the triangular PDF and the
//filtering in one go :-)
currentDither = (double(fpdL)/UINT32_MAX);
ditherL = currentDither;
ditherL -= previousDitherL;
previousDitherL = currentDither;
//TPDF: two 0-1 random noises
currentDither = (double(fpdR)/UINT32_MAX);
ditherR = currentDither;
ditherR -= previousDitherR;
previousDitherR = currentDither;
//TPDF: two 0-1 random noises
if (fabs(ditherL-ditherR) < 0.5) {
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
currentDither = (double(fpdL)/UINT32_MAX);
ditherL = currentDither;
ditherL -= previousDitherL;
previousDitherL = currentDither;
}
if (fabs(ditherL-ditherR) < 0.5) {
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
currentDither = (double(fpdR)/UINT32_MAX);
ditherR = currentDither;
ditherR -= previousDitherR;
previousDitherR = currentDither;
}
if (fabs(ditherL-ditherR) < 0.5) {
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
currentDither = (double(fpdL)/UINT32_MAX);
ditherL = currentDither;
ditherL -= previousDitherL;
previousDitherL = currentDither;
}
inputSampleL = floor(inputSampleL+ditherL);
inputSampleR = floor(inputSampleR+ditherR);
inputSampleL /= 8388608.0;
inputSampleR /= 8388608.0;
break; //PaulWide (brighter neutral that's still TPDF and wide)
case 5:
inputSampleL *= 8388608.0;
inputSampleR *= 8388608.0;
cutbinsL = false;
cutbinsR = false;
drySampleL = inputSampleL;//re-using in NJAD
inputSampleL -= noiseShapingL;
//NJAD L
benfordize = floor(inputSampleL);
while (benfordize >= 1.0) benfordize /= 10;
while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10;
hotbinA = floor(benfordize);
//hotbin becomes the Benford bin value for this number floored
totalA = 0.0;
if ((hotbinA > 0) && (hotbinA < 10))
{
bynL[hotbinA] += 1; if (bynL[hotbinA] > 982) cutbinsL = true;
totalA += (301-bynL[1]); totalA += (176-bynL[2]); totalA += (125-bynL[3]);
totalA += (97-bynL[4]); totalA += (79-bynL[5]); totalA += (67-bynL[6]);
totalA += (58-bynL[7]); totalA += (51-bynL[8]); totalA += (46-bynL[9]); bynL[hotbinA] -= 1;
} else hotbinA = 10;
//produce total number- smaller is closer to Benford real
benfordize = ceil(inputSampleL);
while (benfordize >= 1.0) benfordize /= 10;
while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10;
hotbinB = floor(benfordize);
//hotbin becomes the Benford bin value for this number ceiled
totalB = 0.0;
if ((hotbinB > 0) && (hotbinB < 10))
{
bynL[hotbinB] += 1; if (bynL[hotbinB] > 982) cutbinsL = true;
totalB += (301-bynL[1]); totalB += (176-bynL[2]); totalB += (125-bynL[3]);
totalB += (97-bynL[4]); totalB += (79-bynL[5]); totalB += (67-bynL[6]);
totalB += (58-bynL[7]); totalB += (51-bynL[8]); totalB += (46-bynL[9]); bynL[hotbinB] -= 1;
} else hotbinB = 10;
//produce total number- smaller is closer to Benford real
if (totalA < totalB) {bynL[hotbinA] += 1; outputSample = floor(inputSampleL);}
else {bynL[hotbinB] += 1; outputSample = floor(inputSampleL+1);}
//assign the relevant one to the delay line
//and floor/ceil signal accordingly
if (cutbinsL) {
bynL[1] *= 0.99; bynL[2] *= 0.99; bynL[3] *= 0.99; bynL[4] *= 0.99; bynL[5] *= 0.99;
bynL[6] *= 0.99; bynL[7] *= 0.99; bynL[8] *= 0.99; bynL[9] *= 0.99; bynL[10] *= 0.99;
}
noiseShapingL += outputSample - drySampleL;
if (noiseShapingL > fabs(inputSampleL)) noiseShapingL = fabs(inputSampleL);
if (noiseShapingL < -fabs(inputSampleL)) noiseShapingL = -fabs(inputSampleL);
inputSampleL /= 8388608.0;
if (inputSampleL > 1.0) inputSampleL = 1.0;
if (inputSampleL < -1.0) inputSampleL = -1.0;
//finished NJAD L
//NJAD R
drySampleR = inputSampleR;
inputSampleR -= noiseShapingR;
benfordize = floor(inputSampleR);
while (benfordize >= 1.0) benfordize /= 10;
while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10;
hotbinA = floor(benfordize);
//hotbin becomes the Benford bin value for this number floored
totalA = 0.0;
if ((hotbinA > 0) && (hotbinA < 10))
{
bynR[hotbinA] += 1; if (bynR[hotbinA] > 982) cutbinsR = true;
totalA += (301-bynR[1]); totalA += (176-bynR[2]); totalA += (125-bynR[3]);
totalA += (97-bynR[4]); totalA += (79-bynR[5]); totalA += (67-bynR[6]);
