mirror of
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518 lines
16 KiB
C++
Executable file
518 lines
16 KiB
C++
Executable file
/* ========================================
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* VariMu - VariMu.h
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* Copyright (c) 2016 airwindows, All rights reserved
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* ======================================== */
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#ifndef __VariMu_H
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#include "VariMu.h"
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#endif
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void VariMu::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 = 2.0;
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overallscale /= 44100.0;
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overallscale *= getSampleRate();
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double threshold = 1.001 - (1.0-pow(1.0-A,3));
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double muMakeupGain = sqrt(1.0 / threshold);
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muMakeupGain = (muMakeupGain + sqrt(muMakeupGain))/2.0;
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muMakeupGain = sqrt(muMakeupGain);
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double outGain = sqrt(muMakeupGain);
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//gain settings around threshold
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double release = pow((1.15-B),5)*32768.0;
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release /= overallscale;
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double fastest = sqrt(release);
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//speed settings around release
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double coefficient;
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double output = outGain * C;
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double wet = D;
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long double squaredSampleL;
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long double squaredSampleR;
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// µ µ µ µ µ µ µ µ µ µ µ µ is the kitten song o/~
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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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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 aVariMu. 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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if (fabs(inputSampleL) > fabs(previousL)) squaredSampleL = previousL * previousL;
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else squaredSampleL = inputSampleL * inputSampleL;
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previousL = inputSampleL;
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inputSampleL *= muMakeupGain;
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if (fabs(inputSampleR) > fabs(previousR)) squaredSampleR = previousR * previousR;
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else squaredSampleR = inputSampleR * inputSampleR;
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previousR = inputSampleR;
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inputSampleR *= muMakeupGain;
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//adjust coefficients for L
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if (flip)
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{
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if (fabs(squaredSampleL) > threshold)
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{
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muVaryL = threshold / fabs(squaredSampleL);
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muAttackL = sqrt(fabs(muSpeedAL));
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muCoefficientAL = muCoefficientAL * (muAttackL-1.0);
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if (muVaryL < threshold)
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{
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muCoefficientAL = muCoefficientAL + threshold;
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}
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else
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{
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muCoefficientAL = muCoefficientAL + muVaryL;
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}
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muCoefficientAL = muCoefficientAL / muAttackL;
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}
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else
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{
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muCoefficientAL = muCoefficientAL * ((muSpeedAL * muSpeedAL)-1.0);
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muCoefficientAL = muCoefficientAL + 1.0;
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muCoefficientAL = muCoefficientAL / (muSpeedAL * muSpeedAL);
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}
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muNewSpeedL = muSpeedAL * (muSpeedAL-1);
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muNewSpeedL = muNewSpeedL + fabs(squaredSampleL*release)+fastest;
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muSpeedAL = muNewSpeedL / muSpeedAL;
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}
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else
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{
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if (fabs(squaredSampleL) > threshold)
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{
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muVaryL = threshold / fabs(squaredSampleL);
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muAttackL = sqrt(fabs(muSpeedBL));
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muCoefficientBL = muCoefficientBL * (muAttackL-1);
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if (muVaryL < threshold)
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{
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muCoefficientBL = muCoefficientBL + threshold;
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}
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else
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{
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muCoefficientBL = muCoefficientBL + muVaryL;
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}
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muCoefficientBL = muCoefficientBL / muAttackL;
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}
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else
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{
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muCoefficientBL = muCoefficientBL * ((muSpeedBL * muSpeedBL)-1.0);
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muCoefficientBL = muCoefficientBL + 1.0;
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muCoefficientBL = muCoefficientBL / (muSpeedBL * muSpeedBL);
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}
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muNewSpeedL = muSpeedBL * (muSpeedBL-1);
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muNewSpeedL = muNewSpeedL + fabs(squaredSampleL*release)+fastest;
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muSpeedBL = muNewSpeedL / muSpeedBL;
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}
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//got coefficients, adjusted speeds for L
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//adjust coefficients for R
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if (flip)
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{
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if (fabs(squaredSampleR) > threshold)
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{
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muVaryR = threshold / fabs(squaredSampleR);
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muAttackR = sqrt(fabs(muSpeedAR));
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muCoefficientAR = muCoefficientAR * (muAttackR-1.0);
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if (muVaryR < threshold)
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{
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muCoefficientAR = muCoefficientAR + threshold;
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}
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else
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{
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muCoefficientAR = muCoefficientAR + muVaryR;
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}
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muCoefficientAR = muCoefficientAR / muAttackR;
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}
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else
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{
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muCoefficientAR = muCoefficientAR * ((muSpeedAR * muSpeedAR)-1.0);
