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Diffstat (limited to 'plugins/MacVST/UnBox/source/UnBoxProc.cpp')
| -rwxr-xr-x | plugins/MacVST/UnBox/source/UnBoxProc.cpp | 488 |
1 files changed, 488 insertions, 0 deletions
diff --git a/plugins/MacVST/UnBox/source/UnBoxProc.cpp b/plugins/MacVST/UnBox/source/UnBoxProc.cpp new file mode 100755 index 0000000..12133dc --- /dev/null +++ b/plugins/MacVST/UnBox/source/UnBoxProc.cpp @@ -0,0 +1,488 @@ +/* ======================================== + * UnBox - UnBox.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __UnBox_H +#include "UnBox.h" +#endif + +void UnBox::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 input = A*2.0; + double unbox = B+1.0; + unbox *= unbox; //let's get some more gain into this + double iirAmount = (unbox*0.00052)/overallscale; + double output = C*2.0; + + double treble = unbox; //averaging taps 1-4 + double gain = treble; + if (gain > 1.0) {e[0] = 1.0; gain -= 1.0;} else {e[0] = gain; gain = 0.0;} + if (gain > 1.0) {e[1] = 1.0; gain -= 1.0;} else {e[1] = gain; gain = 0.0;} + if (gain > 1.0) {e[2] = 1.0; gain -= 1.0;} else {e[2] = gain; gain = 0.0;} + if (gain > 1.0) {e[3] = 1.0; gain -= 1.0;} else {e[3] = gain; gain = 0.0;} + if (gain > 1.0) {e[4] = 1.0; gain -= 1.0;} else {e[4] = gain; gain = 0.0;} + //there, now we have a neat little moving average with remainders + if (treble < 1.0) treble = 1.0; + e[0] /= treble; + e[1] /= treble; + e[2] /= treble; + e[3] /= treble; + e[4] /= treble; + //and now it's neatly scaled, too + + treble = unbox*2.0; //averaging taps 1-8 + gain = treble; + if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;} + if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;} + if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;} + if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;} + if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;} + if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;} + if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;} + if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;} + if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;} + if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;} + //there, now we have a neat little moving average with remainders + if (treble < 1.0) treble = 1.0; + f[0] /= treble; + f[1] /= treble; + f[2] /= treble; + f[3] /= treble; + f[4] /= treble; + f[5] /= treble; + f[6] /= treble; + f[7] /= treble; + f[8] /= treble; + f[9] /= treble; + //and now it's neatly scaled, too + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double inputSampleR = *in2; + + if (input != 1.0) {inputSampleL *= input; inputSampleR *= input;} + + static int noisesourceL = 0; + static int noisesourceR = 850010; + int residue; + double applyresidue; + + noisesourceL = noisesourceL % 1700021; noisesourceL++; + residue = noisesourceL * noisesourceL; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL += applyresidue; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + inputSampleL -= applyresidue; + } + + noisesourceR = noisesourceR % 1700021; noisesourceR++; + residue = noisesourceR * noisesourceR; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR += applyresidue; + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + inputSampleR -= applyresidue; + } + //for live air, we always apply the dither noise. Then, if our result is + //effectively digital black, we'll subtract it aUnBox. We want a 'air' hiss + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + aL[4] = aL[3]; aL[3] = aL[2]; aL[2] = aL[1]; + aL[1] = aL[0]; aL[0] = inputSampleL; + inputSampleL *= e[0]; + inputSampleL += (aL[1] * e[1]); + inputSampleL += (aL[2] * e[2]); + inputSampleL += (aL[3] * e[3]); + inputSampleL += (aL[4] * e[4]); + //this is now an average of inputSampleL + + aR[4] = aR[3]; aR[3] = aR[2]; aR[2] = aR[1]; + aR[1] = aR[0]; aR[0] = inputSampleR; + inputSampleR *= e[0]; + inputSampleR += (aR[1] * e[1]); + inputSampleR += (aR[2] * e[2]); + inputSampleR += (aR[3] * e[3]); + inputSampleR += (aR[4] * e[4]); + //this is now an average of inputSampleR + + bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1]; + bL[1] = bL[0]; bL[0] = inputSampleL; + inputSampleL *= e[0]; + inputSampleL += (bL[1] * e[1]); + inputSampleL += (bL[2] * e[2]); + inputSampleL += (bL[3] * e[3]); + inputSampleL += (bL[4] * e[4]); + //this is now an average of an average of inputSampleL. Two poles + + bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1]; + bR[1] = bR[0]; bR[0] = inputSampleR; + inputSampleR *= e[0]; + inputSampleR += (bR[1] * e[1]); + inputSampleR += (bR[2] * e[2]); + inputSampleR += (bR[3] * e[3]); + inputSampleR += (bR[4] * e[4]); + //this is now an average of an average of inputSampleR. Two poles + + inputSampleL *= unbox; + inputSampleR *= unbox; + //clip to 1.2533141373155 to reach maximum output + if (inputSampleL > 1.2533141373155) inputSampleL = 1.2533141373155; + if (inputSampleL < -1.2533141373155) inputSampleL = -1.2533141373155; + inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL)); + + if (inputSampleR > 1.2533141373155) inputSampleR = 1.2533141373155; + if (inputSampleR < -1.2533141373155) inputSampleR = -1.2533141373155; + inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR)); + + inputSampleL /= unbox; + inputSampleR /= unbox; + //now we have a distorted inputSample at the correct volume relative to drySample + + long double accumulatorSampleL = (drySampleL - inputSampleL); + cL[9] = cL[8]; cL[8] = cL[7]; cL[7] = cL[6]; cL[6] = cL[5]; + cL[5] = cL[4]; cL[4] = cL[3]; cL[3] = cL[2]; cL[2] = cL[1]; + cL[1] = cL[0]; cL[0] = accumulatorSampleL; + accumulatorSampleL *= f[0]; + accumulatorSampleL += (cL[1] * f[1]); + accumulatorSampleL += (cL[2] * f[2]); + accumulatorSampleL += (cL[3] * f[3]); + accumulatorSampleL += (cL[4] * f[4]); + accumulatorSampleL += (cL[5] * f[5]); + accumulatorSampleL += (cL[6] * f[6]); + accumulatorSampleL += (cL[7] * f[7]); + accumulatorSampleL += (cL[8] * f[8]); + accumulatorSampleL += (cL[9] * f[9]); + //this is now an average of all the recent variances from dry + + long double accumulatorSampleR = (drySampleR - inputSampleR); + cR[9] = cR[8]; cR[8] = cR[7]; cR[7] = cR[6]; cR[6] = cR[5]; + cR[5] = cR[4]; cR[4] = cR[3]; cR[3] = cR[2]; cR[2] = cR[1]; + cR[1] = cR[0]; cR[0] = accumulatorSampleR; + accumulatorSampleR *= f[0]; + accumulatorSampleR += (cR[1] * f[1]); + accumulatorSampleR += (cR[2] * f[2]); + accumulatorSampleR += (cR[3] * f[3]); + accumulatorSampleR += (cR[4] * f[4]); + accumulatorSampleR += (cR[5] * f[5]); + accumulatorSampleR += (cR[6] * f[6]); + accumulatorSampleR += (cR[7] * f[7]); + accumulatorSampleR += (cR[8] * f[8]); + accumulatorSampleR += (cR[9] * f[9]); + //this is now an average of all the recent variances from dry + + iirSampleAL = (iirSampleAL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount); + accumulatorSampleL -= iirSampleAL; + //two poles of IIR + + iirSampleAR = (iirSampleAR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount); + accumulatorSampleR -= iirSampleAR; + //two poles of IIR + + iirSampleBL = (iirSampleBL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount); + accumulatorSampleL -= iirSampleBL; + //highpass section + + iirSampleBR = (iirSampleBR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount); + accumulatorSampleR -= iirSampleBR; + //highpass section + //this is now a highpassed average of all the recent variances from dry + + inputSampleL = drySampleL - accumulatorSampleL; + inputSampleR = drySampleR - accumulatorSampleR; + //we apply it as one operation, to get the result. + + if (output != 1.0) {inputSampleL *= output; inputSampleR *= output;} + + //noise shaping to 32-bit floating point + float fpTemp = inputSampleL; + fpNShapeL += (inputSampleL-fpTemp); + inputSampleL += fpNShapeL; + //if this confuses you look at the wordlength for fpTemp :) + fpTemp = inputSampleR; + fpNShapeR += (inputSampleR-fpTemp); + inputSampleR += fpNShapeR; + //for deeper space and warmth, we try a non-oscillating noise shaping + //that is kind of ruthless: it will forever retain the rounding errors + //except we'll dial it back a hair at the end of every buffer processed + //end noise shaping on 32 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } + fpNShapeL *= 0.999999; + fpNShapeR *= 0.999999; + //we will just delicately dial