totalA += (58-bynR[7]); totalA += (51-bynR[8]); totalA += (46-bynR[9]); bynR[hotbinA] -= 1;
} else hotbinA = 10;
//produce total number- smaller is closer to Benford real
benfordize = ceil(inputSampleR);
while (benfordize >= 1.0) benfordize /= 10;
while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10;
hotbinB = floor(benfordize);
//hotbin becomes the Benford bin value for this number ceiled
totalB = 0.0;
if ((hotbinB > 0) && (hotbinB < 10))
{
bynR[hotbinB] += 1; if (bynR[hotbinB] > 982) cutbinsR = true;
totalB += (301-bynR[1]); totalB += (176-bynR[2]); totalB += (125-bynR[3]);
totalB += (97-bynR[4]); totalB += (79-bynR[5]); totalB += (67-bynR[6]);
totalB += (58-bynR[7]); totalB += (51-bynR[8]); totalB += (46-bynR[9]); bynR[hotbinB] -= 1;
} else hotbinB = 10;
//produce total number- smaller is closer to Benford real
if (totalA < totalB) {bynR[hotbinA] += 1; outputSample = floor(inputSampleR);}
else {bynR[hotbinB] += 1; outputSample = floor(inputSampleR+1);}
//assign the relevant one to the delay line
//and floor/ceil signal accordingly
if (cutbinsR) {
bynR[1] *= 0.99; bynR[2] *= 0.99; bynR[3] *= 0.99; bynR[4] *= 0.99; bynR[5] *= 0.99;
bynR[6] *= 0.99; bynR[7] *= 0.99; bynR[8] *= 0.99; bynR[9] *= 0.99; bynR[10] *= 0.99;
}
noiseShapingR += outputSample - drySampleR;
if (noiseShapingR > fabs(inputSampleR)) noiseShapingR = fabs(inputSampleR);
if (noiseShapingR < -fabs(inputSampleR)) noiseShapingR = -fabs(inputSampleR);
inputSampleR /= 8388608.0;
if (inputSampleR > 1.0) inputSampleR = 1.0;
if (inputSampleR < -1.0) inputSampleR = -1.0;
break; //NJAD (Monitoring. Brightest)
case 6:
//begin 32 bit stereo floating point dither
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
break; //Bypass for saving floating point files directly
}
*out1 = inputSampleL;
*out2 = inputSampleR;
in1++;
in2++;
out1++;
out2++;
}
}
void Mastering2::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();
double iirSide = pow(A,3) * (0.1 / overallscale);
double threshSinew = (0.25+((1.0-B)*0.333))/overallscale;
double depthSinew = 1.0-pow(1.0-B,2.0);
double trebleZoom = C-0.5;
long double trebleGain = (trebleZoom*fabs(trebleZoom))+1.0;
if (trebleGain > 1.0) trebleGain = pow(trebleGain,3.0+sqrt(overallscale));
//this boost is necessary to adapt to higher sample rates
double midZoom = D-0.5;
long double midGain = (midZoom*fabs(midZoom))+1.0;
double kalMid = 0.35-(D*0.25); //crossover frequency between mid/bass
double kalSub = 0.45+(D*0.25); //crossover frequency between bass/sub
double bassZoom = (E*0.5)-0.25;
long double bassGain = (-bassZoom*fabs(bassZoom))+1.0; //control inverted
long double subGain = ((E*0.25)-0.125)+1.0;
if (subGain < 1.0) subGain = 1.0; //very small sub shift, only pos.
long double driveIn = (F-0.5)+1.0;
long double driveOut = (-(F-0.5)*fabs(F-0.5))+1.0;
int spacing = floor(overallscale); //should give us working basic scaling, usually 2 or 4
if (spacing < 1) spacing = 1; if (spacing > 16) spacing = 16;
int dither = (int) (G*5.999);
int depth = (int)(17.0*overallscale);
if (depth < 3) depth = 3; if (depth > 98) depth = 98; //for Dark
while (--sampleFrames >= 0)
{
long double inputSampleL = *in1;
long 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 *= driveIn;
inputSampleR *= driveIn;
long double mid = inputSampleL + inputSampleR;
long double side = inputSampleL - inputSampleR;
//assign mid and side.Between these sections, you can do mid/side processing
long double temp = side;
if (fpFlip) {
iirA = (iirA * (1.0 - iirSide)) + (temp * iirSide); temp = iirA;
} else {
iirB = (iirB * (1.0 - iirSide)) + (temp * iirSide); temp = iirB;
}
iirC = (iirC * (1.0 - iirSide)) + (temp * iirSide); temp = iirC;
side -= sin(temp);
fpFlip = !fpFlip;
inputSampleL = (mid+side)/2.0;
inputSampleR = (mid-side)/2.0;
//unassign mid and side
long double drySampleL = inputSampleL;
long double drySampleR = inputSampleR;
//begin Air3L
air[pvSL4] = air[pvAL4] - air[pvAL3]; air[pvSL3] = air[pvAL3] - air[pvAL2];
air[pvSL2] = air[pvAL2] - air[pvAL1]; air[pvSL1] = air[pvAL1] - inputSampleL;
air[accSL3] = air[pvSL4] - air[pvSL3]; air[accSL2] = air[pvSL3] - air[pvSL2];
