mirror of
https://github.com/airwindows/airwindows.git
synced 2026-05-17 06:05:56 -06:00
9ecd9a8c01
Note that I am compelled to break VST builds on Windows because my build folders have had some files in 'em: you'll have to find those yourself. Here's hoping anyone trying to develop for Windows VST (or any VST) is able to find what they need, but I cannot help you set up a build environment, I can only give you my code for the audio part.
488 lines
18 KiB
C++
Executable file
488 lines
18 KiB
C++
Executable file
/* ========================================
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* UnBox - UnBox.h
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* Copyright (c) 2016 airwindows, All rights reserved
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* ======================================== */
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#ifndef __UnBox_H
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#include "UnBox.h"
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#endif
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void UnBox::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
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{
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float* in1 = inputs[0];
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float* in2 = inputs[1];
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float* out1 = outputs[0];
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float* out2 = outputs[1];
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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= getSampleRate();
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double input = A*2.0;
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double unbox = B+1.0;
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unbox *= unbox; //let's get some more gain into this
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double iirAmount = (unbox*0.00052)/overallscale;
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double output = C*2.0;
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double treble = unbox; //averaging taps 1-4
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double gain = treble;
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if (gain > 1.0) {e[0] = 1.0; gain -= 1.0;} else {e[0] = gain; gain = 0.0;}
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if (gain > 1.0) {e[1] = 1.0; gain -= 1.0;} else {e[1] = gain; gain = 0.0;}
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if (gain > 1.0) {e[2] = 1.0; gain -= 1.0;} else {e[2] = gain; gain = 0.0;}
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if (gain > 1.0) {e[3] = 1.0; gain -= 1.0;} else {e[3] = gain; gain = 0.0;}
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if (gain > 1.0) {e[4] = 1.0; gain -= 1.0;} else {e[4] = gain; gain = 0.0;}
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//there, now we have a neat little moving average with remainders
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if (treble < 1.0) treble = 1.0;
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e[0] /= treble;
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e[1] /= treble;
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e[2] /= treble;
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e[3] /= treble;
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e[4] /= treble;
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//and now it's neatly scaled, too
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treble = unbox*2.0; //averaging taps 1-8
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gain = treble;
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if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;}
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if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;}
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if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;}
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if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;}
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if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;}
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if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;}
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if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;}
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if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;}
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if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;}
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if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;}
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//there, now we have a neat little moving average with remainders
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if (treble < 1.0) treble = 1.0;
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f[0] /= treble;
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f[1] /= treble;
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f[2] /= treble;
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f[3] /= treble;
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f[4] /= treble;
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f[5] /= treble;
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f[6] /= treble;
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f[7] /= treble;
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f[8] /= treble;
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f[9] /= treble;
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//and now it's neatly scaled, too
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while (--sampleFrames >= 0)
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{
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long double inputSampleL = *in1;
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long double inputSampleR = *in2;
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if (input != 1.0) {inputSampleL *= input; inputSampleR *= input;}
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static int noisesourceL = 0;
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static int noisesourceR = 850010;
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int residue;
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double applyresidue;
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noisesourceL = noisesourceL % 1700021; noisesourceL++;
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residue = noisesourceL * noisesourceL;
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residue = residue % 170003; residue *= residue;
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residue = residue % 17011; residue *= residue;
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residue = residue % 1709; residue *= residue;
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residue = residue % 173; residue *= residue;
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residue = residue % 17;
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applyresidue = residue;
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applyresidue *= 0.00000001;
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applyresidue *= 0.00000001;
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inputSampleL += applyresidue;
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if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
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inputSampleL -= applyresidue;
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}
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noisesourceR = noisesourceR % 1700021; noisesourceR++;
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residue = noisesourceR * noisesourceR;
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residue = residue % 170003; residue *= residue;
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residue = residue % 17011; residue *= residue;
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residue = residue % 1709; residue *= residue;
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residue = residue % 173; residue *= residue;
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residue = residue % 17;
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applyresidue = residue;
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applyresidue *= 0.00000001;
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applyresidue *= 0.00000001;
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inputSampleR += applyresidue;
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if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
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inputSampleR -= applyresidue;
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}