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muCoefficientAR = muCoefficientAR + 1.0;
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muCoefficientAR = muCoefficientAR / (muSpeedAR * muSpeedAR);
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}
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muNewSpeedR = muSpeedAR * (muSpeedAR-1);
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muNewSpeedR = muNewSpeedR + fabs(squaredSampleR*release)+fastest;
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muSpeedAR = muNewSpeedR / muSpeedAR;
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}
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else
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{
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if (fabs(squaredSampleR) > threshold)
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{
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muVaryR = threshold / fabs(squaredSampleR);
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muAttackR = sqrt(fabs(muSpeedBR));
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muCoefficientBR = muCoefficientBR * (muAttackR-1);
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if (muVaryR < threshold)
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{
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muCoefficientBR = muCoefficientBR + threshold;
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}
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else
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{
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muCoefficientBR = muCoefficientBR + muVaryR;
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}
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muCoefficientBR = muCoefficientBR / muAttackR;
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}
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else
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{
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muCoefficientBR = muCoefficientBR * ((muSpeedBR * muSpeedBR)-1.0);
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muCoefficientBR = muCoefficientBR + 1.0;
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muCoefficientBR = muCoefficientBR / (muSpeedBR * muSpeedBR);
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}
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muNewSpeedR = muSpeedBR * (muSpeedBR-1);
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muNewSpeedR = muNewSpeedR + fabs(squaredSampleR*release)+fastest;
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muSpeedBR = muNewSpeedR / muSpeedBR;
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}
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//got coefficients, adjusted speeds for R
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if (flip)
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{
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coefficient = (muCoefficientAL + pow(muCoefficientAL,2))/2.0;
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inputSampleL *= coefficient;
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coefficient = (muCoefficientAR + pow(muCoefficientAR,2))/2.0;
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inputSampleR *= coefficient;
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}
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else
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{
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coefficient = (muCoefficientBL + pow(muCoefficientBL,2))/2.0;
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inputSampleL *= coefficient;
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coefficient = (muCoefficientBR + pow(muCoefficientBR,2))/2.0;
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inputSampleR *= coefficient;
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}
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//applied compression with vari-vari-µ-µ-µ-µ-µ-µ-is-the-kitten-song o/~
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//applied gain correction to control output level- tends to constrain sound rather than inflate it
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flip = !flip;
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if (output < 1.0) {
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inputSampleL *= output;
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inputSampleR *= output;
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}
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if (wet < 1.0) {
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inputSampleL = (drySampleL * (1.0-wet)) + (inputSampleL * wet);
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inputSampleR = (drySampleR * (1.0-wet)) + (inputSampleR * wet);
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}
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//nice little output stage template: if we have another scale of floating point
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//number, we really don't want to meaninglessly multiply that by 1.0.
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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 VariMu::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 = 2.0;
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overallscale /= 44100.0;
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overallscale *= getSampleRate();
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double threshold = 1.001 - (1.0-pow(1.0-A,3));
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double muMakeupGain = sqrt(1.0 / threshold);
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muMakeupGain = (muMakeupGain + sqrt(muMakeupGain))/2.0;
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muMakeupGain = sqrt(muMakeupGain);
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double outGain = sqrt(muMakeupGain);
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//gain settings around threshold
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double release = pow((1.15-B),5)*32768.0;
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release /= overallscale;
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double fastest = sqrt(release);
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//speed settings around release
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double coefficient;
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double output = outGain * C;
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double wet = D;
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long double squaredSampleL;
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long double squaredSampleR;
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// µ µ µ µ µ µ µ µ µ µ µ µ is the kitten song o/~
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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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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 aVariMu. 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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if (fabs(inputSampleL) > fabs(previousL)) squaredSampleL = previousL * previousL;
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else squaredSampleL = inputSampleL * inputSampleL;
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previousL = inputSampleL;
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inputSampleL *= muMakeupGain;
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if (fabs(inputSampleR) > fabs(previousR)) squaredSampleR = previousR * previousR;
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else squaredSampleR = inputSampleR * inputSampleR;
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previousR = inputSampleR;
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inputSampleR *= muMakeupGain;
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//adjust coefficients for L
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if (flip)
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{
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if (fabs(squaredSampleL) > threshold)
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{
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muVaryL = threshold / fabs(squaredSampleL);
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muAttackL = sqrt(fabs(muSpeedAL));
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muCoefficientAL = muCoefficientAL * (muAttackL-1.0);
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if (muVaryL < threshold)
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{
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muCoefficientAL = muCoefficientAL + threshold;
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}
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else
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{
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muCoefficientAL = muCoefficientAL + muVaryL;
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}
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muCoefficientAL = muCoefficientAL / muAttackL;
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}