air[accSL1] = air[pvSL2] - air[pvSL1];
air[acc2SL2] = air[accSL3] - air[accSL2]; air[acc2SL1] = air[accSL2] - air[accSL1];
air[outAL] = -(air[pvAL1] + air[pvSL3] + air[acc2SL2] - ((air[acc2SL2] + air[acc2SL1])*0.5));
air[gainAL] *= 0.5; air[gainAL] += fabs(drySampleL-air[outAL])*0.5;
if (air[gainAL] > 0.3*sqrt(overallscale)) air[gainAL] = 0.3*sqrt(overallscale);
air[pvAL4] = air[pvAL3]; air[pvAL3] = air[pvAL2];
air[pvAL2] = air[pvAL1]; air[pvAL1] = (air[gainAL] * air[outAL]) + drySampleL;
long double midL = drySampleL - ((air[outAL]*0.5)+(drySampleL*(0.457-(0.017*overallscale))));
temp = (midL + air[gndavgL])*0.5; air[gndavgL] = midL; midL = temp;
long double trebleL = drySampleL-midL;
//end Air3L
//begin Air3R
air[pvSR4] = air[pvAR4] - air[pvAR3]; air[pvSR3] = air[pvAR3] - air[pvAR2];
air[pvSR2] = air[pvAR2] - air[pvAR1]; air[pvSR1] = air[pvAR1] - inputSampleR;
air[accSR3] = air[pvSR4] - air[pvSR3]; air[accSR2] = air[pvSR3] - air[pvSR2];
air[accSR1] = air[pvSR2] - air[pvSR1];
air[acc2SR2] = air[accSR3] - air[accSR2]; air[acc2SR1] = air[accSR2] - air[accSR1];
air[outAR] = -(air[pvAR1] + air[pvSR3] + air[acc2SR2] - ((air[acc2SR2] + air[acc2SR1])*0.5));
air[gainAR] *= 0.5; air[gainAR] += fabs(drySampleR-air[outAR])*0.5;
if (air[gainAR] > 0.3*sqrt(overallscale)) air[gainAR] = 0.3*sqrt(overallscale);
air[pvAR4] = air[pvAR3]; air[pvAR3] = air[pvAR2];
air[pvAR2] = air[pvAR1]; air[pvAR1] = (air[gainAR] * air[outAR]) + drySampleR;
long double midR = drySampleR - ((air[outAR]*0.5)+(drySampleR*(0.457-(0.017*overallscale))));
temp = (midR + air[gndavgR])*0.5; air[gndavgR] = midR; midR = temp;
long double trebleR = drySampleR-midR;
//end Air3R
//begin KalmanML
temp = midL;
kalM[prevSlewL3] += kalM[prevSampL3] - kalM[prevSampL2]; kalM[prevSlewL3] *= 0.5;
kalM[prevSlewL2] += kalM[prevSampL2] - kalM[prevSampL1]; kalM[prevSlewL2] *= 0.5;
kalM[prevSlewL1] += kalM[prevSampL1] - midL; kalM[prevSlewL1] *= 0.5;
//make slews from each set of samples used
kalM[accSlewL2] += kalM[prevSlewL3] - kalM[prevSlewL2]; kalM[accSlewL2] *= 0.5;
kalM[accSlewL1] += kalM[prevSlewL2] - kalM[prevSlewL1]; kalM[accSlewL1] *= 0.5;
//differences between slews: rate of change of rate of change
kalM[accSlewL3] += (kalM[accSlewL2] - kalM[accSlewL1]); kalM[accSlewL3] *= 0.5;
//entering the abyss, what even is this
kalM[kalOutL] += kalM[prevSampL1] + kalM[prevSlewL2] + kalM[accSlewL3]; kalM[kalOutL] *= 0.5;
//resynthesizing predicted result (all iir smoothed)
kalM[kalGainL] += fabs(temp-kalM[kalOutL])*kalMid*8.0; kalM[kalGainL] *= 0.5;
//madness takes its toll. Kalman Gain: how much dry to retain
if (kalM[kalGainL] > kalMid*0.5) kalM[kalGainL] = kalMid*0.5;
//attempts to avoid explosions
kalM[kalOutL] += (temp*(1.0-(0.68+(kalMid*0.157))));
//this is for tuning a really complete cancellation up around Nyquist
kalM[prevSampL3] = kalM[prevSampL2]; kalM[prevSampL2] = kalM[prevSampL1];
kalM[prevSampL1] = (kalM[kalGainL] * kalM[kalOutL]) + ((1.0-kalM[kalGainL])*temp);
//feed the chain of previous samples
long double bassL = (kalM[kalOutL]+kalM[kalAvgL])*0.5;
kalM[kalAvgL] = kalM[kalOutL];
midL -= bassL;
//end KalmanML
//begin KalmanMR
temp = midR;
kalM[prevSlewR3] += kalM[prevSampR3] - kalM[prevSampR2]; kalM[prevSlewR3] *= 0.5;
kalM[prevSlewR2] += kalM[prevSampR2] - kalM[prevSampR1]; kalM[prevSlewR2] *= 0.5;
kalM[prevSlewR1] += kalM[prevSampR1] - midR; kalM[prevSlewR1] *= 0.5;
//make slews from each set of samples used
kalM[accSlewR2] += kalM[prevSlewR3] - kalM[prevSlewR2]; kalM[accSlewR2] *= 0.5;
kalM[accSlewR1] += kalM[prevSlewR2] - kalM[prevSlewR1]; kalM[accSlewR1] *= 0.5;
//differences between slews: rate of change of rate of change
kalM[accSlewR3] += (kalM[accSlewR2] - kalM[accSlewR1]); kalM[accSlewR3] *= 0.5;
//entering the abyss, what even is this
kalM[kalOutR] += kalM[prevSampR1] + kalM[prevSlewR2] + kalM[accSlewR3]; kalM[kalOutR] *= 0.5;
//resynthesizing predicted result (all iir smoothed)
kalM[kalGainR] += fabs(temp-kalM[kalOutR])*kalMid*8.0; kalM[kalGainR] *= 0.5;
//madness takes its toll. Kalman Gain: how much dry to retain