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//for live air, we always apply the dither noise. Then, if our result is
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//effectively digital black, we'll subtract it aUnBox. We want a 'air' hiss
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long double drySampleL = inputSampleL;
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long double drySampleR = inputSampleR;
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aL[4] = aL[3]; aL[3] = aL[2]; aL[2] = aL[1];
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aL[1] = aL[0]; aL[0] = inputSampleL;
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inputSampleL *= e[0];
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inputSampleL += (aL[1] * e[1]);
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inputSampleL += (aL[2] * e[2]);
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inputSampleL += (aL[3] * e[3]);
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inputSampleL += (aL[4] * e[4]);
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//this is now an average of inputSampleL
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aR[4] = aR[3]; aR[3] = aR[2]; aR[2] = aR[1];
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aR[1] = aR[0]; aR[0] = inputSampleR;
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inputSampleR *= e[0];
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inputSampleR += (aR[1] * e[1]);
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inputSampleR += (aR[2] * e[2]);
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inputSampleR += (aR[3] * e[3]);
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inputSampleR += (aR[4] * e[4]);
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//this is now an average of inputSampleR
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bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1];
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bL[1] = bL[0]; bL[0] = inputSampleL;
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inputSampleL *= e[0];
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inputSampleL += (bL[1] * e[1]);
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inputSampleL += (bL[2] * e[2]);
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inputSampleL += (bL[3] * e[3]);
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inputSampleL += (bL[4] * e[4]);
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//this is now an average of an average of inputSampleL. Two poles
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bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1];
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bR[1] = bR[0]; bR[0] = inputSampleR;
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inputSampleR *= e[0];
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inputSampleR += (bR[1] * e[1]);
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inputSampleR += (bR[2] * e[2]);
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inputSampleR += (bR[3] * e[3]);
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inputSampleR += (bR[4] * e[4]);
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//this is now an average of an average of inputSampleR. Two poles
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inputSampleL *= unbox;
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inputSampleR *= unbox;
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//clip to 1.2533141373155 to reach maximum output
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if (inputSampleL > 1.2533141373155) inputSampleL = 1.2533141373155;
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if (inputSampleL < -1.2533141373155) inputSampleL = -1.2533141373155;
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inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL));
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if (inputSampleR > 1.2533141373155) inputSampleR = 1.2533141373155;
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if (inputSampleR < -1.2533141373155) inputSampleR = -1.2533141373155;
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inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR));
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inputSampleL /= unbox;
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inputSampleR /= unbox;
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//now we have a distorted inputSample at the correct volume relative to drySample
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long double accumulatorSampleL = (drySampleL - inputSampleL);
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cL[9] = cL[8]; cL[8] = cL[7]; cL[7] = cL[6]; cL[6] = cL[5];
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cL[5] = cL[4]; cL[4] = cL[3]; cL[3] = cL[2]; cL[2] = cL[1];
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cL[1] = cL[0]; cL[0] = accumulatorSampleL;
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accumulatorSampleL *= f[0];
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accumulatorSampleL += (cL[1] * f[1]);
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accumulatorSampleL += (cL[2] * f[2]);
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accumulatorSampleL += (cL[3] * f[3]);
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accumulatorSampleL += (cL[4] * f[4]);
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accumulatorSampleL += (cL[5] * f[5]);
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accumulatorSampleL += (cL[6] * f[6]);
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accumulatorSampleL += (cL[7] * f[7]);
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accumulatorSampleL += (cL[8] * f[8]);
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accumulatorSampleL += (cL[9] * f[9]);
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//this is now an average of all the recent variances from dry
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long double accumulatorSampleR = (drySampleR - inputSampleR);
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cR[9] = cR[8]; cR[8] = cR[7]; cR[7] = cR[6]; cR[6] = cR[5];
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cR[5] = cR[4]; cR[4] = cR[3]; cR[3] = cR[2]; cR[2] = cR[1];
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cR[1] = cR[0]; cR[0] = accumulatorSampleR;
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accumulatorSampleR *= f[0];
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accumulatorSampleR += (cR[1] * f[1]);
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accumulatorSampleR += (cR[2] * f[2]);
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accumulatorSampleR += (cR[3] * f[3]);
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accumulatorSampleR += (cR[4] * f[4]);
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accumulatorSampleR += (cR[5] * f[5]);
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accumulatorSampleR += (cR[6] * f[6]);
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accumulatorSampleR += (cR[7] * f[7]);
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accumulatorSampleR += (cR[8] * f[8]);
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accumulatorSampleR += (cR[9] * f[9]);
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//this is now an average of all the recent variances from dry
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iirSampleAL = (iirSampleAL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount);
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accumulatorSampleL -= iirSampleAL;
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//two poles of IIR
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iirSampleAR = (iirSampleAR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount);
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accumulatorSampleR -= iirSampleAR;
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//two poles of IIR
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iirSampleBL = (iirSampleBL * (1 - iirAmount)) + (accumulatorSampleL * iirAmount);
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accumulatorSampleL -= iirSampleBL;
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//highpass section
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iirSampleBR = (iirSampleBR * (1 - iirAmount)) + (accumulatorSampleR * iirAmount);
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accumulatorSampleR -= iirSampleBR;
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//highpass section
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//this is now a highpassed average of all the recent variances from dry
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inputSampleL = drySampleL - accumulatorSampleL;
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inputSampleR = drySampleR - accumulatorSampleR;
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//we apply it as one operation, to get the result.