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else
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{
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muCoefficientAL = muCoefficientAL * ((muSpeedAL * muSpeedAL)-1.0);
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muCoefficientAL = muCoefficientAL + 1.0;
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muCoefficientAL = muCoefficientAL / (muSpeedAL * muSpeedAL);
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}
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muNewSpeedL = muSpeedAL * (muSpeedAL-1);
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muNewSpeedL = muNewSpeedL + fabs(squaredSampleL*release)+fastest;
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muSpeedAL = muNewSpeedL / muSpeedAL;
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}
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else
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{
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if (fabs(squaredSampleL) > threshold)
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{
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muVaryL = threshold / fabs(squaredSampleL);
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muAttackL = sqrt(fabs(muSpeedBL));
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muCoefficientBL = muCoefficientBL * (muAttackL-1);
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if (muVaryL < threshold)
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{
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muCoefficientBL = muCoefficientBL + threshold;
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}
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else
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{
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muCoefficientBL = muCoefficientBL + muVaryL;
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}
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muCoefficientBL = muCoefficientBL / muAttackL;
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}
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else
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{
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muCoefficientBL = muCoefficientBL * ((muSpeedBL * muSpeedBL)-1.0);
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muCoefficientBL = muCoefficientBL + 1.0;
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muCoefficientBL = muCoefficientBL / (muSpeedBL * muSpeedBL);
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}
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muNewSpeedL = muSpeedBL * (muSpeedBL-1);
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muNewSpeedL = muNewSpeedL + fabs(squaredSampleL*release)+fastest;
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muSpeedBL = muNewSpeedL / muSpeedBL;
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}
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//got coefficients, adjusted speeds for L
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//adjust coefficients for R
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if (flip)
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{
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if (fabs(squaredSampleR) > threshold)
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{
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muVaryR = threshold / fabs(squaredSampleR);
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muAttackR = sqrt(fabs(muSpeedAR));
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muCoefficientAR = muCoefficientAR * (muAttackR-1.0);
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if (muVaryR < threshold)
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{
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muCoefficientAR = muCoefficientAR + threshold;
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}
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else
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{
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muCoefficientAR = muCoefficientAR + muVaryR;
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}
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muCoefficientAR = muCoefficientAR / muAttackR;
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}
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else
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{
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muCoefficientAR = muCoefficientAR * ((muSpeedAR * muSpeedAR)-1.0);
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muCoefficientAR = muCoefficientAR + 1.0;
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muCoefficientAR = muCoefficientAR / (muSpeedAR * muSpeedAR);
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}
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muNewSpeedR = muSpeedAR * (muSpeedAR-1);
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muNewSpeedR = muNewSpeedR + fabs(squaredSampleR*release)+fastest;
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muSpeedAR = muNewSpeedR / muSpeedAR;
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}
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else
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{
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if (fabs(squaredSampleR) > threshold)
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{
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muVaryR = threshold / fabs(squaredSampleR);
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muAttackR = sqrt(fabs(muSpeedBR));
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muCoefficientBR = muCoefficientBR * (muAttackR-1);
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if (muVaryR < threshold)
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{
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muCoefficientBR = muCoefficientBR + threshold;
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}
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else
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{
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muCoefficientBR = muCoefficientBR + muVaryR;
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}
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muCoefficientBR = muCoefficientBR / muAttackR;
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}
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else
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{
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muCoefficientBR = muCoefficientBR * ((muSpeedBR * muSpeedBR)-1.0);
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muCoefficientBR = muCoefficientBR + 1.0;
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muCoefficientBR = muCoefficientBR / (muSpeedBR * muSpeedBR);
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}
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muNewSpeedR = muSpeedBR * (muSpeedBR-1);
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muNewSpeedR = muNewSpeedR + fabs(squaredSampleR*release)+fastest;
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muSpeedBR = muNewSpeedR / muSpeedBR;
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}
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//got coefficients, adjusted speeds for R
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if (flip)
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{
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coefficient = (muCoefficientAL + pow(muCoefficientAL,2))/2.0;
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inputSampleL *= coefficient;
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coefficient = (muCoefficientAR + pow(muCoefficientAR,2))/2.0;
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inputSampleR *= coefficient;
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}
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else
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{
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coefficient = (muCoefficientBL + pow(muCoefficientBL,2))/2.0;
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inputSampleL *= coefficient;
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coefficient = (muCoefficientBR + pow(muCoefficientBR,2))/2.0;
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inputSampleR *= coefficient;
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}
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//applied compression with vari-vari-µ-µ-µ-µ-µ-µ-is-the-kitten-song o/~
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//applied gain correction to control output level- tends to constrain sound rather than inflate it
|
|
flip = !flip;
|
|
|
|
if (output < 1.0) {
|
|
inputSampleL *= output;
|
|
inputSampleR *= output;
|
|
}
|
|
if (wet < 1.0) {
|
|
inputSampleL = (drySampleL * (1.0-wet)) + (inputSampleL * wet);
|
|
inputSampleR = (drySampleR * (1.0-wet)) + (inputSampleR * wet);
|
|
}
|
|
//nice little output stage template: if we have another scale of floating point
|
|
//number, we really don't want to meaninglessly multiply that by 1.0.
|
|
|
|
//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.
|
|
}
|