if (kalM[kalGainR] > kalMid*0.5) kalM[kalGainR] = kalMid*0.5;
//attempts to avoid explosions
kalM[kalOutR] += (temp*(1.0-(0.68+(kalMid*0.157))));
//this is for tuning a really complete cancellation up around Nyquist
kalM[prevSampR3] = kalM[prevSampR2]; kalM[prevSampR2] = kalM[prevSampR1];
kalM[prevSampR1] = (kalM[kalGainR] * kalM[kalOutR]) + ((1.0-kalM[kalGainR])*temp);
//feed the chain of previous samples
long double bassR = (kalM[kalOutR]+kalM[kalAvgR])*0.5;
kalM[kalAvgR] = kalM[kalOutR];
midR -= bassR;
//end KalmanMR
//begin KalmanSL
temp = bassL;
kalS[prevSlewL3] += kalS[prevSampL3] - kalS[prevSampL2]; kalS[prevSlewL3] *= 0.5;
kalS[prevSlewL2] += kalS[prevSampL2] - kalS[prevSampL1]; kalS[prevSlewL2] *= 0.5;
kalS[prevSlewL1] += kalS[prevSampL1] - bassL; kalS[prevSlewL1] *= 0.5;
//make slews from each set of samples used
kalS[accSlewL2] += kalS[prevSlewL3] - kalS[prevSlewL2]; kalS[accSlewL2] *= 0.5;
kalS[accSlewL1] += kalS[prevSlewL2] - kalS[prevSlewL1]; kalS[accSlewL1] *= 0.5;
//differences between slews: rate of change of rate of change
kalS[accSlewL3] += (kalS[accSlewL2] - kalS[accSlewL1]); kalS[accSlewL3] *= 0.5;
//entering the abyss, what even is this
kalS[kalOutL] += kalS[prevSampL1] + kalS[prevSlewL2] + kalS[accSlewL3]; kalS[kalOutL] *= 0.5;
//resynthesizing predicted result (all iir smoothed)
kalS[kalGainL] += fabs(temp-kalS[kalOutL])*kalSub*8.0; kalS[kalGainL] *= 0.5;
//madness takes its toll. Kalman Gain: how much dry to retain
if (kalS[kalGainL] > kalSub*0.5) kalS[kalGainL] = kalSub*0.5;
//attempts to avoid explosions
kalS[kalOutL] += (temp*(1.0-(0.68+(kalSub*0.157))));
//this is for tuning a really complete cancellation up around Nyquist
kalS[prevSampL3] = kalS[prevSampL2]; kalS[prevSampL2] = kalS[prevSampL1];
kalS[prevSampL1] = (kalS[kalGainL] * kalS[kalOutL]) + ((1.0-kalS[kalGainL])*temp);
//feed the chain of previous samples
long double subL = (kalS[kalOutL]+kalS[kalAvgL])*0.5;
kalS[kalAvgL] = kalS[kalOutL];
bassL -= subL;
//end KalmanSL
//begin KalmanSR
temp = bassR;
kalS[prevSlewR3] += kalS[prevSampR3] - kalS[prevSampR2]; kalS[prevSlewR3] *= 0.5;
kalS[prevSlewR2] += kalS[prevSampR2] - kalS[prevSampR1]; kalS[prevSlewR2] *= 0.5;
kalS[prevSlewR1] += kalS[prevSampR1] - bassR; kalS[prevSlewR1] *= 0.5;
//make slews from each set of samples used
kalS[accSlewR2] += kalS[prevSlewR3] - kalS[prevSlewR2]; kalS[accSlewR2] *= 0.5;
kalS[accSlewR1] += kalS[prevSlewR2] - kalS[prevSlewR1]; kalS[accSlewR1] *= 0.5;
//differences between slews: rate of change of rate of change
kalS[accSlewR3] += (kalS[accSlewR2] - kalS[accSlewR1]); kalS[accSlewR3] *= 0.5;
//entering the abyss, what even is this
kalS[kalOutR] += kalS[prevSampR1] + kalS[prevSlewR2] + kalS[accSlewR3]; kalS[kalOutR] *= 0.5;
//resynthesizing predicted result (all iir smoothed)
kalS[kalGainR] += fabs(temp-kalS[kalOutR])*kalSub*8.0; kalS[kalGainR] *= 0.5;
//madness takes its toll. Kalman Gain: how much dry to retain
if (kalS[kalGainR] > kalSub*0.5) kalS[kalGainR] = kalSub*0.5;
//attempts to avoid explosions
kalS[kalOutR] += (temp*(1.0-(0.68+(kalSub*0.157))));
//this is for tuning a really complete cancellation up around Nyquist
kalS[prevSampR3] = kalS[prevSampR2]; kalS[prevSampR2] = kalS[prevSampR1];
kalS[prevSampR1] = (kalS[kalGainR] * kalS[kalOutR]) + ((1.0-kalS[kalGainR])*temp);
//feed the chain of previous samples
long double subR = (kalS[kalOutR]+kalS[kalAvgR])*0.5;
kalS[kalAvgR] = kalS[kalOutR];
bassR -= subR;
//end KalmanSR
inputSampleL = (subL*subGain);
inputSampleR = (subR*subGain);
if (bassZoom > 0.0) {
double closer = bassL * 1.57079633;
if (closer > 1.57079633) closer = 1.57079633;
if (closer < -1.57079633) closer = -1.57079633;
bassL = (bassL*(1.0-bassZoom))+(sin(closer)*bassZoom);
closer = bassR * 1.57079633;
if (closer > 1.57079633) closer = 1.57079633;
if (closer < -1.57079633) closer = -1.57079633;
bassR = (bassR*(1.0-bassZoom))+(sin(closer)*bassZoom);
} //zooming in will make the body of the sound louder: it's just Density
if (bassZoom < 0.0) {
double farther = fabs(bassL) * 1.57079633;