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if (output != 1.0) {inputSampleL *= output; inputSampleR *= output;}
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//noise shaping to 32-bit floating point
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float fpTemp = inputSampleL;
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fpNShapeL += (inputSampleL-fpTemp);
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inputSampleL += fpNShapeL;
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//if this confuses you look at the wordlength for fpTemp :)
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fpTemp = inputSampleR;
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fpNShapeR += (inputSampleR-fpTemp);
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inputSampleR += fpNShapeR;
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//for deeper space and warmth, we try a non-oscillating noise shaping
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//that is kind of ruthless: it will forever retain the rounding errors
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//except we'll dial it back a hair at the end of every buffer processed
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//end noise shaping on 32 bit output
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*out1 = inputSampleL;
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*out2 = inputSampleR;
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*in1++;
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*in2++;
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*out1++;
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*out2++;
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}
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fpNShapeL *= 0.999999;
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fpNShapeR *= 0.999999;
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//we will just delicately dial back the FP noise shaping, not even every sample
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//this is a good place to put subtle 'no runaway' calculations, though bear in mind
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//that it will be called more often when you use shorter sample buffers in the DAW.
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//So, very low latency operation will call these calculations more often.
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}
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void UnBox::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
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{
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double* in1 = inputs[0];
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double* in2 = inputs[1];
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double* out1 = outputs[0];
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double* out2 = outputs[1];
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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= getSampleRate();
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double input = A*2.0;
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double unbox = B+1.0;
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unbox *= unbox; //let's get some more gain into this
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double iirAmount = (unbox*0.00052)/overallscale;
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double output = C*2.0;
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double treble = unbox; //averaging taps 1-4
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double gain = treble;
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if (gain > 1.0) {e[0] = 1.0; gain -= 1.0;} else {e[0] = gain; gain = 0.0;}
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if (gain > 1.0) {e[1] = 1.0; gain -= 1.0;} else {e[1] = gain; gain = 0.0;}
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if (gain > 1.0) {e[2] = 1.0; gain -= 1.0;} else {e[2] = gain; gain = 0.0;}
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if (gain > 1.0) {e[3] = 1.0; gain -= 1.0;} else {e[3] = gain; gain = 0.0;}
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if (gain > 1.0) {e[4] = 1.0; gain -= 1.0;} else {e[4] = gain; gain = 0.0;}
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//there, now we have a neat little moving average with remainders
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if (treble < 1.0) treble = 1.0;
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e[0] /= treble;
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e[1] /= treble;
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e[2] /= treble;
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e[3] /= treble;
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e[4] /= treble;
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//and now it's neatly scaled, too
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treble = unbox*2.0; //averaging taps 1-8
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gain = treble;
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if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;}
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if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;}
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if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;}
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if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;}
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if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;}
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if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;}
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if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;}
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if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;}
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if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;}
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if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;}
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//there, now we have a neat little moving average with remainders
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if (treble < 1.0) treble = 1.0;
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f[0] /= treble;
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f[1] /= treble;
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f[2] /= treble;
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f[3] /= treble;
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f[4] /= treble;
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f[5] /= treble;
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f[6] /= treble;
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f[7] /= treble;
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f[8] /= treble;
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f[9] /= treble;
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//and now it's neatly scaled, too
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while (--sampleFrames >= 0)
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{
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long double inputSampleL = *in1;
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long double inputSampleR = *in2;
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if (input != 1.0) {inputSampleL *= input; inputSampleR *= input;}
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static int noisesourceL = 0;
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static int noisesourceR = 850010;