back the FP noise shaping, not even every sample + //this is a good place to put subtle 'no runaway' calculations, though bear in mind + //that it will be called more often when you use shorter sample buffers in the DAW. + //So, very low latency operation will call these calculations more often. +} + +void UnBox::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 input = A*2.0; + double unbox = B+1.0; + unbox *= unbox; //let's get some more gain into this + double iirAmount = (unbox*0.00052)/overallscale; + double output = C*2.0; + + double treble = unbox; //averaging taps 1-4 + double gain = treble; + if (gain > 1.0) {e[0] = 1.0; gain -= 1.0;} else {e[0] = gain; gain = 0.0;} + if (gain > 1.0) {e[1] = 1.0; gain -= 1.0;} else {e[1] = gain; gain = 0.0;} + if (gain > 1.0) {e[2] = 1.0; gain -= 1.0;} else {e[2] = gain; gain = 0.0;} + if (gain > 1.0) {e[3] = 1.0; gain -= 1.0;} else {e[3] = gain; gain = 0.0;} + if (gain > 1.0) {e[4] = 1.0; gain -= 1.0;} else {e[4] = gain; gain = 0.0;} + //there, now we have a neat little moving average with remainders + if (treble < 1.0) treble = 1.0; + e[0] /= treble; + e[1] /= treble; + e[2] /= treble; + e[3] /= treble; + e[4] /= treble; + //and now it's neatly scaled, too + + treble = unbox*2.0; //averaging taps 1-8 + gain = treble; + if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;} + if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;} + if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;} + if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;} + if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;} + if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;} + if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;} + if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;} + if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;} + if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;} + //there, now we have a neat little moving average with remainders + if (treble < 1.0) treble = 1.0; + f[0] /= treble; + f[1] /= treble; + f[2] /= treble; + f[3] /= treble; + f[4] /= treble; + f[5] /= treble; + f[6] /= treble; + f[7] /= treble; + f[8] /= treble; + f[9] /= treble; + //and now it's neatly scaled, too + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double inputSampleR = *in2; + + if (input != 1.0) {inputSampleL *= input; inputSampleR *= input;} + + static int noisesourceL = 0; + static int noisesourceR = 850010; + int residue; + double applyresidue; + + noisesourceL = noisesourceL % 1700021; noisesourceL++; + residue = noisesourceL * noisesourceL; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL += applyresidue; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + inputSampleL -= applyresidue; + } + + noisesourceR = noisesourceR % 1700021; noisesourceR++; + residue = noisesourceR * noisesourceR; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR += applyresidue; + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + inputSampleR -= applyresidue; + } + //for live air, we always apply the dither noise. Then, if our result is + //effectively digital black, we'll subtract it aUnBox. We want a 'air' hiss + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + aL[4] = aL[3]; aL[3] = aL[2]; aL[2] = aL[1]; + aL[1] = aL[0]; aL[0] = inputSampleL; + inputSampleL *= e[0]; + inputSampleL += (aL[1] * e[1]); + inputSampleL += (aL[2] * e[2]); + inputSampleL += (aL[3] * e[3]); + inputSampleL += (aL[4] * e[4]); + //this is now an average of inputSampleL + + aR[4] = aR[3]; aR[3] = aR[2]; aR[2] = aR[1]; + aR[1] = aR[0]; aR[0] = inputSampleR; + inputSampleR *= e[0]; + inputSampleR += (aR[1] * e[1]); + inputSampleR += (aR[2] * e[2]); + inputSampleR += (aR[3] * e[3]); + inputSampleR += (aR[4] * e[4]); + //this is now an average of inputSampleR + + bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1]; + bL[1] = bL[0]; bL[0] = inputSampleL; + inputSampleL *= e[0]; + inputSampleL += (bL[1] * e[1]); + inputSampleL += (bL[2] * e[2]); + inputSampleL += (bL[3] * e[3]); + inputSampleL += (bL[4] * e[4]); + //this is now an average of an average of inputSampleL. Two poles + + bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1]; + bR[1] = bR[0]; bR[0] = inputSampleR; + inputSampleR *= e[0]; + inputSampleR += (bR[1] * e[1]); + inputSampleR += (bR[2] * e[2]); + inputSampleR += (bR[3] * e[3]); + inputSampleR += (bR[4] * e[4]); + //this is now an average of an average of inputSampleR. Two poles + + inputSampleL *= unbox; + inputSampleR *= unbox; + //clip to 1.2533141373155 to reach maximum output + if (inputSampleL > 1.2533141373155) inputSampleL = 1.2533141373155; + if (inputSampleL < -1.2533141373155) inputSampleL = -1.2533141373155; + inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL)); + + if (inputSampleR > 1.2533141373155) inputSampleR = 1.2533141373155; + if (inputSampleR < -1.2533141373155) inputSampleR = -1.2533141373155; + inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR)); + + inputSampleL /= unbox; + inputSampleR /= unbox; + //now we have a distorted inputSample at the correct volume relative to drySample + + long double accumulatorSampleL = (drySampleL - inputSampleL); + cL[9] = cL[8]; cL[8] = cL[7]; cL[7] = cL[6]; cL[6] = cL[5]; + cL[5] = cL[4]; cL[4] = cL[3]; cL[3] = cL[2]; cL[2] = cL[1]; + cL[1] = cL[0]; cL[0] = accumulatorSampleL; + accumulatorSampleL *= f[0]; + accumulatorSampleL += (cL[1] * f[1]); + accumulatorSampleL += (cL[2] * f[2]); + accumulatorSampleL += (cL[3] * f[3]); + accumulatorSampleL += (cL[4] * f[4]); + accumulatorSampleL += (cL[5] * f[5]); + accumulatorSampleL += (cL[6] * f[6]); + accumulatorSampleL += (cL[7] * f[7]); + accumulatorSampleL += (cL[8] * f[8]); + accumulatorSampleL += (cL[9] * f[9]); + //this is now an average of all the recent variances from dry + + long double accumulatorSampleR = (drySampleR - inputSampleR); + cR[9] = cR[8]; cR[8] = cR[7]; cR[7] = cR[6]; cR[6] = cR[5]; + cR[5] = cR[4]; cR[4] = cR[3]; cR[3] = cR[2]; cR[2] = cR[1]; + cR[1] = cR[0]; cR[0] = accumulatorSampleR; + accumulatorSampleR *= f[0]; + accumulatorSampleR += (cR[1] * f[1]); + accumulatorSampleR += (cR[2] * f[2]); + accumulatorSampleR += (cR[3] * f[3]); + accumulatorSampleR += (cR[4] * f[4]); + accumulatorSampleR += (cR[5] * f[5]); + accumulatorSampleR += (cR[6] * f[6]); + accumulatorSampleR += (cR[7] * f[7]); + accumulatorSampleR += (cR[8] * f[8]); + accumulatorSampleR += (cR[9] * f[9]); + //this is now an average of all the recent variances from dry + + iirSampleAL = (iirSampleAL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount); + accumulatorSampleL -= iirSampleAL; + //two poles of IIR + + iirSampleAR = (iirSampleAR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount); + accumulatorSampleR -= iirSampleAR; + //two poles of IIR + + iirSampleBL = (iirSampleBL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount); + accumulatorSampleL -= iirSampleBL; + //highpass section + + iirSampleBR = (iirSampleBR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount); + accumulatorSampleR -= iirSampleBR; + //highpass section + //this is now a highpassed average of all the recent variances from dry + + inputSampleL = drySampleL - accumulatorSampleL; + inputSampleR = drySampleR - accumulatorSampleR; + //we apply it as one operation, to get the result. + + if (output != 1.0) {inputSampleL *= output; inputSampleR *= output;} + + //noise shaping to 64-bit floating point + double fpTemp = inputSampleL; + fpNShapeL += (inputSampleL-fpTemp); + inputSampleL += fpNShapeL; + //if this confuses you look at the wordlength for fpTemp :) + fpTemp = inputSampleR; + fpNShapeR += (inputSampleR-fpTemp); + inputSampleR += fpNShapeR; + //for deeper space and warmth, we try a non-oscillating noise shaping + //that is kind of ruthless: it will forever retain the rounding errors + //except we'll dial it back a hair at the end of every buffer processed + //end noise shaping on 64 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } + fpNShapeL *= 0.999999; + fpNShapeR *= 0.999999; + //we will just delicately dial back the FP noise shaping, not even every sample + //this is a good place to put subtle 'no runaway' calculations, though bear in mind + //that it will be called more often when you use shorter sample buffers in the DAW. + //So, very low latency operation will call these calculations more often. +} |