if (farther > 1.57079633) farther = 1.0;
else farther = 1.0-cos(farther);
if (bassL > 0.0) bassL = (bassL*(1.0+bassZoom))-(farther*bassZoom*1.57079633);
if (bassL < 0.0) bassL = (bassL*(1.0+bassZoom))+(farther*bassZoom*1.57079633);
farther = fabs(bassR) * 1.57079633;
if (farther > 1.57079633) farther = 1.0;
else farther = 1.0-cos(farther);
if (bassR > 0.0) bassR = (bassR*(1.0+bassZoom))-(farther*bassZoom*1.57079633);
if (bassR < 0.0) bassR = (bassR*(1.0+bassZoom))+(farther*bassZoom*1.57079633);
} //zooming out boosts the hottest peaks but cuts back softer stuff
inputSampleL += (bassL*bassGain);
inputSampleR += (bassR*bassGain);
if (midZoom > 0.0) {
double closer = midL * 1.57079633;
if (closer > 1.57079633) closer = 1.57079633;
if (closer < -1.57079633) closer = -1.57079633;
midL = (midL*(1.0-midZoom))+(sin(closer)*midZoom);
closer = midR * 1.57079633;
if (closer > 1.57079633) closer = 1.57079633;
if (closer < -1.57079633) closer = -1.57079633;
midR = (midR*(1.0-midZoom))+(sin(closer)*midZoom);
} //zooming in will make the body of the sound louder: it's just Density
if (midZoom < 0.0) {
double farther = fabs(midL) * 1.57079633;
if (farther > 1.57079633) farther = 1.0;
else farther = 1.0-cos(farther);
if (midL > 0.0) midL = (midL*(1.0+midZoom))-(farther*midZoom*1.57079633);
if (midL < 0.0) midL = (midL*(1.0+midZoom))+(farther*midZoom*1.57079633);
farther = fabs(midR) * 1.57079633;
if (farther > 1.57079633) farther = 1.0;
else farther = 1.0-cos(farther);
if (midR > 0.0) midR = (midR*(1.0+midZoom))-(farther*midZoom*1.57079633);
if (midR < 0.0) midR = (midR*(1.0+midZoom))+(farther*midZoom*1.57079633);
} //zooming out boosts the hottest peaks but cuts back softer stuff
inputSampleL += (midL*midGain);
inputSampleR += (midR*midGain);
if (trebleZoom > 0.0) {
double closer = trebleL * 1.57079633;
if (closer > 1.57079633) closer = 1.57079633;
if (closer < -1.57079633) closer = -1.57079633;
trebleL = (trebleL*(1.0-trebleZoom))+(sin(closer)*trebleZoom);
closer = trebleR * 1.57079633;
if (closer > 1.57079633) closer = 1.57079633;
if (closer < -1.57079633) closer = -1.57079633;
trebleR = (trebleR*(1.0-trebleZoom))+(sin(closer)*trebleZoom);
} //zooming in will make the body of the sound louder: it's just Density
if (trebleZoom < 0.0) {
double farther = fabs(trebleL) * 1.57079633;
if (farther > 1.57079633) farther = 1.0;
else farther = 1.0-cos(farther);
if (trebleL > 0.0) trebleL = (trebleL*(1.0+trebleZoom))-(farther*trebleZoom*1.57079633);
if (trebleL < 0.0) trebleL = (trebleL*(1.0+trebleZoom))+(farther*trebleZoom*1.57079633);
farther = fabs(trebleR) * 1.57079633;
if (farther > 1.57079633) farther = 1.0;
else farther = 1.0-cos(farther);
if (trebleR > 0.0) trebleR = (trebleR*(1.0+trebleZoom))-(farther*trebleZoom*1.57079633);
if (trebleR < 0.0) trebleR = (trebleR*(1.0+trebleZoom))+(farther*trebleZoom*1.57079633);
} //zooming out boosts the hottest peaks but cuts back softer stuff
inputSampleL += (trebleL*trebleGain);
inputSampleR += (trebleR*trebleGain);
inputSampleL *= driveOut;
inputSampleR *= driveOut;
//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
temp = inputSampleL;
long double sinew = threshSinew * cos(lastSinewL*lastSinewL);
if (inputSampleL - lastSinewL > sinew) temp = lastSinewL + sinew;
if (-(inputSampleL - lastSinewL) > sinew) temp = lastSinewL - sinew;
lastSinewL = temp;
inputSampleL = (inputSampleL * (1.0-depthSinew))+(lastSinewL*depthSinew);
temp = inputSampleR;
sinew = threshSinew * cos(lastSinewR*lastSinewR);
if (inputSampleR - lastSinewR > sinew) temp = lastSinewR + sinew;
if (-(inputSampleR - lastSinewR) > sinew) temp = lastSinewR - sinew;
lastSinewR = temp;
inputSampleR = (inputSampleR * (1.0-depthSinew))+(lastSinewR*depthSinew);
//run Sinew to stop excess slews, but run a dry/wet to allow a range of brights
switch (dither) {
case 1:
//begin Dark
inputSampleL *= 8388608.0;
inputSampleR *= 8388608.0; //we will apply the 24 bit Dark
//We are doing it first Left, then Right, because the loops may run faster if
//they aren't too jammed full of variables. This means re-running code.