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int residue;
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double applyresidue;
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noisesourceL = noisesourceL % 1700021; noisesourceL++;
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residue = noisesourceL * noisesourceL;
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residue = residue % 170003; residue *= residue;
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residue = residue % 17011; residue *= residue;
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residue = residue % 1709; residue *= residue;
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residue = residue % 173; residue *= residue;
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residue = residue % 17;
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applyresidue = residue;
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applyresidue *= 0.00000001;
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applyresidue *= 0.00000001;
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inputSampleL += applyresidue;
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if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
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inputSampleL -= applyresidue;
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}
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noisesourceR = noisesourceR % 1700021; noisesourceR++;
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residue = noisesourceR * noisesourceR;
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residue = residue % 170003; residue *= residue;
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residue = residue % 17011; residue *= residue;
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residue = residue % 1709; residue *= residue;
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residue = residue % 173; residue *= residue;
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residue = residue % 17;
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applyresidue = residue;
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applyresidue *= 0.00000001;
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applyresidue *= 0.00000001;
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inputSampleR += applyresidue;
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if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
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inputSampleR -= applyresidue;
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}
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//for live air, we always apply the dither noise. Then, if our result is
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//effectively digital black, we'll subtract it aUnBox. We want a 'air' hiss
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long double drySampleL = inputSampleL;
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long double drySampleR = inputSampleR;
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aL[4] = aL[3]; aL[3] = aL[2]; aL[2] = aL[1];
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aL[1] = aL[0]; aL[0] = inputSampleL;
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inputSampleL *= e[0];
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inputSampleL += (aL[1] * e[1]);
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inputSampleL += (aL[2] * e[2]);
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inputSampleL += (aL[3] * e[3]);
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inputSampleL += (aL[4] * e[4]);
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//this is now an average of inputSampleL
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aR[4] = aR[3]; aR[3] = aR[2]; aR[2] = aR[1];
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aR[1] = aR[0]; aR[0] = inputSampleR;
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inputSampleR *= e[0];
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inputSampleR += (aR[1] * e[1]);
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inputSampleR += (aR[2] * e[2]);
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inputSampleR += (aR[3] * e[3]);
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inputSampleR += (aR[4] * e[4]);
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//this is now an average of inputSampleR
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bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1];
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bL[1] = bL[0]; bL[0] = inputSampleL;
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inputSampleL *= e[0];
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inputSampleL += (bL[1] * e[1]);
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inputSampleL += (bL[2] * e[2]);
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inputSampleL += (bL[3] * e[3]);
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inputSampleL += (bL[4] * e[4]);
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//this is now an average of an average of inputSampleL. Two poles
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bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1];
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bR[1] = bR[0]; bR[0] = inputSampleR;
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inputSampleR *= e[0];
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inputSampleR += (bR[1] * e[1]);
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inputSampleR += (bR[2] * e[2]);
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inputSampleR += (bR[3] * e[3]);
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inputSampleR += (bR[4] * e[4]);
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//this is now an average of an average of inputSampleR. Two poles
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inputSampleL *= unbox;
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inputSampleR *= unbox;
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//clip to 1.2533141373155 to reach maximum output
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if (inputSampleL > 1.2533141373155) inputSampleL = 1.2533141373155;
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if (inputSampleL < -1.2533141373155) inputSampleL = -1.2533141373155;
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inputSampleL = sin(inputSampleL * fabs(inputSampleL)) / ((inputSampleL == 0.0) ?1:fabs(inputSampleL));
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if (inputSampleR > 1.2533141373155) inputSampleR = 1.2533141373155;
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if (inputSampleR < -1.2533141373155) inputSampleR = -1.2533141373155;
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inputSampleR = sin(inputSampleR * fabs(inputSampleR)) / ((inputSampleR == 0.0) ?1:fabs(inputSampleR));
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inputSampleL /= unbox;
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inputSampleR /= unbox;
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//now we have a distorted inputSample at the correct volume relative to drySample
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|
|
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long double accumulatorSampleL = (drySampleL - inputSampleL);
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cL[9] = cL[8]; cL[8] = cL[7]; cL[7] = cL[6]; cL[6] = cL[5];
|
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cL[5] = cL[4]; cL[4] = cL[3]; cL[3] = cL[2]; cL[2] = cL[1];
|
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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.
|
|
}
|