//begin left
quantA = floor(inputSampleL);
quantB = floor(inputSampleL+1.0);
//to do this style of dither, we quantize in either direction and then
//do a reconstruction of what the result will be for each choice.
//We then evaluate which one we like, and keep a history of what we previously had
expectedSlew = 0;
for(int x = 0; x < depth; x++) {
expectedSlew += (darkSampleL[x+1] - darkSampleL[x]);
}
expectedSlew /= depth; //we have an average of all recent slews
//we are doing that to voice the thing down into the upper mids a bit
//it mustn't just soften the brightest treble, it must smooth high mids too
testA = fabs((darkSampleL[0] - quantA) - expectedSlew);
testB = fabs((darkSampleL[0] - quantB) - expectedSlew);
if (testA < testB) inputSampleL = quantA;
else inputSampleL = quantB;
//select whichever one departs LEAST from the vector of averaged
//reconstructed previous final samples. This will force a kind of dithering
//as it'll make the output end up as smooth as possible
for(int x = depth; x >=0; x--) {
darkSampleL[x+1] = darkSampleL[x];
}
darkSampleL[0] = inputSampleL;
//end Dark left
//begin right
quantA = floor(inputSampleR);
quantB = floor(inputSampleR+1.0);
//to do this style of dither, we quantize in either direction and then
//do a reconstruction of what the result will be for each choice.
//We then evaluate which one we like, and keep a history of what we previously had
expectedSlew = 0;
for(int x = 0; x < depth; x++) {
expectedSlew += (darkSampleR[x+1] - darkSampleR[x]);
}
expectedSlew /= depth; //we have an average of all recent slews
//we are doing that to voice the thing down into the upper mids a bit
//it mustn't just soften the brightest treble, it must smooth high mids too
testA = fabs((darkSampleR[0] - quantA) - expectedSlew);
testB = fabs((darkSampleR[0] - quantB) - expectedSlew);
if (testA < testB) inputSampleR = quantA;
else inputSampleR = quantB;
//select whichever one departs LEAST from the vector of averaged
//reconstructed previous final samples. This will force a kind of dithering
//as it'll make the output end up as smooth as possible
for(int x = depth; x >=0; x--) {
darkSampleR[x+1] = darkSampleR[x];
}
darkSampleR[0] = inputSampleR;
//end Dark right
inputSampleL /= 8388608.0;
inputSampleR /= 8388608.0;
break; //Dark (Monitoring2)
case 2:
//begin Dark for Ten Nines
inputSampleL *= 8388608.0;
inputSampleR *= 8388608.0; //we will apply the 24 bit Dark
//We are doing it first Left, then Right, because the loops may run faster if
//they aren't too jammed full of variables. This means re-running code.
//begin L
correction = 0;
if (flip) {
NSOddL = (NSOddL * 0.9999999999) + prevShapeL;
NSEvenL = (NSEvenL * 0.9999999999) - prevShapeL;
correction = NSOddL;
} else {
NSOddL = (NSOddL * 0.9999999999) - prevShapeL;
NSEvenL = (NSEvenL * 0.9999999999) + prevShapeL;
correction = NSEvenL;
}
shapedSampleL = inputSampleL+correction;
//end Ten Nines L
//begin Dark L
quantA = floor(shapedSampleL);
quantB = floor(shapedSampleL+1.0);
//to do this style of dither, we quantize in either direction and then
//do a reconstruction of what the result will be for each choice.
//We then evaluate which one we like, and keep a history of what we previously had
expectedSlew = 0;
for(int x = 0; x < depth; x++) {
expectedSlew += (darkSampleL[x+1] - darkSampleL[x]);
}
expectedSlew /= depth; //we have an average of all recent slews
//we are doing that to voice the thing down into the upper mids a bit
//it mustn't just soften the brightest treble, it must smooth high mids too
testA = fabs((darkSampleL[0] - quantA) - expectedSlew);
testB = fabs((darkSampleL[0] - quantB) - expectedSlew);
if (testA < testB) inputSampleL = quantA;
else inputSampleL = quantB;
//select whichever one departs LEAST from the vector of averaged
//reconstructed previous final samples. This will force a kind of dithering
//as it'll make the output end up as smooth as possible
for(int x = depth; x >=0; x--) {
darkSampleL[x+1] = darkSampleL[x];
}
darkSampleL[0] = inputSampleL;
//end Dark L
prevShapeL = (floor(shapedSampleL) - inputSampleL)*0.9999999999;
//end Ten Nines L
//begin R
correction = 0;
if (flip) {
NSOddR = (NSOddR * 0.9999999999) + prevShapeR;
NSEvenR = (NSEvenR * 0.9999999999) - prevShapeR;
correction = NSOddR;
} else {
NSOddR = (NSOddR * 0.9999999999) - prevShapeR;
NSEvenR = (NSEvenR * 0.9999999999) + prevShapeR;
correction = NSEvenR;
}
shapedSampleR = inputSampleR+correction;
//end Ten Nines R
//begin Dark R
quantA = floor(shapedSampleR);
quantB = floor(shapedSampleR+1.0);
//to do this style of dither, we quantize in either direction and then
//do a reconstruction of what the result will be for each choice.
//We then evaluate which one we like, and keep a history of what we previously had
expectedSlew = 0;
for(int x = 0; x < depth; x++) {
expectedSlew += (darkSampleR[x+1] - darkSampleR[x]);
}
expectedSlew /= depth; //we have an average of all recent slews
//we are doing that to voice the thing down into the upper mids a bit
//it mustn't just soften the brightest treble, it must smooth high mids too
testA = fabs((darkSampleR[0] - quantA) - expectedSlew);
testB = fabs((darkSampleR[0] - quantB) - expectedSlew);
if (testA < testB) inputSampleR = quantA;
else inputSampleR = quantB;
//select whichever one departs LEAST from the vector of averaged
//reconstructed previous final samples. This will force a kind of dithering
//as it'll make the output end up as smooth as possible
for(int x = depth; x >=0; x--) {
darkSampleR[x+1] = darkSampleR[x];
}
darkSampleR[0] = inputSampleR;
//end Dark R
prevShapeR = (floor(shapedSampleR) - inputSampleR)*0.9999999999;
//end Ten Nines
flip = !flip;
inputSampleL /= 8388608.0;
inputSampleR /= 8388608.0;
break; //Ten Nines (which goes into Dark in Monitoring3)
case 3:
inputSampleL *= 8388608.0;
inputSampleR *= 8388608.0;
ditherL = -1.0;
ditherL += (double(fpdL)/UINT32_MAX);
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
ditherL += (double(fpdL)/UINT32_MAX);
//TPDF: two 0-1 random noises
ditherR = -1.0;
ditherR += (double(fpdR)/UINT32_MAX);
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
ditherR += (double(fpdR)/UINT32_MAX);
//TPDF: two 0-1 random noises
if (fabs(ditherL-ditherR) < 0.5) {
ditherL = -1.0;
ditherL += (double(fpdL)/UINT32_MAX);
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
ditherL += (double(fpdL)/UINT32_MAX);
}
if (fabs(ditherL-ditherR) < 0.5) {
ditherR = -1.0;
ditherR += (double(fpdR)/UINT32_MAX);
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
ditherR += (double(fpdR)/UINT32_MAX);
}
if (fabs(ditherL-ditherR) < 0.5) {
ditherL = -1.0;
ditherL += (double(fpdL)/UINT32_MAX);
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
ditherL += (double(fpdL)/UINT32_MAX);
}
inputSampleL = floor(inputSampleL+ditherL);
inputSampleR = floor(inputSampleR+ditherR);
inputSampleL /= 8388608.0;
inputSampleR /= 8388608.0;
break; //TPDFWide (a good neutral with the width enhancement)
case 4:
inputSampleL *= 8388608.0;
inputSampleR *= 8388608.0;
//Paul Frindle: It's true that the dither itself can sound different
//if it's given a different freq response and you get to hear it.
//The one we use most is triangular single pole high pass dither.
//It's not freq bent enough to sound odd, but is slightly less audible than
//flat dither. It can also be easily made by taking one sample of dither
//away from the previous one - this gives you the triangular PDF and the
//filtering in one go :-)
currentDither = (double(fpdL)/UINT32_MAX);
ditherL = currentDither;
ditherL -= previousDitherL;
previousDitherL = currentDither;
//TPDF: two 0-1 random noises
currentDither = (double(fpdR)/UINT32_MAX);
ditherR = currentDither;
ditherR -= previousDitherR;
previousDitherR = currentDither;
//TPDF: two 0-1 random noises
if (fabs(ditherL-ditherR) < 0.5) {
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
currentDither = (double(fpdL)/UINT32_MAX);
ditherL = currentDither;
ditherL -= previousDitherL;
previousDitherL = currentDither;
}
if (fabs(ditherL-ditherR) < 0.5) {
fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
currentDither = (double(fpdR)/UINT32_MAX);
ditherR = currentDither;
ditherR -= previousDitherR;
previousDitherR = currentDither;
}
if (fabs(ditherL-ditherR) < 0.5) {
fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
currentDither = (double(fpdL)/UINT32_MAX);
ditherL = currentDither;
ditherL -= previousDitherL;
previousDitherL = currentDither;
}
inputSampleL = floor(inputSampleL+ditherL);
inputSampleR = floor(inputSampleR+ditherR);
inputSampleL /= 8388608.0;
inputSampleR /= 8388608.0;
break; //PaulWide (brighter neutral that's still TPDF and wide)
case 5:
inputSampleL *= 8388608.0;
inputSampleR *= 8388608.0;
cutbinsL = false;
cutbinsR = false;
drySampleL = inputSampleL;//re-using in NJAD
inputSampleL -= noiseShapingL;
//NJAD L
benfordize = floor(inputSampleL);
while (benfordize >= 1.0) benfordize /= 10;
while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10;
hotbinA = floor(benfordize);
//hotbin becomes the Benford bin value for this number floored
totalA = 0.0;
if ((hotbinA > 0) && (hotbinA < 10))
{
bynL[hotbinA] += 1; if (bynL[hotbinA] > 982) cutbinsL = true;
totalA += (301-bynL[1]); totalA += (176-bynL[2]); totalA += (125-bynL[3]);
totalA += (97-bynL[4]); totalA += (79-bynL[5]); totalA += (67-bynL[6]);
totalA += (58-bynL[7]); totalA += (51-bynL[8]); totalA += (46-bynL[9]); bynL[hotbinA] -= 1;
} else hotbinA = 10;
//produce total number- smaller is closer to Benford real
benfordize = ceil(inputSampleL);
while (benfordize >= 1.0) benfordize /= 10;
while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10;
hotbinB = floor(benfordize);
//hotbin becomes the Benford bin value for this number ceiled
totalB = 0.0;
if ((hotbinB > 0) && (hotbinB < 10))
{
bynL[hotbinB] += 1; if (bynL[hotbinB] > 982) cutbinsL = true;
totalB += (301-bynL[1]); totalB += (176-bynL[2]); totalB += (125-bynL[3]);
totalB += (97-bynL[4]); totalB += (79-bynL[5]); totalB += (67-bynL[6]);
totalB += (58-bynL[7]); totalB += (51-bynL[8]); totalB += (46-bynL[9]); bynL[hotbinB] -= 1;
} else hotbinB = 10;
//produce total number- smaller is closer to Benford real
if (totalA < totalB) {bynL[hotbinA] += 1; outputSample = floor(inputSampleL);}
else {bynL[hotbinB] += 1; outputSample = floor(inputSampleL+1);}
//assign the relevant one to the delay line
//and floor/ceil signal accordingly
if (cutbinsL) {
bynL[1] *= 0.99; bynL[2] *= 0.99; bynL[3] *= 0.99; bynL[4] *= 0.99; bynL[5] *= 0.99;
bynL[6] *= 0.99; bynL[7] *= 0.99; bynL[8] *= 0.99; bynL[9] *= 0.99; bynL[10] *= 0.99;
}
noiseShapingL += outputSample - drySampleL;
if (noiseShapingL > fabs(inputSampleL)) noiseShapingL = fabs(inputSampleL);
if (noiseShapingL < -fabs(inputSampleL)) noiseShapingL = -fabs(inputSampleL);
inputSampleL /= 8388608.0;
if (inputSampleL > 1.0) inputSampleL = 1.0;
if (inputSampleL < -1.0) inputSampleL = -1.0;
//finished NJAD L
//NJAD R
drySampleR = inputSampleR;
inputSampleR -= noiseShapingR;
benfordize = floor(inputSampleR);
while (benfordize >= 1.0) benfordize /= 10;
while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10;
hotbinA = floor(benfordize);
//hotbin becomes the Benford bin value for this number floored
totalA = 0.0;
if ((hotbinA > 0) && (hotbinA < 10))
{
bynR[hotbinA] += 1; if (bynR[hotbinA] > 982) cutbinsR = true;
totalA += (301-bynR[1]); totalA += (176-bynR[2]); totalA += (125-bynR[3]);
totalA += (97-bynR[4]); totalA += (79-bynR[5]); totalA += (67-bynR[6]);
totalA += (58-bynR[7]); totalA += (51-bynR[8]); totalA += (46-bynR[9]); bynR[hotbinA] -= 1;
} else hotbinA = 10;
//produce total number- smaller is closer to Benford real
benfordize = ceil(inputSampleR);
while (benfordize >= 1.0) benfordize /= 10;
while (benfordize < 1.0 && benfordize > 0.0000001) benfordize *= 10;
hotbinB = floor(benfordize);
//hotbin becomes the Benford bin value for this number ceiled
totalB = 0.0;
if ((hotbinB > 0) && (hotbinB < 10))
{
bynR[hotbinB] += 1; if (bynR[hotbinB] > 982) cutbinsR = true;
totalB += (301-bynR[1]); totalB += (176-bynR[2]); totalB += (125-bynR[3]);
totalB += (97-bynR[4]); totalB += (79-bynR[5]); totalB += (67-bynR[6]);
totalB += (58-bynR[7]); totalB += (51-bynR[8]); totalB += (46-bynR[9]); bynR[hotbinB] -= 1;
} else hotbinB = 10;
//produce total number- smaller is closer to Benford real
if (totalA < totalB) {bynR[hotbinA] += 1; outputSample = floor(inputSampleR);}
else {bynR[hotbinB] += 1; outputSample = floor(inputSampleR+1);}
//assign the relevant one to the delay line
//and floor/ceil signal accordingly
if (cutbinsR) {
bynR[1] *= 0.99; bynR[2] *= 0.99; bynR[3] *= 0.99; bynR[4] *= 0.99; bynR[5] *= 0.99;
bynR[6] *= 0.99; bynR[7] *= 0.99; bynR[8] *= 0.99; bynR[9] *= 0.99; bynR[10] *= 0.99;
}
noiseShapingR += outputSample - drySampleR;
if (noiseShapingR > fabs(inputSampleR)) noiseShapingR = fabs(inputSampleR);
if (noiseShapingR < -fabs(inputSampleR)) noiseShapingR = -fabs(inputSampleR);
inputSampleR /= 8388608.0;
if (inputSampleR > 1.0) inputSampleR = 1.0;
if (inputSampleR < -1.0) inputSampleR = -1.0;
break; //NJAD (Monitoring. Brightest)
case 6:
//begin 64 bit stereo floating point dither
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
break; //Bypass for saving floating point files directly
}
*out1 = inputSampleL;
*out2 = inputSampleR;
in1++;
in2++;
out1++;
out2++;
}
}
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